#include <string.h>
#include <sys/time.h>
#include <stdlib.h>
+#include <utilX.h>
#include <mm_error.h>
#include <mm_attrs.h>
#include "mm_player_ini.h"
#include "mm_player_attrs.h"
#include "mm_player_capture.h"
+#include "mm_player_utils.h"
+#include <sched.h>
+
+#include <mm_sound.h>
+
+#define MM_SMOOTH_STREAMING
/*===========================================================================================
| |
#define TRICK_PLAY_MUTE_THRESHOLD_MIN 0.0
#define MM_VOLUME_FACTOR_DEFAULT 1.0
-#define MM_VOLUME_FACTOR_MIN 0
-#define MM_VOLUME_FACTOR_MAX 1.0
+#define MM_VOLUME_FACTOR_MIN 0
+#define MM_VOLUME_FACTOR_MAX 1.0
#define MM_PLAYER_FADEOUT_TIME_DEFAULT 700000 // 700 msec
#define MM_PLAYER_WMA_VNAME "wmaversion"
#define DEFAULT_PLAYBACK_RATE 1.0
+#define PLAYBACK_RATE_EX_AUDIO_MIN 0.5
+#define PLAYBACK_RATE_EX_AUDIO_MAX 2.0
+#define PLAYBACK_RATE_EX_VIDEO_MIN 0.5
+#define PLAYBACK_RATE_EX_VIDEO_MAX 1.5
-#define GST_QUEUE_DEFAULT_TIME 2
+#define GST_QUEUE_DEFAULT_TIME 4
#define GST_QUEUE_HLS_TIME 8
+#define DEFAULT_AUDIO_CH 0
+
/* video capture callback*/
gulong ahs_appsrc_cb_probe_id = 0;
-#define MMPLAYER_USE_FILE_FOR_BUFFERING(player) (((player)->profile.uri_type != MM_PLAYER_URI_TYPE_HLS) && (PLAYER_INI()->http_file_buffer_path) && (strlen(PLAYER_INI()->http_file_buffer_path) > 0) )
-#define MMPLAYER_PLAY_SUBTITLE(player) ((player)->play_subtitle)
+#define MMPLAYER_USE_FILE_FOR_BUFFERING(player) (((player)->profile.uri_type != MM_PLAYER_URI_TYPE_HLS) && (player->ini.http_file_buffer_path) && (strlen(player->ini.http_file_buffer_path) > 0) )
+
+#define LAZY_PAUSE_TIMEOUT_MSEC 700
+#define MM_PLAYER_NAME "mmplayer"
-#define LAZY_PAUSE_TIMEOUT_MSEC 700
+//#define ENABLE_DRMSRC
/*---------------------------------------------------------------------------
| LOCAL CONSTANT DEFINITIONS: |
/*---------------------------------------------------------------------------
| LOCAL FUNCTION PROTOTYPES: |
---------------------------------------------------------------------------*/
-static gboolean __mmplayer_set_state(mm_player_t* player, int state);
-static int __mmplayer_get_state(mm_player_t* player);
-static int __mmplayer_gst_create_video_pipeline(mm_player_t* player, GstCaps *caps);
+static int __mmplayer_set_state(mm_player_t* player, int state);
+static int __mmplayer_gst_create_video_pipeline(mm_player_t* player, GstCaps *caps, MMDisplaySurfaceType surface_type);
static int __mmplayer_gst_create_audio_pipeline(mm_player_t* player);
static int __mmplayer_gst_create_text_pipeline(mm_player_t* player);
-static int __mmplayer_gst_create_subtitle_pipeline(mm_player_t* player);
+static int __mmplayer_gst_create_subtitle_src(mm_player_t* player);
static int __mmplayer_gst_create_pipeline(mm_player_t* player);
static int __mmplayer_gst_destroy_pipeline(mm_player_t* player);
static int __mmplayer_gst_element_link_bucket(GList* element_bucket);
static gboolean __mmplayer_gst_callback(GstBus *bus, GstMessage *msg, gpointer data);
-static void __mmplayer_gst_decode_callback(GstElement *decodebin, GstPad *pad, gboolean last, gpointer data);
+static void __mmplayer_gst_selector_blocked(GstPad* pad, gboolean blocked, gpointer data);
+static void __mmplayer_gst_decode_pad_added(GstElement* elem, GstPad* pad, gpointer data);
+static void __mmplayer_gst_decode_no_more_pads(GstElement* elem, gpointer data);
+static void __mmplayer_gst_decode_callback(GstElement *decodebin, GstPad *pad, gpointer data);
+static void __mmplayer_gst_decode_unknown_type(GstElement *elem, GstPad* pad, GstCaps *caps, gpointer data);
+static gboolean __mmplayer_gst_decode_autoplug_continue(GstElement *bin, GstPad* pad, GstCaps * caps, gpointer data);
+static gint __mmplayer_gst_decode_autoplug_select(GstElement *bin, GstPad* pad, GstCaps * caps, GstElementFactory* factory, gpointer data);
+//static GValueArray* __mmplayer_gst_decode_autoplug_factories(GstElement *bin, GstPad* pad, GstCaps * caps, gpointer data);
+static GValueArray* __mmplayer_gst_decode_autoplug_sort(GstElement *bin, GstPad* pad, GstCaps * caps, GValueArray *factories, gpointer data);
+static void __mmplayer_gst_decode_pad_removed(GstElement *elem, GstPad* new_pad, gpointer data);
+static void __mmplayer_gst_decode_drained(GstElement *bin, gpointer data);
+static void __mmplayer_gst_element_added(GstElement* bin, GstElement* element, gpointer data);
+static GstElement * __mmplayer_create_decodebin(mm_player_t* player);
+static gboolean __mmplayer_try_to_plug_decodebin(mm_player_t* player, GstPad *srcpad, const GstCaps *caps);
static void __mmplayer_typefind_have_type( GstElement *tf, guint probability, GstCaps *caps, gpointer data);
static gboolean __mmplayer_try_to_plug(mm_player_t* player, GstPad *pad, const GstCaps *caps);
static void __mmplayer_pipeline_complete(GstElement *decodebin, gpointer data);
static gboolean __mmplayer_is_midi_type(gchar* str_caps);
-static gboolean __mmplayer_is_amr_type (gchar *str_caps);
static gboolean __mmplayer_is_only_mp3_type (gchar *str_caps);
+static gboolean __mmplayer_is_omx_decoder_type(mm_player_t* player); // mp3
+static void __mmplayer_set_audio_attrs(mm_player_t* player, GstCaps* caps);
+//static void __mmplayer_check_video_zero_cpoy(mm_player_t* player, GstElementFactory* factory);
static gboolean __mmplayer_close_link(mm_player_t* player, GstPad *srcpad, GstElement *sinkelement, const char *padname, const GList *templlist);
static gboolean __mmplayer_feature_filter(GstPluginFeature *feature, gpointer data);
static void __mmplayer_add_new_pad(GstElement *element, GstPad *pad, gpointer data);
static void __mmplayer_gst_rtp_no_more_pads (GstElement *element, gpointer data);
+static void __mmplayer_gst_wfd_dynamic_pad (GstElement *element, GstPad *pad, gpointer data);
static void __mmplayer_gst_rtp_dynamic_pad (GstElement *element, GstPad *pad, gpointer data);
-static gboolean __mmplayer_update_stream_service_type( mm_player_t* player );
+static gboolean __mmplayer_get_stream_service_type( mm_player_t* player );
static gboolean __mmplayer_update_subtitle( GstElement* object, GstBuffer *buffer, GstPad *pad, gpointer data);
static void __mmplayer_init_factories(mm_player_t* player);
static void __mmplayer_release_factories(mm_player_t* player);
-static void __mmplayer_release_misc(mm_player_t* player);
-static gboolean __mmplayer_gstreamer_init(void);
+static void __mmplayer_release_misc(mm_player_t* player);
+static void __mmplayer_release_misc_post(mm_player_t* player);
+static gboolean __mmplayer_init_gstreamer(mm_player_t* player);
static int __mmplayer_gst_set_state (mm_player_t* player, GstElement * pipeline, GstState state, gboolean async, gint timeout );
-gboolean __mmplayer_post_message(mm_player_t* player, enum MMMessageType msgtype, MMMessageParamType* param);
static gboolean __mmplayer_gst_extract_tag_from_msg(mm_player_t* player, GstMessage *msg);
+static gboolean __mmplayer_gst_handle_duration(mm_player_t* player, GstMessage* msg);
+
int __mmplayer_switch_audio_sink (mm_player_t* player);
static gboolean __mmplayer_gst_remove_fakesink(mm_player_t* player, MMPlayerGstElement* fakesink);
static int __mmplayer_check_state(mm_player_t* player, enum PlayerCommandState command);
static gboolean __mmplayer_audio_stream_probe (GstPad *pad, GstBuffer *buffer, gpointer u_data);
+static gboolean __mmplayer_video_stream_probe (GstPad *pad, GstBuffer *buffer, gpointer u_data);
+static gboolean __mmplayer_subtitle_adjust_position_probe(GstPad *pad, GstBuffer *buffer, gpointer u_data);
+static int __mmplayer_change_selector_pad (mm_player_t* player, MMPlayerTrackType type, int index);
static gboolean __mmplayer_dump_pipeline_state( mm_player_t* player );
static gboolean __mmplayer_check_subtitle( mm_player_t* player );
static gboolean __mmplayer_handle_gst_error ( mm_player_t* player, GstMessage * message, GError* error );
static gboolean __mmplayer_handle_streaming_error ( mm_player_t* player, GstMessage * message );
-static void __mmplayer_post_delayed_eos( mm_player_t* player, int delay_in_ms );
-static void __mmplayer_cancel_delayed_eos( mm_player_t* player );
+static void __mmplayer_handle_eos_delay( mm_player_t* player, int delay_in_ms );
+static void __mmplayer_cancel_eos_timer( mm_player_t* player );
static gboolean __mmplayer_eos_timer_cb(gpointer u_data);
static gboolean __mmplayer_link_decoder( mm_player_t* player,GstPad *srcpad);
static gboolean __mmplayer_link_sink( mm_player_t* player,GstPad *srcpad);
-static int __mmplayer_post_missed_plugin(mm_player_t* player);
-static int __mmplayer_check_not_supported_codec(mm_player_t* player, gchar* mime);
+static int __mmplayer_handle_missed_plugin(mm_player_t* player);
+static int __mmplayer_check_not_supported_codec(mm_player_t* player, const gchar* factory_class, const gchar* mime);
static gboolean __mmplayer_configure_audio_callback(mm_player_t* player);
static void __mmplayer_add_sink( mm_player_t* player, GstElement* sink);
-static void __mmplayer_release_signal_connection(mm_player_t* player);
+static void __mmplayer_del_sink( mm_player_t* player, GstElement* sink);
+static void __mmplayer_release_signal_connection(mm_player_t* player, MMPlayerSignalType type);
static void __mmplayer_set_antishock( mm_player_t* player, gboolean disable_by_force);
+static gpointer __mmplayer_next_play_thread(gpointer data);
static gpointer __mmplayer_repeat_thread(gpointer data);
-int _mmplayer_get_track_count(MMHandleType hplayer, MMPlayerTrackType track_type, int *count);
+static gboolean _mmplayer_update_content_attrs(mm_player_t* player, enum content_attr_flag flag);
+
+
+static gboolean __mmplayer_add_dump_buffer_probe(mm_player_t *player, GstElement *element);
+static gboolean __mmplayer_dump_buffer_probe_cb(GstPad *pad, GstBuffer *buffer, gpointer u_data);
+static void __mmplayer_release_dump_list (GList *dump_list);
static int __gst_realize(mm_player_t* player);
static int __gst_unrealize(mm_player_t* player);
static int __gst_start(mm_player_t* player);
static int __gst_stop(mm_player_t* player);
-int __gst_pause(mm_player_t* player, gboolean async);
-int __gst_resume(mm_player_t* player, gboolean async);
+static int __gst_pause(mm_player_t* player, gboolean async);
+static int __gst_resume(mm_player_t* player, gboolean async);
static gboolean __gst_seek(mm_player_t* player, GstElement * element, gdouble rate,
GstFormat format, GstSeekFlags flags, GstSeekType cur_type,
gint64 cur, GstSeekType stop_type, gint64 stop );
static int __gst_pending_seek ( mm_player_t* player );
-static int __gst_set_position(mm_player_t* player, int format, unsigned long position);
+static int __gst_set_position(mm_player_t* player, int format, unsigned long position, gboolean internal_called);
static int __gst_get_position(mm_player_t* player, int format, unsigned long *position);
static int __gst_get_buffer_position(mm_player_t* player, int format, unsigned long* start_pos, unsigned long* stop_pos);
static int __gst_adjust_subtitle_position(mm_player_t* player, int format, int position);
static int __gst_set_message_callback(mm_player_t* player, MMMessageCallback callback, gpointer user_param);
-static void __gst_set_async_state_change(mm_player_t* player, gboolean async);
static gint __gst_handle_core_error( mm_player_t* player, int code );
static gint __gst_handle_library_error( mm_player_t* player, int code );
static gint __gst_transform_gsterror( mm_player_t* player, GstMessage * message, GError* error);
static gboolean __gst_send_event_to_sink( mm_player_t* player, GstEvent* event );
-static int __mmplayer_get_video_frame_from_buffer(mm_player_t* player, GstBuffer *buffer);
static int __mmplayer_set_pcm_extraction(mm_player_t* player);
static gboolean __mmplayer_can_extract_pcm( mm_player_t* player );
static void __mmplayer_add_new_caps(GstPad* pad, GParamSpec* unused, gpointer data);
static void __mmplayer_set_unlinked_mime_type(mm_player_t* player, GstCaps *caps);
-static void __mmplayer_set_videosink_type(mm_player_t* player);
/* util */
const gchar * __get_state_name ( int state );
static gboolean __is_streaming( mm_player_t* player );
static gboolean __is_rtsp_streaming( mm_player_t* player );
+static gboolean __is_wfd_streaming( mm_player_t* player );
static gboolean __is_live_streaming ( mm_player_t* player );
static gboolean __is_http_streaming( mm_player_t* player );
static gboolean __is_http_live_streaming( mm_player_t* player );
+static gboolean __is_dash_streaming( mm_player_t* player );
+static gboolean __is_smooth_streaming( mm_player_t* player );
static gboolean __is_http_progressive_down(mm_player_t* player);
+static gboolean __has_suffix(mm_player_t * player, const gchar * suffix);
-static gboolean __mmplayer_warm_up_video_codec( mm_player_t* player, GstElementFactory *factory);
static GstBusSyncReply __mmplayer_bus_sync_callback (GstBus * bus, GstMessage * message, gpointer data);
+static int __mmplayer_realize_streaming_ext(mm_player_t* player);
+static int __mmplayer_unrealize_streaming_ext(mm_player_t *player);
+static int __mmplayer_start_streaming_ext(mm_player_t *player);
+static int __mmplayer_destroy_streaming_ext(mm_player_t* player);
+static int __mmplayer_do_change_videosink(mm_player_t* player, const int dec_index, const char *videosink_element, MMDisplaySurfaceType surface_type, void *display_overlay);
+static void __mmplayer_remove_g_source_from_context(GMainContext *context, guint source_id);
+
+static gboolean __mmplayer_verify_next_play_path(mm_player_t *player);
+static void __mmplayer_activate_next_source(mm_player_t *player, GstState target);
+static void __mmplayer_check_pipeline(mm_player_t* player);
+static gboolean __mmplayer_deactivate_selector(mm_player_t *player, MMPlayerTrackType type);
+static void __mmplayer_deactivate_old_path(mm_player_t *player);
+static int __mmplayer_ignore_current_external_display_mode(mm_player_t* player);
+#if 0 // We'll need this in future.
+static int __mmplayer_gst_switching_element(mm_player_t *player, GstElement *search_from, const gchar *removal_name, const gchar *new_element_name);
+#endif
+
+static void __mmplayer_update_buffer_setting(mm_player_t *player, GstMessage *buffering_msg);
+static GstElement *__mmplayer_element_create_and_link(mm_player_t *player, GstPad* pad, const char* name);
+static gboolean __mmplayer_can_do_interrupt(mm_player_t *player);
+
+/* device change post proc */
+void __mmplayer_device_change_post_process(gpointer user);
+void __mmplayer_set_required_cb_score(mm_player_t* player, guint score);
+void __mmplayer_inc_cb_score(mm_player_t* player);
+void __mmplayer_post_proc_reset(mm_player_t* player);
+void __mmplayer_device_change_trigger_post_process(mm_player_t* player);
+static int __mmplayer_gst_create_plain_text_elements(mm_player_t* player);
+
+
/*===========================================================================================
| |
| FUNCTION DEFINITIONS |
{
MMPlayerStateType current_state = MM_PLAYER_STATE_NUM;
MMPlayerStateType pending_state = MM_PLAYER_STATE_NUM;
- MMPlayerStateType target_state = MM_PLAYER_STATE_NUM;
- MMPlayerStateType prev_state = MM_PLAYER_STATE_NUM;
-
- debug_fenter();
+// MMPlayerStateType target_state = MM_PLAYER_STATE_NUM;
+// MMPlayerStateType prev_state = MM_PLAYER_STATE_NUM;
return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ //debug_log("incomming command : %d \n", command );
+
current_state = MMPLAYER_CURRENT_STATE(player);
pending_state = MMPLAYER_PENDING_STATE(player);
- target_state = MMPLAYER_TARGET_STATE(player);
- prev_state = MMPLAYER_PREV_STATE(player);
+// target_state = MMPLAYER_TARGET_STATE(player);
+// prev_state = MMPLAYER_PREV_STATE(player);
MMPLAYER_PRINT_STATE(player);
- debug_log("incomming command : %d", command );
-
switch( command )
{
case MMPLAYER_COMMAND_CREATE:
{
MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_NULL;
-
+
if ( current_state == MM_PLAYER_STATE_NULL ||
current_state == MM_PLAYER_STATE_READY ||
current_state == MM_PLAYER_STATE_PAUSED ||
if ( current_state == MM_PLAYER_STATE_READY )
goto NO_OP;
-
+
/* need playing/paused state to stop */
if ( current_state != MM_PLAYER_STATE_PLAYING &&
current_state != MM_PLAYER_STATE_PAUSED )
case MMPLAYER_COMMAND_RESUME:
{
- if ( MMPLAYER_IS_LIVE_STREAMING(player) )
- goto NO_OP;
if (player->doing_seek)
goto NOT_COMPLETED_SEEK;
default:
break;
}
-
- debug_log("status OK\n");
player->cmd = command;
- debug_fenter();
return MM_ERROR_NONE;
-
INVALID_STATE:
debug_warning("since player is in wrong state(%s). it's not able to apply the command(%d)",
MMPLAYER_STATE_GET_NAME(current_state), command);
return MM_ERROR_PLAYER_INVALID_STATE;
NOT_COMPLETED_SEEK:
+ debug_warning("not completed seek");
return MM_ERROR_PLAYER_DOING_SEEK;
NO_OP:
GstState element_pending_state = GST_STATE_VOID_PENDING;
GstStateChangeReturn ret = GST_STATE_CHANGE_FAILURE;
- debug_fenter();
-
+ MMPLAYER_FENTER();
+
return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
return_val_if_fail ( element, MM_ERROR_INVALID_ARGUMENT );
- debug_log("setting [%s] state to : %d\n", GST_ELEMENT_NAME(element), state);
+ debug_log("setting [%s] element state to : %s\n", GST_ELEMENT_NAME(element), gst_element_state_get_name(state));
/* set state */
ret = gst_element_set_state(element, state);
if ( ret == GST_STATE_CHANGE_FAILURE )
{
- debug_error("failed to set [%s] state to [%d]\n", GST_ELEMENT_NAME(element), state);
+ debug_error("failed to set [%s] state\n", GST_ELEMENT_NAME(element));
+
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+
return MM_ERROR_PLAYER_INTERNAL;
}
-
+
/* return here so state transition to be done in async mode */
if ( async )
{
if ( ret == GST_STATE_CHANGE_FAILURE || ( state != element_state ) )
{
- debug_error("failed to change [%s] state to [%s] within %d sec\n",
- GST_ELEMENT_NAME(element),
+ debug_error("failed to change [%s] element state to [%s] within %d sec\n",
+ GST_ELEMENT_NAME(element),
gst_element_state_get_name(state), timeout );
-
- debug_error(" [%s] state : %s pending : %s \n",
- GST_ELEMENT_NAME(element),
- gst_element_state_get_name(element_state),
+
+ debug_error(" [%s] state : %s pending : %s \n",
+ GST_ELEMENT_NAME(element),
+ gst_element_state_get_name(element_state),
gst_element_state_get_name(element_pending_state) );
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+
return MM_ERROR_PLAYER_INTERNAL;
}
- debug_log("[%s] state has changed to %s \n",
- GST_ELEMENT_NAME(element),
- gst_element_state_get_name(element_state));
+ debug_log("[%s] element state has changed\n", GST_ELEMENT_NAME(element));
+
+ MMPLAYER_FLEAVE();
- debug_fleave();
-
return MM_ERROR_NONE;
}
-
static void
-__mmplayer_videostream_cb(GstElement *element, void *stream,
-int width, int height, gpointer data) // @
+__mmplayer_videostream_cb(GstElement *element, void *data,
+int width, int height, gpointer user_data) // @
{
- mm_player_t* player = (mm_player_t*)data;
- int length = 0;
+ mm_player_t* player = (mm_player_t*)user_data;
return_if_fail ( player );
- if (player->video_stream_cb )
- {
- length = width * height * 4; // for rgb 32bit
- player->video_stream_cb(stream, length, player->video_stream_cb_user_param, width, height);
- }
-}
-
-gboolean
-_mmplayer_update_content_attrs(mm_player_t* player) // @
-{
- GstFormat fmt = GST_FORMAT_TIME;
- gint64 dur_nsec = 0;
- GstStructure* p = NULL;
- MMHandleType attrs = 0;
- gint retry_count = 0;
- gint retry_count_max = 10;
- gchar *path = NULL;
- struct stat sb;
-
- return_val_if_fail ( player, FALSE );
+ MMPLAYER_FENTER();
- if ( ! player->need_update_content_attrs )
+ if (player->is_drm_file)
{
- debug_log("content attributes are already updated");
- return TRUE;
+ MMMessageParamType msg_param = { 0, };
+ debug_warning("not supported in drm file");
+ msg_param.code = MM_ERROR_PLAYER_DRM_OUTPUT_PROTECTION;
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
}
-
- /* get content attribute first */
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
+ else if ( !player->set_mode.media_packet_video_stream && player->video_stream_cb)
{
- debug_error("cannot get content attribute");
- return FALSE;
- }
+ MMPlayerVideoStreamDataType stream;
- /* update duration
- * NOTE : we need to wait for a while until is possible to get duration from pipeline
- * as getting duration timing is depends on behavier of demuxers ( or etc ).
- * we set timeout 100ms * 10 as initial value. fix it if needed.
- */
- if ( player->need_update_content_dur )
- {
- while ( retry_count < retry_count_max)
- {
- if ( FALSE == gst_element_query_duration( player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
- &fmt, &dur_nsec ) )
- {
- /* retry if failed */
- debug_warning("failed to get duraton. waiting 100ms and then retrying...");
- usleep(100000);
- retry_count++;
- continue;
- }
+ /* clear stream data structure */
+ memset(&stream, 0x0, sizeof(MMPlayerVideoStreamDataType));
- if ( dur_nsec == 0 && ( !MMPLAYER_IS_LIVE_STREAMING( player ) ) )
- {
- /* retry if duration is zero in case of not live stream */
- debug_warning("returned duration is zero. but it's not an live stream. retrying...");
- usleep(100000);
- retry_count++;
- continue;
- }
+ stream.data = data;
+ stream.length_total = width * height * 4; // for rgb 32bit
+ stream.height = height;
+ stream.width = width;
+ player->video_stream_cb(&stream, player->video_stream_cb_user_param);
+ }
- break;
- }
+ MMPLAYER_FLEAVE();
+}
- player->duration = dur_nsec;
- debug_log("duration : %lld msec", GST_TIME_AS_MSECONDS(dur_nsec));
+static void
+__mmplayer_videoframe_render_error_cb(GstElement *element, void *error_id, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
- /* try to get streaming service type */
- __mmplayer_update_stream_service_type( player );
+ return_if_fail ( player );
- /* check duration is OK */
- if ( dur_nsec == 0 && !MMPLAYER_IS_LIVE_STREAMING( player ) )
+ MMPLAYER_FENTER();
+
+ if (player->video_frame_render_error_cb )
+ {
+ if (player->attrs)
{
- /* FIXIT : find another way to get duration here. */
- debug_error("finally it's failed to get duration from pipeline. progressbar will not work correctely!");
+ int surface_type = 0;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &surface_type);
+ switch (surface_type)
+ {
+ case MM_DISPLAY_SURFACE_X_EXT:
+ player->video_frame_render_error_cb((unsigned int*)error_id, player->video_frame_render_error_cb_user_param);
+ debug_log("display surface type(X_EXT) : render error callback(%p) is finished", player->video_frame_render_error_cb);
+ break;
+ default:
+ debug_error("video_frame_render_error_cb was set, but this surface type(%d) is not supported", surface_type);
+ break;
+ }
}
else
{
- player->need_update_content_dur = FALSE;
+ debug_error("could not get surface type");
}
-
- /*update duration */
- mm_attrs_set_int_by_name(attrs, "content_duration", GST_TIME_AS_MSECONDS(dur_nsec));
}
else
{
- debug_log("no ready to get or content duration already updated");
+ debug_warning("video_frame_render_error_cb was not set");
}
- /* update rate, channels */
- if ( player->pipeline->audiobin &&
- player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
- {
- GstCaps *caps_a = NULL;
- GstPad* pad = NULL;
- gint samplerate = 0, channels = 0;
+ MMPLAYER_FLEAVE();
+}
- pad = gst_element_get_static_pad(
- player->pipeline->audiobin[MMPLAYER_A_CONV].gst, "sink" );
+void
+__mmplayer_device_change_post_process(gpointer user)
+{
+ mm_player_t* player = (mm_player_t*)user;
+ unsigned long position = 0;
+ MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
+ MMPlayerStateType pending_state = MM_PLAYER_STATE_NONE;
- if ( pad )
- {
- caps_a = gst_pad_get_negotiated_caps( pad );
+ MMPLAYER_FENTER();
- if ( caps_a )
- {
- p = gst_caps_get_structure (caps_a, 0);
+ if (! player ||
+ ! player->pipeline ||
+ ! player->pipeline->mainbin ||
+ ! player->pipeline->mainbin[MMPLAYER_M_PIPE].gst )
+ {
+ goto EXIT;
+ }
- gst_structure_get_int (p, "rate", &samplerate);
- mm_attrs_set_int_by_name(attrs, "content_audio_samplerate", samplerate);
+ current_state = MMPLAYER_CURRENT_STATE(player);
+ pending_state = MMPLAYER_PENDING_STATE(player);
- gst_structure_get_int (p, "channels", &channels);
- mm_attrs_set_int_by_name(attrs, "content_audio_channels", channels);
- gst_caps_unref( caps_a );
- caps_a = NULL;
+ if (player->post_proc.need_pause_and_resume)
+ {
+ debug_log("pausing");
+ if ((pending_state == MM_PLAYER_STATE_PLAYING) ||
+ ((pending_state == MM_PLAYER_STATE_NONE) && (current_state != MM_PLAYER_STATE_PAUSED)))
+ gst_element_set_state(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PAUSED);
+ }
- debug_log("samplerate : %d channels : %d", samplerate, channels);
- }
- else
- {
- debug_warning("not ready to get audio caps");
- }
+ /* seek should be done within pause and resume */
+ if (player->post_proc.need_seek)
+ {
+ debug_log("seeking");
+ __gst_get_position(player, MM_PLAYER_POS_FORMAT_TIME, &position);
+ debug_log(">> seek to current position = %ld ms", position);
+ __gst_set_position(player, MM_PLAYER_POS_FORMAT_TIME, position, TRUE);
+ }
- gst_object_unref( pad );
- }
- else
- {
- debug_warning("failed to get pad from audiosink");
- }
+ if (player->post_proc.need_pause_and_resume)
+ {
+ debug_log("resuming");
+ if ((pending_state == MM_PLAYER_STATE_PLAYING) ||
+ ((pending_state == MM_PLAYER_STATE_NONE) && (current_state != MM_PLAYER_STATE_PAUSED)))
+ gst_element_set_state(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PLAYING);
}
- /* update width, height, framerate */
- if ( player->pipeline->videobin &&
- player->pipeline->videobin[MMPLAYER_V_SINK].gst )
+ /* async */
+ if (player->post_proc.need_async)
{
- GstCaps *caps_v = NULL;
- GstPad* pad = NULL;
- gint tmpNu, tmpDe;
- gint width, height;
+ debug_log("setting async");
+ debug_log(">> player->display_stat = %d", player->display_stat);
- if (player->use_multi_surface)
+ if ((player->display_stat == MMPLAYER_DISPLAY_STATUS_HDMI_ACTIVE) ||
+ (player->display_stat == MMPLAYER_DISPLAY_STATUS_UNKNOWN_ACTIVE))
{
- /* NOTE : if v_stream_caps were deprecated, it might be implemented by using "pad-added" signal callback */
- if (player->v_stream_caps)
- {
- caps_v = player->v_stream_caps;
- p = gst_caps_get_structure (caps_v, 0);
- gst_structure_get_int (p, "width", &width);
- mm_attrs_set_int_by_name(attrs, "content_video_width", width);
+ g_object_set(G_OBJECT(player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "async", TRUE, NULL);
+ }
+ /* TODO : need some comment here */
+ else if (player->display_stat == MMPLAYER_DISPLAY_STATUS_NULL)
+ {
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_FAKE_SINK].gst), "async", TRUE, NULL);
+ }
+ }
- gst_structure_get_int (p, "height", &height);
- mm_attrs_set_int_by_name(attrs, "content_video_height", height);
+EXIT:
+ /* reset all */
+ __mmplayer_post_proc_reset(player);
+ return;
+}
- gst_structure_get_fraction (p, "framerate", &tmpNu, &tmpDe);
+void __mmplayer_set_required_cb_score(mm_player_t* player, guint score)
+{
+ return_if_fail(player);
+ player->post_proc.required_cb_score = score;
+ debug_log("set required score to : %d", score);
+}
- debug_log("width : %d height : %d", width, height );
+void __mmplayer_inc_cb_score(mm_player_t* player)
+{
+ return_if_fail(player);
+ player->post_proc.cb_score++;
+ debug_log("post proc cb score increased to %d", player->post_proc.cb_score);
+}
- if (tmpDe > 0)
- {
- mm_attrs_set_int_by_name(attrs, "content_video_fps", tmpNu / tmpDe);
- debug_log("fps : %d", tmpNu / tmpDe);
- }
- }
- else
- {
- debug_warning("failed to get caps from v_stream_caps when using multi-surface");
- }
- }
- else
+void __mmplayer_post_proc_reset(mm_player_t* player)
+{
+ return_if_fail(player);
+
+ /* check if already triggered */
+ if (player->post_proc.id)
+ {
+ /* TODO : need to consider multiple main context. !!!! */
+ if (FALSE == g_source_remove(player->post_proc.id) )
{
- pad = gst_element_get_static_pad( player->pipeline->videobin[MMPLAYER_V_SINK].gst, "sink" );
- if ( pad )
- {
- caps_v = gst_pad_get_negotiated_caps( pad );
- if (caps_v)
- {
- p = gst_caps_get_structure (caps_v, 0);
- gst_structure_get_int (p, "width", &width);
- mm_attrs_set_int_by_name(attrs, "content_video_width", width);
+ debug_error("failed to remove exist post_proc item");
+ }
+ player->post_proc.id = 0;
+ }
- gst_structure_get_int (p, "height", &height);
- mm_attrs_set_int_by_name(attrs, "content_video_height", height);
+ memset(&player->post_proc, 0, sizeof(mm_player_post_proc_t));
- gst_structure_get_fraction (p, "framerate", &tmpNu, &tmpDe);
+ /* set default required cb score 1 as only audio device has changed in this case.
+ if display status is changed with audio device, required cb score is set 2 in display status callback.
+ this logic bases on the assumption which audio device callback is called after calling display status callback. */
+ player->post_proc.required_cb_score = 1;
+}
- debug_log("width : %d height : %d", width, height );
+void
+__mmplayer_device_change_trigger_post_process(mm_player_t* player)
+{
+ return_if_fail(player);
- gst_caps_unref( caps_v );
- caps_v = NULL;
+ /* check score */
+ if ( player->post_proc.cb_score < player->post_proc.required_cb_score )
+ {
+ /* wait for next turn */
+ debug_log("wait for next turn. required cb score : %d current score : %d\n",
+ player->post_proc.required_cb_score, player->post_proc.cb_score);
+ return;
+ }
- if (tmpDe > 0)
- {
- mm_attrs_set_int_by_name(attrs, "content_video_fps", tmpNu / tmpDe);
- debug_log("fps : %d", tmpNu / tmpDe);
- }
- }
- else
- {
- debug_warning("failed to get negitiated caps from videosink");
- }
- if (!player->use_multi_surface)
- {
- gst_object_unref( pad );
- pad = NULL;
- }
- }
- else
- {
- debug_warning("failed to get pad from videosink");
- }
+ /* check if already triggered */
+ if (player->post_proc.id)
+ {
+ /* TODO : need to consider multiple main context. !!!! */
+ if (FALSE == g_source_remove(player->post_proc.id) )
+ {
+ debug_error("failed to remove exist post_proc item");
}
+ player->post_proc.id = 0;
}
- if (player->duration)
- {
- guint64 data_size = 0;
+ player->post_proc.id = g_idle_add((GSourceFunc)__mmplayer_device_change_post_process, (gpointer)player);
+}
- if (!MMPLAYER_IS_STREAMING(player) && (player->can_support_codec & FOUND_PLUGIN_VIDEO))
- {
- mm_attrs_get_string_by_name(attrs, "profile_uri", &path);
+/* NOTE : Sound module has different latency according to output device So,
+ * synchronization problem can be happened whenever device is changed.
+ * To avoid this issue, we do reset avsystem or seek as workaroud.
+ */
+static void
+__mmplayer_sound_device_info_changed_cb_func (MMSoundDevice_t device_h, int changed_info_type, void *user_data)
+{
+ int ret;
+ mm_sound_device_type_e device_type;
+ mm_player_t* player = (mm_player_t*) user_data;
- if (stat(path, &sb) == 0)
- {
- data_size = (guint64)sb.st_size;
- }
- }
- else if (MMPLAYER_IS_HTTP_STREAMING(player))
- {
- data_size = player->http_content_size;
- }
+ return_if_fail( player );
- if (data_size)
- {
- guint64 bitrate = 0;
- guint64 msec_dur = 0;
+ debug_warning("device_info_changed_cb is called, device_h[0x%x], changed_info_type[%d]\n", device_h, changed_info_type);
- msec_dur = GST_TIME_AS_MSECONDS(player->duration);
- bitrate = data_size * 8 * 1000 / msec_dur;
- debug_log("file size : %u, video bitrate = %llu", data_size, bitrate);
- mm_attrs_set_int_by_name(attrs, "content_video_bitrate", bitrate);
- }
- }
+ __mmplayer_inc_cb_score(player);
+ /* get device type with device_h*/
+ ret = mm_sound_get_device_type(device_h, &device_type);
+ if (ret) {
+ debug_error("failed to mm_sound_get_device_type()\n");
+ }
- /* validate all */
- if ( mmf_attrs_commit ( attrs ) )
+ /* do pause and resume only if video is playing */
+ if ( player->videodec_linked && MMPLAYER_CURRENT_STATE(player) == MM_PLAYER_STATE_PLAYING )
{
- debug_error("failed to update attributes\n");
- return FALSE;
- }
+ switch (device_type)
+ {
+ case MM_SOUND_DEVICE_TYPE_BLUETOOTH:
+ case MM_SOUND_DEVICE_TYPE_AUDIOJACK:
+ case MM_SOUND_DEVICE_TYPE_BUILTIN_SPEAKER:
+ case MM_SOUND_DEVICE_TYPE_HDMI:
+ case MM_SOUND_DEVICE_TYPE_MIRRORING:
+ {
+ player->post_proc.need_pause_and_resume = TRUE;
+ }
+ break;
- player->need_update_content_attrs = FALSE;
+ default:
+ debug_log("do nothing");
+ }
+ }
+ debug_warning("dispatched");
- return TRUE;
+ __mmplayer_device_change_trigger_post_process(player);
}
-gboolean __mmplayer_update_stream_service_type( mm_player_t* player )
+static void
+__mmplayer_display_status_cb(GstElement *element, MMPlayerDisplayStatus stat, gpointer data)
{
- MMHandleType attrs = 0;
- gint streaming_type = STREAMING_SERVICE_NONE;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*)data;
+ return_if_fail ( player );
+ MMPLAYER_FENTER();
- return_val_if_fail ( player &&
- player->pipeline &&
- player->pipeline->mainbin &&
- player->pipeline->mainbin[MMPLAYER_M_SRC].gst,
- FALSE );
+ guint required_cb_score = 2;
- /* streaming service type if streaming */
- if ( ! MMPLAYER_IS_STREAMING(player) );
- return FALSE;
+ __mmplayer_inc_cb_score(player);
+ __mmplayer_set_required_cb_score(player, required_cb_score);
- if (MMPLAYER_IS_RTSP_STREAMING(player))
+ if ((player->pipeline->textbin == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_TEXT_IDENTITY].gst == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst == NULL))
{
- /* get property from rtspsrc element */
- g_object_get(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_SRC].gst), "service_type", &streaming_type, NULL);
+ debug_warning("text bin is not ready yet");
+ goto EXIT;
}
- else if (MMPLAYER_IS_HTTP_STREAMING(player))
+
+ if (player->is_subtitle_force_drop == TRUE)
{
- if ( player->duration <= 0)
- streaming_type = STREAMING_SERVICE_LIVE;
- else
- streaming_type = STREAMING_SERVICE_VOD;
+ debug_warning("subtitle is dropping");
+ player->display_stat = stat;
+ goto EXIT;
}
-
- player->streaming_type = streaming_type;
- if ( player->streaming_type == STREAMING_SERVICE_LIVE)
+ if (stat != util_get_is_connected_external_display())
{
- debug_log("It's live streaming. pause/resume/seek are not working.\n");
+ debug_warning("stat is wrong!! (%d from xvimagesink)", stat);
+ goto EXIT;
}
- else if (player->streaming_type == STREAMING_SERVICE_LIVE)
+
+ if ((stat == MMPLAYER_DISPLAY_STATUS_HDMI_ACTIVE) || (stat == MMPLAYER_DISPLAY_STATUS_UNKNOWN_ACTIVE))
{
- debug_log("It's vod streaming. pause/resume/seek are working.\n");
+ MMMessageParamType msg = {0, };
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst), "drop-probability", (gfloat)1.0, NULL);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_TEXT_IDENTITY].gst), "drop-probability", (gfloat)0.0, NULL);
+ debug_log("external display is ACTIVE");
+
+ msg.data = NULL;
+ msg.subtitle.duration = 0;
+
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_UPDATE_SUBTITLE, &msg );
}
- else
+ else if (stat == MMPLAYER_DISPLAY_STATUS_NULL)
{
- debug_warning("fail to determine streaming type. pause/resume/seek may not working properly if stream is live stream\n");
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst), "drop-probability", (gfloat)0.0, NULL);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_TEXT_IDENTITY].gst), "drop-probability", (gfloat)1.0, NULL); //If it is not ACTIVE...
+ debug_log("external display is NULL");
}
- /* get profile attribute */
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
+ if (player->display_stat != stat)
{
- debug_error("cannot get content attribute\n");
- return FALSE;
- }
+ player->display_stat = stat;
- mm_attrs_set_int_by_name ( attrs, "streaming_type", streaming_type );
- /* validate all */
- if ( mmf_attrs_commit ( attrs ) )
- {
- debug_warning("updating streaming service type failed. pause/resume/seek may not working properly if stream is live stream\n");
- return FALSE;
+ g_object_set(G_OBJECT(player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "async", FALSE, NULL);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_FAKE_SINK].gst), "async", FALSE, NULL);
+
+ /* setup post processing */
+ player->post_proc.need_async = TRUE;
+ player->post_proc.need_seek = TRUE;
+ player->post_proc.need_pause_and_resume = TRUE;
}
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return TRUE;
+EXIT:
+ __mmplayer_device_change_trigger_post_process(player);
+ return;
}
-
-/* this function sets the player state and also report
- * it to applicaton by calling callback function
- */
-static gboolean
-__mmplayer_set_state(mm_player_t* player, int state) // @
+static void
+__mmplayer_external_resolution_cb(GstElement *element, int width, int height, gpointer data)
{
- MMMessageParamType msg = {0, };
- int asm_result = MM_ERROR_NONE;
- int new_state = state;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*)data;
+ return_if_fail ( player );
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, FALSE );
+ if ((player->pipeline->textbin == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_RENDER].gst == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst == NULL))
+ {
+ debug_warning("text bin is not ready yet");
+ return;
+ }
- if ( MMPLAYER_CURRENT_STATE(player) == new_state )
+ if((width==0 || height==0) && util_get_is_connected_external_display()!=MMPLAYER_DISPLAY_STATUS_NULL)
{
- debug_warning("already same state(%s)\n", MMPLAYER_STATE_GET_NAME(state));
- return TRUE;
+ debug_warning("stat is wrong!! skip setting external resolution to 0");
+ return;
}
+ debug_log("set external width and height : %d * %d", width, height);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_RENDER].gst), "external-width", width, NULL);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_RENDER].gst), "external-height", height, NULL);
- /* update player states */
- MMPLAYER_PREV_STATE(player) = MMPLAYER_CURRENT_STATE(player);
- MMPLAYER_CURRENT_STATE(player) = new_state;
- if ( MMPLAYER_CURRENT_STATE(player) == MMPLAYER_PENDING_STATE(player) )
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
-
- /* print state */
- MMPLAYER_PRINT_STATE(player);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "external-width", width, NULL);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "external-height", height, NULL);
- /* post message to application */
- if (MMPLAYER_TARGET_STATE(player) == new_state)
- {
- /* fill the message with state of player */
- msg.state.previous = MMPLAYER_PREV_STATE(player);
- msg.state.current = MMPLAYER_CURRENT_STATE(player);
+ MMPLAYER_FLEAVE();
+}
- /* state changed by asm callback */
- if ( player->sm.by_asm_cb )
- {
- msg.union_type = MM_MSG_UNION_CODE;
- msg.code = player->sm.event_src;
- MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_INTERRUPTED, &msg );
- }
- /* state changed by usecase */
- else
- {
- MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_CHANGED, &msg );
- }
+static void
+__mmplayer_hided_window_cb(GstElement *element, gboolean is_hided , gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ return_if_fail ( player );
+ MMPLAYER_FENTER();
- debug_log ("player reach the target state, then do something in each state(%s).\n",
- MMPLAYER_STATE_GET_NAME(MMPLAYER_TARGET_STATE(player)));
- }
- else
+ if ((player->pipeline->textbin == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst == NULL))
{
- debug_log ("intermediate state, do nothing.\n");
- return TRUE;
+ debug_warning("text_xvimagesink is not ready yet");
+ return;
}
-
- switch ( MMPLAYER_TARGET_STATE(player) )
- {
- case MM_PLAYER_STATE_NULL:
- case MM_PLAYER_STATE_READY:
- {
- if (player->cmd == MMPLAYER_COMMAND_STOP)
- {
- asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_STOP);
- if ( asm_result != MM_ERROR_NONE )
- {
- debug_error("failed to set asm state to stop\n");
- return FALSE;
- }
- }
- }
- break;
-
- case MM_PLAYER_STATE_PAUSED:
- {
- /* special care for local playback. normaly we can get some content attribute
- * when the demuxer is changed to PAUSED. so we are trying it. it will be tried again
- * when PLAYING state has signalled if failed.
- * note that this is only happening pause command has come before the state of pipeline
- * reach to the PLAYING.
- */
- _mmplayer_update_content_attrs( player );
-
- /* add audio callback probe if condition is satisfied */
- if ( ! player->audio_cb_probe_id && player->is_sound_extraction )
- __mmplayer_configure_audio_callback(player);
-
- asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_PAUSE);
- if ( asm_result )
- {
- debug_error("failed to set asm state to PAUSE\n");
- return FALSE;
- }
- }
- break;
-
- case MM_PLAYER_STATE_PLAYING:
- {
- /* update attributes which are only available on playing status */
- player->need_update_content_attrs = TRUE;
- _mmplayer_update_content_attrs ( player );
-
- if ( player->cmd == MMPLAYER_COMMAND_START && !player->sent_bos )
- {
- __mmplayer_post_missed_plugin ( player );
-
- /* update video resource status */
- if ( ( player->can_support_codec & 0x02) == FOUND_PLUGIN_VIDEO )
- {
- asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_PLAYING);
- if ( asm_result )
- {
- MMMessageParamType msg = {0, };
-
- debug_error("failed to go ahead because of video conflict\n");
-
- msg.union_type = MM_MSG_UNION_CODE;
- msg.code = MM_ERROR_POLICY_INTERRUPTED;
- MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_INTERRUPTED, &msg);
-
- _mmplayer_unrealize((MMHandleType)player);
-
- return FALSE;
- }
- }
- }
-
- if ( player->resumed_by_rewind && player->playback_rate < 0.0 )
- {
- /* initialize because auto resume is done well. */
- player->resumed_by_rewind = FALSE;
- player->playback_rate = 1.0;
- }
-
- if ( !player->sent_bos )
- {
- /* check audio codec field is set or not
- * we can get it from typefinder or codec's caps.
- */
- gchar *audio_codec = NULL;
- mm_attrs_get_string_by_name(player->attrs, "content_audio_codec", &audio_codec);
-
- /* The codec format can't be sent for audio only case like amr, mid etc.
- * Because, parser don't make related TAG.
- * So, if it's not set yet, fill it with found data.
- */
- if ( ! audio_codec )
- {
- if ( g_strrstr(player->type, "audio/midi"))
- {
- audio_codec = g_strdup("MIDI");
-
- }
- else if ( g_strrstr(player->type, "audio/x-amr"))
- {
- audio_codec = g_strdup("AMR");
- }
- else if ( g_strrstr(player->type, "audio/mpeg") && !g_strrstr(player->type, "mpegversion=(int)1"))
- {
- audio_codec = g_strdup("AAC");
- }
- else
- {
- audio_codec = g_strdup("unknown");
- }
- mm_attrs_set_string_by_name(player->attrs, "content_audio_codec", audio_codec);
-
- MMPLAYER_FREEIF(audio_codec);
- mmf_attrs_commit(player->attrs);
- debug_log("set audio codec type with caps\n");
- }
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "hided-window", is_hided, NULL);
+ debug_log("update window-hide status : %d", is_hided);
- MMTA_ACUM_ITEM_END("[KPI] start media player service", FALSE);
- MMTA_ACUM_ITEM_END("[KPI] media player service create->playing", FALSE);
+ MMPLAYER_FLEAVE();
+}
- MMPLAYER_POST_MSG ( player, MM_MESSAGE_BEGIN_OF_STREAM, NULL );
- player->sent_bos = TRUE;
- }
- }
- break;
+static void
+__mmplayer_quick_panel_on_cb(GstElement *element, gboolean quick_panel_on , gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ return_if_fail ( player );
+ MMPLAYER_FENTER();
- case MM_PLAYER_STATE_NONE:
- default:
- debug_warning("invalid target state, there is nothing to do.\n");
- break;
+ if ((player->pipeline->textbin == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst == NULL))
+ {
+ debug_warning("text_xvimagesink is not ready yet");
+ return;
}
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "quick-panel-on", quick_panel_on, NULL);
+ debug_log("update quick panel status : %d", quick_panel_on);
- debug_fleave();
-
- return TRUE;
+ MMPLAYER_FLEAVE();
}
-
-gboolean
-__mmplayer_post_message(mm_player_t* player, enum MMMessageType msgtype, MMMessageParamType* param) // @
+static void
+__mmplayer_multiwindow_active_cb(GstElement *element, gboolean multiwindow_active , gpointer data)
{
- return_val_if_fail( player, FALSE );
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*)data;
+ return_if_fail ( player );
- if ( !player->msg_cb )
+ if ((player->pipeline->textbin == NULL) ||
+ (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst == NULL))
{
- debug_warning("no msg callback. can't post\n");
- return FALSE;
+ return;
+ }
+ if (player->last_multiwin_status != multiwindow_active)
+ {
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst), "multiwindow-active", multiwindow_active, NULL);
+ player->last_multiwin_status = multiwindow_active;
+ debug_log("update multiwindow-active status : %d", multiwindow_active);
}
-
- //debug_log("Message (type : %d) will be posted using msg-cb(%p). \n", msgtype, player->msg_cb);
-
- player->msg_cb(msgtype, param, player->msg_cb_param);
-
- debug_fleave();
-
- return TRUE;
}
-static int
-__mmplayer_get_state(mm_player_t* player) // @
+/* This function should be called after the pipeline goes PAUSED or higher
+state. */
+gboolean
+_mmplayer_update_content_attrs(mm_player_t* player, enum content_attr_flag flag) // @
{
- int state = MM_PLAYER_STATE_NONE;
-
- debug_fenter();
-
- return_val_if_fail ( player, MM_PLAYER_STATE_NONE );
-
- state = MMPLAYER_CURRENT_STATE(player);
-
- debug_log("player state is %s.\n", MMPLAYER_STATE_GET_NAME(state));
+ static gboolean has_duration = FALSE;
+ static gboolean has_video_attrs = FALSE;
+ static gboolean has_audio_attrs = FALSE;
+ static gboolean has_bitrate = FALSE;
+ gboolean missing_only = FALSE;
+ gboolean all = FALSE;
- debug_fleave();
-
- return state;
-}
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 dur_nsec = 0;
+ GstStructure* p = NULL;
+ MMHandleType attrs = 0;
+ gchar *path = NULL;
+ gint stream_service_type = STREAMING_SERVICE_NONE;
+ struct stat sb;
-static void
-__gst_set_async_state_change(mm_player_t* player, gboolean async)
-{
- debug_fenter();
-
- return_if_fail( player && player->pipeline && player->pipeline->mainbin );
+ MMPLAYER_FENTER();
- /* need only when we are using decodebin */
- if ( ! PLAYER_INI()->use_decodebin )
- return;
+ return_val_if_fail ( player, FALSE );
- /* audio sink */
- if ( player->pipeline->audiobin &&
- player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
+ /* check player state here */
+ if ( MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PAUSED &&
+ MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PLAYING )
{
- debug_log("audiosink async : %d\n", async);
- g_object_set (G_OBJECT (player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "async", async, NULL);
+ /* give warning now only */
+ debug_warning("be careful. content attributes may not available in this state ");
}
- /* video sink */
- if ( player->pipeline->videobin &&
- player->pipeline->videobin[MMPLAYER_V_SINK].gst )
+ /* get content attribute first */
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
{
- debug_log("videosink async : %d\n", async);
- g_object_set (G_OBJECT (player->pipeline->videobin[MMPLAYER_V_SINK].gst), "async", async, NULL);
+ debug_error("cannot get content attribute");
+ return FALSE;
}
- /* decodebin if enabled */
- if ( PLAYER_INI()->use_decodebin )
+ /* get update flag */
+
+ if ( flag & ATTR_MISSING_ONLY )
{
- debug_log("decodebin async : %d\n", async);
- g_object_set (G_OBJECT (player->pipeline->mainbin[MMPLAYER_M_AUTOPLUG].gst), "async-handling", async, NULL);
+ missing_only = TRUE;
+ debug_log("updating missed attr only");
}
- debug_fleave();
-}
+ if ( flag & ATTR_ALL )
+ {
+ all = TRUE;
+ has_duration = FALSE;
+ has_video_attrs = FALSE;
+ has_audio_attrs = FALSE;
+ has_bitrate = FALSE;
-static gpointer __mmplayer_repeat_thread(gpointer data)
-{
- mm_player_t* player = (mm_player_t*) data;
- gboolean ret_value = FALSE;
- MMHandleType attrs = 0;
- gint count = 0;
+ debug_log("updating all attrs");
+ }
- return_val_if_fail ( player, NULL );
+ if ( missing_only && all )
+ {
+ debug_warning("cannot use ATTR_MISSING_ONLY and ATTR_ALL. ignoring ATTR_MISSING_ONLY flag!");
+ missing_only = FALSE;
+ }
- while ( ! player->repeat_thread_exit )
+ if ( (flag & ATTR_DURATION) || (!has_duration && missing_only) || all )
{
- debug_log("repeat thread started. waiting for signal.\n");
- g_cond_wait( player->repeat_thread_cond, player->repeat_thread_mutex );
+ debug_log("try to update duration");
+ has_duration = FALSE;
- if ( player->repeat_thread_exit )
+ if (gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &dur_nsec ))
{
- debug_log("exiting repeat thread\n");
- break;
+ player->duration = dur_nsec;
+ debug_warning("duration : %lld msec", GST_TIME_AS_MSECONDS(dur_nsec));
}
- /* lock */
- MMPLAYER_CMD_LOCK( player );
-
- return_val_if_fail( player, NULL );
-
- attrs = MMPLAYER_GET_ATTRS(player);
+ /* try to get streaming service type */
+ stream_service_type = __mmplayer_get_stream_service_type( player );
+ mm_attrs_set_int_by_name ( attrs, "streaming_type", stream_service_type );
- if (mm_attrs_get_int_by_name(attrs, "profile_play_count", &count) != MM_ERROR_NONE)
+ /* check duration is OK */
+ if ( dur_nsec == 0 && !MMPLAYER_IS_LIVE_STREAMING( player ) )
{
- debug_error("can not get play count\n");
- break;
+ /* FIXIT : find another way to get duration here. */
+ debug_error("finally it's failed to get duration from pipeline. progressbar will not work correctely!");
}
-
- if ( player->section_repeat )
+ else
{
- ret_value = _mmplayer_activate_section_repeat((MMHandleType)player, player->section_repeat_start, player->section_repeat_end);
+ /*update duration */
+ mm_attrs_set_int_by_name(attrs, "content_duration", GST_TIME_AS_MSECONDS(dur_nsec));
+ has_duration = TRUE;
}
- else
+ }
+
+ if ( (flag & ATTR_AUDIO) || (!has_audio_attrs && missing_only) || all )
+ {
+ /* update audio params
+ NOTE : We need original audio params and it can be only obtained from src pad of audio
+ decoder. Below code only valid when we are not using 'resampler' just before
+ 'audioconverter'. */
+
+ debug_log("try to update audio attrs");
+ has_audio_attrs = FALSE;
+
+ if ( player->pipeline->audiobin &&
+ player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
{
- if ( player->playback_rate < 0.0 )
+ GstCaps *caps_a = NULL;
+ GstPad* pad = NULL;
+ gint samplerate = 0, channels = 0;
+
+ pad = gst_element_get_static_pad(
+ player->pipeline->audiobin[MMPLAYER_A_CONV].gst, "sink" );
+
+ if ( pad )
{
- player->resumed_by_rewind = TRUE;
- _mmplayer_set_mute((MMHandleType)player, 0);
- MMPLAYER_POST_MSG( player, MM_MESSAGE_RESUMED_BY_REW, NULL );
- }
+ caps_a = gst_pad_get_negotiated_caps( pad );
- ret_value = __gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, 1.0,
- GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET,
- 0, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);
+ if ( caps_a )
+ {
+ p = gst_caps_get_structure (caps_a, 0);
- /* initialize */
- player->sent_bos = FALSE;
- }
+ mm_attrs_get_int_by_name(attrs, "content_audio_samplerate", &samplerate);
- if ( ! ret_value )
- {
- debug_error("failed to set position to zero for rewind\n");
- continue;
- }
+ gst_structure_get_int (p, "rate", &samplerate);
+ mm_attrs_set_int_by_name(attrs, "content_audio_samplerate", samplerate);
- /* decrease play count */
- if ( count > 1 )
- {
- /* we successeded to rewind. update play count and then wait for next EOS */
- count--;
+ gst_structure_get_int (p, "channels", &channels);
+ mm_attrs_set_int_by_name(attrs, "content_audio_channels", channels);
- mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
+ secure_debug_log("samplerate : %d channels : %d", samplerate, channels);
- /* commit attribute */
- if ( mmf_attrs_commit ( attrs ) )
+ gst_caps_unref( caps_a );
+ caps_a = NULL;
+
+ has_audio_attrs = TRUE;
+ }
+ else
+ {
+ debug_warning("not ready to get audio caps");
+ }
+
+ gst_object_unref( pad );
+ }
+ else
{
- debug_error("failed to commit attribute\n");
+ debug_warning("failed to get pad from audiosink");
}
}
+ }
- /* unlock */
- MMPLAYER_CMD_UNLOCK( player );
- }
-
- return NULL;
-}
-
-static void
-__mmplayer_handle_buffering_message ( mm_player_t* player )
-{
- MMPlayerStateType prev_state = MM_PLAYER_STATE_NONE;
- MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
- MMPlayerStateType target_state = MM_PLAYER_STATE_NONE;
- MMPlayerStateType pending_state = MM_PLAYER_STATE_NONE;
-
- return_if_fail ( player );
-
- prev_state = MMPLAYER_PREV_STATE(player),
- current_state = MMPLAYER_CURRENT_STATE(player);
- target_state = MMPLAYER_TARGET_STATE(player);
- pending_state = MMPLAYER_PENDING_STATE(player);
-
- if ( !player->streamer->is_buffering )
+ if ( (flag & ATTR_VIDEO) || (!has_video_attrs && missing_only) || all )
{
- debug_log( "player state : prev %s, current %s, pending %s, target %s \n",
- MMPLAYER_STATE_GET_NAME(prev_state),
- MMPLAYER_STATE_GET_NAME(current_state),
- MMPLAYER_STATE_GET_NAME(pending_state),
- MMPLAYER_STATE_GET_NAME(target_state));
+ debug_log("try to update video attrs");
+ has_video_attrs = FALSE;
- /* NOTE : if buffering has done, player has to go to target state. */
- switch ( target_state )
+ if ( player->pipeline->videobin &&
+ player->pipeline->videobin[MMPLAYER_V_SINK].gst )
{
- case MM_PLAYER_STATE_PAUSED :
- {
- if ( MMPLAYER_IS_RTSP_STREAMING(player) )
- return;
-
- switch ( pending_state )
- {
- case MM_PLAYER_STATE_PLAYING:
- {
- __gst_pause ( player, TRUE );
- }
- break;
-
- case MM_PLAYER_STATE_PAUSED:
- {
- debug_log("player is already going to paused state, there is nothing to do.\n");
- }
- break;
-
- case MM_PLAYER_STATE_NONE:
- case MM_PLAYER_STATE_NULL:
- case MM_PLAYER_STATE_READY:
- default :
- {
- debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
- }
- break;
- }
- }
- break;
+ GstCaps *caps_v = NULL;
+ GstPad* pad = NULL;
+ gint tmpNu, tmpDe;
+ gint width, height;
- case MM_PLAYER_STATE_PLAYING :
+ pad = gst_element_get_static_pad( player->pipeline->videobin[MMPLAYER_V_SINK].gst, "sink" );
+ if ( pad )
{
- if ( MMPLAYER_IS_RTSP_STREAMING(player) )
+ caps_v = gst_pad_get_negotiated_caps( pad );
+ if (caps_v)
{
- __gst_resume ( player, TRUE );
+ p = gst_caps_get_structure (caps_v, 0);
+ gst_structure_get_int (p, "width", &width);
+ mm_attrs_set_int_by_name(attrs, "content_video_width", width);
- /* now we can overcome 'state-lost' situation */
- player->state_lost = FALSE;
+ gst_structure_get_int (p, "height", &height);
+ mm_attrs_set_int_by_name(attrs, "content_video_height", height);
- return;
- }
+ gst_structure_get_fraction (p, "framerate", &tmpNu, &tmpDe);
- switch ( pending_state )
- {
- case MM_PLAYER_STATE_NONE:
- {
- if (current_state != MM_PLAYER_STATE_PLAYING)
- __gst_resume ( player, TRUE );
- }
- break;
+ secure_debug_log("width : %d height : %d", width, height );
- case MM_PLAYER_STATE_PAUSED:
- {
- __gst_resume ( player, TRUE );
- }
- break;
+ gst_caps_unref( caps_v );
+ caps_v = NULL;
- case MM_PLAYER_STATE_PLAYING:
+ if (tmpDe > 0)
{
- debug_log("player is already going to playing state, there is nothing to do.\n");
+ mm_attrs_set_int_by_name(attrs, "content_video_fps", tmpNu / tmpDe);
+ secure_debug_log("fps : %d", tmpNu / tmpDe);
}
- break;
- case MM_PLAYER_STATE_NULL:
- case MM_PLAYER_STATE_READY:
- default :
- {
- debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
- }
- break;
+ has_video_attrs = TRUE;
+ }
+ else
+ {
+ debug_log("no negitiated caps from videosink");
}
+ gst_object_unref( pad );
+ pad = NULL;
}
- break;
-
- case MM_PLAYER_STATE_NULL :
- case MM_PLAYER_STATE_READY :
- case MM_PLAYER_STATE_NONE :
- default:
+ else
{
- debug_warning("invalid target state [%s].\n", MMPLAYER_STATE_GET_NAME(target_state) );
+ debug_log("no videosink sink pad");
}
- break;
}
}
- else
+
+
+ if ( (flag & ATTR_BITRATE) || (!has_bitrate && missing_only) || all )
{
- /* NOTE : in case of rtsp streaming, the src plugin provide the pipeline clock.
- * the src plugin is buffering, the pipeline clock stop automatically.
- * so don't need to pause the player.
- */
- if ( MMPLAYER_IS_RTSP_STREAMING(player) )
- {
- player->state_lost = TRUE;
- return;
- }
+ has_bitrate = FALSE;
- /* NOTE : during the buffering, pause the player for stopping pipeline clock.
- * it's for stopping the pipeline clock to prevent dropping the data in sink element.
- */
- switch ( pending_state )
+ /* FIXIT : please make it clear the dependancy with duration/codec/uritype */
+ if (player->duration)
{
- case MM_PLAYER_STATE_NONE:
+ guint64 data_size = 0;
+
+ if (!MMPLAYER_IS_STREAMING(player) && (player->can_support_codec & FOUND_PLUGIN_VIDEO))
{
- if (current_state != MM_PLAYER_STATE_PAUSED)
- __gst_pause ( player, TRUE );
- }
- break;
+ mm_attrs_get_string_by_name(attrs, "profile_uri", &path);
- case MM_PLAYER_STATE_PLAYING:
+ if (stat(path, &sb) == 0)
+ {
+ data_size = (guint64)sb.st_size;
+ }
+ }
+ else if (MMPLAYER_IS_HTTP_STREAMING(player))
{
- __gst_pause ( player, TRUE );
+ data_size = player->http_content_size;
}
- break;
+ debug_log("try to update bitrate : data_size = %lld", data_size);
- case MM_PLAYER_STATE_PAUSED:
+ if (data_size)
{
- debug_log("player is already going to paused state, there is nothing to do.\n");
+ guint64 bitrate = 0;
+ guint64 msec_dur = 0;
+
+ msec_dur = GST_TIME_AS_MSECONDS(player->duration);
+ bitrate = data_size * 8 * 1000 / msec_dur;
+ secure_debug_log("file size : %u, video bitrate = %llu", data_size, bitrate);
+ mm_attrs_set_int_by_name(attrs, "content_video_bitrate", bitrate);
+
+ has_bitrate = TRUE;
}
- break;
- case MM_PLAYER_STATE_NULL:
- case MM_PLAYER_STATE_READY:
- default :
+ if (MMPLAYER_IS_RTSP_STREAMING(player))
{
- debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
+ if(player->total_bitrate)
+ {
+ mm_attrs_set_int_by_name(attrs, "content_video_bitrate", player->total_bitrate);
+ has_bitrate = TRUE;
+ }
}
- break;
}
}
+
+ /* validate all */
+ if ( mmf_attrs_commit ( attrs ) )
+ {
+ debug_error("failed to update attributes\n");
+ return FALSE;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
}
-static gboolean
-__mmplayer_gst_callback(GstBus *bus, GstMessage *msg, gpointer data) // @
+gint __mmplayer_get_stream_service_type( mm_player_t* player )
{
- mm_player_t* player = (mm_player_t*) data;
- gboolean ret = TRUE;
- static gboolean async_done = FALSE;
+ gint streaming_type = STREAMING_SERVICE_NONE;
- return_val_if_fail ( player, FALSE );
- return_val_if_fail ( msg && GST_IS_MESSAGE(msg), FALSE );
+ MMPLAYER_FENTER();
- switch ( GST_MESSAGE_TYPE( msg ) )
+ return_val_if_fail ( player &&
+ player->pipeline &&
+ player->pipeline->mainbin &&
+ player->pipeline->mainbin[MMPLAYER_M_SRC].gst,
+ FALSE );
+
+ /* streaming service type if streaming */
+ if ( ! MMPLAYER_IS_STREAMING(player) )
+ return STREAMING_SERVICE_NONE;
+
+ if (MMPLAYER_IS_RTSP_STREAMING(player))
{
- case GST_MESSAGE_UNKNOWN:
- debug_warning("unknown message received\n");
+ /* get property from rtspsrc element */
+ g_object_get(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_SRC].gst),
+ "service_type", &streaming_type, NULL);
+ }
+ else if (MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ streaming_type = (player->duration == 0) ?
+ STREAMING_SERVICE_LIVE : STREAMING_SERVICE_VOD;
+ }
+
+ switch ( streaming_type )
+ {
+ case STREAMING_SERVICE_LIVE:
+ debug_log("it's live streaming");
+ break;
+ case STREAMING_SERVICE_VOD:
+ debug_log("it's vod streaming");
break;
+ case STREAMING_SERVICE_NONE:
+ debug_error("should not get here");
+ break;
+ default:
+ debug_error("should not get here");
+ }
- case GST_MESSAGE_EOS:
- {
- MMHandleType attrs = 0;
- gint count = 0;
+ player->streaming_type = streaming_type;
+ MMPLAYER_FLEAVE();
- debug_log("GST_MESSAGE_EOS received\n");
+ return streaming_type;
+}
- /* NOTE : EOS event is comming multiple time. watch out it */
- /* check state. we only process EOS when pipeline state goes to PLAYING */
- if ( ! (player->cmd == MMPLAYER_COMMAND_START || player->cmd == MMPLAYER_COMMAND_RESUME) )
- {
- debug_warning("EOS received on non-playing state. ignoring it\n");
- break;
- }
- if ( (player->audio_stream_cb) && (player->is_sound_extraction) )
- {
- GstPad *pad = NULL;
-
- pad = gst_element_get_static_pad (player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "sink");
+/* this function sets the player state and also report
+ * it to applicaton by calling callback function
+ */
+static int
+__mmplayer_set_state(mm_player_t* player, int state) // @
+{
+ MMMessageParamType msg = {0, };
+ int asm_result = MM_ERROR_NONE;
+ gboolean post_bos = FALSE;
+ gboolean interrupted_by_asm = FALSE;
+ int ret = MM_ERROR_NONE;
- debug_error("release audio callback\n");
-
- /* release audio callback */
- gst_pad_remove_buffer_probe (pad, player->audio_cb_probe_id);
- player->audio_cb_probe_id = 0;
- /* audio callback should be free because it can be called even though probe remove.*/
- player->audio_stream_cb = NULL;
- player->audio_stream_cb_user_param = NULL;
+ return_val_if_fail ( player, FALSE );
- }
+ if ( MMPLAYER_CURRENT_STATE(player) == state )
+ {
+ debug_warning("already same state(%s)\n", MMPLAYER_STATE_GET_NAME(state));
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
+ return ret;
+ }
- /* rewind if repeat count is greater then zero */
- /* get play count */
- attrs = MMPLAYER_GET_ATTRS(player);
+ /* update player states */
+ MMPLAYER_PREV_STATE(player) = MMPLAYER_CURRENT_STATE(player);
+ MMPLAYER_CURRENT_STATE(player) = state;
- if ( attrs )
- {
- gboolean smooth_repeat = FALSE;
+ /* FIXIT : it's better to do like below code
+ if ( MMPLAYER_CURRENT_STATE(player) == MMPLAYER_TARGET_STATE(player) )
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
+ and add more code to handling PENDING_STATE.
+ */
+ if ( MMPLAYER_CURRENT_STATE(player) == MMPLAYER_PENDING_STATE(player) )
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
- mm_attrs_get_int_by_name(attrs, "profile_play_count", &count);
- mm_attrs_get_int_by_name(attrs, "profile_smooth_repeat", &smooth_repeat);
+ /* print state */
+ MMPLAYER_PRINT_STATE(player);
- debug_log("remaining play count: %d, playback rate: %f\n", count, player->playback_rate);
+ /* do some FSM stuffs before posting new state to application */
+ interrupted_by_asm = player->sm.by_asm_cb;
- if ( count > 1 || count == -1 || player->playback_rate < 0.0 ) /* default value is 1 */
+ switch ( MMPLAYER_CURRENT_STATE(player) )
+ {
+ case MM_PLAYER_STATE_NULL:
+ case MM_PLAYER_STATE_READY:
+ {
+ if (player->cmd == MMPLAYER_COMMAND_STOP)
+ {
+ asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_STOP, FALSE);
+ if ( asm_result != MM_ERROR_NONE )
{
- if ( smooth_repeat )
+ debug_error("failed to set asm state to stop\n");
+ return MM_ERROR_POLICY_INTERNAL;
+ }
+ }
+ }
+ break;
+
+ case MM_PLAYER_STATE_PAUSED:
+ {
+ if ( ! player->sent_bos )
+ {
+ int found = 0;
+ #define MMPLAYER_MAX_SOUND_PRIORITY 3
+
+ /* it's first time to update all content attrs. */
+ _mmplayer_update_content_attrs( player, ATTR_ALL );
+ /* set max sound priority to keep own sound and not to mute other's one */
+ mm_attrs_get_int_by_name(player->attrs, "content_video_found", &found);
+ if (found)
+ {
+ mm_attrs_get_int_by_name(player->attrs, "content_audio_found", &found);
+ if (found)
{
- debug_log("smooth repeat enabled. seeking operation will be excuted in new thread\n");
+ debug_log("set max audio priority");
+ g_object_set(player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "priority", MMPLAYER_MAX_SOUND_PRIORITY, NULL);
+ }
+ }
- g_cond_signal( player->repeat_thread_cond );
+ }
- break;
- }
- else
- {
- gint ret_value = 0;
-
- if ( player->section_repeat )
- {
- ret_value = _mmplayer_activate_section_repeat((MMHandleType)player, player->section_repeat_start, player->section_repeat_end);
- }
- else
- {
+ /* add audio callback probe if condition is satisfied */
+ if ( ! player->audio_cb_probe_id && player->set_mode.pcm_extraction )
+ {
+ __mmplayer_configure_audio_callback(player);
+ /* FIXIT : handle return value */
+ }
- if ( player->playback_rate < 0.0 )
- {
- player->resumed_by_rewind = TRUE;
- _mmplayer_set_mute((MMHandleType)player, 0);
- MMPLAYER_POST_MSG( player, MM_MESSAGE_RESUMED_BY_REW, NULL );
- }
+ if (!MMPLAYER_IS_STREAMING(player) || (player->streamer && !player->streamer->is_buffering))
+ {
+ asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_PAUSE, FALSE);
+ if ( asm_result )
+ {
+ debug_error("failed to set asm state to PAUSE\n");
+ return MM_ERROR_POLICY_INTERNAL;
+ }
+ }
+ }
+ break;
- ret_value = __gst_set_position( player, MM_PLAYER_POS_FORMAT_TIME, 0 );
+ case MM_PLAYER_STATE_PLAYING:
+ {
+ /* try to get content metadata */
+ if ( ! player->sent_bos )
+ {
+ /* NOTE : giving ATTR_MISSING_ONLY may have dependency with
+ * c-api since c-api doesn't use _start() anymore. It may not work propery with
+ * legacy mmfw-player api */
+ _mmplayer_update_content_attrs( player, ATTR_MISSING_ONLY);
+ }
- /* initialize */
- player->sent_bos = FALSE;
- }
+ if ( (player->cmd == MMPLAYER_COMMAND_START) || (player->cmd == MMPLAYER_COMMAND_RESUME) )
+ {
+ if (!player->sent_bos)
+ {
+ __mmplayer_handle_missed_plugin ( player );
+ }
- if ( MM_ERROR_NONE != ret_value )
- {
- debug_error("failed to set position to zero for rewind\n");
- }
- else
- {
- if ( count > 1 )
- {
- /* we successeded to rewind. update play count and then wait for next EOS */
- count--;
+ /* update ASM state for video and streaming buffering */
+ asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_PLAYING, TRUE);
+ if (asm_result != MM_ERROR_NONE)
+ {
+ if (player->pipeline->videobin)
+ {
+ MMMessageParamType msg = {0, };
- mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
+ debug_error("failed to go ahead because of video conflict\n");
- if ( mmf_attrs_commit ( attrs ) )
- debug_error("failed to commit attrs\n");
- }
- }
+ msg.union_type = MM_MSG_UNION_CODE;
+ msg.code = MM_ERROR_POLICY_INTERRUPTED;
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_INTERRUPTED, &msg);
- break;
+ _mmplayer_unrealize((MMHandleType)player);
+ }
+ else
+ {
+ debug_error("failed to play by ASM error : 0x%X\n", asm_result);
+ _mmplayer_pause((MMHandleType)player);
+ return asm_result;
}
+
+ return MM_ERROR_POLICY_INTERNAL;
}
}
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-eos" );
-
- /* post eos message to application */
- __mmplayer_post_delayed_eos( player, PLAYER_INI()->eos_delay );
-
- /* reset last position */
- player->last_position = 0;
- }
- break;
-
- case GST_MESSAGE_ERROR:
- {
- GError *error = NULL;
- gchar* debug = NULL;
- gchar *msg_src_element = NULL;
-
- /* generating debug info before returning error */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-error" );
-
- /* get error code */
- gst_message_parse_error( msg, &error, &debug );
+ if ( player->resumed_by_rewind && player->playback_rate < 0.0 )
+ {
+ /* initialize because auto resume is done well. */
+ player->resumed_by_rewind = FALSE;
+ player->playback_rate = 1.0;
+ }
- msg_src_element = GST_ELEMENT_NAME( GST_ELEMENT_CAST( msg->src ) );
- if ( gst_structure_has_name ( msg->structure, "streaming_error" ) )
+ if ( !player->sent_bos )
{
- /* Note : the streaming error from the streaming source is handled
- * using __mmplayer_handle_streaming_error.
+ /* check audio codec field is set or not
+ * we can get it from typefinder or codec's caps.
*/
- __mmplayer_handle_streaming_error ( player, msg );
+ gchar *audio_codec = NULL;
+ mm_attrs_get_string_by_name(player->attrs, "content_audio_codec", &audio_codec);
- /* dump state of all element */
- __mmplayer_dump_pipeline_state( player );
- }
- else
- {
- /* traslate gst error code to msl error code. then post it
- * to application if needed
+ /* The codec format can't be sent for audio only case like amr, mid etc.
+ * Because, parser don't make related TAG.
+ * So, if it's not set yet, fill it with found data.
*/
- __mmplayer_handle_gst_error( player, msg, error );
+ if ( ! audio_codec )
+ {
+ if ( g_strrstr(player->type, "audio/midi"))
+ {
+ audio_codec = g_strdup("MIDI");
- /* dump state of all element */
- __mmplayer_dump_pipeline_state( player );
+ }
+ else if ( g_strrstr(player->type, "audio/x-amr"))
+ {
+ audio_codec = g_strdup("AMR");
+ }
+ else if ( g_strrstr(player->type, "audio/mpeg") && !g_strrstr(player->type, "mpegversion=(int)1"))
+ {
+ audio_codec = g_strdup("AAC");
+ }
+ else
+ {
+ audio_codec = g_strdup("unknown");
+ }
+ mm_attrs_set_string_by_name(player->attrs, "content_audio_codec", audio_codec);
- }
+ MMPLAYER_FREEIF(audio_codec);
+ mmf_attrs_commit(player->attrs);
+ debug_log("set audio codec type with caps\n");
+ }
- if (MMPLAYER_IS_HTTP_PD(player))
- {
- _mmplayer_pd_stop ((MMHandleType)player);
+ post_bos = TRUE;
}
-
- MMPLAYER_FREEIF( debug );
- g_error_free( error );
}
break;
- case GST_MESSAGE_WARNING:
- {
- char* debug = NULL;
- GError* error = NULL;
-
- gst_message_parse_warning(msg, &error, &debug);
-
- debug_warning("warning : %s\n", error->message);
- debug_warning("debug : %s\n", debug);
+ case MM_PLAYER_STATE_NONE:
+ default:
+ debug_warning("invalid target state, there is nothing to do.\n");
+ break;
+ }
- MMPLAYER_POST_MSG( player, MM_MESSAGE_WARNING, NULL );
- MMPLAYER_FREEIF( debug );
- g_error_free( error );
- }
- break;
+ /* post message to application */
+ if (MMPLAYER_TARGET_STATE(player) == state)
+ {
+ /* fill the message with state of player */
+ msg.state.previous = MMPLAYER_PREV_STATE(player);
+ msg.state.current = MMPLAYER_CURRENT_STATE(player);
- case GST_MESSAGE_INFO: debug_log("GST_MESSAGE_STATE_DIRTY\n"); break;
+ debug_log ("player reach the target state (%s)", MMPLAYER_STATE_GET_NAME(MMPLAYER_TARGET_STATE(player)));
- case GST_MESSAGE_TAG:
+ /* state changed by asm callback */
+ if ( interrupted_by_asm )
{
- debug_log("GST_MESSAGE_TAG\n");
- if ( ! __mmplayer_gst_extract_tag_from_msg( player, msg ) )
- {
- debug_warning("failed to extract tags from gstmessage\n");
- }
+ msg.union_type = MM_MSG_UNION_CODE;
+ msg.code = player->sm.event_src;
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_INTERRUPTED, &msg );
}
- break;
-
- case GST_MESSAGE_BUFFERING:
+ /* state changed by usecase */
+ else
{
- MMMessageParamType msg_param = {0, };
- gboolean update_buffering_percent = TRUE;
-
- if ( !MMPLAYER_IS_STREAMING(player) || (player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS) ) // pure hlsdemux case, don't consider buffering of msl currently
- break;
-
- __mm_player_streaming_buffering (player->streamer, msg);
-
- __mmplayer_handle_buffering_message ( player );
-
- update_buffering_percent = player->pipeline_is_constructed || MMPLAYER_IS_RTSP_STREAMING(player) || MMPLAYER_IS_HTTP_LIVE_STREAMING(player);
- if (update_buffering_percent)
- {
- msg_param.connection.buffering = player->streamer->buffering_percent;
- MMPLAYER_POST_MSG ( player, MM_MESSAGE_BUFFERING, &msg_param );
- }
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_CHANGED, &msg );
}
- break;
+ }
+ else
+ {
+ debug_log ("intermediate state, do nothing.\n");
+ MMPLAYER_PRINT_STATE(player);
+ return ret;
+ }
- case GST_MESSAGE_STATE_CHANGED:
- {
- MMPlayerGstElement *mainbin;
- const GValue *voldstate, *vnewstate, *vpending;
- GstState oldstate, newstate, pending;
+ if ( post_bos )
+ {
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_BEGIN_OF_STREAM, NULL );
+ player->sent_bos = TRUE;
+ }
- if ( ! ( player->pipeline && player->pipeline->mainbin ) )
- {
- debug_error("player pipeline handle is null");
+ return ret;
+}
- break;
- }
+gboolean
+__mmplayer_post_message(mm_player_t* player, enum MMMessageType msgtype, MMMessageParamType* param) // @
+{
+ return_val_if_fail( player, FALSE );
- mainbin = player->pipeline->mainbin;
+ if ( !player->msg_cb )
+ {
+ return FALSE;
+ }
- /* get state info from msg */
- voldstate = gst_structure_get_value (msg->structure, "old-state");
- vnewstate = gst_structure_get_value (msg->structure, "new-state");
- vpending = gst_structure_get_value (msg->structure, "pending-state");
+ //debug_log("Message (type : %d) will be posted using msg-cb(%p). \n", msgtype, player->msg_cb);
- oldstate = (GstState)voldstate->data[0].v_int;
- newstate = (GstState)vnewstate->data[0].v_int;
- pending = (GstState)vpending->data[0].v_int;
+ player->msg_cb(msgtype, param, player->msg_cb_param);
- if (oldstate == newstate)
- break;
+ return TRUE;
+}
- debug_log("state changed [%s] : %s ---> %s final : %s\n",
- GST_OBJECT_NAME(GST_MESSAGE_SRC(msg)),
- gst_element_state_get_name( (GstState)oldstate ),
- gst_element_state_get_name( (GstState)newstate ),
- gst_element_state_get_name( (GstState)pending ) );
+static gpointer __mmplayer_next_play_thread(gpointer data)
+{
+ mm_player_t* player = (mm_player_t*) data;
+ MMPlayerGstElement *mainbin = NULL;
- /* we only handle messages from pipeline */
- if( msg->src != (GstObject *)mainbin[MMPLAYER_M_PIPE].gst )
- break;
+ return_val_if_fail ( player, NULL );
- switch(newstate)
- {
- case GST_STATE_VOID_PENDING:
- break;
+ while ( ! player->next_play_thread_exit )
+ {
+ debug_log("next play thread started. waiting for signal.\n");
+ g_cond_wait( player->next_play_thread_cond, player->next_play_thread_mutex );
- case GST_STATE_NULL:
- break;
+ debug_log("re building pipeline for next play.\n");
- case GST_STATE_READY:
- break;
+ if ( player->next_play_thread_exit )
+ {
+ MMPLAYER_PLAYBACK_UNLOCK(player);
- case GST_STATE_PAUSED:
- {
- if ( ! player->audio_cb_probe_id && player->is_sound_extraction )
- __mmplayer_configure_audio_callback(player);
-
+ debug_log("exiting next play thread\n");
+ break;
+ }
- if ( MMPLAYER_IS_STREAMING(player) )
- {
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PAUSED );
+ mainbin = player->pipeline->mainbin;
- __mm_player_streaming_set_content_bitrate(player->streamer, player->total_maximum_bitrate, player->total_bitrate);
+ MMPLAYER_RELEASE_ELEMENT(player, mainbin, MMPLAYER_M_MUXED_S_BUFFER);
+ MMPLAYER_RELEASE_ELEMENT(player, mainbin, MMPLAYER_M_ID3DEMUX);
+ MMPLAYER_RELEASE_ELEMENT(player, mainbin, MMPLAYER_M_AUTOPLUG);
+ MMPLAYER_RELEASE_ELEMENT(player, mainbin, MMPLAYER_M_TYPEFIND);
+ MMPLAYER_RELEASE_ELEMENT(player, mainbin, MMPLAYER_M_SRC);
- /* check pending seek and do now */
- if ( player->pending_seek.is_pending )
- __gst_pending_seek ( player );
- }
+ __mmplayer_activate_next_source(player, GST_STATE_PLAYING);
+ }
- player->need_update_content_dur = TRUE;
- }
- break;
+ return NULL;
+}
- case GST_STATE_PLAYING:
- {
+static gpointer __mmplayer_repeat_thread(gpointer data)
+{
+ mm_player_t* player = (mm_player_t*) data;
+ gboolean ret_value = FALSE;
+ MMHandleType attrs = 0;
+ gint count = 0;
- gboolean is_async = FALSE;
+ return_val_if_fail ( player, NULL );
- if (player->doing_seek && async_done)
- {
- player->doing_seek = FALSE;
- async_done = FALSE;
- MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
- }
+ while ( ! player->repeat_thread_exit )
+ {
+ debug_log("repeat thread started. waiting for signal.\n");
+ g_cond_wait( player->repeat_thread_cond, player->repeat_thread_mutex );
- /* It should be called in case of async start only.
- * don't need to call in the case of resume and sync start (currently, resume is sync.)
- */
- mm_attrs_get_int_by_name(player->attrs,"profile_async_start", &is_async);
- if (is_async && player->cmd == MMPLAYER_COMMAND_START)
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PLAYING );
- }
- break;
+ if ( player->repeat_thread_exit )
+ {
+ debug_log("exiting repeat thread\n");
+ break;
+ }
- default:
- break;
- }
+ if ( !player->cmd_lock )
+ {
+ debug_log("can't get cmd lock\n");
+ return NULL;
}
- break;
- case GST_MESSAGE_STATE_DIRTY: debug_log("GST_MESSAGE_STATE_DIRTY\n"); break;
- case GST_MESSAGE_STEP_DONE: debug_log("GST_MESSAGE_STEP_DONE\n"); break;
- case GST_MESSAGE_CLOCK_PROVIDE: debug_log("GST_MESSAGE_CLOCK_PROVIDE\n"); break;
-
- case GST_MESSAGE_CLOCK_LOST:
- {
- GstClock *clock = NULL;
- gst_message_parse_clock_lost (msg, &clock);
- debug_log("GST_MESSAGE_CLOCK_LOST : %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
- g_print ("GST_MESSAGE_CLOCK_LOST : %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
+ /* lock */
+ g_mutex_lock(player->cmd_lock);
- if (PLAYER_INI()->provide_clock)
- {
- debug_log ("Provide clock is TRUE, do pause->resume\n");
- __gst_pause(player, FALSE);
- __gst_resume(player, FALSE);
- }
- }
+ attrs = MMPLAYER_GET_ATTRS(player);
+
+ if (mm_attrs_get_int_by_name(attrs, "profile_play_count", &count) != MM_ERROR_NONE)
+ {
+ debug_error("can not get play count\n");
break;
-
- case GST_MESSAGE_NEW_CLOCK:
+ }
+
+ if ( player->section_repeat )
+ {
+ ret_value = _mmplayer_activate_section_repeat((MMHandleType)player, player->section_repeat_start, player->section_repeat_end);
+ }
+ else
+ {
+ if ( player->playback_rate < 0.0 )
{
- GstClock *clock = NULL;
- gst_message_parse_new_clock (msg, &clock);
- debug_log("GST_MESSAGE_NEW_CLOCK : %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
+ player->resumed_by_rewind = TRUE;
+ _mmplayer_set_mute((MMHandleType)player, 0);
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_RESUMED_BY_REW, NULL );
}
- break;
- case GST_MESSAGE_STRUCTURE_CHANGE: debug_log("GST_MESSAGE_STRUCTURE_CHANGE\n"); break;
- case GST_MESSAGE_STREAM_STATUS: debug_log("GST_MESSAGE_STREAM_STATUS\n"); break;
- case GST_MESSAGE_APPLICATION: debug_log("GST_MESSAGE_APPLICATION\n"); break;
+ ret_value = __gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, player->playback_rate,
+ GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET,
+ 0, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);
- case GST_MESSAGE_ELEMENT:
- {
- debug_log("GST_MESSAGE_ELEMENT\n");
+ /* initialize */
+ player->sent_bos = FALSE;
}
- break;
-
- case GST_MESSAGE_SEGMENT_START: debug_log("GST_MESSAGE_SEGMENT_START\n"); break;
- case GST_MESSAGE_SEGMENT_DONE: debug_log("GST_MESSAGE_SEGMENT_DONE\n"); break;
-
- case GST_MESSAGE_DURATION:
+
+ if ( ! ret_value )
{
- debug_log("GST_MESSAGE_DURATION\n");
+ debug_error("failed to set position to zero for rewind\n");
+ continue;
}
- break;
-
- case GST_MESSAGE_LATENCY: debug_log("GST_MESSAGE_LATENCY\n"); break;
- case GST_MESSAGE_ASYNC_START: debug_log("GST_MESSAGE_ASYNC_DONE : %s\n", gst_element_get_name(GST_MESSAGE_SRC(msg))); break;
-
- case GST_MESSAGE_ASYNC_DONE:
+
+ /* decrease play count */
+ if ( count > 1 )
{
- debug_log("GST_MESSAGE_ASYNC_DONE : %s\n", gst_element_get_name(GST_MESSAGE_SRC(msg)));
+ /* we successeded to rewind. update play count and then wait for next EOS */
+ count--;
- if (player->doing_seek)
+ mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
+
+ /* commit attribute */
+ if ( mmf_attrs_commit ( attrs ) )
{
- if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PAUSED)
- {
- player->doing_seek = FALSE;
- MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
- }
- else if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PLAYING)
- {
- async_done = TRUE;
- }
+ debug_error("failed to commit attribute\n");
}
}
- break;
-
- case GST_MESSAGE_REQUEST_STATE: debug_log("GST_MESSAGE_REQUEST_STATE\n"); break;
- case GST_MESSAGE_STEP_START: debug_log("GST_MESSAGE_STEP_START\n"); break;
- case GST_MESSAGE_QOS: debug_log("GST_MESSAGE_QOS\n"); break;
- case GST_MESSAGE_PROGRESS: debug_log("GST_MESSAGE_PROGRESS\n"); break;
- case GST_MESSAGE_ANY: debug_log("GST_MESSAGE_ANY\n"); break;
- default:
- debug_warning("unhandled message\n");
- break;
+ /* unlock */
+ g_mutex_unlock(player->cmd_lock);
}
- /* FIXIT : this cause so many warnings/errors from glib/gstreamer. we should not call it since
- * gst_element_post_message api takes ownership of the message.
- */
- //gst_message_unref( msg );
-
- return ret;
+ return NULL;
}
-static gboolean
-__mmplayer_gst_extract_tag_from_msg(mm_player_t* player, GstMessage* msg) // @
+static void
+__mmplayer_update_buffer_setting(mm_player_t *player, GstMessage *buffering_msg)
{
+ MMHandleType attrs = 0;
+ guint64 data_size = 0;
+ gchar* path = NULL;
+ unsigned long pos_msec = 0;
+ struct stat sb;
-/* macro for better code readability */
-#define MMPLAYER_UPDATE_TAG_STRING(gsttag, attribute, playertag) \
-if (gst_tag_list_get_string(tag_list, gsttag, &string)) \
-{\
- if (string != NULL)\
- {\
- debug_log ( "update tag string : %s\n", string); \
- mm_attrs_set_string_by_name(attribute, playertag, string); \
- g_free(string);\
- string = NULL;\
- }\
-}
-
-#define MMPLAYER_UPDATE_TAG_IMAGE(gsttag, attribute, playertag) \
-value = gst_tag_list_get_value_index(tag_list, gsttag, index); \
-if (value) \
-{\
- buffer = gst_value_get_buffer (value); \
- debug_log ( "update album cover data : %p, size : %d\n", GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer)); \
- player->album_art = (gchar *)g_malloc(GST_BUFFER_SIZE(buffer)); \
- if (player->album_art); \
- { \
- memcpy(player->album_art, GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer)); \
- mm_attrs_set_data_by_name(attribute, playertag, (void *)player->album_art, GST_BUFFER_SIZE(buffer)); \
- } \
-}
+ return_if_fail( player && player->pipeline && player->pipeline->mainbin);
-#define MMPLAYER_UPDATE_TAG_UINT(gsttag, attribute, playertag) \
-if (gst_tag_list_get_uint(tag_list, gsttag, &v_uint))\
-{\
- if(v_uint)\
- {\
- if(gsttag==GST_TAG_BITRATE)\
- {\
- if (player->updated_bitrate_count == 0) \
- mm_attrs_set_int_by_name(attribute, "content_audio_bitrate", v_uint); \
- if (player->updated_bitrate_count<MM_PLAYER_STREAM_COUNT_MAX) \
- {\
- player->bitrate[player->updated_bitrate_count] = v_uint;\
- player->total_bitrate += player->bitrate[player->updated_maximum_bitrate_count]; \
- player->updated_bitrate_count++; \
- mm_attrs_set_int_by_name(attribute, playertag, player->total_bitrate);\
- debug_log ( "update bitrate %d[bps] of stream #%d.\n", v_uint, player->updated_bitrate_count);\
- }\
- }\
- else if (gsttag==GST_TAG_MAXIMUM_BITRATE)\
- {\
- if (player->updated_maximum_bitrate_count<MM_PLAYER_STREAM_COUNT_MAX) \
- {\
- player->maximum_bitrate[player->updated_maximum_bitrate_count] = v_uint;\
- player->total_maximum_bitrate += player->maximum_bitrate[player->updated_maximum_bitrate_count]; \
- player->updated_maximum_bitrate_count++; \
- mm_attrs_set_int_by_name(attribute, playertag, player->total_maximum_bitrate); \
- debug_log ( "update maximum bitrate %d[bps] of stream #%d\n", v_uint, player->updated_maximum_bitrate_count);\
- }\
- }\
- else\
- {\
- mm_attrs_set_int_by_name(attribute, playertag, v_uint); \
- }\
- v_uint = 0;\
- }\
-}
+ __gst_get_position(player, MM_PLAYER_POS_FORMAT_TIME, &pos_msec); // update last_position
-#define MMPLAYER_UPDATE_TAG_DATE(gsttag, attribute, playertag) \
-if (gst_tag_list_get_date(tag_list, gsttag, &date))\
-{\
- if (date != NULL)\
- {\
- string = g_strdup_printf("%d", g_date_get_year(date));\
- mm_attrs_set_string_by_name(attribute, playertag, string);\
- debug_log ( "metainfo year : %s\n", string);\
- MMPLAYER_FREEIF(string);\
- g_date_free(date);\
- }\
-}
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("fail to get attributes.\n");
+ return;
+ }
-#define MMPLAYER_UPDATE_TAG_UINT64(gsttag, attribute, playertag) \
-if(gst_tag_list_get_uint64(tag_list, gsttag, &v_uint64))\
-{\
- if(v_uint64)\
- {\
- /* FIXIT : don't know how to store date */\
- g_assert(1);\
- v_uint64 = 0;\
- }\
-}
+ if (!MMPLAYER_IS_STREAMING(player) && (player->can_support_codec & FOUND_PLUGIN_VIDEO))
+ {
+ mm_attrs_get_string_by_name(attrs, "profile_uri", &path);
-#define MMPLAYER_UPDATE_TAG_DOUBLE(gsttag, attribute, playertag) \
-if(gst_tag_list_get_double(tag_list, gsttag, &v_double))\
-{\
- if(v_double)\
- {\
- /* FIXIT : don't know how to store date */\
- g_assert(1);\
- v_double = 0;\
- }\
-}
+ if (stat(path, &sb) == 0)
+ {
+ data_size = (guint64)sb.st_size;
+ }
+ }
+ else if (MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ data_size = player->http_content_size;
+ }
- /* function start */
- GstTagList* tag_list = NULL;
+ __mm_player_streaming_buffering( player->streamer,
+ buffering_msg,
+ data_size,
+ player->last_position,
+ player->duration);
- MMHandleType attrs = 0;
+ __mm_player_streaming_sync_property(player->streamer, player->pipeline->mainbin[MMPLAYER_M_AUTOPLUG].gst);
- char *string = NULL;
- guint v_uint = 0;
- GDate *date = NULL;
- /* album cover */
- GstBuffer *buffer = NULL;
- gint index = 0;
- const GValue *value;
+ return;
+}
- /* currently not used. but those are needed for above macro */
- //guint64 v_uint64 = 0;
- //gdouble v_double = 0;
- return_val_if_fail( player && msg, FALSE );
+static void
+__mmplayer_handle_buffering_message ( mm_player_t* player )
+{
+ MMPlayerStateType prev_state = MM_PLAYER_STATE_NONE;
+ MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
+ MMPlayerStateType target_state = MM_PLAYER_STATE_NONE;
+ MMPlayerStateType pending_state = MM_PLAYER_STATE_NONE;
- attrs = MMPLAYER_GET_ATTRS(player);
+ return_if_fail ( player );
- return_val_if_fail( attrs, FALSE );
+ prev_state = MMPLAYER_PREV_STATE(player);
+ current_state = MMPLAYER_CURRENT_STATE(player);
+ target_state = MMPLAYER_TARGET_STATE(player);
+ pending_state = MMPLAYER_PENDING_STATE(player);
- /* get tag list from gst message */
- gst_message_parse_tag(msg, &tag_list);
+ if (MMPLAYER_IS_LIVE_STREAMING(player))
+ return;
- /* store tags to player attributes */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_TITLE, attrs, "tag_title");
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_TITLE_SORTNAME, ?, ?); */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ARTIST, attrs, "tag_artist");
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ARTIST_SORTNAME, ?, ?); */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ALBUM, attrs, "tag_album");
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ALBUM_SORTNAME, ?, ?); */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COMPOSER, attrs, "tag_author");
- MMPLAYER_UPDATE_TAG_DATE(GST_TAG_DATE, attrs, "tag_date");
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_GENRE, attrs, "tag_genre");
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COMMENT, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_EXTENDED_COMMENT, ?, ?); */
- MMPLAYER_UPDATE_TAG_UINT(GST_TAG_TRACK_NUMBER, attrs, "tag_track_num");
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_TRACK_COUNT, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_ALBUM_VOLUME_NUMBER, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_ALBUM_VOLUME_COUNT, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LOCATION, ?, ?); */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_DESCRIPTION, attrs, "tag_description");
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_VERSION, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ISRC, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ORGANIZATION, ?, ?); */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COPYRIGHT, attrs, "tag_copyright");
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COPYRIGHT_URI, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_CONTACT, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LICENSE, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LICENSE_URI, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_PERFORMER, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_UINT64(GST_TAG_DURATION, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_CODEC, ?, ?); */
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_VIDEO_CODEC, attrs, "content_video_codec");
- MMPLAYER_UPDATE_TAG_STRING(GST_TAG_AUDIO_CODEC, attrs, "content_audio_codec");
- MMPLAYER_UPDATE_TAG_UINT(GST_TAG_BITRATE, attrs, "content_bitrate");
- MMPLAYER_UPDATE_TAG_UINT(GST_TAG_MAXIMUM_BITRATE, attrs, "content_max_bitrate");
- MMPLAYER_UPDATE_TAG_IMAGE(GST_TAG_IMAGE, attrs, "tag_album_cover");
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_NOMINAL_BITRATE, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_MINIMUM_BITRATE, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_SERIAL, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ENCODER, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_ENCODER_VERSION, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_TRACK_GAIN, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_TRACK_PEAK, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_ALBUM_GAIN, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_ALBUM_PEAK, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_REFERENCE_LEVEL, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LANGUAGE_CODE, ?, ?); */
- /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_BEATS_PER_MINUTE, ?, ?); */
+ if ( !player->streamer->is_buffering )
+ {
+ debug_log( "player state : prev %s, current %s, pending %s, target %s \n",
+ MMPLAYER_STATE_GET_NAME(prev_state),
+ MMPLAYER_STATE_GET_NAME(current_state),
+ MMPLAYER_STATE_GET_NAME(pending_state),
+ MMPLAYER_STATE_GET_NAME(target_state));
- if ( mmf_attrs_commit ( attrs ) )
- debug_error("failed to commit.\n");
+ /* NOTE : if buffering has done, player has to go to target state. */
+ switch ( target_state )
+ {
+ case MM_PLAYER_STATE_PAUSED :
+ {
+ switch ( pending_state )
+ {
+ case MM_PLAYER_STATE_PLAYING:
+ {
+ __gst_pause ( player, TRUE );
+ }
+ break;
- gst_tag_list_free(tag_list);
+ case MM_PLAYER_STATE_PAUSED:
+ {
+ debug_log("player is already going to paused state, there is nothing to do.\n");
+ }
+ break;
- return TRUE;
-}
+ case MM_PLAYER_STATE_NONE:
+ case MM_PLAYER_STATE_NULL:
+ case MM_PLAYER_STATE_READY:
+ default :
+ {
+ debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
+ }
+ break;
+ }
+ }
+ break;
-static void
-__mmplayer_gst_rtp_no_more_pads (GstElement *element, gpointer data) // @
-{
- mm_player_t* player = (mm_player_t*) data;
+ case MM_PLAYER_STATE_PLAYING :
+ {
+ switch ( pending_state )
+ {
+ case MM_PLAYER_STATE_NONE:
+ {
+ if (current_state != MM_PLAYER_STATE_PLAYING)
+ __gst_resume ( player, TRUE );
+ }
+ break;
- debug_fenter();
+ case MM_PLAYER_STATE_PAUSED:
+ {
+ /* NOTE: It should be worked as asynchronously.
+ * Because, buffering can be completed during autoplugging when pipeline would try to go playing state directly.
+ */
+ __gst_resume ( player, TRUE );
+ }
+ break;
- /* NOTE : we can remove fakesink here if there's no rtp_dynamic_pad. because whenever
- * we connect autoplugging element to the pad which is just added to rtspsrc, we increase
- * num_dynamic_pad. and this is no-more-pad situation which means mo more pad will be added.
- * So we can say this. if num_dynamic_pad is zero, it must be one of followings
+ case MM_PLAYER_STATE_PLAYING:
+ {
+ debug_log("player is already going to playing state, there is nothing to do.\n");
+ }
+ break;
- * [1] audio and video will be dumped with filesink.
- * [2] autoplugging is done by just using pad caps.
- * [3] typefinding has happend in audio but audiosink is created already before no-more-pad signal
- * and the video will be dumped via filesink.
- */
- if ( player->num_dynamic_pad == 0 )
- {
- debug_log("it seems pad caps is directely used for autoplugging. removing fakesink now\n");
+ case MM_PLAYER_STATE_NULL:
+ case MM_PLAYER_STATE_READY:
+ default :
+ {
+ debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
+ }
+ break;
+ }
+ }
+ break;
- if ( ! __mmplayer_gst_remove_fakesink( player,
- &player->pipeline->mainbin[MMPLAYER_M_SRC_FAKESINK]) );
+ case MM_PLAYER_STATE_NULL :
+ case MM_PLAYER_STATE_READY :
+ case MM_PLAYER_STATE_NONE :
+ default:
+ {
+ debug_warning("invalid target state [%s].\n", MMPLAYER_STATE_GET_NAME(target_state) );
+ }
+ break;
+ }
}
+ else
+ {
+ /* NOTE : during the buffering, pause the player for stopping pipeline clock.
+ * it's for stopping the pipeline clock to prevent dropping the data in sink element.
+ */
+ switch ( pending_state )
+ {
+ case MM_PLAYER_STATE_NONE:
+ {
+ if (current_state != MM_PLAYER_STATE_PAUSED)
+ {
+ debug_log("set pause state during buffering\n");
+ __gst_pause ( player, TRUE );
- /* create dot before error-return. for debugging */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-no-more-pad" );
+ // to cover the weak-signal environment.
+ if (MMPLAYER_IS_RTSP_STREAMING(player))
+ {
+ unsigned long position = 0;
+ gint64 pos_msec = 0;
- /* NOTE : if rtspsrc goes to PLAYING before adding it's src pads, a/v sink elements will
- * not goes to PLAYING. they will just remain in PAUSED state. simply we are giving
- * PLAYING state again.
- */
- __mmplayer_gst_set_state(player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PLAYING, TRUE, 5000 );
+ debug_log("[RTSP] seek to the buffering start point\n");
- player->no_more_pad = TRUE;
+ if (__gst_get_position(player, MM_PLAYER_POS_FORMAT_TIME, &position ))
+ {
+ debug_error("failed to get position\n");
+ break;
+ }
- debug_fleave();
-}
+ /* key unit seek */
+ pos_msec = position * G_GINT64_CONSTANT(1000000);
-static gboolean
-__mmplayer_gst_remove_fakesink(mm_player_t* player, MMPlayerGstElement* fakesink) // @
-{
- GstElement* parent = NULL;
+ __gst_seek(player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, 1.0,
+ GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET,
+ pos_msec, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);
+ }
+ }
+ }
+ break;
- return_val_if_fail(player && player->pipeline && fakesink, FALSE);
+ case MM_PLAYER_STATE_PLAYING:
+ {
+ __gst_pause ( player, TRUE );
+ }
+ break;
- /* lock */
- g_mutex_lock( player->fsink_lock );
+ case MM_PLAYER_STATE_PAUSED:
+ {
+ }
+ break;
- if ( ! fakesink->gst )
- {
- goto ERROR;
+ case MM_PLAYER_STATE_NULL:
+ case MM_PLAYER_STATE_READY:
+ default :
+ {
+ debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
+ }
+ break;
+ }
}
+}
- /* get parent of fakesink */
- parent = (GstElement*)gst_object_get_parent( (GstObject*)fakesink->gst );
- if ( ! parent )
- {
- debug_log("fakesink already removed\n");
- goto ERROR;
- }
+static void
+__mmplayer_drop_subtitle(mm_player_t* player, gboolean is_drop)
+{
+ MMPlayerGstElement *textbin;
+ MMPLAYER_FENTER();
- gst_element_set_locked_state( fakesink->gst, TRUE );
+ return_if_fail ( player &&
+ player->pipeline &&
+ player->pipeline->textbin);
- /* setting the state to NULL never returns async
- * so no need to wait for completion of state transiton
- */
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state (fakesink->gst, GST_STATE_NULL) )
- {
- debug_error("fakesink state change failure!\n");
+ return_if_fail ( player->pipeline->textbin[MMPLAYER_T_VIDEO_SINK].gst &&
+ player->pipeline->textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst &&
+ player->pipeline->textbin[MMPLAYER_T_TEXT_IDENTITY].gst);
- /* FIXIT : should I return here? or try to proceed to next? */
- /* return FALSE; */
- }
+ textbin = player->pipeline->textbin;
- /* remove fakesink from it's parent */
- if ( ! gst_bin_remove( GST_BIN( parent ), fakesink->gst ) )
+ if (is_drop)
{
- debug_error("failed to remove fakesink\n");
+ debug_log("Drop subtitle text after getting EOS\n");
- gst_object_unref( parent );
+ g_object_set(textbin[MMPLAYER_T_FAKE_SINK].gst, "async", FALSE, NULL);
+ g_object_set(textbin[MMPLAYER_T_VIDEO_SINK].gst, "async", FALSE, NULL);
+ g_object_set(textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst, "drop-probability", (gfloat)1.0, NULL);
+ g_object_set(textbin[MMPLAYER_T_TEXT_IDENTITY].gst, "drop-probability", (gfloat)1.0, NULL);
- goto ERROR;
+ player->is_subtitle_force_drop = TRUE;
}
+ else
+ {
+ if (player->is_subtitle_force_drop == TRUE)
+ {
+ debug_log("Enable subtitle data path without drop\n");
- gst_object_unref( parent );
-
- /* FIXIT : releasing fakesink takes too much time (around 700ms)
- * we need to figure out the reason why. just for now, fakesink will be released
- * in __mmplayer_gst_destroy_pipeline()
- */
- // gst_object_unref ( fakesink->gst );
- // fakesink->gst = NULL;
+ // player->display_stat = util_get_is_connected_external_display();
- debug_log("state-holder removed\n");
+ /* if connected with external display */
+ if ((player->display_stat == MMPLAYER_DISPLAY_STATUS_HDMI_ACTIVE) ||
+ (player->display_stat == MMPLAYER_DISPLAY_STATUS_UNKNOWN_ACTIVE))
+ {
+ g_object_set(textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst, "drop-probability", (gfloat)1.0, NULL);
+ g_object_set(textbin[MMPLAYER_T_TEXT_IDENTITY].gst, "drop-probability", (gfloat)0.0, NULL);
+ g_object_set(textbin[MMPLAYER_T_VIDEO_SINK].gst, "async", TRUE, NULL);
+ g_object_set(textbin[MMPLAYER_T_FAKE_SINK].gst, "async", FALSE, NULL);
- gst_element_set_locked_state( fakesink->gst, FALSE );
+ debug_log ("connected with external display");
+ }
+ else
+ {
+ g_object_set(textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst, "drop-probability", (gfloat)0.0, NULL);
+ g_object_set(textbin[MMPLAYER_T_TEXT_IDENTITY].gst, "drop-probability", (gfloat)1.0, NULL);
+ g_object_set(textbin[MMPLAYER_T_VIDEO_SINK].gst, "async", FALSE, NULL);
+ g_object_set(textbin[MMPLAYER_T_FAKE_SINK].gst, "async", TRUE, NULL);
- g_mutex_unlock( player->fsink_lock );
- return TRUE;
+ debug_log ("non-connected with external display");
+ }
-ERROR:
- if ( fakesink->gst )
- {
- gst_element_set_locked_state( fakesink->gst, FALSE );
+ player->is_subtitle_force_drop = FALSE;
+ }
}
-
- g_mutex_unlock( player->fsink_lock );
- return FALSE;
}
-
-static void
-__mmplayer_gst_rtp_dynamic_pad (GstElement *element, GstPad *pad, gpointer data) // @
+static gboolean
+__mmplayer_gst_callback(GstBus *bus, GstMessage *msg, gpointer data) // @
{
- GstPad *sinkpad = NULL;
- GstCaps* caps = NULL;
- GstElement* new_element = NULL;
- enum MainElementID element_id = MMPLAYER_M_NUM;
-
mm_player_t* player = (mm_player_t*) data;
+ gboolean ret = TRUE;
+ static gboolean async_done = FALSE;
- debug_fenter();
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( msg && GST_IS_MESSAGE(msg), FALSE );
- return_if_fail( element && pad );
- return_if_fail( player &&
- player->pipeline &&
- player->pipeline->mainbin );
-
-
- /* first, check pt type which we are supporting or not */
- if ( MMPLAYER_PT_IS_VIDEO( GST_PAD_NAME( pad ) ) )
- {
- /* video packet */
- debug_log("is video stream. take it.\n");
- element_id = MMPLAYER_M_S_VDEC;
- }
- else if ( MMPLAYER_PT_IS_AUDIO( GST_PAD_NAME( pad ) ) )
- {
- /* audio packet */
- debug_log("is audio stream. take it.\n");
- element_id = MMPLAYER_M_S_ADEC;
- }
- else
+ switch ( GST_MESSAGE_TYPE( msg ) )
{
- /* FIXIT : application should know this. need one more message type */
- debug_error ("This Payload type is unknown!\n");
- goto ERROR;
- }
-
- /* payload type is recognizable. increase num_dynamic and wait for sinkbin creation.
- * num_dynamic_pad will decreased after creating a sinkbin.
- */
- player->num_dynamic_pad++;
- debug_log("stream count inc : %d\n", player->num_dynamic_pad);
+ case GST_MESSAGE_UNKNOWN:
+ debug_log("unknown message received\n");
+ break;
- /* perform autoplugging if dump is disabled */
- if ( ! new_element )
- {
- if ( PLAYER_INI()->rtsp_do_typefinding )
+ case GST_MESSAGE_EOS:
{
- /* create autoplugging element */
- if( PLAYER_INI()->use_decodebin )
- {
- /* create decodebin */
- new_element = gst_element_factory_make("decodebin", NULL);
+ MMHandleType attrs = 0;
+ gint count = 0;
- g_object_set(G_OBJECT(new_element), "async-handling", TRUE, NULL);
+ debug_log("GST_MESSAGE_EOS received\n");
- if ( ! new_element )
+ /* NOTE : EOS event is comming multiple time. watch out it */
+ /* check state. we only process EOS when pipeline state goes to PLAYING */
+ if ( ! (player->cmd == MMPLAYER_COMMAND_START || player->cmd == MMPLAYER_COMMAND_RESUME) )
{
- debug_error("failed to create autoplug element\n");
- goto ERROR;
+ debug_log("EOS received on non-playing state. ignoring it\n");
+ break;
}
- /* set signal handler */
- MMPLAYER_SIGNAL_CONNECT( player,
- G_OBJECT(new_element),
- "new-decoded-pad",
- G_CALLBACK(__mmplayer_gst_decode_callback),
- (gpointer)player );
- }
- else
+ __mmplayer_drop_subtitle(player, TRUE);
+
+ if ( (player->audio_stream_cb) && (player->set_mode.pcm_extraction) )
{
- /* create typefind */
- new_element = gst_element_factory_make( "typefind", NULL );
- if ( ! new_element )
- {
- debug_error("failed to create typefind\n");
- goto ERROR;
- }
+ GstPad *pad = NULL;
- MMPLAYER_SIGNAL_CONNECT( player,
- G_OBJECT(new_element),
- "have-type",
- G_CALLBACK(__mmplayer_typefind_have_type),
- (gpointer)player);
+ pad = gst_element_get_static_pad (player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "sink");
- /* FIXIT : try to remove it */
- player->have_dynamic_pad = FALSE;
- }
- }
- else /* NOTE : use pad's caps directely. if enabled. what I am assuming is there's no elemnt has dynamic pad */
- {
- debug_log("using pad caps to autopluging instead of doing typefind\n");
+ debug_log("release audio callback\n");
- caps = gst_pad_get_caps( pad );
+ /* release audio callback */
+ gst_pad_remove_buffer_probe (pad, player->audio_cb_probe_id);
+ player->audio_cb_probe_id = 0;
+ /* audio callback should be free because it can be called even though probe remove.*/
+ player->audio_stream_cb = NULL;
+ player->audio_stream_cb_user_param = NULL;
- MMPLAYER_CHECK_NULL( caps );
+ }
- /* clear previous result*/
- player->have_dynamic_pad = FALSE;
+ /* rewind if repeat count is greater then zero */
+ /* get play count */
+ attrs = MMPLAYER_GET_ATTRS(player);
- if ( ! __mmplayer_try_to_plug( player, pad, caps ) )
+ if ( attrs )
{
- debug_error("failed to autoplug for caps : %s\n", gst_caps_to_string( caps ) );
- goto ERROR;
- }
+ gboolean smooth_repeat = FALSE;
- /* check if there's dynamic pad*/
- if( player->have_dynamic_pad )
- {
- debug_error("using pad caps assums there's no dynamic pad !\n");
- debug_error("try with enalbing rtsp_do_typefinding\n");
- goto ERROR;
- }
+ mm_attrs_get_int_by_name(attrs, "profile_play_count", &count);
+ mm_attrs_get_int_by_name(attrs, "profile_smooth_repeat", &smooth_repeat);
- gst_caps_unref( caps );
- caps = NULL;
- }
- }
+ player->play_count = count;
- /* excute new_element if created*/
- if ( new_element )
- {
- debug_log("adding new element to pipeline\n");
+ debug_log("remaining play count: %d, playback rate: %f\n", count, player->playback_rate);
- /* set state to READY before add to bin */
- MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_READY );
+ if ( count > 1 || count == -1 || player->playback_rate < 0.0 ) /* default value is 1 */
+ {
+ if ( smooth_repeat )
+ {
+ debug_log("smooth repeat enabled. seeking operation will be excuted in new thread\n");
- /* add new element to the pipeline */
- if ( FALSE == gst_bin_add( GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), new_element) )
- {
- debug_error("failed to add autoplug element to bin\n");
- goto ERROR;
- }
+ g_cond_signal( player->repeat_thread_cond );
- /* get pad from element */
- sinkpad = gst_element_get_static_pad ( GST_ELEMENT(new_element), "sink" );
- if ( !sinkpad )
- {
- debug_error("failed to get sinkpad from autoplug element\n");
- goto ERROR;
- }
+ break;
+ }
+ else
+ {
+ gint ret_value = 0;
- /* link it */
- if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
- {
- debug_error("failed to link autoplug element\n");
- goto ERROR;
- }
+ if ( player->section_repeat )
+ {
+ ret_value = _mmplayer_activate_section_repeat((MMHandleType)player, player->section_repeat_start, player->section_repeat_end);
+ }
+ else
+ {
+ if ( player->playback_rate < 0.0 )
+ {
+ player->resumed_by_rewind = TRUE;
+ _mmplayer_set_mute((MMHandleType)player, 0);
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_RESUMED_BY_REW, NULL );
+ }
- gst_object_unref (sinkpad);
- sinkpad = NULL;
+ __mmplayer_handle_eos_delay( player, player->ini.delay_before_repeat );
- /* run. setting PLAYING here since streamming source is live source */
- MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_PLAYING );
- }
+ /* initialize */
+ player->sent_bos = FALSE;
+ }
- /* store handle to futher manipulation */
- player->pipeline->mainbin[element_id].id = element_id;
- player->pipeline->mainbin[element_id].gst = new_element;
+ if ( MM_ERROR_NONE != ret_value )
+ {
+ debug_error("failed to set position to zero for rewind\n");
+ }
- debug_fleave();
+ /* not posting eos when repeating */
+ break;
+ }
+ }
+ }
- return;
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-eos" );
-STATE_CHANGE_FAILED:
-ERROR:
- /* FIXIT : take care if new_element has already added to pipeline */
- if ( new_element )
- gst_object_unref(GST_OBJECT(new_element));
+ /* post eos message to application */
+ __mmplayer_handle_eos_delay( player, player->ini.eos_delay );
- if ( sinkpad )
- gst_object_unref(GST_OBJECT(sinkpad));
+ /* reset last position */
+ player->last_position = 0;
+ }
+ break;
- if ( caps )
- gst_object_unref(GST_OBJECT(caps));
+ case GST_MESSAGE_ERROR:
+ {
+ GError *error = NULL;
+ gchar* debug = NULL;
- /* FIXIT : how to inform this error to MSL ????? */
- /* FIXIT : I think we'd better to use g_idle_add() to destroy pipeline and
- * then post an error to application
- */
-}
+ /* generating debug info before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-error" );
+ /* get error code */
+ gst_message_parse_error( msg, &error, &debug );
-static void
-__mmplayer_gst_decode_callback(GstElement *decodebin, GstPad *pad, gboolean last, gpointer data) // @
-{
- mm_player_t* player = NULL;
- MMHandleType attrs = 0;
- GstElement* pipeline = NULL;
- GstCaps* caps = NULL;
- GstStructure* str = NULL;
- const gchar* name = NULL;
- GstPad* sinkpad = NULL;
- GstElement* sinkbin = NULL;
+ if ( gst_structure_has_name ( msg->structure, "streaming_error" ) )
+ {
+ /* Note : the streaming error from the streaming source is handled
+ * using __mmplayer_handle_streaming_error.
+ */
+ __mmplayer_handle_streaming_error ( player, msg );
- /* check handles */
- player = (mm_player_t*) data;
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+ }
+ else
+ {
+ /* traslate gst error code to msl error code. then post it
+ * to application if needed
+ */
+ __mmplayer_handle_gst_error( player, msg, error );
- return_if_fail( decodebin && pad );
- return_if_fail(player && player->pipeline && player->pipeline->mainbin);
+ if (debug)
+ {
+ debug_error ("error debug : %s", debug);
+ }
- pipeline = player->pipeline->mainbin[MMPLAYER_M_PIPE].gst;
+ }
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
- {
- debug_error("cannot get content attribute\n");
- goto ERROR;
- }
+ if (MMPLAYER_IS_HTTP_PD(player))
+ {
+ _mmplayer_unrealize_pd_downloader ((MMHandleType)player);
+ }
- /* get mimetype from caps */
- caps = gst_pad_get_caps( pad );
- if ( !caps )
- {
- debug_error("cannot get caps from pad.\n");
- goto ERROR;
- }
+ MMPLAYER_FREEIF( debug );
+ g_error_free( error );
+ }
+ break;
- str = gst_caps_get_structure( caps, 0 );
- if ( ! str )
- {
- debug_error("cannot get structure from capse.\n");
- goto ERROR;
- }
+ case GST_MESSAGE_WARNING:
+ {
+ char* debug = NULL;
+ GError* error = NULL;
- name = gst_structure_get_name(str);
- if ( ! name )
- {
- debug_error("cannot get mimetype from structure.\n");
- goto ERROR;
- }
+ gst_message_parse_warning(msg, &error, &debug);
- debug_log("detected mimetype : %s\n", name);
+ debug_log("warning : %s\n", error->message);
+ debug_log("debug : %s\n", debug);
- if (strstr(name, "audio"))
- {
- if (player->pipeline->audiobin == NULL)
- {
- __ta__("__mmplayer_gst_create_audio_pipeline",
- if (MM_ERROR_NONE != __mmplayer_gst_create_audio_pipeline(player))
- {
- debug_error("failed to create audiobin. continuing without audio\n");
- goto ERROR;
- }
- )
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_WARNING, NULL );
- sinkbin = player->pipeline->audiobin[MMPLAYER_A_BIN].gst;
- debug_log("creating audiosink bin success\n");
+ MMPLAYER_FREEIF( debug );
+ g_error_free( error );
}
- else
+ break;
+
+ case GST_MESSAGE_TAG:
{
- sinkbin = player->pipeline->audiobin[MMPLAYER_A_BIN].gst;
- debug_log("re-using audiobin\n");
+ debug_log("GST_MESSAGE_TAG\n");
+ if ( ! __mmplayer_gst_extract_tag_from_msg( player, msg ) )
+ {
+ debug_warning("failed to extract tags from gstmessage\n");
+ }
}
+ break;
- /* FIXIT : track number shouldn't be hardcoded */
- mm_attrs_set_int_by_name(attrs, "content_audio_track_num", 1);
+ case GST_MESSAGE_BUFFERING:
+ {
+ MMMessageParamType msg_param = {0, };
+ int asm_result = MM_ERROR_NONE;
- player->audiosink_linked = 1;
- debug_msg("player->audsink_linked set to 1\n");
+ if (!MMPLAYER_IS_STREAMING(player))
+ break;
- }
- else if (strstr(name, "video"))
- {
- if (player->pipeline->videobin == NULL)
- {
- __mmplayer_set_videosink_type(player);
+ /* ignore the prev buffering message */
+ if ((player->streamer) && (player->streamer->is_buffering == FALSE) && (player->streamer->is_buffering_done == TRUE))
+ {
+ gint buffer_percent = 0;
- /* NOTE : not make videobin because application dose not want to play it even though file has video stream.
- */
- if (PLAYER_INI()->video_surface == MM_DISPLAY_SURFACE_NULL)
- {
- debug_log("not make videobin because it dose not want\n");
- goto ERROR;
- }
+ gst_message_parse_buffering (msg, &buffer_percent);
- __ta__("__mmplayer_gst_create_video_pipeline",
- if (MM_ERROR_NONE != __mmplayer_gst_create_video_pipeline(player, caps) )
+ if (buffer_percent == MAX_BUFFER_PERCENT)
{
- debug_error("failed to create videobin. continuing without video\n");
- goto ERROR;
+ debug_log ("Ignored all the previous buffering msg! (got %d%%)\n", buffer_percent);
+ player->streamer->is_buffering_done = FALSE;
}
- )
-
- sinkbin = player->pipeline->videobin[MMPLAYER_V_BIN].gst;
- debug_log("creating videosink bin success\n");
- }
- else
- {
- sinkbin = player->pipeline->videobin[MMPLAYER_V_BIN].gst;
- debug_log("re-using videobin\n");
- }
-
- /* FIXIT : track number shouldn't be hardcoded */
- mm_attrs_set_int_by_name(attrs, "content_video_track_num", 1);
- player->videosink_linked = 1;
- debug_msg("player->videosink_linked set to 1\n");
+ break;
+ }
- }
- else if (strstr(name, "text"))
- {
- if (player->pipeline->textbin == NULL)
- {
- __ta__("__mmplayer_gst_create_text_pipeline",
- if (MM_ERROR_NONE != __mmplayer_gst_create_text_pipeline(player))
+ /* update ASM state to ASM_STATE_PLAYING */
+ /* fixed ASM_STATE_WAITING -> ASM_STATE_PLAYING for Samsunlink issue*/
+ if ((player->streamer) && (player->streamer->is_buffering == FALSE) && (MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PAUSED))
+ {
+ asm_result = _mmplayer_asm_set_state((MMHandleType)player, ASM_STATE_WAITING, TRUE);
+ if ( asm_result != MM_ERROR_NONE )
{
- debug_error("failed to create textbin. continuing without text\n");
- goto ERROR;
+ debug_warning("failed to set asm state to waiting, but keep going...\n");
}
- )
-
- sinkbin = player->pipeline->textbin[MMPLAYER_T_BIN].gst;
- debug_log("creating textink bin success\n");
- }
- else
- {
- sinkbin = player->pipeline->textbin[MMPLAYER_T_BIN].gst;
- debug_log("re-using textbin\n");
- }
+ }
- /* FIXIT : track number shouldn't be hardcoded */
- mm_attrs_set_int_by_name(attrs, "content_text_track_num", 1);
+ __mmplayer_update_buffer_setting(player, msg);
- player->textsink_linked = 1;
- debug_msg("player->textsink_linked set to 1\n");
- }
- else
- {
- debug_warning("unknown type of elementary stream! ignoring it...\n");
- goto ERROR;
- }
+ __mmplayer_handle_buffering_message ( player );
- if ( sinkbin )
- {
- /* warm up */
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state( sinkbin, GST_STATE_READY ) )
- {
- debug_error("failed to set state(READY) to sinkbin\n");
- goto ERROR;
+ msg_param.connection.buffering = player->streamer->buffering_percent;
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_BUFFERING, &msg_param );
+ if (MMPLAYER_IS_RTSP_STREAMING(player) && (player->streamer->buffering_percent >= MAX_BUFFER_PERCENT))
+ {
+ if (player->doing_seek)
+ {
+ if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PAUSED)
+ {
+ player->doing_seek = FALSE;
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
+ }
+ else if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PLAYING)
+ {
+ async_done = TRUE;
+ }
+ }
+ }
}
+ break;
- /* add */
- if ( FALSE == gst_bin_add( GST_BIN(pipeline), sinkbin ) )
+ case GST_MESSAGE_STATE_CHANGED:
{
- debug_error("failed to add sinkbin to pipeline\n");
- goto ERROR;
- }
+ MMPlayerGstElement *mainbin;
+ const GValue *voldstate, *vnewstate, *vpending;
+ GstState oldstate, newstate, pending;
- sinkpad = gst_element_get_static_pad( GST_ELEMENT(sinkbin), "sink" );
+ if ( ! ( player->pipeline && player->pipeline->mainbin ) )
+ {
+ debug_error("player pipeline handle is null");
+ break;
+ }
- if ( !sinkpad )
- {
- debug_error("failed to get pad from sinkbin\n");
- goto ERROR;
- }
+ mainbin = player->pipeline->mainbin;
- /* link */
- if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
- {
- debug_error("failed to get pad from sinkbin\n");
- goto ERROR;
- }
+ /* we only handle messages from pipeline */
+ if( msg->src != (GstObject *)mainbin[MMPLAYER_M_PIPE].gst )
+ break;
- /* run */
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state( sinkbin, GST_STATE_PAUSED ) )
- {
- debug_error("failed to set state(PLAYING) to sinkbin\n");
- goto ERROR;
- }
+ /* get state info from msg */
+ voldstate = gst_structure_get_value (msg->structure, "old-state");
+ vnewstate = gst_structure_get_value (msg->structure, "new-state");
+ vpending = gst_structure_get_value (msg->structure, "pending-state");
- gst_object_unref( sinkpad );
- sinkpad = NULL;
- }
+ oldstate = (GstState)voldstate->data[0].v_int;
+ newstate = (GstState)vnewstate->data[0].v_int;
+ pending = (GstState)vpending->data[0].v_int;
- /* update track number attributes */
- if ( mmf_attrs_commit ( attrs ) )
- debug_error("failed to commit attrs\n");
+ debug_log("state changed [%s] : %s ---> %s final : %s\n",
+ GST_OBJECT_NAME(GST_MESSAGE_SRC(msg)),
+ gst_element_state_get_name( (GstState)oldstate ),
+ gst_element_state_get_name( (GstState)newstate ),
+ gst_element_state_get_name( (GstState)pending ) );
- debug_log("linking sink bin success\n");
+ if (oldstate == newstate)
+ {
+ debug_log("pipeline reports state transition to old state");
+ break;
+ }
+ switch(newstate)
+ {
+ case GST_STATE_VOID_PENDING:
+ break;
- /* FIXIT : we cannot hold callback for 'no-more-pad' signal because signal was emitted in
- * streaming task. if the task blocked, then buffer will not flow to the next element
- * ( autoplugging element ). so this is special hack for streaming. please try to remove it
- */
- /* dec stream count. we can remove fakesink if it's zero */
- player->num_dynamic_pad--;
+ case GST_STATE_NULL:
+ break;
- debug_log("stream count dec : %d (num of dynamic pad)\n", player->num_dynamic_pad);
+ case GST_STATE_READY:
+ break;
- if ( ( player->no_more_pad ) && ( player->num_dynamic_pad == 0 ) )
- {
- __mmplayer_pipeline_complete( NULL, player );
- }
+ case GST_STATE_PAUSED:
+ {
+ gboolean prepare_async = FALSE;
+ gboolean is_drm = FALSE;
-ERROR:
- if ( caps )
- gst_caps_unref( caps );
+ if ( ! player->audio_cb_probe_id && player->set_mode.pcm_extraction)
+ __mmplayer_configure_audio_callback(player);
- if ( sinkpad )
- gst_object_unref(GST_OBJECT(sinkpad));
+ if ( ! player->sent_bos && oldstate == GST_STATE_READY) // managed prepare async case
+ {
+ mm_attrs_get_int_by_name(player->attrs, "profile_prepare_async", &prepare_async);
+ debug_log("checking prepare mode for async transition - %d", prepare_async);
+ }
- return;
-}
+ if ( MMPLAYER_IS_STREAMING(player) || prepare_async )
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PAUSED );
-int
-_mmplayer_update_video_param(mm_player_t* player) // @
-{
- MMHandleType attrs = 0;
- gint videosink_idx_x = 0;
- gint videosink_idx_evas = 0;
- gboolean is_first_v_sink_x = FALSE;
+ if (MMPLAYER_IS_STREAMING(player) && (player->streamer))
+ {
+ __mm_player_streaming_set_content_bitrate(player->streamer,
+ player->total_maximum_bitrate, player->total_bitrate);
+ }
+ }
- debug_fenter();
+ /* NOTE : should consider streaming case */
+ /* check if drm file */
+ if ((player->pipeline->mainbin[MMPLAYER_M_SRC].gst) &&
+ (g_object_class_find_property(G_OBJECT_GET_CLASS(player->pipeline->mainbin[MMPLAYER_M_SRC].gst), "is-drm")))
+ {
+ g_object_get(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_SRC].gst), "is-drm", &is_drm, NULL);
- /* check video sinkbin is created */
- return_val_if_fail ( player &&
- player->pipeline &&
- player->pipeline->videobin &&
- player->pipeline->videobin[MMPLAYER_V_BIN].gst &&
- player->pipeline->videobin[MMPLAYER_V_SINK].gst,
- MM_ERROR_PLAYER_NOT_INITIALIZED );
+ if (is_drm)
+ {
+ player->is_drm_file = TRUE;
+ }
+ }
+ }
+ break;
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
- {
- debug_error("cannot get content attribute");
- return MM_ERROR_PLAYER_INTERNAL;
- }
+ case GST_STATE_PLAYING:
+ {
+ if ( MMPLAYER_IS_STREAMING(player) ) // managed prepare async case when buffering is completed
+ {
+ // pending state should be reset oyherwise, it's still playing even though it's resumed after bufferging.
+ if ((MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PLAYING) ||
+ (MMPLAYER_PENDING_STATE(player) == MM_PLAYER_STATE_PLAYING))
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PLAYING);
+ }
+ }
- /* update display surface */
- __mmplayer_set_videosink_type(player);
+ if (player->src_changed)
+ {
+ _mmplayer_update_content_attrs(player, ATTR_ALL);
+ player->src_changed = FALSE;
+ }
- /* check video stream callback is used */
- if( player->use_video_stream )
- {
- if (player->is_nv12_tiled)
- {
- gchar *ename = NULL;
- int degree, width, height;
- degree = width = height = 0;
-
- mm_attrs_get_int_by_name(attrs, "display_width", &width);
- mm_attrs_get_int_by_name(attrs, "display_height", &height);
- mm_attrs_get_int_by_name(attrs, "display_rotation", °ree);
+ if (player->doing_seek && async_done)
+ {
+ player->doing_seek = FALSE;
+ async_done = FALSE;
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
+ }
+ }
+ break;
- /* resize video frame with requested values for fimcconvert */
- ename = GST_PLUGIN_FEATURE_NAME(gst_element_get_factory(player->pipeline->videobin[MMPLAYER_V_CONV].gst));
+ default:
+ break;
+ }
+ }
+ break;
- if (g_strrstr(ename, "fimcconvert"))
+ case GST_MESSAGE_CLOCK_LOST:
{
- if (width)
- g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "src-width", width, NULL);
-
- if (height)
- g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "src-height", height, NULL);
+ GstClock *clock = NULL;
+ gst_message_parse_clock_lost (msg, &clock);
+ debug_log("GST_MESSAGE_CLOCK_LOST : %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
+ g_print ("GST_MESSAGE_CLOCK_LOST : %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
- switch (degree)
+ if (((player->ini.provide_clock_for_music) && (!player->videodec_linked)) ||
+ ((player->ini.provide_clock_for_movie) && (player->videodec_linked)))
{
- case MM_DISPLAY_ROTATION_NONE:
- degree = 0;
- break;
-
- case MM_DISPLAY_ROTATION_90:
- degree = 90;
- break;
-
- case MM_DISPLAY_ROTATION_180:
- degree = 180;
- break;
-
- case MM_DISPLAY_ROTATION_270:
- degree = 270;
- break;
-
- default:
- degree = 0;
- break;
+ debug_log ("Provide clock is TRUE, do pause->resume\n");
+ __gst_pause(player, FALSE);
+ __gst_resume(player, FALSE);
}
-
- g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "rotate", degree, NULL);
-
- debug_log("updating fimcconvert - r[%d], w[%d], h[%d]", degree, width, height);
}
- else
+ break;
+
+ case GST_MESSAGE_NEW_CLOCK:
{
- debug_error("no available video converter");
+ GstClock *clock = NULL;
+ gst_message_parse_new_clock (msg, &clock);
+ debug_log("GST_MESSAGE_NEW_CLOCK : %s\n", (clock ? GST_OBJECT_NAME (clock) : "NULL"));
}
- }
- else
- {
- int rotate, width, height, orientation, rotation;
+ break;
- rotate = width = height = orientation = rotation = 0;
+ case GST_MESSAGE_ELEMENT:
+ {
+ const gchar *structure_name;
+ gint count = 0;
+ MMHandleType attrs = 0;
- debug_log("using video stream callback with memsink. player handle : [%p]", player);
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute");
+ ret = FALSE;
+ break;
+ }
- mm_attrs_get_int_by_name(attrs, "display_width", &width);
- mm_attrs_get_int_by_name(attrs, "display_height", &height);
- mm_attrs_get_int_by_name(attrs, "display_rotation", &rotation);
- mm_attrs_get_int_by_name(attrs, "display_orientation", &orientation);
+ if(msg->structure == NULL)
+ break;
- if(rotation == MM_DISPLAY_ROTATION_NONE) rotate = 0;
- else if(rotation ==MM_DISPLAY_ROTATION_90) rotate = 90;
- else if(rotation ==MM_DISPLAY_ROTATION_180) rotate = 180;
- else if(rotation ==MM_DISPLAY_ROTATION_270) rotate = 270;
+ structure_name = gst_structure_get_name(msg->structure);
+ if(!strcmp(structure_name, "Language_list"))
+ {
+ const GValue *lang_list = NULL;
+ lang_list = gst_structure_get_value (msg->structure, "lang_list");
+ if(lang_list != NULL)
+ {
+ count = g_list_length((GList *)g_value_get_pointer (lang_list));
+ if (count > 1)
+ debug_log("Total audio tracks (from parser) = %d \n",count);
+ }
+ }
- if(orientation == 1) rotate = 90;
- else if(orientation == 2) rotate = 180;
- else if(orientation == 3) rotate = 270;
-
- if (width)
- g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst, "width", width, NULL);
+ if (!strcmp (structure_name, "Ext_Sub_Language_List"))
+ {
+ const GValue *lang_list = NULL;
+ MMPlayerLangStruct *temp = NULL;
- if (height)
- g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst, "height", height, NULL);
+ lang_list = gst_structure_get_value (msg->structure, "lang_list");
+ if (lang_list != NULL)
+ {
+ count = g_list_length ((GList *)g_value_get_pointer (lang_list));
+ if (count)
+ {
+ player->subtitle_language_list = (GList *)g_value_get_pointer (lang_list);
+ mm_attrs_set_int_by_name(attrs, "content_text_track_num", (gint)count);
+ if (mmf_attrs_commit (attrs))
+ debug_error("failed to commit.\n");
+ debug_log("Total external subtitle tracks = %d \n", count);
+ }
+ while (count)
+ {
+ temp = g_list_nth_data (player->subtitle_language_list, count - 1);
+ debug_log ("value of lang_key is %s and lang_code is %s",
+ temp->language_key, temp->language_code);
+ count--;
+ }
+ }
+ }
+ if (!strcmp (structure_name, "Int_Sub_Language_List"))
+ {
+ const GValue *lang_list = NULL;
+ MMPlayerLangStruct *temp = NULL;
+ lang_list = gst_structure_get_value (msg->structure, "lang_list");
+ if (lang_list != NULL && !player->is_external_subtitle_present)
+ {
+ count = g_list_length ((GList *)g_value_get_pointer (lang_list));
+ if (count)
+ {
+ player->subtitle_language_list = (GList *)g_value_get_pointer (lang_list);
+ g_object_set (G_OBJECT (player->pipeline->mainbin[MMPLAYER_M_T_SUBMUX_INTERNAL].gst), "lang-list", player->subtitle_language_list, NULL);
+ mm_attrs_set_int_by_name(attrs, "content_text_track_num", (gint)count);
+ if (mmf_attrs_commit (attrs))
+ debug_error("failed to commit.\n");
+ debug_log("Total internal subtitle tracks = %d \n", count);
+ }
+ while (count)
+ {
+ temp = g_list_nth_data (player->subtitle_language_list, count - 1);
+ debug_log ("value of lang_key is %s and lang_code is %s",
+ temp->language_key, temp->language_code);
+ count--;
+ }
+ } else if (lang_list != NULL) {
+ GList* internal_lang_list;
+ internal_lang_list = (GList *)g_value_get_pointer (lang_list);
+ debug_log ("it is internal & external case simultaneously");
+ g_object_set (G_OBJECT (player->pipeline->mainbin[MMPLAYER_M_T_SUBMUX_INTERNAL].gst), "lang-list", internal_lang_list, NULL);
+ } else {
+ debug_error ("language list is coming null");
+ }
+ }
- g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst, "rotate", rotate,NULL);
- }
-
- return MM_ERROR_NONE;
- }
-
- if (player->use_multi_surface)
- {
- is_first_v_sink_x = !strncmp(GST_ELEMENT_NAME(player->pipeline->videobin[MMPLAYER_V_SINK_EXT].gst), "videosink_ext_evas", 18);
- if (is_first_v_sink_x)
- {
- videosink_idx_x = MMPLAYER_V_SINK;
- videosink_idx_evas = MMPLAYER_V_SINK_EXT;
- }
- else
- {
- videosink_idx_x = MMPLAYER_V_SINK_EXT;
- videosink_idx_evas = MMPLAYER_V_SINK;
- }
- debug_log("use multi-surface : videosink_idx_x(%d), videosink_idx_evas(%d)", videosink_idx_x,videosink_idx_evas);
- }
- else
- {
- videosink_idx_x = videosink_idx_evas = MMPLAYER_V_SINK;
- }
-
- /* configuring display */
- switch ( PLAYER_INI()->video_surface )
- {
- case MM_DISPLAY_SURFACE_X:
- {
- /* ximagesink or xvimagesink */
- void *xid = NULL;
- int zoom = 0;
- int degree = 0;
- int display_method = 0;
- int roi_x = 0;
- int roi_y = 0;
- int roi_w = 0;
- int roi_h = 0;
- int force_aspect_ratio = 0;
-
- gboolean visible = TRUE;
-
- /* common case if using multi-surface */
- if (player->use_multi_surface)
- {
- void *evas_image_object = NULL;
- if (is_first_v_sink_x)
- {
- mm_attrs_get_data_by_name(attrs, "display_overlay", &xid);
- mm_attrs_get_data_by_name(attrs, "display_overlay_ext", &evas_image_object);
- }
- else
- {
- mm_attrs_get_data_by_name(attrs, "display_overlay", &evas_image_object);
- mm_attrs_get_data_by_name(attrs, "display_overlay_ext", &xid);
- }
- if ( xid )
- {
- debug_log("use multi-surface : xid %d", *(int*)xid);
- gst_x_overlay_set_xwindow_id( GST_X_OVERLAY( player->pipeline->videobin[videosink_idx_x].gst ), *(int*)xid );
- }
- else
- {
- debug_error("no xwindow");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- if (evas_image_object)
- {
- g_object_set(player->pipeline->videobin[videosink_idx_evas].gst,
- "evas-object", evas_image_object,
- "visible", FALSE,
- NULL);
- }
- else
- {
- debug_error("no evas object");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- debug_log("use multi-surface : evas_image_object (%x)", evas_image_object);
- debug_log("use multi-surface : evas visible %d", FALSE);
- }
- /* common case if using x surface */
- else
- {
- mm_attrs_get_data_by_name(attrs, "display_overlay", &xid);
- if ( xid )
- {
- debug_log("set video param : xid %d", *(int*)xid);
- gst_x_overlay_set_xwindow_id( GST_X_OVERLAY( player->pipeline->videobin[MMPLAYER_V_SINK].gst ), *(int*)xid );
- }
- else
- {
- /* FIXIT : is it error case? */
- debug_warning("still we don't have xid on player attribute. create it's own surface.");
+ /* custom message */
+ if (!strcmp (structure_name, "audio_codec_not_supported")) {
+ MMMessageParamType msg_param = {0,};
+ msg_param.code = MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
+ MMPLAYER_POST_MSG(player, MM_MESSAGE_ERROR, &msg_param);
}
}
+ break;
- /* if xvimagesink */
- if (!strcmp(PLAYER_INI()->videosink_element_x,"xvimagesink"))
+ case GST_MESSAGE_DURATION:
+ {
+ debug_log("GST_MESSAGE_DURATION\n");
+ ret = __mmplayer_gst_handle_duration(player, msg);
+ if (!ret)
{
- mm_attrs_get_int_by_name(attrs, "display_force_aspect_ration", &force_aspect_ratio);
- mm_attrs_get_int_by_name(attrs, "display_zoom", &zoom);
- mm_attrs_get_int_by_name(attrs, "display_rotation", °ree);
- mm_attrs_get_int_by_name(attrs, "display_method", &display_method);
- mm_attrs_get_int_by_name(attrs, "display_roi_x", &roi_x);
- mm_attrs_get_int_by_name(attrs, "display_roi_y", &roi_y);
- mm_attrs_get_int_by_name(attrs, "display_roi_width", &roi_w);
- mm_attrs_get_int_by_name(attrs, "display_roi_height", &roi_h);
- mm_attrs_get_int_by_name(attrs, "display_visible", &visible);
+ debug_warning("failed to update duration");
+ }
+ }
- g_object_set(player->pipeline->videobin[videosink_idx_x].gst,
- "force-aspect-ratio", force_aspect_ratio,
- "zoom", zoom,
- "rotate", degree,
- "handle-events", TRUE,
- "display-geometry-method", display_method,
- "draw-borders", FALSE,
- "dst-roi-x", roi_x,
- "dst-roi-y", roi_y,
- "dst-roi-w", roi_w,
- "dst-roi-h", roi_h,
- "visible", visible,
- NULL );
+ break;
- debug_log("set video param : zoom %d", zoom);
- debug_log("set video param : rotate %d", degree);
- debug_log("set video param : method %d", display_method);
- debug_log("set video param : dst-roi-x: %d, dst-roi-y: %d, dst-roi-w: %d, dst-roi-h: %d",
- roi_x, roi_y, roi_w, roi_h );
- debug_log("set video param : visible %d", visible);
- debug_log("set video param : force aspect ratio %d", force_aspect_ratio);
- }
+ case GST_MESSAGE_ASYNC_START:
+ {
+ debug_log("GST_MESSAGE_ASYNC_START : %s\n", GST_ELEMENT_NAME(GST_MESSAGE_SRC(msg)));
}
break;
- case MM_DISPLAY_SURFACE_EVAS:
+
+ case GST_MESSAGE_ASYNC_DONE:
{
- void *object = NULL;
- int scaling = 0;
- gboolean visible = TRUE;
+ debug_log("GST_MESSAGE_ASYNC_DONE : %s\n", GST_ELEMENT_NAME(GST_MESSAGE_SRC(msg)));
- /* common case if using multi-surface */
- if (player->use_multi_surface)
- {
- void *xid = NULL;
- mm_attrs_get_int_by_name(attrs, "display_visible", &visible);
- if (is_first_v_sink_x)
- {
- mm_attrs_get_data_by_name(attrs, "display_overlay", &xid);
- mm_attrs_get_data_by_name(attrs, "display_overlay_ext", &object);
- }
- else
- {
- mm_attrs_get_data_by_name(attrs, "display_overlay", &object);
- mm_attrs_get_data_by_name(attrs, "display_overlay_ext", &xid);
- }
- if (object)
- {
- g_object_set(player->pipeline->videobin[videosink_idx_evas].gst,
- "evas-object", object,
- "visible", visible,
- NULL);
- debug_log("use multi-surface : evas-object %x", object);
- debug_log("use multi-surface : evas visible %d", object);
- }
- else
- {
- debug_error("no evas object");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- if (xid)
- {
- gst_x_overlay_set_xwindow_id( GST_X_OVERLAY( player->pipeline->videobin[videosink_idx_x].gst ), *(int*)xid );
- }
- else
- {
- debug_error("no xwindow");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- /* if xvimagesink */
- if (!strcmp(PLAYER_INI()->videosink_element_x,"xvimagesink"))
- {
- g_object_set(player->pipeline->videobin[videosink_idx_x].gst,
- "visible", FALSE,
- NULL );
- }
- debug_log("use multi-surface : xid %d", *(int*)xid);
- debug_log("use multi-surface : x visible %d", FALSE);
- }
- /* common case if using evas surface */
- else
- {
- mm_attrs_get_data_by_name(attrs, "display_overlay", &object);
- mm_attrs_get_int_by_name(attrs, "display_visible", &visible);
- mm_attrs_get_int_by_name(attrs, "display_evas_do_scaling", &scaling);
- if (object)
- {
- g_object_set(player->pipeline->videobin[videosink_idx_evas].gst,
- "evas-object", object,
- "visible", visible,
- NULL);
- debug_log("set video param : evas-object %x", object);
- debug_log("set video param : visible %d", visible);
- }
- else
- {
- debug_error("no evas object");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- }
+ /* we only handle messages from pipeline */
+ if( msg->src != (GstObject *)player->pipeline->mainbin[MMPLAYER_M_PIPE].gst )
+ break;
- /* if evasimagesink */
- if (!strcmp(PLAYER_INI()->videosink_element_evas,"evasimagesink") && player->is_nv12_tiled)
+ if( !MMPLAYER_IS_RTSP_STREAMING(player) )
{
- int width = 0;
- int height = 0;
- int no_scaling = !scaling;
-
- mm_attrs_get_int_by_name(attrs, "display_width", &width);
- mm_attrs_get_int_by_name(attrs, "display_height", &height);
-
- if (no_scaling)
- {
- /* no-scaling order to fimcconvert, original width, height size of media src will be passed to sink plugin */
- g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst,
- "src-width", 0, /* setting 0, output video width will be media src's width */
- "src-height", 0, /* setting 0, output video height will be media src's height */
- NULL);
- }
- else
+ if (player->doing_seek)
{
- /* scaling order to fimcconvert */
- if (width)
+ if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PAUSED)
{
- g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "src-width", width, NULL);
+ player->doing_seek = FALSE;
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
}
- if (height)
+ else if (MMPLAYER_TARGET_STATE(player) == MM_PLAYER_STATE_PLAYING)
{
- g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "src-height", height, NULL);
- }
- debug_log("set video param : video frame scaling down to width(%d) height(%d)", width, height);
- }
- debug_log("set video param : display_evas_do_scaling %d", scaling);
- }
-
- /* if evaspixmapsink */
- if (!strcmp(PLAYER_INI()->videosink_element_evas,"evaspixmapsink"))
- {
- int display_method = 0;
- int roi_x = 0;
- int roi_y = 0;
- int roi_w = 0;
- int roi_h = 0;
- int force_aspect_ratio = 0;
- int origin_size = !scaling;
+ if ((MMPLAYER_IS_HTTP_STREAMING(player)) &&
+ (player->streamer) &&
+ (player->streamer->streaming_buffer_type == BUFFER_TYPE_MUXED) &&
+ (player->streamer->is_buffering == FALSE))
+ {
+ GstQuery *query = NULL;
+ gboolean busy = FALSE;
+ gint percent = 0;
- mm_attrs_get_int_by_name(attrs, "display_force_aspect_ration", &force_aspect_ratio);
- mm_attrs_get_int_by_name(attrs, "display_method", &display_method);
- mm_attrs_get_int_by_name(attrs, "display_roi_x", &roi_x);
- mm_attrs_get_int_by_name(attrs, "display_roi_y", &roi_y);
- mm_attrs_get_int_by_name(attrs, "display_roi_width", &roi_w);
- mm_attrs_get_int_by_name(attrs, "display_roi_height", &roi_h);
+ if (player->streamer->buffer_handle[BUFFER_TYPE_MUXED].buffer)
+ {
+ query = gst_query_new_buffering ( GST_FORMAT_PERCENT );
+ if ( gst_element_query (player->streamer->buffer_handle[BUFFER_TYPE_MUXED].buffer, query ) )
+ {
+ gst_query_parse_buffering_percent ( query, &busy, &percent);
+ }
+ gst_query_unref (query);
+
+ debug_log("buffered percent(%s): %d\n",
+ GST_ELEMENT_NAME(player->streamer->buffer_handle[BUFFER_TYPE_MUXED].buffer), percent);
+ }
- g_object_set(player->pipeline->videobin[videosink_idx_evas].gst,
- "force-aspect-ratio", force_aspect_ratio,
- "origin-size", origin_size,
- "dst-roi-x", roi_x,
- "dst-roi-y", roi_y,
- "dst-roi-w", roi_w,
- "dst-roi-h", roi_h,
- "display-geometry-method", display_method,
- NULL );
+ if (percent >= 100)
+ {
+ player->streamer->is_buffering = FALSE;
+ __mmplayer_handle_buffering_message(player);
+ }
+ }
- debug_log("set video param : method %d", display_method);
- debug_log("set video param : dst-roi-x: %d, dst-roi-y: %d, dst-roi-w: %d, dst-roi-h: %d",
- roi_x, roi_y, roi_w, roi_h );
- debug_log("set video param : force aspect ratio %d", force_aspect_ratio);
- debug_log("set video param : display_evas_do_scaling %d (origin-size %d)", scaling, origin_size);
+ async_done = TRUE;
+ }
+ }
}
}
break;
- case MM_DISPLAY_SURFACE_NULL:
- {
- /* do nothing */
- }
+
+ #if 0 /* delete unnecessary logs */
+ case GST_MESSAGE_REQUEST_STATE: debug_log("GST_MESSAGE_REQUEST_STATE\n"); break;
+ case GST_MESSAGE_STEP_START: debug_log("GST_MESSAGE_STEP_START\n"); break;
+ case GST_MESSAGE_QOS: debug_log("GST_MESSAGE_QOS\n"); break;
+ case GST_MESSAGE_PROGRESS: debug_log("GST_MESSAGE_PROGRESS\n"); break;
+ case GST_MESSAGE_ANY: debug_log("GST_MESSAGE_ANY\n"); break;
+ case GST_MESSAGE_INFO: debug_log("GST_MESSAGE_STATE_DIRTY\n"); break;
+ case GST_MESSAGE_STATE_DIRTY: debug_log("GST_MESSAGE_STATE_DIRTY\n"); break;
+ case GST_MESSAGE_STEP_DONE: debug_log("GST_MESSAGE_STEP_DONE\n"); break;
+ case GST_MESSAGE_CLOCK_PROVIDE: debug_log("GST_MESSAGE_CLOCK_PROVIDE\n"); break;
+ case GST_MESSAGE_STRUCTURE_CHANGE: debug_log("GST_MESSAGE_STRUCTURE_CHANGE\n"); break;
+ case GST_MESSAGE_STREAM_STATUS: debug_log("GST_MESSAGE_STREAM_STATUS\n"); break;
+ case GST_MESSAGE_APPLICATION: debug_log("GST_MESSAGE_APPLICATION\n"); break;
+ case GST_MESSAGE_SEGMENT_START: debug_log("GST_MESSAGE_SEGMENT_START\n"); break;
+ case GST_MESSAGE_SEGMENT_DONE: debug_log("GST_MESSAGE_SEGMENT_DONE\n"); break;
+ case GST_MESSAGE_LATENCY: debug_log("GST_MESSAGE_LATENCY\n"); break;
+ #endif
+
+ default:
break;
}
- debug_fleave();
+ /* FIXIT : this cause so many warnings/errors from glib/gstreamer. we should not call it since
+ * gst_element_post_message api takes ownership of the message.
+ */
+ //gst_message_unref( msg );
- return MM_ERROR_NONE;
+ return ret;
}
-static int
-__mmplayer_gst_element_link_bucket(GList* element_bucket) // @
+static gboolean
+__mmplayer_gst_handle_duration(mm_player_t* player, GstMessage* msg)
{
- GList* bucket = element_bucket;
- MMPlayerGstElement* element = NULL;
- MMPlayerGstElement* prv_element = NULL;
- gint successful_link_count = 0;
+ GstFormat format;
+ gint64 bytes = 0;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail(element_bucket, -1);
+ return_val_if_fail(player, FALSE);
+ return_val_if_fail(msg, FALSE);
- prv_element = (MMPlayerGstElement*)bucket->data;
- bucket = bucket->next;
+ gst_message_parse_duration (msg, &format, &bytes);
- for ( ; bucket; bucket = bucket->next )
+ if (MMPLAYER_IS_HTTP_STREAMING(player) && format == GST_FORMAT_BYTES )
{
- element = (MMPlayerGstElement*)bucket->data;
-
- if ( element && element->gst )
- {
- if ( GST_ELEMENT_LINK(GST_ELEMENT(prv_element->gst), GST_ELEMENT(element->gst)) )
- {
- debug_log("linking [%s] to [%s] success\n",
- GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
- GST_ELEMENT_NAME(GST_ELEMENT(element->gst)) );
- successful_link_count ++;
- }
- else
- {
- debug_log("linking [%s] to [%s] failed\n",
- GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
- GST_ELEMENT_NAME(GST_ELEMENT(element->gst)) );
- return -1;
- }
-
- }
-
- prv_element = element;
+ debug_log("data total size of http content: %lld", bytes);
+ player->http_content_size = bytes;
+ }
+ else if (format == GST_FORMAT_TIME)
+ {
+ /* handling audio clip which has vbr. means duration is keep changing */
+ _mmplayer_update_content_attrs (player, ATTR_DURATION );
+ }
+ else
+ {
+ debug_warning("duration is neither BYTES or TIME");
+ return FALSE;
}
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return successful_link_count;
+ return TRUE;
}
-static int
-__mmplayer_gst_element_add_bucket_to_bin(GstBin* bin, GList* element_bucket) // @
-{
- GList* bucket = element_bucket;
- MMPlayerGstElement* element = NULL;
- int successful_add_count = 0;
-
- debug_fenter();
-
- return_val_if_fail(element_bucket, 0);
- return_val_if_fail(bin, 0);
- for ( ; bucket; bucket = bucket->next )
- {
- element = (MMPlayerGstElement*)bucket->data;
-
- if ( element && element->gst )
- {
- if( !gst_bin_add(bin, GST_ELEMENT(element->gst)) )
- {
- debug_log("__mmplayer_gst_element_link_bucket : Adding element [%s] to bin [%s] failed\n",
- GST_ELEMENT_NAME(GST_ELEMENT(element->gst)),
- GST_ELEMENT_NAME(GST_ELEMENT(bin) ) );
- return 0;
- }
- successful_add_count ++;
- }
- }
+static gboolean
+__mmplayer_gst_extract_tag_from_msg(mm_player_t* player, GstMessage* msg) // @
+{
- debug_fleave();
+/* macro for better code readability */
+#define MMPLAYER_UPDATE_TAG_STRING(gsttag, attribute, playertag) \
+if (gst_tag_list_get_string(tag_list, gsttag, &string)) \
+{\
+ if (string != NULL)\
+ {\
+ secure_debug_log ( "update tag string : %s\n", string); \
+ mm_attrs_set_string_by_name(attribute, playertag, string); \
+ g_free(string);\
+ string = NULL;\
+ }\
+}
- return successful_add_count;
+#define MMPLAYER_UPDATE_TAG_IMAGE(gsttag, attribute, playertag) \
+value = gst_tag_list_get_value_index(tag_list, gsttag, index); \
+if (value) \
+{\
+ buffer = gst_value_get_buffer (value); \
+ secure_debug_log ( "update album cover data : %p, size : %d\n", GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer)); \
+ MMPLAYER_FREEIF(player->album_art); \
+ player->album_art = (gchar *)g_malloc(GST_BUFFER_SIZE(buffer)); \
+ if (player->album_art) \
+ { \
+ memcpy(player->album_art, GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer)); \
+ mm_attrs_set_data_by_name(attribute, playertag, (void *)player->album_art, GST_BUFFER_SIZE(buffer)); \
+ if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player)) \
+ { \
+ msg_param.data = (void *)player->album_art; \
+ msg_param.size = GST_BUFFER_SIZE(buffer); \
+ MMPLAYER_POST_MSG (player, MM_MESSAGE_IMAGE_BUFFER, &msg_param); \
+ secure_debug_log ( "post message image buffer data : %p, size : %d\n", GST_BUFFER_DATA(buffer), GST_BUFFER_SIZE(buffer)); \
+ } \
+ } \
}
+#define MMPLAYER_UPDATE_TAG_UINT(gsttag, attribute, playertag) \
+if (gst_tag_list_get_uint(tag_list, gsttag, &v_uint))\
+{\
+ if(v_uint)\
+ {\
+ if (!strncmp(gsttag, GST_TAG_BITRATE, strlen(GST_TAG_BITRATE))) \
+ {\
+ if (player->updated_bitrate_count == 0) \
+ mm_attrs_set_int_by_name(attribute, "content_audio_bitrate", v_uint); \
+ if (player->updated_bitrate_count<MM_PLAYER_STREAM_COUNT_MAX) \
+ {\
+ player->bitrate[player->updated_bitrate_count] = v_uint;\
+ player->total_bitrate += player->bitrate[player->updated_maximum_bitrate_count]; \
+ player->updated_bitrate_count++; \
+ mm_attrs_set_int_by_name(attribute, playertag, player->total_bitrate);\
+ secure_debug_log ( "update bitrate %d[bps] of stream #%d.\n", v_uint, player->updated_bitrate_count);\
+ }\
+ }\
+ else if (!strncmp(gsttag, GST_TAG_MAXIMUM_BITRATE, strlen(GST_TAG_MAXIMUM_BITRATE))) \
+ {\
+ if (player->updated_maximum_bitrate_count<MM_PLAYER_STREAM_COUNT_MAX) \
+ {\
+ player->maximum_bitrate[player->updated_maximum_bitrate_count] = v_uint;\
+ player->total_maximum_bitrate += player->maximum_bitrate[player->updated_maximum_bitrate_count]; \
+ player->updated_maximum_bitrate_count++; \
+ mm_attrs_set_int_by_name(attribute, playertag, player->total_maximum_bitrate); \
+ secure_debug_log ( "update maximum bitrate %d[bps] of stream #%d\n", v_uint, player->updated_maximum_bitrate_count);\
+ }\
+ }\
+ else\
+ {\
+ mm_attrs_set_int_by_name(attribute, playertag, v_uint); \
+ }\
+ v_uint = 0;\
+ }\
+}
+#define MMPLAYER_UPDATE_TAG_DATE(gsttag, attribute, playertag) \
+if (gst_tag_list_get_date(tag_list, gsttag, &date))\
+{\
+ if (date != NULL)\
+ {\
+ string = g_strdup_printf("%d", g_date_get_year(date));\
+ mm_attrs_set_string_by_name(attribute, playertag, string);\
+ secure_debug_log ( "metainfo year : %s\n", string);\
+ MMPLAYER_FREEIF(string);\
+ g_date_free(date);\
+ }\
+}
-/**
- * This function is to create audio pipeline for playing.
- *
- * @param player [in] handle of player
- *
- * @return This function returns zero on success.
- * @remark
- * @see __mmplayer_gst_create_midi_pipeline, __mmplayer_gst_create_video_pipeline
- */
-#define MMPLAYER_CREATEONLY_ELEMENT(x_bin, x_id, x_factory, x_name) \
-x_bin[x_id].id = x_id;\
-x_bin[x_id].gst = gst_element_factory_make(x_factory, x_name);\
-if ( ! x_bin[x_id].gst )\
+#define MMPLAYER_UPDATE_TAG_UINT64(gsttag, attribute, playertag) \
+if(gst_tag_list_get_uint64(tag_list, gsttag, &v_uint64))\
{\
- debug_critical("failed to create %s \n", x_factory);\
- goto ERROR;\
-}\
+ if(v_uint64)\
+ {\
+ /* FIXIT : don't know how to store date */\
+ g_assert(1);\
+ v_uint64 = 0;\
+ }\
+}
-/* macro for code readability. just for sinkbin-creation functions */
-#define MMPLAYER_CREATE_ELEMENT(x_bin, x_id, x_factory, x_name, x_add_bucket) \
-do \
-{ \
- x_bin[x_id].id = x_id;\
- x_bin[x_id].gst = gst_element_factory_make(x_factory, x_name);\
- if ( ! x_bin[x_id].gst )\
+#define MMPLAYER_UPDATE_TAG_DOUBLE(gsttag, attribute, playertag) \
+if(gst_tag_list_get_double(tag_list, gsttag, &v_double))\
+{\
+ if(v_double)\
{\
- debug_critical("failed to create %s \n", x_factory);\
- goto ERROR;\
+ /* FIXIT : don't know how to store date */\
+ g_assert(1);\
+ v_double = 0;\
}\
- if ( x_add_bucket )\
- element_bucket = g_list_append(element_bucket, &x_bin[x_id]);\
-} while(0);
+}
+ /* function start */
+ GstTagList* tag_list = NULL;
-/**
- * AUDIO PIPELINE
- * - Local playback : audioconvert !volume ! capsfilter ! dnse ! audiosink
- * - Streaming : audioconvert !volume ! audiosink
- * - PCM extraction : audioconvert ! audioresample ! capsfilter ! fakesink
- */
-static int
-__mmplayer_gst_create_audio_pipeline(mm_player_t* player)
-{
- MMPlayerGstElement* first_element = NULL;
- MMPlayerGstElement* audiobin = NULL;
MMHandleType attrs = 0;
- GstPad *pad = NULL;
- GstPad *ghostpad = NULL;
- GList* element_bucket = NULL;
- char *device_name = NULL;
- gboolean link_audio_sink_now = TRUE;
- int i =0;
- debug_fenter();
+ char *string = NULL;
+ guint v_uint = 0;
+ GDate *date = NULL;
+ /* album cover */
+ GstBuffer *buffer = NULL;
+ gint index = 0;
+ const GValue *value;
+ MMMessageParamType msg_param = {0, };
- return_val_if_fail( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ /* currently not used. but those are needed for above macro */
+ //guint64 v_uint64 = 0;
+ //gdouble v_double = 0;
- /* alloc handles */
- audiobin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_A_NUM);
- audiobin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_A_NUM);
- if ( ! audiobin )
- {
- debug_error("failed to allocate memory for audiobin\n");
- return MM_ERROR_PLAYER_NO_FREE_SPACE;
- }
+ return_val_if_fail( player && msg, FALSE );
attrs = MMPLAYER_GET_ATTRS(player);
- /* create bin */
- audiobin[MMPLAYER_A_BIN].id = MMPLAYER_A_BIN;
- audiobin[MMPLAYER_A_BIN].gst = gst_bin_new("audiobin");
- if ( !audiobin[MMPLAYER_A_BIN].gst )
- {
- debug_critical("failed to create audiobin\n");
- goto ERROR;
- }
-
- /* take it */
- player->pipeline->audiobin = audiobin;
-
- player->is_sound_extraction = __mmplayer_can_extract_pcm(player);
-
- /* Adding audiotp plugin for reverse trickplay feature */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_TP, "audiotp", "audiotrickplay", TRUE);
+ return_val_if_fail( attrs, FALSE );
- /* converter */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV, "audioconvert", "audioconverter", TRUE);
+ /* get tag list from gst message */
+ gst_message_parse_tag(msg, &tag_list);
- if ( ! player->is_sound_extraction )
- {
- /* for logical volume control */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_VOL, "volume", "volume", TRUE);
- g_object_set(G_OBJECT (audiobin[MMPLAYER_A_VOL].gst), "volume", player->sound.volume, NULL);
+ /* store tags to player attributes */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_TITLE, attrs, "tag_title");
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_TITLE_SORTNAME, ?, ?); */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ARTIST, attrs, "tag_artist");
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ARTIST_SORTNAME, ?, ?); */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ALBUM, attrs, "tag_album");
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ALBUM_SORTNAME, ?, ?); */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COMPOSER, attrs, "tag_author");
+ MMPLAYER_UPDATE_TAG_DATE(GST_TAG_DATE, attrs, "tag_date");
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_GENRE, attrs, "tag_genre");
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COMMENT, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_EXTENDED_COMMENT, ?, ?); */
+ MMPLAYER_UPDATE_TAG_UINT(GST_TAG_TRACK_NUMBER, attrs, "tag_track_num");
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_TRACK_COUNT, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_ALBUM_VOLUME_NUMBER, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_ALBUM_VOLUME_COUNT, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LOCATION, ?, ?); */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_DESCRIPTION, attrs, "tag_description");
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_VERSION, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ISRC, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ORGANIZATION, ?, ?); */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COPYRIGHT, attrs, "tag_copyright");
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_COPYRIGHT_URI, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_CONTACT, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LICENSE, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LICENSE_URI, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_PERFORMER, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_UINT64(GST_TAG_DURATION, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_CODEC, ?, ?); */
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_VIDEO_CODEC, attrs, "content_video_codec");
+ MMPLAYER_UPDATE_TAG_STRING(GST_TAG_AUDIO_CODEC, attrs, "content_audio_codec");
+ MMPLAYER_UPDATE_TAG_UINT(GST_TAG_BITRATE, attrs, "content_bitrate");
+ MMPLAYER_UPDATE_TAG_UINT(GST_TAG_MAXIMUM_BITRATE, attrs, "content_max_bitrate");
+ MMPLAYER_UPDATE_TAG_IMAGE(GST_TAG_IMAGE, attrs, "tag_album_cover");
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_NOMINAL_BITRATE, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_MINIMUM_BITRATE, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_SERIAL, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_ENCODER, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_UINT(GST_TAG_ENCODER_VERSION, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_TRACK_GAIN, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_TRACK_PEAK, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_ALBUM_GAIN, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_ALBUM_PEAK, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_REFERENCE_LEVEL, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_STRING(GST_TAG_LANGUAGE_CODE, ?, ?); */
+ /* MMPLAYER_UPDATE_TAG_DOUBLE(GST_TAG_BEATS_PER_MINUTE, ?, ?); */
- if (player->sound.mute)
- {
- debug_log("mute enabled\n");
- g_object_set(G_OBJECT (audiobin[MMPLAYER_A_VOL].gst), "mute", player->sound.mute, NULL);
- }
+ if ( mmf_attrs_commit ( attrs ) )
+ debug_error("failed to commit.\n");
- /* NOTE : streaming doesn't need capsfilter and sound effect*/
- if ( ! MMPLAYER_IS_RTSP_STREAMING( player ) )
- {
- GstCaps* caps = NULL;
+ gst_tag_list_free(tag_list);
- /*capsfilter */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_DEFAULT, "capsfilter", "audiocapsfilter", TRUE);
+ return TRUE;
+}
- caps = gst_caps_from_string( "audio/x-raw-int, "
- "endianness = (int) LITTLE_ENDIAN, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16" );
+static void
+__mmplayer_gst_rtp_no_more_pads (GstElement *element, gpointer data) // @
+{
+ mm_player_t* player = (mm_player_t*) data;
- g_object_set (GST_ELEMENT(audiobin[MMPLAYER_A_CAPS_DEFAULT].gst), "caps", caps, NULL );
+ MMPLAYER_FENTER();
- gst_caps_unref( caps );
+ /* NOTE : we can remove fakesink here if there's no rtp_dynamic_pad. because whenever
+ * we connect autoplugging element to the pad which is just added to rtspsrc, we increase
+ * num_dynamic_pad. and this is no-more-pad situation which means mo more pad will be added.
+ * So we can say this. if num_dynamic_pad is zero, it must be one of followings
- /* audio filter. if enabled */
- if ( PLAYER_INI()->use_audio_filter_preset || PLAYER_INI()->use_audio_filter_custom )
- {
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER, "soundalive", "audiofilter", TRUE);
- }
- }
- /* create audio sink */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, PLAYER_INI()->name_of_audiosink,
- "audiosink", link_audio_sink_now);
-
- /* sync on */
- if (MMPLAYER_IS_RTSP_STREAMING (player) )
- g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "sync", FALSE, NULL); /* sync off */
- else
- g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "sync", TRUE, NULL); /* sync on */
-
- /* qos on */
- g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "qos", TRUE, NULL); /* qos on */
-
- /* FIXIT : using system clock. isn't there another way? */
- g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "provide-clock", PLAYER_INI()->provide_clock, NULL);
-
- __mmplayer_add_sink( player, audiobin[MMPLAYER_A_SINK].gst );
-
- if(player->audio_buffer_cb)
- {
- g_object_set(audiobin[MMPLAYER_A_SINK].gst, "audio-handle", player->audio_buffer_cb_user_param, NULL);
- g_object_set(audiobin[MMPLAYER_A_SINK].gst, "audio-callback", player->audio_buffer_cb, NULL);
- }
+ * [1] audio and video will be dumped with filesink.
+ * [2] autoplugging is done by just using pad caps.
+ * [3] typefinding has happend in audio but audiosink is created already before no-more-pad signal
+ * and the video will be dumped via filesink.
+ */
+ if ( player->num_dynamic_pad == 0 )
+ {
+ debug_log("it seems pad caps is directely used for autoplugging. removing fakesink now\n");
- if ( g_strrstr(PLAYER_INI()->name_of_audiosink, "avsysaudiosink") )
+ if ( ! __mmplayer_gst_remove_fakesink( player,
+ &player->pipeline->mainbin[MMPLAYER_M_SRC_FAKESINK]) )
{
- gint volume_type = 0;
- gint audio_route = 0;
- gint sound_priority = FALSE;
- gint is_spk_out_only = 0;
-
- /* set volume table
- * It should be set after player creation through attribute.
- * But, it can not be changed during playing.
+ /* NOTE : __mmplayer_pipeline_complete() can be called several time. because
+ * signaling mechanism ( pad-added, no-more-pad, new-decoded-pad ) from various
+ * source element are not same. To overcome this situation, this function will called
+ * several places and several times. Therefore, this is not an error case.
*/
- mm_attrs_get_int_by_name(attrs, "sound_volume_type", &volume_type);
- mm_attrs_get_int_by_name(attrs, "sound_route", &audio_route);
- mm_attrs_get_int_by_name(attrs, "sound_priority", &sound_priority);
- mm_attrs_get_int_by_name(attrs, "sound_spk_out_only", &is_spk_out_only);
-
- g_object_set(audiobin[MMPLAYER_A_SINK].gst,
- "volumetype", volume_type,
- "audio-route", audio_route,
- "priority", sound_priority,
- "user-route", is_spk_out_only,
- NULL);
-
- debug_log("audiosink property status...volume type:%d, route:%d, priority=%d, user-route=%d\n",
- volume_type, audio_route, sound_priority, is_spk_out_only);
+ return;
}
-
- /* Antishock can be enabled when player is resumed by soundCM.
- * But, it's not used in MMS, setting and etc.
- * Because, player start seems like late.
- */
- __mmplayer_set_antishock( player , FALSE );
}
- else // pcm extraction only and no sound output
- {
- int dst_samplerate = 0;
- int dst_channels = 0;
- int dst_depth = 0;
- char *caps_type = NULL;
- GstCaps* caps = NULL;
-
- /* resampler */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_RESAMPLER, "audioresample", "resampler", TRUE);
-
- /* get conf. values */
- mm_attrs_multiple_get(player->attrs,
- NULL,
- "pcm_extraction_samplerate", &dst_samplerate,
- "pcm_extraction_channels", &dst_channels,
- "pcm_extraction_depth", &dst_depth,
- NULL);
- /* capsfilter */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_DEFAULT, "capsfilter", "audiocapsfilter", TRUE);
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, dst_samplerate,
- "channels", G_TYPE_INT, dst_channels,
- "depth", G_TYPE_INT, dst_depth,
- NULL);
+ /* create dot before error-return. for debugging */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-no-more-pad" );
- caps_type = gst_caps_to_string(caps);
- debug_log("resampler new caps : %s\n", caps_type);
+ player->no_more_pad = TRUE;
- g_object_set (GST_ELEMENT(audiobin[MMPLAYER_A_CAPS_DEFAULT].gst), "caps", caps, NULL );
+ MMPLAYER_FLEAVE();
+}
- /* clean */
- gst_caps_unref( caps );
- MMPLAYER_FREEIF( caps_type );
+static gboolean
+__mmplayer_gst_remove_fakesink(mm_player_t* player, MMPlayerGstElement* fakesink) // @
+{
+ GstElement* parent = NULL;
- /* fake sink */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, "fakesink", "fakesink", TRUE);
+ return_val_if_fail(player && player->pipeline, FALSE);
- /* set sync */
- g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "sync", FALSE, NULL);
+ /* if we have no fakesink. this meas we are using decodebin2 which doesn'
+ t need to add extra fakesink */
+ return_val_if_fail(fakesink, TRUE);
- __mmplayer_add_sink( player, audiobin[MMPLAYER_A_SINK].gst );
- }
+ /* lock */
+ g_mutex_lock( player->fsink_lock );
- /* adding created elements to bin */
- debug_log("adding created elements to bin\n");
- if( !__mmplayer_gst_element_add_bucket_to_bin( GST_BIN(audiobin[MMPLAYER_A_BIN].gst), element_bucket ))
+ if ( ! fakesink->gst )
{
- debug_error("failed to add elements\n");
goto ERROR;
}
- /* linking elements in the bucket by added order. */
- debug_log("Linking elements in the bucket by added order.\n");
- if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ /* get parent of fakesink */
+ parent = (GstElement*)gst_object_get_parent( (GstObject*)fakesink->gst );
+ if ( ! parent )
{
- debug_error("failed to link elements\n");
+ debug_log("fakesink already removed\n");
goto ERROR;
}
- /* get first element's sinkpad for creating ghostpad */
- first_element = (MMPlayerGstElement *)element_bucket->data;
+ gst_element_set_locked_state( fakesink->gst, TRUE );
- pad = gst_element_get_static_pad(GST_ELEMENT(first_element->gst), "sink");
- if ( ! pad )
+ /* setting the state to NULL never returns async
+ * so no need to wait for completion of state transiton
+ */
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state (fakesink->gst, GST_STATE_NULL) )
{
- debug_error("failed to get pad from first element of audiobin\n");
- goto ERROR;
- }
+ debug_error("fakesink state change failure!\n");
- ghostpad = gst_ghost_pad_new("sink", pad);
- if ( ! ghostpad )
- {
- debug_error("failed to create ghostpad\n");
- goto ERROR;
+ /* FIXIT : should I return here? or try to proceed to next? */
+ /* return FALSE; */
}
- if ( FALSE == gst_element_add_pad(audiobin[MMPLAYER_A_BIN].gst, ghostpad) )
+ /* remove fakesink from it's parent */
+ if ( ! gst_bin_remove( GST_BIN( parent ), fakesink->gst ) )
{
- debug_error("failed to add ghostpad to audiobin\n");
- goto ERROR;
- }
+ debug_error("failed to remove fakesink\n");
- gst_object_unref(pad);
+ gst_object_unref( parent );
- if ( !player->bypass_sound_effect && (PLAYER_INI()->use_audio_filter_preset || PLAYER_INI()->use_audio_filter_custom) )
- {
- if ( player->audio_filter_info.filter_type == MM_AUDIO_FILTER_TYPE_PRESET )
- {
- if (!_mmplayer_sound_filter_preset_apply(player, player->audio_filter_info.preset))
- {
- debug_msg("apply sound effect(preset:%d) setting success\n",player->audio_filter_info.preset);
- }
- }
- else if ( player->audio_filter_info.filter_type == MM_AUDIO_FILTER_TYPE_CUSTOM )
- {
- if (!_mmplayer_sound_filter_custom_apply(player))
- {
- debug_msg("apply sound effect(custom) setting success\n");
- }
- }
+ goto ERROR;
}
- /* done. free allocated variables */
- MMPLAYER_FREEIF( device_name );
- g_list_free(element_bucket);
+ gst_object_unref( parent );
- mm_attrs_set_int_by_name(attrs, "content_audio_found", TRUE);
- if ( mmf_attrs_commit ( attrs ) ) /* return -1 if error */
- debug_error("failed to commit attribute ""content_audio_found"".\n");
+ debug_log("state-holder removed\n");
- debug_fleave();
+ gst_element_set_locked_state( fakesink->gst, FALSE );
- return MM_ERROR_NONE;
+ g_mutex_unlock( player->fsink_lock );
+ return TRUE;
ERROR:
+ if ( fakesink->gst )
+ {
+ gst_element_set_locked_state( fakesink->gst, FALSE );
+ }
- debug_log("ERROR : releasing audiobin\n");
+ g_mutex_unlock( player->fsink_lock );
+ return FALSE;
+}
- MMPLAYER_FREEIF( device_name );
- if ( pad )
- gst_object_unref(GST_OBJECT(pad));
+static void
+__mmplayer_gst_rtp_dynamic_pad (GstElement *element, GstPad *pad, gpointer data) // @
+{
+ GstPad *sinkpad = NULL;
+ GstCaps* caps = NULL;
+ GstElement* new_element = NULL;
+ GstStructure* str = NULL;
+ const gchar* name = NULL;
- if ( ghostpad )
- gst_object_unref(GST_OBJECT(ghostpad));
+ mm_player_t* player = (mm_player_t*) data;
- g_list_free( element_bucket );
+ MMPLAYER_FENTER();
+ return_if_fail( element && pad );
+ return_if_fail( player &&
+ player->pipeline &&
+ player->pipeline->mainbin );
- /* release element which are not added to bin */
- for ( i = 1; i < MMPLAYER_A_NUM; i++ ) /* NOTE : skip bin */
- {
- if ( audiobin[i].gst )
- {
- GstObject* parent = NULL;
- parent = gst_element_get_parent( audiobin[i].gst );
- if ( !parent )
- {
- gst_object_unref(GST_OBJECT(audiobin[i].gst));
- audiobin[i].gst = NULL;
- }
- else
- {
- gst_object_unref(GST_OBJECT(parent));
- }
- }
- }
+ /* payload type is recognizable. increase num_dynamic and wait for sinkbin creation.
+ * num_dynamic_pad will decreased after creating a sinkbin.
+ */
+ player->num_dynamic_pad++;
+ debug_log("stream count inc : %d\n", player->num_dynamic_pad);
- /* release audiobin with it's childs */
- if ( audiobin[MMPLAYER_A_BIN].gst )
+ /* perform autoplugging if dump is disabled */
+ if ( player->ini.rtsp_do_typefinding )
{
- gst_object_unref(GST_OBJECT(audiobin[MMPLAYER_A_BIN].gst));
- }
+ /* create typefind */
+ new_element = gst_element_factory_make( "typefind", NULL );
+ if ( ! new_element )
+ {
+ debug_error("failed to create typefind\n");
+ goto ERROR;
+ }
- MMPLAYER_FREEIF( audiobin );
+ MMPLAYER_SIGNAL_CONNECT( player,
+ G_OBJECT(new_element),
+ MM_PLAYER_SIGNAL_TYPE_AUTOPLUG,
+ "have-type",
+ G_CALLBACK(__mmplayer_typefind_have_type),
+ (gpointer)player);
- player->pipeline->audiobin = NULL;
+ /* FIXIT : try to remove it */
+ player->have_dynamic_pad = FALSE;
+ }
+ else /* NOTE : use pad's caps directely. if enabled. what I am assuming is there's no elemnt has dynamic pad */
+ {
+ debug_log("using pad caps to autopluging instead of doing typefind\n");
- return MM_ERROR_PLAYER_INTERNAL;
-}
+ caps = gst_pad_get_caps( pad );
-static gboolean
-__mmplayer_audio_stream_probe (GstPad *pad, GstBuffer *buffer, gpointer u_data)
-{
- mm_player_t* player = (mm_player_t*) u_data;
- gint size;
- guint8 *data;
+ MMPLAYER_CHECK_NULL( caps );
- data = GST_BUFFER_DATA(buffer);
- size = GST_BUFFER_SIZE(buffer);
+ /* clear previous result*/
+ player->have_dynamic_pad = FALSE;
- if (player->audio_stream_cb && size && data)
- player->audio_stream_cb((void *)data, size, player->audio_stream_cb_user_param);
+ str = gst_caps_get_structure(caps, 0);
- return TRUE;
-}
+ if ( ! str )
+ {
+ debug_error ("cannot get structure from capse.\n");
+ goto ERROR;
+ }
-gboolean
-__mmplayer_ahs_appsrc_probe (GstPad *pad, GstBuffer *buffer, gpointer u_data)
-{
- mm_player_t* player = (mm_player_t*) u_data;
+ name = gst_structure_get_name (str);
+ if ( ! name )
+ {
+ debug_error ("cannot get mimetype from structure.\n");
+ goto ERROR;
+ }
- //g_print ("appsrc buf : timestamp = %"GST_TIME_FORMAT", duration = %"GST_TIME_FORMAT"\n",
- // GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)), GST_TIME_ARGS (GST_BUFFER_DURATION(buffer)));
+ if (strstr(name, "video"))
+ {
+ gint stype = 0;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &stype);
- if (ahs_check_allow_cache (player->ahs_player))
- {
- ahs_store_media_presentation (player->ahs_player, GST_BUFFER_DATA (buffer), GST_BUFFER_SIZE (buffer));
- }
- return TRUE;
-}
+ if (stype == MM_DISPLAY_SURFACE_NULL)
+ {
+ if (player->v_stream_caps)
+ {
+ gst_caps_unref(player->v_stream_caps);
+ player->v_stream_caps = NULL;
+ }
-/**
- * This function is to create video pipeline.
- *
- * @param player [in] handle of player
- * caps [in] src caps of decoder
- *
- * @return This function returns zero on success.
- * @remark
- * @see __mmplayer_gst_create_audio_pipeline, __mmplayer_gst_create_midi_pipeline
- */
-/**
- * VIDEO PIPELINE
- * - x surface (arm/x86) : xvimagesink
- * - evas surface (arm) : evaspixmapsink
- * fimcconvert ! evasimagesink
- * - evas surface (x86) : videoconvertor ! evasimagesink
- * - multi-surface (arm) : tee name=tee ! xvimagesink tee. ! evaspixmapsink
- */
-static int
-__mmplayer_gst_create_video_pipeline(mm_player_t* player, GstCaps* caps)
-{
- GstPad *pad = NULL;
- MMHandleType attrs;
- GList*element_bucket = NULL;
- MMPlayerGstElement* first_element = NULL;
- MMPlayerGstElement* videobin = NULL;
- gchar* vconv_factory = NULL;
- char *err_name = NULL;
- gboolean use_multi_surface = FALSE;
- char *videosink_element = NULL;
- char *videosink_element_ext = NULL;
+ new_element = gst_element_factory_make("fakesink", NULL);
+ player->num_dynamic_pad--;
+ goto NEW_ELEMENT;
+ }
+ }
- debug_fenter();
+ /* clear previous result*/
+ player->have_dynamic_pad = FALSE;
- return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ if ( ! __mmplayer_try_to_plug( player, pad, caps ) )
+ {
+ debug_error("failed to autoplug for caps");
+ goto ERROR;
+ }
- /* alloc handles */
- videobin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_V_NUM);
- if ( !videobin )
- return MM_ERROR_PLAYER_NO_FREE_SPACE;
+ /* check if there's dynamic pad*/
+ if( player->have_dynamic_pad )
+ {
+ debug_error("using pad caps assums there's no dynamic pad !\n");
+ debug_error("try with enalbing rtsp_do_typefinding\n");
+ goto ERROR;
+ }
- player->pipeline->videobin = videobin;
+ gst_caps_unref( caps );
+ caps = NULL;
+ }
- attrs = MMPLAYER_GET_ATTRS(player);
+ NEW_ELEMENT:
- /* create bin */
- videobin[MMPLAYER_V_BIN].id = MMPLAYER_V_BIN;
- videobin[MMPLAYER_V_BIN].gst = gst_bin_new("videobin");
- if ( !videobin[MMPLAYER_V_BIN].gst )
+ /* excute new_element if created*/
+ if ( new_element )
{
- debug_critical("failed to create audiobin\n");
- goto ERROR;
- }
-
- mm_player_get_attribute(player, &err_name, "display_surface_use_multi", &use_multi_surface, NULL);
- player->use_multi_surface = use_multi_surface;
-
- if( player->use_video_stream ) // video stream callack, so send raw video data to application
- {
- GstStructure *str = NULL;
- guint32 fourcc = 0;
- gint ret = 0;
+ debug_log("adding new element to pipeline\n");
- debug_log("using memsink\n");
+ /* set state to READY before add to bin */
+ MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_READY );
- /* first, create colorspace convert */
- if (player->is_nv12_tiled)
- {
- vconv_factory = "fimcconvert";
- }
- else // get video converter from player ini file
+ /* add new element to the pipeline */
+ if ( FALSE == gst_bin_add( GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), new_element) )
{
- if (strlen(PLAYER_INI()->name_of_video_converter) > 0)
- {
- vconv_factory = PLAYER_INI()->name_of_video_converter;
- }
+ debug_error("failed to add autoplug element to bin\n");
+ goto ERROR;
}
- if (vconv_factory)
+ /* get pad from element */
+ sinkpad = gst_element_get_static_pad ( GST_ELEMENT(new_element), "sink" );
+ if ( !sinkpad )
{
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CONV, vconv_factory, "video converter", TRUE);
+ debug_error("failed to get sinkpad from autoplug element\n");
+ goto ERROR;
}
-
- /* then, create video scale to resize if needed */
- str = gst_caps_get_structure (caps, 0);
- if ( ! str )
+ /* link it */
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
{
- debug_error("cannot get structure\n");
+ debug_error("failed to link autoplug element\n");
goto ERROR;
}
- MMPLAYER_LOG_GST_CAPS_TYPE(caps);
+ gst_object_unref (sinkpad);
+ sinkpad = NULL;
- ret = gst_structure_get_fourcc (str, "format", &fourcc);
+ /* run. setting PLAYING here since streamming source is live source */
+ MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_PLAYING );
+ }
- if ( !ret )
- debug_log("not fixed format at this point, and not consider this case\n")
+ MMPLAYER_FLEAVE();
- /* NOTE : if the width of I420 format is not multiple of 8, it should be resize before colorspace conversion.
- * so, video scale is required for this case only.
- */
- if ( GST_MAKE_FOURCC ('I', '4', '2', '0') == fourcc )
- {
- gint width = 0; //width of video
- gint height = 0; //height of video
- gint framerate_n = 0; //numerator of frame rate
- gint framerate_d = 0; //denominator of frame rate
- GstCaps* video_caps = NULL;
- GValue *fps = NULL;
+ return;
- /* video scale */
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SCALE, "videoscale", "videoscale", TRUE);
+STATE_CHANGE_FAILED:
+ERROR:
+ /* FIXIT : take care if new_element has already added to pipeline */
+ if ( new_element )
+ gst_object_unref(GST_OBJECT(new_element));
- /*to limit width as multiple of 8 */
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CAPS, "capsfilter", "videocapsfilter", TRUE);
+ if ( sinkpad )
+ gst_object_unref(GST_OBJECT(sinkpad));
- /* get video stream caps parsed by demuxer */
- str = gst_caps_get_structure (player->v_stream_caps, 0);
- if ( ! str )
- {
- debug_error("cannot get structure\n");
- goto ERROR;
- }
+ if ( caps )
+ gst_object_unref(GST_OBJECT(caps));
- /* check the width if it's a multiple of 8 or not */
- ret = gst_structure_get_int (str, "width", &width);
- if ( ! ret )
- {
- debug_error("cannot get width\n");
- goto ERROR;
- }
- width = GST_ROUND_UP_8(width);
+ /* FIXIT : how to inform this error to MSL ????? */
+ /* FIXIT : I think we'd better to use g_idle_add() to destroy pipeline and
+ * then post an error to application
+ */
+}
- ret = gst_structure_get_int(str, "height", &height);
- if ( ! ret )
- {
- debug_error("cannot get height\n");
- goto ERROR;
- }
- fps = gst_structure_get_value (str, "framerate");
- if ( ! fps )
- {
- debug_error("cannot get fps\n");
- goto ERROR;
- }
- framerate_n = gst_value_get_fraction_numerator (fps);
- framerate_d = gst_value_get_fraction_denominator (fps);
- video_caps = gst_caps_new_simple( "video/x-raw-yuv",
- "format", GST_TYPE_FOURCC, GST_MAKE_FOURCC ('I', '4', '2', '0'),
- "width", G_TYPE_INT, width,
- "height", G_TYPE_INT, height,
- "framerate", GST_TYPE_FRACTION, framerate_n, framerate_d,
- NULL);
+/* FIXIT : check indent */
+static void
+__mmplayer_gst_wfd_dynamic_pad (GstElement *element, GstPad *pad, gpointer data) // @
+{
+ GstPad *sinkpad = NULL;
+ GstCaps* caps = NULL;
+ GstElement* new_element = NULL;
+ enum MainElementID element_id = MMPLAYER_M_NUM;
+
+ mm_player_t* player = (mm_player_t*) data;
+
+ MMPLAYER_FENTER();
+
+ return_if_fail( element && pad );
+ return_if_fail( player &&
+ player->pipeline &&
+ player->pipeline->mainbin );
+
+ debug_log("stream count inc : %d\n", player->num_dynamic_pad);
+
+ {
+ debug_log("using pad caps to autopluging instead of doing typefind\n");
+ caps = gst_pad_get_caps( pad );
+ MMPLAYER_CHECK_NULL( caps );
+ /* clear previous result*/
+ player->have_dynamic_pad = FALSE;
+ new_element = gst_element_factory_make("rtpmp2tdepay", "wfd_rtp_depay");
+ if ( !new_element )
+ {
+ debug_error ( "failed to create wfd rtp depay element\n" );
+ goto ERROR;
+ }
+ MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_READY );
+ /* add new element to the pipeline */
+ if ( FALSE == gst_bin_add( GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), new_element) )
+ {
+ debug_log("failed to add autoplug element to bin\n");
+ goto ERROR;
+ }
+ /* get pad from element */
+ sinkpad = gst_element_get_static_pad ( GST_ELEMENT(new_element), "sink" );
+ if ( !sinkpad )
+ {
+ debug_log("failed to get sinkpad from autoplug element\n");
+ goto ERROR;
+ }
+ /* link it */
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
+ {
+ debug_log("failed to link autoplug element\n");
+ goto ERROR;
+ }
+ gst_object_unref (sinkpad);
+ sinkpad = NULL;
+ pad = gst_element_get_static_pad ( GST_ELEMENT(new_element), "src" );
+ caps = gst_pad_get_caps( pad );
+ MMPLAYER_CHECK_NULL( caps );
+ MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_PLAYING );
+ /* create typefind */
+ new_element = gst_element_factory_make( "typefind", NULL );
+ if ( ! new_element )
+ {
+ debug_log("failed to create typefind\n");
+ goto ERROR;
+ }
+
+ MMPLAYER_SIGNAL_CONNECT( player,
+ G_OBJECT(new_element),
+ MM_PLAYER_SIGNAL_TYPE_AUTOPLUG,
+ "have-type",
+ G_CALLBACK(__mmplayer_typefind_have_type),
+ (gpointer)player);
+
+ player->have_dynamic_pad = FALSE;
+ }
+
+ /* excute new_element if created*/
+ if ( new_element )
+ {
+ debug_log("adding new element to pipeline\n");
+
+ /* set state to READY before add to bin */
+ MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_READY );
+
+ /* add new element to the pipeline */
+ if ( FALSE == gst_bin_add( GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), new_element) )
+ {
+ debug_log("failed to add autoplug element to bin\n");
+ goto ERROR;
+ }
+
+ /* get pad from element */
+ sinkpad = gst_element_get_static_pad ( GST_ELEMENT(new_element), "sink" );
+ if ( !sinkpad )
+ {
+ debug_log("failed to get sinkpad from autoplug element\n");
+ goto ERROR;
+ }
+
+ /* link it */
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
+ {
+ debug_log("failed to link autoplug element\n");
+ goto ERROR;
+ }
+
+ gst_object_unref (sinkpad);
+ sinkpad = NULL;
+
+ /* run. setting PLAYING here since streamming source is live source */
+ MMPLAYER_ELEMENT_SET_STATE( new_element, GST_STATE_PLAYING );
+ }
+
+ /* store handle to futher manipulation */
+ player->pipeline->mainbin[element_id].id = element_id;
+ player->pipeline->mainbin[element_id].gst = new_element;
+
+ MMPLAYER_FLEAVE();
+
+ return;
- g_object_set (GST_ELEMENT(videobin[MMPLAYER_V_CAPS].gst), "caps", video_caps, NULL );
+STATE_CHANGE_FAILED:
+ERROR:
+ /* FIXIT : take care if new_element has already added to pipeline */
+ if ( new_element )
+ gst_object_unref(GST_OBJECT(new_element));
- gst_caps_unref( video_caps );
- }
+ if ( sinkpad )
+ gst_object_unref(GST_OBJECT(sinkpad));
- /* finally, create video sink. its oupput should be BGRX8888 for application like cario surface. */
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, "avsysmemsink", "videosink", TRUE);
+ if ( caps )
+ gst_object_unref(GST_OBJECT(caps));
- MMPLAYER_SIGNAL_CONNECT( player,
- videobin[MMPLAYER_V_SINK].gst,
- "video-stream",
- G_CALLBACK(__mmplayer_videostream_cb),
- player );
+ /* FIXIT : how to inform this error to MSL ????? */
+ /* FIXIT : I think we'd better to use g_idle_add() to destroy pipeline and
+ * then post an error to application
+ */
+}
+
+static void
+__mmplayer_gst_selector_blocked (GstPad* pad, gboolean blocked, gpointer data)
+{
+ debug_log ("pad blocked callback, blocked: %d", blocked);
+}
+
+static void
+__mmplayer_gst_decode_pad_added (GstElement *elem, GstPad *pad, gpointer data)
+{
+ mm_player_t* player = NULL;
+ GstElement* pipeline = NULL;
+ GstElement* selector = NULL;
+ GstElement* fakesink = NULL;
+ GstCaps* caps = NULL;
+ GstStructure* str = NULL;
+ const gchar* name = NULL;
+ GstPad* sinkpad = NULL;
+ GstPad* srcpad = NULL;
+ gboolean first_track = FALSE;
+
+ enum MainElementID elemId = MMPLAYER_M_NUM;
+
+ /* check handles */
+ player = (mm_player_t*)data;
+
+ return_if_fail (elem && pad);
+ return_if_fail (player && player->pipeline && player->pipeline->mainbin);
+
+ //debug_log ("pad-added signal handling\n");
+
+ pipeline = player->pipeline->mainbin[MMPLAYER_M_PIPE].gst;
+
+ /* get mimetype from caps */
+ caps = gst_pad_get_caps (pad);
+ if ( !caps )
+ {
+ debug_error ("cannot get caps from pad.\n");
+ goto ERROR;
}
- else // render video data using sink pugin like xvimagesink
+
+ str = gst_caps_get_structure (caps, 0);
+ if ( ! str )
{
- debug_log("using videosink");
-
- /*set video converter */
- if (strlen(PLAYER_INI()->name_of_video_converter) > 0)
- {
- vconv_factory = PLAYER_INI()->name_of_video_converter;
- gboolean nv12t_hw_enabled = FALSE;
+ debug_error ("cannot get structure from capse.\n");
+ goto ERROR;
+ }
+
+ name = gst_structure_get_name (str);
+ if ( ! name )
+ {
+ debug_error ("cannot get mimetype from structure.\n");
+ goto ERROR;
+ }
+
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
+ //debug_log ("detected mimetype : %s\n", name);
- if (player->is_nv12_tiled && PLAYER_INI()->use_video_hw_accel)
- nv12t_hw_enabled = TRUE;
-
- if ( (nv12t_hw_enabled && (PLAYER_INI()->video_surface == MM_DISPLAY_SURFACE_EVAS) && !strcmp(PLAYER_INI()->videosink_element_evas, "evasimagesink")) || (nv12t_hw_enabled && use_multi_surface && !strcmp(PLAYER_INI()->videosink_element_evas, "evasimagesink") ) )
+ if (strstr(name, "video"))
+ {
+ gint stype = 0;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &stype);
+
+ /* don't make video because of not required, and not support multiple track */
+ if (stype == MM_DISPLAY_SURFACE_NULL)
+ {
+ debug_log ("no video sink by null surface or multiple track");
+ gchar *caps_str = gst_caps_to_string(caps);
+ if (strstr(caps_str, "ST12") || strstr(caps_str, "SN12"))
{
- vconv_factory = "fimcconvert";
+ player->set_mode.video_zc = TRUE;
}
- else if (nv12t_hw_enabled)
+ MMPLAYER_FREEIF( caps_str );
+
+ if (player->v_stream_caps)
{
- vconv_factory = NULL;
+ gst_caps_unref(player->v_stream_caps);
+ player->v_stream_caps = NULL;
}
- if (vconv_factory && !use_multi_surface)
+ debug_log ("create fakesink instead of videobin");
+
+ /* fake sink */
+ fakesink = gst_element_factory_make ("fakesink", NULL);
+ if (fakesink == NULL)
{
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CONV, vconv_factory, "video converter", TRUE);
- debug_log("using video converter: %s", vconv_factory);
+ debug_error ("ERROR : fakesink create error\n");
+ goto ERROR;
}
- }
- /* videoscaler */ /* NOTE : ini parsing method seems to be more suitable rather than define method */
- #if !defined(__arm__)
- if (!use_multi_surface) /* NOTE : at now, we did not consider using multi-surface in x86 case */
- {
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SCALE, "videoscale", "videoscaler", TRUE);
- }
- #endif
+ player->video_fakesink = fakesink;
- /* set video sink */
- switch (PLAYER_INI()->video_surface)
- {
- case MM_DISPLAY_SURFACE_X:
- if (strlen(PLAYER_INI()->videosink_element_x) > 0)
- videosink_element = PLAYER_INI()->videosink_element_x;
- else
- goto ERROR;
- break;
- case MM_DISPLAY_SURFACE_EVAS:
- if (strlen(PLAYER_INI()->videosink_element_evas) > 0)
- videosink_element = PLAYER_INI()->videosink_element_evas;
- else
- goto ERROR;
- break;
- case MM_DISPLAY_SURFACE_NULL:
- if (strlen(PLAYER_INI()->videosink_element_fake) > 0)
- videosink_element = PLAYER_INI()->videosink_element_fake;
- else
- goto ERROR;
- break;
- default:
- debug_error("unidentified surface type");
- goto ERROR;
- }
+ gst_bin_add (GST_BIN(pipeline), fakesink);
- if (use_multi_surface)
- {
- if ( (strlen(PLAYER_INI()->videosink_element_x) > 0 ) && (strlen(PLAYER_INI()->videosink_element_evas) > 0 ) )
+ // link
+ sinkpad = gst_element_get_static_pad (fakesink, "sink");
+
+ if (GST_PAD_LINK_OK != gst_pad_link(pad, sinkpad))
{
- /* create elements for multi sink usage */
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_TEE, "tee", "tee", TRUE);
- debug_log("using tee");
+ debug_warning ("failed to link fakesink\n");
+ gst_object_unref (GST_OBJECT(fakesink));
+ goto ERROR;
+ }
- if (vconv_factory)
- {
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CONV, vconv_factory, "video converter", TRUE);
- debug_log("using video converter: %s", vconv_factory);
- }
+ if (player->set_mode.media_packet_video_stream)
+ player->video_cb_probe_id = gst_pad_add_buffer_probe (sinkpad, G_CALLBACK (__mmplayer_video_stream_probe), player);
- if (PLAYER_INI()->video_surface == MM_DISPLAY_SURFACE_X)
- {
- videosink_element_ext = PLAYER_INI()->videosink_element_evas;
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK_EXT, videosink_element_ext, "videosink_ext_evas", TRUE);
- }
- else if (PLAYER_INI()->video_surface == MM_DISPLAY_SURFACE_EVAS)
- {
- videosink_element_ext = PLAYER_INI()->videosink_element_x;
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK_EXT, videosink_element_ext, "videosink_ext_x", TRUE);
- }
- else
- {
- debug_error("not normal case..multi-surface mode is only available with X and EVAS surface");
- goto ERROR;
- }
- debug_log("selected videosink_ext name: %s", videosink_element_ext);
- }
+ g_object_set (G_OBJECT (fakesink), "async", TRUE, NULL);
+ g_object_set (G_OBJECT (fakesink), "sync", TRUE, NULL);
+ gst_element_set_state (fakesink, GST_STATE_PAUSED);
+
+ goto DONE;
}
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, videosink_element, "videosink", TRUE);
- debug_log("selected videosink name: %s", videosink_element);
+
+ debug_log ("video selector \n");
+ elemId = MMPLAYER_M_V_INPUT_SELECTOR;
}
+ else
+ {
+ if (strstr(name, "audio"))
+ {
+ gint samplerate = 0;
+ gint channels = 0;
- if ( _mmplayer_update_video_param(player) != MM_ERROR_NONE)
- goto ERROR;
+ debug_log ("audio selector \n");
+ elemId = MMPLAYER_M_A_INPUT_SELECTOR;
- /* qos on */
- g_object_set (G_OBJECT (videobin[MMPLAYER_V_SINK].gst), "qos", TRUE, NULL);
+ gst_structure_get_int (str, "rate", &samplerate);
+ gst_structure_get_int (str, "channels", &channels);
- if (use_multi_surface)
- {
- g_object_set (G_OBJECT (videobin[MMPLAYER_V_SINK_EXT].gst), "qos", TRUE, NULL);
- g_object_set (G_OBJECT (videobin[MMPLAYER_V_SINK].gst), "async", FALSE, NULL);
- g_object_set (G_OBJECT (videobin[MMPLAYER_V_SINK_EXT].gst), "async", FALSE, NULL);
- }
+ if ((channels > 0 && samplerate == 0)) {//exclude audio decoding
+ /* fake sink */
+ fakesink = gst_element_factory_make ("fakesink", NULL);
+ if (fakesink == NULL)
+ {
+ debug_error ("ERROR : fakesink create error\n");
+ goto ERROR;
+ }
- /* store it as it's sink element */
- __mmplayer_add_sink( player, videobin[MMPLAYER_V_SINK].gst );
- if (use_multi_surface)
- {
- __mmplayer_add_sink( player, videobin[MMPLAYER_V_SINK_EXT].gst );
- }
+ gst_bin_add (GST_BIN(pipeline), fakesink);
- /* adding created elements to bin */
- if( ! __mmplayer_gst_element_add_bucket_to_bin(GST_BIN(videobin[MMPLAYER_V_BIN].gst), element_bucket) )
- {
- debug_error("failed to add elements\n");
- goto ERROR;
- }
+ /* link */
+ sinkpad = gst_element_get_static_pad (fakesink, "sink");
- if (!use_multi_surface)
- {
- /* Linking elements in the bucket by added order */
- if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ if (GST_PAD_LINK_OK != gst_pad_link(pad, sinkpad))
+ {
+ debug_warning ("failed to link fakesink\n");
+ gst_object_unref (GST_OBJECT(fakesink));
+ goto ERROR;
+ }
+
+ g_object_set (G_OBJECT (fakesink), "async", TRUE, NULL);
+ g_object_set (G_OBJECT (fakesink), "sync", TRUE, NULL);
+ gst_element_set_state (fakesink, GST_STATE_PAUSED);
+
+ goto DONE;
+ }
+ }
+ else if (strstr(name, "text"))
+ {
+ debug_log ("text selector \n");
+ elemId = MMPLAYER_M_T_SUBMUX_INTERNAL;
+ }
+ else
{
- debug_error("failed to link elements\n");
+ debug_error ("wrong elem id \n");
goto ERROR;
}
}
- else
+
+ selector = player->pipeline->mainbin[elemId].gst;
+
+ if (selector == NULL)
{
- GstPad *sinkpad;
- int i = 0;
- for (i = 0 ; i < 2 ; i++)
+ if (strstr(name, "audio") || strstr(name, "video")) {
+ selector = gst_element_factory_make ("input-selector", NULL);
+ debug_log ("Creating input-selector\n");
+ if (selector == NULL)
+ {
+ debug_error ("ERROR : input-selector create error\n");
+ goto ERROR;
+ }
+ g_object_set (selector, "sync-streams", TRUE, NULL);
+ }
+ else
{
- if (!vconv_factory)
+ if (player->textsink_linked)
{
- sinkpad = gst_element_get_static_pad (videobin[MMPLAYER_V_SINK+i].gst, "sink");
- debug_log("do not need vconv");
+ selector = gst_element_factory_make ("input-selector", "text-inselector");
+ debug_log ("Creating input-selector\n");
}
else
{
- if (PLAYER_INI()->video_surface == MM_DISPLAY_SURFACE_X)
- {
- if (i == 0)
- {
- sinkpad = gst_element_get_static_pad (videobin[MMPLAYER_V_SINK].gst, "sink");
- }
- else
- {
- sinkpad = gst_element_get_static_pad (videobin[MMPLAYER_V_CONV].gst, "sink");
- GST_ELEMENT_LINK(GST_ELEMENT(videobin[MMPLAYER_V_CONV].gst), GST_ELEMENT(videobin[MMPLAYER_V_SINK_EXT].gst));
- }
- }
- else
- {
- if (i == 0)
- {
- sinkpad = gst_element_get_static_pad (videobin[MMPLAYER_V_CONV].gst, "sink");
- GST_ELEMENT_LINK(GST_ELEMENT(videobin[MMPLAYER_V_CONV].gst), GST_ELEMENT(videobin[MMPLAYER_V_SINK].gst));
- }
- else
- {
- sinkpad = gst_element_get_static_pad (videobin[MMPLAYER_V_SINK_EXT].gst, "sink");
- }
- }
+ selector = gst_element_factory_make ("submux", "internal-submux");
+ g_object_set (G_OBJECT (selector), "lang-list", player->subtitle_language_list, NULL);
+ debug_log ("Creating submux\n");
}
- player->tee_src_pad[i] = gst_element_get_request_pad (videobin[MMPLAYER_V_TEE].gst, "src%d");
- if (gst_pad_link (player->tee_src_pad[i], sinkpad) != GST_PAD_LINK_OK) {
- debug_error("failed to link tee element with sink element(%d)",i);
+ if (selector == NULL)
+ {
+ debug_error ("ERROR : input-selector create error\n");
goto ERROR;
}
- gst_object_unref (sinkpad);
+ debug_log ("Created submux %p\n", selector);
}
- }
+ gst_bin_add (GST_BIN(pipeline), selector);
+ gst_element_set_state (selector, GST_STATE_PAUSED);
- /* get first element's sinkpad for creating ghostpad */
- first_element = (MMPlayerGstElement *)element_bucket->data;
- if ( !first_element )
+ player->pipeline->mainbin[elemId].id = elemId;
+ player->pipeline->mainbin[elemId].gst = selector;
+
+ first_track = TRUE;
+ // player->selector[stream_type].active_pad_index = DEFAULT_TRACK; // default
+
+ srcpad = gst_element_get_static_pad (selector, "src");
+
+ debug_log ("blocking %" GST_PTR_FORMAT, srcpad);
+ gst_pad_set_blocked_async (srcpad, TRUE, __mmplayer_gst_selector_blocked, NULL);
+ }
+ else
{
- debug_error("failed to get first element from bucket\n");
- goto ERROR;
+ debug_log ("input-selector is already created.\n");
+ selector = player->pipeline->mainbin[elemId].gst;
}
- pad = gst_element_get_static_pad(GST_ELEMENT(first_element->gst), "sink");
- if ( !pad )
+ // link
+ debug_log ("Calling request pad with selector %p \n", selector);
+ sinkpad = gst_element_get_request_pad (selector, "sink%d");
+
+ debug_log ("got pad %s:%s from selector", GST_DEBUG_PAD_NAME (sinkpad));
+
+ if (GST_PAD_LINK_OK != gst_pad_link(pad, sinkpad))
{
- debug_error("failed to get pad from first element\n");
+ debug_warning ("failed to link selector\n");
+ gst_object_unref (GST_OBJECT(selector));
goto ERROR;
}
- /* create ghostpad */
- if ( FALSE == gst_element_add_pad(videobin[MMPLAYER_V_BIN].gst, gst_ghost_pad_new("sink", pad)) )
+ if (first_track && (strstr(name, "audio") || strstr(name, "video")))
{
- debug_error("failed to add ghostpad to videobin\n");
- goto ERROR;
+ debug_log ("this is first track --> active track \n");
+ g_object_set (selector, "active-pad", sinkpad, NULL);
+ } else {
+ debug_log ("subtitle is of internal type \n");
+ if(!player->textsink_linked) {
+ g_object_set (selector, "is-internal", TRUE, NULL);
+ } else {
+ debug_log ("this is first track --> active track \n");
+ g_object_set (selector, "active-pad", sinkpad, NULL);
+ }
}
- gst_object_unref(pad);
-
- /* done. free allocated variables */
- g_list_free(element_bucket);
- mm_attrs_set_int_by_name(attrs, "content_video_found", TRUE);
- if ( mmf_attrs_commit ( attrs ) ) /* return -1 if error */
- debug_error("failed to commit attribute ""content_video_found"".\n");
+DONE:
+ERROR:
- debug_fleave();
+ if (caps)
+ {
+ gst_caps_unref (caps);
+ }
- return MM_ERROR_NONE;
+ if (sinkpad)
+ {
+ gst_object_unref (GST_OBJECT(sinkpad));
+ sinkpad = NULL;
+ }
-ERROR:
- debug_error("ERROR : releasing videobin\n");
+ if (srcpad)
+ {
+ gst_object_unref (GST_OBJECT(srcpad));
+ srcpad = NULL;
+ }
- g_list_free( element_bucket );
+ return;
+}
- if (pad)
- gst_object_unref(GST_OBJECT(pad));
+static void __mmplayer_handle_text_decode_path(mm_player_t* player, GstElement* text_selector)
+{
+ #define DEFAULT_TRACK 0
+ GstPad* srcpad = NULL;
+ MMHandleType attrs = 0;
+ gint active_index = 0;
- /* release videobin with it's childs */
- if ( videobin[MMPLAYER_V_BIN].gst )
+ // [link] input-selector :: textbin
+ srcpad = gst_element_get_static_pad (text_selector, "src");
+ if (!srcpad)
{
- gst_object_unref(GST_OBJECT(videobin[MMPLAYER_V_BIN].gst));
+ debug_error("failed to get srcpad from selector\n");
+ return;
}
+ debug_log ("got pad %s:%s from text selector\n", GST_DEBUG_PAD_NAME(srcpad));
- MMPLAYER_FREEIF( videobin );
-
- player->pipeline->videobin = NULL;
+ active_index = player->selector[MM_PLAYER_TRACK_TYPE_TEXT].active_pad_index;
+ if ((active_index != DEFAULT_TRACK) &&
+ (__mmplayer_change_selector_pad(player, MM_PLAYER_TRACK_TYPE_TEXT, active_index) != MM_ERROR_NONE))
+ {
+ debug_warning("failed to change text track\n");
+ player->selector[MM_PLAYER_TRACK_TYPE_TEXT].active_pad_index = DEFAULT_TRACK;
+ }
- return MM_ERROR_PLAYER_INTERNAL;
-}
+ player->no_more_pad = TRUE;
+ __mmplayer_gst_decode_callback (text_selector, srcpad, player);
-static int __mmplayer_gst_create_text_pipeline(mm_player_t* player)
-{
- MMPlayerGstElement* first_element = NULL;
- MMPlayerGstElement* textbin = NULL;
- GList* element_bucket = NULL;
- GstPad *pad = NULL;
- GstPad *ghostpad = NULL;
- gint i = 0;
+ debug_log ("unblocking %" GST_PTR_FORMAT, srcpad);
+ gst_pad_set_blocked_async (srcpad, FALSE, __mmplayer_gst_selector_blocked, NULL);
- debug_fenter();
+ debug_log("Total text tracks = %d \n", player->selector[MM_PLAYER_TRACK_TYPE_TEXT].total_track_num);
- return_val_if_fail( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ if (player->selector[MM_PLAYER_TRACK_TYPE_TEXT].total_track_num > 0)
+ player->has_closed_caption = TRUE;
- /* alloc handles */
- textbin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_T_NUM);
- if ( ! textbin )
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( attrs )
{
- debug_error("failed to allocate memory for textbin\n");
- return MM_ERROR_PLAYER_NO_FREE_SPACE;
+ mm_attrs_set_int_by_name(attrs, "content_text_track_num",(gint)player->selector[MM_PLAYER_TRACK_TYPE_TEXT].total_track_num);
+ if (mmf_attrs_commit (attrs))
+ debug_error("failed to commit.\n");
+ }
+ else
+ {
+ debug_error("cannot get content attribute");
}
- /* create bin */
- textbin[MMPLAYER_T_BIN].id = MMPLAYER_T_BIN;
- textbin[MMPLAYER_T_BIN].gst = gst_bin_new("textbin");
- if ( !textbin[MMPLAYER_T_BIN].gst )
+ if (srcpad)
{
- debug_critical("failed to create textbin\n");
- goto ERROR;
+ gst_object_unref ( GST_OBJECT(srcpad) );
+ srcpad = NULL;
}
+}
- /* take it */
- player->pipeline->textbin = textbin;
+int _mmplayer_gst_set_audio_channel(MMHandleType hplayer, MMPlayerAudioChannel ch_idx)
+{
+ int result = MM_ERROR_NONE;
- /* fakesink */
- MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_SINK, "fakesink", "text_sink", TRUE);
+ mm_player_t* player = (mm_player_t*)hplayer;
+ MMPlayerGstElement* mainbin = NULL;
+ gchar* change_pad_name = NULL;
+ GstPad* sinkpad = NULL;
+ GstCaps* caps = NULL;
- g_object_set (G_OBJECT (textbin[MMPLAYER_T_SINK].gst), "sync", TRUE, NULL);
- g_object_set (G_OBJECT (textbin[MMPLAYER_T_SINK].gst), "async", FALSE, NULL);
- g_object_set (G_OBJECT (textbin[MMPLAYER_T_SINK].gst), "signal-handoffs", TRUE, NULL);
+ MMPLAYER_FENTER();
- MMPLAYER_SIGNAL_CONNECT( player,
- G_OBJECT(textbin[MMPLAYER_T_SINK].gst),
- "handoff",
- G_CALLBACK(__mmplayer_update_subtitle),
- (gpointer)player );
+ return_val_if_fail (player, MM_ERROR_PLAYER_NOT_INITIALIZED);
- __mmplayer_add_sink (player, GST_ELEMENT(textbin[MMPLAYER_T_SINK].gst));
+ debug_log ("Change Audio mode to %d\n", ch_idx);
+ player->use_deinterleave = TRUE;
- /* adding created elements to bin */
- debug_log("adding created elements to bin\n");
- if( !__mmplayer_gst_element_add_bucket_to_bin( GST_BIN(textbin[MMPLAYER_T_BIN].gst), element_bucket ))
+ if ((!player->pipeline) || (!player->pipeline->mainbin))
{
- debug_error("failed to add elements\n");
- goto ERROR;
+ debug_log ("pre setting : %d\n", ch_idx);
+
+ player->audio_mode.active_pad_index = ch_idx;
+ return result;
}
- /* linking elements in the bucket by added order. */
- debug_log("Linking elements in the bucket by added order.\n");
- if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ mainbin = player->pipeline->mainbin;
+
+ if (mainbin[MMPLAYER_M_A_SELECTOR].gst == NULL)
{
- debug_error("failed to link elements\n");
- goto ERROR;
+ if (player->max_audio_channels < 2)
+ {
+ debug_log ("mono channel track only\n");
+ return result;
+ }
+
+ debug_warning ("selector doesn't exist\n");
+ return result; /* keep playing */
}
- /* get first element's sinkpad for creating ghostpad */
- first_element = (MMPlayerGstElement *)element_bucket->data;
+ debug_log ("total_ch_num : %d\n", player->audio_mode.total_track_num);
- pad = gst_element_get_static_pad(GST_ELEMENT(first_element->gst), "sink");
- if ( ! pad )
+ if (player->audio_mode.total_track_num < 2)
{
- debug_error("failed to get pad from first element of textbin\n");
- goto ERROR;
+ debug_warning ("there is no another audio path\n");
+ return result; /* keep playing */
}
- ghostpad = gst_ghost_pad_new("sink", pad);
- if ( ! ghostpad )
+ if ((ch_idx < 0) || (ch_idx >= player->audio_mode.total_track_num))
{
- debug_error("failed to create ghostpad\n");
- goto ERROR;
+ debug_warning ("Not a proper ch_idx : %d \n", ch_idx);
+ return result; /* keep playing */
}
- if ( FALSE == gst_element_add_pad(textbin[MMPLAYER_T_BIN].gst, ghostpad) )
+ /*To get the new pad from the selector*/
+ change_pad_name = g_strdup_printf ("sink%d", ch_idx);
+ if (change_pad_name == NULL)
{
- debug_error("failed to add ghostpad to textbin\n");
- goto ERROR;
+ debug_warning ("Pad does not exists\n");
+ goto ERROR; /* keep playing */
}
- gst_object_unref(pad);
+ debug_log ("new active pad name: %s\n", change_pad_name);
+ sinkpad = gst_element_get_static_pad (mainbin[MMPLAYER_M_A_SELECTOR].gst, change_pad_name);
+ if (sinkpad == NULL)
+ {
+ //result = MM_ERROR_PLAYER_INTERNAL;
+ goto ERROR; /* keep playing */
+ }
- /* done. free allocated variables */
- g_list_free(element_bucket);
+ debug_log ("Set Active Pad - %s:%s\n", GST_DEBUG_PAD_NAME(sinkpad));
+ g_object_set (mainbin[MMPLAYER_M_A_SELECTOR].gst, "active-pad", sinkpad, NULL);
- debug_fleave();
+ caps = GST_PAD_CAPS(sinkpad);
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
- return MM_ERROR_NONE;
+ __mmplayer_set_audio_attrs (player, caps);
+ player->audio_mode.active_pad_index = ch_idx;
ERROR:
- debug_log("ERROR : releasing textbin\n");
+ if (sinkpad)
+ gst_object_unref (sinkpad);
- if ( pad )
- gst_object_unref(GST_OBJECT(pad));
+ MMPLAYER_FREEIF(change_pad_name);
- if ( ghostpad )
- gst_object_unref(GST_OBJECT(ghostpad));
+ MMPLAYER_FLEAVE();
+ return result;
+}
- g_list_free( element_bucket );
- /* release element which are not added to bin */
- for ( i = 1; i < MMPLAYER_T_NUM; i++ ) /* NOTE : skip bin */
- {
- if ( textbin[i].gst )
- {
- GstObject* parent = NULL;
- parent = gst_element_get_parent( textbin[i].gst );
+static void
+__mmplayer_gst_deinterleave_pad_added(GstElement *elem, GstPad *pad, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ GstElement* selector = NULL;
+ GstElement* queue = NULL;
- if ( !parent )
- {
- gst_object_unref(GST_OBJECT(textbin[i].gst));
- textbin[i].gst = NULL;
- }
- else
- {
- gst_object_unref(GST_OBJECT(parent));
- }
- }
- }
+ GstPad* srcpad = NULL;
+ GstPad* sinkpad = NULL;
+ gchar* caps_str= NULL;
- /* release textbin with it's childs */
- if ( textbin[MMPLAYER_T_BIN].gst )
+ MMPLAYER_FENTER();
+ return_if_fail (player && player->pipeline && player->pipeline->mainbin);
+
+ caps_str = gst_caps_to_string(GST_PAD_CAPS(pad));
+ debug_log ("deinterleave new caps : %s\n", caps_str);
+ MMPLAYER_FREEIF(caps_str);
+
+ if ((queue = __mmplayer_element_create_and_link(player, pad, "queue")) == NULL)
{
- gst_object_unref(GST_OBJECT(textbin[MMPLAYER_T_BIN].gst));
+ debug_error ("ERROR : queue create error\n");
+ goto ERROR;
}
- MMPLAYER_FREEIF( textbin );
+ g_object_set(G_OBJECT(queue),
+ "max-size-buffers", 10,
+ "max-size-bytes", 0,
+ "max-size-time", (guint64)0,
+ NULL);
- player->pipeline->textbin = NULL;
+ selector = player->pipeline->mainbin[MMPLAYER_M_A_SELECTOR].gst;
- return MM_ERROR_PLAYER_INTERNAL;
-}
+ if (!selector)
+ {
+ debug_error("there is no audio channel selector.\n");
+ goto ERROR;
+ }
+ srcpad = gst_element_get_static_pad (queue, "src");
+ sinkpad = gst_element_get_request_pad (selector, "sink%d");
-static int
-__mmplayer_gst_create_subtitle_pipeline(mm_player_t* player)
-{
- GstBus *bus = NULL;
- MMPlayerGstElement* subtitlebin = NULL;
- MMHandleType attrs = 0;
- gchar *subtitle_uri =NULL;
- GList*element_bucket = NULL;
+ debug_log ("link (%s:%s - %s:%s)\n", GST_DEBUG_PAD_NAME(srcpad), GST_DEBUG_PAD_NAME(sinkpad));
- #define USE_MESSAGE_FOR_PLAYING_SUBTITLE
-#ifndef USE_MESSAGE_FOR_PLAYING_SUBTITLE
- void *xid = NULL;
- gint width =0, height = 0;
- gboolean silent=FALSE;
-#endif
+ if (GST_PAD_LINK_OK != gst_pad_link(srcpad, sinkpad))
+ {
+ debug_warning ("failed to link deinterleave - selector\n");
+ goto ERROR;
+ }
- debug_fenter();
+ gst_element_set_state (queue, GST_STATE_PAUSED);
+ player->audio_mode.total_track_num++;
- /* get mainbin */
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ERROR:
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
+ if (srcpad)
{
- debug_error("cannot get content attribute\n");
- return MM_ERROR_PLAYER_INTERNAL;
+ gst_object_unref ( GST_OBJECT(srcpad) );
+ srcpad = NULL;
}
- mm_attrs_get_string_by_name ( attrs, "subtitle_uri", &subtitle_uri );
- if ( !subtitle_uri || strlen(subtitle_uri) < 1)
+ if (sinkpad)
{
- debug_error("subtitle uri is not proper filepath.\n");
- return MM_ERROR_PLAYER_INVALID_URI;
+ gst_object_unref ( GST_OBJECT(sinkpad) );
+ sinkpad = NULL;
}
- debug_log("subtitle file path is [%s].\n", subtitle_uri);
+ MMPLAYER_FLEAVE();
+ return;
+}
- /* alloc handles */
- subtitlebin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_SUB_NUM);
- if ( !subtitlebin )
- {
- debug_error("failed to allocate memory\n");
- return MM_ERROR_PLAYER_NO_FREE_SPACE;
- }
+static void
+__mmplayer_gst_deinterleave_no_more_pads (GstElement *elem, gpointer data)
+{
+ mm_player_t* player = NULL;
+ GstElement* selector = NULL;
+ GstPad* sinkpad = NULL;
+ gint active_index = 0;
+ gchar* change_pad_name = NULL;
+ GstCaps* caps = NULL; // no need to unref
- /* create bin */
- subtitlebin[MMPLAYER_SUB_PIPE].id = MMPLAYER_SUB_PIPE;
- subtitlebin[MMPLAYER_SUB_PIPE].gst = gst_pipeline_new("subtitlebin");
- if ( !subtitlebin[MMPLAYER_SUB_PIPE].gst )
+ MMPLAYER_FENTER();
+ player = (mm_player_t*) data;
+
+ selector = player->pipeline->mainbin[MMPLAYER_M_A_SELECTOR].gst;
+
+ if (!selector)
{
- debug_error("failed to create text pipeline\n");
+ debug_error("there is no audio channel selector.\n");
goto ERROR;
}
- player->pipeline->subtitlebin = subtitlebin;
- /* create the text file source */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_SRC, "filesrc", "subtitle_source", TRUE);
- g_object_set(G_OBJECT (subtitlebin[MMPLAYER_SUB_SRC].gst), "location", subtitle_uri, NULL);
+ active_index = player->audio_mode.active_pad_index;
- /* queue */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_QUEUE, "queue", NULL, TRUE);
+ if (active_index != DEFAULT_AUDIO_CH)
+ {
+ gint audio_ch = DEFAULT_AUDIO_CH;
- /* subparse */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_SUBPARSE, "subparse", "subtitle_parser", TRUE);
+ /*To get the new pad from the selector*/
+ change_pad_name = g_strdup_printf ("sink%d", active_index);
+ if (change_pad_name != NULL)
+ {
+ sinkpad = gst_element_get_static_pad (selector, change_pad_name);
+ if (sinkpad != NULL)
+ {
+ debug_log ("Set Active Pad - %s:%s\n", GST_DEBUG_PAD_NAME(sinkpad));
+ g_object_set (selector, "active-pad", sinkpad, NULL);
-#ifndef USE_MESSAGE_FOR_PLAYING_SUBTITLE
- /* textrender */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_TEXTRENDER, "textrender", "subtitle_render", TRUE);
+ audio_ch = active_index;
- mm_attrs_get_int_by_name(attrs,"width", &width);
- mm_attrs_get_int_by_name(attrs,"height", &height);
- mm_attrs_get_int_by_name(attrs,"silent", &silent);
- g_object_set ( G_OBJECT (subtitlebin[MMPLAYER_SUB_TEXTRENDER].gst),"width", width, NULL);
- g_object_set ( G_OBJECT (subtitlebin[MMPLAYER_SUB_TEXTRENDER].gst),"height", height, NULL);
- g_object_set ( G_OBJECT (subtitlebin[MMPLAYER_SUB_TEXTRENDER].gst),"silent", silent, NULL);
+ caps = GST_PAD_CAPS(sinkpad);
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
- debug_log ( "subtitle winow size is [%dX%d].\n", width, height );
- debug_log ( "subtitle silent is [%d].\n", silent );
+ __mmplayer_set_audio_attrs (player, caps);
+ }
+ }
- /* converter1 */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_CONV1, "ffmpegcolorspace", "subtitle_converter1", TRUE);
+ player->audio_mode.active_pad_index = audio_ch;
+ debug_log("audio LR info (0:stereo) = %d\n", player->audio_mode.active_pad_index);
+ }
- /* videofliper */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_FLIP, "videoflip", "subtitle_fliper", TRUE);
+ERROR:
- /* converter2 */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_CONV2, "ffmpegcolorspace", "subtitle_converter2", TRUE);
+ if (sinkpad)
+ gst_object_unref (sinkpad);
- /* text sink */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_SINK, "ximagesink", "subtitle_sink", TRUE);
+ MMPLAYER_FLEAVE();
+ return;
+}
- mm_attrs_get_data_by_name(attrs, "xid", &xid);
- if ( xid )
- {
- debug_log("setting subtitle xid = %d\n", *(int*)xid);
- gst_x_overlay_set_xwindow_id(GST_X_OVERLAY(subtitlebin[MMPLAYER_SUB_SINK].gst), *(int*)xid);
- }
- else
- {
- /* FIXIT : is it error case? */
- debug_warning("still we don't have xid on player attribute. create it's own surface.\n");
- }
-#else
- /* text sink */
- MMPLAYER_CREATE_ELEMENT(subtitlebin, MMPLAYER_SUB_SINK, "fakesink", "subtitle_sink", TRUE);
+static void
+__mmplayer_gst_build_deinterleave_path (GstElement *elem, GstPad *pad, gpointer data)
+{
+ mm_player_t* player = NULL;
+ MMPlayerGstElement *mainbin = NULL;
- g_object_set (G_OBJECT (subtitlebin[MMPLAYER_SUB_SINK].gst), "sync", TRUE, NULL);
- g_object_set (G_OBJECT (subtitlebin[MMPLAYER_SUB_SINK].gst), "async", FALSE, NULL);
- g_object_set (G_OBJECT (subtitlebin[MMPLAYER_SUB_SINK].gst), "signal-handoffs", TRUE, NULL);
+ GstElement* tee = NULL;
+ GstElement* stereo_queue = NULL;
+ GstElement* mono_queue = NULL;
+ GstElement* conv = NULL;
+ GstElement* filter = NULL;
+ GstElement* deinterleave = NULL;
+ GstElement* selector = NULL;
- MMPLAYER_SIGNAL_CONNECT( player,
- G_OBJECT(subtitlebin[MMPLAYER_SUB_SINK].gst),
- "handoff",
- G_CALLBACK(__mmplayer_update_subtitle),
- (gpointer)player );
-#endif
+ GstPad* srcpad = NULL;
+ GstPad* selector_srcpad = NULL;
+ GstPad* sinkpad = NULL;
+ GstCaps* caps = NULL;
- /* adding created elements to bin */
- if( ! __mmplayer_gst_element_add_bucket_to_bin(GST_BIN(subtitlebin[MMPLAYER_SUB_PIPE].gst), element_bucket) )
- {
- debug_error("failed to add elements\n");
- goto ERROR;
- }
+ MMPLAYER_FENTER();
- /* Linking elements in the bucket by added order */
- if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
- {
- debug_error("failed to link elements\n");
- goto ERROR;
- }
+ /* check handles */
+ player = (mm_player_t*) data;
- /* done. free allocated variables */
- g_list_free(element_bucket);
+ return_if_fail( elem && pad );
+ return_if_fail( player && player->pipeline && player->pipeline->mainbin );
- player->play_subtitle = TRUE;
+ mainbin = player->pipeline->mainbin;
- debug_fleave();
-
- return MM_ERROR_NONE;
+ /* tee */
+ if ((tee = __mmplayer_element_create_and_link(player, pad, "tee")) == NULL)
+ {
+ debug_error ("ERROR : tee create error\n");
+ goto ERROR;
+ }
+ mainbin[MMPLAYER_M_A_TEE].id = MMPLAYER_M_A_TEE;
+ mainbin[MMPLAYER_M_A_TEE].gst = tee;
-ERROR:
- debug_error("ERROR : releasing text pipeline\n");
+ gst_element_set_state (tee, GST_STATE_PAUSED);
- g_list_free( element_bucket );
+ /* queue */
+ srcpad = gst_element_get_request_pad (tee, "src%d");
+ if ((stereo_queue = __mmplayer_element_create_and_link(player, srcpad, "queue")) == NULL)
+ {
+ debug_error ("ERROR : stereo queue create error\n");
+ goto ERROR;
+ }
+
+ g_object_set(G_OBJECT(stereo_queue),
+ "max-size-buffers", 10,
+ "max-size-bytes", 0,
+ "max-size-time", (guint64)0,
+ NULL);
- /* release subtitlebin with it's childs */
- if ( subtitlebin[MMPLAYER_SUB_PIPE].gst )
+ player->pipeline->mainbin[MMPLAYER_M_A_Q1].id = MMPLAYER_M_A_Q1;
+ player->pipeline->mainbin[MMPLAYER_M_A_Q1].gst = stereo_queue;
+
+ if (srcpad)
{
- gst_object_unref(GST_OBJECT(subtitlebin[MMPLAYER_SUB_PIPE].gst));
+ gst_object_unref (GST_OBJECT(srcpad));
+ srcpad = NULL;
}
- MMPLAYER_FREEIF( subtitlebin );
+ srcpad = gst_element_get_request_pad (tee, "src%d");
- player->pipeline->subtitlebin = NULL;
+ if ((mono_queue = __mmplayer_element_create_and_link(player, srcpad, "queue")) == NULL)
+ {
+ debug_error ("ERROR : mono queue create error\n");
+ goto ERROR;
+ }
- return MM_ERROR_PLAYER_INTERNAL;
-}
+ g_object_set(G_OBJECT(mono_queue),
+ "max-size-buffers", 10,
+ "max-size-bytes", 0,
+ "max-size-time", (guint64)0,
+ NULL);
-gboolean
-__mmplayer_update_subtitle( GstElement* object, GstBuffer *buffer, GstPad *pad, gpointer data)
-{
- mm_player_t* player = (mm_player_t*) data;
- MMMessageParamType msg = {0, };
- GstClockTime duration = 0;
- guint8 *text = NULL;
- gboolean ret = TRUE;
+ player->pipeline->mainbin[MMPLAYER_M_A_Q2].id = MMPLAYER_M_A_Q2;
+ player->pipeline->mainbin[MMPLAYER_M_A_Q2].gst = mono_queue;
- debug_fenter();
+ gst_element_set_state (stereo_queue, GST_STATE_PAUSED);
+ gst_element_set_state (mono_queue, GST_STATE_PAUSED);
- return_val_if_fail ( player, FALSE );
- return_val_if_fail ( buffer, FALSE );
+ /* audioconvert */
+ srcpad = gst_element_get_static_pad (mono_queue, "src");
+ if ((conv = __mmplayer_element_create_and_link(player, srcpad, "audioconvert")) == NULL)
+ {
+ debug_error ("ERROR : audioconvert create error\n");
+ goto ERROR;
+ }
- text = GST_BUFFER_DATA(buffer);
- duration = GST_BUFFER_DURATION(buffer);
+ player->pipeline->mainbin[MMPLAYER_M_A_CONV].id = MMPLAYER_M_A_CONV;
+ player->pipeline->mainbin[MMPLAYER_M_A_CONV].gst = conv;
- if ( player->is_subtitle_off )
+ /* caps filter */
+ if (srcpad)
{
- debug_log("subtitle is OFF.\n" );
- return TRUE;
+ gst_object_unref (GST_OBJECT(srcpad));
+ srcpad = NULL;
}
+ srcpad = gst_element_get_static_pad (conv, "src");
- if ( !text )
+ if ((filter = __mmplayer_element_create_and_link(player, srcpad, "capsfilter")) == NULL)
{
- debug_log("There is no subtitle to be displayed.\n" );
- return TRUE;
+ debug_error ("ERROR : capsfilter create error\n");
+ goto ERROR;
}
- msg.data = (void *) text;
- msg.subtitle.duration = GST_TIME_AS_MSECONDS(duration);
-
- debug_warning("update subtitle : [%ld msec] %s\n'", msg.subtitle.duration, (char*)msg.data );
-
- MMPLAYER_POST_MSG( player, MM_MESSAGE_UPDATE_SUBTITLE, &msg );
-
- debug_fleave();
-
- return ret;
-}
+ player->pipeline->mainbin[MMPLAYER_M_A_FILTER].id = MMPLAYER_M_A_FILTER;
+ player->pipeline->mainbin[MMPLAYER_M_A_FILTER].gst = filter;
+ caps = gst_caps_from_string( "audio/x-raw-int, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "channels = (int) 2");
-static int __gst_adjust_subtitle_position(mm_player_t* player, int format, int position)
-{
- GstEvent* event = NULL;
- gint64 current_pos = 0;
- gint64 adusted_pos = 0;
- gboolean ret = TRUE;
+ g_object_set (GST_ELEMENT(player->pipeline->mainbin[MMPLAYER_M_A_FILTER].gst), "caps", caps, NULL );
+ gst_caps_unref( caps );
- debug_fenter();
+ gst_element_set_state (conv, GST_STATE_PAUSED);
+ gst_element_set_state (filter, GST_STATE_PAUSED);
- /* check player and subtitlebin are created */
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( MMPLAYER_PLAY_SUBTITLE(player), MM_ERROR_PLAYER_NOT_INITIALIZED );
-
- if (position == 0)
+ /* deinterleave */
+ if (srcpad)
{
- debug_log("adjusted values is 0, no need to adjust subtitle position.\n");
- return MM_ERROR_NONE;
+ gst_object_unref (GST_OBJECT(srcpad));
+ srcpad = NULL;
}
+ srcpad = gst_element_get_static_pad (filter, "src");
- switch (format)
+ if ((deinterleave = __mmplayer_element_create_and_link(player, srcpad, "deinterleave")) == NULL)
{
- case MM_PLAYER_POS_FORMAT_TIME:
- {
- /* check current postion */
- ret = gst_element_query_position( GST_ELEMENT(player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst), &format, ¤t_pos );
- if ( !ret )
- {
- debug_warning("fail to query current postion.\n");
- return MM_ERROR_PLAYER_SEEK;
- }
- else
- {
- adusted_pos = current_pos + ((gint64)position * G_GINT64_CONSTANT(1000000));
- if (adusted_pos < 0)
- adusted_pos = G_GINT64_CONSTANT(0);
- debug_log("adjust subtitle postion : %lu -> %lu [msec]\n", GST_TIME_AS_MSECONDS(current_pos), GST_TIME_AS_MSECONDS(adusted_pos));
- }
-
- event = gst_event_new_seek (1.0, GST_FORMAT_TIME,
- ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- GST_SEEK_TYPE_SET, adusted_pos,
- GST_SEEK_TYPE_SET, -1);
- }
- break;
+ debug_error ("ERROR : deinterleave create error\n");
+ goto ERROR;
+ }
- case MM_PLAYER_POS_FORMAT_PERCENT:
- {
- debug_warning("percent format is not supported yet.\n");
- return MM_ERROR_INVALID_ARGUMENT;
- }
- break;
+ g_object_set (deinterleave, "keep-positions", TRUE, NULL);
- default:
- {
- debug_warning("invalid format.\n");
- return MM_ERROR_INVALID_ARGUMENT;
- }
- }
+ MMPLAYER_SIGNAL_CONNECT (player, deinterleave, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added",
+ G_CALLBACK (__mmplayer_gst_deinterleave_pad_added), player);
- /* keep ref to the event */
- gst_event_ref (event);
+ MMPLAYER_SIGNAL_CONNECT (player, deinterleave, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "no-more-pads",
+ G_CALLBACK (__mmplayer_gst_deinterleave_no_more_pads), player);
- debug_log("sending event[%s] to sink element [%s]\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(player->pipeline->subtitlebin[MMPLAYER_SUB_SINK].gst) );
+ player->pipeline->mainbin[MMPLAYER_M_A_DEINTERLEAVE].id = MMPLAYER_M_A_DEINTERLEAVE;
+ player->pipeline->mainbin[MMPLAYER_M_A_DEINTERLEAVE].gst = deinterleave;
- if ( ret = gst_element_send_event (player->pipeline->subtitlebin[MMPLAYER_SUB_SINK].gst, event) )
+ /* selector */
+ selector = gst_element_factory_make ("input-selector", "audio-channel-selector");
+ if (selector == NULL)
{
- debug_log("sending event[%s] to sink element [%s] success!\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(player->pipeline->subtitlebin[MMPLAYER_SUB_SINK].gst) );
+ debug_error ("ERROR : audio-selector create error\n");
+ goto ERROR;
}
- /* unref to the event */
- gst_event_unref (event);
-
-
- debug_fleave();
+ g_object_set (selector, "sync-streams", TRUE, NULL);
+ gst_bin_add (GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), selector);
- return MM_ERROR_NONE;
+ player->pipeline->mainbin[MMPLAYER_M_A_SELECTOR].id = MMPLAYER_M_A_SELECTOR;
+ player->pipeline->mainbin[MMPLAYER_M_A_SELECTOR].gst = selector;
-}
+ selector_srcpad = gst_element_get_static_pad (selector, "src");
-static void
-__gst_appsrc_feed_data_mem(GstElement *element, guint size, gpointer user_data) // @
-{
- GstElement *appsrc = element;
- tBuffer *buf = (tBuffer *)user_data;
- GstBuffer *buffer = NULL;
- GstFlowReturn ret = GST_FLOW_OK;
- gint len = size;
+ debug_log ("blocking %" GST_PTR_FORMAT, selector_srcpad);
+ gst_pad_set_blocked_async (selector_srcpad, TRUE, __mmplayer_gst_selector_blocked, NULL);
- return_if_fail ( element );
- return_if_fail ( buf );
+ if (srcpad)
+ {
+ gst_object_unref (GST_OBJECT(srcpad));
+ srcpad = NULL;
+ }
- buffer = gst_buffer_new ();
+ srcpad = gst_element_get_static_pad(stereo_queue, "src");
+ sinkpad = gst_element_get_request_pad (selector, "sink%d");
- if (buf->offset >= buf->len)
+ if (GST_PAD_LINK_OK != gst_pad_link(srcpad, sinkpad))
{
- debug_log("call eos appsrc\n");
- g_signal_emit_by_name (appsrc, "end-of-stream", &ret);
- return;
+ debug_warning ("failed to link queue_stereo - selector\n");
+ goto ERROR;
}
- if ( buf->len - buf->offset < size)
- {
- len = buf->len - buf->offset + buf->offset;
- }
+ player->audio_mode.total_track_num++;
- GST_BUFFER_DATA(buffer) = (guint8*)(buf->buf + buf->offset);
- GST_BUFFER_SIZE(buffer) = len;
- GST_BUFFER_OFFSET(buffer) = buf->offset;
- GST_BUFFER_OFFSET_END(buffer) = buf->offset + len;
+ g_object_set (selector, "active-pad", sinkpad, NULL);
+ gst_element_set_state (deinterleave, GST_STATE_PAUSED);
+ gst_element_set_state (selector, GST_STATE_PAUSED);
- debug_log("feed buffer %p, offset %u-%u length %u\n", buffer, buf->offset, buf->len,len);
- g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+ __mmplayer_gst_decode_callback (selector, selector_srcpad, player);
- buf->offset += len;
-}
+ debug_log ("unblocking %" GST_PTR_FORMAT, selector_srcpad);
+ gst_pad_set_blocked_async (selector_srcpad, FALSE, __mmplayer_gst_selector_blocked, NULL);
-static gboolean
-__gst_appsrc_seek_data_mem(GstElement *element, guint64 size, gpointer user_data) // @
-{
- tBuffer *buf = (tBuffer *)user_data;
+ERROR:
+ if (sinkpad)
+ {
+ gst_object_unref (GST_OBJECT(sinkpad));
+ sinkpad = NULL;
+ }
- return_val_if_fail ( buf, FALSE );
+ if (srcpad)
+ {
+ gst_object_unref (GST_OBJECT(srcpad));
+ srcpad = NULL;
+ }
- buf->offset = (int)size;
+ if (selector_srcpad)
+ {
+ gst_object_unref (GST_OBJECT(selector_srcpad));
+ selector_srcpad = NULL;
+ }
- return TRUE;
+ MMPLAYER_FLEAVE();
+ return;
}
static void
-__gst_appsrc_feed_data(GstElement *element, guint size, gpointer user_data) // @
+__mmplayer_gst_decode_no_more_pads (GstElement *elem, gpointer data)
{
- mm_player_t *player = (mm_player_t*)user_data;
+ mm_player_t* player = NULL;
+ GstPad* srcpad = NULL;
+ GstElement* video_selector = NULL;
+ GstElement* audio_selector = NULL;
+ GstElement* text_selector = NULL;
+ MMHandleType attrs = 0;
+ gint active_index = 0;
+ GstFormat fmt = GST_FORMAT_BYTES;
+ gint64 dur_bytes = 0L;
- return_if_fail ( player );
+ player = (mm_player_t*) data;
- debug_msg("app-src: feed data\n");
-
- if(player->need_data_cb)
- player->need_data_cb(size, player->buffer_cb_user_param);
-}
+ debug_log("no-more-pad signal handling\n");
-static gboolean
-__gst_appsrc_seek_data(GstElement *element, guint64 offset, gpointer user_data) // @
-{
- mm_player_t *player = (mm_player_t*)user_data;
+ if ((player->cmd == MMPLAYER_COMMAND_DESTROY) ||
+ (player->cmd == MMPLAYER_COMMAND_UNREALIZE))
+ {
+ debug_warning("no need to go more");
- return_val_if_fail ( player, FALSE );
+ if (player->pp_rebuilding)
+ {
+ player->pp_rebuilding = FALSE;
+ MMPLAYER_PLAYBACK_UNLOCK(player);
+ }
- debug_msg("app-src: seek data\n");
+ return;
+ }
- if(player->seek_data_cb)
- player->seek_data_cb(offset, player->buffer_cb_user_param);
+ if ((!MMPLAYER_IS_HTTP_PD(player)) &&
+ (MMPLAYER_IS_HTTP_STREAMING(player)) &&
+ (!player->pipeline->mainbin[MMPLAYER_M_DEMUXED_S_BUFFER].gst) &&
+ (player->pipeline->mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst))
+ {
+ #define ESTIMATED_BUFFER_UNIT (1*1024*1024)
- return TRUE;
-}
+ if (NULL == player->streamer)
+ {
+ debug_warning("invalid state for buffering");
+ goto ERROR;
+ }
+ gdouble init_buffering_time = (gdouble)player->streamer->buffering_req.initial_second;
+ guint buffer_bytes = init_buffering_time * ESTIMATED_BUFFER_UNIT;
-static gboolean
-__gst_appsrc_enough_data(GstElement *element, gpointer user_data) // @
-{
- mm_player_t *player = (mm_player_t*)user_data;
+ buffer_bytes = MAX(buffer_bytes, player->streamer->buffer_handle[BUFFER_TYPE_MUXED].buffering_bytes);
+ debug_log("[Decodebin2] set use-buffering on Q2 (pre buffer time: %d sec, buffer size : %d)\n", (gint)init_buffering_time, buffer_bytes);
- return_val_if_fail ( player, FALSE );
+ init_buffering_time = (init_buffering_time != 0)?(init_buffering_time):(player->ini.http_buffering_time);
- debug_msg("app-src: enough data:%p\n", player->enough_data_cb);
-
- if(player->enough_data_cb)
- player->enough_data_cb(player->buffer_cb_user_param);
+ if ( !gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, &fmt, &dur_bytes))
+ debug_error("fail to get duration.\n");
- return TRUE;
-}
+ // enable use-buffering on queue2 instead of multiqueue (ex)audio only streaming
+ // use file information was already set on Q2 when it was created.
+ __mm_player_streaming_set_queue2(player->streamer,
+ player->pipeline->mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst,
+ TRUE, // use_buffering
+ buffer_bytes,
+ init_buffering_time,
+ 1.0, // low percent
+ player->ini.http_buffering_limit, // high percent
+ FALSE,
+ NULL,
+ ((dur_bytes>0)?((guint64)dur_bytes):0));
+ }
-int
-_mmplayer_push_buffer(MMHandleType hplayer, unsigned char *buf, int size) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- GstBuffer *buffer = NULL;
- GstFlowReturn gst_ret = GST_FLOW_OK;
- int ret = MM_ERROR_NONE;
+ video_selector = player->pipeline->mainbin[MMPLAYER_M_V_INPUT_SELECTOR].gst;
+ audio_selector = player->pipeline->mainbin[MMPLAYER_M_A_INPUT_SELECTOR].gst;
+ text_selector = player->pipeline->mainbin[MMPLAYER_M_T_SUBMUX_INTERNAL].gst;
+
+ if (video_selector)
+ {
+ // the first track can be played only. not support track changing yet.
+ // [link] input-selector :: videobin
+ srcpad = gst_element_get_static_pad (video_selector, "src");
+ if (!srcpad)
+ {
+ debug_error("failed to get srcpad from video selector\n");
+ goto ERROR;
+ }
- debug_fenter();
+ debug_log ("got pad %s:%s from selector\n", GST_DEBUG_PAD_NAME(srcpad));
+ if ((!text_selector) && (!audio_selector))
+ player->no_more_pad = TRUE;
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ __mmplayer_gst_decode_callback (video_selector, srcpad, player);
- /* check current state */
-// MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_START );
+ debug_log ("unblocking %" GST_PTR_FORMAT, srcpad);
+ gst_pad_set_blocked_async (srcpad, FALSE, __mmplayer_gst_selector_blocked, NULL);
+ debug_log("Total video tracks = %d \n", player->selector[MM_PLAYER_TRACK_TYPE_VIDEO].total_track_num);
- /* NOTE : we should check and create pipeline again if not created as we destroy
- * whole pipeline when stopping in streamming playback
- */
- if ( ! player->pipeline )
+ gst_object_unref ( GST_OBJECT(srcpad) );
+ srcpad = NULL;
+ }
+ else
{
- if ( MM_ERROR_NONE != __gst_realize( player ) )
+ if ((player->pipeline->videobin) && (player->pipeline->videobin[MMPLAYER_V_BIN].gst))
{
- debug_error("failed to realize before starting. only in streamming\n");
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_log ("There is no video track : remove videobin");
+
+ __mmplayer_release_signal_connection( player, MM_PLAYER_SIGNAL_TYPE_VIDEOBIN );
+ __mmplayer_del_sink ( player, player->pipeline->videobin[MMPLAYER_V_SINK].gst );
+
+ MMPLAYER_RELEASE_ELEMENT ( player, player->pipeline->videobin, MMPLAYER_V_BIN );
+ MMPLAYER_FREEIF ( player->pipeline->videobin )
}
}
- debug_msg("app-src: pushing data\n");
+ if (audio_selector)
+ {
+ #define DEFAULT_TRACK 0
+ active_index = player->selector[MM_PLAYER_TRACK_TYPE_AUDIO].active_pad_index;
+ if ((active_index != DEFAULT_TRACK) &&
+ (__mmplayer_change_selector_pad(player, MM_PLAYER_TRACK_TYPE_AUDIO, active_index) != MM_ERROR_NONE))
+ {
+ debug_warning("failed to change audio track\n");
+ player->selector[MM_PLAYER_TRACK_TYPE_AUDIO].active_pad_index = DEFAULT_TRACK;
+ }
- if ( buf == NULL )
- {
- debug_error("buf is null\n");
- return MM_ERROR_NONE;
- }
+ // [link] input-selector :: audiobin
+ srcpad = gst_element_get_static_pad (audio_selector, "src");
+ if (!srcpad)
+ {
+ debug_error("failed to get srcpad from selector\n");
+ goto ERROR;
+ }
- buffer = gst_buffer_new ();
+ debug_log ("got pad %s:%s from selector\n", GST_DEBUG_PAD_NAME(srcpad));
+ if (!text_selector)
+ player->no_more_pad = TRUE;
- if (size <= 0)
- {
- debug_log("call eos appsrc\n");
- g_signal_emit_by_name (player->pipeline->mainbin[MMPLAYER_M_SRC].gst, "end-of-stream", &gst_ret);
- return MM_ERROR_NONE;
- }
+ if ((player->use_deinterleave == TRUE) && (player->max_audio_channels >= 2))
+ {
+ debug_log ("unblocking %" GST_PTR_FORMAT, srcpad);
+ gst_pad_set_blocked_async (srcpad, FALSE, __mmplayer_gst_selector_blocked, NULL);
- GST_BUFFER_DATA(buffer) = (guint8*)(buf);
- GST_BUFFER_SIZE(buffer) = size;
+ __mmplayer_gst_build_deinterleave_path(audio_selector, srcpad, player);
+ }
+ else
+ {
+ __mmplayer_gst_decode_callback (audio_selector, srcpad, player);
- debug_log("feed buffer %p, length %u\n", buf, size);
- g_signal_emit_by_name (player->pipeline->mainbin[MMPLAYER_M_SRC].gst, "push-buffer", buffer, &gst_ret);
+ debug_log ("unblocking %" GST_PTR_FORMAT, srcpad);
+ gst_pad_set_blocked_async (srcpad, FALSE, __mmplayer_gst_selector_blocked, NULL);
+ }
- debug_fleave();
+ debug_log("Total audio tracks = %d \n", player->selector[MM_PLAYER_TRACK_TYPE_AUDIO].total_track_num);
- return ret;
-}
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( attrs )
+ {
+ mm_attrs_set_int_by_name(attrs, "content_audio_track_num",(gint)player->selector[MM_PLAYER_TRACK_TYPE_AUDIO].total_track_num);
+ if (mmf_attrs_commit (attrs))
+ debug_error("failed to commit.\n");
+ }
+ else
+ {
+ debug_error("cannot get content attribute");
+ }
+ }
+ else
+ {
+ if ((player->pipeline->audiobin) && (player->pipeline->audiobin[MMPLAYER_A_BIN].gst))
+ {
+ debug_log ("There is no audio track : remove audiobin");
-static GstBusSyncReply
-__mmplayer_bus_sync_callback (GstBus * bus, GstMessage * message, gpointer data)
-{
- mm_player_t *player = (mm_player_t *)data;
- GstElement *sender = (GstElement *) GST_MESSAGE_SRC (message);
- const gchar *name = gst_element_get_name (sender);
+ __mmplayer_release_signal_connection( player, MM_PLAYER_SIGNAL_TYPE_AUDIOBIN );
+ __mmplayer_del_sink ( player, player->pipeline->audiobin[MMPLAYER_A_SINK].gst );
- switch (GST_MESSAGE_TYPE (message))
+ MMPLAYER_RELEASE_ELEMENT ( player, player->pipeline->audiobin, MMPLAYER_A_BIN );
+ MMPLAYER_FREEIF ( player->pipeline->audiobin )
+ }
+ }
+
+ if (text_selector)
{
- case GST_MESSAGE_TAG:
- __mmplayer_gst_extract_tag_from_msg(player, message);
- break;
+ __mmplayer_handle_text_decode_path(player, text_selector);
+ }
- case GST_MESSAGE_DURATION:
- if (MMPLAYER_IS_STREAMING(player))
- {
- GstFormat format;
- gint64 bytes = 0;
+ MMPLAYER_FLEAVE();
- gst_message_parse_duration (message, &format, &bytes);
- if (format == GST_FORMAT_BYTES)
- {
- debug_log("data total size of http content: %lld", bytes);
- player->http_content_size = bytes;
- }
- }
- else
- {
- player->need_update_content_attrs = TRUE;
- _mmplayer_update_content_attrs(player);
- }
- default:
- return GST_BUS_PASS;
+ERROR:
+ if (srcpad)
+ {
+ gst_object_unref ( GST_OBJECT(srcpad) );
+ srcpad = NULL;
}
- debug_log("GST_MESSAGE_TAG from %s", name);
- gst_message_unref (message);
-
- return GST_BUS_DROP;
+ if (player->pp_rebuilding)
+ {
+ player->pp_rebuilding = FALSE;
+ MMPLAYER_PLAYBACK_UNLOCK(player);
+ }
}
-/**
- * This function is to create audio or video pipeline for playing.
- *
- * @param player [in] handle of player
- *
- * @return This function returns zero on success.
- * @remark
- * @see
- */
-static int
-__mmplayer_gst_create_pipeline(mm_player_t* player) // @
+static void
+__mmplayer_gst_decode_callback(GstElement *elem, GstPad *pad, gpointer data) // @
{
- GstBus *bus = NULL;
- MMPlayerGstElement *mainbin = NULL;
+ mm_player_t* player = NULL;
MMHandleType attrs = 0;
- GstElement* element = NULL;
- GList* element_bucket = NULL;
- gboolean need_state_holder = TRUE;
- gint i = 0;
+ GstElement* pipeline = NULL;
+ GstCaps* caps = NULL;
+ gchar* caps_str = NULL;
+ GstStructure* str = NULL;
+ const gchar* name = NULL;
+ GstPad* sinkpad = NULL;
+ GstElement* sinkbin = NULL;
+ gboolean reusing = FALSE;
+ GstElement *text_selector = NULL;
+
+ /* check handles */
+ player = (mm_player_t*) data;
- debug_fenter();
+ return_if_fail( elem && pad );
+ return_if_fail(player && player->pipeline && player->pipeline->mainbin);
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ pipeline = player->pipeline->mainbin[MMPLAYER_M_PIPE].gst;
- /* get profile attribute */
attrs = MMPLAYER_GET_ATTRS(player);
if ( !attrs )
{
debug_error("cannot get content attribute\n");
- goto INIT_ERROR;
+ goto ERROR;
}
- /* create pipeline handles */
- if ( player->pipeline )
+ /* get mimetype from caps */
+ caps = gst_pad_get_caps( pad );
+ if ( !caps )
{
- debug_warning("pipeline should be released before create new one\n");
- goto INIT_ERROR;
+ debug_error("cannot get caps from pad.\n");
+ goto ERROR;
}
+ caps_str = gst_caps_to_string(caps);
- player->pipeline = (MMPlayerGstPipelineInfo*) g_malloc0( sizeof(MMPlayerGstPipelineInfo) );
- if (player->pipeline == NULL)
- goto INIT_ERROR;
-
- memset( player->pipeline, 0, sizeof(MMPlayerGstPipelineInfo) );
-
+ str = gst_caps_get_structure( caps, 0 );
+ if ( ! str )
+ {
+ debug_error("cannot get structure from capse.\n");
+ goto ERROR;
+ }
- /* create mainbin */
- mainbin = (MMPlayerGstElement*) g_malloc0( sizeof(MMPlayerGstElement) * MMPLAYER_M_NUM );
- if (mainbin == NULL)
- goto INIT_ERROR;
-
- memset( mainbin, 0, sizeof(MMPlayerGstElement) * MMPLAYER_M_NUM);
-
-
- /* create pipeline */
- mainbin[MMPLAYER_M_PIPE].id = MMPLAYER_M_PIPE;
- mainbin[MMPLAYER_M_PIPE].gst = gst_pipeline_new("player");
- if ( ! mainbin[MMPLAYER_M_PIPE].gst )
+ name = gst_structure_get_name(str);
+ if ( ! name )
{
- debug_error("failed to create pipeline\n");
- goto INIT_ERROR;
+ debug_error("cannot get mimetype from structure.\n");
+ goto ERROR;
}
+ //debug_log("detected mimetype : %s\n", name);
- /* create source element */
- switch ( player->profile.uri_type )
+ if (strstr(name, "audio"))
{
- /* rtsp streamming */
- case MM_PLAYER_URI_TYPE_URL_RTSP:
+ if (player->pipeline->audiobin == NULL)
{
- gint udp_timeout, network_bandwidth;
- gchar *user_agent, *wap_profile;
-
- element = gst_element_factory_make(PLAYER_INI()->name_of_rtspsrc, "streaming_source");
-
- if ( !element )
+ if (MM_ERROR_NONE != __mmplayer_gst_create_audio_pipeline(player))
{
- debug_critical("failed to create streaming source element\n");
- break;
+ debug_error("failed to create audiobin. continuing without audio\n");
+ goto ERROR;
}
- debug_log("using streamming source [%s].\n", PLAYER_INI()->name_of_rtspsrc);
+ sinkbin = player->pipeline->audiobin[MMPLAYER_A_BIN].gst;
+ debug_log("creating audiosink bin success\n");
+ }
+ else
+ {
+ reusing = TRUE;
+ sinkbin = player->pipeline->audiobin[MMPLAYER_A_BIN].gst;
+ debug_log("reusing audiobin\n");
+ _mmplayer_update_content_attrs( player, ATTR_AUDIO);
+ }
- /* make it zero */
- udp_timeout = network_bandwidth = 0;
- user_agent = wap_profile = NULL;
+ if (player->selector[MM_PLAYER_TRACK_TYPE_AUDIO].total_track_num <= 0) // should not update if content have multi audio tracks
+ mm_attrs_set_int_by_name(attrs, "content_audio_track_num", 1);
- /* get attribute */
- mm_attrs_get_int_by_name ( attrs,"streaming_udp_timeout", &udp_timeout );
- mm_attrs_get_string_by_name ( attrs, "streaming_user_agent", &user_agent );
- mm_attrs_get_string_by_name ( attrs,"streaming_wap_profile", &wap_profile );
- mm_attrs_get_int_by_name ( attrs, "streaming_network_bandwidth", &network_bandwidth );
+ player->audiosink_linked = 1;
- debug_log("setting streaming source ----------------\n");
- debug_log("udp_timeout : %d\n", udp_timeout);
- debug_log("user_agent : %s\n", user_agent);
- debug_log("wap_profile : %s\n", wap_profile);
- debug_log("network_bandwidth : %d\n", network_bandwidth);
- debug_log("buffering time : %d\n", PLAYER_INI()->rtsp_buffering_time);
- debug_log("rebuffering time : %d\n", PLAYER_INI()->rtsp_rebuffering_time);
- debug_log("-----------------------------------------\n");
+ sinkpad = gst_element_get_static_pad( GST_ELEMENT(sinkbin), "sink" );
+ if ( !sinkpad )
+ {
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
+ }
+ }
+ else if (strstr(name, "video"))
+ {
+ if (strstr(caps_str, "ST12") || strstr(caps_str, "SN12"))
+ {
+ player->set_mode.video_zc = TRUE;
+ }
- /* setting property to streaming source */
- g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
- g_object_set(G_OBJECT(element), "timeout", udp_timeout, NULL);
- g_object_set(G_OBJECT(element), "bandwidth", network_bandwidth, NULL);
- g_object_set(G_OBJECT(element), "buffering_time", PLAYER_INI()->rtsp_buffering_time, NULL);
- g_object_set(G_OBJECT(element), "rebuffering_time", PLAYER_INI()->rtsp_rebuffering_time, NULL);
- if ( user_agent )
- g_object_set(G_OBJECT(element), "user_agent", user_agent, NULL);
- if ( wap_profile )
- g_object_set(G_OBJECT(element), "wap_profile", wap_profile, NULL);
+ if (player->pipeline->videobin == NULL)
+ {
+ /* NOTE : not make videobin because application dose not want to play it even though file has video stream. */
+ /* get video surface type */
+ int surface_type = 0;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &surface_type);
- MMPLAYER_SIGNAL_CONNECT ( player, G_OBJECT(element), "pad-added",
- G_CALLBACK (__mmplayer_gst_rtp_dynamic_pad), player );
- MMPLAYER_SIGNAL_CONNECT ( player, G_OBJECT(element), "no-more-pads",
- G_CALLBACK (__mmplayer_gst_rtp_no_more_pads), player );
+ if (surface_type == MM_DISPLAY_SURFACE_NULL)
+ {
+ debug_log("not make videobin because it dose not want\n");
+ goto ERROR;
+ }
- player->no_more_pad = FALSE;
- player->num_dynamic_pad = 0;
+ if (MM_ERROR_NONE != __mmplayer_gst_create_video_pipeline(player, caps, surface_type) )
+ {
+ debug_error("failed to create videobin. continuing without video\n");
+ goto ERROR;
+ }
- /* NOTE : we cannot determine it yet. this filed will be filled by
- * _mmplayer_update_content_attrs() after START.
- */
- player->streaming_type = STREAMING_SERVICE_NONE;
+ sinkbin = player->pipeline->videobin[MMPLAYER_V_BIN].gst;
+ debug_log("creating videosink bin success\n");
}
- break;
-
- /* http streaming*/
- case MM_PLAYER_URI_TYPE_URL_HTTP:
+ else
{
- gchar *user_agent, *proxy, *cookies, **cookie_list;
- user_agent = proxy = cookies = NULL;
- cookie_list = NULL;
- gint mode = MM_PLAYER_PD_MODE_NONE;
-
- mm_attrs_get_int_by_name ( attrs, "pd_mode", &mode );
-
- player->pd_mode = mode;
+ reusing = TRUE;
+ sinkbin = player->pipeline->videobin[MMPLAYER_V_BIN].gst;
+ debug_log("re-using videobin\n");
+ _mmplayer_update_content_attrs( player, ATTR_VIDEO);
+ }
- debug_log("http playback, PD mode : %d\n", player->pd_mode);
+ /* FIXIT : track number shouldn't be hardcoded */
+ mm_attrs_set_int_by_name(attrs, "content_video_track_num", 1);
+ player->videosink_linked = 1;
- if ( ! MMPLAYER_IS_HTTP_PD(player) )
+ /* NOTE : intermediate code before doing H/W subtitle compositon */
+ if ( player->use_textoverlay && player->play_subtitle )
+ {
+ debug_log("using textoverlay for external subtitle");
+ /* check text bin has created well */
+ if ( player->pipeline && player->pipeline->textbin )
{
- element = gst_element_factory_make(PLAYER_INI()->name_of_httpsrc, "http_streaming_source");
- if ( !element )
+ /* get sinkpad from textoverlay */
+ sinkpad = gst_element_get_static_pad(
+ GST_ELEMENT(player->pipeline->textbin[MMPLAYER_T_BIN].gst),
+ "video_sink" );
+ if ( ! sinkpad )
{
- debug_critical("failed to create http streaming source element[%s].\n", PLAYER_INI()->name_of_httpsrc);
- break;
+ debug_error("failed to get sink pad from textoverlay");
+ goto ERROR;
}
- debug_log("using http streamming source [%s].\n", PLAYER_INI()->name_of_httpsrc);
-
- /* get attribute */
- mm_attrs_get_string_by_name ( attrs, "streaming_cookie", &cookies );
- mm_attrs_get_string_by_name ( attrs, "streaming_user_agent", &user_agent );
- mm_attrs_get_string_by_name ( attrs, "streaming_proxy", &proxy );
-
- /* get attribute */
- debug_log("setting http streaming source ----------------\n");
- debug_log("location : %s\n", player->profile.uri);
- debug_log("cookies : %s\n", cookies);
- debug_log("proxy : %s\n", proxy);
- debug_log("user_agent : %s\n", user_agent);
- debug_log("timeout : %d\n", PLAYER_INI()->http_timeout);
- debug_log("-----------------------------------------\n");
- /* setting property to streaming source */
- g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
- g_object_set(G_OBJECT(element), "timeout", PLAYER_INI()->http_timeout, NULL);
- /* check if prosy is vailid or not */
- if ( util_check_valid_url ( proxy ) )
- g_object_set(G_OBJECT(element), "proxy", proxy, NULL);
- /* parsing cookies */
- if ( ( cookie_list = util_get_cookie_list ((const char*)cookies) ) )
- g_object_set(G_OBJECT(element), "cookies", cookie_list, NULL);
- if ( user_agent )
- g_object_set(G_OBJECT(element), "user_agent", user_agent, NULL);
- }
- else // progressive download
- {
- if (player->pd_mode == MM_PLAYER_PD_MODE_URI)
+ /* link new pad with textoverlay first */
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
{
- gchar *path = NULL;
-
- mm_attrs_get_string_by_name ( attrs, "pd_location", &path );
-
- MMPLAYER_FREEIF(player->pd_file_location);
-
- debug_log("PD Location : %s\n", path);
-
- if ( path )
- {
- player->pd_file_location = g_strdup(path);
- }
- else
- {
- debug_error("can't find pd location so, it should be set \n");
- return MM_ERROR_PLAYER_FILE_NOT_FOUND;
- }
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
}
- element = gst_element_factory_make("pdpushsrc", "PD pushsrc");
- if ( !element )
+ gst_object_unref(sinkpad);
+ sinkpad = NULL;
+
+ /* alright, override pad to textbin.src for futher link */
+ pad = gst_element_get_static_pad(
+ GST_ELEMENT(player->pipeline->textbin[MMPLAYER_T_BIN].gst),
+ "src" );
+ if ( ! pad )
{
- debug_critical("failed to create PD push source element[%s].\n", "pdpushsrc");
- break;
+ debug_error("failed to get sink pad from textoverlay");
+ goto ERROR;
}
-
- g_object_set(G_OBJECT(element), "location", player->pd_file_location, NULL);
}
-
- player->streaming_type = STREAMING_SERVICE_NONE;
+ else
+ {
+ debug_error("should not reach here.");
+ goto ERROR;
+ }
}
- break;
- /* file source */
- case MM_PLAYER_URI_TYPE_FILE:
+ sinkpad = gst_element_get_static_pad( GST_ELEMENT(sinkbin), "sink" );
+ if ( !sinkpad )
{
- char* drmsrc = PLAYER_INI()->name_of_drmsrc;
-
- debug_log("using [%s] for 'file://' handler.\n", drmsrc);
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
+ }
+ }
+ else if (strstr(name, "text"))
+ {
+ if (player->pipeline->textbin == NULL)
+ {
+ MMPlayerGstElement* mainbin = NULL;
- element = gst_element_factory_make(drmsrc, "source");
- if ( !element )
+ if (MM_ERROR_NONE != __mmplayer_gst_create_text_pipeline(player))
{
- debug_critical("failed to create %s\n", drmsrc);
- break;
+ debug_error("failed to create textbin. continuing without text\n");
+ goto ERROR;
}
- g_object_set(G_OBJECT(element), "location", (player->profile.uri)+7, NULL); /* uri+7 -> remove "file:// */
- //g_object_set(G_OBJECT(element), "use-mmap", TRUE, NULL);
- }
- break;
+ sinkbin = player->pipeline->textbin[MMPLAYER_T_BIN].gst;
+ debug_log("creating textink bin success\n");
-#ifdef NO_USE_GST_HLSDEMUX
- case MM_PLAYER_URI_TYPE_HLS:
- {
- guint64 stream_type = GST_APP_STREAM_TYPE_STREAM;
+ /* FIXIT : track number shouldn't be hardcoded */
+ mm_attrs_set_int_by_name(attrs, "content_text_track_num", 1);
- debug_log("hsl src is selected\n");
+ player->textsink_linked = 1;
+ debug_msg("player->textsink_linked set to 1\n");
- element = gst_element_factory_make("appsrc", "hls-source");
- if ( !element )
+ sinkpad = gst_element_get_static_pad( GST_ELEMENT(sinkbin), "text_sink" );
+ if ( !sinkpad )
{
- debug_critical("failed to create appsrc element\n");
- break;
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
}
- g_object_set( element, "stream-type", stream_type, NULL );
- // TODO: need to set time based limit instead of bytes (or) remove constraint.... need to investigate more
- g_object_set( element, "max-bytes", 500000, NULL ); // Naveen made queue as un-limited
+ mainbin = player->pipeline->mainbin;
+
+ if (!mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst)
+ {
+ /* input selector */
+ text_selector = gst_element_factory_make("input-selector", "subtitle_inselector");
+ if ( !text_selector )
+ {
+ debug_error ( "failed to create subtitle input selector element\n" );
+ goto ERROR;
+ }
+ g_object_set (text_selector, "sync-streams", TRUE, NULL);
+
+ mainbin[MMPLAYER_M_T_INPUT_SELECTOR].id = MMPLAYER_M_T_INPUT_SELECTOR;
+ mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst = text_selector;
+
+ /* warm up */
+ if (GST_STATE_CHANGE_FAILURE == gst_element_set_state (text_selector, GST_STATE_READY))
+ {
+ debug_error("failed to set state(READY) to sinkbin\n");
+ goto ERROR;
+ }
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), text_selector))
+ {
+ debug_warning("failed to add subtitle input selector\n");
+ goto ERROR;
+ }
+
+ debug_log ("created element input-selector");
+
+ }
+ else
+ {
+ debug_log ("already having subtitle input selector");
+ text_selector = mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst;
+ }
}
- break;
-#endif
- /* appsrc */
- case MM_PLAYER_URI_TYPE_BUFF:
+ else
{
- guint64 stream_type = GST_APP_STREAM_TYPE_STREAM;
+ if (!player->textsink_linked)
+ {
+ debug_log("re-using textbin\n");
- debug_log("mem src is selected\n");
+ reusing = TRUE;
+ sinkbin = player->pipeline->textbin[MMPLAYER_T_BIN].gst;
- element = gst_element_factory_make("appsrc", "buff-source");
- if ( !element )
+ player->textsink_linked = 1;
+ debug_msg("player->textsink_linked set to 1\n");
+ }
+ else
{
- debug_critical("failed to create appsrc element\n");
- break;
+ debug_log("ignoring internal subtutle since external subtitle is available");
}
+ }
+ }
+ else
+ {
+ debug_warning("unknown type of elementary stream! ignoring it...\n");
+ goto ERROR;
+ }
- g_object_set( element, "stream-type", stream_type, NULL );
- //g_object_set( element, "size", player->mem_buf.len, NULL );
- //g_object_set( element, "blocksize", (guint64)20480, NULL );
+ if ( sinkbin )
+ {
+ if(!reusing)
+ {
+ /* warm up */
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state( sinkbin, GST_STATE_READY ) )
+ {
+ debug_error("failed to set state(READY) to sinkbin\n");
+ goto ERROR;
+ }
- MMPLAYER_SIGNAL_CONNECT( player, element, "seek-data",
- G_CALLBACK(__gst_appsrc_seek_data), player);
- MMPLAYER_SIGNAL_CONNECT( player, element, "need-data",
- G_CALLBACK(__gst_appsrc_feed_data), player);
- MMPLAYER_SIGNAL_CONNECT( player, element, "enough-data",
- G_CALLBACK(__gst_appsrc_enough_data), player);
+ /* Added for multi audio support to avoid adding audio bin again*/
+ /* add */
+ if ( FALSE == gst_bin_add( GST_BIN(pipeline), sinkbin ) )
+ {
+ debug_error("failed to add sinkbin to pipeline\n");
+ goto ERROR;
+ }
}
- break;
- /* appsrc */
- case MM_PLAYER_URI_TYPE_MEM:
+ if (text_selector)
{
- guint64 stream_type = GST_APP_STREAM_TYPE_RANDOM_ACCESS;
+ GstPad *srcpad = NULL;
+
+ srcpad = gst_element_get_static_pad(player->pipeline->mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst,"src");
+ if (!srcpad)
+ {
+ debug_error ("failed to get source pad");
+ goto ERROR;
+ }
+
+ /* link input selector & textbin */
+ if (gst_pad_link(srcpad,sinkpad)!= 0)
+ {
+ debug_warning("failed to link input selector and textbin ghost pad\n");
+ goto ERROR;
+ }
+
+ sinkpad = gst_element_get_request_pad (player->pipeline->mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "sink0");
+ if (sinkpad == NULL)
+ {
+ debug_error ("failed to get request pad from input selector");;
+ goto ERROR;
+ }
+ }
- debug_log("mem src is selected\n");
+ /* link */
+ if (GST_PAD_LINK_OK != GST_PAD_LINK (pad, sinkpad))
+ {
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
+ }
- element = gst_element_factory_make("appsrc", "mem-source");
- if ( !element )
+ if (!reusing)
+ {
+ /* run */
+ if (GST_STATE_CHANGE_FAILURE == gst_element_set_state (sinkbin, GST_STATE_PAUSED))
{
- debug_critical("failed to create appsrc element\n");
- break;
+ debug_error("failed to set state(PLAYING) to sinkbin\n");
+ goto ERROR;
}
- g_object_set( element, "stream-type", stream_type, NULL );
- g_object_set( element, "size", player->mem_buf.len, NULL );
- g_object_set( element, "blocksize", (guint64)20480, NULL );
-
- MMPLAYER_SIGNAL_CONNECT( player, element, "seek-data",
- G_CALLBACK(__gst_appsrc_seek_data_mem), &player->mem_buf );
- MMPLAYER_SIGNAL_CONNECT( player, element, "need-data",
- G_CALLBACK(__gst_appsrc_feed_data_mem), &player->mem_buf );
+ if (text_selector)
+ {
+ if (GST_STATE_CHANGE_FAILURE == gst_element_set_state (text_selector, GST_STATE_PAUSED))
+ {
+ debug_error("failed to set state(READY) to sinkbin\n");
+ goto ERROR;
+ }
+ }
}
- break;
- case MM_PLAYER_URI_TYPE_URL:
- break;
-
- case MM_PLAYER_URI_TYPE_TEMP:
- break;
- case MM_PLAYER_URI_TYPE_NONE:
- default:
- break;
+ gst_object_unref (sinkpad);
+ sinkpad = NULL;
}
- /* check source element is OK */
- if ( ! element )
- {
- debug_critical("no source element was created.\n");
- goto INIT_ERROR;
- }
+ debug_log ("linking sink bin success\n");
- /* take source element */
- mainbin[MMPLAYER_M_SRC].id = MMPLAYER_M_SRC;
- mainbin[MMPLAYER_M_SRC].gst = element;
- element_bucket = g_list_append(element_bucket, &mainbin[MMPLAYER_M_SRC]);
+ /* FIXIT : we cannot hold callback for 'no-more-pad' signal because signal was emitted in
+ * streaming task. if the task blocked, then buffer will not flow to the next element
+ * ( autoplugging element ). so this is special hack for streaming. please try to remove it
+ */
+ /* dec stream count. we can remove fakesink if it's zero */
+ if (player->num_dynamic_pad)
+ player->num_dynamic_pad--;
- if (MMPLAYER_IS_STREAMING(player) || (player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS))
- {
- player->streamer = __mm_player_streaming_create();
- __mm_player_streaming_initialize(player->streamer);
- }
+ debug_log ("no more pads: %d stream count dec : %d (num of dynamic pad)\n", player->no_more_pad, player->num_dynamic_pad);
- if ( player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS )
+ if ((player->no_more_pad) && (player->num_dynamic_pad == 0))
{
- debug_log ("adding appsrc's pad probe...\n");
- GstPad *pad = NULL;
- pad = gst_element_get_static_pad(mainbin[MMPLAYER_M_SRC].gst, "src" );
-
- /* register probe */
- ahs_appsrc_cb_probe_id = gst_pad_add_buffer_probe (pad,
- G_CALLBACK (__mmplayer_ahs_appsrc_probe), player);
-
- gst_object_unref(pad);
- pad = NULL;
- }
-
- if ( MMPLAYER_IS_HTTP_PD(player) )
- {
- debug_log ("Picked queue2 element....\n");
- element = gst_element_factory_make("queue2", "hls_stream_buffer");
- if ( !element )
- {
- debug_critical ( "failed to create http streaming buffer element\n" );
- goto INIT_ERROR;
- }
-
- /* take it */
- mainbin[MMPLAYER_M_S_BUFFER].id = MMPLAYER_M_S_BUFFER;
- mainbin[MMPLAYER_M_S_BUFFER].gst = element;
- element_bucket = g_list_append(element_bucket, &mainbin[MMPLAYER_M_S_BUFFER]);
-
- __mm_player_streaming_set_buffer(player->streamer,
- element,
- TRUE,
- PLAYER_INI()->http_max_size_bytes,
- 1.0,
- PLAYER_INI()->http_buffering_limit,
- PLAYER_INI()->http_buffering_time,
- FALSE,
- NULL,
- 0);
+ __mmplayer_pipeline_complete (NULL, player);
}
- /* create autoplugging element if src element is not a streamming src */
- if ( player->profile.uri_type != MM_PLAYER_URI_TYPE_URL_RTSP )
+ /* FIXIT : please leave a note why this code is needed */
+ if(MMPLAYER_IS_WFD_STREAMING( player ))
{
- element = NULL;
+ player->no_more_pad = TRUE;
+ }
- if( PLAYER_INI()->use_decodebin )
- {
- /* create decodebin */
- element = gst_element_factory_make("decodebin", "decodebin");
+ERROR:
- g_object_set(G_OBJECT(element), "async-handling", TRUE, NULL);
+ MMPLAYER_FREEIF(caps_str);
- /* set signal handler */
- MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(element), "new-decoded-pad",
- G_CALLBACK(__mmplayer_gst_decode_callback), player);
+ if ( caps )
+ gst_caps_unref( caps );
- /* we don't need state holder, bcz decodebin is doing well by itself */
- need_state_holder = FALSE;
- }
- else
- {
- element = gst_element_factory_make("typefind", "typefinder");
- MMPLAYER_SIGNAL_CONNECT( player, element, "have-type",
- G_CALLBACK(__mmplayer_typefind_have_type), (gpointer)player );
- }
+ if ( sinkpad )
+ gst_object_unref(GST_OBJECT(sinkpad));
- /* check autoplug element is OK */
- if ( ! element )
- {
- debug_critical("can not create autoplug element\n");
- goto INIT_ERROR;
- }
+ /* flusing out new attributes */
+ if ( mmf_attrs_commit ( attrs ) )
+ {
+ debug_error("failed to comit attributes\n");
+ }
- mainbin[MMPLAYER_M_AUTOPLUG].id = MMPLAYER_M_AUTOPLUG;
- mainbin[MMPLAYER_M_AUTOPLUG].gst = element;
+ return;
+}
- element_bucket = g_list_append(element_bucket, &mainbin[MMPLAYER_M_AUTOPLUG]);
- }
+static gboolean
+__mmplayer_get_property_value_for_rotation(mm_player_t* player, int rotation_angle, int *value)
+{
+ int pro_value = 0; // in the case of expection, default will be returned.
+ int dest_angle = rotation_angle;
+ int rotation_type = -1;
+ #define ROTATION_USING_SINK 0
+ #define ROTATION_USING_CUSTOM 1
+ #define ROTATION_USING_FLIP 2
+ return_val_if_fail(player, FALSE);
+ return_val_if_fail(value, FALSE);
+ return_val_if_fail(rotation_angle >= 0, FALSE);
- /* add elements to pipeline */
- if( !__mmplayer_gst_element_add_bucket_to_bin(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), element_bucket))
+ if (rotation_angle >= 360)
{
- debug_error("Failed to add elements to pipeline\n");
- goto INIT_ERROR;
+ dest_angle = rotation_angle - 360;
}
-
- /* linking elements in the bucket by added order. */
- if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ /* chech if supported or not */
+ if ( dest_angle % 90 )
{
- debug_error("Failed to link some elements\n");
- goto INIT_ERROR;
+ debug_log("not supported rotation angle = %d", rotation_angle);
+ return FALSE;
}
-
- /* create fakesink element for keeping the pipeline state PAUSED. if needed */
- if ( need_state_holder )
+ /*
+ * xvimagesink only (A)
+ * custom_convert - no xv (e.g. memsink, evasimagesink (B)
+ * videoflip - avsysmemsink (C)
+ */
+ if (player->set_mode.video_zc)
{
- /* create */
- mainbin[MMPLAYER_M_SRC_FAKESINK].id = MMPLAYER_M_SRC_FAKESINK;
- mainbin[MMPLAYER_M_SRC_FAKESINK].gst = gst_element_factory_make ("fakesink", "state-holder");
-
- if (!mainbin[MMPLAYER_M_SRC_FAKESINK].gst)
+ if (player->pipeline->videobin[MMPLAYER_V_CONV].gst) // B
{
- debug_error ("fakesink element could not be created\n");
- goto INIT_ERROR;
+ rotation_type = ROTATION_USING_CUSTOM;
}
- GST_OBJECT_FLAG_UNSET (mainbin[MMPLAYER_M_SRC_FAKESINK].gst, GST_ELEMENT_IS_SINK);
-
- /* take ownership of fakesink. we are reusing it */
- gst_object_ref( mainbin[MMPLAYER_M_SRC_FAKESINK].gst );
-
- /* add */
- if ( FALSE == gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst),
- mainbin[MMPLAYER_M_SRC_FAKESINK].gst) )
+ else // A
{
- debug_error("failed to add fakesink to bin\n");
- goto INIT_ERROR;
+ rotation_type = ROTATION_USING_SINK;
}
}
-
- /* now we have completed mainbin. take it */
- player->pipeline->mainbin = mainbin;
-
- /* connect bus callback */
- bus = gst_pipeline_get_bus(GST_PIPELINE(mainbin[MMPLAYER_M_PIPE].gst));
- if ( !bus )
+ else
{
- debug_error ("cannot get bus from pipeline.\n");
- goto INIT_ERROR;
- }
- player->bus_watcher = gst_bus_add_watch(bus, (GstBusFunc)__mmplayer_gst_callback, player);
+ int surface_type = 0;
+ rotation_type = ROTATION_USING_FLIP;
- /* Note : check whether subtitle atrribute uri is set. If uri is set, then create the text pipeline */
- if ( __mmplayer_check_subtitle ( player ) )
- {
- debug_log("try to create subtitle pipeline \n");
+ mm_attrs_get_int_by_name(player->attrs, "display_surface_type", &surface_type);
+ debug_log("check display surface type attribute: %d", surface_type);
- if ( MM_ERROR_NONE != __mmplayer_gst_create_subtitle_pipeline(player) )
- debug_error("fail to create subtitle pipeline")
+ if ((surface_type == MM_DISPLAY_SURFACE_X) ||
+ (surface_type == MM_DISPLAY_SURFACE_EVAS && !strcmp(player->ini.videosink_element_evas, "evaspixmapsink")))
+ {
+ rotation_type = ROTATION_USING_SINK;
+ }
else
- debug_log("subtitle pipeline is created successfully\n");
- }
-
- /* set sync handler to get tag synchronously */
- gst_bus_set_sync_handler(bus, __mmplayer_bus_sync_callback, player);
-
-
- /* finished */
- gst_object_unref(GST_OBJECT(bus));
- g_list_free(element_bucket);
-
- debug_fleave();
-
- return MM_ERROR_NONE;
-
-INIT_ERROR:
+ {
+ rotation_type = ROTATION_USING_FLIP; //C
+ }
- __mmplayer_gst_destroy_pipeline(player);
- g_list_free(element_bucket);
+ debug_log("using %d type for rotation", rotation_type);
+ }
- /* release element which are not added to bin */
- for ( i = 1; i < MMPLAYER_M_NUM; i++ ) /* NOTE : skip pipeline */
+ /* get property value for setting */
+ switch(rotation_type)
{
- if ( mainbin[i].gst )
- {
- GstObject* parent = NULL;
- parent = gst_element_get_parent( mainbin[i].gst );
-
- if ( !parent )
+ case ROTATION_USING_SINK: // xvimagesink, pixmap
{
- gst_object_unref(GST_OBJECT(mainbin[i].gst));
- mainbin[i].gst = NULL;
+ switch (dest_angle)
+ {
+ case 0:
+ break;
+ case 90:
+ pro_value = 3; // clockwise 90
+ break;
+ case 180:
+ pro_value = 2;
+ break;
+ case 270:
+ pro_value = 1; // counter-clockwise 90
+ break;
+ }
}
- else
+ break;
+ case ROTATION_USING_CUSTOM:
{
- gst_object_unref(GST_OBJECT(parent));
+ gchar *ename = NULL;
+ ename = GST_PLUGIN_FEATURE_NAME(gst_element_get_factory(player->pipeline->videobin[MMPLAYER_V_CONV].gst));
+
+ if (g_strrstr(ename, "fimcconvert"))
+ {
+ switch (dest_angle)
+ {
+ case 0:
+ break;
+ case 90:
+ pro_value = 90; // clockwise 90
+ break;
+ case 180:
+ pro_value = 180;
+ break;
+ case 270:
+ pro_value = 270; // counter-clockwise 90
+ break;
+ }
+ }
}
- }
- }
+ break;
+ case ROTATION_USING_FLIP: // videoflip
+ {
+ switch (dest_angle)
+ {
- /* release pipeline with it's childs */
- if ( mainbin[MMPLAYER_M_PIPE].gst )
- {
- gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_PIPE].gst));
+ case 0:
+ break;
+ case 90:
+ pro_value = 1; // clockwise 90
+ break;
+ case 180:
+ pro_value = 2;
+ break;
+ case 270:
+ pro_value = 3; // counter-clockwise 90
+ break;
+ }
+ }
+ break;
}
- MMPLAYER_FREEIF( player->pipeline );
- MMPLAYER_FREEIF( mainbin );
+ debug_log("setting rotation property value : %d, used rotation type : %d", pro_value, rotation_type);
- return MM_ERROR_PLAYER_INTERNAL;
-}
+ *value = pro_value;
+ return TRUE;
+}
-static int
-__mmplayer_gst_destroy_pipeline(mm_player_t* player) // @
+int
+_mmplayer_update_video_param(mm_player_t* player) // @
{
- gint timeout = 0;
- int ret = MM_ERROR_NONE;
- int i = 0;
+ MMHandleType attrs = 0;
+ int surface_type = 0;
+ int org_angle = 0; // current supported angle values are 0, 90, 180, 270
+ int user_angle = 0;
+ int user_angle_type= 0;
+ int rotation_value = 0;
- debug_fenter();
-
- return_val_if_fail ( player, MM_ERROR_INVALID_HANDLE );
+ MMPLAYER_FENTER();
- /* cleanup stuffs */
- MMPLAYER_FREEIF(player->type);
- player->have_dynamic_pad = FALSE;
- player->no_more_pad = FALSE;
- player->num_dynamic_pad = 0;
- player->last_position = 0;
- player->duration = 0;
- player->http_content_size = 0;
- player->not_supported_codec = MISSING_PLUGIN_NONE;
- player->can_support_codec = FOUND_PLUGIN_NONE;
- player->not_found_demuxer = 0;
- if (player->v_stream_caps)
+ /* check video sinkbin is created */
+ return_val_if_fail ( player &&
+ player->pipeline &&
+ player->pipeline->videobin &&
+ player->pipeline->videobin[MMPLAYER_V_BIN].gst &&
+ player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
{
- gst_caps_unref(player->v_stream_caps);
- player->v_stream_caps = NULL;
+ debug_error("cannot get content attribute");
+ return MM_ERROR_PLAYER_INTERNAL;
}
-
- player->state_lost = FALSE;
- player->need_update_content_dur = FALSE;
- player->pending_seek.is_pending = FALSE;
- player->pending_seek.format = MM_PLAYER_POS_FORMAT_TIME;
- player->pending_seek.pos = 0;
+ /* update user roation */
+ mm_attrs_get_int_by_name(attrs, "display_rotation", &user_angle_type);
- if (ahs_appsrc_cb_probe_id )
+ /* get angle with user type */
+ switch(user_angle_type)
{
- GstPad *pad = NULL;
- pad = gst_element_get_static_pad(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, "src" );
-
- gst_pad_remove_buffer_probe (pad, ahs_appsrc_cb_probe_id);
- gst_object_unref(pad);
- pad = NULL;
- ahs_appsrc_cb_probe_id = 0;
+ case MM_DISPLAY_ROTATION_NONE:
+ user_angle = 0;
+ break;
+ case MM_DISPLAY_ROTATION_90: // counter-clockwise 90
+ user_angle = 270;
+ break;
+ case MM_DISPLAY_ROTATION_180:
+ user_angle = 180;
+ break;
+ case MM_DISPLAY_ROTATION_270: // clockwise 90
+ user_angle = 90;
+ break;
}
- /* free request pads for multi-surface */
- if (player->use_multi_surface && player->pipeline && player->pipeline->videobin[MMPLAYER_V_TEE].gst)
+ /* get original orientation */
+ if (player->v_stream_caps)
{
- for (i = 0 ; i < 2 ; i++)
+ GstStructure *str = NULL;
+
+ str = gst_caps_get_structure (player->v_stream_caps, 0);
+ if ( !gst_structure_get_int (str, "orientation", &org_angle))
{
- if (player->tee_src_pad[i])
- {
- gst_element_release_request_pad (player->pipeline->videobin[MMPLAYER_V_TEE].gst, player->tee_src_pad[i]);
- gst_object_unref(player->tee_src_pad[i]);
- }
+ debug_log ("missing 'orientation' field in video caps");
}
- player->use_multi_surface = FALSE;
- debug_log("release request pads from TEE, and unref TEE's src pads");
}
- if ( player->sink_elements )
- g_list_free ( player->sink_elements );
- player->sink_elements = NULL;
+ debug_log("check user angle: %d, orientation: %d", user_angle, org_angle);
- /* cleanup unlinked mime type */
- MMPLAYER_FREEIF(player->unlinked_audio_mime);
- MMPLAYER_FREEIF(player->unlinked_video_mime);
- MMPLAYER_FREEIF(player->unlinked_demuxer_mime);
+ /* check video stream callback is used */
+ if(!player->set_mode.media_packet_video_stream && player->use_video_stream )
+ {
+ if (player->set_mode.video_zc)
+ {
+ gchar *ename = NULL;
+ int width = 0;
+ int height = 0;
- /* cleanup running stuffs */
- __mmplayer_cancel_delayed_eos( player );
+ mm_attrs_get_int_by_name(attrs, "display_width", &width);
+ mm_attrs_get_int_by_name(attrs, "display_height", &height);
- /* cleanup gst stuffs */
- if ( player->pipeline )
- {
- MMPlayerGstElement* mainbin = player->pipeline->mainbin;
- GstTagList* tag_list = player->pipeline->tag_list;
- GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (mainbin[MMPLAYER_M_PIPE].gst));
+ /* resize video frame with requested values for fimcconvert */
+ ename = GST_PLUGIN_FEATURE_NAME(gst_element_get_factory(player->pipeline->videobin[MMPLAYER_V_CONV].gst));
- /* first we need to disconnect all signal hander */
- __mmplayer_release_signal_connection( player );
+ if (ename && g_strrstr(ename, "fimcconvert"))
+ {
+ if (width)
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "dst-width", width, NULL);
- /* disconnecting bus watch */
- if ( player->bus_watcher )
- g_source_remove( player->bus_watcher );
- player->bus_watcher = 0;
+ if (height)
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "dst-height", height, NULL);
- gst_bus_set_sync_handler (bus, NULL, NULL);
+ /* NOTE: fimcconvert does not manage index of src buffer from upstream src-plugin, decoder gives frame information in output buffer with no ordering */
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "src-rand-idx", TRUE, NULL);
- if ( mainbin )
- {
- MMPlayerGstElement* audiobin = player->pipeline->audiobin;
- MMPlayerGstElement* videobin = player->pipeline->videobin;
- MMPlayerGstElement* textbin = player->pipeline->textbin;
- MMPlayerGstElement* subtitlebin = player->pipeline->subtitlebin;
+ /* get rotation value to set */
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
- debug_log("pipeline status before set state to NULL\n");
- __mmplayer_dump_pipeline_state( player );
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "rotate", rotation_value, NULL);
- timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
- ret = __mmplayer_gst_set_state ( player, mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_NULL, FALSE, timeout );
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("fail to change state to NULL\n");
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_log("updating fimcconvert - r[%d], w[%d], h[%d]", rotation_value, width, height);
}
+ }
+ else
+ {
+ debug_log("using video stream callback with memsink. player handle : [%p]", player);
- debug_log("pipeline status before unrefering pipeline\n");
- __mmplayer_dump_pipeline_state( player );
+ /* get rotation value to set */
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
- gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_PIPE].gst));
-
- /* free fakesink */
- if ( mainbin[MMPLAYER_M_SRC_FAKESINK].gst )
- gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_SRC_FAKESINK].gst));
-
- /* free avsysaudiosink
- avsysaudiosink should be unref when destory pipeline just after start play with BT.
- Because audiosink is created but never added to bin, and therefore it will not be unref when pipeline is destroyed.
- */
- MMPLAYER_FREEIF( audiobin );
- MMPLAYER_FREEIF( videobin );
- MMPLAYER_FREEIF( textbin );
- MMPLAYER_FREEIF( mainbin );
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_FLIP].gst, "method", rotation_value, NULL);
}
- if ( tag_list )
- gst_tag_list_free(tag_list);
-
- MMPLAYER_FREEIF( player->pipeline );
+ return MM_ERROR_NONE;
}
- player->pipeline_is_constructed = FALSE;
-
- debug_fleave();
-
- return ret;
-}
-
-static int __gst_realize(mm_player_t* player) // @
-{
- gint timeout = 0;
- int ret = MM_ERROR_NONE;
+ /* update display surface */
+ mm_attrs_get_int_by_name(attrs, "display_surface_type", &surface_type);
+ debug_log("check display surface type attribute: %d", surface_type);
- debug_fenter();
+ /* configuring display */
+ switch ( surface_type )
+ {
+ case MM_DISPLAY_SURFACE_X:
+ {
+ /* ximagesink or xvimagesink */
+ void *xid = NULL;
+ double zoom = 0;
+ int display_method = 0;
+ int roi_x = 0;
+ int roi_y = 0;
+ int roi_w = 0;
+ int roi_h = 0;
+ int src_crop_x = 0;
+ int src_crop_y = 0;
+ int src_crop_w = 0;
+ int src_crop_h = 0;
+ int force_aspect_ratio = 0;
+ gboolean visible = TRUE;
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ /* common case if using x surface */
+ mm_attrs_get_data_by_name(attrs, "display_overlay", &xid);
+ if ( xid )
+ {
+ int xwin_id = 0;
+ xwin_id = *(int*)xid;
+ debug_log("set video param : xid %d", *(int*)xid);
+ if (xwin_id)
+ {
+ gst_x_overlay_set_xwindow_id( GST_X_OVERLAY( player->pipeline->videobin[MMPLAYER_V_SINK].gst ), *(int*)xid );
+ }
+ }
+ else
+ {
+ /* FIXIT : is it error case? */
+ debug_warning("still we don't have xid on player attribute. create it's own surface.");
+ }
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_READY;
- MMPLAYER_PRINT_STATE(player);
+ /* if xvimagesink */
+ if (!strcmp(player->ini.videosink_element_x,"xvimagesink"))
+ {
+ mm_attrs_get_int_by_name(attrs, "display_force_aspect_ration", &force_aspect_ratio);
+ mm_attrs_get_double_by_name(attrs, "display_zoom", &zoom);
+ mm_attrs_get_int_by_name(attrs, "display_method", &display_method);
+ mm_attrs_get_int_by_name(attrs, "display_src_crop_x", &src_crop_x);
+ mm_attrs_get_int_by_name(attrs, "display_src_crop_y", &src_crop_y);
+ mm_attrs_get_int_by_name(attrs, "display_src_crop_width", &src_crop_w);
+ mm_attrs_get_int_by_name(attrs, "display_src_crop_height", &src_crop_h);
+ mm_attrs_get_int_by_name(attrs, "display_roi_x", &roi_x);
+ mm_attrs_get_int_by_name(attrs, "display_roi_y", &roi_y);
+ mm_attrs_get_int_by_name(attrs, "display_roi_width", &roi_w);
+ mm_attrs_get_int_by_name(attrs, "display_roi_height", &roi_h);
+ mm_attrs_get_int_by_name(attrs, "display_visible", &visible);
+ #define DEFAULT_DISPLAY_MODE 2 // TV only, PRI_VIDEO_OFF_AND_SEC_VIDEO_FULL_SCREEN
- __ta__("__mmplayer_gst_create_pipeline",
- ret = __mmplayer_gst_create_pipeline(player);
- if ( ret )
- {
- debug_critical("failed to create pipeline\n");
- return ret;
- }
- )
+ /* setting for cropping media source */
+ if (src_crop_w && src_crop_h)
+ {
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "src-crop-x", src_crop_x,
+ "src-crop-y", src_crop_y,
+ "src-crop-w", src_crop_w,
+ "src-crop-h", src_crop_h,
+ NULL );
+ }
- /* set pipeline state to READY */
- /* NOTE : state change to READY must be performed sync. */
- timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_READY, FALSE, timeout);
+ /* setting for ROI mode */
+ if (display_method == 5) // 5 for ROI mode
+ {
+ int roi_mode = 0;
+ mm_attrs_get_int_by_name(attrs, "display_roi_mode", &roi_mode);
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "dst-roi-mode", roi_mode,
+ "dst-roi-x", roi_x,
+ "dst-roi-y", roi_y,
+ "dst-roi-w", roi_w,
+ "dst-roi-h", roi_h,
+ NULL );
+ /* get rotation value to set,
+ do not use org_angle because ROI mode in xvimagesink needs both a rotation value and an orientation value */
+ __mmplayer_get_property_value_for_rotation(player, user_angle, &rotation_value);
+ }
+ else
+ {
+ /* get rotation value to set */
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
+ }
- if (MMPLAYER_PLAY_SUBTITLE(player))
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst, GST_STATE_READY, FALSE, timeout);
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "force-aspect-ratio", force_aspect_ratio,
+ "zoom", (float)zoom,
+ "orientation", org_angle/90, // setting for orientation of media, it is used for ROI/ZOOM feature in xvimagesink
+ "rotate", rotation_value,
+ "handle-events", TRUE,
+ "display-geometry-method", display_method,
+ "draw-borders", FALSE,
+ "visible", visible,
+ "display-mode", DEFAULT_DISPLAY_MODE,
+ NULL );
- if ( ret != MM_ERROR_NONE )
- {
- /* return error if failed to set state */
- debug_error("failed to set state PAUSED (live : READY).\n");
+ debug_log("set video param : zoom %lf, rotate %d, method %d visible %d", zoom, rotation_value, display_method, visible);
+ debug_log("set video param : dst-roi-x: %d, dst-roi-y: %d, dst-roi-w: %d, dst-roi-h: %d", roi_x, roi_y, roi_w, roi_h );
+ debug_log("set video param : force aspect ratio %d, display mode %d", force_aspect_ratio, DEFAULT_DISPLAY_MODE);
+ }
+ }
+ break;
+ case MM_DISPLAY_SURFACE_EVAS:
+ {
+ void *object = NULL;
+ int scaling = 0;
+ gboolean visible = TRUE;
+ int display_method = 0;
- /* dump state of all element */
- __mmplayer_dump_pipeline_state( player );
+ /* common case if using evas surface */
+ mm_attrs_get_data_by_name(attrs, "display_overlay", &object);
+ mm_attrs_get_int_by_name(attrs, "display_visible", &visible);
+ mm_attrs_get_int_by_name(attrs, "display_evas_do_scaling", &scaling);
+ mm_attrs_get_int_by_name(attrs, "display_method", &display_method);
- return ret;
- }
- else
- {
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_READY );
- }
+ /* if evasimagesink */
+ if (!strcmp(player->ini.videosink_element_evas,"evasimagesink"))
+ {
+ if (object)
+ {
+ /* if it is evasimagesink, we are not supporting rotation */
+ if (user_angle_type!=MM_DISPLAY_ROTATION_NONE)
+ {
+ mm_attrs_set_int_by_name(attrs, "display_rotation", MM_DISPLAY_ROTATION_NONE);
+ if (mmf_attrs_commit (attrs)) /* return -1 if error */
+ debug_error("failed to commit\n");
+ debug_warning("unsupported feature");
+ return MM_ERROR_NOT_SUPPORT_API;
+ }
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "evas-object", object,
+ "visible", visible,
+ "display-geometry-method", display_method,
+ "rotate", rotation_value,
+ NULL);
+ debug_log("set video param : method %d", display_method);
+ debug_log("set video param : evas-object %x, visible %d", object, visible);
+ debug_log("set video param : evas-object %x, rotate %d", object, rotation_value);
+ }
+ else
+ {
+ debug_error("no evas object");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- /* create dot before error-return. for debugging */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-realize" );
- debug_fleave();
+ /* if evasimagesink using converter */
+ if (player->set_mode.video_zc && player->pipeline->videobin[MMPLAYER_V_CONV].gst)
+ {
+ int width = 0;
+ int height = 0;
+ int no_scaling = !scaling;
- return ret;
-}
+ mm_attrs_get_int_by_name(attrs, "display_width", &width);
+ mm_attrs_get_int_by_name(attrs, "display_height", &height);
-static int __gst_unrealize(mm_player_t* player) // @
-{
- int ret = MM_ERROR_NONE;
+ /* NOTE: fimcconvert does not manage index of src buffer from upstream src-plugin, decoder gives frame information in output buffer with no ordering */
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "src-rand-idx", TRUE, NULL);
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "dst-buffer-num", 5, NULL);
- debug_fenter();
+ if (no_scaling)
+ {
+ /* no-scaling order to fimcconvert, original width, height size of media src will be passed to sink plugin */
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst,
+ "dst-width", 0, /* setting 0, output video width will be media src's width */
+ "dst-height", 0, /* setting 0, output video height will be media src's height */
+ NULL);
+ }
+ else
+ {
+ /* scaling order to fimcconvert */
+ if (width)
+ {
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "dst-width", width, NULL);
+ }
+ if (height)
+ {
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_CONV].gst, "dst-height", height, NULL);
+ }
+ debug_log("set video param : video frame scaling down to width(%d) height(%d)", width, height);
+ }
+ debug_log("set video param : display_evas_do_scaling %d", scaling);
+ }
+ }
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ /* if evaspixmapsink */
+ if (!strcmp(player->ini.videosink_element_evas,"evaspixmapsink"))
+ {
+ if (object)
+ {
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "evas-object", object,
+ "visible", visible,
+ "display-geometry-method", display_method,
+ "rotate", rotation_value,
+ NULL);
+ debug_log("set video param : method %d", display_method);
+ debug_log("set video param : evas-object %x, visible %d", object, visible);
+ debug_log("set video param : evas-object %x, rotate %d", object, rotation_value);
+ }
+ else
+ {
+ debug_error("no evas object");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NULL;
- MMPLAYER_PRINT_STATE(player);
+ int display_method = 0;
+ int roi_x = 0;
+ int roi_y = 0;
+ int roi_w = 0;
+ int roi_h = 0;
+ int origin_size = !scaling;
- player->use_video_stream = FALSE;
- player->video_stream_cb = NULL;
- player->video_stream_cb_user_param = NULL;
+ mm_attrs_get_int_by_name(attrs, "display_method", &display_method);
+ mm_attrs_get_int_by_name(attrs, "display_roi_x", &roi_x);
+ mm_attrs_get_int_by_name(attrs, "display_roi_y", &roi_y);
+ mm_attrs_get_int_by_name(attrs, "display_roi_width", &roi_w);
+ mm_attrs_get_int_by_name(attrs, "display_roi_height", &roi_h);
- player->audio_stream_cb = NULL;
- player->audio_stream_cb_user_param = NULL;
+ /* get rotation value to set */
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
- player->audio_buffer_cb = NULL;
- player->audio_buffer_cb_user_param = NULL;
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "origin-size", origin_size,
+ "rotate", rotation_value,
+ "dst-roi-x", roi_x,
+ "dst-roi-y", roi_y,
+ "dst-roi-w", roi_w,
+ "dst-roi-h", roi_h,
+ "display-geometry-method", display_method,
+ NULL );
- player->sent_bos = FALSE;
- player->playback_rate = DEFAULT_PLAYBACK_RATE;
+ debug_log("set video param : method %d", display_method);
+ debug_log("set video param : dst-roi-x: %d, dst-roi-y: %d, dst-roi-w: %d, dst-roi-h: %d",
+ roi_x, roi_y, roi_w, roi_h );
+ debug_log("set video param : display_evas_do_scaling %d (origin-size %d)", scaling, origin_size);
+ }
+ }
+ break;
+ case MM_DISPLAY_SURFACE_X_EXT: /* NOTE : this surface type is used for the videoTexture(canvasTexture) overlay */
+ {
+ void *pixmap_id_cb = NULL;
+ void *pixmap_id_cb_user_data = NULL;
+ int display_method = 0;
+ gboolean visible = TRUE;
- /* clean found parsers */
- if (player->parsers)
- {
- g_list_free(player->parsers);
- player->parsers = NULL;
- }
+ /* if xvimagesink */
+ if (strcmp(player->ini.videosink_element_x,"xvimagesink"))
+ {
+ debug_error("videosink is not xvimagesink");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- MMPLAYER_FREEIF(player->album_art);
+ /* get information from attributes */
+ mm_attrs_get_data_by_name(attrs, "display_overlay", &pixmap_id_cb);
+ mm_attrs_get_data_by_name(attrs, "display_overlay_user_data", &pixmap_id_cb_user_data);
+ mm_attrs_get_int_by_name(attrs, "display_method", &display_method);
+ mm_attrs_get_int_by_name(attrs, "display_visible", &visible);
- /* destroy pipeline */
- ret = __mmplayer_gst_destroy_pipeline( player );
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("failed to destory pipeline\n");
- return ret;
- }
- else
- {
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_NULL );
+ if ( pixmap_id_cb )
+ {
+ debug_log("set video param : display_overlay(0x%x)", pixmap_id_cb);
+ if (pixmap_id_cb_user_data)
+ {
+ debug_log("set video param : display_overlay_user_data(0x%x)", pixmap_id_cb_user_data);
+ }
+ }
+ else
+ {
+ debug_error("failed to set pixmap-id-callback");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ /* get rotation value to set */
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
+
+ debug_log("set video param : rotate %d, method %d, visible %d", rotation_value, display_method, visible);
+
+ /* set properties of videosink plugin */
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "display-geometry-method", display_method,
+ "draw-borders", FALSE,
+ "visible", visible,
+ "rotate", rotation_value,
+ "pixmap-id-callback", pixmap_id_cb,
+ "pixmap-id-callback-userdata", pixmap_id_cb_user_data,
+ NULL );
+ }
+ break;
+ case MM_DISPLAY_SURFACE_NULL:
+ {
+ /* do nothing */
+ }
+ break;
}
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return ret;
+ return MM_ERROR_NONE;
}
-static int __gst_pending_seek ( mm_player_t* player )
+static int
+__mmplayer_gst_element_link_bucket(GList* element_bucket) // @
{
- MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
- MMPlayerStateType pending_state = MM_PLAYER_STATE_NONE;
- int ret = MM_ERROR_NONE;
-
- debug_fenter();
+ GList* bucket = element_bucket;
+ MMPlayerGstElement* element = NULL;
+ MMPlayerGstElement* prv_element = NULL;
+ gint successful_link_count = 0;
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ MMPLAYER_FENTER();
- if ( !player->pending_seek.is_pending )
- {
- debug_log("pending seek is not reserved. nothing to do.\n" );
- return ret;
- }
+ return_val_if_fail(element_bucket, -1);
- /* check player state if player could pending seek or not. */
- current_state = MMPLAYER_CURRENT_STATE(player);
- pending_state = MMPLAYER_PENDING_STATE(player);
+ prv_element = (MMPlayerGstElement*)bucket->data;
+ bucket = bucket->next;
- if ( current_state != MM_PLAYER_STATE_PAUSED && current_state != MM_PLAYER_STATE_PLAYING )
+ for ( ; bucket; bucket = bucket->next )
{
- debug_warning("try to pending seek in %s state, try next time. \n",
- MMPLAYER_STATE_GET_NAME(current_state));
- return ret;
- }
-
- debug_log("trying to play from (%lu) pending position\n", player->pending_seek.pos);
-
- ret = __gst_set_position ( player, player->pending_seek.format, player->pending_seek.pos );
-
- if ( MM_ERROR_NONE != ret )
- debug_error("failed to seek pending postion. just keep staying current position.\n");
+ element = (MMPlayerGstElement*)bucket->data;
- player->pending_seek.is_pending = FALSE;
-
- debug_fleave();
-
- return ret;
-}
-
-
-static int __gst_start(mm_player_t* player) // @
-{
- gboolean sound_extraction = 0;
- gboolean async = FALSE;
- gint timeout = 0;
- int ret = MM_ERROR_NONE;
-
- debug_fenter();
-
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
-
- /* get sound_extraction property */
- mm_attrs_get_int_by_name(player->attrs, "pcm_extraction", &sound_extraction);
- /* NOTE : if SetPosition was called before Start. do it now */
- /* streaming doesn't support it. so it should be always sync */
- /* !! create one more api to check if there is pending seek rather than checking variables */
-
- if ( (player->pending_seek.is_pending || sound_extraction) && !MMPLAYER_IS_STREAMING(player))
- {
- ret = __gst_pause(player, FALSE);
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("failed to set state to PAUSED for pending seek\n");
- return ret;
- }
-
- if ( sound_extraction )
+ if ( element && element->gst )
{
- debug_log("setting pcm extraction\n");
-
- ret = __mmplayer_set_pcm_extraction(player);
- if ( MM_ERROR_NONE != ret )
+ if ( GST_ELEMENT_LINK(GST_ELEMENT(prv_element->gst), GST_ELEMENT(element->gst)) )
{
- debug_warning("failed to set pcm extraction\n");
- return ret;
+ debug_log("linking [%s] to [%s] success\n",
+ GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
+ GST_ELEMENT_NAME(GST_ELEMENT(element->gst)) );
+ successful_link_count ++;
}
- }
- else
- {
- if ( MM_ERROR_NONE != __gst_pending_seek(player) )
+ else
{
- debug_warning("failed to seek pending postion. starting from the begin of content.\n");
+ debug_log("linking [%s] to [%s] failed\n",
+ GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
+ GST_ELEMENT_NAME(GST_ELEMENT(element->gst)) );
+ return -1;
}
}
- }
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PLAYING;
+ prv_element = element;
+ }
- MMPLAYER_PRINT_STATE(player);
+ MMPLAYER_FLEAVE();
- /* NOTE : note that current variant of rtspsrc has their own buffering mechanism.
- * their keeping rtp packets without gst timestamping. there's no easy way to export buffering mechanism to plugin-level. and
- * application also not ready. thus, just for now. state transition to PLAYING will done in asyn mode.
- * please do not confused with async property hack which done by __gst_set_async_state_change()
- */
- /* NOTE : testing async start when local playback also */
+ return successful_link_count;
+}
- mm_attrs_get_int_by_name(player->attrs,"profile_async_start", &async);
- debug_log("start mode : %s", (async ? "async" : "sync"));
+static int
+__mmplayer_gst_element_add_bucket_to_bin(GstBin* bin, GList* element_bucket) // @
+{
+ GList* bucket = element_bucket;
+ MMPlayerGstElement* element = NULL;
+ int successful_add_count = 0;
- /* set pipeline state to PLAYING */
- timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PLAYING, async, timeout );
+ MMPLAYER_FENTER();
- if (MMPLAYER_PLAY_SUBTITLE(player))
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst, GST_STATE_PLAYING, FALSE, timeout );
+ return_val_if_fail(element_bucket, 0);
+ return_val_if_fail(bin, 0);
- if (ret == MM_ERROR_NONE)
+ for ( ; bucket; bucket = bucket->next )
{
- if (!async)
- {
- guint duration = 0;
- debug_log("sync state changing is completed");
+ element = (MMPlayerGstElement*)bucket->data;
- mm_attrs_get_int_by_name(player->attrs, "content_duration", &duration);
- if (!duration)
+ if ( element && element->gst )
+ {
+ if( !gst_bin_add(bin, GST_ELEMENT(element->gst)) )
{
- debug_warning("try to update duration more");
- player->need_update_content_attrs = TRUE;
- player->need_update_content_dur = TRUE;
- _mmplayer_update_content_attrs(player);
+ debug_log("__mmplayer_gst_element_link_bucket : Adding element [%s] to bin [%s] failed\n",
+ GST_ELEMENT_NAME(GST_ELEMENT(element->gst)),
+ GST_ELEMENT_NAME(GST_ELEMENT(bin) ) );
+ return 0;
}
-
- MMPLAYER_SET_STATE(player, MM_PLAYER_STATE_PLAYING);
+ successful_add_count ++;
}
}
- else
- {
- debug_error("failed to set state to PLAYING");
-
- /* dump state of all element */
- __mmplayer_dump_pipeline_state( player );
-
- return ret;
- }
-
- /* FIXIT : analyze so called "async problem" */
- /* set async off */
- __gst_set_async_state_change( player, FALSE );
-
- /* generating debug info before returning error */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-start" );
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return ret;
+ return successful_add_count;
}
-static void __mmplayer_do_sound_fadedown(mm_player_t* player, unsigned int time)
-{
- debug_fenter();
-
- return_if_fail(player
- && player->pipeline
- && player->pipeline->audiobin
- && player->pipeline->audiobin[MMPLAYER_A_SINK].gst);
- g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "mute", 2, NULL);
-
- usleep(time);
- debug_fleave();
-}
+/**
+ * This function is to create audio pipeline for playing.
+ *
+ * @param player [in] handle of player
+ *
+ * @return This function returns zero on success.
+ * @remark
+ * @see __mmplayer_gst_create_midi_pipeline, __mmplayer_gst_create_video_pipeline
+ */
+#define MMPLAYER_CREATEONLY_ELEMENT(x_bin, x_id, x_factory, x_name) \
+x_bin[x_id].id = x_id;\
+x_bin[x_id].gst = gst_element_factory_make(x_factory, x_name);\
+if ( ! x_bin[x_id].gst )\
+{\
+ debug_error("failed to create %s \n", x_factory);\
+ goto ERROR;\
+}\
-static void __mmplayer_undo_sound_fadedown(mm_player_t* player)
-{
- debug_fenter();
-
- return_if_fail(player
- && player->pipeline
- && player->pipeline->audiobin
- && player->pipeline->audiobin[MMPLAYER_A_SINK].gst);
-
- g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "mute", 0, NULL);
+#define MMPLAYER_CREATE_ELEMENT_ADD_BIN(x_bin, x_id, x_factory, x_name, y_bin, x_player) \
+x_bin[x_id].id = x_id;\
+x_bin[x_id].gst = gst_element_factory_make(x_factory, x_name);\
+if ( ! x_bin[x_id].gst )\
+{\
+ debug_error("failed to create %s \n", x_factory);\
+ goto ERROR;\
+}\
+else\
+{\
+ if (x_player->ini.set_dump_element_flag)\
+ __mmplayer_add_dump_buffer_probe(x_player, x_bin[x_id].gst);\
+}\
+if( !gst_bin_add(GST_BIN(y_bin), GST_ELEMENT(x_bin[x_id].gst)))\
+{\
+ debug_log("__mmplayer_gst_element_link_bucket : Adding element [%s] to bin [%s] failed\n",\
+ GST_ELEMENT_NAME(GST_ELEMENT(x_bin[x_id].gst)),\
+ GST_ELEMENT_NAME(GST_ELEMENT(y_bin) ) );\
+ goto ERROR;\
+}\
- debug_fleave();
-}
+/* macro for code readability. just for sinkbin-creation functions */
+#define MMPLAYER_CREATE_ELEMENT(x_bin, x_id, x_factory, x_name, x_add_bucket, x_player) \
+do \
+{ \
+ x_bin[x_id].id = x_id;\
+ x_bin[x_id].gst = gst_element_factory_make(x_factory, x_name);\
+ if ( ! x_bin[x_id].gst )\
+ {\
+ debug_error("failed to create %s \n", x_factory);\
+ goto ERROR;\
+ }\
+ else\
+ {\
+ if (x_player->ini.set_dump_element_flag)\
+ __mmplayer_add_dump_buffer_probe(x_player, x_bin[x_id].gst);\
+ }\
+ if ( x_add_bucket )\
+ element_bucket = g_list_append(element_bucket, &x_bin[x_id]);\
+} while(0);
-static int __gst_stop(mm_player_t* player) // @
+/**
+ * AUDIO PIPELINE
+ * - Local playback : audiotp !audioconvert ! volume ! capsfilter ! audioeffect ! audioeffect_sec ! audiosink
+ * - Streaming : audiotp !audioconvert !volume ! audiosink
+ * - PCM extraction : audiotp !audioconvert ! audioresample ! capsfilter ! fakesink
+ */
+static int
+__mmplayer_gst_create_audio_pipeline(mm_player_t* player)
{
- GstStateChangeReturn change_ret = GST_STATE_CHANGE_SUCCESS;
+ MMPlayerGstElement* first_element = NULL;
+ MMPlayerGstElement* audiobin = NULL;
MMHandleType attrs = 0;
- gboolean fadewown = FALSE;
- gboolean rewind = FALSE;
- gint timeout = 0;
- int ret = MM_ERROR_NONE;
-
- debug_fenter();
+ GstPad *pad = NULL;
+ GstPad *ghostpad = NULL;
+ GList* element_bucket = NULL;
+ gboolean link_audio_sink_now = TRUE;
+ int i =0;
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ MMPLAYER_FENTER();
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_READY;
- MMPLAYER_PRINT_STATE(player);
+ return_val_if_fail( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
+ /* alloc handles */
+ audiobin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_A_NUM);
+ if ( ! audiobin )
{
- debug_error("cannot get content attribute\n");
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_error("failed to allocate memory for audiobin\n");
+ return MM_ERROR_PLAYER_NO_FREE_SPACE;
}
- mm_attrs_get_int_by_name(attrs,"sound_fadedown", &fadewown);
-
- /* enable fadedown */
- if (fadewown)
- __mmplayer_do_sound_fadedown(player, MM_PLAYER_FADEOUT_TIME_DEFAULT);
+ attrs = MMPLAYER_GET_ATTRS(player);
- /* Just set state to PAUESED and the rewind. it's usual player behavior. */
- timeout = MMPLAYER_STATE_CHANGE_TIMEOUT ( player );
- if ( player->profile.uri_type == MM_PLAYER_URI_TYPE_BUFF )
+ /* create bin */
+ audiobin[MMPLAYER_A_BIN].id = MMPLAYER_A_BIN;
+ audiobin[MMPLAYER_A_BIN].gst = gst_bin_new("audiobin");
+ if ( !audiobin[MMPLAYER_A_BIN].gst )
{
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_READY, FALSE, timeout );
+ debug_error("failed to create audiobin\n");
+ goto ERROR;
}
- else
- {
- ret = __mmplayer_gst_set_state( player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PAUSED, FALSE, timeout );
- if (MMPLAYER_PLAY_SUBTITLE(player))
- ret = __mmplayer_gst_set_state( player,
- player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst, GST_STATE_PAUSED, FALSE, timeout );
+ /* take it */
+ player->pipeline->audiobin = audiobin;
- if ( !MMPLAYER_IS_STREAMING(player))
- rewind = TRUE;
- }
+ player->set_mode.pcm_extraction = __mmplayer_can_extract_pcm(player);
- /* disable fadeout */
- if (fadewown)
- __mmplayer_undo_sound_fadedown(player);
+ /* Adding audiotp plugin for reverse trickplay feature */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_TP, "audiotp", "audio trickplay", TRUE, player);
+
+ /* converter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV, "audioconvert", "audio converter", TRUE, player);
+ /* resampler */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_RESAMPLER, player->ini.name_of_audio_resampler, "audio resampler", TRUE, player);
- /* return if set_state has failed */
- if ( ret != MM_ERROR_NONE )
+ if (player->set_mode.pcm_extraction) // pcm extraction only and no sound output
{
- debug_error("failed to set state.\n");
+ int dst_samplerate = 0;
+ int dst_channels = 0;
+ int dst_depth = 0;
+ char *caps_str = NULL;
+ GstCaps* caps = NULL;
- /* dump state of all element. don't care it success or not */
- __mmplayer_dump_pipeline_state( player );
+ /* get conf. values */
+ mm_attrs_multiple_get(player->attrs,
+ NULL,
+ "pcm_extraction_samplerate", &dst_samplerate,
+ "pcm_extraction_channels", &dst_channels,
+ "pcm_extraction_depth", &dst_depth,
+ NULL);
- return ret;
- }
+ /* capsfilter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_DEFAULT, "capsfilter", "audio capsfilter", TRUE, player);
+ caps = gst_caps_new_simple ("audio/x-raw-int",
+ "rate", G_TYPE_INT, dst_samplerate,
+ "channels", G_TYPE_INT, dst_channels,
+ "depth", G_TYPE_INT, dst_depth,
+ NULL);
+ caps_str = gst_caps_to_string(caps);
+ debug_log("new caps : %s\n", caps_str);
- /* rewind */
- if ( rewind )
- {
- if ( ! __gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, 1.0,
- GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, 0,
- GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE) )
- {
- debug_warning("failed to rewind\n");
- ret = MM_ERROR_PLAYER_SEEK;
- }
- }
+ g_object_set (GST_ELEMENT(audiobin[MMPLAYER_A_CAPS_DEFAULT].gst), "caps", caps, NULL );
- /* initialize */
- player->sent_bos = FALSE;
+ /* clean */
+ gst_caps_unref( caps );
+ MMPLAYER_FREEIF( caps_str );
- /* wait for seek to complete */
- change_ret = gst_element_get_state (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, NULL, NULL, timeout * GST_SECOND);
- if (MMPLAYER_PLAY_SUBTITLE(player))
- change_ret = gst_element_get_state (player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst, NULL, NULL, timeout * GST_SECOND);
+ /* fake sink */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, "fakesink", "fakesink", TRUE, player);
- if ( change_ret == GST_STATE_CHANGE_SUCCESS || change_ret == GST_STATE_CHANGE_NO_PREROLL )
- {
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_READY );
+ /* set sync */
+ g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "sync", FALSE, NULL);
}
- else
+ else // normal playback
{
- debug_error("fail to stop player.\n");
- ret = MM_ERROR_PLAYER_INTERNAL;
- }
-
- /* generate dot file if enabled */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-stop" );
+ typedef enum
+ {
+ GST_BASE_AUDIO_SINK_SLAVE_RESAMPLE = 0,
+ GST_BASE_AUDIO_SINK_SLAVE_SKEW,
+ GST_BASE_AUDIO_SINK_SLAVE_NONE
+ } GstBaseAudioSinkSlaveMethod;
- debug_fleave();
+ //GstCaps* caps = NULL;
+ gint channels = 0;
- return ret;
-}
+ /* for logical volume control */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_VOL, "volume", "volume", TRUE, player);
+ g_object_set(G_OBJECT (audiobin[MMPLAYER_A_VOL].gst), "volume", player->sound.volume, NULL);
+ if (player->sound.mute)
+ {
+ debug_log("mute enabled\n");
+ g_object_set(G_OBJECT (audiobin[MMPLAYER_A_VOL].gst), "mute", player->sound.mute, NULL);
+ }
-int __gst_pause(mm_player_t* player, gboolean async) // @
-{
- gint timeout = 0;
- int ret = MM_ERROR_NONE;
+#if 0
+ /*capsfilter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_DEFAULT, "capsfilter", "audiocapsfilter", TRUE, player);
+ caps = gst_caps_from_string( "audio/x-raw-int, "
+ "endianness = (int) LITTLE_ENDIAN, "
+ "signed = (boolean) true, "
+ "width = (int) 16, "
+ "depth = (int) 16" );
+ g_object_set (GST_ELEMENT(audiobin[MMPLAYER_A_CAPS_DEFAULT].gst), "caps", caps, NULL );
+ gst_caps_unref( caps );
+#endif
- debug_fenter();
+ /* chech if multi-chennels */
+ if (player->pipeline->mainbin && player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst)
+ {
+ GstPad *srcpad = NULL;
+ GstCaps *caps = NULL;
- return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ if ((srcpad = gst_element_get_static_pad(player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst, "src")))
+ {
+ if ((caps = gst_pad_get_caps(srcpad)))
+ {
+ //MMPLAYER_LOG_GST_CAPS_TYPE(caps);
+ GstStructure *str = gst_caps_get_structure(caps, 0);
+ if (str)
+ gst_structure_get_int (str, "channels", &channels);
+ gst_caps_unref(caps);
+ }
+ gst_object_unref(srcpad);
+ }
+ }
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PAUSED;
- MMPLAYER_PRINT_STATE(player);
+ /* audio effect element. if audio effect is enabled */
+ if ( (strcmp(player->ini.name_of_audio_effect, ""))
+ && (channels <= 2)
+ && player->ini.use_audio_effect_custom)
+ {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER, player->ini.name_of_audio_effect, "audio effect filter", TRUE, player);
- if ( async )
- debug_log("do async state transition to PAUSE.\n");
+ debug_log("audio effect config. bypass = %d, effect type = %d", player->bypass_audio_effect, player->audio_effect_info.effect_type);
- /* set pipeline status to PAUSED */
- timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PAUSED, async, timeout );
+ if ( (!player->bypass_audio_effect)
+ && player->ini.use_audio_effect_custom)
+ {
+ if ( MM_AUDIO_EFFECT_TYPE_CUSTOM == player->audio_effect_info.effect_type )
+ {
+ if (!_mmplayer_audio_effect_custom_apply(player))
+ {
+ debug_msg("apply audio effect(custom) setting success\n");
+ }
+ }
+ }
- if (MMPLAYER_PLAY_SUBTITLE(player))
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst, GST_STATE_PAUSED, async, timeout );
+ if ( (strcmp(player->ini.name_of_audio_effect_sec, ""))
+ && (player->set_mode.rich_audio) )
+ {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER_SEC, player->ini.name_of_audio_effect_sec, "audio effect filter", TRUE, player);
+ }
+ }
+ if (!MMPLAYER_IS_RTSP_STREAMING(player))
+ {
+ if (player->set_mode.rich_audio && channels <= 2)
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_VSP, "audiovsp", "x-speed", TRUE, player);
+ }
- /* NOTE : here we are setting state PAUSED to streaming source element again. because
- * main pipeline will not set the state of it's all childs if state of the pipeline
- * is already PAUSED for some reason. this situaition can happen when rebuffering or
- * buffering after seeking. in both case streaming source will send flush event and then
- * sink elements will lost it's state and it will set the state to PAUSED by theyself.
- */
- if ( MMPLAYER_IS_RTSP_STREAMING( player ) && player->state_lost )
- {
- debug_log("setting PAUSED to streaming source again.\n");
- gst_element_set_state(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, GST_STATE_PAUSED);
- }
+ /* create audio sink */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, player->ini.name_of_audiosink, "audiosink", link_audio_sink_now, player);
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("failed to set state to PAUSED\n");
+ /* qos on */
+ g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "qos", TRUE, NULL); /* qos on */
+ g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "slave-method", GST_BASE_AUDIO_SINK_SLAVE_NONE, NULL);
- /* dump state of all element */
- __mmplayer_dump_pipeline_state( player );
+ /* FIXIT : using system clock. isn't there another way? */
+ if (player->videodec_linked)
+ {
+ debug_log("provide clock for movie = %s", (player->ini.provide_clock_for_movie)?"audio_clock":"sys_clock");
+ g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "provide-clock", player->ini.provide_clock_for_movie, NULL);
+ }
+ else
+ {
+ debug_log("provide clock for music = %s", (player->ini.provide_clock_for_music)?"audio_clock":"sys_clock");
+ g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "provide-clock", player->ini.provide_clock_for_music, NULL);
+ }
- return ret;
- }
- else
- {
- if ( async == FALSE )
+ if ( g_strrstr(player->ini.name_of_audiosink, "avsysaudiosink") || g_strrstr(player->ini.name_of_audiosink, "pulsesink") )
{
- debug_log("sync state changing is completed.");
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PAUSED );
+ gint volume_type = 0;
+ gint audio_route = 0;
+ gint sound_priority = FALSE;
+ gint route_path = 0;
+ gint latency_mode = 0;
+ gint close_handle = 0;
+
+ /* set volume table
+ * It should be set after player creation through attribute.
+ * But, it can not be changed during playing.
+ */
+ mm_attrs_get_int_by_name(attrs, "sound_volume_type", &volume_type);
+ mm_attrs_get_int_by_name(attrs, "sound_route", &audio_route);
+ mm_attrs_get_int_by_name(attrs, "sound_priority", &sound_priority);
+ mm_attrs_get_int_by_name(attrs, "sound_spk_out_only", &route_path);
+ mm_attrs_get_int_by_name(attrs, "sound_latency_mode", &latency_mode);
+
+ if (player->sm.user_route_policy != 0) {
+ route_path = player->sm.user_route_policy;
+ }
+
+ g_object_set(audiobin[MMPLAYER_A_SINK].gst,
+ "volumetype", volume_type,
+ "audio-route", audio_route,
+ "priority", sound_priority,
+ "user-route", route_path,
+ "latency", latency_mode,
+ "close-handle-on-prepare", close_handle,
+ NULL);
+
+ debug_log("audiosink property status...volume type:%d, route:%d, priority=%d, user-route=%d, latency=%d, close-resource=%d\n",
+ volume_type, audio_route, sound_priority, route_path, latency_mode, close_handle);
}
- }
- /* FIXIT : analyze so called "async problem" */
- /* set async off */
- __gst_set_async_state_change( player, TRUE);
+ /* Antishock can be enabled when player is resumed by soundCM.
+ * But, it's not used in MMS, setting and etc.
+ * Because, player start seems like late.
+ */
+ __mmplayer_set_antishock( player , FALSE );
+ }
+ __mmplayer_add_sink( player, audiobin[MMPLAYER_A_SINK].gst );
- /* generate dot file before returning error */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-pause" );
+ /* adding created elements to bin */
+ debug_log("adding created elements to bin\n");
+ if( !__mmplayer_gst_element_add_bucket_to_bin( GST_BIN(audiobin[MMPLAYER_A_BIN].gst), element_bucket ))
+ {
+ debug_error("failed to add elements\n");
+ goto ERROR;
+ }
- debug_fleave();
+ /* linking elements in the bucket by added order. */
+ debug_log("Linking elements in the bucket by added order.\n");
+ if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ {
+ debug_error("failed to link elements\n");
+ goto ERROR;
+ }
- return ret;
-}
+ /* get first element's sinkpad for creating ghostpad */
+ first_element = (MMPlayerGstElement *)element_bucket->data;
-int __gst_resume(mm_player_t* player, gboolean async) // @
-{
- int ret = MM_ERROR_NONE;
- gint timeout = 0;
+ pad = gst_element_get_static_pad(GST_ELEMENT(first_element->gst), "sink");
+ if ( ! pad )
+ {
+ debug_error("failed to get pad from first element of audiobin\n");
+ goto ERROR;
+ }
- debug_fenter();
+ ghostpad = gst_ghost_pad_new("sink", pad);
+ if ( ! ghostpad )
+ {
+ debug_error("failed to create ghostpad\n");
+ goto ERROR;
+ }
- return_val_if_fail(player && player->pipeline,
- MM_ERROR_PLAYER_NOT_INITIALIZED);
+ if ( FALSE == gst_element_add_pad(audiobin[MMPLAYER_A_BIN].gst, ghostpad) )
+ {
+ debug_error("failed to add ghostpad to audiobin\n");
+ goto ERROR;
+ }
+ gst_object_unref(pad);
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PLAYING;
- MMPLAYER_PRINT_STATE(player);
+ g_list_free(element_bucket);
- /* generate dot file before returning error */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-resume" );
+ mm_attrs_set_int_by_name(attrs, "content_audio_found", TRUE);
- __mmplayer_set_antishock( player , FALSE );
+ MMPLAYER_FLEAVE();
- if ( async )
- debug_log("do async state transition to PLAYING.\n");
+ return MM_ERROR_NONE;
- /* set pipeline state to PLAYING */
- timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PLAYING, async, timeout );
+ERROR:
- if (MMPLAYER_PLAY_SUBTITLE(player))
- ret = __mmplayer_gst_set_state(player,
- player->pipeline->subtitlebin[MMPLAYER_SUB_PIPE].gst, GST_STATE_PLAYING, async, timeout );
+ debug_log("ERROR : releasing audiobin\n");
- /* NOTE : same reason when pausing */
- if ( MMPLAYER_IS_RTSP_STREAMING(player) && player->state_lost )
- {
- debug_log("setting PLAYING to streaming source again.\n");
- gst_element_set_state(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, GST_STATE_PLAYING);
- }
+ if ( pad )
+ gst_object_unref(GST_OBJECT(pad));
- if (ret != MM_ERROR_NONE)
- {
- debug_error("failed to set state to PLAYING\n");
+ if ( ghostpad )
+ gst_object_unref(GST_OBJECT(ghostpad));
- /* dump state of all element */
- __mmplayer_dump_pipeline_state( player );
+ g_list_free( element_bucket );
- return ret;
- }
- else
+ /* release element which are not added to bin */
+ for ( i = 1; i < MMPLAYER_A_NUM; i++ ) /* NOTE : skip bin */
{
- if (async == FALSE)
- {
- debug_log("sync state changing is completed.");
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PLAYING );
+ if ( audiobin[i].gst )
+ {
+ GstObject* parent = NULL;
+ parent = gst_element_get_parent( audiobin[i].gst );
+
+ if ( !parent )
+ {
+ gst_object_unref(GST_OBJECT(audiobin[i].gst));
+ audiobin[i].gst = NULL;
+ }
+ else
+ {
+ gst_object_unref(GST_OBJECT(parent));
+ }
}
}
-
- /* FIXIT : analyze so called "async problem" */
- /* set async off */
- __gst_set_async_state_change( player, FALSE );
- /* generate dot file before returning error */
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-resume" );
+ /* release audiobin with it's childs */
+ if ( audiobin[MMPLAYER_A_BIN].gst )
+ {
+ gst_object_unref(GST_OBJECT(audiobin[MMPLAYER_A_BIN].gst));
+ }
- debug_fleave();
+ MMPLAYER_FREEIF( audiobin );
- return ret;
+ player->pipeline->audiobin = NULL;
+
+ return MM_ERROR_PLAYER_INTERNAL;
}
-static int
-__gst_set_position(mm_player_t* player, int format, unsigned long position) // @
+static gboolean
+__mmplayer_audio_stream_probe (GstPad *pad, GstBuffer *buffer, gpointer u_data)
{
- MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
- MMPlayerStateType pending_state = MM_PLAYER_STATE_NONE;
- GstFormat fmt = GST_FORMAT_TIME;
- unsigned long dur = 0;
- gint64 dur_msec = 0;
- gint64 pos_msec = 0;
- gboolean ret = TRUE;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*) u_data;
+ gint size;
+ guint8 *data;
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( !MMPLAYER_IS_LIVE_STREAMING(player), MM_ERROR_PLAYER_NO_OP );
+ data = GST_BUFFER_DATA(buffer);
+ size = GST_BUFFER_SIZE(buffer);
- /* check player state if player could seek or not. */
- current_state = MMPLAYER_CURRENT_STATE(player);
- pending_state = MMPLAYER_PENDING_STATE(player);
-
- if ( current_state != MM_PLAYER_STATE_PLAYING && current_state != MM_PLAYER_STATE_PAUSED )
- goto PENDING;
+ if (player->audio_stream_cb && size && data)
+ player->audio_stream_cb((void *)data, size, player->audio_stream_cb_user_param);
- /* check duration */
- /* NOTE : duration cannot be zero except live streaming.
- * Since some element could have some timing problemn with quering duration, try again.
- */
- if ( player->duration == 0 )
- {
- ret = gst_element_query_duration( player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &dur_msec );
- if ( !ret )
- goto SEEK_ERROR;
- else
- player->duration = dur_msec;
- }
+ return TRUE;
+}
- if ( player->duration )
- {
- dur = GST_TIME_AS_MSECONDS(player->duration);
- debug_log("duration is %d [msec]. \n", dur);
- }
- else
- {
- debug_error("could not get the duration. fail to seek.\n");
- goto SEEK_ERROR;
+static gboolean
+__mmplayer_video_stream_probe (GstPad *pad, GstBuffer *buffer, gpointer user_data)
+{
+ GstCaps *caps = NULL;
+ MMPlayerVideoStreamDataType stream;
+ MMPlayerMPlaneImage *scmn_imgb = NULL;
+ GstStructure *structure = NULL;
+ unsigned int fourcc = 0;
+ mm_player_t* player = (mm_player_t*)user_data;
+
+ caps = gst_buffer_get_caps(buffer);
+ if (caps == NULL) {
+ debug_error( "Caps is NULL." );
+ return TRUE;
}
-
- /* do seek */
- switch ( format )
- {
- case MM_PLAYER_POS_FORMAT_TIME:
- {
- /* check position is valid or not */
- if ( position > dur )
- goto INVALID_ARGS;
- debug_log("seeking to (%lu) msec\n", position);
+ //MMPLAYER_LOG_GST_CAPS_TYPE(caps);
- if (player->doing_seek)
- {
- debug_log("not completed seek");
- return MM_ERROR_PLAYER_DOING_SEEK;
- }
- player->doing_seek = TRUE;
+ /* clear stream data structure */
+ memset(&stream, 0x0, sizeof(MMPlayerVideoStreamDataType));
- pos_msec = position * G_GINT64_CONSTANT(1000000);
- ret = __gst_seek ( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, 1.0,
- GST_FORMAT_TIME, ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- GST_SEEK_TYPE_SET, pos_msec, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE );
+ structure = gst_caps_get_structure( caps, 0 );
+ gst_structure_get_int(structure, "width", &(stream.width));
+ gst_structure_get_int(structure, "height", &(stream.height));
+ gst_structure_get_fourcc(structure, "format", &fourcc);
+ stream.format = util_get_pixtype(fourcc);
+ gst_caps_unref( caps );
+ caps = NULL;
- if ( !ret )
- {
- debug_error("failed to set position. dur[%lud] pos[%lud] pos_msec[%llud]\n", dur, position, pos_msec);
- goto SEEK_ERROR;
- }
- }
- break;
+ /*
+ debug_log( "Call video steramCb, data[%p], Width[%d],Height[%d], Format[%d]",
+ GST_BUFFER_DATA(buffer), stream.width, stream.height, stream.format );
+ */
- case MM_PLAYER_POS_FORMAT_PERCENT:
- {
- if ( position < 0 && position > 100 )
- goto INVALID_ARGS;
+ if (stream.width == 0 || stream.height == 0) {
+ debug_error("Wrong condition!!");
+ return TRUE;
+ }
- debug_log("seeking to (%lu)%% \n", position);
+ /* set size and timestamp */
+ stream.length_total = GST_BUFFER_SIZE(buffer);
+ stream.timestamp = (unsigned int)(GST_BUFFER_TIMESTAMP(buffer)/1000000); /* nano sec -> mili sec */
- if (player->doing_seek)
- {
- debug_log("not completed seek");
- return MM_ERROR_PLAYER_DOING_SEEK;
- }
- player->doing_seek = TRUE;
+ //check zero-copy
+ if (player->set_mode.video_zc
+ && player->set_mode.media_packet_video_stream
+ && GST_BUFFER_MALLOCDATA(buffer)) {
+ scmn_imgb = (MMPlayerMPlaneImage *)GST_BUFFER_MALLOCDATA(buffer);
+ }
- /* FIXIT : why don't we use 'GST_FORMAT_PERCENT' */
- pos_msec = (gint64) ( ( position * player->duration ) / 100 );
- ret = __gst_seek ( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, 1.0,
- GST_FORMAT_TIME, ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- GST_SEEK_TYPE_SET, pos_msec, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE );
+ /* set tbm bo */
+ if (scmn_imgb) {
+ /* copy pointer of tbm bo, stride, elevation */
+ memcpy(stream.bo, scmn_imgb->bo, sizeof(void *) * MM_PLAYER_IMGB_MPLANE_MAX);
+ memcpy(stream.stride, scmn_imgb->s, sizeof(int) * MM_PLAYER_IMGB_MPLANE_MAX);
+ memcpy(stream.elevation, scmn_imgb->e, sizeof(int) * MM_PLAYER_IMGB_MPLANE_MAX);
- if ( !ret )
- {
- debug_error("failed to set position. dur[%lud] pos[%lud] pos_msec[%llud]\n", dur, position, pos_msec);
- goto SEEK_ERROR;
- }
+ /*
+ int i = 0;
+ for (i = 0 ; i < BUFFER_MAX_PLANE_NUM ; i++) {
+ debug_log("w %d, h %d, s %d, e %d, bo %p",
+ scmn_imgb->w[i],scmn_imgb->h[i],scmn_imgb->s[i],scmn_imgb->e[i],
+ scmn_imgb->bo[i]);
+ if(scmn_imgb->bo[i])
+ debug_log("bo size %d", tbm_bo_size(scmn_imgb->bo[i]));
}
- break;
+ */
- default:
- goto INVALID_ARGS;
-
+ /* set gst buffer */
+ stream.internal_buffer = buffer;
+ } else {
+ stream.data = GST_BUFFER_DATA(buffer);
}
+ if (player->video_stream_cb) {
+ player->video_stream_cb(&stream, player->video_stream_cb_user_param);
+ }
- /* NOTE : store last seeking point to overcome some bad operation
- * ( returning zero when getting current position ) of some elements
- */
- player->last_position = pos_msec;
-
- debug_log("playback rate: %f\n", player->playback_rate);
-
- /* MSL should guarante playback rate when seek is selected during trick play of fast forward. */
- if ( player->playback_rate > 1.0 )
- _mmplayer_set_playspeed ( (MMHandleType)player, player->playback_rate );
-
- debug_fleave();
+ return TRUE;
+}
- return MM_ERROR_NONE;
+static int
+__mmplayer_gst_create_video_filters(mm_player_t* player, GList** bucket, gboolean use_video_stream)
+{
+ gchar* video_csc = "ffmpegcolorspace"; // default colorspace converter
+ GList* element_bucket = *bucket;
-PENDING:
- player->pending_seek.is_pending = TRUE;
- player->pending_seek.format = format;
- player->pending_seek.pos = position;
-
- debug_warning("player current-state : %s, pending-state : %s, just preserve pending position(%lu).\n",
- MMPLAYER_STATE_GET_NAME(current_state), MMPLAYER_STATE_GET_NAME(pending_state), player->pending_seek.pos);
-
- return MM_ERROR_NONE;
-
-INVALID_ARGS:
- debug_error("invalid arguments, position : %ld dur : %ld format : %d \n", position, dur, format);
-
- return MM_ERROR_PLAYER_INTERNAL;
+ return_val_if_fail(player && player->pipeline && player->pipeline->videobin, MM_ERROR_PLAYER_NOT_INITIALIZED);
-SEEK_ERROR:
- player->doing_seek = FALSE;
- return MM_ERROR_PLAYER_SEEK;
+ MMPLAYER_FENTER();
-}
+ if (!player->set_mode.media_packet_video_stream && use_video_stream)
+ {
+ if (player->set_mode.video_zc && strlen(player->ini.name_of_video_converter) > 0)
+ {
+ video_csc = player->ini.name_of_video_converter;
+ }
-#define TRICKPLAY_OFFSET GST_MSECOND
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_CONV, video_csc, "video converter", TRUE, player);
+ debug_log("using video converter: %s", video_csc);
-static int
-__gst_get_position(mm_player_t* player, int format, unsigned long* position) // @
-{
- MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
- GstFormat fmt = GST_FORMAT_TIME;
- signed long long pos_msec = 0;
- gboolean ret = TRUE;
+ if ( !player->set_mode.video_zc)
+ {
+ GstStructure *str = NULL;
+ gint width = 0; //width of video
+ gint height = 0; //height of video
+ GstCaps* video_caps = NULL;
+ GstStructure *structure = NULL;
- return_val_if_fail( player && position && player->pipeline && player->pipeline->mainbin,
- MM_ERROR_PLAYER_NOT_INITIALIZED );
+ /* rotator, scaler and capsfilter */
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_FLIP, "videoflip", "video rotator", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_SCALE, "videoscale", "video scaler", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_CAPS, "capsfilter", "videocapsfilter", TRUE, player);
- current_state = MMPLAYER_CURRENT_STATE(player);
+ /* get video stream caps parsed by demuxer */
+ str = gst_caps_get_structure (player->v_stream_caps, 0);
+ if ( !str )
+ {
+ debug_error("cannot get structure");
+ goto ERROR;
+ }
- /* NOTE : query position except paused state to overcome some bad operation
- * please refer to below comments in details
- */
- if ( current_state != MM_PLAYER_STATE_PAUSED )
- {
- ret = gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &pos_msec);
- }
+ mm_attrs_get_int_by_name(player->attrs, "display_width", &width);
- /* NOTE : get last point to overcome some bad operation of some elements
- * ( returning zero when getting current position in paused state
- * and when failed to get postion during seeking
- */
- if ( ( current_state == MM_PLAYER_STATE_PAUSED )
- || ( ! ret ))
- //|| ( player->last_position != 0 && pos_msec == 0 ) )
- {
- debug_warning(" Playback rate is %f\n", player->playback_rate);
- debug_warning ("pos_msec = %"GST_TIME_FORMAT" and ret = %d and state = %d", GST_TIME_ARGS (pos_msec), ret, current_state);
+ if (width) {
+ structure = gst_structure_new("video/x-raw-rgb", "width", G_TYPE_INT, width, NULL);
+ }
- if(player->playback_rate < 0.0)
- pos_msec = player->last_position - TRICKPLAY_OFFSET;
- else
- pos_msec = player->last_position;
+ mm_attrs_get_int_by_name(player->attrs, "display_height", &height);
- if (!ret)
- pos_msec = player->last_position;
- else
- player->last_position = pos_msec;
+ if (height ) {
+ if (structure) {
+ gst_structure_set (structure, "height", G_TYPE_INT, height, NULL);
+ }else {
+ structure = gst_structure_new("video/x-raw-rgb", "height", G_TYPE_INT, height, NULL);
+ }
+ }
- debug_warning("returning last point : %"GST_TIME_FORMAT, GST_TIME_ARGS(pos_msec));
+ if (width ||height) {
+ video_caps = gst_caps_new_full(structure, NULL);
+ gst_structure_free(structure);
+ g_object_set (GST_ELEMENT(player->pipeline->videobin[MMPLAYER_V_CAPS].gst), "caps", video_caps, NULL );
+ MMPLAYER_LOG_GST_CAPS_TYPE(video_caps);
+ gst_caps_unref(video_caps);
+ }
+ }
}
else
{
- player->last_position = pos_msec;
- }
-
- switch (format) {
- case MM_PLAYER_POS_FORMAT_TIME:
- *position = GST_TIME_AS_MSECONDS(pos_msec);
- break;
+ MMDisplaySurfaceType surface_type = MM_DISPLAY_SURFACE_NULL;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", (int *)&surface_type);
- case MM_PLAYER_POS_FORMAT_PERCENT:
+ if (player->set_mode.video_zc)
{
- int dur = 0;
- int pos = 0;
-
- dur = player->duration / GST_SECOND;
- if (dur <= 0)
+ if ( (surface_type == MM_DISPLAY_SURFACE_EVAS) && ( !strcmp(player->ini.videosink_element_evas, "evasimagesink")) )
{
- debug_log ("duration is [%d], so returning position 0\n",dur);
- *position = 0;
+ video_csc = player->ini.name_of_video_converter;
}
else
{
- pos = pos_msec / GST_SECOND;
- *position = pos * 100 / dur;
+ video_csc = "";
}
- break;
}
- default:
- return MM_ERROR_PLAYER_INTERNAL;
- }
- debug_log("current position : %lu\n", *position);
+ if (video_csc && (strcmp(video_csc, "")))
+ {
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_CONV, video_csc, "video converter", TRUE, player);
+ debug_log("using video converter: %s", video_csc);
+ }
+
+ /* set video rotator */
+ if ( !player->set_mode.video_zc )
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_FLIP, "videoflip", "video rotator", TRUE, player);
+
+ /* videoscaler */
+ #if !defined(__arm__)
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_SCALE, "videoscale", "videoscaler", TRUE, player);
+ #endif
+ }
-
+ *bucket = element_bucket;
+ MMPLAYER_FLEAVE();
return MM_ERROR_NONE;
-}
+ERROR:
+ *bucket = NULL;
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+}
-static int __gst_get_buffer_position(mm_player_t* player, int format, unsigned long* start_pos, unsigned long* stop_pos)
+/**
+ * This function is to create video pipeline.
+ *
+ * @param player [in] handle of player
+ * caps [in] src caps of decoder
+ * surface_type [in] surface type for video rendering
+ *
+ * @return This function returns zero on success.
+ * @remark
+ * @see __mmplayer_gst_create_audio_pipeline, __mmplayer_gst_create_midi_pipeline
+ */
+/**
+ * VIDEO PIPELINE
+ * - x surface (arm/x86) : xvimagesink
+ * - evas surface (arm) : evaspixmapsink
+ * fimcconvert ! evasimagesink
+ * - evas surface (x86) : videoconvertor ! videoflip ! evasimagesink
+ */
+static int
+__mmplayer_gst_create_video_pipeline(mm_player_t* player, GstCaps* caps, MMDisplaySurfaceType surface_type)
{
- GstElement *element = NULL;
- GstQuery *query = NULL;
+ GstPad *pad = NULL;
+ MMHandleType attrs;
+ GList*element_bucket = NULL;
+ MMPlayerGstElement* first_element = NULL;
+ MMPlayerGstElement* videobin = NULL;
+ gchar *videosink_element = NULL;
- return_val_if_fail( player &&
- player->pipeline &&
- player->pipeline->mainbin,
- MM_ERROR_PLAYER_NOT_INITIALIZED );
+ MMPLAYER_FENTER();
- return_val_if_fail( start_pos && stop_pos, MM_ERROR_INVALID_ARGUMENT );
+ return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
- if ( MMPLAYER_IS_HTTP_STREAMING ( player ) || MMPLAYER_IS_HTTP_LIVE_STREAMING ( player ))
+ /* alloc handles */
+ videobin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_V_NUM);
+ if ( !videobin )
{
- /* Note : In case of http streaming or HLS, the buffering queue [ queue2 ] could handle buffering query. */
- element = GST_ELEMENT ( player->pipeline->mainbin[MMPLAYER_M_S_BUFFER].gst );
+ return MM_ERROR_PLAYER_NO_FREE_SPACE;
}
- else if ( MMPLAYER_IS_RTSP_STREAMING ( player ) )
+
+ player->pipeline->videobin = videobin;
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
{
- debug_warning ( "it's not supported yet.\n" );
- return MM_ERROR_NONE;
+ debug_error("cannot get content attribute");
+ return MM_ERROR_PLAYER_INTERNAL;
}
- else
+
+ /* create bin */
+ videobin[MMPLAYER_V_BIN].id = MMPLAYER_V_BIN;
+ videobin[MMPLAYER_V_BIN].gst = gst_bin_new("videobin");
+ if ( !videobin[MMPLAYER_V_BIN].gst )
{
- debug_warning ( "it's only used for streaming case.\n" );
- return MM_ERROR_NONE;
+ debug_error("failed to create videobin");
+ goto ERROR;
}
- *start_pos = 0;
- *stop_pos = 0;
-
- switch ( format )
+ if( player->use_video_stream ) // video stream callback, so send raw video data to application
{
- case MM_PLAYER_POS_FORMAT_PERCENT :
- {
- query = gst_query_new_buffering ( GST_FORMAT_PERCENT );
- if ( gst_element_query ( element, query ) )
- {
- gint64 start, stop;
- GstFormat format;
- gboolean busy;
- gint percent;
+ debug_log("using memsink\n");
- gst_query_parse_buffering_percent ( query, &busy, &percent);
- gst_query_parse_buffering_range ( query, &format, &start, &stop, NULL );
+ if ( __mmplayer_gst_create_video_filters(player, &element_bucket, TRUE) != MM_ERROR_NONE)
+ goto ERROR;
- debug_log ( "buffering start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT "\n", start, stop);
+ /* finally, create video sink. output will be BGRA8888. */
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, "avsysmemsink", "videosink", TRUE, player);
- if ( start != -1)
- *start_pos = 100 * start / GST_FORMAT_PERCENT_MAX;
- else
- *start_pos = 0;
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "video-stream",
+ G_CALLBACK(__mmplayer_videostream_cb),
+ player );
+ }
+ else // render video data using sink plugin like xvimagesink
+ {
+ if ( __mmplayer_gst_create_video_filters(player, &element_bucket, FALSE) != MM_ERROR_NONE)
+ goto ERROR;
- if ( stop != -1)
- *stop_pos = 100 * stop / GST_FORMAT_PERCENT_MAX;
- else
- *stop_pos = 0;
- }
- gst_query_unref (query);
+ /* set video sink */
+ switch (surface_type)
+ {
+ case MM_DISPLAY_SURFACE_X:
+ if (strlen(player->ini.videosink_element_x) > 0)
+ videosink_element = player->ini.videosink_element_x;
+ else
+ goto ERROR;
+ break;
+ case MM_DISPLAY_SURFACE_EVAS:
+ if (strlen(player->ini.videosink_element_evas) > 0)
+ videosink_element = player->ini.videosink_element_evas;
+ else
+ goto ERROR;
+ break;
+ case MM_DISPLAY_SURFACE_X_EXT:
+ {
+ void *pixmap_id_cb = NULL;
+ mm_attrs_get_data_by_name(attrs, "display_overlay", &pixmap_id_cb);
+ if (pixmap_id_cb) /* this is used for the videoTextue(canvasTexture) overlay */
+ {
+ videosink_element = player->ini.videosink_element_x;
+ }
+ else
+ {
+ debug_error("something wrong.. callback function for getting pixmap id is null");
+ goto ERROR;
+ }
+ break;
+ }
+ case MM_DISPLAY_SURFACE_NULL:
+ if (strlen(player->ini.videosink_element_fake) > 0)
+ videosink_element = player->ini.videosink_element_fake;
+ else
+ goto ERROR;
+ break;
+ default:
+ debug_error("unidentified surface type");
+ goto ERROR;
}
- break;
- case MM_PLAYER_POS_FORMAT_TIME :
- debug_warning ( "Time format is not supported yet.\n" );
- break;
-
- default :
- break;
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, videosink_element, videosink_element, TRUE, player);
+ debug_log("selected videosink name: %s", videosink_element);
+ if(surface_type==MM_DISPLAY_SURFACE_X)
+ {
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "display-status",
+ G_CALLBACK(__mmplayer_display_status_cb),
+ player );
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "external-resolution",
+ G_CALLBACK(__mmplayer_external_resolution_cb),
+ player );
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "hided-window",
+ G_CALLBACK(__mmplayer_hided_window_cb),
+ player );
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "quick-panel-on",
+ G_CALLBACK(__mmplayer_quick_panel_on_cb),
+ player );
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "multiwindow-active",
+ G_CALLBACK(__mmplayer_multiwindow_active_cb),
+ player );
+ }
+ /* connect signal handlers for sink plug-in */
+ switch (surface_type) {
+ case MM_DISPLAY_SURFACE_X_EXT:
+ MMPLAYER_SIGNAL_CONNECT( player,
+ player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ MM_PLAYER_SIGNAL_TYPE_VIDEOBIN,
+ "frame-render-error",
+ G_CALLBACK(__mmplayer_videoframe_render_error_cb),
+ player );
+ debug_log("videoTexture usage, connect a signal handler for pixmap rendering error");
+ break;
+ default:
+ break;
+ }
}
- debug_log("current buffer position : %lu~%lu \n", *start_pos, *stop_pos );
+ if ( _mmplayer_update_video_param(player) != MM_ERROR_NONE)
+ goto ERROR;
- return MM_ERROR_NONE;
-}
+ /* store it as it's sink element */
+ __mmplayer_add_sink( player, videobin[MMPLAYER_V_SINK].gst );
-static int
-__gst_set_message_callback(mm_player_t* player, MMMessageCallback callback, gpointer user_param) // @
-{
- debug_fenter();
+ /* adding created elements to bin */
+ if( ! __mmplayer_gst_element_add_bucket_to_bin(GST_BIN(videobin[MMPLAYER_V_BIN].gst), element_bucket) )
+ {
+ debug_error("failed to add elements\n");
+ goto ERROR;
+ }
- if ( !player )
+ /* Linking elements in the bucket by added order */
+ if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
{
- debug_warning("set_message_callback is called with invalid player handle\n");
- return MM_ERROR_PLAYER_NOT_INITIALIZED;
+ debug_error("failed to link elements\n");
+ goto ERROR;
}
- player->msg_cb = callback;
- player->msg_cb_param = user_param;
+ /* get first element's sinkpad for creating ghostpad */
+ first_element = (MMPlayerGstElement *)element_bucket->data;
+ if ( !first_element )
+ {
+ debug_error("failed to get first element from bucket\n");
+ goto ERROR;
+ }
- debug_log("msg_cb : 0x%x msg_cb_param : 0x%x\n", (guint)callback, (guint)user_param);
+ pad = gst_element_get_static_pad(GST_ELEMENT(first_element->gst), "sink");
+ if ( !pad )
+ {
+ debug_error("failed to get pad from first element\n");
+ goto ERROR;
+ }
- debug_fleave();
+ /* create ghostpad */
+ player->ghost_pad_for_videobin = gst_ghost_pad_new("sink", pad);
+ if ( FALSE == gst_element_add_pad(videobin[MMPLAYER_V_BIN].gst, player->ghost_pad_for_videobin) )
+ {
+ debug_error("failed to add ghostpad to videobin\n");
+ goto ERROR;
+ }
+ gst_object_unref(pad);
- return MM_ERROR_NONE;
-}
+ /* done. free allocated variables */
+ g_list_free(element_bucket);
-static gboolean __mmfplayer_parse_profile(const char *uri, void *param, MMPlayerParseProfile* data) // @
-{
- gboolean ret = FALSE;
- char *path = NULL;
+ mm_attrs_set_int_by_name(attrs, "content_video_found", TRUE);
- debug_fenter();
+ MMPLAYER_FLEAVE();
- return_val_if_fail ( uri , FALSE);
- return_val_if_fail ( data , FALSE);
- return_val_if_fail ( ( strlen(uri) <= MM_MAX_URL_LEN ), FALSE );
+ return MM_ERROR_NONE;
- memset(data, 0, sizeof(MMPlayerParseProfile));
-
- if ((path = strstr(uri, "file://")))
- {
- if (util_exist_file_path(path + 7)) {
- strncpy(data->uri, path, MM_MAX_URL_LEN-1);
+ERROR:
+ debug_error("ERROR : releasing videobin\n");
- if ( util_is_sdp_file ( path ) )
- {
- debug_log("uri is actually a file but it's sdp file. giving it to rtspsrc\n");
- data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
- }
- else
- {
- data->uri_type = MM_PLAYER_URI_TYPE_FILE;
- }
- ret = TRUE;
- }
- else
- {
- debug_warning("could access %s.\n", path);
- }
- }
- else if ((path = strstr(uri, "buff://")))
- {
- data->uri_type = MM_PLAYER_URI_TYPE_BUFF;
- ret = TRUE;
- }
- else if ((path = strstr(uri, "rtsp://")))
- {
- if (strlen(path)) {
- strcpy(data->uri, uri);
- data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
- ret = TRUE;
- }
- }
- else if ((path = strstr(uri, "http://")))
- {
- if (strlen(path)) {
- strcpy(data->uri, uri);
-
-#ifdef NO_USE_GST_HLSDEMUX
- if ((path = strstr(uri, "m3u")))
- {
- debug_log ("++++++++++++++++++++ M3U +++++++++++++++++++++\n");
- debug_log ("\n\n\n\n \t HLS profile \n\n\n");
- data->uri_type = MM_PLAYER_URI_TYPE_HLS;
- }
- else
-#endif
- data->uri_type = MM_PLAYER_URI_TYPE_URL_HTTP;
+ g_list_free( element_bucket );
- ret = TRUE;
- }
- }
- else if ((path = strstr(uri, "https://")))
- {
- if (strlen(path)) {
- strcpy(data->uri, uri);
+ if (pad)
+ gst_object_unref(GST_OBJECT(pad));
-#ifdef NO_USE_GST_HLSDEMUX
- if ((path = strstr(uri, "m3u")))
- {
- debug_log ("++++++++++++++++++++ M3U +++++++++++++++++++++\n");
- debug_log ("\n\n\n\n \t HLS profile \n\n\n");
- data->uri_type = MM_PLAYER_URI_TYPE_HLS;
- }
- else
-#endif
- data->uri_type = MM_PLAYER_URI_TYPE_URL_HTTP;
-
- ret = TRUE;
- }
- }
- else if ((path = strstr(uri, "rtspu://")))
+ /* release videobin with it's childs */
+ if ( videobin[MMPLAYER_V_BIN].gst )
{
- if (strlen(path)) {
- strcpy(data->uri, uri);
- data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
- ret = TRUE;
- }
+ gst_object_unref(GST_OBJECT(videobin[MMPLAYER_V_BIN].gst));
}
- else if ((path = strstr(uri, "rtspr://")))
- {
- strcpy(data->uri, path);
- char *separater =strstr(path, "*");
- if (separater) {
- int urgent_len = 0;
- char *urgent = separater + strlen("*");
- if ((urgent_len = strlen(urgent))) {
- data->uri[strlen(path) - urgent_len - strlen("*")] = '\0';
- strcpy(data->urgent, urgent);
- data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
- ret = TRUE;
- }
- }
- }
- else if ((path = strstr(uri, "mms://")))
- {
- if (strlen(path)) {
- strcpy(data->uri, uri);
- data->uri_type = MM_PLAYER_URI_TYPE_URL_MMS;
- ret = TRUE;
- }
- }
- else if ((path = strstr(uri, "mem://")))
- {
- if (strlen(path)) {
- int mem_size = 0;
- char *buffer = NULL;
- char *seperator = strchr(path, ',');
- char ext[100] = {0,}, size[100] = {0,};
+ MMPLAYER_FREEIF( videobin );
- if (seperator) {
- if ((buffer = strstr(path, "ext="))) {
- buffer += strlen("ext=");
+ player->pipeline->videobin = NULL;
- if (strlen(buffer)) {
- strcpy(ext, buffer);
+ return MM_ERROR_PLAYER_INTERNAL;
+}
- if ((seperator = strchr(ext, ','))
- || (seperator = strchr(ext, ' '))
- || (seperator = strchr(ext, '\0'))) {
- seperator[0] = '\0';
- }
- }
- }
+static int __mmplayer_gst_create_plain_text_elements(mm_player_t* player)
+{
+ GList *element_bucket = NULL;
+ MMPlayerGstElement *textbin = player->pipeline->textbin;
- if ((buffer = strstr(path, "size="))) {
- buffer += strlen("size=");
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEXT_FAKE_QUEUE, "queue", "text_f_queue", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_FAKE_SINK, "fakesink", "text_fakesink", TRUE, player);
+ MMPLAYER_SIGNAL_CONNECT( player,
+ G_OBJECT(textbin[MMPLAYER_T_FAKE_SINK].gst),
+ MM_PLAYER_SIGNAL_TYPE_TEXTBIN,
+ "handoff",
+ G_CALLBACK(__mmplayer_update_subtitle),
+ (gpointer)player );
- if (strlen(buffer) > 0) {
- strcpy(size, buffer);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "async", TRUE, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "sync", TRUE, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "signal-handoffs", TRUE, NULL);
- if ((seperator = strchr(size, ','))
- || (seperator = strchr(size, ' '))
- || (seperator = strchr(size, '\0'))) {
- seperator[0] = '\0';
- }
+ if (!player->play_subtitle)
+ {
+ debug_log ("add textbin sink as sink element of whole pipeline.\n");
+ __mmplayer_add_sink (player, GST_ELEMENT(textbin[MMPLAYER_T_FAKE_SINK].gst));
+ }
- mem_size = atoi(size);
- }
- }
- }
+ /* adding created elements to bin */
+ debug_log("adding created elements to bin\n");
+ if( !__mmplayer_gst_element_add_bucket_to_bin( GST_BIN(textbin[MMPLAYER_T_BIN].gst), element_bucket ))
+ {
+ debug_error("failed to add elements\n");
+ goto ERROR;
+ }
- debug_log("ext: %s, mem_size: %d, mmap(param): %p\n", ext, mem_size, param);
- if ( mem_size && param) {
- data->mem = param;
- data->mem_size = mem_size;
- data->uri_type = MM_PLAYER_URI_TYPE_MEM;
- ret = TRUE;
- }
- }
+ /* unset sink flag from textbin. not to hold eos when video data is shorter than subtitle */
+ GST_OBJECT_FLAG_UNSET (textbin[MMPLAYER_T_BIN].gst, GST_ELEMENT_IS_SINK);
+ GST_OBJECT_FLAG_UNSET (textbin[MMPLAYER_T_FAKE_SINK].gst, GST_ELEMENT_IS_SINK);
+
+ /* linking elements in the bucket by added order. */
+ debug_log("Linking elements in the bucket by added order.\n");
+ if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ {
+ debug_error("failed to link elements\n");
+ goto ERROR;
}
- else
+
+ /* done. free allocated variables */
+ g_list_free(element_bucket);
+
+ if (textbin[MMPLAYER_T_TEXT_FAKE_QUEUE].gst)
{
- /* if no protocol prefix exist. check file existence and then give file:// as it's prefix */
- if (util_exist_file_path(uri))
+ GstPad *pad = NULL;
+ GstPad *ghostpad = NULL;
+
+ pad = gst_element_get_static_pad(GST_ELEMENT(textbin[MMPLAYER_T_TEXT_FAKE_QUEUE].gst), "sink");
+ if (!pad)
{
- debug_warning("uri has no protocol-prefix. giving 'file://' by default.\n");
+ debug_error("failed to get video pad of textbin\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- sprintf( data->uri, "file://%s", uri );
+ ghostpad = gst_ghost_pad_new("text_sink", pad);
+ gst_object_unref(pad);
- if ( util_is_sdp_file( (char*)uri ) )
- {
- debug_log("uri is actually a file but it's sdp file. giving it to rtspsrc\n");
- data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
- }
- else
- {
- data->uri_type = MM_PLAYER_URI_TYPE_FILE;
- }
- ret = TRUE;
+ if (!ghostpad)
+ {
+ debug_error("failed to create ghostpad of textbin\n");
+ goto ERROR;
}
- else
+
+ if (!gst_element_add_pad(textbin[MMPLAYER_T_BIN].gst, ghostpad))
{
- debug_error ("invalid uri, could not play..\n");
- data->uri_type = MM_PLAYER_URI_TYPE_NONE;
+ debug_error("failed to add ghostpad to textbin\n");
+ goto ERROR;
}
}
-
- if (data->uri_type == MM_PLAYER_URI_TYPE_NONE) {
- ret = FALSE;
- }
+ return MM_ERROR_NONE;
- /* dump parse result */
- debug_log("profile parsing result ---\n");
- debug_log("incomming uri : %s\n", uri);
- debug_log("uri : %s\n", data->uri);
- debug_log("uri_type : %d\n", data->uri_type);
- debug_log("play_mode : %d\n", data->play_mode);
- debug_log("mem : 0x%x\n", (guint)data->mem);
- debug_log("mem_size : %d\n", data->mem_size);
- debug_log("urgent : %s\n", data->urgent);
- debug_log("--------------------------\n");
-
- debug_fleave();
+ERROR:
+ g_list_free(element_bucket);
- return ret;
+ return MM_ERROR_PLAYER_INTERNAL;
}
-gboolean _asm_postmsg(gpointer *data)
-{
- mm_player_t* player = (mm_player_t*)data;
- MMMessageParamType msg = {0, };
-
- debug_fenter();
-
- return_val_if_fail ( player, FALSE );
-
- msg.union_type = MM_MSG_UNION_CODE;
- msg.code = player->sm.event_src;
-
- MMPLAYER_POST_MSG( player, MM_MESSAGE_READY_TO_RESUME, &msg);
-
- return FALSE;
-}
-gboolean _asm_lazy_pause(gpointer *data)
+static int __mmplayer_gst_create_text_pipeline(mm_player_t* player)
{
- mm_player_t* player = (mm_player_t*)data;
- int ret = MM_ERROR_NONE;
+ MMPlayerGstElement *textbin = NULL;
+ GList *element_bucket = NULL;
+ GstPad *pad = NULL;
+ GstPad *ghostpad = NULL;
+ gint i = 0;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, FALSE );
+ return_val_if_fail( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- if (MMPLAYER_CURRENT_STATE(player) == MM_PLAYER_STATE_PLAYING)
+ /* alloc handles */
+ textbin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_T_NUM);
+ if ( ! textbin )
{
- debug_log ("Ready to proceed lazy pause\n");
- ret = _mmplayer_pause((MMHandleType)player);
- if(MM_ERROR_NONE != ret)
- {
- debug_error("MMPlayer pause failed in ASM callback lazy pause\n");
- }
+ debug_error("failed to allocate memory for textbin\n");
+ return MM_ERROR_PLAYER_NO_FREE_SPACE;
}
- else
+
+ /* create bin */
+ textbin[MMPLAYER_T_BIN].id = MMPLAYER_T_BIN;
+ textbin[MMPLAYER_T_BIN].gst = gst_bin_new("textbin");
+ if ( !textbin[MMPLAYER_T_BIN].gst )
{
- debug_log ("Invalid state to proceed lazy pause\n");
+ debug_error("failed to create textbin\n");
+ goto ERROR;
}
- /* unset mute */
- if (player->pipeline && player->pipeline->audiobin)
- g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "mute", 0, NULL);
+ /* take it */
+ player->pipeline->textbin = textbin;
- player->sm.by_asm_cb = 0; //should be reset here
+ /* fakesink */
+ if (player->use_textoverlay)
+ {
+ debug_log ("use textoverlay for displaying \n");
- debug_fleave();
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_TEXT_QUEUE, "queue", "text_t_queue", textbin[MMPLAYER_T_BIN].gst, player);
- return FALSE;
-}
-ASM_cb_result_t
-__mmplayer_asm_callback(int handle, ASM_event_sources_t event_src, ASM_sound_commands_t command, unsigned int sound_status, void* cb_data)
-{
- mm_player_t* player = (mm_player_t*) cb_data;
- ASM_cb_result_t cb_res = ASM_CB_RES_IGNORE;
- MMHandleType attrs = 0;
- int result = MM_ERROR_NONE;
- gboolean lazy_pause = FALSE;
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_VIDEO_QUEUE, "queue", "text_v_queue", textbin[MMPLAYER_T_BIN].gst, player);
- debug_fenter();
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_VIDEO_CONVERTER, "fimcconvert", "text_v_converter", textbin[MMPLAYER_T_BIN].gst, player);
- return_val_if_fail ( player && player->pipeline, ASM_CB_RES_IGNORE );
- return_val_if_fail ( player->attrs, MM_ERROR_PLAYER_INTERNAL );
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_OVERLAY, "textoverlay", "text_overlay", textbin[MMPLAYER_T_BIN].gst, player);
- if (player->is_sound_extraction)
- {
- debug_log("sound extraction is working...so, asm command is ignored.\n");
- return result;
- }
-
- player->sm.by_asm_cb = 1; // it should be enabled for player state transition with called application command
- player->sm.event_src = event_src;
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_VIDEO_QUEUE].gst, "src", textbin[MMPLAYER_T_VIDEO_CONVERTER].gst, "sink"))
+ {
+ debug_error("failed to link queue and converter\n");
+ goto ERROR;
+ }
- if(event_src == ASM_EVENT_SOURCE_OTHER_PLAYER_APP)
- {
- player->sm.event_src = ASM_EVENT_SOURCE_OTHER_APP;
- }
- else if(event_src == ASM_EVENT_SOURCE_EARJACK_UNPLUG )
- {
- int stop_by_asm = 0;
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_VIDEO_CONVERTER].gst, "src", textbin[MMPLAYER_T_OVERLAY].gst, "video_sink"))
+ {
+ debug_error("failed to link queue and textoverlay\n");
+ goto ERROR;
+ }
- mm_attrs_get_int_by_name(player->attrs, "sound_stop_when_unplugged", &stop_by_asm);
- if (!stop_by_asm)
- return cb_res;
- }
- else if (event_src == ASM_EVENT_SOURCE_RESOURCE_CONFLICT)
- {
- /* can use video overlay simultaneously */
- /* video resource conflict */
- if(player->pipeline->videobin)
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_TEXT_QUEUE].gst, "src", textbin[MMPLAYER_T_OVERLAY].gst, "text_sink"))
+ {
+ debug_error("failed to link queue and textoverlay\n");
+ goto ERROR;
+ }
+
+ if (textbin[MMPLAYER_T_TEXT_QUEUE].gst)
{
- if (PLAYER_INI()->multiple_codec_supported)
+ pad = gst_element_get_static_pad(GST_ELEMENT(textbin[MMPLAYER_T_TEXT_QUEUE].gst), "sink");
+ if (!pad)
{
- debug_log("video conflict but, can support to use video overlay simultaneously");
- result = _mmplayer_pause((MMHandleType)player);
- cb_res = ASM_CB_RES_PAUSE;
+ debug_error("failed to get src pad of textbin\n");
+ goto ERROR;
}
- else
+
+ ghostpad = gst_ghost_pad_new("sink", pad);
+ if (!ghostpad)
{
- debug_log("video conflict, can't support for multiple codec instance");
- result = _mmplayer_unrealize((MMHandleType)player);
- cb_res = ASM_CB_RES_STOP;
+ debug_error("failed to create ghostpad of textbin\n");
+ goto ERROR;
+ }
+
+ if (!gst_element_add_pad(textbin[MMPLAYER_T_BIN].gst, ghostpad))
+ {
+ debug_error("failed to add ghostpad to textbin\n");
+ goto ERROR;
}
}
- return cb_res;
}
-
- switch(command)
+ else
{
- case ASM_COMMAND_PLAY:
- case ASM_COMMAND_STOP:
- debug_warning ("Got unexpected asm command (%d)", command);
- break;
-
- case ASM_COMMAND_PAUSE:
+ int surface_type = 0;
+ GstCaps* caps = NULL;
+
+ debug_log ("use subtitle message for displaying \n");
+
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &surface_type);
+
+ switch(surface_type)
{
- debug_log("Got msg from asm to Pause");
-
- if(event_src == ASM_EVENT_SOURCE_CALL_START
- || event_src == ASM_EVENT_SOURCE_ALARM_START
- || event_src == ASM_EVENT_SOURCE_OTHER_APP)
- {
- //hold 0.7 second to excute "fadedown mute" effect
- debug_log ("do fade down->pause->undo fade down");
-
- __mmplayer_do_sound_fadedown(player, MM_PLAYER_FADEOUT_TIME_DEFAULT);
-
- result = _mmplayer_pause((MMHandleType)player);
- if (result != MM_ERROR_NONE)
+ case MM_DISPLAY_SURFACE_X:
+ {
+ /* check whether external subtitle supported or not */
+ if (!player->ini.external_subtitle)
{
- debug_warning("fail to set Pause state by asm");
- cb_res = ASM_CB_RES_IGNORE;
+ debug_log ("not supported external subtitle\n");
+ if (__mmplayer_gst_create_plain_text_elements(player) != MM_ERROR_NONE)
+ {
+ debug_error("failed to make plain text elements\n");
+ goto ERROR;
+ }
break;
}
- __mmplayer_undo_sound_fadedown(player);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEE, "tee", "text_tee", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEXT_FAKE_QUEUE, "queue", "text_f_queue", TRUE, player);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_FAKE_QUEUE].gst),
+ "max-size-time", (guint64)0,
+ "max-size-bytes", (guint)0,
+ "max-size-buffers", (guint)5,
+ NULL);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEXT_FAKE_IDENTITY, "identity", "text_f_identity", TRUE, player);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst), "signal-handoffs", FALSE, NULL);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_FAKE_SINK, "fakesink", "text_fakesink", TRUE, player);
+ MMPLAYER_SIGNAL_CONNECT( player,
+ G_OBJECT(textbin[MMPLAYER_T_FAKE_SINK].gst),
+ MM_PLAYER_SIGNAL_TYPE_TEXTBIN,
+ "handoff",
+ G_CALLBACK(__mmplayer_update_subtitle),
+ (gpointer)player );
+
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "async", FALSE, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "sync", TRUE, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "signal-handoffs", TRUE, NULL);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEXT_QUEUE, "queue", "text_queue", TRUE, player);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_QUEUE].gst),
+ "max-size-time", (guint64)0,
+ "max-size-bytes", (guint)0,
+ "max-size-buffers", (guint)5,
+ NULL);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEXT_IDENTITY, "identity", "text_identity", TRUE, player);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_IDENTITY].gst), "signal-handoffs", FALSE, NULL);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_RENDER, "subrender", "text_subrender", TRUE, player);
+
+ if (player->font_desc == NULL)
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_RENDER].gst), "font-desc", "Tizen 42", NULL);
+ else
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_RENDER].gst), "font-desc", player->font_desc, NULL);
+
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_RENDER].gst), "font-color", player->font_color, NULL);
+ g_object_set (G_OBJECT (player->pipeline->textbin[MMPLAYER_T_RENDER].gst), "font-bg-color", player->font_bg_color, NULL);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_CAPSFILTER, "capsfilter", "text_capsfilter", TRUE, player);
+ caps = gst_caps_from_string( "video/x-raw-rgb, "
+ "width = (int) [ 1, MAX ], "
+ "height = (int) [ 1, MAX ], "
+ "framerate = (fraction) [ 0, MAX ], "
+ "subtitle = (boolean) true" );
+ g_object_set (G_OBJECT(textbin[MMPLAYER_T_CAPSFILTER].gst), "caps", caps, NULL );
+ gst_caps_unref( caps );
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_VIDEO_SINK, "xvimagesink", "text_xvimagesink", TRUE, player);
+ g_object_set(G_OBJECT(textbin[MMPLAYER_T_VIDEO_SINK].gst), "subpicture", TRUE, NULL);
+
+ player->display_stat = util_get_is_connected_external_display();
+
+ if(player->display_stat == MMPLAYER_DISPLAY_STATUS_UNKNOWN_ACTIVE)
+ {
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_RENDER].gst), "external-width", player->ini.mirroring_width, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_RENDER].gst), "external-height", player->ini.mirroring_height, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_VIDEO_SINK].gst), "external-width", player->ini.mirroring_width, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_VIDEO_SINK].gst), "external-height", player->ini.mirroring_height, NULL);
+ }
+ g_object_set(G_OBJECT(textbin[MMPLAYER_T_VIDEO_SINK].gst), "show-preroll-frame", FALSE, NULL);
+
+ /* if connected with external display */
+ if ((player->display_stat == MMPLAYER_DISPLAY_STATUS_HDMI_ACTIVE) ||
+ (player->display_stat == MMPLAYER_DISPLAY_STATUS_UNKNOWN_ACTIVE)) {
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst), "drop-probability", (gfloat)1.0, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_IDENTITY].gst), "drop-probability", (gfloat)0.0, NULL);
+ debug_warning ("connected with external display");
+ } else {
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst), "drop-probability", (gfloat)0.0, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_TEXT_IDENTITY].gst), "drop-probability", (gfloat)1.0, NULL);
+ g_object_set(G_OBJECT(textbin[MMPLAYER_T_VIDEO_SINK].gst), "async", FALSE, NULL);
+ g_object_set (G_OBJECT (textbin[MMPLAYER_T_FAKE_SINK].gst), "async", FALSE, NULL);
+
+ debug_warning ("non-connected status");
+ }
+ if (!player->play_subtitle)
+ {
+ debug_log ("add textbin sink as sink element of whole pipeline.\n");
+ __mmplayer_add_sink (player, GST_ELEMENT(textbin[MMPLAYER_T_FAKE_SINK].gst));
+ __mmplayer_add_sink (player, GST_ELEMENT(textbin[MMPLAYER_T_VIDEO_SINK].gst));
+ }
+
+ /* adding created elements to bin */
+ if( !__mmplayer_gst_element_add_bucket_to_bin( GST_BIN(textbin[MMPLAYER_T_BIN].gst), element_bucket ))
+ {
+ debug_error("failed to add elements\n");
+ goto ERROR;
+ }
+
+ /* unset sink flag from textbin. not to hold eos when video data is shorter than subtitle */
+ GST_OBJECT_FLAG_UNSET (textbin[MMPLAYER_T_BIN].gst, GST_ELEMENT_IS_SINK);
+ GST_OBJECT_FLAG_UNSET (textbin[MMPLAYER_T_FAKE_SINK].gst, GST_ELEMENT_IS_SINK);
+ GST_OBJECT_FLAG_UNSET (textbin[MMPLAYER_T_VIDEO_SINK].gst, GST_ELEMENT_IS_SINK);
+
+ /* linking elements in the bucket by added order. */
+ GstPad *pad, *rpad;
+
+ rpad = gst_element_get_request_pad(textbin[MMPLAYER_T_TEE].gst, "src%d");
+ pad = gst_element_get_static_pad(textbin[MMPLAYER_T_TEXT_FAKE_QUEUE].gst, "sink");
+ gst_pad_link(rpad, pad);
+ gst_object_unref(rpad);
+ gst_object_unref(pad);
+
+ rpad = gst_element_get_request_pad(textbin[MMPLAYER_T_TEE].gst, "src%d");
+ pad = gst_element_get_static_pad(textbin[MMPLAYER_T_TEXT_QUEUE].gst, "sink");
+ gst_pad_link(rpad, pad);
+ gst_object_unref(rpad);
+ gst_object_unref(pad);
+
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_TEXT_FAKE_QUEUE].gst, "src", textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst, "sink")) {
+ debug_error ("fake queue and identity could not be linked.\n");
+ goto ERROR;
+ }
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_TEXT_FAKE_IDENTITY].gst, "src", textbin[MMPLAYER_T_FAKE_SINK].gst, "sink")) {
+ debug_error ("indentity and fake sink could not be linked.\n");
+ goto ERROR;
+ }
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_TEXT_QUEUE].gst, "src", textbin[MMPLAYER_T_TEXT_IDENTITY].gst, "sink")) {
+ debug_error ("queue and identity could not be linked.\n");
+ goto ERROR;
+ }
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_TEXT_IDENTITY].gst, "src", textbin[MMPLAYER_T_RENDER].gst, "sink")) {
+ debug_error ("identity and subrender could not be linked.\n");
+ goto ERROR;
+ }
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_RENDER].gst, "src", textbin[MMPLAYER_T_CAPSFILTER].gst, "sink")) {
+ debug_error ("subrender and capsfilter could not be linked.\n");
+ goto ERROR;
+ }
+ if (!gst_element_link_pads (textbin[MMPLAYER_T_CAPSFILTER].gst, "src", textbin[MMPLAYER_T_VIDEO_SINK].gst, "sink")) {
+ debug_error ("capsfilter and xvimagesink could not be linked.\n");
+ goto ERROR;
+ }
+
+ /* done. free allocated variables */
+ g_list_free(element_bucket);
+ if (textbin[MMPLAYER_T_TEE].gst)
+ {
+ pad = gst_element_get_static_pad(GST_ELEMENT(textbin[MMPLAYER_T_TEE].gst), "sink");
+ if (!pad)
+ {
+ debug_error("failed to get text pad of textbin\n");
+ goto ERROR;
+ }
+
+ ghostpad = gst_ghost_pad_new("text_sink", pad);
+ if (!ghostpad)
+ {
+ debug_error("failed to create ghostpad of textbin\n");
+ goto ERROR;
+ }
+
+ if ( FALSE == gst_element_add_pad(textbin[MMPLAYER_T_BIN].gst, ghostpad) )
+ {
+ debug_error("failed to add ghostpad to textbin\n");
+ goto ERROR;
+ }
+ }
+
+ break;
}
- else if(event_src == ASM_EVENT_SOURCE_OTHER_PLAYER_APP)
- {
- lazy_pause = TRUE; // return as soon as possible, for fast start of other app
- if ( player->pipeline->audiobin && player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
- g_object_set( player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "mute", 2, NULL);
+ case MM_DISPLAY_SURFACE_EVAS:
+ case MM_DISPLAY_SURFACE_GL:
+ case MM_DISPLAY_SURFACE_NULL:
+ case MM_DISPLAY_SURFACE_X_EXT:
+ if (__mmplayer_gst_create_plain_text_elements(player) != MM_ERROR_NONE)
+ {
+ debug_error("failed to make plain text elements\n");
+ goto ERROR;
+ }
+ break;
+
+ default:
+ break;
+ }
+ }
+ gst_object_unref(pad);
+
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+
+ERROR:
+
+ debug_log("ERROR : releasing textbin\n");
+
+ if ( pad )
+ gst_object_unref(GST_OBJECT(pad));
+
+ if ( ghostpad )
+ gst_object_unref(GST_OBJECT(ghostpad));
+
+ g_list_free( element_bucket );
+
+ /* release element which are not added to bin */
+ for ( i = 1; i < MMPLAYER_T_NUM; i++ ) /* NOTE : skip bin */
+ {
+ if ( textbin[i].gst )
+ {
+ GstObject* parent = NULL;
+ parent = gst_element_get_parent( textbin[i].gst );
- player->lazy_pause_event_id = g_timeout_add(LAZY_PAUSE_TIMEOUT_MSEC, (GSourceFunc)_asm_lazy_pause, (gpointer)player);
- debug_log ("set lazy pause timer (id=[%d], timeout=[%d ms])", player->lazy_pause_event_id, LAZY_PAUSE_TIMEOUT_MSEC);
+ if ( !parent )
+ {
+ gst_object_unref(GST_OBJECT(textbin[i].gst));
+ textbin[i].gst = NULL;
}
else
{
- //immediate pause
- debug_log ("immediate pause");
- result = _mmplayer_pause((MMHandleType)player);
+ gst_object_unref(GST_OBJECT(parent));
}
- cb_res = ASM_CB_RES_PAUSE;
- }
- break;
-
- case ASM_COMMAND_RESUME:
- {
- debug_log("Got msg from asm to Resume. So, application can resume. code (%d) \n", event_src);
- player->sm.by_asm_cb = 0;
- //ASM server is single thread daemon. So use g_idle_add() to post resume msg
- g_idle_add((GSourceFunc)_asm_postmsg, (gpointer)player);
- cb_res = ASM_CB_RES_IGNORE;
}
- break;
+ }
- default:
- break;
+ /* release textbin with it's childs */
+ if ( textbin[MMPLAYER_T_BIN].gst )
+ {
+ gst_object_unref(GST_OBJECT(textbin[MMPLAYER_T_BIN].gst));
}
- if (!lazy_pause)
- player->sm.by_asm_cb = 0;
+ MMPLAYER_FREEIF( textbin );
- debug_fleave();
+ player->pipeline->textbin = NULL;
- return cb_res;
+ return MM_ERROR_PLAYER_INTERNAL;
}
-int
-_mmplayer_create_player(MMHandleType handle) // @
-{
- mm_player_t* player = MM_PLAYER_CAST(handle);
- gint i;
-
- debug_fenter();
-
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- MMTA_ACUM_ITEM_BEGIN("[KPI] media player service create->playing", FALSE);
+static int
+__mmplayer_gst_create_subtitle_src(mm_player_t* player)
+{
+ MMPlayerGstElement* mainbin = NULL;
+ MMHandleType attrs = 0;
+ GstElement *subsrc = NULL;
+ GstElement *subparse = NULL;
+ gchar *subtitle_uri =NULL;
+ const gchar *charset = NULL;
+ GstPad *sinkpad = NULL;
+ GstPad *srcpad = NULL;
+ GstPad *pad = NULL;
- /* initialize player state */
- MMPLAYER_CURRENT_STATE(player) = MM_PLAYER_STATE_NONE;
- MMPLAYER_PREV_STATE(player) = MM_PLAYER_STATE_NONE;
- MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
- MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_NONE;
- MMPLAYER_PRINT_STATE(player);
+ MMPLAYER_FENTER();
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL ( player, MMPLAYER_COMMAND_CREATE );
+ /* get mainbin */
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
- /* construct attributes */
- player->attrs = _mmplayer_construct_attribute(handle);
+ mainbin = player->pipeline->mainbin;
- if ( !player->attrs )
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
{
- debug_critical("Failed to construct attributes\n");
- goto ERROR;
+ debug_error("cannot get content attribute\n");
+ return MM_ERROR_PLAYER_INTERNAL;
}
- /* initialize gstreamer with configured parameter */
- if ( ! __mmplayer_gstreamer_init() )
+ mm_attrs_get_string_by_name ( attrs, "subtitle_uri", &subtitle_uri );
+ if ( !subtitle_uri || strlen(subtitle_uri) < 1)
{
- debug_critical("Initializing gstreamer failed\n");
- goto ERROR;
+ debug_error("subtitle uri is not proper filepath.\n");
+ return MM_ERROR_PLAYER_INVALID_URI;
}
+ debug_log("subtitle file path is [%s].\n", subtitle_uri);
- /* initialize factories if not using decodebin */
- if ( FALSE == PLAYER_INI()->use_decodebin )
- {
- if( player->factories == NULL )
- __mmplayer_init_factories(player);
- }
- /* create lock. note that g_tread_init() has already called in gst_init() */
- player->fsink_lock = g_mutex_new();
- if ( ! player->fsink_lock )
+ /* create the subtitle source */
+ subsrc = gst_element_factory_make("filesrc", "subtitle_source");
+ if ( !subsrc )
{
- debug_critical("Cannot create mutex for command lock\n");
+ debug_error ( "failed to create filesrc element\n" );
goto ERROR;
}
+ g_object_set(G_OBJECT (subsrc), "location", subtitle_uri, NULL);
- /* create repeat mutex */
- player->repeat_thread_mutex = g_mutex_new();
- if ( ! player->repeat_thread_mutex )
+ mainbin[MMPLAYER_M_SUBSRC].id = MMPLAYER_M_SUBSRC;
+ mainbin[MMPLAYER_M_SUBSRC].gst = subsrc;
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), subsrc))
{
- debug_critical("Cannot create repeat mutex\n");
+ debug_warning("failed to add queue\n");
goto ERROR;
}
- /* create repeat cond */
- player->repeat_thread_cond = g_cond_new();
- if ( ! player->repeat_thread_cond )
+ /* subparse */
+ subparse = gst_element_factory_make("submux", "external-submux");
+ if ( !subparse )
{
- debug_critical("Cannot create repeat cond\n");
+ debug_error ( "failed to create subparse element\n" );
goto ERROR;
}
- /* create repeat thread */
- player->repeat_thread =
- g_thread_create (__mmplayer_repeat_thread, (gpointer)player, TRUE, NULL);
- if ( ! player->repeat_thread )
+ charset = util_get_charset(subtitle_uri);
+ if (charset)
{
- goto ERROR;
+ debug_log ("detected charset is %s\n", charset );
+ g_object_set (G_OBJECT (subparse), "subtitle-encoding", charset, NULL);
}
- if ( MM_ERROR_NONE != _mmplayer_initialize_video_capture(player))
+ mainbin[MMPLAYER_M_T_SUBMUX_EXTERNAL].id = MMPLAYER_M_T_SUBMUX_EXTERNAL;
+ mainbin[MMPLAYER_M_T_SUBMUX_EXTERNAL].gst = subparse;
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), subparse))
{
- debug_error("failed to initialize video capture\n");
+ debug_warning("failed to add subparse\n");
goto ERROR;
}
+ sinkpad = gst_element_get_request_pad (subparse, "sink%d");
+ srcpad = gst_element_get_pad(subsrc,"src");
- /* register to asm */
- if ( MM_ERROR_NONE != _mmplayer_asm_register(&player->sm, (ASM_sound_cb_t)__mmplayer_asm_callback, (void*)player) )
+ /* link filesrc & submux */
+ if (gst_pad_link (srcpad, sinkpad) != 0)
{
- /* NOTE : we are dealing it as an error since we cannot expect it's behavior */
- debug_error("failed to register asm server\n");
- return MM_ERROR_POLICY_INTERNAL;
+ debug_warning("failed to link subsrc and subparse\n");
+ goto ERROR;
}
- if (MMPLAYER_IS_HTTP_PD(player))
+ player->play_subtitle = TRUE;
+ player->adjust_subtitle_pos = 0;
+
+ debug_log ("play subtitle using subtitle file\n");
+
+ if (player->pipeline->textbin == NULL)
{
- player->pd_downloader = NULL;
- player->pd_file_location = NULL;
- }
+ /* if textbin is null, assuming input selector also not created */
+ GstElement *selector = NULL;
- /* give default value of sound effect setting */
- player->audio_filter_info.filter_type = MM_AUDIO_FILTER_TYPE_NONE;
- player->bypass_sound_effect = TRUE;
+ /* input selector */
+ selector = gst_element_factory_make("input-selector", "subtitle_inselector");
+ if (!selector)
+ {
+ debug_error ( "failed to create subtitle input selector element\n" );
+ goto ERROR;
+ }
+ g_object_set (selector, "sync-streams", TRUE, NULL);
- player->sound.volume = MM_VOLUME_FACTOR_DEFAULT;
+ mainbin[MMPLAYER_M_T_INPUT_SELECTOR].id = MMPLAYER_M_T_INPUT_SELECTOR;
+ mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst = selector;
- /* initialize last position */
- player->last_position = 0;
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), selector))
+ {
+ debug_warning("failed to add subtitle input selector\n");
+ goto ERROR;
+ }
- /* initialize missing plugin */
- player->not_supported_codec = MISSING_PLUGIN_NONE;
+ if (MM_ERROR_NONE != __mmplayer_gst_create_text_pipeline(player))
+ {
+ debug_error("failed to create textbin. continuing without text\n");
+ goto ERROR;
+ }
- /* initialize found plugin */
- player->can_support_codec = FOUND_PLUGIN_NONE;
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), GST_ELEMENT(player->pipeline->textbin[MMPLAYER_T_BIN].gst)))
+ {
+ debug_warning("failed to add textbin\n");
+ goto ERROR;
+ }
- player->posted_msg = FALSE;
+ debug_log ("link text input selector and textbin ghost pad");
+ if (!gst_element_link_pads (mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "src", player->pipeline->textbin[MMPLAYER_T_BIN].gst, "text_sink"))
+ {
+ debug_warning("failed to link subparse and textbin\n");
+ goto ERROR;
+ }
-#if 0
- /* initialize application client id for dnse */
- player->app_id_set_up_dnse = MM_AUDIO_FILTER_CLIENT_NONE;
-#endif
- /* set player state to ready */
- MMPLAYER_SET_STATE(player, MM_PLAYER_STATE_NULL);
+ sinkpad = gst_element_get_request_pad (mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "sink%d");
+ if (sinkpad == NULL)
+ {
+ debug_error ("failed to get request pad");;
+ goto ERROR;
+ }
- player->section_repeat = FALSE;
- player->section_repeat_start = 0;
- player->section_repeat_end = 0;
- player->is_sound_extraction = FALSE;
- player->playback_rate = DEFAULT_PLAYBACK_RATE;
- player->resumed_by_rewind = FALSE;
- player->is_nv12_tiled = FALSE;
- player->has_many_types = FALSE;
- player->no_more_pad = TRUE;
- player->parsers = NULL;
- player->pipeline_is_constructed = FALSE;
-
- /*initialize adaptive http streaming handle */
- player->ahs_player = NULL;
-
- /* set buffering parameter for streaming */
- for(i=0; i<MM_PLAYER_STREAM_COUNT_MAX; i++)
+ debug_log ("Set Active Pad - %s\n", GST_DEBUG_PAD_NAME(sinkpad));
+ g_object_set (player->pipeline->mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "active-pad", sinkpad, NULL);
+
+ player->textsink_linked = 1;
+ player->external_text_idx = 0;
+ debug_msg("player->textsink_linked set to 1\n");
+ }
+ else
{
- player->bitrate[i] = 0;
- player->maximum_bitrate[i] = 0;
+ debug_log("text bin has been created. reuse it.");
+
+ sinkpad = gst_element_get_request_pad (mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "sink%d");
+ if (sinkpad == NULL)
+ {
+ debug_error ("failed to get request pad");;
+ goto ERROR;
+ }
+
+ player->external_text_idx = 1;
}
- player->updated_bitrate_count = 0;
- player->total_bitrate = 0;
- player->updated_maximum_bitrate_count = 0;
- player->total_maximum_bitrate = 0;
- /* initialize unlinked audio/video mime type */
- player->unlinked_audio_mime = NULL;
- player->unlinked_video_mime = NULL;
- player->unlinked_demuxer_mime = NULL;
- player->not_found_demuxer = 0;
-
- player->sent_bos = FALSE;
+ srcpad = gst_element_get_pad (subparse,"src");
- player->doing_seek = FALSE;
+ /* link submux & input selector */
+ if (gst_pad_link (srcpad, sinkpad) != 0)
+ {
+ debug_warning("failed to link submux and input selector\n");
+ goto ERROR;
+ }
- player->lazy_pause_event_id = 0;
-
- player->streamer = NULL;
+ pad = gst_element_get_static_pad (player->pipeline->textbin[MMPLAYER_T_FAKE_SINK].gst, "sink");
- player->play_subtitle = FALSE;
+ if (!pad)
+ {
+ debug_error("failed to get sink pad from textsink to probe data");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- player->v_stream_caps = NULL;
-
- /* initialize pending seek */
- player->pending_seek.is_pending = FALSE;
- player->pending_seek.format = MM_PLAYER_POS_FORMAT_TIME;
- player->pending_seek.pos = 0;
+ gst_pad_add_buffer_probe(pad, G_CALLBACK(__mmplayer_subtitle_adjust_position_probe), player);
+ gst_object_unref(pad);
+ pad=NULL;
- /* set player state to null */
- MMPLAYER_STATE_CHANGE_TIMEOUT(player) = PLAYER_INI()->localplayback_state_change_timeout;
- MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_NULL );
- debug_fleave();
+ MMPLAYER_FLEAVE();
return MM_ERROR_NONE;
ERROR:
- /* free lock */
- if ( player->fsink_lock )
- g_mutex_free( player->fsink_lock );
- player->fsink_lock = NULL;
+ return MM_ERROR_PLAYER_INTERNAL;
+}
- /* free thread */
- if ( player->repeat_thread_cond &&
- player->repeat_thread_mutex &&
- player->repeat_thread )
- {
- player->repeat_thread_exit = TRUE;
- g_cond_signal( player->repeat_thread_cond );
+gboolean
+__mmplayer_update_subtitle( GstElement* object, GstBuffer *buffer, GstPad *pad, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*) data;
+ MMMessageParamType msg = {0, };
+ GstClockTime duration = 0;
+ guint8 *text = NULL;
+ guint text_size = 0;
+ gboolean ret = TRUE;
- g_thread_join( player->repeat_thread );
- player->repeat_thread = NULL;
+ MMPLAYER_FENTER();
- g_mutex_free ( player->repeat_thread_mutex );
- player->repeat_thread_mutex = NULL;
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( buffer, FALSE );
- g_cond_free ( player->repeat_thread_cond );
- player->repeat_thread_cond = NULL;
+ text = GST_BUFFER_DATA(buffer);
+ text_size = GST_BUFFER_SIZE(buffer);
+ duration = GST_BUFFER_DURATION(buffer);
+
+ if ( player->set_mode.subtitle_off )
+ {
+ debug_log("subtitle is OFF.\n" );
+ return TRUE;
}
- /* clear repeat thread mutex/cond if still alive
- * this can happen if only thread creating has failed
- */
- if ( player->repeat_thread_mutex )
- g_mutex_free ( player->repeat_thread_mutex );
- if ( player->repeat_thread_cond )
- g_cond_free ( player->repeat_thread_cond );
+ if ( !text || (text_size == 0))
+ {
+ debug_log("There is no subtitle to be displayed.\n" );
+ return TRUE;
+ }
- /* release attributes */
- _mmplayer_deconstruct_attribute(handle);
+ msg.data = (void *) text;
+ msg.subtitle.duration = GST_TIME_AS_MSECONDS(duration);
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_log("update subtitle : [%ld msec] %s\n'", msg.subtitle.duration, (char*)msg.data );
+
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_UPDATE_SUBTITLE, &msg );
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
}
static gboolean
-__mmplayer_gstreamer_init(void) // @
+__mmplayer_subtitle_adjust_position_probe(GstPad *pad, GstBuffer *buffer, gpointer u_data)
{
- static gboolean initialized = FALSE;
- static const int max_argc = 50;
- gint* argc = NULL;
- gchar** argv = NULL;
- GError *err = NULL;
- int i = 0;
+ mm_player_t *player = (mm_player_t *) u_data;
+ GstClockTime cur_timestamp = 0;
+ gint64 adjusted_timestamp = 0;
- debug_fenter();
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( buffer, FALSE );
- if ( initialized )
+ if ( player->set_mode.subtitle_off )
{
- debug_log("gstreamer already initialized.\n");
+ debug_log("subtitle is OFF.\n" );
return TRUE;
}
- /* alloc */
- argc = malloc( sizeof(int) );
- argv = malloc( sizeof(gchar*) * max_argc );
-
- if ( !argc || !argv )
- goto ERROR;
+ if (player->adjust_subtitle_pos == 0 )
+ {
+ debug_log("nothing to do");
+ return TRUE;
+ }
- memset( argv, 0, sizeof(gchar*) * max_argc );
+ cur_timestamp = GST_BUFFER_TIMESTAMP(buffer);
- /* add initial */
- *argc = 1;
- argv[0] = g_strdup( "mmplayer" );
+ adjusted_timestamp = (gint64) cur_timestamp + ((gint64) player->adjust_subtitle_pos * G_GINT64_CONSTANT(1000000));
- /* add gst_param */
- for ( i = 0; i < 5; i++ ) /* FIXIT : num of param is now fixed to 5. make it dynamic */
+ if ( adjusted_timestamp < 0)
{
- if ( strlen( PLAYER_INI()->gst_param[i] ) > 0 )
- {
- argv[*argc] = g_strdup( PLAYER_INI()->gst_param[i] );
- (*argc)++;
- }
+ debug_log("adjusted_timestamp under zero");
+ MMPLAYER_FLEAVE();
+ return FALSE;
}
- /* we would not do fork for scanning plugins */
- argv[*argc] = g_strdup("--gst-disable-registry-fork");
- (*argc)++;
+ GST_BUFFER_TIMESTAMP(buffer) = (GstClockTime) adjusted_timestamp;
+ debug_log("buffer timestamp changed %" GST_TIME_FORMAT " -> %" GST_TIME_FORMAT "",
+ GST_TIME_ARGS(cur_timestamp),
+ GST_TIME_ARGS(GST_BUFFER_TIMESTAMP(buffer)));
- /* check disable registry scan */
- if ( PLAYER_INI()->skip_rescan )
+ return TRUE;
+}
+static int __gst_adjust_subtitle_position(mm_player_t* player, int format, int position)
+{
+ MMPLAYER_FENTER();
+
+ /* check player and subtitlebin are created */
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( player->play_subtitle, MM_ERROR_NOT_SUPPORT_API );
+
+ if (position == 0)
{
- argv[*argc] = g_strdup("--gst-disable-registry-update");
- (*argc)++;
+ debug_log ("nothing to do\n");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
}
- /* check disable segtrap */
- if ( PLAYER_INI()->disable_segtrap )
+ switch (format)
{
- argv[*argc] = g_strdup("--gst-disable-segtrap");
- (*argc)++;
- }
+ case MM_PLAYER_POS_FORMAT_TIME:
+ {
+ /* check current postion */
+ player->adjust_subtitle_pos = position;
- debug_log("initializing gstreamer with following parameter\n");
- debug_log("argc : %d\n", *argc);
+ debug_log("save adjust_subtitle_pos in player") ;
+ }
+ break;
- for ( i = 0; i < *argc; i++ )
- {
- debug_log("argv[%d] : %s\n", i, argv[i]);
+ default:
+ {
+ debug_warning("invalid format.\n");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
}
+ MMPLAYER_FLEAVE();
- /* initializing gstreamer */
- __ta__("gst_init time",
+ return MM_ERROR_NONE;
+}
- if ( ! gst_init_check (argc, &argv, &err))
- {
- debug_error("Could not initialize GStreamer: %s\n", err ? err->message : "unknown error occurred");
- if (err)
- {
- g_error_free (err);
- }
+static void
+__gst_appsrc_feed_data_mem(GstElement *element, guint size, gpointer user_data) // @
+{
+ GstElement *appsrc = element;
+ tBuffer *buf = (tBuffer *)user_data;
+ GstBuffer *buffer = NULL;
+ GstFlowReturn ret = GST_FLOW_OK;
+ gint len = size;
- goto ERROR;
- }
- );
+ return_if_fail ( element );
+ return_if_fail ( buf );
- /* release */
- for ( i = 0; i < *argc; i++ )
+ buffer = gst_buffer_new ();
+ if(buffer == NULL)
+ return;
+
+ if (buf->offset >= buf->len)
{
- MMPLAYER_FREEIF( argv[i] );
+ debug_log("call eos appsrc\n");
+ g_signal_emit_by_name (appsrc, "end-of-stream", &ret);
+ return;
}
- MMPLAYER_FREEIF( argv );
- MMPLAYER_FREEIF( argc );
+ if ( buf->len - buf->offset < size)
+ {
+ len = buf->len - buf->offset + buf->offset;
+ }
- /* done */
- initialized = TRUE;
+ GST_BUFFER_DATA(buffer) = (guint8*)(buf->buf + buf->offset);
+ GST_BUFFER_SIZE(buffer) = len;
+ GST_BUFFER_OFFSET(buffer) = buf->offset;
+ GST_BUFFER_OFFSET_END(buffer) = buf->offset + len;
+
+ //debug_log("feed buffer %p, offset %u-%u length %u\n", buffer, buf->offset, buf->len,len);
+ g_signal_emit_by_name (appsrc, "push-buffer", buffer, &ret);
+
+ buf->offset += len;
+}
+
+static gboolean
+__gst_appsrc_seek_data_mem(GstElement *element, guint64 size, gpointer user_data) // @
+{
+ tBuffer *buf = (tBuffer *)user_data;
+
+ return_val_if_fail ( buf, FALSE );
- debug_fleave();
+ buf->offset = (int)size;
return TRUE;
+}
-ERROR:
+static void
+__gst_appsrc_feed_data(GstElement *element, guint size, gpointer user_data) // @
+{
+ mm_player_t *player = (mm_player_t*)user_data;
- MMPLAYER_FREEIF( argv );
- MMPLAYER_FREEIF( argc );
+ return_if_fail ( player );
- return FALSE;
+ debug_msg("app-src: feed data\n");
+
+ if(player->need_data_cb)
+ player->need_data_cb(size, player->buffer_cb_user_param);
}
-int
-__mmplayer_release_extended_streaming(mm_player_t* player)
+static gboolean
+__gst_appsrc_seek_data(GstElement *element, guint64 offset, gpointer user_data) // @
{
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ mm_player_t *player = (mm_player_t*)user_data;
- if (player->pd_downloader && MMPLAYER_IS_HTTP_PD(player))
- {
- _mmplayer_pd_stop ((MMHandleType)player);
- _mmplayer_pd_deinitialize ((MMHandleType)player);
- _mmplayer_pd_destroy((MMHandleType)player);
- }
+ return_val_if_fail ( player, FALSE );
-#ifdef NO_USE_GST_HLSDEMUX
- /* destroy can called at anytime */
- if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
- {
- if (player->ahs_player)
- {
- __mm_player_ahs_stop (player->ahs_player);
- __mm_player_ahs_deinitialize (player->ahs_player);
- __mm_player_ahs_destroy(player->ahs_player);
- player->ahs_player = NULL;
- }
- }
-#endif
+ debug_msg("app-src: seek data\n");
- if (MMPLAYER_IS_STREAMING(player))
- {
- if (player->streamer)
- {
- __mm_player_streaming_deinitialize (player->streamer);
- __mm_player_streaming_destroy(player->streamer);
- player->streamer = NULL;
- }
- }
+ if(player->seek_data_cb)
+ player->seek_data_cb(offset, player->buffer_cb_user_param);
+
+ return TRUE;
+}
+
+
+static gboolean
+__gst_appsrc_enough_data(GstElement *element, gpointer user_data) // @
+{
+ mm_player_t *player = (mm_player_t*)user_data;
+
+ return_val_if_fail ( player, FALSE );
+
+ debug_msg("app-src: enough data:%p\n", player->enough_data_cb);
+
+ if(player->enough_data_cb)
+ player->enough_data_cb(player->buffer_cb_user_param);
+
+ return TRUE;
}
int
-_mmplayer_destroy(MMHandleType handle) // @
+_mmplayer_push_buffer(MMHandleType hplayer, unsigned char *buf, int size) // @
{
- mm_player_t* player = MM_PLAYER_CAST(handle);
+ mm_player_t* player = (mm_player_t*)hplayer;
+ GstBuffer *buffer = NULL;
+ GstFlowReturn gst_ret = GST_FLOW_OK;
+ int ret = MM_ERROR_NONE;
- debug_fenter();
+ MMPLAYER_FENTER();
- /* check player handle */
return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- /* destroy can called at anytime */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL ( player, MMPLAYER_COMMAND_DESTROY );
+ /* check current state */
+// MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_START );
- __mmplayer_release_extended_streaming(player);
- /* release repeat thread */
- if ( player->repeat_thread_cond &&
- player->repeat_thread_mutex &&
- player->repeat_thread )
+ /* NOTE : we should check and create pipeline again if not created as we destroy
+ * whole pipeline when stopping in streamming playback
+ */
+ if ( ! player->pipeline )
{
- player->repeat_thread_exit = TRUE;
- g_cond_signal( player->repeat_thread_cond );
-
- debug_log("waitting for repeat thread exit\n");
- g_thread_join ( player->repeat_thread );
- g_mutex_free ( player->repeat_thread_mutex );
- g_cond_free ( player->repeat_thread_cond );
- debug_log("repeat thread released\n");
+ if ( MM_ERROR_NONE != __gst_realize( player ) )
+ {
+ debug_error("failed to realize before starting. only in streamming\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
}
- if (MM_ERROR_NONE != _mmplayer_release_video_capture(player))
+ debug_msg("app-src: pushing data\n");
+
+ if ( buf == NULL )
{
- debug_error("failed to release video capture\n");
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_error("buf is null\n");
+ return MM_ERROR_NONE;
}
- /* withdraw asm */
- if ( MM_ERROR_NONE != _mmplayer_asm_deregister(&player->sm) )
- {
- debug_error("failed to deregister asm server\n");
+ buffer = gst_buffer_new ();
+ if(buffer == NULL)
return MM_ERROR_PLAYER_INTERNAL;
- }
- /* release pipeline */
- if ( MM_ERROR_NONE != __mmplayer_gst_destroy_pipeline( player ) )
+ if (size <= 0)
{
- debug_error("failed to destory pipeline\n");
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_log("call eos appsrc\n");
+ g_signal_emit_by_name (player->pipeline->mainbin[MMPLAYER_M_SRC].gst, "end-of-stream", &gst_ret);
+ return MM_ERROR_NONE;
}
- /* release attributes */
- _mmplayer_deconstruct_attribute( handle );
+ GST_BUFFER_DATA(buffer) = (guint8*)(buf);
+ GST_BUFFER_SIZE(buffer) = size;
- /* release factories */
- __mmplayer_release_factories( player );
+ debug_log("feed buffer %p, length %u\n", buf, size);
+ g_signal_emit_by_name (player->pipeline->mainbin[MMPLAYER_M_SRC].gst, "push-buffer", buffer, &gst_ret);
- /* release miscellaneous information */
- __mmplayer_release_misc( player );
+ MMPLAYER_FLEAVE();
- /* release lock */
- if ( player->fsink_lock )
- g_mutex_free( player->fsink_lock );
+ return ret;
+}
- if ( player->msg_cb_lock )
- g_mutex_free( player->msg_cb_lock );
+/* if retval is FALSE, it will be dropped for perfomance. */
+static gboolean
+__mmplayer_check_useful_message(mm_player_t *player, GstMessage * message)
+{
+ gboolean retval = FALSE;
- if (player->lazy_pause_event_id)
+ if ( !(player->pipeline && player->pipeline->mainbin) )
{
- g_source_remove (player->lazy_pause_event_id);
- player->lazy_pause_event_id = 0;
+ debug_error("player pipeline handle is null");
+ return TRUE;
}
- debug_fleave();
-
- return MM_ERROR_NONE;
-}
-
-
-int
-__mmplayer_init_extended_streaming(mm_player_t* player)
-{
- int ret = MM_ERROR_NONE;
-
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
-
-#ifdef NO_USE_GST_HLSDEMUX
- /* prepare adaptive http streaming */
- if (player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS)
+ switch (GST_MESSAGE_TYPE (message))
{
- player->ahs_player = __mm_player_ahs_create ();
-
- if (NULL == player->ahs_player)
- {
- debug_error ("Unable to create AHS player\n");
- ret = MM_ERROR_PLAYER_NO_FREE_SPACE;
- }
- else
+ case GST_MESSAGE_TAG:
+ case GST_MESSAGE_EOS:
+ case GST_MESSAGE_ERROR:
+ case GST_MESSAGE_WARNING:
+ case GST_MESSAGE_CLOCK_LOST:
+ case GST_MESSAGE_NEW_CLOCK:
+ case GST_MESSAGE_ELEMENT:
+ case GST_MESSAGE_DURATION:
+ case GST_MESSAGE_ASYNC_START:
+ retval = TRUE;
+ break;
+ case GST_MESSAGE_ASYNC_DONE:
+ case GST_MESSAGE_STATE_CHANGED:
+ /* we only handle messages from pipeline */
+ if(( message->src == (GstObject *)player->pipeline->mainbin[MMPLAYER_M_PIPE].gst ) && (!player->pp_rebuilding))
+ retval = TRUE;
+ else
+ retval = FALSE;
+ break;
+ case GST_MESSAGE_BUFFERING:
{
- if ( !__mm_player_ahs_initialize (player->ahs_player, player->profile.uri_type,
- player->profile.uri, player->pipeline->mainbin[MMPLAYER_M_SRC].gst) )
+ gint buffer_percent = 0;
+
+ gst_message_parse_buffering (message, &buffer_percent);
+
+ if ((MMPLAYER_IS_STREAMING(player)) &&
+ (player->streamer) &&
+ (player->streamer->is_buffering == TRUE) &&
+ (buffer_percent == MAX_BUFFER_PERCENT))
{
- debug_error ("failed to initialize ahs player\n");
- ret = MM_ERROR_PLAYER_NO_FREE_SPACE;
+ debug_log (">>> [%s] Buffering DONE is detected !!\n", GST_OBJECT_NAME(GST_MESSAGE_SRC(message)));
+ player->streamer->is_buffering_done = TRUE;
}
+
+ retval = TRUE;
+ break;
}
+ default:
+ retval = FALSE;
+ break;
}
-#endif
- if (MMPLAYER_IS_HTTP_PD(player))
+ return retval;
+}
+
+static GstBusSyncReply
+__mmplayer_bus_sync_callback (GstBus * bus, GstMessage * message, gpointer data)
+{
+ mm_player_t *player = (mm_player_t *)data;
+ GstBusSyncReply reply = GST_BUS_DROP;
+
+ if ( ! ( player->pipeline && player->pipeline->mainbin ) )
{
- gboolean bret = FALSE;
- gchar *src_uri = NULL;
+ debug_error("player pipeline handle is null");
+ return GST_BUS_PASS;
+ }
- player->pd_downloader = _mmplayer_pd_create ();
+ if (!__mmplayer_check_useful_message(player, message))
+ {
+ gst_message_unref (message);
+ return GST_BUS_DROP;
+ }
- if ( !player->pd_downloader )
- {
- debug_error ("Unable to create PD Downloader...");
- ret = MM_ERROR_PLAYER_NO_FREE_SPACE;
- }
-
- if (player->pd_mode == MM_PLAYER_PD_MODE_URI)
- src_uri = player->profile.uri;
-
- bret = _mmplayer_pd_initialize ((MMHandleType)player, src_uri, player->pd_file_location, player->pipeline->mainbin[MMPLAYER_M_SRC].gst);
-
- if (FALSE == bret)
- {
- debug_error ("Unable to create PD Downloader...");
- ret = MM_ERROR_PLAYER_NOT_INITIALIZED;
- }
+ switch (GST_MESSAGE_TYPE (message))
+ {
+ case GST_MESSAGE_STATE_CHANGED:
+ /* post directly for fast launch */
+ if (player->sync_handler) {
+ __mmplayer_gst_callback(NULL, message, player);
+ reply = GST_BUS_DROP;
+ }
+ else {
+ reply = GST_BUS_PASS;
+ }
+ break;
+ case GST_MESSAGE_TAG:
+ __mmplayer_gst_extract_tag_from_msg(player, message);
+ break;
+ case GST_MESSAGE_DURATION:
+ __mmplayer_gst_handle_duration(player, message);
+ break;
+ case GST_MESSAGE_ASYNC_DONE:
+ /* NOTE:Don't call gst_callback directly
+ * because previous frame can be showed even though this message is received for seek.
+ */
+ default:
+ reply = GST_BUS_PASS;
+ break;
}
- return ret;
+ if (reply == GST_BUS_DROP)
+ gst_message_unref (message);
+
+ return reply;
}
-int
-_mmplayer_realize(MMHandleType hplayer) // @
+/**
+ * This function is to create audio or video pipeline for playing.
+ *
+ * @param player [in] handle of player
+ *
+ * @return This function returns zero on success.
+ * @remark
+ * @see
+ */
+static int
+__mmplayer_gst_create_pipeline(mm_player_t* player) // @
{
- mm_player_t* player = (mm_player_t*)hplayer;
- char *uri =NULL;
- void *param = NULL;
- int application_pid = -1;
- gboolean update_registry = FALSE;
+ GstBus *bus = NULL;
+ MMPlayerGstElement *mainbin = NULL;
MMHandleType attrs = 0;
- int ret = MM_ERROR_NONE;
-
- debug_fenter();
+ GstElement* element = NULL;
+ GList* element_bucket = NULL;
+ gboolean need_state_holder = TRUE;
+ gint i = 0;
- /* check player handle */
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED )
+ MMPLAYER_FENTER();
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_REALIZE );
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ /* get profile attribute */
attrs = MMPLAYER_GET_ATTRS(player);
if ( !attrs )
{
- debug_error("fail to get attributes.\n");
- return MM_ERROR_PLAYER_INTERNAL;
- }
-
- mm_attrs_get_int_by_name(attrs, "sound_application_pid", &application_pid );
- player->sm.pid = application_pid;
-
- mm_attrs_get_string_by_name(attrs, "profile_uri", &uri);
- mm_attrs_get_data_by_name(attrs, "profile_user_param", ¶m);
-
- if (! __mmfplayer_parse_profile((const char*)uri, param, &player->profile) )
- {
- debug_error("failed to parse profile\n");
- return MM_ERROR_PLAYER_INVALID_URI;
- }
-
- /* FIXIT : we can use thouse in player->profile directly */
- if (player->profile.uri_type == MM_PLAYER_URI_TYPE_MEM)
- {
- player->mem_buf.buf = (char *)player->profile.mem;
- player->mem_buf.len = player->profile.mem_size;
- player->mem_buf.offset = 0;
+ debug_error("cannot get content attribute\n");
+ goto INIT_ERROR;
}
- if (player->profile.uri_type == MM_PLAYER_URI_TYPE_URL_MMS)
+ /* create pipeline handles */
+ if ( player->pipeline )
{
- debug_warning("mms protocol is not supported format.\n");
- return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+ debug_warning("pipeline should be released before create new one\n");
+ goto INIT_ERROR;
}
- if (MMPLAYER_IS_STREAMING(player))
- MMPLAYER_STATE_CHANGE_TIMEOUT(player) = PLAYER_INI()->live_state_change_timeout;
- else
- MMPLAYER_STATE_CHANGE_TIMEOUT(player) = PLAYER_INI()->localplayback_state_change_timeout;
-
- player->videodec_linked = 0;
- player->videosink_linked = 0;
- player->audiodec_linked = 0;
- player->audiosink_linked = 0;
- player->textsink_linked = 0;
-
- /* set the subtitle ON default */
- player->is_subtitle_off = FALSE;
+ player->pipeline = (MMPlayerGstPipelineInfo*) g_malloc0( sizeof(MMPlayerGstPipelineInfo) );
+ if (player->pipeline == NULL)
+ goto INIT_ERROR;
- /* we need to update content attrs only the content has changed */
- player->need_update_content_attrs = TRUE;
- player->need_update_content_dur = FALSE;
+ memset( player->pipeline, 0, sizeof(MMPlayerGstPipelineInfo) );
- /* registry should be updated for downloadable codec */
- mm_attrs_get_int_by_name(attrs, "profile_update_registry", &update_registry);
- if ( update_registry )
- {
- debug_log("updating registry...\n");
- gst_update_registry();
+ /* create mainbin */
+ mainbin = (MMPlayerGstElement*) g_malloc0( sizeof(MMPlayerGstElement) * MMPLAYER_M_NUM );
+ if (mainbin == NULL)
+ goto INIT_ERROR;
- /* then we have to rebuild factories */
- __mmplayer_release_factories( player );
- __mmplayer_init_factories(player);
- }
+ memset( mainbin, 0, sizeof(MMPlayerGstElement) * MMPLAYER_M_NUM);
- /* realize pipeline */
- ret = __gst_realize( player );
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("fail to realize the player.\n");
- }
- else
+ /* create pipeline */
+ mainbin[MMPLAYER_M_PIPE].id = MMPLAYER_M_PIPE;
+ mainbin[MMPLAYER_M_PIPE].gst = gst_pipeline_new("player");
+ if ( ! mainbin[MMPLAYER_M_PIPE].gst )
{
- __mmplayer_init_extended_streaming(player);
+ debug_error("failed to create pipeline\n");
+ goto INIT_ERROR;
}
+ player->demux_pad_index = 0;
+ player->subtitle_language_list = NULL;
- debug_fleave();
+ player->font_color = player->ini.font_color;
+ player->font_bg_color = player->ini.font_background_color;
- return ret;
-}
+ player->display_stat = MMPLAYER_DISPLAY_STATUS_NULL;
+ player->is_subtitle_force_drop = FALSE;
+ player->last_multiwin_status = FALSE;
-int
-__mmplayer_deinit_extended_streaming(mm_player_t *player)
-{
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
-
- /* destroy can called at anytime */
- if (player->pd_downloader && MMPLAYER_IS_HTTP_PD(player))
+ /* create source element */
+ switch ( player->profile.uri_type )
{
- //__mmplayer_pd_deinitalize (player->ahs_player);
- _mmplayer_pd_stop ((MMHandleType)player);
- }
+ /* rtsp streamming */
+ case MM_PLAYER_URI_TYPE_URL_RTSP:
+ {
+ gint network_bandwidth;
+ gchar *user_agent, *wap_profile;
-#ifdef NO_USE_GST_HLSDEMUX
- /* destroy can called at anytime */
- if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
- {
- __mm_player_ahs_stop (player->ahs_player);
- //__mm_player_ahs_deinitalize (player->ahs_player);
- }
-#endif
- return MM_ERROR_NONE;
+ element = gst_element_factory_make(player->ini.name_of_rtspsrc, "streaming_source");
-}
+ if ( !element )
+ {
+ debug_error("failed to create streaming source element\n");
+ break;
+ }
-int
-_mmplayer_unrealize(MMHandleType hplayer) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- int ret = MM_ERROR_NONE;
+ debug_log("using streamming source [%s].\n", player->ini.name_of_rtspsrc);
- debug_fenter();
+ /* make it zero */
+ network_bandwidth = 0;
+ user_agent = wap_profile = NULL;
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED )
+ /* get attribute */
+ mm_attrs_get_string_by_name ( attrs, "streaming_user_agent", &user_agent );
+ mm_attrs_get_string_by_name ( attrs,"streaming_wap_profile", &wap_profile );
+ mm_attrs_get_int_by_name ( attrs, "streaming_network_bandwidth", &network_bandwidth );
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_UNREALIZE );
+ secure_debug_log("user_agent : %s\n", user_agent);
+ secure_debug_log("wap_profile : %s\n", wap_profile);
+ debug_log("network_bandwidth : %d\n", network_bandwidth);
+ debug_log("buffering time : %d\n", player->ini.rtsp_buffering_time);
+ debug_log("rebuffering time : %d\n", player->ini.rtsp_rebuffering_time);
- __mmplayer_deinit_extended_streaming(player);
+ /* setting property to streaming source */
+ g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
+ g_object_set(G_OBJECT(element), "bandwidth", network_bandwidth, NULL);
+ g_object_set(G_OBJECT(element), "buffering_time", player->ini.rtsp_buffering_time, NULL);
+ g_object_set(G_OBJECT(element), "rebuffering_time", player->ini.rtsp_rebuffering_time, NULL);
+ if ( user_agent )
+ g_object_set(G_OBJECT(element), "user_agent", user_agent, NULL);
+ if ( wap_profile )
+ g_object_set(G_OBJECT(element), "wap_profile", wap_profile, NULL);
- /* unrealize pipeline */
- ret = __gst_unrealize( player );
+ MMPLAYER_SIGNAL_CONNECT ( player, G_OBJECT(element), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added",
+ G_CALLBACK (__mmplayer_gst_rtp_dynamic_pad), player );
+ MMPLAYER_SIGNAL_CONNECT ( player, G_OBJECT(element), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "no-more-pads",
+ G_CALLBACK (__mmplayer_gst_rtp_no_more_pads), player );
- /* set player state if success */
- if ( MM_ERROR_NONE == ret )
- {
- ret = _mmplayer_asm_set_state(hplayer, ASM_STATE_STOP);
- if ( ret )
- {
- debug_error("failed to set asm state to STOP\n");
- return ret;
+ player->use_decodebin = FALSE;
}
- }
+ break;
- debug_fleave();
+ /* WFD streamming */
+ case MM_PLAYER_URI_TYPE_URL_WFD:
+ {
+ element = gst_element_factory_make("wfdrtspsrc", "wfd_source");
+ if ( !element )
+ {
+ debug_error("failed to create wfd streaming source element\n");
+ break;
+ }
+ debug_log("using wfd streamming source wfdrtspsrc.\n");
+ g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
+ g_object_set(G_OBJECT(element), "debug", TRUE, NULL);
+ g_object_set(G_OBJECT(element), "latency", 0, NULL);
+ MMPLAYER_SIGNAL_CONNECT ( player, G_OBJECT(element), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added",
+ G_CALLBACK (__mmplayer_gst_wfd_dynamic_pad), player );
- return ret;
-}
+ player->use_decodebin = FALSE;
+ }
+ break;
-int
-_mmplayer_set_message_callback(MMHandleType hplayer, MMMessageCallback callback, gpointer user_param) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
+ /* http streaming*/
+ case MM_PLAYER_URI_TYPE_URL_HTTP:
+ {
+ gchar *user_agent, *proxy, *cookies, **cookie_list;
+ gint http_timeout = DEFAULT_HTTP_TIMEOUT;
+ user_agent = proxy = cookies = NULL;
+ cookie_list = NULL;
+ gint mode = MM_PLAYER_PD_MODE_NONE;
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ mm_attrs_get_int_by_name ( attrs, "pd_mode", &mode );
- return __gst_set_message_callback(player, callback, user_param);
-}
+ player->pd_mode = mode;
-int
-_mmplayer_get_state(MMHandleType hplayer, int* state) // @
-{
- mm_player_t *player = (mm_player_t*)hplayer;
+ debug_log("http playback, PD mode : %d\n", player->pd_mode);
- return_val_if_fail(state, MM_ERROR_INVALID_ARGUMENT);
+ if ( ! MMPLAYER_IS_HTTP_PD(player) )
+ {
+ element = gst_element_factory_make(player->ini.name_of_httpsrc, "http_streaming_source");
+ if ( !element )
+ {
+ debug_error("failed to create http streaming source element[%s].\n", player->ini.name_of_httpsrc);
+ break;
+ }
+ debug_log("using http streamming source [%s].\n", player->ini.name_of_httpsrc);
- *state = MMPLAYER_CURRENT_STATE(player);
+ /* get attribute */
+ mm_attrs_get_string_by_name ( attrs, "streaming_cookie", &cookies );
+ mm_attrs_get_string_by_name ( attrs, "streaming_user_agent", &user_agent );
+ mm_attrs_get_string_by_name ( attrs, "streaming_proxy", &proxy );
+ mm_attrs_get_int_by_name ( attrs, "streaming_timeout", &http_timeout );
- return MM_ERROR_NONE;
-}
+ if ((http_timeout == DEFAULT_HTTP_TIMEOUT) &&
+ (player->ini.http_timeout != DEFAULT_HTTP_TIMEOUT))
+ {
+ debug_log("get timeout from ini\n");
+ http_timeout = player->ini.http_timeout;
+ }
+ /* get attribute */
+ secure_debug_log("location : %s\n", player->profile.uri);
+ secure_debug_log("cookies : %s\n", cookies);
+ secure_debug_log("proxy : %s\n", proxy);
+ secure_debug_log("user_agent : %s\n", user_agent);
+ debug_log("timeout : %d\n", http_timeout);
-int
-_mmplayer_set_volume(MMHandleType hplayer, MMPlayerVolumeType volume) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
- GstElement* vol_element = NULL;
- int i = 0;
+ /* setting property to streaming source */
+ g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
+ g_object_set(G_OBJECT(element), "timeout", http_timeout, NULL);
+ g_object_set(G_OBJECT(element), "blocksize", (unsigned long)(64*1024), NULL);
- debug_fenter();
+ /* check if prosy is vailid or not */
+ if ( util_check_valid_url ( proxy ) )
+ g_object_set(G_OBJECT(element), "proxy", proxy, NULL);
+ /* parsing cookies */
+ if ( ( cookie_list = util_get_cookie_list ((const char*)cookies) ) )
+ g_object_set(G_OBJECT(element), "cookies", cookie_list, NULL);
+ if ( user_agent )
+ g_object_set(G_OBJECT(element), "user_agent", user_agent, NULL);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ if ( MMPLAYER_URL_HAS_DASH_SUFFIX(player) )
+ {
+ debug_warning("it's dash. and it's still experimental feature.");
+ }
+ else if (MMPLAYER_URL_HAS_HLS_SUFFIX(player) )
+ {
+ debug_warning("it's hls. using decodebin");
+ }
+ }
+ else // progressive download
+ {
+ gchar* location = NULL;
- debug_log("volume [L]=%f:[R]=%f\n",
- volume.level[MM_VOLUME_CHANNEL_LEFT], volume.level[MM_VOLUME_CHANNEL_RIGHT]);
+ if (player->pd_mode == MM_PLAYER_PD_MODE_URI)
+ {
+ gchar *path = NULL;
- /* invalid factor range or not */
- for ( i = 0; i < MM_VOLUME_CHANNEL_NUM; i++ )
- {
- if (volume.level[i] < MM_VOLUME_FACTOR_MIN || volume.level[i] > MM_VOLUME_FACTOR_MAX) {
- debug_error("Invalid factor! (valid factor:0~1.0)\n");
- return MM_ERROR_INVALID_ARGUMENT;
- }
- }
+ mm_attrs_get_string_by_name ( attrs, "pd_location", &path );
- /* Save volume to handle. Currently the first array element will be saved. */
- player->sound.volume = volume.level[0];
+ MMPLAYER_FREEIF(player->pd_file_save_path);
- /* check pipeline handle */
- if ( ! player->pipeline || ! player->pipeline->audiobin )
- {
- debug_log("audiobin is not created yet\n");
- debug_log("but, current stored volume will be set when it's created.\n");
+ debug_log("PD Location : %s\n", path);
- /* NOTE : stored volume will be used in create_audiobin
- * returning MM_ERROR_NONE here makes application to able to
- * set volume at anytime.
- */
- return MM_ERROR_NONE;
- }
+ if ( path )
+ {
+ player->pd_file_save_path = g_strdup(path);
+ }
+ else
+ {
+ debug_error("can't find pd location so, it should be set \n");
+ return MM_ERROR_PLAYER_FILE_NOT_FOUND;
+ }
+ }
- /* setting volume to volume element */
- vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+ element = gst_element_factory_make("pdpushsrc", "PD pushsrc");
+ if ( !element )
+ {
+ debug_error("failed to create PD push source element[%s].\n", "pdpushsrc");
+ break;
+ }
- if ( vol_element )
- {
- debug_log("volume is set [%f]\n", player->sound.volume);
- g_object_set(vol_element, "volume", player->sound.volume, NULL);
- }
+ if (player->pd_mode == MM_PLAYER_PD_MODE_URI)
+ g_object_set(G_OBJECT(element), "location", player->pd_file_save_path, NULL);
+ else
+ g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
- debug_fleave();
+ g_object_get(element, "location", &location, NULL);
+ debug_log("PD_LOCATION [%s].\n", location);
+ if (location)
+ g_free (location);
+ }
+ }
+ break;
- return MM_ERROR_NONE;
-}
+ /* file source */
+ case MM_PLAYER_URI_TYPE_FILE:
+ {
+#ifdef ENABLE_DRMSRC
-int
-_mmplayer_get_volume(MMHandleType hplayer, MMPlayerVolumeType* volume)
-{
- mm_player_t* player = (mm_player_t*) hplayer;
- int i = 0;
+ char* drmsrc = player->ini.name_of_drmsrc;
- debug_fenter();
+ debug_log("using [%s] for 'file://' handler.\n", drmsrc);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail( volume, MM_ERROR_INVALID_ARGUMENT );
+ element = gst_element_factory_make(drmsrc, "source");
+ if ( !element )
+ {
+ debug_error("failed to create %s\n", drmsrc);
+ break;
+ }
+#else
+ debug_log("using filesrc for 'file://' handler.\n");
- /* returning stored volume */
- for (i = 0; i < MM_VOLUME_CHANNEL_NUM; i++)
- volume->level[i] = player->sound.volume;
+ element = gst_element_factory_make("filesrc", "source");
- debug_fleave();
+ if ( !element )
+ {
+ debug_error("failed to create filesrc\n");
+ break;
+ }
+#endif
+ g_object_set(G_OBJECT(element), "location", (player->profile.uri)+7, NULL); /* uri+7 -> remove "file:// */
+ //g_object_set(G_OBJECT(element), "use-mmap", TRUE, NULL);
+ }
+ break;
- return MM_ERROR_NONE;
-}
+ case MM_PLAYER_URI_TYPE_SS:
+ {
+ gint http_timeout = DEFAULT_HTTP_TIMEOUT;
+ element = gst_element_factory_make("souphttpsrc", "http streaming source");
+ if ( !element )
+ {
+ debug_error("failed to create http streaming source element[%s]", player->ini.name_of_httpsrc);
+ break;
+ }
+ mm_attrs_get_int_by_name ( attrs, "streaming_timeout", &http_timeout );
+ if ((http_timeout == DEFAULT_HTTP_TIMEOUT) &&
+ (player->ini.http_timeout != DEFAULT_HTTP_TIMEOUT))
+ {
+ debug_log("get timeout from ini\n");
+ http_timeout = player->ini.http_timeout;
+ }
-int
-_mmplayer_set_mute(MMHandleType hplayer, int mute) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
- GstElement* vol_element = NULL;
+ /* setting property to streaming source */
+ g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
+ g_object_set(G_OBJECT(element), "timeout", http_timeout, NULL);
+ }
+ break;
- debug_fenter();
+ /* appsrc */
+ case MM_PLAYER_URI_TYPE_BUFF:
+ {
+ guint64 stream_type = GST_APP_STREAM_TYPE_STREAM;
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ debug_log("mem src is selected\n");
- debug_log("mute : %d\n", mute);
+ element = gst_element_factory_make("appsrc", "buff-source");
+ if ( !element )
+ {
+ debug_error("failed to create appsrc element\n");
+ break;
+ }
- /* mute value shoud 0 or 1 */
- if ( mute != 0 && mute != 1 )
- {
- debug_error("bad mute value\n");
+ g_object_set( element, "stream-type", stream_type, NULL );
+ //g_object_set( element, "size", player->mem_buf.len, NULL );
+ //g_object_set( element, "blocksize", (guint64)20480, NULL );
- /* FIXIT : definitly, we need _BAD_PARAM error code */
- return MM_ERROR_INVALID_ARGUMENT;
- }
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_OTHERS, "seek-data",
+ G_CALLBACK(__gst_appsrc_seek_data), player);
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_OTHERS, "need-data",
+ G_CALLBACK(__gst_appsrc_feed_data), player);
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_OTHERS, "enough-data",
+ G_CALLBACK(__gst_appsrc_enough_data), player);
+ }
+ break;
+ /* appsrc */
+ case MM_PLAYER_URI_TYPE_MEM:
+ {
+ guint64 stream_type = GST_APP_STREAM_TYPE_RANDOM_ACCESS;
- /* just hold mute value if pipeline is not ready */
- if ( !player->pipeline || !player->pipeline->audiobin )
- {
- debug_log("pipeline is not ready. holding mute value\n");
- player->sound.mute = mute;
- return MM_ERROR_NONE;
- }
+ debug_log("mem src is selected\n");
+ element = gst_element_factory_make("appsrc", "mem-source");
+ if ( !element )
+ {
+ debug_error("failed to create appsrc element\n");
+ break;
+ }
- vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+ g_object_set( element, "stream-type", stream_type, NULL );
+ g_object_set( element, "size", player->mem_buf.len, NULL );
+ g_object_set( element, "blocksize", (guint64)20480, NULL );
- /* NOTE : volume will only created when the bt is enabled */
- if ( vol_element )
- {
- g_object_set(vol_element, "mute", mute, NULL);
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_OTHERS, "seek-data",
+ G_CALLBACK(__gst_appsrc_seek_data_mem), &player->mem_buf );
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_OTHERS, "need-data",
+ G_CALLBACK(__gst_appsrc_feed_data_mem), &player->mem_buf );
+ }
+ break;
+ case MM_PLAYER_URI_TYPE_URL:
+ break;
+
+ case MM_PLAYER_URI_TYPE_TEMP:
+ break;
+
+ case MM_PLAYER_URI_TYPE_NONE:
+ default:
+ break;
}
- else
+
+ /* check source element is OK */
+ if ( ! element )
{
- debug_log("volume elemnet is not created. using volume in audiosink\n");
+ debug_error("no source element was created.\n");
+ goto INIT_ERROR;
}
- player->sound.mute = mute;
+ /* take source element */
+ mainbin[MMPLAYER_M_SRC].id = MMPLAYER_M_SRC;
+ mainbin[MMPLAYER_M_SRC].gst = element;
+ element_bucket = g_list_append(element_bucket, &mainbin[MMPLAYER_M_SRC]);
- debug_fleave();
+ if ((MMPLAYER_IS_STREAMING(player)) && (player->streamer == NULL))
+ {
+ player->streamer = __mm_player_streaming_create();
+ __mm_player_streaming_initialize(player->streamer);
+ }
- return MM_ERROR_NONE;
-}
+ if ( MMPLAYER_IS_HTTP_PD(player) )
+ {
+ gdouble pre_buffering_time = (gdouble)player->streamer->buffering_req.initial_second;
-int
-_mmplayer_get_mute(MMHandleType hplayer, int* pmute) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
- GstElement* vol_element = NULL;
+ debug_log ("Picked queue2 element(pre buffer : %d sec)....\n", pre_buffering_time);
+ element = gst_element_factory_make("queue2", "queue2");
+ if ( !element )
+ {
+ debug_error ( "failed to create http streaming buffer element\n" );
+ goto INIT_ERROR;
+ }
- debug_fenter();
+ /* take it */
+ mainbin[MMPLAYER_M_MUXED_S_BUFFER].id = MMPLAYER_M_MUXED_S_BUFFER;
+ mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst = element;
+ element_bucket = g_list_append(element_bucket, &mainbin[MMPLAYER_M_MUXED_S_BUFFER]);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( pmute, MM_ERROR_INVALID_ARGUMENT );
+ pre_buffering_time = (pre_buffering_time > 0)?(pre_buffering_time):(player->ini.http_buffering_time);
- /* just hold mute value if pipeline is not ready */
- if ( !player->pipeline || !player->pipeline->audiobin )
- {
- debug_log("pipeline is not ready. returning stored value\n");
- *pmute = player->sound.mute;
- return MM_ERROR_NONE;
+ __mm_player_streaming_set_queue2(player->streamer,
+ element,
+ TRUE,
+ player->ini.http_max_size_bytes,
+ pre_buffering_time,
+ 1.0,
+ player->ini.http_buffering_limit,
+ FALSE,
+ NULL,
+ 0);
}
+ /* create autoplugging element if src element is not a rtsp src */
+ if ((player->profile.uri_type != MM_PLAYER_URI_TYPE_URL_RTSP) &&
+ (player->profile.uri_type != MM_PLAYER_URI_TYPE_URL_WFD))
+ {
+ element = NULL;
+ enum MainElementID elemId = MMPLAYER_M_NUM;
- vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+ if ((player->use_decodebin) &&
+ ((MMPLAYER_IS_HTTP_PD(player)) ||
+ (!MMPLAYER_IS_HTTP_STREAMING(player))))
+ {
+ elemId = MMPLAYER_M_AUTOPLUG;
+ element = __mmplayer_create_decodebin(player);
+ need_state_holder = FALSE;
+ }
+ else
+ {
+ elemId = MMPLAYER_M_TYPEFIND;
+ element = gst_element_factory_make("typefind", "typefinder");
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "have-type",
+ G_CALLBACK(__mmplayer_typefind_have_type), (gpointer)player );
+ }
- if ( vol_element )
+
+ /* check autoplug element is OK */
+ if ( ! element )
+ {
+ debug_error("can not create element (%d)\n", elemId);
+ goto INIT_ERROR;
+ }
+
+ mainbin[elemId].id = elemId;
+ mainbin[elemId].gst = element;
+
+ element_bucket = g_list_append(element_bucket, &mainbin[elemId]);
+ }
+
+ /* add elements to pipeline */
+ if( !__mmplayer_gst_element_add_bucket_to_bin(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), element_bucket))
{
- g_object_get(vol_element, "mute", pmute, NULL);
- debug_log("mute=%d\n\n", *pmute);
+ debug_error("Failed to add elements to pipeline\n");
+ goto INIT_ERROR;
}
- else
+
+
+ /* linking elements in the bucket by added order. */
+ if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
{
- *pmute = player->sound.mute;
+ debug_error("Failed to link some elements\n");
+ goto INIT_ERROR;
}
- debug_fleave();
- return MM_ERROR_NONE;
-}
+ /* create fakesink element for keeping the pipeline state PAUSED. if needed */
+ if ( need_state_holder )
+ {
+ /* create */
+ mainbin[MMPLAYER_M_SRC_FAKESINK].id = MMPLAYER_M_SRC_FAKESINK;
+ mainbin[MMPLAYER_M_SRC_FAKESINK].gst = gst_element_factory_make ("fakesink", "state-holder");
-int
-_mmplayer_set_videostream_cb(MMHandleType hplayer, mm_player_video_stream_callback callback, void *user_param) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
+ if (!mainbin[MMPLAYER_M_SRC_FAKESINK].gst)
+ {
+ debug_error ("fakesink element could not be created\n");
+ goto INIT_ERROR;
+ }
+ GST_OBJECT_FLAG_UNSET (mainbin[MMPLAYER_M_SRC_FAKESINK].gst, GST_ELEMENT_IS_SINK);
- debug_fenter();
+ /* take ownership of fakesink. we are reusing it */
+ gst_object_ref( mainbin[MMPLAYER_M_SRC_FAKESINK].gst );
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( callback, MM_ERROR_INVALID_ARGUMENT );
+ /* add */
+ if ( FALSE == gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst),
+ mainbin[MMPLAYER_M_SRC_FAKESINK].gst) )
+ {
+ debug_error("failed to add fakesink to bin\n");
+ goto INIT_ERROR;
+ }
+ }
- player->video_stream_cb = callback;
- player->video_stream_cb_user_param = user_param;
- player->use_video_stream = TRUE;
- debug_log("Stream cb Handle value is %p : %p\n", player, player->video_stream_cb);
+ /* now we have completed mainbin. take it */
+ player->pipeline->mainbin = mainbin;
- debug_fleave();
+ /* connect bus callback */
+ bus = gst_pipeline_get_bus(GST_PIPELINE(mainbin[MMPLAYER_M_PIPE].gst));
+ if ( !bus )
+ {
+ debug_error ("cannot get bus from pipeline.\n");
+ goto INIT_ERROR;
+ }
- return MM_ERROR_NONE;
-}
+ player->bus_watcher = gst_bus_add_watch(bus, (GstBusFunc)__mmplayer_gst_callback, player);
-int
-_mmplayer_set_audiostream_cb(MMHandleType hplayer, mm_player_audio_stream_callback callback, void *user_param) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
+ player->context.thread_default = g_main_context_get_thread_default();
- debug_fenter();
+ if (NULL == player->context.thread_default)
+ {
+ player->context.thread_default = g_main_context_default();
+ debug_log("thread-default context is the global default context");
+ }
+ debug_warning("bus watcher thread context = %p, watcher : %d", player->context.thread_default, player->bus_watcher);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+ /* Note : check whether subtitle atrribute uri is set. If uri is set, then try to play subtitle file */
+ if ( __mmplayer_check_subtitle ( player ) )
+ {
+ if ( MM_ERROR_NONE != __mmplayer_gst_create_subtitle_src(player) )
+ debug_error("fail to create subtitle src\n");
+ }
- player->audio_stream_cb = callback;
- player->audio_stream_cb_user_param = user_param;
- debug_log("Audio Stream cb Handle value is %p : %p\n", player, player->audio_stream_cb);
+ /* set sync handler to get tag synchronously */
+ gst_bus_set_sync_handler(bus, __mmplayer_bus_sync_callback, player);
+
+ /* finished */
+ gst_object_unref(GST_OBJECT(bus));
+ g_list_free(element_bucket);
- debug_fleave();
+ MMPLAYER_FLEAVE();
return MM_ERROR_NONE;
-}
-int
-_mmplayer_set_audiobuffer_cb(MMHandleType hplayer, mm_player_audio_stream_callback callback, void *user_param) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
+INIT_ERROR:
- debug_fenter();
+ __mmplayer_gst_destroy_pipeline(player);
+ g_list_free(element_bucket);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+ /* release element which are not added to bin */
+ for ( i = 1; i < MMPLAYER_M_NUM; i++ ) /* NOTE : skip pipeline */
+ {
+ if ( mainbin[i].gst )
+ {
+ GstObject* parent = NULL;
+ parent = gst_element_get_parent( mainbin[i].gst );
+
+ if ( !parent )
+ {
+ gst_object_unref(GST_OBJECT(mainbin[i].gst));
+ mainbin[i].gst = NULL;
+ }
+ else
+ {
+ gst_object_unref(GST_OBJECT(parent));
+ }
+ }
+ }
- player->audio_buffer_cb = callback;
- player->audio_buffer_cb_user_param = user_param;
- debug_log("Audio Stream cb Handle value is %p : %p\n", player, player->audio_buffer_cb);
+ /* release pipeline with it's childs */
+ if ( mainbin[MMPLAYER_M_PIPE].gst )
+ {
+ gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_PIPE].gst));
+ }
- debug_fleave();
+ MMPLAYER_FREEIF( player->pipeline );
+ MMPLAYER_FREEIF( mainbin );
- return MM_ERROR_NONE;
+ return MM_ERROR_PLAYER_INTERNAL;
}
-int
-_mmplayer_set_buffer_need_data_cb(MMHandleType hplayer, mm_player_buffer_need_data_callback callback, void *user_param) // @
+void __mmplayer_remove_g_source_from_context(GMainContext *context, guint source_id)
{
- mm_player_t* player = (mm_player_t*) hplayer;
-
- debug_fenter();
-
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+ GSource *source = NULL;
- player->need_data_cb = callback;
- player->buffer_cb_user_param = user_param;
+ MMPLAYER_FENTER();
- debug_log("buffer need dataHandle value is %p : %p\n", player, player->need_data_cb);
+ source = g_main_context_find_source_by_id (context, source_id);
- debug_fleave();
+ if (source != NULL)
+ {
+ debug_warning("context: %p, source id: %d, source: %p", context, source_id, source);
+ g_source_destroy(source);
+ }
- return MM_ERROR_NONE;
+ MMPLAYER_FLEAVE();
}
-int
-_mmplayer_set_buffer_enough_data_cb(MMHandleType hplayer, mm_player_buffer_enough_data_callback callback, void *user_param) // @
+static int
+__mmplayer_gst_destroy_pipeline(mm_player_t* player) // @
{
- mm_player_t* player = (mm_player_t*) hplayer;
-
- debug_fenter();
+ gint timeout = 0;
+ int ret = MM_ERROR_NONE;
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+ MMPLAYER_FENTER();
- player->enough_data_cb = callback;
- player->buffer_cb_user_param = user_param;
+ return_val_if_fail ( player, MM_ERROR_INVALID_HANDLE );
- debug_log("buffer enough data cb Handle value is %p : %p\n", player, player->enough_data_cb);
+ /* cleanup stuffs */
+ MMPLAYER_FREEIF(player->type);
+ player->have_dynamic_pad = FALSE;
+ player->no_more_pad = FALSE;
+ player->num_dynamic_pad = 0;
+ player->demux_pad_index = 0;
+ player->subtitle_language_list = NULL;
+ player->use_deinterleave = FALSE;
+ player->pp_rebuilding = FALSE;
+ player->max_audio_channels = 0;
+ player->video_share_api_delta = 0;
+ player->video_share_clock_delta = 0;
+ player->video_hub_download_mode = 0;
- debug_fleave();
-
- return MM_ERROR_NONE;
-}
+ MMPLAYER_FREEIF(player->font_desc);
+ player->font_color = 0;
+ player->font_bg_color = 0;
-int
-_mmplayer_set_buffer_seek_data_cb(MMHandleType hplayer, mm_player_buffer_seek_data_callback callback, void *user_param) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
+ __mmplayer_post_proc_reset(player);
- debug_fenter();
+ if (player->streamer)
+ {
+ __mm_player_streaming_deinitialize (player->streamer);
+ __mm_player_streaming_destroy(player->streamer);
+ player->streamer = NULL;
+ }
- return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+ /* cleanup unlinked mime type */
+ MMPLAYER_FREEIF(player->unlinked_audio_mime);
+ MMPLAYER_FREEIF(player->unlinked_video_mime);
+ MMPLAYER_FREEIF(player->unlinked_demuxer_mime);
- player->seek_data_cb = callback;
- player->buffer_cb_user_param = user_param;
+ /* cleanup running stuffs */
+ __mmplayer_cancel_eos_timer( player );
- debug_log("buffer seek data cb Handle value is %p : %p\n", player, player->seek_data_cb);
+ /* remove sound cb */
+ if ( MM_ERROR_NONE != mm_sound_remove_active_device_changed_callback(MM_PLAYER_NAME))
+ {
+ debug_error("failed to mm_sound_remove_active_device_changed_callback");
+ }
- debug_fleave();
+ /* cleanup gst stuffs */
+ if ( player->pipeline )
+ {
+ MMPlayerGstElement* mainbin = player->pipeline->mainbin;
+ GstTagList* tag_list = player->pipeline->tag_list;
- return MM_ERROR_NONE;
-}
+ /* first we need to disconnect all signal hander */
+ __mmplayer_release_signal_connection( player, MM_PLAYER_SIGNAL_TYPE_ALL );
-int __mmplayer_start_extended_streaming(mm_player_t *player)
-{
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ /* disconnecting bus watch */
+ if ( player->bus_watcher )
+ __mmplayer_remove_g_source_from_context(player->context.thread_default, player->bus_watcher);
+ player->bus_watcher = 0;
- if (MMPLAYER_IS_HTTP_PD(player) && player->pd_downloader)
- {
- if (player->pd_mode == MM_PLAYER_PD_MODE_URI)
+ if ( mainbin )
{
- gboolean bret = FALSE;
-
- bret = _mmplayer_pd_start ((MMHandleType)player);
- if (FALSE == bret)
+ MMPlayerGstElement* audiobin = player->pipeline->audiobin;
+ MMPlayerGstElement* videobin = player->pipeline->videobin;
+ MMPlayerGstElement* textbin = player->pipeline->textbin;
+ GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE (mainbin[MMPLAYER_M_PIPE].gst));
+ gst_bus_set_sync_handler (bus, NULL, NULL);
+ gst_object_unref(bus);
+
+ timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
+ ret = __mmplayer_gst_set_state ( player, mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_NULL, FALSE, timeout );
+ if ( ret != MM_ERROR_NONE )
{
- debug_error ("ERROR while starting PD...\n");
- return MM_ERROR_PLAYER_NOT_INITIALIZED;
+ debug_error("fail to change state to NULL\n");
+ return MM_ERROR_PLAYER_INTERNAL;
}
- }
- }
-
-#ifdef NO_USE_GST_HLSDEMUX
- if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
- {
- if ( !__mm_player_ahs_start (player->ahs_player))
- {
- debug_error("failed to start ahs\n");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- }
-#endif
- return MM_ERROR_NONE;
-}
-int
-_mmplayer_start(MMHandleType hplayer) // @
-{
- mm_player_t* player = (mm_player_t*) hplayer;
- gint ret = MM_ERROR_NONE;
-
- debug_fenter();
+ debug_warning("succeeded in chaning state to NULL\n");
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_PIPE].gst));
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_START );
+ /* free fakesink */
+ if ( mainbin[MMPLAYER_M_SRC_FAKESINK].gst )
+ gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_SRC_FAKESINK].gst));
- ret = _mmplayer_asm_set_state(hplayer, ASM_STATE_PLAYING);
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("failed to set asm state to PLAYING\n");
- return ret;
- }
-
- /* NOTE : we should check and create pipeline again if not created as we destroy
- * whole pipeline when stopping in streamming playback
- */
- if ( ! player->pipeline )
- {
- ret = __gst_realize( player );
- if ( MM_ERROR_NONE != ret )
- {
- debug_error("failed to realize before starting. only in streamming\n");
- return ret;
+ /* free avsysaudiosink
+ avsysaudiosink should be unref when destory pipeline just after start play with BT.
+ Because audiosink is created but never added to bin, and therefore it will not be unref when pipeline is destroyed.
+ */
+ MMPLAYER_FREEIF( audiobin );
+ MMPLAYER_FREEIF( videobin );
+ MMPLAYER_FREEIF( textbin );
+ MMPLAYER_FREEIF( mainbin );
}
+
+ if ( tag_list )
+ gst_tag_list_free(tag_list);
+
+ MMPLAYER_FREEIF( player->pipeline );
}
+ MMPLAYER_FREEIF(player->album_art);
- if (__mmplayer_start_extended_streaming(player) != MM_ERROR_NONE)
- return MM_ERROR_PLAYER_INTERNAL;
-
- /* start pipeline */
- ret = __gst_start( player );
- if ( ret != MM_ERROR_NONE )
+ if (player->v_stream_caps)
{
- debug_error("failed to start player.\n");
+ gst_caps_unref(player->v_stream_caps);
+ player->v_stream_caps = NULL;
}
- debug_fleave();
-
- return ret;
-}
+ if ( player->sink_elements )
+ g_list_free ( player->sink_elements );
+ player->sink_elements = NULL;
+ debug_warning("finished destroy pipeline\n");
+ MMPLAYER_FLEAVE();
-/* NOTE: post "not supported codec message" to application
- * when one codec is not found during AUTOPLUGGING in MSL.
- * So, it's separated with error of __mmplayer_gst_callback().
- * And, if any codec is not found, don't send message here.
- * Because GST_ERROR_MESSAGE is posted by other plugin internally.
- */
-int
-__mmplayer_post_missed_plugin(mm_player_t* player)
+ return ret;
+}
+
+static int __gst_realize(mm_player_t* player) // @
{
- MMMessageParamType msg_param;
- memset (&msg_param, 0, sizeof(MMMessageParamType));
- gboolean post_msg_direct = FALSE;
+ gint timeout = 0;
+ int ret = MM_ERROR_NONE;
- debug_fenter();
+ MMPLAYER_FENTER();
return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
- debug_log("not_supported_codec = 0x%02x, can_support_codec = 0x%02x\n",
- player->not_supported_codec, player->can_support_codec);
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_READY;
- if( player->not_found_demuxer )
+ ret = __mmplayer_gst_create_pipeline(player);
+ if ( ret )
{
- msg_param.code = MM_ERROR_PLAYER_CODEC_NOT_FOUND;
- msg_param.data = g_strdup_printf("%s", player->unlinked_demuxer_mime);
+ debug_error("failed to create pipeline\n");
+ return ret;
+ }
- MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
- MMPLAYER_FREEIF(msg_param.data);
+ /* set pipeline state to READY */
+ /* NOTE : state change to READY must be performed sync. */
+ timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
+ ret = __mmplayer_gst_set_state(player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_READY, FALSE, timeout);
- return MM_ERROR_NONE;
+ if ( ret != MM_ERROR_NONE )
+ {
+ /* return error if failed to set state */
+ debug_error("failed to set READY state");
+ return ret;
}
-
- if (player->not_supported_codec)
+ else
{
- if ( player->can_support_codec ) // There is one codec to play
- {
- post_msg_direct = TRUE;
- }
- else
- {
- if ( player->pipeline->audiobin ) // Some content has only PCM data in container.
- post_msg_direct = TRUE;
- }
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_READY );
+ }
- if ( post_msg_direct )
- {
- MMMessageParamType msg_param;
- memset (&msg_param, 0, sizeof(MMMessageParamType));
+ /* create dot before error-return. for debugging */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-realize" );
- if ( player->not_supported_codec == MISSING_PLUGIN_AUDIO )
- {
- debug_warning("not found AUDIO codec, posting error code to application.\n");
+ MMPLAYER_FLEAVE();
- msg_param.code = MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
- msg_param.data = g_strdup_printf("%s", player->unlinked_audio_mime);
- }
- else if ( player->not_supported_codec == MISSING_PLUGIN_VIDEO )
- {
- debug_warning("not found VIDEO codec, posting error code to application.\n");
+ return ret;
+}
- msg_param.code = MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
- msg_param.data = g_strdup_printf("%s", player->unlinked_video_mime);
- }
+static int __gst_unrealize(mm_player_t* player) // @
+{
+ int ret = MM_ERROR_NONE;
- MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+ MMPLAYER_FENTER();
- MMPLAYER_FREEIF(msg_param.data);
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
- return MM_ERROR_NONE;
- }
- else // no any supported codec case
- {
- debug_warning("not found any codec, posting error code to application.\n");
-
- if ( player->not_supported_codec == MISSING_PLUGIN_AUDIO )
- {
- msg_param.code = MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
- msg_param.data = g_strdup_printf("%s", player->unlinked_audio_mime);
- }
- else
- {
- msg_param.code = MM_ERROR_PLAYER_CODEC_NOT_FOUND;
- msg_param.data = g_strdup_printf("%s, %s", player->unlinked_video_mime, player->unlinked_audio_mime);
- }
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NULL;
+ MMPLAYER_PRINT_STATE(player);
- MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
-
- MMPLAYER_FREEIF(msg_param.data);
- }
+ /* release miscellaneous information */
+ __mmplayer_release_misc( player );
+
+ /* destroy pipeline */
+ ret = __mmplayer_gst_destroy_pipeline( player );
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to destory pipeline\n");
+ return ret;
}
-
- debug_fleave();
- return MM_ERROR_NONE;
+ /* release miscellaneous information.
+ these info needs to be released after pipeline is destroyed. */
+ __mmplayer_release_misc_post( player );
+
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_NULL );
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
}
-/* NOTE : it should be able to call 'stop' anytime*/
-int
-_mmplayer_stop(MMHandleType hplayer) // @
+static int __gst_pending_seek ( mm_player_t* player )
{
- mm_player_t* player = (mm_player_t*)hplayer;
+ MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
int ret = MM_ERROR_NONE;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ if ( !player->pending_seek.is_pending )
+ {
+ debug_log("pending seek is not reserved. nothing to do.\n" );
+ return ret;
+ }
+
+ /* check player state if player could pending seek or not. */
+ current_state = MMPLAYER_CURRENT_STATE(player);
- if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
+ if ( current_state != MM_PLAYER_STATE_PAUSED && current_state != MM_PLAYER_STATE_PLAYING )
{
- __mm_player_ahs_stop (player->ahs_player);
+ debug_warning("try to pending seek in %s state, try next time. \n",
+ MMPLAYER_STATE_GET_NAME(current_state));
+ return ret;
}
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_STOP );
+ debug_log("trying to play from (%lu) pending position\n", player->pending_seek.pos);
- /* NOTE : application should not wait for EOS after calling STOP */
- __mmplayer_cancel_delayed_eos( player );
+ ret = __gst_set_position ( player, player->pending_seek.format, player->pending_seek.pos, FALSE );
- /* stop pipeline */
- ret = __gst_stop( player );
+ if ( MM_ERROR_NONE != ret )
+ debug_error("failed to seek pending postion. just keep staying current position.\n");
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("failed to stop player.\n");
- }
+ player->pending_seek.is_pending = FALSE;
- debug_fleave();
+ MMPLAYER_FLEAVE();
return ret;
}
-
-int
-_mmplayer_pause(MMHandleType hplayer) // @
+static int __gst_start(mm_player_t* player) // @
{
- mm_player_t* player = (mm_player_t*)hplayer;
- MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
- GstFormat fmt = GST_FORMAT_TIME;
- signed long long pos_msec = 0;
+ gboolean sound_extraction = 0;
int ret = MM_ERROR_NONE;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_PAUSE );
+ /* get sound_extraction property */
+ mm_attrs_get_int_by_name(player->attrs, "pcm_extraction", &sound_extraction);
- /* NOTE : store current point to overcome some bad operation
- * ( returning zero when getting current position in paused state) of some
- * elements
- */
- current_state = MMPLAYER_CURRENT_STATE(player);
- if ( current_state == MM_PLAYER_STATE_PLAYING )
+ /* NOTE : if SetPosition was called before Start. do it now */
+ /* streaming doesn't support it. so it should be always sync */
+ /* !! create one more api to check if there is pending seek rather than checking variables */
+ if ( (player->pending_seek.is_pending || sound_extraction) && !MMPLAYER_IS_STREAMING(player))
{
- ret = gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &pos_msec);
- if ( ! ret )
- debug_warning("getting current position failed in paused\n");
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_PAUSED;
+ ret = __gst_pause(player, FALSE);
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to set state to PAUSED for pending seek\n");
+ return ret;
+ }
+
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_PLAYING;
+
+ if ( sound_extraction )
+ {
+ debug_log("setting pcm extraction\n");
- player->last_position = pos_msec;
+ ret = __mmplayer_set_pcm_extraction(player);
+ if ( MM_ERROR_NONE != ret )
+ {
+ debug_warning("failed to set pcm extraction\n");
+ return ret;
+ }
+ }
+ else
+ {
+ if ( MM_ERROR_NONE != __gst_pending_seek(player) )
+ {
+ debug_warning("failed to seek pending postion. starting from the begin of content.\n");
+ }
+ }
}
- /* pause pipeline */
- ret = __gst_pause( player, FALSE );
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PLAYING;
+ MMPLAYER_PRINT_STATE(player);
- if ( MMPLAYER_IS_HTTP_LIVE_STREAMING ( player ))
+ /* set pipeline state to PLAYING */
+ ret = __mmplayer_gst_set_state(player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PLAYING, FALSE, MMPLAYER_STATE_CHANGE_TIMEOUT(player) );
+ if (ret == MM_ERROR_NONE)
{
- //__mm_player_ahs_pause (player->ahs_player); // Not yet implemented
+ MMPLAYER_SET_STATE(player, MM_PLAYER_STATE_PLAYING);
}
-
- if ( ret != MM_ERROR_NONE )
+ else
{
- debug_error("failed to pause player.\n");
+ debug_error("failed to set state to PLAYING");
+ return ret;
}
- debug_fleave();
+ /* generating debug info before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-start" );
+
+ MMPLAYER_FLEAVE();
return ret;
}
-int
-_mmplayer_resume(MMHandleType hplayer)
+static void __mmplayer_do_sound_fadedown(mm_player_t* player, unsigned int time)
{
- mm_player_t* player = (mm_player_t*)hplayer;
- int ret = MM_ERROR_NONE;
+ MMPLAYER_FENTER();
- debug_fenter();
+ return_if_fail(player
+ && player->pipeline
+ && player->pipeline->audiobin
+ && player->pipeline->audiobin[MMPLAYER_A_SINK].gst);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "mute", 2, NULL);
- ret = _mmplayer_asm_set_state(hplayer, ASM_STATE_PLAYING);
- if ( ret )
- {
- debug_error("failed to set asm state to PLAYING\n");
- return ret;
- }
-
- /* check current state */
- MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_RESUME );
+ usleep(time);
- /* resume pipeline */
- ret = __gst_resume( player, FALSE );
+ MMPLAYER_FLEAVE();
+}
- if ( ret != MM_ERROR_NONE )
- {
- debug_error("failed to resume player.\n");
- }
+static void __mmplayer_undo_sound_fadedown(mm_player_t* player)
+{
+ MMPLAYER_FENTER();
+ return_if_fail(player
+ && player->pipeline
+ && player->pipeline->audiobin
+ && player->pipeline->audiobin[MMPLAYER_A_SINK].gst);
- debug_fleave();
+ g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "mute", 0, NULL);
- return ret;
+ MMPLAYER_FLEAVE();
}
-int
-__mmplayer_set_play_count(mm_player_t* player, gint count)
+static int __gst_stop(mm_player_t* player) // @
{
+ GstStateChangeReturn change_ret = GST_STATE_CHANGE_SUCCESS;
MMHandleType attrs = 0;
+ gboolean fadedown = FALSE;
+ gboolean rewind = FALSE;
+ gint timeout = 0;
+ int ret = MM_ERROR_NONE;
+ GstState state;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ return_val_if_fail ( player->pipeline->mainbin, MM_ERROR_PLAYER_NOT_INITIALIZED);
- attrs = MMPLAYER_GET_ATTRS(player);
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_READY;
+ MMPLAYER_PRINT_STATE(player);
+
+ attrs = MMPLAYER_GET_ATTRS(player);
if ( !attrs )
{
- debug_error("fail to get attributes.\n");
+ debug_error("cannot get content attribute\n");
return MM_ERROR_PLAYER_INTERNAL;
}
- mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
- if ( mmf_attrs_commit ( attrs ) ) /* return -1 if error */
- debug_error("failed to commit\n");
-
- debug_fleave();
+ mm_attrs_get_int_by_name(attrs, "sound_fadedown", &fadedown);
- return MM_ERROR_NONE;
-}
+ /* enable fadedown */
+ if (fadedown || player->sm.by_asm_cb)
+ __mmplayer_do_sound_fadedown(player, MM_PLAYER_FADEOUT_TIME_DEFAULT);
-int
-_mmplayer_activate_section_repeat(MMHandleType hplayer, unsigned long start, unsigned long end)
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- gint64 start_pos = 0;
- gint64 end_pos = 0;
- gint infinity = -1;
+ /* Just set state to PAUESED and the rewind. it's usual player behavior. */
+ timeout = MMPLAYER_STATE_CHANGE_TIMEOUT ( player );
- debug_fenter();
+ if (player->profile.uri_type == MM_PLAYER_URI_TYPE_BUFF ||
+ player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS)
+ {
+ state = GST_STATE_READY;
+ }
+ else
+ {
+ state = GST_STATE_PAUSED;
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( end <= GST_TIME_AS_MSECONDS(player->duration), MM_ERROR_INVALID_ARGUMENT );
+ if ( ! MMPLAYER_IS_STREAMING(player) ||
+ (player->streaming_type == STREAMING_SERVICE_VOD && player->videodec_linked)) {
+ rewind = TRUE;
+ }
+ }
- player->section_repeat = TRUE;
- player->section_repeat_start = start;
- player->section_repeat_end = end;
+ /* set gst state */
+ ret = __mmplayer_gst_set_state( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, state, FALSE, timeout );
- start_pos = player->section_repeat_start * G_GINT64_CONSTANT(1000000);
- end_pos = player->section_repeat_end * G_GINT64_CONSTANT(1000000);
+ /* disable fadeout */
+ if (fadedown || player->sm.by_asm_cb)
+ __mmplayer_undo_sound_fadedown(player);
- __mmplayer_set_play_count( player, infinity );
+ /* return if set_state has failed */
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to set state.\n");
+ return ret;
+ }
- if ( (!__gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
- 1.0,
- GST_FORMAT_TIME,
- ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- GST_SEEK_TYPE_SET, start_pos,
- GST_SEEK_TYPE_SET, end_pos)))
+ /* rewind */
+ if ( rewind )
{
- debug_error("failed to activate section repeat\n");
+ if ( ! __gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, player->playback_rate,
+ GST_FORMAT_TIME, GST_SEEK_FLAG_FLUSH, GST_SEEK_TYPE_SET, 0,
+ GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE) )
+ {
+ debug_warning("failed to rewind\n");
+ ret = MM_ERROR_PLAYER_SEEK;
+ }
+ }
- return MM_ERROR_PLAYER_SEEK;
+ /* initialize */
+ player->sent_bos = FALSE;
+
+ /* wait for seek to complete */
+ change_ret = gst_element_get_state (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, NULL, NULL, timeout * GST_SECOND);
+ if ( change_ret == GST_STATE_CHANGE_SUCCESS || change_ret == GST_STATE_CHANGE_NO_PREROLL )
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_READY );
+ }
+ else
+ {
+ debug_error("fail to stop player.\n");
+ ret = MM_ERROR_PLAYER_INTERNAL;
+ __mmplayer_dump_pipeline_state(player);
}
- debug_log("succeeded to set section repeat from %d to %d\n",
- player->section_repeat_start, player->section_repeat_end);
+ /* generate dot file if enabled */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-stop" );
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return MM_ERROR_NONE;
+ return ret;
}
-static int
-__mmplayer_set_pcm_extraction(mm_player_t* player)
+int __gst_pause(mm_player_t* player, gboolean async) // @
{
- guint64 start_nsec = 0;
- guint64 end_nsec = 0;
- guint64 dur_nsec = 0;
- guint64 dur_msec = 0;
- GstFormat fmt = GST_FORMAT_TIME;
- int required_start = 0;
- int required_end = 0;
- int ret = 0;
-
- debug_fenter();
-
- return_val_if_fail( player, FALSE );
+ int ret = MM_ERROR_NONE;
- mm_attrs_multiple_get(player->attrs,
- NULL,
- "pcm_extraction_start_msec", &required_start,
- "pcm_extraction_end_msec", &required_end,
- NULL);
+ MMPLAYER_FENTER();
- debug_log("pcm extraction required position is from [%d] to [%d] (msec)\n", required_start, required_end);
+ return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ return_val_if_fail(player->pipeline->mainbin, MM_ERROR_PLAYER_NOT_INITIALIZED);
- if (required_start == 0 && required_end == 0)
- {
- debug_log("extracting entire stream");
- return MM_ERROR_NONE;
- }
- else if (required_start < 0 || required_start > required_end || required_end < 0 )
- {
- debug_log("invalid range for pcm extraction");
- return MM_ERROR_INVALID_ARGUMENT;
- }
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PAUSED;
+ MMPLAYER_PRINT_STATE(player);
- /* get duration */
- ret = gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &dur_nsec);
- if ( !ret )
- {
- debug_error("failed to get duration");
- return MM_ERROR_PLAYER_INTERNAL;
- }
- dur_msec = GST_TIME_AS_MSECONDS(dur_nsec);
+ /* set pipeline status to PAUSED */
+ player->ignore_asyncdone = TRUE;
- if (dur_msec < required_end) // FIXME
- {
- debug_log("invalid end pos for pcm extraction");
- return MM_ERROR_INVALID_ARGUMENT;
- }
+ ret = __mmplayer_gst_set_state(player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PAUSED, async, MMPLAYER_STATE_CHANGE_TIMEOUT(player));
- start_nsec = required_start * G_GINT64_CONSTANT(1000000);
- end_nsec = required_end * G_GINT64_CONSTANT(1000000);
+ player->ignore_asyncdone = FALSE;
- if ( (!__gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
- 1.0,
- GST_FORMAT_TIME,
- ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- GST_SEEK_TYPE_SET, start_nsec,
- GST_SEEK_TYPE_SET, end_nsec)))
+ if ( FALSE == async )
{
- debug_error("failed to seek for pcm extraction\n");
+ if ( ret != MM_ERROR_NONE )
+ {
+ GstMessage *msg = NULL;
+ GTimer *timer = NULL;
+ gdouble MAX_TIMEOUT_SEC = 3;
- return MM_ERROR_PLAYER_SEEK;
- }
+ debug_error("failed to set state to PAUSED");
- debug_log("succeeded to set up segment extraction from [%llu] to [%llu] (nsec)\n", start_nsec, end_nsec);
+ timer = g_timer_new();
+ g_timer_start(timer);
- debug_fleave();
+ GstBus *bus = gst_pipeline_get_bus (GST_PIPELINE(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst));
+ gboolean got_msg = FALSE;
+ /* check if gst error posted or not */
+ do
+ {
+ msg = gst_bus_timed_pop(bus, GST_SECOND /2);
+ if (msg)
+ {
+ if (GST_MESSAGE_TYPE(msg) == GST_MESSAGE_ERROR)
+ {
+ GError *error = NULL;
- return MM_ERROR_NONE;
-}
+ /* parse error code */
+ gst_message_parse_error(msg, &error, NULL);
-static
-void __mmplayer_set_videosink_type(mm_player_t* player)
-{
- int type = 0;
+ if ( gst_structure_has_name ( msg->structure, "streaming_error" ) )
+ {
+ /* Note : the streaming error from the streaming source is handled
+ * using __mmplayer_handle_streaming_error.
+ */
+ __mmplayer_handle_streaming_error ( player, msg );
- debug_fenter();
+ }
+ else if (error)
+ {
+ debug_error("paring error posted from bus, domain : %s, code : %d", g_quark_to_string(error->domain), error->code);
- return_if_fail( player );
+ if (error->domain == GST_STREAM_ERROR)
+ {
+ ret = __gst_handle_stream_error( player, error, msg );
+ }
+ else if (error->domain == GST_RESOURCE_ERROR)
+ {
+ ret = __gst_handle_resource_error( player, error->code );
+ }
+ else if (error->domain == GST_LIBRARY_ERROR)
+ {
+ ret = __gst_handle_library_error( player, error->code );
+ }
+ else if (error->domain == GST_CORE_ERROR)
+ {
+ ret = __gst_handle_core_error( player, error->code );
+ }
+ }
- mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &type);
-
- debug_log("check display surface attribute: %d\n", type);
+ got_msg = TRUE;
+ player->msg_posted = TRUE;
+ }
+ gst_message_unref(msg);
+ }
+ } while (!got_msg && (g_timer_elapsed(timer, NULL) < MAX_TIMEOUT_SEC));
+ /* clean */
+ gst_object_unref(bus);
+ g_timer_stop (timer);
+ g_timer_destroy (timer);
- if (type >=MM_DISPLAY_SURFACE_X)
- {
- PLAYER_INI()->video_surface = type;
+ return ret;
+ }
+ else if ( (!player->pipeline->videobin) && (!player->pipeline->audiobin) )
+ {
+ if (MMPLAYER_IS_RTSP_STREAMING(player))
+ return ret;
+ return MM_ERROR_PLAYER_CODEC_NOT_FOUND;
+ }
+ else if ( ret== MM_ERROR_NONE)
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PAUSED );
+ }
}
- debug_fleave();
+ /* generate dot file before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-pause" );
+
+ MMPLAYER_FLEAVE();
- return;
+ return ret;
}
-int
-_mmplayer_deactivate_section_repeat(MMHandleType hplayer)
+int __gst_resume(mm_player_t* player, gboolean async) // @
{
- mm_player_t* player = (mm_player_t*)hplayer;
- gint64 cur_pos = 0;
- GstFormat fmt = GST_FORMAT_TIME;
- gint onetime = 1;
+ int ret = MM_ERROR_NONE;
+ gint timeout = 0;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(player && player->pipeline,
+ MM_ERROR_PLAYER_NOT_INITIALIZED);
- player->section_repeat = FALSE;
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PLAYING;
+ MMPLAYER_PRINT_STATE(player);
- __mmplayer_set_play_count( player, onetime );
+ /* generate dot file before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-resume" );
- gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &cur_pos);
+ __mmplayer_set_antishock( player , FALSE );
- if ( (!__gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
- 1.0,
- GST_FORMAT_TIME,
- ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- GST_SEEK_TYPE_SET, cur_pos,
- GST_SEEK_TYPE_SET, player->duration )))
- {
- debug_error("failed to deactivate section repeat\n");
+ if ( async )
+ debug_log("do async state transition to PLAYING.\n");
- return MM_ERROR_PLAYER_SEEK;
+ /* set pipeline state to PLAYING */
+ timeout = MMPLAYER_STATE_CHANGE_TIMEOUT(player);
+
+ ret = __mmplayer_gst_set_state(player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PLAYING, async, timeout );
+ if (ret != MM_ERROR_NONE)
+ {
+ debug_error("failed to set state to PLAYING\n");
+ return ret;
+ }
+ else
+ {
+ if (async == FALSE)
+ {
+ // MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PLAYING );
+ debug_log("update state machine to %d\n", MM_PLAYER_STATE_PLAYING);
+ ret = __mmplayer_set_state(player, MM_PLAYER_STATE_PLAYING);
+ }
}
- debug_fenter();
+ /* generate dot file before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-resume" );
+
+ MMPLAYER_FLEAVE();
- return MM_ERROR_NONE;
+ return ret;
}
-int
-_mmplayer_set_playspeed(MMHandleType hplayer, gdouble rate)
+static int
+__gst_set_position(mm_player_t* player, int format, unsigned long position, gboolean internal_called) // @
{
- mm_player_t* player = (mm_player_t*)hplayer;
- signed long long pos_msec = 0;
- int ret = MM_ERROR_NONE;
- int mute = FALSE;
- GstFormat format =GST_FORMAT_TIME;
- MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
- debug_fenter();
+ GstFormat fmt = GST_FORMAT_TIME;
+ unsigned long dur_msec = 0;
+ gint64 dur_nsec = 0;
+ gint64 pos_nsec = 0;
+ gboolean ret = TRUE;
+ gboolean accurated = FALSE;
+ GstSeekFlags seek_flags = GST_SEEK_FLAG_FLUSH;
+ MMPLAYER_FENTER();
return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( !MMPLAYER_IS_STREAMING(player), MM_ERROR_NOT_SUPPORT_API );
+ return_val_if_fail ( !MMPLAYER_IS_LIVE_STREAMING(player), MM_ERROR_PLAYER_NO_OP );
- /* The sound of video is not supported under 0.0 and over 2.0. */
- if(rate >= TRICK_PLAY_MUTE_THRESHOLD_MAX || rate < TRICK_PLAY_MUTE_THRESHOLD_MIN)
+ if ( MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PLAYING
+ && MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PAUSED )
+ goto PENDING;
+
+ /* check duration */
+ /* NOTE : duration cannot be zero except live streaming.
+ * Since some element could have some timing problemn with quering duration, try again.
+ */
+ if ( !player->duration )
{
- if (player->can_support_codec & FOUND_PLUGIN_VIDEO)
- mute = TRUE;
+ if ( !gst_element_query_duration( player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &dur_nsec ))
+ {
+ goto SEEK_ERROR;
+ }
+ player->duration = dur_nsec;
}
- _mmplayer_set_mute(hplayer, mute);
-
- if (player->playback_rate == rate)
- return MM_ERROR_NONE;
-
- /* If the position is reached at start potion during fast backward, EOS is posted.
- * So, This EOS have to be classified with it which is posted at reaching the end of stream.
- * */
- player->playback_rate = rate;
-
- current_state = MMPLAYER_CURRENT_STATE(player);
- if ( current_state != MM_PLAYER_STATE_PAUSED )
- ret = gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &format, &pos_msec);
+ if ( player->duration )
+ {
+ dur_msec = GST_TIME_AS_MSECONDS(player->duration);
+ }
+ else
+ {
+ debug_error("could not get the duration. fail to seek.\n");
+ goto SEEK_ERROR;
+ }
- debug_log ("pos_msec = %"GST_TIME_FORMAT" and ret = %d and state = %d", GST_TIME_ARGS (pos_msec), ret, current_state);
+ debug_log("playback rate: %f\n", player->playback_rate);
- if ( ( current_state == MM_PLAYER_STATE_PAUSED )
- || ( ! ret ))
- //|| ( player->last_position != 0 && pos_msec == 0 ) )
+ mm_attrs_get_int_by_name(player->attrs,"accurate_seek", &accurated);
+ if (accurated)
{
- debug_warning("returning last point : %lld\n", player->last_position );
- pos_msec = player->last_position;
+ seek_flags |= GST_SEEK_FLAG_ACCURATE;
}
-
- if ((!gst_element_seek (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
- rate,
- GST_FORMAT_TIME,
- ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
- //( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT),
- GST_SEEK_TYPE_SET, pos_msec,
- //GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE,
- GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)))
+ else
{
- debug_error("failed to set speed playback\n");
- return MM_ERROR_PLAYER_SEEK;
+ seek_flags |= GST_SEEK_FLAG_KEY_UNIT;
}
- debug_log("succeeded to set speed playback as %fl\n", rate);
+ /* do seek */
+ switch ( format )
+ {
+ case MM_PLAYER_POS_FORMAT_TIME:
+ {
+ /* check position is valid or not */
+ if ( position > dur_msec )
+ goto INVALID_ARGS;
- debug_fleave();
+ debug_log("seeking to (%lu) msec, duration is %d msec\n", position, dur_msec);
- return MM_ERROR_NONE;;
-}
+ if (player->doing_seek)
+ {
+ debug_log("not completed seek");
+ return MM_ERROR_PLAYER_DOING_SEEK;
+ }
-int
-_mmplayer_set_position(MMHandleType hplayer, int format, int position) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- int ret = MM_ERROR_NONE;
+ if ( !internal_called)
+ player->doing_seek = TRUE;
- debug_fenter();
+ pos_nsec = position * G_GINT64_CONSTANT(1000000);
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ if ((MMPLAYER_IS_HTTP_STREAMING(player)) && (!player->videodec_linked))
+ {
+ gint64 cur_time = 0;
+ GstFormat format = GST_FORMAT_TIME;
+
+ /* get current position */
+ gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &format, &cur_time);
+
+ /* flush */
+ GstEvent *event = gst_event_new_seek (1.0,
+ GST_FORMAT_TIME,
+ (GstSeekFlags)GST_SEEK_FLAG_FLUSH,
+ GST_SEEK_TYPE_SET, cur_time,
+ GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE);
+ if(event) {
+ __gst_send_event_to_sink(player, event);
+ }
- ret = __gst_set_position ( player, format, (unsigned long)position );
+ __gst_pause( player, FALSE );
+ }
- debug_fleave();
+ ret = __gst_seek ( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, player->playback_rate,
+ GST_FORMAT_TIME, seek_flags,
+ GST_SEEK_TYPE_SET, pos_nsec, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE );
+ if ( !ret )
+ {
+ debug_error("failed to set position. dur[%lu] pos[%lu] pos_msec[%llu]\n", dur_msec, position, pos_nsec);
+ goto SEEK_ERROR;
+ }
+ }
+ break;
- return ret;
-}
+ case MM_PLAYER_POS_FORMAT_PERCENT:
+ {
+ debug_log("seeking to (%lu)%% \n", position);
-int
-_mmplayer_get_position(MMHandleType hplayer, int format, unsigned long *position) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- int ret = MM_ERROR_NONE;
+ if (player->doing_seek)
+ {
+ debug_log("not completed seek");
+ return MM_ERROR_PLAYER_DOING_SEEK;
+ }
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ if ( !internal_called)
+ player->doing_seek = TRUE;
- ret = __gst_get_position ( player, format, position );
-
- return ret;
-}
+ /* FIXIT : why don't we use 'GST_FORMAT_PERCENT' */
+ pos_nsec = (gint64) ( ( position * player->duration ) / 100 );
+ ret = __gst_seek ( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, player->playback_rate,
+ GST_FORMAT_TIME, seek_flags,
+ GST_SEEK_TYPE_SET, pos_nsec, GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE );
+ if ( !ret )
+ {
+ debug_error("failed to set position. dur[%lud] pos[%lud] pos_msec[%llud]\n", dur_msec, position, pos_nsec);
+ goto SEEK_ERROR;
+ }
+ }
+ break;
-int
-_mmplayer_get_buffer_position(MMHandleType hplayer, int format, unsigned long* start_pos, unsigned long* stop_pos) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- int ret = MM_ERROR_NONE;
+ default:
+ goto INVALID_ARGS;
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ }
- ret = __gst_get_buffer_position ( player, format, start_pos, stop_pos );
+ /* NOTE : store last seeking point to overcome some bad operation
+ * ( returning zero when getting current position ) of some elements
+ */
+ player->last_position = pos_nsec;
- return ret;
-}
+ /* MSL should guarante playback rate when seek is selected during trick play of fast forward. */
+ if ( player->playback_rate > 1.0 )
+ _mmplayer_set_playspeed ( (MMHandleType)player, player->playback_rate );
-int
-_mmplayer_adjust_subtitle_postion(MMHandleType hplayer, int format, int position) // @
-{
- mm_player_t* player = (mm_player_t*)hplayer;
- int ret = MM_ERROR_NONE;
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
- debug_fenter();
+PENDING:
+ player->pending_seek.is_pending = TRUE;
+ player->pending_seek.format = format;
+ player->pending_seek.pos = position;
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ debug_warning("player current-state : %s, pending-state : %s, just preserve pending position(%lu).\n",
+ MMPLAYER_STATE_GET_NAME(MMPLAYER_CURRENT_STATE(player)), MMPLAYER_STATE_GET_NAME(MMPLAYER_PENDING_STATE(player)), player->pending_seek.pos);
- ret = __gst_adjust_subtitle_position(player, format, position);
+ return MM_ERROR_NONE;
- debug_fleave();
+INVALID_ARGS:
+ debug_error("invalid arguments, position : %ld dur : %ld format : %d \n", position, dur_msec, format);
+ return MM_ERROR_INVALID_ARGUMENT;
- return ret;
+SEEK_ERROR:
+ player->doing_seek = FALSE;
+ return MM_ERROR_PLAYER_SEEK;
}
-static gboolean
-__mmplayer_is_midi_type( gchar* str_caps)
-{
- if ( ( g_strrstr(str_caps, "audio/midi") ) ||
- ( g_strrstr(str_caps, "application/x-gst_ff-mmf") ) ||
- ( g_strrstr(str_caps, "application/x-smaf") ) ||
- ( g_strrstr(str_caps, "audio/x-imelody") ) ||
- ( g_strrstr(str_caps, "audio/mobile-xmf") ) ||
- ( g_strrstr(str_caps, "audio/xmf") ) ||
- ( g_strrstr(str_caps, "audio/mxmf") ) )
- {
- debug_log("midi\n");
+#define TRICKPLAY_OFFSET GST_MSECOND
- return TRUE;
- }
+static int
+__gst_get_position(mm_player_t* player, int format, unsigned long* position) // @
+{
+ MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
+ GstFormat fmt = GST_FORMAT_TIME;
+ signed long long pos_msec = 0;
+ gboolean ret = TRUE;
- debug_log("not midi.\n");
+ return_val_if_fail( player && position && player->pipeline && player->pipeline->mainbin,
+ MM_ERROR_PLAYER_NOT_INITIALIZED );
- return FALSE;
-}
+ current_state = MMPLAYER_CURRENT_STATE(player);
-static gboolean
-__mmplayer_is_amr_type (gchar *str_caps)
-{
- if ((g_strrstr(str_caps, "AMR")) ||
- (g_strrstr(str_caps, "amr")))
+ /* NOTE : query position except paused state to overcome some bad operation
+ * please refer to below comments in details
+ */
+ if ( current_state != MM_PLAYER_STATE_PAUSED )
{
- return TRUE;
+ ret = gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &pos_msec);
}
- return FALSE;
-}
-static gboolean
-__mmplayer_is_only_mp3_type (gchar *str_caps)
-{
- if (g_strrstr(str_caps, "application/x-id3") ||
- (g_strrstr(str_caps, "audio/mpeg") && g_strrstr(str_caps, "mpegversion=(int)1")))
+ /* NOTE : get last point to overcome some bad operation of some elements
+ * ( returning zero when getting current position in paused state
+ * and when failed to get postion during seeking
+ */
+ if ( ( current_state == MM_PLAYER_STATE_PAUSED )
+ || ( ! ret ))
+ //|| ( player->last_position != 0 && pos_msec == 0 ) )
{
- return TRUE;
- }
- return FALSE;
-}
-
-static void
-__mmplayer_typefind_have_type( GstElement *tf, guint probability, // @
-GstCaps *caps, gpointer data)
-{
- mm_player_t* player = (mm_player_t*)data;
- GstPad* pad = NULL;
+ debug_log ("pos_msec = %"GST_TIME_FORMAT" and ret = %d and state = %d", GST_TIME_ARGS (pos_msec), ret, current_state);
- debug_fenter();
+ if(player->playback_rate < 0.0)
+ pos_msec = player->last_position - TRICKPLAY_OFFSET;
+ else
+ pos_msec = player->last_position;
- return_if_fail( player && tf && caps );
+ if (!ret)
+ pos_msec = player->last_position;
+ else
+ player->last_position = pos_msec;
- /* store type string */
- MMPLAYER_FREEIF(player->type);
- player->type = gst_caps_to_string(caps);
- if (player->type)
- debug_log("meida type %s found, probability %d%% / %d\n", player->type, probability, gst_caps_get_size(caps));
+ debug_log("returning last point : %"GST_TIME_FORMAT, GST_TIME_ARGS(pos_msec));
- /* midi type should be stored because it will be used to set audio gain in avsysauiosink */
- if ( __mmplayer_is_midi_type(player->type))
- {
- player->profile.play_mode = MM_PLAYER_MODE_MIDI;
}
- else if (__mmplayer_is_amr_type(player->type))
+ else
{
- player->bypass_sound_effect = FALSE;
- if ( (PLAYER_INI()->use_audio_filter_preset || PLAYER_INI()->use_audio_filter_custom) )
+ if (player->duration > 0 && pos_msec > player->duration) {
+ pos_msec = player->duration;
+ }
+
+ if (player->sm.keep_last_pos) {
+ debug_log("return last pos as stop by asm, %"GST_TIME_FORMAT, GST_TIME_ARGS(player->last_position));
+ pos_msec = player->last_position;
+ }
+ else {
+ player->last_position = pos_msec;
+ }
+ }
+
+ switch (format) {
+ case MM_PLAYER_POS_FORMAT_TIME:
+ *position = GST_TIME_AS_MSECONDS(pos_msec);
+ break;
+
+ case MM_PLAYER_POS_FORMAT_PERCENT:
{
- if ( player->audio_filter_info.filter_type == MM_AUDIO_FILTER_TYPE_PRESET )
+ gint64 dur = 0;
+ gint64 pos = 0;
+
+ dur = player->duration / GST_SECOND;
+ if (dur <= 0)
{
- if (!_mmplayer_sound_filter_preset_apply(player, player->audio_filter_info.preset))
- {
- debug_msg("apply sound effect(preset:%d) setting success\n",player->audio_filter_info.preset);
- }
+ debug_log ("duration is [%d], so returning position 0\n",dur);
+ *position = 0;
}
- else if ( player->audio_filter_info.filter_type == MM_AUDIO_FILTER_TYPE_CUSTOM )
+ else
{
- if (!_mmplayer_sound_filter_custom_apply(player))
- {
- debug_msg("apply sound effect(custom) setting success\n");
- }
- }
- }
- }
- else if ( g_strrstr(player->type, "application/x-hls"))
- {
- /* If it can't know exact type when it parses uri because of redirection case,
- * it will be fixed by typefinder here.
- */
- player->profile.uri_type = MM_PLAYER_URI_TYPE_HLS;
- }
-
- pad = gst_element_get_static_pad(tf, "src");
- if ( !pad )
- {
- debug_error("fail to get typefind src pad.\n");
- return;
- }
-
-
- /* try to plug */
- if ( ! __mmplayer_try_to_plug( player, pad, caps ) )
- {
- debug_error("failed to autoplug for type : %s\n", player->type);
-
- if ( ( PLAYER_INI()->async_start ) &&
- ( ! MMPLAYER_IS_RTSP_STREAMING ( player ) ) &&
- ( player->posted_msg == FALSE ) )
- {
- __mmplayer_post_missed_plugin( player );
- }
-
- goto DONE;
- }
-
- /* finish autopluging if no dynamic pad waiting */
- if( ( ! player->have_dynamic_pad) && ( ! player->has_many_types) )
- {
- if ( ! MMPLAYER_IS_RTSP_STREAMING( player ) )
- {
- __mmplayer_pipeline_complete( NULL, (gpointer)player );
+ pos = pos_msec / GST_SECOND;
+ *position = pos * 100 / dur;
+ }
+ break;
}
+ default:
+ return MM_ERROR_PLAYER_INTERNAL;
}
-DONE:
- gst_object_unref( GST_OBJECT(pad) );
-
- debug_fleave();
-
- return;
+ return MM_ERROR_NONE;
}
-static gboolean
-__mmplayer_warm_up_video_codec( mm_player_t* player, GstElementFactory *factory)
-{
- GstElement *element;
- GstStateChangeReturn ret;
- gboolean usable = TRUE;
-
- return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( factory, MM_ERROR_COMMON_INVALID_ARGUMENT );
-
- element = gst_element_factory_create (factory, NULL);
-
- ret = gst_element_set_state (element, GST_STATE_READY);
-
- if (ret != GST_STATE_CHANGE_SUCCESS)
- {
- debug_error ("resource conflict so, %s unusable\n", GST_PLUGIN_FEATURE_NAME (factory));
- usable = FALSE;
- }
-
- gst_element_set_state (element, GST_STATE_NULL);
- gst_object_unref (element);
-
- return usable;
-}
-/* it will return first created element */
-static gboolean
-__mmplayer_try_to_plug(mm_player_t* player, GstPad *pad, const GstCaps *caps) // @
+static int __gst_get_buffer_position(mm_player_t* player, int format, unsigned long* start_pos, unsigned long* stop_pos)
{
- MMPlayerGstElement* mainbin = NULL;
- const char* mime = NULL;
- const GList* item = NULL;
- const gchar* klass = NULL;
- GstCaps* res = NULL;
- gboolean skip = FALSE;
- GstPad* queue_pad = NULL;
- GstElement* queue = NULL;
- GstElement *element = NULL;
+#define STREAMING_IS_FINISHED 0
+#define BUFFERING_MAX_PER 100
- debug_fenter();
+ GstQuery *query = NULL;
return_val_if_fail( player &&
player->pipeline &&
player->pipeline->mainbin,
- FALSE );
+ MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( start_pos && stop_pos, MM_ERROR_INVALID_ARGUMENT );
- mainbin = player->pipeline->mainbin;
+ if (!MMPLAYER_IS_HTTP_STREAMING ( player ))
+ {
+ /* and rtsp is not ready yet. */
+ debug_warning ( "it's only used for http streaming case.\n" );
+ return MM_ERROR_NONE;
+ }
- mime = gst_structure_get_name(gst_caps_get_structure(caps, 0));
+ *start_pos = 0;
+ *stop_pos = 0;
- /* return if we got raw output */
- if(g_str_has_prefix(mime, "video/x-raw") || g_str_has_prefix(mime, "audio/x-raw") ||g_str_has_prefix(mime, "text/plain") )
- {
+ switch ( format )
+ {
+ case MM_PLAYER_POS_FORMAT_PERCENT :
+ {
+ gint start_per = -1, stop_per = -1;
+ gint64 buffered_total = 0;
- element = (GstElement*)gst_pad_get_parent(pad);
+ unsigned long position = 0;
+ guint curr_size_bytes = 0;
+ gint64 buffering_left = -1;
+ gint buffered_sec = -1;
+ gint64 content_duration = player->duration;
+ guint64 content_size = player->http_content_size;
-/* NOTE : When no decoder has added during autoplugging. like a simple wave playback.
- * No queue will be added. I think it can caused breaking sound when playing raw audio
- * frames but there's no different. Decodebin also doesn't add with those wav fils.
- * Anyway, currentely raw-queue seems not necessary.
- */
-#if 1
+ if (content_duration > 0)
+ {
+ if (!__gst_get_position(player, MM_PLAYER_POS_FORMAT_TIME, &position))
+ {
+ debug_log ("[Time] pos %d ms / dur %d sec / %lld bytes", position, (guint)(content_duration/GST_SECOND), content_size);
+ start_per = 100 * (position*GST_MSECOND) / content_duration;
- /* NOTE : check if previously linked element is demuxer/depayloader/parse means no decoder
- * has linked. if so, we need to add queue for quality of output. note that
- * decodebin also has same problem.
- */
+ /* buffered size info from multiqueue */
+ if (player->pipeline->mainbin[MMPLAYER_M_DEMUXED_S_BUFFER].gst)
+ {
+ g_object_get(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_DEMUXED_S_BUFFER].gst), "curr-size-bytes", &curr_size_bytes, NULL);
+ debug_log ("[MQ] curr_size_bytes = %d", curr_size_bytes);
- klass = gst_element_factory_get_klass( gst_element_get_factory(element) );
+ buffered_total += curr_size_bytes;
+ }
- /* add queue if needed */
- if( g_strrstr(klass, "Demux") ||
- g_strrstr(klass, "Depayloader") ||
- g_strrstr(klass, "Parse") )
- {
- debug_log("adding raw queue\n");
+ /* buffered size info from queue2 */
+ if (player->pipeline->mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst)
+ {
+ query = gst_query_new_buffering ( GST_FORMAT_BYTES );
+ if (gst_element_query(player->pipeline->mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst, query))
+ {
+ GstBufferingMode mode;
+ gint byte_in_rate = 0, byte_out_rate = 0;
+ gint64 start_byte = 0, stop_byte = 0;
+ guint num_of_ranges = 0;
+ guint idx = 0;
+
+ num_of_ranges = gst_query_get_n_buffering_ranges(query);
+ for ( idx=0 ; idx<num_of_ranges ; idx++ )
+ {
+ gst_query_parse_nth_buffering_range (query, idx, &start_byte, &stop_byte);
+ debug_log ("[Q2][range %d] %lld ~ %lld\n", idx, start_byte, stop_byte);
- queue = gst_element_factory_make("queue", NULL);
- if ( ! queue )
- {
- debug_warning("failed to create queue\n");
- goto ERROR;
- }
+ buffered_total += (stop_byte - start_byte);
+ }
- /* warmup */
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(queue, GST_STATE_READY) )
- {
- debug_warning("failed to set state READY to queue\n");
- goto ERROR;
+ gst_query_parse_buffering_stats(query, &mode, &byte_in_rate, &byte_out_rate, &buffering_left);
+ debug_log ("[Q2] in_rate %d, out_rate %d, left %lld\n", byte_in_rate, byte_out_rate, buffering_left);
+ }
+ gst_query_unref (query);
+ }
+
+ if (buffering_left == STREAMING_IS_FINISHED)
+ {
+ stop_per = BUFFERING_MAX_PER;
+ }
+ else
+ {
+ guint dur_sec = (guint)(content_duration/GST_SECOND);
+ guint avg_byterate = (dur_sec>0)?((guint)(content_size/dur_sec)):(0);
+
+ if (avg_byterate > 0)
+ buffered_sec = (gint)(buffered_total/avg_byterate);
+ else if (player->total_maximum_bitrate > 0)
+ buffered_sec = (gint)(GET_BIT_FROM_BYTE(buffered_total)/(gint64)player->total_maximum_bitrate);
+ else if (player->total_bitrate > 0)
+ buffered_sec = (gint)(GET_BIT_FROM_BYTE(buffered_total)/(gint64)player->total_bitrate);
+
+ if ((buffered_sec >= 0) && (dur_sec > 0))
+ stop_per = start_per + (100 * buffered_sec / dur_sec);
+ }
+
+ debug_log ("[Buffered Total] %lld bytes, %d sec, per %d~%d\n", buffered_total, buffered_sec, start_per, stop_per);
+ }
}
- /* add to pipeline */
- if ( ! gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue) )
+ if (((buffered_total == 0) || (start_per < 0) || (stop_per < 0)) &&
+ (player->pipeline->mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst))
{
- debug_warning("failed to add queue\n");
- goto ERROR;
- }
+ query = gst_query_new_buffering ( GST_FORMAT_PERCENT );
+ if ( gst_element_query ( player->pipeline->mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst, query ) )
+ {
+ GstFormat format;
+ gint64 range_start_per = -1, range_stop_per = -1;
- /* link queue */
- queue_pad = gst_element_get_static_pad(queue, "sink");
+ gst_query_parse_buffering_range ( query, &format, &range_start_per, &range_stop_per, NULL );
- if ( GST_PAD_LINK_OK != gst_pad_link(pad, queue_pad) )
- {
- debug_warning("failed to link queue\n");
- goto ERROR;
- }
- gst_object_unref ( GST_OBJECT(queue_pad) );
- queue_pad = NULL;
+ debug_log ("[Q2] range start %" G_GINT64_FORMAT " ~ stop %" G_GINT64_FORMAT "\n", range_start_per , range_stop_per);
- /* running */
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(queue, GST_STATE_PAUSED) )
- {
- debug_warning("failed to set state READY to queue\n");
- goto ERROR;
- }
+ if (range_start_per != -1)
+ start_per = (gint)(100 * range_start_per / GST_FORMAT_PERCENT_MAX);
- /* replace given pad to queue:src */
- pad = gst_element_get_static_pad(queue, "src");
- if ( ! pad )
- {
- debug_warning("failed to get pad from queue\n");
- goto ERROR;
+ if (range_stop_per != -1)
+ stop_per = (gint)(100 * range_stop_per / GST_FORMAT_PERCENT_MAX);
+ }
+ gst_query_unref (query);
}
- }
-#endif
- /* check if player can do start continually */
- MMPLAYER_CHECK_CMD_IF_EXIT(player);
- if(__mmplayer_link_sink(player,pad))
- __mmplayer_gst_decode_callback(element, pad, FALSE, player);
-
- gst_object_unref( GST_OBJECT(element));
- element = NULL;
+ if ( start_per > 0)
+ *start_pos = (start_per < 100)?(start_per):(100);
+ else
+ *start_pos = 0;
+ if ( stop_per > 0)
+ *stop_pos = (stop_per < 100)?(stop_per):(100);
+ else
+ *stop_pos = 0;
- return TRUE;
- }
+ break;
+ }
+ case MM_PLAYER_POS_FORMAT_TIME :
+ debug_warning ( "Time format is not supported yet.\n" );
+ break;
- item = player->factories;
- for(; item != NULL ; item = item->next)
- {
+ default :
+ break;
+ }
- GstElementFactory *factory = GST_ELEMENT_FACTORY(item->data);
- const GList *pads;
- gint idx = 0;
+ debug_log("current buffer position : %lu~%lu \n", *start_pos, *stop_pos );
- skip = FALSE;
+ return MM_ERROR_NONE;
+}
- /* filtering exclude keyword */
- for ( idx = 0; PLAYER_INI()->exclude_element_keyword[idx][0] != '\0'; idx++ )
- {
- if ( g_strrstr(GST_PLUGIN_FEATURE_NAME (factory),
- PLAYER_INI()->exclude_element_keyword[idx] ) )
- {
- debug_warning("skipping [%s] by exculde keyword [%s]\n",
- GST_PLUGIN_FEATURE_NAME (factory),
- PLAYER_INI()->exclude_element_keyword[idx] );
+static int
+__gst_set_message_callback(mm_player_t* player, MMMessageCallback callback, gpointer user_param) // @
+{
+ MMPLAYER_FENTER();
- skip = TRUE;
- break;
- }
- }
+ if ( !player )
+ {
+ debug_warning("set_message_callback is called with invalid player handle\n");
+ return MM_ERROR_PLAYER_NOT_INITIALIZED;
+ }
- if ( skip ) continue;
+ player->msg_cb = callback;
+ player->msg_cb_param = user_param;
+ debug_log("msg_cb : 0x%x msg_cb_param : 0x%x\n", (guint)callback, (guint)user_param);
- /* check factory class for filtering */
- klass = gst_element_factory_get_klass(GST_ELEMENT_FACTORY(factory));
+ MMPLAYER_FLEAVE();
- if ( MMPLAYER_IS_RTSP_STREAMING( player ) )
- {
- if ( g_strrstr(klass, "Parse") )
- {
- debug_log("streaming doesn't need any parser. skipping [%s]\n",
- GST_PLUGIN_FEATURE_NAME (factory) );
+ return MM_ERROR_NONE;
+}
- continue;
- }
- }
+static int __mmfplayer_parse_profile(const char *uri, void *param, MMPlayerParseProfile* data) // @
+{
+ int ret = MM_ERROR_PLAYER_INVALID_URI;
+ char *path = NULL;
- /* NOTE : msl don't need to use image plugins.
- * So, those plugins should be skipped for error handling.
- */
- if ( g_strrstr(klass, "Codec/Decoder/Image") )
- {
- debug_log("player doesn't need [%s] so, skipping it\n",
- GST_PLUGIN_FEATURE_NAME (factory) );
+ MMPLAYER_FENTER();
- continue;
- }
+ return_val_if_fail ( uri , FALSE);
+ return_val_if_fail ( data , FALSE);
+ return_val_if_fail ( ( strlen(uri) <= MM_MAX_URL_LEN ), FALSE );
+ memset(data, 0, sizeof(MMPlayerParseProfile));
- /* check pad compatability */
- for(pads = gst_element_factory_get_static_pad_templates(factory);
- pads != NULL; pads=pads->next)
- {
- GstStaticPadTemplate *temp1 = pads->data;
- GstCaps* static_caps = NULL;
+ if ((path = strstr(uri, "file://")))
+ {
+ int file_stat = MM_ERROR_NONE;
- if( temp1->direction != GST_PAD_SINK ||
- temp1->presence != GST_PAD_ALWAYS)
- continue;
+ file_stat = util_exist_file_path(path + 7);
+ if (file_stat == MM_ERROR_NONE)
+ {
+ strncpy(data->uri, path, MM_MAX_URL_LEN-1);
- if ( GST_IS_CAPS( &temp1->static_caps.caps) )
+ if ( util_is_sdp_file ( path ) )
{
- /* using existing caps */
- static_caps = gst_caps_ref( &temp1->static_caps.caps );
+ debug_log("uri is actually a file but it's sdp file. giving it to rtspsrc\n");
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
}
else
{
- /* create one */
- static_caps = gst_caps_from_string ( temp1->static_caps.string );
+ data->uri_type = MM_PLAYER_URI_TYPE_FILE;
}
+ ret = MM_ERROR_NONE;
+ }
+ else if (file_stat == MM_ERROR_PLAYER_PERMISSION_DENIED)
+ {
+ data->uri_type = MM_PLAYER_URI_TYPE_NO_PERMISSION;
+ }
+ else
+ {
+ debug_warning("could access %s.\n", path);
+ }
+ }
+ else if ((path = strstr(uri, "buff://")))
+ {
+ data->uri_type = MM_PLAYER_URI_TYPE_BUFF;
+ ret = MM_ERROR_NONE;
+ }
+ else if ((path = strstr(uri, "rtsp://")))
+ {
+ if (strlen(path)) {
+ if((path = strstr(uri, "/wfd1.0/"))) {
+ strncpy(data->uri, uri, MM_MAX_URL_LEN-1);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_WFD;
+ ret = MM_ERROR_NONE;
+ debug_log("uri is actually a wfd client path. giving it to wfdrtspsrc\n");
+ }
+ else {
+ strncpy(data->uri, uri, MM_MAX_URL_LEN-1);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
+ ret = MM_ERROR_NONE;
+ }
+ }
+ }
+ else if ((path = strstr(uri, "http://")))
+ {
+ if (strlen(path)) {
+ strncpy(data->uri, uri, MM_MAX_URL_LEN-1);
+#ifdef MM_SMOOTH_STREAMING
+ if (g_str_has_suffix (g_ascii_strdown(uri, strlen(uri)), ".ism/manifest") ||
+ g_str_has_suffix (g_ascii_strdown(uri, strlen(uri)), ".isml/manifest"))
+ {
+ data->uri_type = MM_PLAYER_URI_TYPE_SS;
+ }
+ else
+#endif
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_HTTP;
- res = gst_caps_intersect(caps, static_caps);
-
- gst_caps_unref( static_caps );
- static_caps = NULL;
+ ret = MM_ERROR_NONE;
+ }
+ }
+ else if ((path = strstr(uri, "https://")))
+ {
+ if (strlen(path)) {
+ strncpy(data->uri, uri, MM_MAX_URL_LEN -1);
+#ifdef MM_SMOOTH_STREAMING
+ if (g_str_has_suffix (g_ascii_strdown(uri, strlen(uri)), ".ism/manifest") ||
+ g_str_has_suffix (g_ascii_strdown(uri, strlen(uri)), ".isml/manifest"))
+ {
+ data->uri_type = MM_PLAYER_URI_TYPE_SS;
+ }
+#endif
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_HTTP;
- if( res && !gst_caps_is_empty(res) )
- {
- GstElement *new_element;
- GList *elements = player->parsers;
- char *name_template = g_strdup(temp1->name_template);
- gchar *name_to_plug = GST_PLUGIN_FEATURE_NAME(factory);
+ ret = MM_ERROR_NONE;
+ }
+ }
+ else if ((path = strstr(uri, "rtspu://")))
+ {
+ if (strlen(path)) {
+ strncpy(data->uri, uri, MM_MAX_URL_LEN -1);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
+ ret = MM_ERROR_NONE;
+ }
+ }
+ else if ((path = strstr(uri, "rtspr://")))
+ {
+ strncpy(data->uri, path, MM_MAX_URL_LEN -1);
+ char *separater =strstr(path, "*");
- gst_caps_unref(res);
+ if (separater) {
+ int urgent_len = 0;
+ char *urgent = separater + strlen("*");
- debug_log("found %s to plug\n", name_to_plug);
+ if ((urgent_len = strlen(urgent))) {
+ data->uri[strlen(path) - urgent_len - strlen("*")] = '\0';
+ strncpy(data->urgent, urgent, MM_MAX_FILENAME_LEN -1);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
+ ret = MM_ERROR_NONE;
+ }
+ }
+ }
+ else if ((path = strstr(uri, "mms://")))
+ {
+ if (strlen(path)) {
+ strncpy(data->uri, uri, MM_MAX_URL_LEN -1);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_MMS;
+ ret = MM_ERROR_NONE;
+ }
+ }
+ else if ((path = strstr(uri, "mem://")))
+ {
+ if (strlen(path)) {
+ #define MM_MAX_EXT_SIZE 100
+ int mem_size = 0;
+ char *buffer = NULL;
+ char *seperator = strchr(path, ',');
+ char ext[MM_MAX_EXT_SIZE] = {0,}, size[MM_MAX_EXT_SIZE] = {0,};
- new_element = gst_element_factory_create(GST_ELEMENT_FACTORY(factory), NULL);
- if ( ! new_element )
- {
- debug_error("failed to create element [%s]. continue with next.\n",
- GST_PLUGIN_FEATURE_NAME (factory));
+ if (seperator) {
+ if ((buffer = strstr(path, "ext="))) {
+ buffer += strlen("ext=");
- MMPLAYER_FREEIF(name_template);
+ if (strlen(buffer)) {
+ strncpy(ext, buffer, MM_MAX_EXT_SIZE -1);
- continue;
+ if ((seperator = strchr(ext, ','))
+ || (seperator = strchr(ext, ' '))
+ || (seperator = strchr(ext, '\0'))) {
+ seperator[0] = '\0';
+ }
+ }
}
- /* check and skip it if it was already used. Otherwise, it can be an infinite loop
- * because parser can accept its own output as input.
- */
- if (g_strrstr(klass, "Parser"))
- {
- gchar *selected = NULL;
+ if ((buffer = strstr(path, "size="))) {
+ buffer += strlen("size=");
- for ( ; elements; elements = g_list_next(elements))
- {
- gchar *element_name = elements->data;
+ if (strlen(buffer) > 0) {
+ strncpy(size, buffer, MM_MAX_EXT_SIZE -1);
- if (g_strrstr(element_name, name_to_plug))
- {
- debug_log("but, %s already linked, so skipping it\n", name_to_plug);
- skip = TRUE;
+ if ((seperator = strchr(size, ','))
+ || (seperator = strchr(size, ' '))
+ || (seperator = strchr(size, '\0'))) {
+ seperator[0] = '\0';
}
- }
-
- if (skip) continue;
- selected = g_strdup(name_to_plug);
-
- player->parsers = g_list_append(player->parsers, selected);
+ mem_size = atoi(size);
+ }
}
+ }
- /* store specific handles for futher control */
- if(g_strrstr(klass, "Demux") || g_strrstr(klass, "Parse"))
- {
- /* FIXIT : first value will be overwritten if there's more
- * than 1 demuxer/parser
- */
- debug_log("plugged element is demuxer. take it\n");
- mainbin[MMPLAYER_M_DEMUX].id = MMPLAYER_M_DEMUX;
- mainbin[MMPLAYER_M_DEMUX].gst = new_element;
- }
- else if(g_strrstr(klass, "Decoder") && __mmplayer_link_decoder(player,pad))
- {
- if(mainbin[MMPLAYER_M_DEC1].gst == NULL)
- {
- debug_log("plugged element is decoder. take it[MMPLAYER_M_DEC1]\n");
- mainbin[MMPLAYER_M_DEC1].id = MMPLAYER_M_DEC1;
- mainbin[MMPLAYER_M_DEC1].gst = new_element;
- }
- else if(mainbin[MMPLAYER_M_DEC2].gst == NULL)
- {
- debug_log("plugged element is decoder. take it[MMPLAYER_M_DEC2]\n");
- mainbin[MMPLAYER_M_DEC2].id = MMPLAYER_M_DEC2;
- mainbin[MMPLAYER_M_DEC2].gst = new_element;
- }
+ debug_log("ext: %s, mem_size: %d, mmap(param): %p\n", ext, mem_size, param);
+ if ( mem_size && param)
+ {
+ data->mem = param;
+ data->mem_size = mem_size;
+ data->uri_type = MM_PLAYER_URI_TYPE_MEM;
+ ret = MM_ERROR_NONE;
+ }
+ }
+ }
+ else
+ {
+ int file_stat = MM_ERROR_NONE;
- /* NOTE : IF one codec is found, add it to supported_codec and remove from
- * missing plugin. Both of them are used to check what's supported codec
- * before returning result of play start. And, missing plugin should be
- * updated here for multi track files.
- */
- if(g_str_has_prefix(mime, "video"))
- {
- GstPad *src_pad = NULL;
- GstPadTemplate *pad_templ = NULL;
- GstCaps *caps = NULL;
- gchar *caps_type = NULL;
-
- debug_log("found VIDEO decoder\n");
- player->not_supported_codec &= MISSING_PLUGIN_AUDIO;
- player->can_support_codec |= FOUND_PLUGIN_VIDEO;
+ file_stat = util_exist_file_path(uri);
- src_pad = gst_element_get_static_pad (new_element, "src");
- pad_templ = gst_pad_get_pad_template (src_pad);
- caps = GST_PAD_TEMPLATE_CAPS(pad_templ);
+ /* if no protocol prefix exist. check file existence and then give file:// as it's prefix */
+ if (file_stat == MM_ERROR_NONE)
+ {
+ g_snprintf(data->uri, MM_MAX_URL_LEN, "file://%s", uri);
- caps_type = gst_caps_to_string(caps);
+ if ( util_is_sdp_file( (char*)uri ) )
+ {
+ debug_log("uri is actually a file but it's sdp file. giving it to rtspsrc\n");
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
+ }
+ else
+ {
+ data->uri_type = MM_PLAYER_URI_TYPE_FILE;
+ }
+ ret = MM_ERROR_NONE;
+ }
+ else if (file_stat == MM_ERROR_PLAYER_PERMISSION_DENIED)
+ {
+ data->uri_type = MM_PLAYER_URI_TYPE_NO_PERMISSION;
+ }
+ else
+ {
+ debug_error ("invalid uri, could not play..\n");
+ data->uri_type = MM_PLAYER_URI_TYPE_NONE;
+ }
+ }
- if ( g_strrstr( caps_type, "ST12") )
- player->is_nv12_tiled = TRUE;
+ if (data->uri_type == MM_PLAYER_URI_TYPE_NONE) {
+ ret = MM_ERROR_PLAYER_FILE_NOT_FOUND;
+ } else if (data->uri_type == MM_PLAYER_URI_TYPE_NO_PERMISSION){
+ ret = MM_ERROR_PLAYER_PERMISSION_DENIED;
+ }
- /* clean */
- MMPLAYER_FREEIF( caps_type );
- gst_object_unref (src_pad);
+ /* dump parse result */
+ secure_debug_warning("incomming uri : %s\n", uri);
+ debug_log("uri_type : %d, play_mode : %d, mem : 0x%x, mem_size : %d, urgent : %s\n",
+ data->uri_type, data->play_mode, (guint)data->mem, data->mem_size, data->urgent);
- g_object_set( G_OBJECT (new_element), "hw-accel", PLAYER_INI()->use_video_hw_accel, NULL);
- g_object_set( G_OBJECT (new_element), "err-conceal", TRUE, NULL);
- }
- else if (g_str_has_prefix(mime, "audio"))
- {
- debug_log("found AUDIO decoder\n");
- player->not_supported_codec &= MISSING_PLUGIN_VIDEO;
- player->can_support_codec |= FOUND_PLUGIN_AUDIO;
- }
- }
- if ( ! __mmplayer_close_link(player, pad, new_element,
- name_template,gst_element_factory_get_static_pad_templates(factory)) )
- {
- if (player->keep_detecting_vcodec)
- continue;
+ MMPLAYER_FLEAVE();
- /* Link is failed even though a supportable codec is found. */
- __mmplayer_check_not_supported_codec(player, (gchar *)mime);
+ return ret;
+}
- MMPLAYER_FREEIF(name_template);
- debug_error("failed to call _close_link\n");
- return FALSE;
- }
+gboolean _asm_postmsg(gpointer *data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ MMMessageParamType msg = {0, };
- MMPLAYER_FREEIF(name_template);
- return TRUE;
- }
+ MMPLAYER_FENTER();
+ return_val_if_fail ( player, FALSE );
+ debug_warning("get notified");
- gst_caps_unref(res);
+ if ((player->cmd == MMPLAYER_COMMAND_DESTROY) ||
+ (player->cmd == MMPLAYER_COMMAND_UNREALIZE))
+ {
+ debug_warning("dispatched");
+ return FALSE;
+ }
- break;
- }
- }
- /* There is no any found codec. */
- __mmplayer_check_not_supported_codec(player,(gchar *)mime);
+ msg.union_type = MM_MSG_UNION_CODE;
+ msg.code = player->sm.event_src;
- debug_error("failed to autoplug\n");
+#if 0 // should remove
+ if (player->sm.event_src == ASM_EVENT_SOURCE_RESUMABLE_CANCELED)
+ {
+ /* fill the message with state of player */
+ msg.state.current = MMPLAYER_CURRENT_STATE(player);
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_STATE_INTERRUPTED, &msg);
+ player->resumable_cancel_id = 0;
+ }
+ else
+#endif
+ {
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_READY_TO_RESUME, &msg);
+ player->resume_event_id = 0;
+ }
- debug_fleave();
-
+ debug_warning("dispatched");
return FALSE;
+}
+
+gboolean _asm_lazy_pause(gpointer *data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ int ret = MM_ERROR_NONE;
+ MMPLAYER_FENTER();
-ERROR:
+ return_val_if_fail ( player, FALSE );
- /* release */
- if ( queue )
- gst_object_unref( queue );
+ if (MMPLAYER_CURRENT_STATE(player) == MM_PLAYER_STATE_PLAYING)
+ {
+ debug_log ("Ready to proceed lazy pause\n");
+ ret = _mmplayer_pause((MMHandleType)player);
+ if(MM_ERROR_NONE != ret)
+ {
+ debug_error("MMPlayer pause failed in ASM callback lazy pause\n");
+ }
+ }
+ else
+ {
+ debug_log ("Invalid state to proceed lazy pause\n");
+ }
+ /* unset mute */
+ if (player->pipeline && player->pipeline->audiobin)
+ g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "mute", 0, NULL);
- if ( queue_pad )
- gst_object_unref( queue_pad );
+ player->sm.by_asm_cb = FALSE; //should be reset here
- if ( element )
- gst_object_unref ( element );
+ MMPLAYER_FLEAVE();
return FALSE;
}
-
-static
-int __mmplayer_check_not_supported_codec(mm_player_t* player, gchar* mime)
+static gboolean
+__mmplayer_can_do_interrupt(mm_player_t *player)
{
- debug_fenter();
-
- return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
- return_val_if_fail ( mime, MM_ERROR_INVALID_ARGUMENT );
-
- debug_log("mimetype to check: %s\n", mime );
-
- if ( MMPLAYER_IS_RTSP_STREAMING(player) )
- goto DONE;
-
- /* add missing plugin */
- /* NOTE : msl should check missing plugin for image mime type.
- * Some motion jpeg clips can have playable audio track.
- * So, msl have to play audio after displaying popup written video format not supported.
- */
- if ( !( player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst ) )
+ if (!player || !player->pipeline || !player->attrs)
{
- if ( !( player->can_support_codec | player->videodec_linked | player->audiodec_linked ) )
- {
- debug_log("not found demuxer\n");
- player->not_found_demuxer = TRUE;
- player->unlinked_demuxer_mime = g_strdup_printf ( "%s", mime );
-
- goto DONE;
- }
+ debug_warning("not initialized");
+ goto FAILED;
}
- if( ( g_str_has_prefix(mime, "video") ) ||( g_str_has_prefix(mime, "image") ) )
+ if ((player->sm.exit_cb) || (player->set_mode.pcm_extraction))
{
- debug_log("can support codec=%d, vdec_linked=%d, adec_linked=%d\n",
- player->can_support_codec, player->videodec_linked, player->audiodec_linked);
+ debug_warning("leave from asm cb right now, %d, %d", player->sm.exit_cb, player->set_mode.pcm_extraction);
+ goto FAILED;
+ }
- /* check that clip have multi tracks or not */
- if ( ( player->can_support_codec & FOUND_PLUGIN_VIDEO ) && ( player->videodec_linked ) )
- {
- debug_log("video plugin is already linked\n");
- }
- else
- {
- debug_warning("add VIDEO to missing plugin\n");
- player->not_supported_codec |= MISSING_PLUGIN_VIDEO;
- }
+ /* check if seeking */
+ if (player->doing_seek)
+ {
+ MMMessageParamType msg_param;
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
+ msg_param.code = MM_ERROR_PLAYER_SEEK;
+ player->doing_seek = FALSE;
+ MMPLAYER_POST_MSG(player, MM_MESSAGE_ERROR, &msg_param);
+ goto FAILED;
}
- else if ( g_str_has_prefix(mime, "audio") )
+
+ /* check other thread */
+ if (player->cmd_lock)
{
- if ( ( player->can_support_codec & FOUND_PLUGIN_AUDIO ) && ( player->audiodec_linked ) )
+ if (!g_mutex_trylock(player->cmd_lock))
{
- debug_log("audio plugin is already linked\n");
+ debug_warning("locked already, cmd state : %d", player->cmd);
+
+ /* check application command */
+ if (player->cmd == MMPLAYER_COMMAND_START || player->cmd == MMPLAYER_COMMAND_RESUME)
+ {
+ debug_warning("playing.. should wait cmd lock then, will be interrupted");
+ g_mutex_lock(player->cmd_lock);
+ goto INTERRUPT;
+ }
+ debug_warning("nothing to do");
+ goto FAILED;
}
else
{
- debug_warning("add AUDIO to missing plugin\n");
- player->not_supported_codec |= MISSING_PLUGIN_AUDIO;
+ debug_warning("can interrupt immediately");
+ goto INTERRUPT;
}
}
-DONE:
- debug_fleave();
+FAILED:
+ return FALSE;
- return MM_ERROR_NONE;
+INTERRUPT:
+ return TRUE;
}
-
-static void __mmplayer_pipeline_complete(GstElement *decodebin, gpointer data) // @
+ASM_cb_result_t
+__mmplayer_asm_callback(int handle, ASM_event_sources_t event_src, ASM_sound_commands_t command, unsigned int sound_status, void* cb_data)
{
- mm_player_t* player = (mm_player_t*)data;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*) cb_data;
+ ASM_cb_result_t cb_res = ASM_CB_RES_IGNORE;
+ int result = MM_ERROR_NONE;
+ gboolean lazy_pause = FALSE;
- return_if_fail( player );
+ debug_warning("get notified");
- /* remove fakesink */
- if ( ! __mmplayer_gst_remove_fakesink( player,
- &player->pipeline->mainbin[MMPLAYER_M_SRC_FAKESINK]) )
+ if (!__mmplayer_can_do_interrupt(player))
{
- /* NOTE : __mmplayer_pipeline_complete() can be called several time. because
- * signaling mechanism ( pad-added, no-more-pad, new-decoded-pad ) from various
- * source element are not same. To overcome this situation, this function will called
- * several places and several times. Therefore, this is not an error case.
- */
- return;
- }
- debug_log("pipeline has completely constructed\n");
-
- player->pipeline_is_constructed = TRUE;
-
- if ( ( PLAYER_INI()->async_start ) &&
- ( ! MMPLAYER_IS_RTSP_STREAMING ( player ) ) &&
- ( player->posted_msg == FALSE ) )
- {
- __mmplayer_post_missed_plugin( player );
+ debug_warning("no need to interrupt, so leave");
+ goto EXIT;
}
-
- MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-complate" );
-}
-static gboolean __mmplayer_configure_audio_callback(mm_player_t* player)
-{
- debug_fenter();
-
- return_val_if_fail ( player, FALSE );
+ player->sm.cb_pending = TRUE;
+ debug_warning("asm event src type : %d, command : 0x%x", event_src, command);
+ player->sm.by_asm_cb = TRUE;
+ player->sm.event_src = event_src;
- if ( MMPLAYER_IS_STREAMING(player) )
- return FALSE;
-
- /* This callback can be set to music player only. */
- if((player->can_support_codec & 0x02) == FOUND_PLUGIN_VIDEO)
+ /* first, check event source */
+ if(event_src == ASM_EVENT_SOURCE_EARJACK_UNPLUG)
{
- debug_warning("audio callback is not supported for video");
- return FALSE;
+ int stop_by_asm = 0;
+ mm_attrs_get_int_by_name(player->attrs, "sound_stop_when_unplugged", &stop_by_asm);
+ if (!stop_by_asm)
+ goto DONE;
}
-
- if (player->audio_stream_cb)
+ else if (event_src == ASM_EVENT_SOURCE_RESOURCE_CONFLICT)
{
+ /* can use video overlay simultaneously */
+ /* video resource conflict */
+ if(player->pipeline->videobin)
{
- GstPad *pad = NULL;
+ if (player->ini.multiple_codec_supported)
+ {
+ debug_log("video conflict but, can support multiple video");
+ result = _mmplayer_pause((MMHandleType)player);
+ cb_res = ASM_CB_RES_PAUSE;
+ }
+ else
+ {
+ debug_log("video conflict, can't support multiple video");
+ result = _mmplayer_unrealize((MMHandleType)player);
+ cb_res = ASM_CB_RES_STOP;
+ }
+ }
+ else if (player->pipeline->audiobin)
+ {
+ debug_log("audio resource conflict");
+ result = _mmplayer_pause((MMHandleType)player);
+ if (result != MM_ERROR_NONE)
+ {
+ debug_warning("fail to set pause by asm");
+ }
+ cb_res = ASM_CB_RES_PAUSE;
+ }
+ goto DONE;
+ }
+#if 0 // should remove
+ else if (event_src == ASM_EVENT_SOURCE_RESUMABLE_CANCELED)
+ {
+ debug_warning("Got msg from asm for resumable canceled.\n");
+ player->sm.antishock = TRUE;
+ player->sm.by_asm_cb = FALSE;
- pad = gst_element_get_static_pad (player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "sink");
+ player->resumable_cancel_id = g_idle_add((GSourceFunc)_asm_postmsg, (gpointer)player);
+ cb_res = ASM_CB_RES_IGNORE;
+ goto DONE;
+ }
+#endif
+ /* then, check command */
+ switch(command)
+ {
+ case ASM_COMMAND_PLAY:
+ debug_warning ("Got unexpected asm command (%d)", command);
+ break;
- if ( !pad )
+ case ASM_COMMAND_STOP: // notification case
{
- debug_error("failed to get sink pad from audiosink to probe data\n");
- return FALSE;
+ GstFormat format = GST_FORMAT_TIME;
+ debug_warning("Got msg from asm to stop");
+
+ if (!gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &format, &player->last_position)) {
+ debug_error("failed to get position");
+ player->last_position = 0;
+ }
+
+ debug_log ("pos_msec = %"GST_TIME_FORMAT"", GST_TIME_ARGS(player->last_position));
+
+ result = _mmplayer_stop((MMHandleType)player);
+ if (result != MM_ERROR_NONE)
+ {
+ debug_warning("fail to set stop state by asm");
+ cb_res = ASM_CB_RES_IGNORE;
+ }
+ else
+ {
+ cb_res = ASM_CB_RES_STOP;
+ }
+ player->sm.by_asm_cb = FALSE; // reset because no message any more from asm
+ player->sm.keep_last_pos = TRUE;
}
+ break;
- player->audio_cb_probe_id = gst_pad_add_buffer_probe (pad,
- G_CALLBACK (__mmplayer_audio_stream_probe), player);
+ case ASM_COMMAND_PAUSE:
+ {
+ debug_warning("Got msg from asm to Pause");
+ if(event_src == ASM_EVENT_SOURCE_CALL_START ||
+ event_src == ASM_EVENT_SOURCE_NOTIFY_START ||
+ event_src == ASM_EVENT_SOURCE_ALARM_START)
+ {
+ if(__mmplayer_ignore_current_external_display_mode(player) == MM_ERROR_NONE)
+ debug_log("screen will not display to external display");
+ }
+ if(event_src == ASM_EVENT_SOURCE_CALL_START ||
+ event_src == ASM_EVENT_SOURCE_ALARM_START ||
+ event_src == ASM_EVENT_SOURCE_MEDIA)
+ {
+ if ( ! MMPLAYER_IS_RTSP_STREAMING(player) ) {
+ //hold 0.7 second to excute "fadedown mute" effect
+ debug_warning ("do fade down->pause->undo fade down");
- gst_object_unref (pad);
+ __mmplayer_do_sound_fadedown(player, MM_PLAYER_FADEOUT_TIME_DEFAULT);
- pad = NULL;
- }
+ result = _mmplayer_pause((MMHandleType)player);
+ if (result != MM_ERROR_NONE)
+ {
+ debug_warning("fail to set Pause state by asm");
+ cb_res = ASM_CB_RES_IGNORE;
+ break;
+ }
+ __mmplayer_undo_sound_fadedown(player);
+ } else {
+ /* rtsp should connect again in specific network becasue tcp session can't be kept any more */
+ _mmplayer_unrealize((MMHandleType)player);
+ }
+ }
+#ifdef USE_LAZY_PAUSE // if enabled, should consider event id and context when removed
+ else if(event_src == ASM_EVENT_SOURCE_OTHER_PLAYER_APP)
+ {
+ lazy_pause = TRUE; // return as soon as possible, for fast start of other app
+
+ if ( player->pipeline->audiobin && player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
+ g_object_set( player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "mute", 2, NULL);
+
+ player->lazy_pause_event_id = g_timeout_add(LAZY_PAUSE_TIMEOUT_MSEC, (GSourceFunc)_asm_lazy_pause, (gpointer)player);
+ debug_warning ("set lazy pause timer (id=[%d], timeout=[%d ms])", player->lazy_pause_event_id, LAZY_PAUSE_TIMEOUT_MSEC);
+ }
+#endif
+ else
+ {
+ debug_warning ("pause immediately");
+ result = _mmplayer_pause((MMHandleType)player);
+ }
+ cb_res = ASM_CB_RES_PAUSE;
+ }
+ break;
+
+ case ASM_COMMAND_RESUME:
+ {
+ debug_warning("Got msg from asm to Resume. So, application can resume. code (%d) \n", event_src);
+ player->sm.antishock = TRUE;
+ player->sm.by_asm_cb = FALSE;
+
+ //ASM server is single thread daemon. So use g_idle_add() to post resume msg
+ player->resume_event_id = g_idle_add((GSourceFunc)_asm_postmsg, (gpointer)player);
+ cb_res = ASM_CB_RES_IGNORE;
+ goto DONE;
+ }
+ break;
+
+ default:
+ break;
}
- else
+
+DONE:
+ if ( !lazy_pause )
{
- debug_error("There is no audio callback to configure.\n");
- return FALSE;
+ player->sm.by_asm_cb = FALSE;
}
+ player->sm.cb_pending = FALSE;
+ MMPLAYER_CMD_UNLOCK( player );
- debug_fleave();
-
- return TRUE;
+EXIT:
+ debug_warning("dispatched");
+ return cb_res;
}
-static void
-__mmplayer_init_factories(mm_player_t* player) // @
+int
+_mmplayer_create_player(MMHandleType handle) // @
{
- debug_fenter();
+ mm_player_t* player = MM_PLAYER_CAST(handle);
- return_if_fail ( player );
+ MMPLAYER_FENTER();
- player->factories = gst_registry_feature_filter(gst_registry_get_default(),
- (GstPluginFeatureFilter)__mmplayer_feature_filter, FALSE, NULL);
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- player->factories = g_list_sort(player->factories, (GCompareFunc)util_factory_rank_compare);
+ /* initialize player state */
+ MMPLAYER_CURRENT_STATE(player) = MM_PLAYER_STATE_NONE;
+ MMPLAYER_PREV_STATE(player) = MM_PLAYER_STATE_NONE;
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_NONE;
- debug_fleave();
-}
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL ( player, MMPLAYER_COMMAND_CREATE );
-static void
-__mmplayer_release_factories(mm_player_t* player) // @
-{
- debug_fenter();
+ /* construct attributes */
+ player->attrs = _mmplayer_construct_attribute(handle);
- return_if_fail ( player );
+ if ( !player->attrs )
+ {
+ debug_error("Failed to construct attributes\n");
+ goto ERROR;
+ }
- if (player->factories)
+ /* initialize gstreamer with configured parameter */
+ if ( ! __mmplayer_init_gstreamer(player) )
{
- gst_plugin_feature_list_free (player->factories);
- player->factories = NULL;
+ debug_error("Initializing gstreamer failed\n");
+ goto ERROR;
}
- debug_fleave();
-}
+ /* initialize factories if not using decodebin */
+ if( player->factories == NULL )
+ __mmplayer_init_factories(player);
-static void
-__mmplayer_release_misc(mm_player_t* player)
-{
- debug_fenter();
+ /* create lock. note that g_tread_init() has already called in gst_init() */
+ player->fsink_lock = g_mutex_new();
+ if ( ! player->fsink_lock )
+ {
+ debug_error("Cannot create mutex for command lock\n");
+ goto ERROR;
+ }
- return_if_fail ( player );
+ /* create repeat mutex */
+ player->repeat_thread_mutex = g_mutex_new();
+ if ( ! player->repeat_thread_mutex )
+ {
+ debug_error("Cannot create repeat mutex\n");
+ goto ERROR;
+ }
- /* free memory related to sound effect */
- if(player->audio_filter_info.custom_ext_level_for_plugin)
+ /* create repeat cond */
+ player->repeat_thread_cond = g_cond_new();
+ if ( ! player->repeat_thread_cond )
{
- free(player->audio_filter_info.custom_ext_level_for_plugin);
+ debug_error("Cannot create repeat cond\n");
+ goto ERROR;
}
- debug_fleave();
-}
+ /* create repeat thread */
+ player->repeat_thread =
+ g_thread_create (__mmplayer_repeat_thread, (gpointer)player, TRUE, NULL);
+ if ( ! player->repeat_thread )
+ {
+ debug_error("failed to create repeat thread");
+ goto ERROR;
+ }
-static GstElement *__mmplayer_element_create_and_link(mm_player_t *player, GstPad* pad, const char* name)
-{
- GstElement *element = NULL;
- GstPad *sinkpad;
+ /* create next play mutex */
+ player->next_play_thread_mutex = g_mutex_new();
+ if ( ! player->next_play_thread_mutex )
+ {
+ debug_error("Cannot create next_play mutex\n");
+ goto ERROR;
+ }
- debug_log("creating %s to plug\n", name);
-
- element = gst_element_factory_make(name, NULL);
- if ( ! element )
+ /* create next play cond */
+ player->next_play_thread_cond = g_cond_new();
+ if ( ! player->next_play_thread_cond )
{
- debug_error("failed to create queue\n");
- return NULL;
+ debug_error("Cannot create next_play cond\n");
+ goto ERROR;
}
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(element, GST_STATE_READY) )
+ /* create next play thread */
+ player->next_play_thread =
+ g_thread_create (__mmplayer_next_play_thread, (gpointer)player, TRUE, NULL);
+ if ( ! player->next_play_thread )
{
- debug_error("failed to set state READY to %s\n", name);
- return NULL;
+ debug_error("failed to create next play thread");
+ goto ERROR;
}
- if ( ! gst_bin_add(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, element) )
+ if ( MM_ERROR_NONE != _mmplayer_initialize_video_capture(player))
{
- debug_error("failed to add %s\n", name);
- return NULL;
+ debug_error("failed to initialize video capture\n");
+ goto ERROR;
}
- sinkpad = gst_element_get_static_pad(element, "sink");
+ /* register to asm */
+ if ( MM_ERROR_NONE != _mmplayer_asm_register(&player->sm, (ASM_sound_cb_t)__mmplayer_asm_callback, (void*)player) )
+ {
+ /* NOTE : we are dealing it as an error since we cannot expect it's behavior */
+ debug_error("failed to register asm server\n");
+ return MM_ERROR_POLICY_INTERNAL;
+ }
- if ( GST_PAD_LINK_OK != gst_pad_link(pad, sinkpad) )
+ /* to add active device callback */
+ if ( MM_ERROR_NONE != mm_sound_add_device_information_changed_callback(MM_SOUND_DEVICE_STATE_ACTIVATED_FLAG, __mmplayer_sound_device_info_changed_cb_func, (void*)player))
{
- debug_error("failed to link %s\n", name);
- gst_object_unref (sinkpad);
-
- return NULL;
+ debug_error("failed mm_sound_add_device_information_changed_callback \n");
}
- debug_log("linked %s to pipeline successfully\n", name);
+ if (MMPLAYER_IS_HTTP_PD(player))
+ {
+ player->pd_downloader = NULL;
+ player->pd_file_save_path = NULL;
+ }
- gst_object_unref (sinkpad);
+ player->streaming_type = STREAMING_SERVICE_NONE;
- return element;
-}
+ /* give default value of audio effect setting */
+ player->sound.volume = MM_VOLUME_FACTOR_DEFAULT;
+ player->playback_rate = DEFAULT_PLAYBACK_RATE;
-static gboolean
-__mmplayer_close_link(mm_player_t* player, GstPad *srcpad, GstElement *sinkelement,
-const char *padname, const GList *templlist)
-{
- GstPad *pad = NULL;
- gboolean has_dynamic_pads = FALSE;
- gboolean has_many_types = FALSE;
- const char *klass = NULL;
- GstStaticPadTemplate *padtemplate = NULL;
- GstElementFactory *factory = NULL;
- GstElement* queue = NULL;
- GstElement* parser = NULL;
- GstPad *pssrcpad = NULL;
- GstPad *qsrcpad = NULL, *qsinkpad = NULL;
- MMPlayerGstElement *mainbin = NULL;
- GstStructure* str = NULL;
- GstCaps* srccaps = NULL;
- GstState warmup = GST_STATE_READY;
- gboolean isvideo_decoder = FALSE;
- guint q_max_size_time = 0;
+ player->play_subtitle = FALSE;
+ player->use_textoverlay = FALSE;
+ player->play_count = 0;
+ player->use_decodebin = TRUE;
+ player->ignore_asyncdone = FALSE;
+ player->use_deinterleave = FALSE;
+ player->pp_rebuilding = FALSE;
+ player->max_audio_channels = 0;
+ player->video_share_api_delta = 0;
+ player->video_share_clock_delta = 0;
+ player->has_closed_caption = FALSE;
- debug_fenter();
+ __mmplayer_post_proc_reset(player);
- return_val_if_fail ( player &&
- player->pipeline &&
- player->pipeline->mainbin,
- FALSE );
+ if (player->ini.dump_element_keyword[0][0] == '\0')
+ {
+ player->ini.set_dump_element_flag= FALSE;
+ }
+ else
+ {
+ player->ini.set_dump_element_flag = TRUE;
+ }
- mainbin = player->pipeline->mainbin;
+ /* set player state to null */
+ MMPLAYER_STATE_CHANGE_TIMEOUT(player) = player->ini.localplayback_state_change_timeout;
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_NULL );
- debug_log("plugging pad %s:%s to newly create %s:%s\n",
- GST_ELEMENT_NAME( GST_PAD_PARENT ( srcpad ) ),
- GST_PAD_NAME( srcpad ),
- GST_ELEMENT_NAME( sinkelement ),
- padname);
+ return MM_ERROR_NONE;
+ERROR:
+ /* free lock */
+ if ( player->fsink_lock )
+ g_mutex_free( player->fsink_lock );
+ player->fsink_lock = NULL;
- factory = gst_element_get_factory(sinkelement);
- klass = gst_element_factory_get_klass(factory);
+ /* free thread */
+ if ( player->repeat_thread_cond &&
+ player->repeat_thread_mutex &&
+ player->repeat_thread )
+ {
+ player->repeat_thread_exit = TRUE;
+ g_cond_signal( player->repeat_thread_cond );
- /* check if player can do start continually */
- MMPLAYER_CHECK_CMD_IF_EXIT(player);
+ g_thread_join( player->repeat_thread );
+ player->repeat_thread = NULL;
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(sinkelement, warmup) )
- {
- if (isvideo_decoder)
- player->keep_detecting_vcodec = TRUE;
+ g_mutex_free ( player->repeat_thread_mutex );
+ player->repeat_thread_mutex = NULL;
- debug_error("failed to set %d state to %s\n", warmup, GST_ELEMENT_NAME( sinkelement ));
- goto ERROR;
+ g_cond_free ( player->repeat_thread_cond );
+ player->repeat_thread_cond = NULL;
}
+ /* clear repeat thread mutex/cond if still alive
+ * this can happen if only thread creating has failed
+ */
+ if ( player->repeat_thread_mutex )
+ g_mutex_free ( player->repeat_thread_mutex );
- /* add to pipeline */
- if ( ! gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), sinkelement) )
+ if ( player->repeat_thread_cond )
+ g_cond_free ( player->repeat_thread_cond );
+
+ /* free next play thread */
+ if ( player->next_play_thread_cond &&
+ player->next_play_thread_mutex &&
+ player->next_play_thread )
{
- debug_error("failed to add %s to mainbin\n", GST_ELEMENT_NAME( sinkelement ));
- goto ERROR;
- }
+ player->next_play_thread_exit = TRUE;
+ g_cond_signal( player->next_play_thread_cond );
- debug_log("element klass : %s\n", klass);
+ g_thread_join( player->next_play_thread );
+ player->next_play_thread = NULL;
- /* added to support multi track files */
- /* only decoder case and any of the video/audio still need to link*/
- if(g_strrstr(klass, "Decoder") && __mmplayer_link_decoder(player,srcpad))
- {
- gchar *name = NULL;
+ g_mutex_free ( player->next_play_thread_mutex );
+ player->next_play_thread_mutex = NULL;
- name = g_strdup(GST_ELEMENT_NAME( GST_PAD_PARENT ( srcpad )));
-
- if (g_strrstr(name, "mpegtsdemux"))
- {
- gchar *demux_caps = NULL;
- gchar *parser_name = NULL;
- GstCaps *dcaps = NULL;
+ g_cond_free ( player->next_play_thread_cond );
+ player->next_play_thread_cond = NULL;
+ }
+ /* clear next play thread mutex/cond if still alive
+ * this can happen if only thread creating has failed
+ */
+ if ( player->next_play_thread_mutex )
+ g_mutex_free ( player->next_play_thread_mutex );
- dcaps = gst_pad_get_caps(srcpad);
- demux_caps = gst_caps_to_string(dcaps);
-
- if (g_strrstr(demux_caps, "video/x-h264"))
- {
- parser_name = g_strdup("h264parse");
- }
- else if (g_strrstr(demux_caps, "video/mpeg"))
- {
- parser_name = g_strdup("mpeg4videoparse");
- }
-
- gst_caps_unref(dcaps);
- MMPLAYER_FREEIF( demux_caps );
+ if ( player->next_play_thread_cond )
+ g_cond_free ( player->next_play_thread_cond );
- if (parser_name)
- {
- parser = __mmplayer_element_create_and_link(player, srcpad, parser_name);
+ /* release attributes */
+ _mmplayer_deconstruct_attribute(handle);
- MMPLAYER_FREEIF(parser_name);
-
- if ( ! parser )
- {
- debug_error("failed to create parser\n");
- }
- else
- {
- /* update srcpad if parser is created */
- pssrcpad = gst_element_get_static_pad(parser, "src");
- srcpad = pssrcpad;
- }
- }
- }
- MMPLAYER_FREEIF(name);
+ MMPLAYER_FLEAVE();
- queue = __mmplayer_element_create_and_link(player, srcpad, "queue"); // parser - queue or demuxer - queue
- if ( ! queue )
- {
- debug_error("failed to create queue\n");
- goto ERROR;
- }
+ return MM_ERROR_PLAYER_INTERNAL;
+}
- /* update srcpad to link with decoder */
- qsrcpad = gst_element_get_static_pad(queue, "src");
- srcpad = qsrcpad;
+static gboolean
+__mmplayer_init_gstreamer(mm_player_t* player) // @
+{
+ static gboolean initialized = FALSE;
+ static const int max_argc = 50;
+ gint* argc = NULL;
+ gchar** argv = NULL;
+ gchar** argv2 = NULL;
+ GError *err = NULL;
+ int i = 0;
+ int arg_count = 0;
- if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
- q_max_size_time = GST_QUEUE_HLS_TIME;
- else
- q_max_size_time = GST_QUEUE_DEFAULT_TIME;
+ if ( initialized )
+ {
+ debug_log("gstreamer already initialized.\n");
+ return TRUE;
+ }
- /* assigning queue handle for futher manipulation purpose */
- /* FIXIT : make it some kind of list so that msl can support more then two stream (text, data, etc...) */
- if(mainbin[MMPLAYER_M_Q1].gst == NULL)
- {
- mainbin[MMPLAYER_M_Q1].id = MMPLAYER_M_Q1;
- mainbin[MMPLAYER_M_Q1].gst = queue;
+ /* alloc */
+ argc = malloc( sizeof(int) );
+ argv = malloc( sizeof(gchar*) * max_argc );
+ argv2 = malloc( sizeof(gchar*) * max_argc );
- g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q1].gst), "max-size-time", q_max_size_time * GST_SECOND, NULL);
- }
- else if(mainbin[MMPLAYER_M_Q2].gst == NULL)
- {
- mainbin[MMPLAYER_M_Q2].id = MMPLAYER_M_Q2;
- mainbin[MMPLAYER_M_Q2].gst = queue;
+ if ( !argc || !argv || !argv2)
+ goto ERROR;
- g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q2].gst), "max-size-time", q_max_size_time * GST_SECOND, NULL);
- }
- else
- {
- debug_critical("Not supporting more then two elementary stream\n");
- g_assert(1);
- }
+ memset( argv, 0, sizeof(gchar*) * max_argc );
+ memset( argv2, 0, sizeof(gchar*) * max_argc );
- pad = gst_element_get_static_pad(sinkelement, padname);
+ /* add initial */
+ *argc = 1;
+ argv[0] = g_strdup( "mmplayer" );
- if ( ! pad )
+ /* add gst_param */
+ for ( i = 0; i < 5; i++ ) /* FIXIT : num of param is now fixed to 5. make it dynamic */
+ {
+ if ( strlen( player->ini.gst_param[i] ) > 0 )
{
- debug_warning("failed to get pad(%s) from %s. retrying with [sink]\n",
- padname, GST_ELEMENT_NAME(sinkelement) );
-
- pad = gst_element_get_static_pad(sinkelement, "sink");
- if ( ! pad )
- {
- debug_error("failed to get pad(sink) from %s. \n",
- GST_ELEMENT_NAME(sinkelement) );
- goto ERROR;
- }
+ argv[*argc] = g_strdup( player->ini.gst_param[i] );
+ (*argc)++;
}
+ }
- /* to check the video/audio type set the proper flag*/
- {
- srccaps = gst_pad_get_caps( srcpad );
- if ( !srccaps )
- goto ERROR;
+ /* we would not do fork for scanning plugins */
+ argv[*argc] = g_strdup("--gst-disable-registry-fork");
+ (*argc)++;
- str = gst_caps_get_structure( srccaps, 0 );
- if ( ! str )
- goto ERROR;
+ /* check disable registry scan */
+ if ( player->ini.skip_rescan )
+ {
+ argv[*argc] = g_strdup("--gst-disable-registry-update");
+ (*argc)++;
+ }
- name = gst_structure_get_name(str);
- if ( ! name )
- goto ERROR;
- }
+ /* check disable segtrap */
+ if ( player->ini.disable_segtrap )
+ {
+ argv[*argc] = g_strdup("--gst-disable-segtrap");
+ (*argc)++;
+ }
- /* link queue and decoder. so, it will be queue - decoder. */
- if ( GST_PAD_LINK_OK != gst_pad_link(srcpad, pad) )
- {
- gst_object_unref(GST_OBJECT(pad));
- debug_error("failed to link (%s) to pad(%s)\n", GST_ELEMENT_NAME( sinkelement ), padname );
+ debug_log("initializing gstreamer with following parameter\n");
+ debug_log("argc : %d\n", *argc);
+ arg_count = *argc;
- /* reconstitute supportable codec */
- {
- if (strstr(name, "video"))
- {
- player->can_support_codec ^= FOUND_PLUGIN_VIDEO;
- }
+ for ( i = 0; i < arg_count; i++ )
+ {
+ argv2[i] = argv[i];
+ debug_log("argv[%d] : %s\n", i, argv2[i]);
+ }
- if (strstr(name, "audio"))
- {
- player->can_support_codec ^= FOUND_PLUGIN_AUDIO;
- }
- }
- goto ERROR;
+
+ /* initializing gstreamer */
+ if ( ! gst_init_check (argc, &argv, &err))
+ {
+ debug_error("Could not initialize GStreamer: %s\n", err ? err->message : "unknown error occurred");
+ if (err)
+ {
+ g_error_free (err);
}
- gst_object_unref(GST_OBJECT(pad));
+ goto ERROR;
+ }
+ /* release */
+ for ( i = 0; i < arg_count; i++ )
+ {
+ //debug_log("release - argv[%d] : %s\n", i, argv2[i]);
+ MMPLAYER_FREEIF( argv2[i] );
+ }
- {
- if (strstr(name, "video"))
- {
- player->videodec_linked = 1;
- debug_msg("player->videodec_linked set to 1\n");
+ MMPLAYER_FREEIF( argv );
+ MMPLAYER_FREEIF( argv2 );
+ MMPLAYER_FREEIF( argc );
- }
- if (strstr(name, "audio"))
- {
- player->audiodec_linked = 1;
- debug_msg("player->auddiodec_linked set to 1\n");
- }
+ /* done */
+ initialized = TRUE;
- gst_caps_unref(GST_CAPS(srccaps));
- srccaps = NULL;
- }
- }
+ return TRUE;
- if ( !MMPLAYER_IS_HTTP_PD(player) )
+ERROR:
+
+ /* release */
+ for ( i = 0; i < arg_count; i++ )
{
- if( (g_strrstr(klass, "Demux") && !g_strrstr(klass, "Metadata")) || (g_strrstr(klass, "Parser") ) )
- {
- if (MMPLAYER_IS_HTTP_STREAMING(player) || MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
- {
- GstFormat fmt = GST_FORMAT_BYTES;
- gint64 dur_bytes = 0L;
- gchar *file_buffering_path = NULL;
- gboolean use_file_buffer = FALSE;
+ debug_log("free[%d] : %s\n", i, argv2[i]);
+ MMPLAYER_FREEIF( argv2[i] );
+ }
- if ( !mainbin[MMPLAYER_M_S_BUFFER].gst)
- {
- debug_log("creating http streaming buffering queue\n");
+ MMPLAYER_FREEIF( argv );
+ MMPLAYER_FREEIF( argv2 );
+ MMPLAYER_FREEIF( argc );
- queue = gst_element_factory_make("queue2", "http_streaming_buffer");
- if ( ! queue )
- {
- debug_critical ( "failed to create buffering queue element\n" );
- goto ERROR;
- }
+ return FALSE;
+}
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(queue, GST_STATE_READY) )
- {
- debug_error("failed to set state READY to buffering queue\n");
- goto ERROR;
- }
+int
+__mmplayer_destroy_streaming_ext(mm_player_t* player)
+{
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- if ( !gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue) )
- {
- debug_error("failed to add buffering queue\n");
- goto ERROR;
- }
+ if (player->pd_downloader)
+ {
+ _mmplayer_unrealize_pd_downloader((MMHandleType)player);
+ MMPLAYER_FREEIF(player->pd_downloader);
+ }
- qsinkpad = gst_element_get_static_pad(queue, "sink");
- qsrcpad = gst_element_get_static_pad(queue, "src");
+ if (MMPLAYER_IS_HTTP_PD(player))
+ {
+ _mmplayer_destroy_pd_downloader((MMHandleType)player);
+ MMPLAYER_FREEIF(player->pd_file_save_path);
+ }
- if ( GST_PAD_LINK_OK != gst_pad_link(srcpad, qsinkpad) )
- {
- debug_error("failed to link buffering queue\n");
- goto ERROR;
- }
- srcpad = qsrcpad;
+ return MM_ERROR_NONE;
+}
+int
+_mmplayer_destroy(MMHandleType handle) // @
+{
+ mm_player_t* player = MM_PLAYER_CAST(handle);
- mainbin[MMPLAYER_M_S_BUFFER].id = MMPLAYER_M_S_BUFFER;
- mainbin[MMPLAYER_M_S_BUFFER].gst = queue;
+ MMPLAYER_FENTER();
- if ( !MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
- {
- if ( !gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, &fmt, &dur_bytes))
- debug_error("fail to get duration.\n");
+ /* check player handle */
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- if (dur_bytes>0)
- {
- use_file_buffer = MMPLAYER_USE_FILE_FOR_BUFFERING(player);
- file_buffering_path = g_strdup(PLAYER_INI()->http_file_buffer_path);
- }
- }
+ /* destroy can called at anytime */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL ( player, MMPLAYER_COMMAND_DESTROY );
- __mm_player_streaming_set_buffer(player->streamer,
- queue,
- TRUE,
- PLAYER_INI()->http_max_size_bytes,
- 1.0,
- PLAYER_INI()->http_buffering_limit,
- PLAYER_INI()->http_buffering_time,
- use_file_buffer,
- file_buffering_path,
- dur_bytes);
+ __mmplayer_destroy_streaming_ext(player);
- MMPLAYER_FREEIF(file_buffering_path);
- }
- }
- }
+ /* release repeat thread */
+ if ( player->repeat_thread_cond &&
+ player->repeat_thread_mutex &&
+ player->repeat_thread )
+ {
+ player->repeat_thread_exit = TRUE;
+ g_cond_signal( player->repeat_thread_cond );
+
+ debug_log("waitting for repeat thread exit\n");
+ g_thread_join ( player->repeat_thread );
+ g_mutex_free ( player->repeat_thread_mutex );
+ g_cond_free ( player->repeat_thread_cond );
+ debug_log("repeat thread released\n");
}
- /* if it is not decoder or */
- /* in decoder case any of the video/audio still need to link*/
- if(!g_strrstr(klass, "Decoder"))
+
+ /* release next play thread */
+ if ( player->next_play_thread_cond &&
+ player->next_play_thread_mutex &&
+ player->next_play_thread )
{
+ player->next_play_thread_exit = TRUE;
+ g_cond_signal( player->next_play_thread_cond );
- pad = gst_element_get_static_pad(sinkelement, padname);
- if ( ! pad )
- {
- debug_warning("failed to get pad(%s) from %s. retrying with [sink]\n",
- padname, GST_ELEMENT_NAME(sinkelement) );
+ debug_log("waitting for next play thread exit\n");
+ g_thread_join ( player->next_play_thread );
+ g_mutex_free ( player->next_play_thread_mutex );
+ g_cond_free ( player->next_play_thread_cond );
+ debug_log("next play thread released\n");
+ }
- pad = gst_element_get_static_pad(sinkelement, "sink");
+ _mmplayer_release_video_capture(player);
- if ( ! pad )
- {
- debug_error("failed to get pad(sink) from %s. \n",
- GST_ELEMENT_NAME(sinkelement) );
- goto ERROR;
- }
- }
+ /* flush any pending asm_cb */
+ if (player->sm.cb_pending)
+ {
+ /* set a flag for make sure asm_cb to be returned immediately */
+ debug_warning("asm cb has pending state");
+ player->sm.exit_cb = TRUE;
- if ( GST_PAD_LINK_OK != gst_pad_link(srcpad, pad) )
- {
- gst_object_unref(GST_OBJECT(pad));
- debug_error("failed to link (%s) to pad(%s)\n", GST_ELEMENT_NAME( sinkelement ), padname );
- goto ERROR;
- }
+ /* make sure to release any pending asm_cb which locked by cmd_lock */
+ MMPLAYER_CMD_UNLOCK(player);
+ sched_yield();
+ MMPLAYER_CMD_LOCK(player);
+ }
- gst_object_unref(GST_OBJECT(pad));
+ /* withdraw asm */
+ if ( MM_ERROR_NONE != _mmplayer_asm_unregister(&player->sm) )
+ {
+ debug_error("failed to deregister asm server\n");
}
- for(;templlist != NULL; templlist = templlist->next)
+#ifdef USE_LAZY_PAUSE
+ if (player->lazy_pause_event_id)
{
- padtemplate = templlist->data;
+ __mmplayer_remove_g_source_from_context(player->context.global_default, player->lazy_pause_event_id);
+ player->lazy_pause_event_id = 0;
+ }
+#endif
- debug_log ("director = [%d], presence = [%d]\n", padtemplate->direction, padtemplate->presence);
+ if (player->resume_event_id)
+ {
+ g_source_remove (player->resume_event_id);
+ player->resume_event_id = 0;
+ }
- if( padtemplate->direction != GST_PAD_SRC ||
- padtemplate->presence == GST_PAD_REQUEST )
- continue;
+ if (player->resumable_cancel_id)
+ {
+ g_source_remove (player->resumable_cancel_id);
+ player->resumable_cancel_id = 0;
+ }
- switch(padtemplate->presence)
- {
- case GST_PAD_ALWAYS:
- {
- GstPad *srcpad = gst_element_get_static_pad(sinkelement, "src");
- GstCaps *caps = gst_pad_get_caps(srcpad);
+ /* release pipeline */
+ if ( MM_ERROR_NONE != __mmplayer_gst_destroy_pipeline( player ) )
+ {
+ debug_error("failed to destory pipeline\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- /* Check whether caps has many types */
- if ( gst_caps_get_size (caps) > 1 && g_strrstr(klass, "Parser")) {
- debug_log ("has_many_types for this caps [%s]\n", gst_caps_to_string(caps));
- has_many_types = TRUE;
- break;
- }
+ if (player->is_external_subtitle_present && player->subtitle_language_list)
+ {
+ g_list_free (player->subtitle_language_list);
+ player->subtitle_language_list = NULL;
+ }
- if ( ! __mmplayer_try_to_plug(player, srcpad, caps) )
- {
- gst_object_unref(GST_OBJECT(srcpad));
- gst_caps_unref(GST_CAPS(caps));
+ __mmplayer_release_dump_list (player->dump_list);
- debug_error("failed to plug something after %s\n", GST_ELEMENT_NAME( sinkelement ));
- goto ERROR;
- }
+ /* release miscellaneous information.
+ these info needs to be released after pipeline is destroyed. */
+ __mmplayer_release_misc_post( player );
- gst_caps_unref(GST_CAPS(caps));
- gst_object_unref(GST_OBJECT(srcpad));
+ /* release attributes */
+ _mmplayer_deconstruct_attribute( handle );
- }
- break;
+ /* release factories */
+ __mmplayer_release_factories( player );
+ /* release lock */
+ if ( player->fsink_lock )
+ g_mutex_free( player->fsink_lock );
- case GST_PAD_SOMETIMES:
- has_dynamic_pads = TRUE;
- break;
+ if ( player->msg_cb_lock )
+ g_mutex_free( player->msg_cb_lock );
- default:
- break;
- }
- }
+ MMPLAYER_FLEAVE();
- /* check if player can do start continually */
- MMPLAYER_CHECK_CMD_IF_EXIT(player);
+ return MM_ERROR_NONE;
+}
- if( has_dynamic_pads )
+int
+__mmplayer_realize_streaming_ext(mm_player_t* player)
+{
+ int ret = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ if (MMPLAYER_IS_HTTP_PD(player))
{
- player->have_dynamic_pad = TRUE;
- MMPLAYER_SIGNAL_CONNECT ( player, sinkelement, "pad-added",
- G_CALLBACK(__mmplayer_add_new_pad), player);
+ gboolean bret = FALSE;
- /* for streaming, more then one typefind will used for each elementary stream
- * so this doesn't mean the whole pipeline completion
- */
- if ( ! MMPLAYER_IS_RTSP_STREAMING( player ) )
+ player->pd_downloader = _mmplayer_create_pd_downloader();
+ if ( !player->pd_downloader )
{
- MMPLAYER_SIGNAL_CONNECT( player, sinkelement, "no-more-pads",
- G_CALLBACK(__mmplayer_pipeline_complete), player);
+ debug_error ("Unable to create PD Downloader...");
+ ret = MM_ERROR_PLAYER_NO_FREE_SPACE;
}
- }
- if (has_many_types)
- {
- GstPad *pad = NULL;
+ bret = _mmplayer_realize_pd_downloader((MMHandleType)player, player->profile.uri, player->pd_file_save_path, player->pipeline->mainbin[MMPLAYER_M_SRC].gst);
- player->has_many_types = has_many_types;
-
- pad = gst_element_get_static_pad(sinkelement, "src");
- MMPLAYER_SIGNAL_CONNECT (player, pad, "notify::caps", G_CALLBACK(__mmplayer_add_new_caps), player);
- gst_object_unref (GST_OBJECT(pad));
+ if (FALSE == bret)
+ {
+ debug_error ("Unable to create PD Downloader...");
+ ret = MM_ERROR_PLAYER_NOT_INITIALIZED;
+ }
}
+ MMPLAYER_FLEAVE();
+ return ret;
+}
- /* check if player can do start continually */
- MMPLAYER_CHECK_CMD_IF_EXIT(player);
+int
+_mmplayer_realize(MMHandleType hplayer) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ char *uri =NULL;
+ void *param = NULL;
+ int application_pid = -1;
+ gboolean update_registry = FALSE;
+ MMHandleType attrs = 0;
+ int ret = MM_ERROR_NONE;
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(sinkelement, GST_STATE_PAUSED) )
+ MMPLAYER_FENTER();
+
+ /* check player handle */
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED )
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_REALIZE );
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
{
- debug_error("failed to set state PAUSED to %s\n", GST_ELEMENT_NAME( sinkelement ));
- goto ERROR;
+ debug_error("fail to get attributes.\n");
+ return MM_ERROR_PLAYER_INTERNAL;
}
- if ( queue )
- {
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state (queue, GST_STATE_PAUSED) )
- {
- debug_error("failed to set state PAUSED to queue\n");
- goto ERROR;
- }
+ mm_attrs_get_int_by_name(attrs, "sound_application_pid", &application_pid );
+ player->sm.pid = application_pid;
- queue = NULL;
+ mm_attrs_get_string_by_name(attrs, "profile_uri", &uri);
+ mm_attrs_get_data_by_name(attrs, "profile_user_param", ¶m);
- gst_object_unref (GST_OBJECT(qsrcpad));
- qsrcpad = NULL;
- }
+ ret = __mmfplayer_parse_profile((const char*)uri, param, &player->profile);
- if ( parser )
+ if (ret != MM_ERROR_NONE)
{
- if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state (parser, GST_STATE_PAUSED) )
- {
- debug_error("failed to set state PAUSED to queue\n");
- goto ERROR;
- }
+ debug_error("failed to parse profile\n");
+ return ret;
+ }
- parser = NULL;
+ /* FIXIT : we can use thouse in player->profile directly */
+ if (player->profile.uri_type == MM_PLAYER_URI_TYPE_MEM)
+ {
+ player->mem_buf.buf = (char *)player->profile.mem;
+ player->mem_buf.len = player->profile.mem_size;
+ player->mem_buf.offset = 0;
+ }
- gst_object_unref (GST_OBJECT(pssrcpad));
- pssrcpad = NULL;
+ if (player->profile.uri_type == MM_PLAYER_URI_TYPE_URL_MMS)
+ {
+ debug_warning("mms protocol is not supported format.\n");
+ return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
}
- debug_fleave();
+ if (MMPLAYER_IS_STREAMING(player))
+ MMPLAYER_STATE_CHANGE_TIMEOUT(player) = player->ini.live_state_change_timeout;
+ else
+ MMPLAYER_STATE_CHANGE_TIMEOUT(player) = player->ini.localplayback_state_change_timeout;
- return TRUE;
+ player->needed_v_parser = FALSE;
+ player->smooth_streaming = FALSE;
+ player->videodec_linked = 0;
+ player->videosink_linked = 0;
+ player->audiodec_linked = 0;
+ player->audiosink_linked = 0;
+ player->textsink_linked = 0;
+ player->is_external_subtitle_present = FALSE;
+ /* set the subtitle ON default */
+ player->is_subtitle_off = FALSE;
-ERROR:
+ /* registry should be updated for downloadable codec */
+ mm_attrs_get_int_by_name(attrs, "profile_update_registry", &update_registry);
- if ( queue )
+ if ( update_registry )
{
- gst_object_unref(GST_OBJECT(qsrcpad));
+ debug_log("updating registry...\n");
+ gst_update_registry();
- /* NOTE : Trying to dispose element queue0, but it is in READY instead of the NULL state.
- * You need to explicitly set elements to the NULL state before
- * dropping the final reference, to allow them to clean up.
- */
- gst_element_set_state(queue, GST_STATE_NULL);
- /* And, it still has a parent "player".
- * You need to let the parent manage the object instead of unreffing the object directly.
- */
+ /* then we have to rebuild factories */
+ __mmplayer_release_factories( player );
+ __mmplayer_init_factories(player);
+ }
- gst_bin_remove (GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue);
- //gst_object_unref( queue );
+ /* realize pipeline */
+ ret = __gst_realize( player );
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("fail to realize the player.\n");
+ }
+ else
+ {
+ ret = __mmplayer_realize_streaming_ext(player);
}
- if ( srccaps )
- gst_caps_unref(GST_CAPS(srccaps));
+ MMPLAYER_FLEAVE();
- return FALSE;
+ return ret;
}
-static gboolean __mmplayer_feature_filter(GstPluginFeature *feature, gpointer data) // @
+int
+__mmplayer_unrealize_streaming_ext(mm_player_t *player)
{
- const gchar *klass;
- //const gchar *name;
+ MMPLAYER_FENTER();
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- /* we only care about element factories */
- if (!GST_IS_ELEMENT_FACTORY(feature))
- return FALSE;
-
- /* only parsers, demuxers and decoders */
- klass = gst_element_factory_get_klass(GST_ELEMENT_FACTORY(feature));
- //name = gst_element_factory_get_longname(GST_ELEMENT_FACTORY(feature));
+ /* destroy can called at anytime */
+ if (player->pd_downloader && MMPLAYER_IS_HTTP_PD(player))
+ {
+ _mmplayer_unrealize_pd_downloader ((MMHandleType)player);
+ MMPLAYER_FREEIF(player->pd_downloader);
+ }
- if( g_strrstr(klass, "Demux") == NULL &&
- g_strrstr(klass, "Codec/Decoder") == NULL &&
- g_strrstr(klass, "Depayloader") == NULL &&
- g_strrstr(klass, "Parse") == NULL)
- {
- return FALSE;
- }
- return TRUE;
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
}
-
-static void __mmplayer_add_new_caps(GstPad* pad, GParamSpec* unused, gpointer data)
+int
+_mmplayer_unrealize(MMHandleType hplayer)
{
- mm_player_t* player = (mm_player_t*) data;
- GstCaps *caps = NULL;
- GstStructure *str = NULL;
- const char *name;
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_if_fail ( pad )
- return_if_fail ( unused )
- return_if_fail ( data )
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED )
- caps = gst_pad_get_caps(pad);
- if ( !caps )
- return;
-
- str = gst_caps_get_structure(caps, 0);
- if ( !str )
- return;
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_UNREALIZE );
- name = gst_structure_get_name(str);
- if ( !name )
- return;
- debug_log("name=%s\n", name);
+ __mmplayer_unrealize_streaming_ext(player);
- if ( ! __mmplayer_try_to_plug(player, pad, caps) )
+ /* unrealize pipeline */
+ ret = __gst_unrealize( player );
+
+ /* set asm stop if success */
+ if (MM_ERROR_NONE == ret)
{
- debug_error("failed to autoplug for type (%s)\n", name);
- gst_caps_unref(caps);
- return;
+ if (player->sm.state != ASM_STATE_STOP)
+ {
+ /* NOTE : Stop asm after pipeline unrealize. Keep this sequence. */
+ ret = _mmplayer_asm_set_state(hplayer, ASM_STATE_STOP, FALSE);
+ if ( ret )
+ {
+ debug_error("failed to set asm state to STOP");
+ return ret;
+ }
+ }
+ }
+ else
+ {
+ debug_error("failed and don't change asm state to stop");
}
- gst_caps_unref(caps);
-
- __mmplayer_pipeline_complete( NULL, (gpointer)player );
-
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return;
+ return ret;
}
-static void __mmplayer_set_unlinked_mime_type(mm_player_t* player, GstCaps *caps)
+int
+_mmplayer_set_message_callback(MMHandleType hplayer, MMMessageCallback callback, gpointer user_param) // @
{
- GstStructure *str;
- gint version = 0;
- const char *stream_type;
- gchar *version_field = NULL;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*)hplayer;
- return_if_fail ( player );
- return_if_fail ( caps );
-
- str = gst_caps_get_structure(caps, 0);
- if ( !str )
- return;
-
- stream_type = gst_structure_get_name(str);
- if ( !stream_type )
- return;
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ return __gst_set_message_callback(player, callback, user_param);
+}
- /* set unlinked mime type for downloadable codec */
- if (g_str_has_prefix(stream_type, "video/"))
- {
- if (g_str_has_prefix(stream_type, "video/mpeg"))
- {
- gst_structure_get_int (str, MM_PLAYER_MPEG_VNAME, &version);
- version_field = MM_PLAYER_MPEG_VNAME;
- }
- else if (g_str_has_prefix(stream_type, "video/x-wmv"))
- {
- gst_structure_get_int (str, MM_PLAYER_WMV_VNAME, &version);
- version_field = MM_PLAYER_WMV_VNAME;
-
- }
- else if (g_str_has_prefix(stream_type, "video/x-divx"))
- {
- gst_structure_get_int (str, MM_PLAYER_DIVX_VNAME, &version);
- version_field = MM_PLAYER_DIVX_VNAME;
- }
+int
+_mmplayer_get_state(MMHandleType hplayer, int* state) // @
+{
+ mm_player_t *player = (mm_player_t*)hplayer;
- if (version)
- {
- player->unlinked_video_mime = g_strdup_printf("%s, %s=%d", stream_type, version_field, version);
- }
- else
- {
- player->unlinked_video_mime = g_strdup_printf("%s", stream_type);
- }
- }
- else if (g_str_has_prefix(stream_type, "audio/"))
- {
- if (g_str_has_prefix(stream_type, "audio/mpeg")) // mp3 or aac
- {
- gst_structure_get_int (str, MM_PLAYER_MPEG_VNAME, &version);
- version_field = MM_PLAYER_MPEG_VNAME;
- }
- else if (g_str_has_prefix(stream_type, "audio/x-wma"))
- {
- gst_structure_get_int (str, MM_PLAYER_WMA_VNAME, &version);
- version_field = MM_PLAYER_WMA_VNAME;
- }
+ return_val_if_fail(state, MM_ERROR_INVALID_ARGUMENT);
- if (version)
- {
- player->unlinked_audio_mime = g_strdup_printf("%s, %s=%d", stream_type, version_field, version);
- }
- else
- {
- player->unlinked_audio_mime = g_strdup_printf("%s", stream_type);
- }
- }
+ *state = MMPLAYER_CURRENT_STATE(player);
- debug_fleave();
+ return MM_ERROR_NONE;
}
-static void __mmplayer_add_new_pad(GstElement *element, GstPad *pad, gpointer data)
+
+int
+_mmplayer_set_volume(MMHandleType hplayer, MMPlayerVolumeType volume) // @
{
- mm_player_t* player = (mm_player_t*) data;
- GstCaps *caps = NULL;
- GstStructure *str = NULL;
- const char *name;
+ mm_player_t* player = (mm_player_t*) hplayer;
+ GstElement* vol_element = NULL;
+ int i = 0;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_if_fail ( player );
- return_if_fail ( pad );
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- GST_OBJECT_LOCK (pad);
- if ((caps = GST_PAD_CAPS(pad)))
- gst_caps_ref(caps);
- GST_OBJECT_UNLOCK (pad);
+ debug_log("volume [L]=%f:[R]=%f\n",
+ volume.level[MM_VOLUME_CHANNEL_LEFT], volume.level[MM_VOLUME_CHANNEL_RIGHT]);
- if ( NULL == caps )
+ /* invalid factor range or not */
+ for ( i = 0; i < MM_VOLUME_CHANNEL_NUM; i++ )
{
- caps = gst_pad_get_caps(pad);
- if ( !caps ) return;
+ if (volume.level[i] < MM_VOLUME_FACTOR_MIN || volume.level[i] > MM_VOLUME_FACTOR_MAX) {
+ debug_error("Invalid factor! (valid factor:0~1.0)\n");
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
}
- MMPLAYER_LOG_GST_CAPS_TYPE(caps);
-
- str = gst_caps_get_structure(caps, 0);
- if ( !str )
- return;
-
- name = gst_structure_get_name(str);
- if ( !name )
- return;
+ /* not support to set other value into each channel */
+ if ((volume.level[MM_VOLUME_CHANNEL_LEFT] != volume.level[MM_VOLUME_CHANNEL_RIGHT]))
+ return MM_ERROR_INVALID_ARGUMENT;
- player->num_dynamic_pad++;
- debug_log("stream count inc : %d\n", player->num_dynamic_pad);
+ /* Save volume to handle. Currently the first array element will be saved. */
+ player->sound.volume = volume.level[MM_VOLUME_CHANNEL_LEFT];
- if (__mmplayer_is_amr_type(name))
- {
- /* store type string */
- MMPLAYER_FREEIF(player->type);
- player->type = gst_caps_to_string(caps);
- player->bypass_sound_effect = FALSE;
- if ( (PLAYER_INI()->use_audio_filter_preset || PLAYER_INI()->use_audio_filter_custom) )
- {
- if ( player->audio_filter_info.filter_type == MM_AUDIO_FILTER_TYPE_PRESET )
- {
- if (!_mmplayer_sound_filter_preset_apply(player, player->audio_filter_info.preset))
- {
- debug_msg("apply sound effect(preset:%d) setting success\n",player->audio_filter_info.preset);
- }
- }
- else if ( player->audio_filter_info.filter_type == MM_AUDIO_FILTER_TYPE_CUSTOM )
- {
- if (!_mmplayer_sound_filter_custom_apply(player))
- {
- debug_msg("apply sound effect(custom) setting success\n");
- }
- }
- }
- }
- /* Note : If the stream is the subtitle, we try not to play it. Just close the demuxer subtitle pad.
- * If want to play it, remove this code.
- */
- else if (g_strrstr(name, "application"))
- {
- if (g_strrstr(name, "x-id3") || g_strrstr(name, "x-apetag"))
- {
- /* If id3/ape tag comes, keep going */
- debug_log("application mime exception : id3/ape tag\n");
- }
- else
- {
- /* Otherwise, we assume that this stream is subtile. */
- debug_log(" application mime type pad is closed. \n");
- return;
- }
- }
- else if (g_strrstr(name, "video"))
+ /* check pipeline handle */
+ if ( ! player->pipeline || ! player->pipeline->audiobin )
{
- int stype;
- mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &stype);
-
- /* don't make video because of not required */
- if (stype == MM_DISPLAY_SURFACE_NULL)
- {
- debug_log("no video because it's not required\n");
- return;
- }
+ debug_log("audiobin is not created yet\n");
+ debug_log("but, current stored volume will be set when it's created.\n");
- player->v_stream_caps = gst_caps_copy(caps); //if needed, video caps is required when videobin is created
+ /* NOTE : stored volume will be used in create_audiobin
+ * returning MM_ERROR_NONE here makes application to able to
+ * set volume at anytime.
+ */
+ return MM_ERROR_NONE;
}
- if ( ! __mmplayer_try_to_plug(player, pad, caps) )
+ /* setting volume to volume element */
+ vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+
+ if ( vol_element )
{
- debug_error("failed to autoplug for type (%s)\n", name);
-
- __mmplayer_set_unlinked_mime_type(player, caps);
+ debug_log("volume is set [%f]\n", player->sound.volume);
+ g_object_set(vol_element, "volume", player->sound.volume, NULL);
}
- gst_caps_unref(caps);
-
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return;
+ return MM_ERROR_NONE;
}
-/* test API for tuning audio gain. this API should be
- * deprecated before the day of final release
- */
+
int
-_mmplayer_set_volume_tune(MMHandleType hplayer, MMPlayerVolumeType volume)
+_mmplayer_get_volume(MMHandleType hplayer, MMPlayerVolumeType* volume)
{
mm_player_t* player = (mm_player_t*) hplayer;
- gint error = MM_ERROR_NONE;
- gint vol_max = 0;
- gboolean isMidi = FALSE;
- gint i = 0;
+ int i = 0;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail( player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED )
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( volume, MM_ERROR_INVALID_ARGUMENT );
- debug_log("clip type=%d(1-midi, 0-others), volume [L]=%d:[R]=%d\n",
- player->profile.play_mode, volume.level[0], volume.level[1]);
+ /* returning stored volume */
+ for (i = 0; i < MM_VOLUME_CHANNEL_NUM; i++)
+ volume->level[i] = player->sound.volume;
- isMidi = ( player->profile.play_mode == MM_PLAYER_MODE_MIDI ) ? TRUE : FALSE;
+ MMPLAYER_FLEAVE();
- if ( isMidi )
- vol_max = 1000;
- else
- vol_max = 100;
+ return MM_ERROR_NONE;
+}
- /* is it proper volume level? */
- for (i = 0; i < MM_VOLUME_CHANNEL_NUM; ++i)
+
+
+int
+_mmplayer_set_mute(MMHandleType hplayer, int mute) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ GstElement* vol_element = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* mute value shoud 0 or 1 */
+ if ( mute != 0 && mute != 1 )
{
- if (volume.level[i] < 0 || volume.level[i] > vol_max) {
- debug_log("Invalid Volume level!!!! \n");
- return MM_ERROR_INVALID_ARGUMENT;
- }
+ debug_error("bad mute value\n");
+
+ /* FIXIT : definitly, we need _BAD_PARAM error code */
+ return MM_ERROR_INVALID_ARGUMENT;
}
- if ( isMidi )
+ player->sound.mute = mute;
+
+ /* just hold mute value if pipeline is not ready */
+ if ( !player->pipeline || !player->pipeline->audiobin )
{
- if ( player->pipeline->mainbin )
- {
- GstElement *midi_element = player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst;
+ debug_log("pipeline is not ready. holding mute value\n");
+ return MM_ERROR_NONE;
+ }
- if ( midi_element && ( strstr(GST_ELEMENT_NAME(midi_element), "midiparse")) )
- {
- debug_log("setting volume (%d) level to midi plugin\n", volume.level[0]);
+ vol_element = player->pipeline->audiobin[MMPLAYER_A_SINK].gst;
- g_object_set(midi_element, "volume", volume.level[0], NULL);
- }
- }
+ /* NOTE : volume will only created when the bt is enabled */
+ if ( vol_element )
+ {
+ debug_log("mute : %d\n", mute);
+ g_object_set(vol_element, "mute", mute, NULL);
}
else
{
- if ( player->pipeline->audiobin )
- {
- GstElement *sink_element = player->pipeline->audiobin[MMPLAYER_A_SINK].gst;
+ debug_log("volume elemnet is not created. using volume in audiosink\n");
+ }
- /* Set to Avsysaudiosink element */
- if ( sink_element )
- {
- gint vol_value = 0;
- gboolean mute = FALSE;
- vol_value = volume.level[0];
+ MMPLAYER_FLEAVE();
- g_object_set(G_OBJECT(sink_element), "tuningvolume", vol_value, NULL);
+ return MM_ERROR_NONE;
+}
- mute = (vol_value == 0)? TRUE:FALSE;
+int
+_mmplayer_get_mute(MMHandleType hplayer, int* pmute) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
- g_object_set(G_OBJECT(sink_element), "mute", mute, NULL);
- }
+ MMPLAYER_FENTER();
- }
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( pmute, MM_ERROR_INVALID_ARGUMENT );
+
+ /* just hold mute value if pipeline is not ready */
+ if ( !player->pipeline || !player->pipeline->audiobin )
+ {
+ debug_log("pipeline is not ready. returning stored value\n");
+ *pmute = player->sound.mute;
+ return MM_ERROR_NONE;
}
- debug_fleave();
+ *pmute = player->sound.mute;
- return error;
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
}
-gboolean
-__mmplayer_dump_pipeline_state( mm_player_t* player )
+int
+_mmplayer_set_videostream_cb(MMHandleType hplayer, mm_player_video_stream_callback callback, void *user_param) // @
{
- GstIterator*iter = NULL;
- gboolean done = FALSE;
+ mm_player_t* player = (mm_player_t*) hplayer;
- GstElement *item = NULL;
- GstElementFactory *factory = NULL;
+ MMPLAYER_FENTER();
- GstState state = GST_STATE_VOID_PENDING;
- GstState pending = GST_STATE_VOID_PENDING;
- GstClockTime time = 200*GST_MSECOND;
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- debug_fenter();
+ player->video_stream_cb = callback;
+ player->video_stream_cb_user_param = user_param;
+ player->use_video_stream = TRUE;
+ debug_log("Stream cb Handle value is %p : %p\n", player, player->video_stream_cb);
- return_val_if_fail ( player &&
- player->pipeline &&
- player->pipeline->mainbin,
- FALSE );
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+}
- iter = gst_bin_iterate_recurse(GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst) );
+int
+_mmplayer_set_audiostream_cb(MMHandleType hplayer, mm_player_audio_stream_callback callback, void *user_param) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
- if ( iter != NULL )
- {
- while (!done) {
- switch ( gst_iterator_next (iter, (gpointer)&item) )
- {
- case GST_ITERATOR_OK:
- gst_element_get_state(GST_ELEMENT (item),&state, &pending,time);
+ MMPLAYER_FENTER();
- factory = gst_element_get_factory (item) ;
- debug_log("%s:%s : From:%s To:%s refcount : %d\n", GST_OBJECT_NAME(factory) , GST_ELEMENT_NAME(item) ,
- gst_element_state_get_name(state), gst_element_state_get_name(pending) , GST_OBJECT_REFCOUNT_VALUE(item));
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ player->audio_stream_cb = callback;
+ player->audio_stream_cb_user_param = user_param;
+ debug_log("Audio Stream cb Handle value is %p : %p\n", player, player->audio_stream_cb);
- gst_object_unref (item);
- break;
- case GST_ITERATOR_RESYNC:
- gst_iterator_resync (iter);
- break;
- case GST_ITERATOR_ERROR:
- done = TRUE;
- break;
- case GST_ITERATOR_DONE:
- done = TRUE;
- break;
- }
- }
- }
+ MMPLAYER_FLEAVE();
- item = GST_ELEMENT(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst);
+ return MM_ERROR_NONE;
+}
- gst_element_get_state(GST_ELEMENT (item),&state, &pending,time);
+int
+_mmplayer_set_buffer_need_data_cb(MMHandleType hplayer, mm_player_buffer_need_data_callback callback, void *user_param) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
- factory = gst_element_get_factory (item) ;
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
- debug_log("%s:%s : From:%s To:%s refcount : %d\n",
- GST_OBJECT_NAME(factory),
- GST_ELEMENT_NAME(item),
- gst_element_state_get_name(state),
- gst_element_state_get_name(pending),
- GST_OBJECT_REFCOUNT_VALUE(item) );
+ player->need_data_cb = callback;
+ player->buffer_cb_user_param = user_param;
- if ( iter )
- gst_iterator_free (iter);
+ debug_log("buffer need dataHandle value is %p : %p\n", player, player->need_data_cb);
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return FALSE;
+ return MM_ERROR_NONE;
}
-
-gboolean
-__mmplayer_check_subtitle( mm_player_t* player )
+int
+_mmplayer_set_buffer_enough_data_cb(MMHandleType hplayer, mm_player_buffer_enough_data_callback callback, void *user_param) // @
{
- MMHandleType attrs = 0;
- char *subtitle_uri = NULL;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*) hplayer;
- return_val_if_fail( player, FALSE );
+ MMPLAYER_FENTER();
- /* get subtitle attribute */
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
- return FALSE;
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
- mm_attrs_get_string_by_name(attrs, "subtitle_uri", &subtitle_uri);
- if ( !subtitle_uri || !strlen(subtitle_uri))
- return FALSE;
+ player->enough_data_cb = callback;
+ player->buffer_cb_user_param = user_param;
- debug_log ("subtite uri is %s[%d]\n", subtitle_uri, strlen(subtitle_uri));
+ debug_log("buffer enough data cb Handle value is %p : %p\n", player, player->enough_data_cb);
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return TRUE;
+ return MM_ERROR_NONE;
}
-static gboolean
-__mmplayer_can_extract_pcm( mm_player_t* player )
+int
+_mmplayer_set_buffer_seek_data_cb(MMHandleType hplayer, mm_player_buffer_seek_data_callback callback, void *user_param) // @
{
- MMHandleType attrs = 0;
- gboolean is_drm = FALSE;
- gboolean sound_extraction = FALSE;
+ mm_player_t* player = (mm_player_t*) hplayer;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
- {
- debug_error("fail to get attributes.");
- return FALSE;
- }
-
- /* check file is drm or not */
- g_object_get(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_SRC].gst), "is-drm", &is_drm, NULL);
+ player->seek_data_cb = callback;
+ player->buffer_cb_user_param = user_param;
- /* get sound_extraction property */
- mm_attrs_get_int_by_name(attrs, "pcm_extraction", &sound_extraction);
+ debug_log("buffer seek data cb Handle value is %p : %p\n", player, player->seek_data_cb);
- if ( ! sound_extraction || is_drm )
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
+
+int
+__mmplayer_start_streaming_ext(mm_player_t *player)
+{
+ gint ret = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ if (MMPLAYER_IS_HTTP_PD(player))
{
- debug_log("pcm extraction param.. is drm = %d, extraction mode = %d", is_drm, sound_extraction);
- return FALSE;
- }
+ if ( !player->pd_downloader )
+ {
+ ret = __mmplayer_realize_streaming_ext(player);
- debug_fleave();
+ if ( ret != MM_ERROR_NONE)
+ {
+ debug_error ("failed to realize streaming ext\n");
+ return ret;
+ }
+ }
- return TRUE;
+ if (player->pd_downloader && player->pd_mode == MM_PLAYER_PD_MODE_URI)
+ {
+ ret = _mmplayer_start_pd_downloader ((MMHandleType)player);
+ if ( !ret )
+ {
+ debug_error ("ERROR while starting PD...\n");
+ return MM_ERROR_PLAYER_NOT_INITIALIZED;
+ }
+ ret = MM_ERROR_NONE;
+ }
+ }
+
+ MMPLAYER_FLEAVE();
+ return ret;
}
-static gboolean
-__mmplayer_handle_gst_error ( mm_player_t* player, GstMessage * message, GError* error )
+int
+_mmplayer_start(MMHandleType hplayer) // @
{
- MMMessageParamType msg_param;
- gchar *msg_src_element;
-
- debug_fenter();
+ mm_player_t* player = (mm_player_t*) hplayer;
+ gint ret = MM_ERROR_NONE;
- return_val_if_fail( player, FALSE );
- return_val_if_fail( error, FALSE );
+ MMPLAYER_FENTER();
- /* NOTE : do somthing necessary inside of __gst_handle_XXX_error. not here */
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- memset (&msg_param, 0, sizeof(MMMessageParamType));
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_START );
- if ( error->domain == GST_CORE_ERROR )
+ ret = _mmplayer_asm_set_state(hplayer, ASM_STATE_PLAYING, TRUE);
+ if ( ret != MM_ERROR_NONE )
{
- msg_param.code = __gst_handle_core_error( player, error->code );
+ debug_error("failed to set asm state to PLAYING\n");
+ return ret;
}
- else if ( error->domain == GST_LIBRARY_ERROR )
+
+ /* NOTE : we should check and create pipeline again if not created as we destroy
+ * whole pipeline when stopping in streamming playback
+ */
+ if ( ! player->pipeline )
{
- msg_param.code = __gst_handle_library_error( player, error->code );
- }
- else if ( error->domain == GST_RESOURCE_ERROR )
+ ret = __gst_realize( player );
+ if ( MM_ERROR_NONE != ret )
+ {
+ debug_error("failed to realize before starting. only in streamming\n");
+ /* unlock */
+ return ret;
+ }
+ }
+
+ ret = __mmplayer_start_streaming_ext(player);
+ if ( ret != MM_ERROR_NONE )
{
- msg_param.code = __gst_handle_resource_error( player, error->code );
+ debug_error("failed to start streaming ext \n");
}
- else if ( error->domain == GST_STREAM_ERROR )
+
+ /* start pipeline */
+ ret = __gst_start( player );
+ if ( ret != MM_ERROR_NONE )
{
- msg_param.code = __gst_handle_stream_error( player, error, message );
+ debug_error("failed to start player.\n");
}
- else
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
+}
+
+/* NOTE: post "not supported codec message" to application
+ * when one codec is not found during AUTOPLUGGING in MSL.
+ * So, it's separated with error of __mmplayer_gst_callback().
+ * And, if any codec is not found, don't send message here.
+ * Because GST_ERROR_MESSAGE is posted by other plugin internally.
+ */
+int
+__mmplayer_handle_missed_plugin(mm_player_t* player)
+{
+ MMMessageParamType msg_param;
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
+ gboolean post_msg_direct = FALSE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ debug_log("not_supported_codec = 0x%02x, can_support_codec = 0x%02x\n",
+ player->not_supported_codec, player->can_support_codec);
+
+ if( player->not_found_demuxer )
{
- debug_warning("This error domain is not defined.\n");
+ msg_param.code = MM_ERROR_PLAYER_CODEC_NOT_FOUND;
+ msg_param.data = g_strdup_printf("%s", player->unlinked_demuxer_mime);
- /* we treat system error as an internal error */
- msg_param.code = MM_ERROR_PLAYER_INVALID_STREAM;
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+ MMPLAYER_FREEIF(msg_param.data);
+
+ return MM_ERROR_NONE;
}
- if ( message->src )
+ if (player->not_supported_codec)
{
- msg_src_element = GST_ELEMENT_NAME( GST_ELEMENT_CAST( message->src ) );
+ if ( player->can_support_codec ) // There is one codec to play
+ {
+ post_msg_direct = TRUE;
+ }
+ else
+ {
+ if ( player->pipeline->audiobin ) // Some content has only PCM data in container.
+ post_msg_direct = TRUE;
+ }
- msg_param.data = (void *) error->message;
+ if ( post_msg_direct )
+ {
+ MMMessageParamType msg_param;
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
- debug_error("-Msg src : [%s] Domain : [%s] Error : [%s] Code : [%d] is tranlated to error code : [0x%x]\n",
- msg_src_element, g_quark_to_string (error->domain), error->message, error->code, msg_param.code);
+ if ( player->not_supported_codec == MISSING_PLUGIN_AUDIO )
+ {
+ debug_warning("not found AUDIO codec, posting error code to application.\n");
+
+ msg_param.code = MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
+ msg_param.data = g_strdup_printf("%s", player->unlinked_audio_mime);
+ }
+ else if ( player->not_supported_codec == MISSING_PLUGIN_VIDEO )
+ {
+ debug_warning("not found VIDEO codec, posting error code to application.\n");
+
+ msg_param.code = MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
+ msg_param.data = g_strdup_printf("%s", player->unlinked_video_mime);
+ }
+
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+
+ MMPLAYER_FREEIF(msg_param.data);
+
+ return MM_ERROR_NONE;
+ }
+ else // no any supported codec case
+ {
+ debug_warning("not found any codec, posting error code to application.\n");
+
+ if ( player->not_supported_codec == MISSING_PLUGIN_AUDIO )
+ {
+ msg_param.code = MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
+ msg_param.data = g_strdup_printf("%s", player->unlinked_audio_mime);
+ }
+ else
+ {
+ msg_param.code = MM_ERROR_PLAYER_CODEC_NOT_FOUND;
+ msg_param.data = g_strdup_printf("%s, %s", player->unlinked_video_mime, player->unlinked_audio_mime);
+ }
+
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+
+ MMPLAYER_FREEIF(msg_param.data);
+ }
}
- if (MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
+
+static void __mmplayer_check_pipeline(mm_player_t* player)
+{
+ GstState element_state = GST_STATE_VOID_PENDING;
+ GstState element_pending_state = GST_STATE_VOID_PENDING;
+ gint timeout = 0;
+ int ret = MM_ERROR_NONE;
+
+ if (player->pp_rebuilding)
{
- __mm_player_ahs_stop (player->ahs_player);
+ debug_warning("pipeline is under construction.\n");
+
+ MMPLAYER_PLAYBACK_LOCK(player);
+ MMPLAYER_PLAYBACK_UNLOCK(player);
+
+ timeout = MMPLAYER_STATE_CHANGE_TIMEOUT ( player );
+
+ /* wait for state transition */
+ ret = gst_element_get_state( player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &element_state, &element_pending_state, timeout * GST_SECOND );
+
+ if ( ret == GST_STATE_CHANGE_FAILURE )
+ {
+ debug_error("failed to change pipeline state within %d sec\n", timeout );
+ }
}
+}
- /* post error to application */
- if ( ! player->posted_msg )
+/* NOTE : it should be able to call 'stop' anytime*/
+int
+_mmplayer_stop(MMHandleType hplayer) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_STOP );
+
+ /* check pipline building state */
+ __mmplayer_check_pipeline(player);
+
+ /* NOTE : application should not wait for EOS after calling STOP */
+ __mmplayer_cancel_eos_timer( player );
+
+ __mmplayer_unrealize_streaming_ext(player);
+
+ /* reset */
+ player->doing_seek = FALSE;
+
+ /* stop pipeline */
+ ret = __gst_stop( player );
+
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to stop player.\n");
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
+}
+
+int
+_mmplayer_pause(MMHandleType hplayer) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint64 pos_msec = 0;
+ gboolean async = FALSE;
+ gint ret = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_PAUSE );
+
+ /* check pipline building state */
+ __mmplayer_check_pipeline(player);
+
+ switch (MMPLAYER_CURRENT_STATE(player))
{
- if (msg_param.code == MM_MESSAGE_DRM_NOT_AUTHORIZED)
+ case MM_PLAYER_STATE_READY:
{
- MMPLAYER_POST_MSG( player, MM_MESSAGE_DRM_NOT_AUTHORIZED, NULL );
+ /* check prepare async or not.
+ * In the case of streaming playback, it's recommned to avoid blocking wait.
+ */
+ mm_attrs_get_int_by_name(player->attrs, "profile_prepare_async", &async);
+ debug_log("prepare working mode : %s", (async ? "async" : "sync"));
}
- else
+ break;
+
+ case MM_PLAYER_STATE_PLAYING:
{
- MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
- }
+ /* NOTE : store current point to overcome some bad operation
+ * ( returning zero when getting current position in paused state) of some
+ * elements
+ */
+ if ( !gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &pos_msec))
+ debug_warning("getting current position failed in paused\n");
- /* don't post more if one was sent already */
- player->posted_msg = TRUE;
+ player->last_position = pos_msec;
+ }
+ break;
}
- else
+
+ /* pause pipeline */
+ ret = __gst_pause( player, async );
+
+ if ( ret != MM_ERROR_NONE )
{
- debug_log("skip error post because it's sent already.\n");
+ debug_error("failed to pause player. ret : 0x%x\n", ret);
}
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return TRUE;
+ return ret;
}
-static gboolean
-__mmplayer_handle_streaming_error ( mm_player_t* player, GstMessage * message )
+int
+_mmplayer_resume(MMHandleType hplayer)
{
- debug_log("\n");
- MMMessageParamType msg_param;
- gchar *msg_src_element = NULL;
- GstStructure *s = NULL;
- guint error_id = 0;
- gchar *error_string = NULL;
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+ gboolean async = FALSE;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail ( player, FALSE );
- return_val_if_fail ( message, FALSE );
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- s = malloc( sizeof(GstStructure) );
- memcpy ( s, gst_message_get_structure ( message ), sizeof(GstStructure));
+ ret = _mmplayer_asm_set_state(hplayer, ASM_STATE_PLAYING, TRUE);
+ if ( ret )
+ {
+ debug_error("failed to set asm state to PLAYING\n");
+ return ret;
+ }
- if ( !gst_structure_get_uint (s, "error_id", &error_id) )
- error_id = MMPLAYER_STREAMING_ERROR_NONE;
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_RESUME );
- switch ( error_id )
+ /* resume pipeline */
+ ret = __gst_resume( player, async );
+
+ if ( ret != MM_ERROR_NONE )
{
- case MMPLAYER_STREAMING_ERROR_UNSUPPORTED_AUDIO:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_UNSUPPORTED_AUDIO;
- break;
- case MMPLAYER_STREAMING_ERROR_UNSUPPORTED_VIDEO:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_UNSUPPORTED_VIDEO;
- break;
- case MMPLAYER_STREAMING_ERROR_CONNECTION_FAIL:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_CONNECTION_FAIL;
- break;
- case MMPLAYER_STREAMING_ERROR_DNS_FAIL:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_DNS_FAIL;
- break;
- case MMPLAYER_STREAMING_ERROR_SERVER_DISCONNECTED:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_SERVER_DISCONNECTED;
- break;
- case MMPLAYER_STREAMING_ERROR_BAD_SERVER:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_BAD_SERVER;
- break;
- case MMPLAYER_STREAMING_ERROR_INVALID_PROTOCOL:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_INVALID_PROTOCOL;
- break;
- case MMPLAYER_STREAMING_ERROR_INVALID_URL:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_INVALID_URL;
- break;
- case MMPLAYER_STREAMING_ERROR_UNEXPECTED_MSG:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_UNEXPECTED_MSG;
- break;
- case MMPLAYER_STREAMING_ERROR_OUT_OF_MEMORIES:
- msg_param.code = MM_ERROR_PLAYER_STREAMING_OUT_OF_MEMORIES;
- break;
+ debug_error("failed to resume player.\n");
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
+}
+
+int
+__mmplayer_set_play_count(mm_player_t* player, gint count)
+{
+ MMHandleType attrs = 0;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("fail to get attributes.\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
+ if ( mmf_attrs_commit ( attrs ) ) /* return -1 if error */
+ debug_error("failed to commit\n");
+
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_activate_section_repeat(MMHandleType hplayer, unsigned long start, unsigned long end)
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ gint64 start_pos = 0;
+ gint64 end_pos = 0;
+ gint infinity = -1;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( end <= GST_TIME_AS_MSECONDS(player->duration), MM_ERROR_INVALID_ARGUMENT );
+
+ player->section_repeat = TRUE;
+ player->section_repeat_start = start;
+ player->section_repeat_end = end;
+
+ start_pos = player->section_repeat_start * G_GINT64_CONSTANT(1000000);
+ end_pos = player->section_repeat_end * G_GINT64_CONSTANT(1000000);
+
+ __mmplayer_set_play_count( player, infinity );
+
+ if ( (!__gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
+ player->playback_rate,
+ GST_FORMAT_TIME,
+ ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
+ GST_SEEK_TYPE_SET, start_pos,
+ GST_SEEK_TYPE_SET, end_pos)))
+ {
+ debug_error("failed to activate section repeat\n");
+
+ return MM_ERROR_PLAYER_SEEK;
+ }
+
+ debug_log("succeeded to set section repeat from %d to %d\n",
+ player->section_repeat_start, player->section_repeat_end);
+
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
+
+static int
+__mmplayer_set_pcm_extraction(mm_player_t* player)
+{
+ gint64 start_nsec = 0;
+ gint64 end_nsec = 0;
+ gint64 dur_nsec = 0;
+ gint64 dur_msec = 0;
+ GstFormat fmt = GST_FORMAT_TIME;
+ int required_start = 0;
+ int required_end = 0;
+ int ret = 0;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, FALSE );
+
+ mm_attrs_multiple_get(player->attrs,
+ NULL,
+ "pcm_extraction_start_msec", &required_start,
+ "pcm_extraction_end_msec", &required_end,
+ NULL);
+
+ debug_log("pcm extraction required position is from [%d] to [%d] (msec)\n", required_start, required_end);
+
+ if (required_start == 0 && required_end == 0)
+ {
+ debug_log("extracting entire stream");
+ return MM_ERROR_NONE;
+ }
+ else if (required_start < 0 || required_start > required_end || required_end < 0 )
+ {
+ debug_log("invalid range for pcm extraction");
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
+
+ /* get duration */
+ ret = gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &dur_nsec);
+ if ( !ret )
+ {
+ debug_error("failed to get duration");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ dur_msec = GST_TIME_AS_MSECONDS(dur_nsec);
+
+ if (dur_msec < required_end) // FIXME
+ {
+ debug_log("invalid end pos for pcm extraction");
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
+
+ start_nsec = required_start * G_GINT64_CONSTANT(1000000);
+ end_nsec = required_end * G_GINT64_CONSTANT(1000000);
+
+ if ( (!__gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
+ 1.0,
+ GST_FORMAT_TIME,
+ ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
+ GST_SEEK_TYPE_SET, start_nsec,
+ GST_SEEK_TYPE_SET, end_nsec)))
+ {
+ debug_error("failed to seek for pcm extraction\n");
+
+ return MM_ERROR_PLAYER_SEEK;
+ }
+
+ debug_log("succeeded to set up segment extraction from [%llu] to [%llu] (nsec)\n", start_nsec, end_nsec);
+
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_deactivate_section_repeat(MMHandleType hplayer)
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ gint64 cur_pos = 0;
+ GstFormat fmt = GST_FORMAT_TIME;
+ gint onetime = 1;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ player->section_repeat = FALSE;
+
+ __mmplayer_set_play_count( player, onetime );
+
+ gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &cur_pos);
+
+ if ( (!__gst_seek( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
+ 1.0,
+ GST_FORMAT_TIME,
+ ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
+ GST_SEEK_TYPE_SET, cur_pos,
+ GST_SEEK_TYPE_SET, player->duration )))
+ {
+ debug_error("failed to deactivate section repeat\n");
+
+ return MM_ERROR_PLAYER_SEEK;
+ }
+
+ MMPLAYER_FENTER();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_set_playspeed(MMHandleType hplayer, float rate)
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ signed long long pos_msec = 0;
+ int ret = MM_ERROR_NONE;
+ int mute = FALSE;
+ GstFormat format =GST_FORMAT_TIME;
+ MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( !MMPLAYER_IS_STREAMING(player), MM_ERROR_NOT_SUPPORT_API );
+
+ /* The sound of video is not supported under 0.0 and over 2.0. */
+ if(rate >= TRICK_PLAY_MUTE_THRESHOLD_MAX || rate < TRICK_PLAY_MUTE_THRESHOLD_MIN)
+ {
+ if (player->can_support_codec & FOUND_PLUGIN_VIDEO)
+ mute = TRUE;
+ }
+ _mmplayer_set_mute(hplayer, mute);
+
+ if (player->playback_rate == rate)
+ return MM_ERROR_NONE;
+
+ /* If the position is reached at start potion during fast backward, EOS is posted.
+ * So, This EOS have to be classified with it which is posted at reaching the end of stream.
+ * */
+ player->playback_rate = rate;
+
+ current_state = MMPLAYER_CURRENT_STATE(player);
+
+ if ( current_state != MM_PLAYER_STATE_PAUSED )
+ ret = gst_element_query_position(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &format, &pos_msec);
+
+ debug_log ("pos_msec = %"GST_TIME_FORMAT" and ret = %d and state = %d", GST_TIME_ARGS (pos_msec), ret, current_state);
+
+ if ( ( current_state == MM_PLAYER_STATE_PAUSED )
+ || ( ! ret ))
+ //|| ( player->last_position != 0 && pos_msec == 0 ) )
+ {
+ debug_warning("returning last point : %lld\n", player->last_position );
+ pos_msec = player->last_position;
+ }
+
+ if ((!gst_element_seek (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst,
+ rate,
+ GST_FORMAT_TIME,
+ ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
+ //( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE | GST_SEEK_FLAG_KEY_UNIT),
+ GST_SEEK_TYPE_SET, pos_msec,
+ //GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE,
+ GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE)))
+ {
+ debug_error("failed to set speed playback\n");
+ return MM_ERROR_PLAYER_SEEK;
+ }
+
+ debug_log("succeeded to set speed playback as %0.1f\n", rate);
+
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;;
+}
+
+int
+_mmplayer_set_position(MMHandleType hplayer, int format, int position) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ ret = __gst_set_position ( player, format, (unsigned long)position, FALSE );
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
+}
+
+int
+_mmplayer_get_position(MMHandleType hplayer, int format, unsigned long *position) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ ret = __gst_get_position ( player, format, position );
+
+ return ret;
+}
+
+int
+_mmplayer_get_buffer_position(MMHandleType hplayer, int format, unsigned long* start_pos, unsigned long* stop_pos) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ ret = __gst_get_buffer_position ( player, format, start_pos, stop_pos );
+
+ return ret;
+}
+
+int
+_mmplayer_adjust_subtitle_postion(MMHandleType hplayer, int format, int position) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ ret = __gst_adjust_subtitle_position(player, format, position);
+
+ MMPLAYER_FLEAVE();
+
+ return ret;
+}
+
+static gboolean
+__mmplayer_is_midi_type( gchar* str_caps)
+{
+ if ( ( g_strrstr(str_caps, "audio/midi") ) ||
+ ( g_strrstr(str_caps, "application/x-gst_ff-mmf") ) ||
+ ( g_strrstr(str_caps, "application/x-smaf") ) ||
+ ( g_strrstr(str_caps, "audio/x-imelody") ) ||
+ ( g_strrstr(str_caps, "audio/mobile-xmf") ) ||
+ ( g_strrstr(str_caps, "audio/xmf") ) ||
+ ( g_strrstr(str_caps, "audio/mxmf") ) )
+ {
+ debug_log("midi\n");
+
+ return TRUE;
+ }
+
+ return FALSE;
+}
+
+static gboolean
+__mmplayer_is_only_mp3_type (gchar *str_caps)
+{
+ if (g_strrstr(str_caps, "application/x-id3") ||
+ (g_strrstr(str_caps, "audio/mpeg") && g_strrstr(str_caps, "mpegversion=(int)1")))
+ {
+ return TRUE;
+ }
+ return FALSE;
+}
+
+/* try to use ALP decoder first instead of selected decoder */
+static gboolean
+__mmplayer_is_omx_decoder_type (mm_player_t* player)
+{
+#define MIN_THRESHOLD_SIZE 320 * 1024 // 320K
+ gboolean ret = FALSE;
+ gchar* path = NULL;
+ guint64 data_size = 0;
+ struct stat sb;
+
+ return_val_if_fail (player, FALSE);
+
+ /* consider mp3 audio only */
+ if ((!MMPLAYER_IS_STREAMING(player)) &&
+ (__mmplayer_is_only_mp3_type(player->type)))
+ {
+ mm_attrs_get_string_by_name(player->attrs, "profile_uri", &path);
+
+ if (stat(path, &sb) == 0)
+ {
+ data_size = (guint64)sb.st_size;
+
+ if (data_size > MIN_THRESHOLD_SIZE)
+ {
+ ret = TRUE;
+ }
+ }
+ }
+
+ debug_log ("need to select omx_decoder ? [%s]\n", (ret)?"YES":"NO");
+ return ret;
+}
+
+static void
+__mmplayer_set_audio_attrs (mm_player_t* player, GstCaps* caps)
+{
+ GstStructure* caps_structure = NULL;
+ gint samplerate = 0;
+ gint channels = 0;
+
+ MMPLAYER_FENTER();
+ return_if_fail (player && caps);
+
+ caps_structure = gst_caps_get_structure(caps, 0);
+
+ /* set stream information */
+ gst_structure_get_int (caps_structure, "rate", &samplerate);
+ mm_attrs_set_int_by_name (player->attrs, "content_audio_samplerate", samplerate);
+
+ gst_structure_get_int (caps_structure, "channels", &channels);
+ mm_attrs_set_int_by_name (player->attrs, "content_audio_channels", channels);
+
+ debug_log ("audio samplerate : %d channels : %d\n", samplerate, channels);
+}
+
+static void
+__mmplayer_typefind_have_type( GstElement *tf, guint probability, // @
+GstCaps *caps, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ GstPad* pad = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_if_fail( player && tf && caps );
+
+ /* store type string */
+ MMPLAYER_FREEIF(player->type);
+ player->type = gst_caps_to_string(caps);
+ if (player->type)
+ debug_log("meida type %s found, probability %d%% / %d\n", player->type, probability, gst_caps_get_size(caps));
+
+ if ( (!MMPLAYER_IS_WFD_STREAMING( player )) &&
+ (!MMPLAYER_IS_RTSP_STREAMING( player )) &&
+ (g_strrstr(player->type, "audio/x-raw-int")))
+ {
+ debug_error("not support media format\n");
+
+ if (player->msg_posted == FALSE)
+ {
+ MMMessageParamType msg_param;
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
+
+ msg_param.code = MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+
+ /* don't post more if one was sent already */
+ player->msg_posted = TRUE;
+ }
+ return;
+ }
+
+ /* midi type should be stored because it will be used to set audio gain in avsysaudiosink */
+ if ( __mmplayer_is_midi_type(player->type))
+ {
+ player->profile.play_mode = MM_PLAYER_MODE_MIDI;
+ player->bypass_audio_effect = TRUE;
+ }
+ else if ( g_strrstr(player->type, "application/x-hls"))
+ {
+ /* If it can't know exact type when it parses uri because of redirection case,
+ * it will be fixed by typefinder here.
+ */
+ player->profile.uri_type = MM_PLAYER_URI_TYPE_HLS;
+ }
+ else if ( g_strrstr(player->type, "application/dash+xml"))
+ {
+ player->profile.uri_type = MM_PLAYER_URI_TYPE_DASH;
+ }
+
+ pad = gst_element_get_static_pad(tf, "src");
+ if ( !pad )
+ {
+ debug_error("fail to get typefind src pad.\n");
+ return;
+ }
+
+ if (player->use_decodebin)
+ {
+ if(!__mmplayer_try_to_plug_decodebin( player, pad, caps ))
+ {
+ gboolean async = FALSE;
+ debug_error("failed to autoplug %s\n", player->type);
+
+ mm_attrs_get_int_by_name(player->attrs, "profile_prepare_async", &async);
+
+ if ( async && player->msg_posted == FALSE )
+ {
+ __mmplayer_handle_missed_plugin( player );
+ }
+
+ goto DONE;
+ }
+ }
+ else
+ {
+ /* try to plug */
+ if ( ! __mmplayer_try_to_plug( player, pad, caps ) )
+ {
+ gboolean async = FALSE;
+ debug_error("failed to autoplug %s\n", player->type);
+
+ mm_attrs_get_int_by_name(player->attrs, "profile_prepare_async", &async);
+
+ if ( async && player->msg_posted == FALSE )
+ {
+ __mmplayer_handle_missed_plugin( player );
+ }
+
+ goto DONE;
+ }
+
+ /* finish autopluging if no dynamic pad waiting */
+ if( ( ! player->have_dynamic_pad) && ( ! player->has_many_types) )
+ {
+ if ( ! MMPLAYER_IS_RTSP_STREAMING( player ) )
+ {
+ __mmplayer_pipeline_complete( NULL, (gpointer)player );
+ }
+ }
+ }
+
+DONE:
+ gst_object_unref( GST_OBJECT(pad) );
+
+ MMPLAYER_FLEAVE();
+
+ return;
+}
+
+#ifdef _MM_PLAYER_ALP_PARSER
+void check_name (void *data, void *user_data)
+{
+ mm_player_t* player = user_data;
+
+ if (g_strrstr((gchar*)data, "mpegaudioparse"))
+ {
+ debug_log("mpegaudioparse - set alp-mp3dec\n");
+ g_object_set(player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst, "alp-mp3dec", TRUE, NULL);
+ }
+}
+#endif
+
+static GstElement *
+__mmplayer_create_decodebin (mm_player_t* player)
+{
+ GstElement *decodebin = NULL;
+
+ MMPLAYER_FENTER();
+
+ /* create decodebin */
+ decodebin = gst_element_factory_make("decodebin2", "decodebin2");
+
+ if (!decodebin)
+ {
+ debug_error("fail to create decodebin\n");
+ goto ERROR;
+ }
+
+ /* raw pad handling signal */
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added",
+ G_CALLBACK(__mmplayer_gst_decode_pad_added), player);
+
+ /* no-more-pad pad handling signal */
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "no-more-pads",
+ G_CALLBACK(__mmplayer_gst_decode_no_more_pads), player);
+
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-removed",
+ G_CALLBACK(__mmplayer_gst_decode_pad_removed), player);
+
+ /* This signal is emitted when a pad for which there is no further possible
+ decoding is added to the decodebin.*/
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "unknown-type",
+ G_CALLBACK(__mmplayer_gst_decode_unknown_type), player );
+
+ /* This signal is emitted whenever decodebin2 finds a new stream. It is emitted
+ before looking for any elements that can handle that stream.*/
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "autoplug-continue",
+ G_CALLBACK(__mmplayer_gst_decode_autoplug_continue), player);
+
+ /* This signal is emitted whenever decodebin2 finds a new stream. It is emitted
+ before looking for any elements that can handle that stream.*/
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "autoplug-select",
+ G_CALLBACK(__mmplayer_gst_decode_autoplug_select), player);
+
+ /* Once decodebin2 has found the possible GstElementFactory objects to try for
+ caps on pad, this signal is emited. The purpose of the signal is for the
+ application to perform additional sorting or filtering on the element factory
+ array.*/
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "autoplug-sort",
+ G_CALLBACK(__mmplayer_gst_decode_autoplug_sort), player);
+
+ /* This signal is emitted once decodebin2 has finished decoding all the data.*/
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "drained",
+ G_CALLBACK(__mmplayer_gst_decode_drained), player);
+
+ /* This signal is emitted when a element is added to the bin.*/
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(decodebin), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "element-added",
+ G_CALLBACK(__mmplayer_gst_element_added), player);
+
+ERROR:
+ return decodebin;
+}
+
+static void
+__mmplayer_gst_id3_pad_added(GstElement *element, GstPad *pad, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ GstCaps *caps = NULL;
+ gchar *caps_str = NULL;
+
+ MMPLAYER_FENTER();
+ return_if_fail (player && pad);
+
+ caps = gst_pad_get_caps(pad);
+ caps_str = gst_caps_to_string(caps);
+
+ debug_log("pad %s:%s, caps %s", GST_DEBUG_PAD_NAME(pad), caps_str);
+ __mmplayer_try_to_plug_decodebin(player, pad, caps);
+
+ MMPLAYER_FREEIF( caps_str );
+ gst_caps_unref(caps);
+
+ MMPLAYER_FLEAVE();
+}
+
+static gboolean
+__mmplayer_try_to_plug_decodebin(mm_player_t* player, GstPad *srcpad, const GstCaps *caps)
+{
+ MMPlayerGstElement* mainbin = NULL;
+ GstElement* decodebin2 = NULL;
+ GstElement* queue2 = NULL;
+ GstElement* id3demux = NULL;
+
+ GstPad* sinkpad = NULL;
+ GstPad* qsrcpad= NULL;
+ gchar *caps_str = NULL;
+
+ GstFormat fmt = GST_FORMAT_BYTES;
+ gint64 dur_bytes = 0L;
+ gchar* file_buffering_path = NULL;
+ gboolean use_file_buffer = FALSE;
+
+ guint max_buffer_size_bytes = 0;
+ gdouble init_buffering_time = (gdouble)player->streamer->buffering_req.initial_second;
+
+ MMPLAYER_FENTER();
+ return_val_if_fail (player && player->pipeline && player->pipeline->mainbin, FALSE);
+
+ mainbin = player->pipeline->mainbin;
+
+ if ((!MMPLAYER_IS_HTTP_PD(player)) &&
+ (MMPLAYER_IS_HTTP_STREAMING(player)))
+ {
+ debug_log ("creating http streaming buffering queue (queue2)\n");
+
+ if (mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst)
+ {
+ debug_error ("MMPLAYER_M_MUXED_S_BUFFER is not null\n");
+ }
+ else
+ {
+ if (!mainbin[MMPLAYER_M_ID3DEMUX].gst)
+ {
+ caps_str = gst_caps_to_string(caps);
+ if (g_strrstr(caps_str, "application/x-id3"))
+ {
+ debug_log("create id3demux in front of queue2 \n");
+
+ id3demux = gst_element_factory_make ("id3demux", NULL);
+ if (!id3demux)
+ {
+ debug_error ("failed to create buffering id3demux element\n");
+ goto ERROR;
+ }
+
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(id3demux), MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added",
+ G_CALLBACK(__mmplayer_gst_id3_pad_added), player);
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), id3demux))
+ {
+ debug_error("failed to add id3demux\n");
+ goto ERROR;
+ }
+
+ sinkpad = gst_element_get_static_pad(id3demux, "sink");
+
+ if (GST_PAD_LINK_OK != gst_pad_link(srcpad, sinkpad))
+ {
+ debug_error("failed to link buffering queue\n");
+ goto ERROR;
+ }
+
+ if (GST_STATE_CHANGE_FAILURE == gst_element_sync_state_with_parent (id3demux))
+ {
+ debug_error("failed to sync id3demux state with parent\n");
+ goto ERROR;
+ }
+
+ mainbin[MMPLAYER_M_ID3DEMUX].id = MMPLAYER_M_ID3DEMUX;
+ mainbin[MMPLAYER_M_ID3DEMUX].gst = id3demux;
+
+ gst_object_unref(GST_OBJECT(sinkpad));
+ MMPLAYER_FREEIF( caps_str );
+
+ MMPLAYER_FLEAVE();
+ return TRUE;
+ }
+
+ MMPLAYER_FREEIF( caps_str );
+ }
+
+ queue2 = gst_element_factory_make ("queue2", "queue2");
+ if (!queue2)
+ {
+ debug_error ("failed to create buffering queue element\n");
+ goto ERROR;
+ }
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue2))
+ {
+ debug_error("failed to add buffering queue\n");
+ goto ERROR;
+ }
+
+ sinkpad = gst_element_get_static_pad(queue2, "sink");
+ qsrcpad = gst_element_get_static_pad(queue2, "src");
+
+ if (GST_PAD_LINK_OK != gst_pad_link(srcpad, sinkpad))
+ {
+ debug_error("failed to link buffering queue\n");
+ goto ERROR;
+ }
+
+ // if ( !MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
+ {
+ if ( !gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, &fmt, &dur_bytes))
+ debug_error("fail to get duration.\n");
+
+ debug_log("dur_bytes = %lld\n", dur_bytes);
+
+ if (dur_bytes > 0)
+ {
+ use_file_buffer = MMPLAYER_USE_FILE_FOR_BUFFERING(player);
+ file_buffering_path = g_strdup(player->ini.http_file_buffer_path);
+ }
+ else
+ {
+ dur_bytes = 0;
+ }
+ }
+
+ /* NOTE : we cannot get any duration info from ts container in case of streaming */
+ // if(!g_strrstr(GST_ELEMENT_NAME(sinkelement), "mpegtsdemux"))
+ if(!g_strrstr(player->type, "video/mpegts"))
+ {
+ max_buffer_size_bytes = (use_file_buffer)?(player->ini.http_max_size_bytes):(5*1024*1024);
+ debug_log("max_buffer_size_bytes = %d\n", max_buffer_size_bytes);
+
+ __mm_player_streaming_set_queue2(player->streamer,
+ queue2,
+ FALSE,
+ max_buffer_size_bytes,
+ player->ini.http_buffering_time,
+ 1.0, // no meaning
+ player->ini.http_buffering_limit, // no meaning
+ use_file_buffer,
+ file_buffering_path,
+ (guint64)dur_bytes);
+ }
+
+ MMPLAYER_FREEIF(file_buffering_path);
+ if (GST_STATE_CHANGE_FAILURE == gst_element_sync_state_with_parent (queue2))
+ {
+ debug_error("failed to sync queue2 state with parent\n");
+ goto ERROR;
+ }
+
+ srcpad = qsrcpad;
+
+ gst_object_unref(GST_OBJECT(sinkpad));
+
+ mainbin[MMPLAYER_M_MUXED_S_BUFFER].id = MMPLAYER_M_MUXED_S_BUFFER;
+ mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst = queue2;
+ }
+ }
+
+ /* create decodebin */
+ decodebin2 = __mmplayer_create_decodebin(player);
+
+ if (!decodebin2)
+ {
+ debug_error("can not create autoplug element\n");
+ goto ERROR;
+ }
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), decodebin2))
+ {
+ debug_error("failed to add decodebin2\n");
+ goto ERROR;
+ }
+
+ /* to force caps on the decodebin element and avoid reparsing stuff by
+ * typefind. It also avoids a deadlock in the way typefind activates pads in
+ * the state change */
+ g_object_set (decodebin2, "sink-caps", caps, NULL);
+
+ sinkpad = gst_element_get_static_pad(decodebin2, "sink");
+
+ if (GST_PAD_LINK_OK != gst_pad_link(srcpad, sinkpad))
+ {
+ debug_error("failed to link decodebin2\n");
+ goto ERROR;
+ }
+
+ gst_object_unref(GST_OBJECT(sinkpad));
+
+ mainbin[MMPLAYER_M_AUTOPLUG].id = MMPLAYER_M_AUTOPLUG;
+ mainbin[MMPLAYER_M_AUTOPLUG].gst = decodebin2;
+
+ if ( ((!MMPLAYER_IS_HTTP_PD(player)) &&
+ (MMPLAYER_IS_HTTP_STREAMING(player))) || MMPLAYER_IS_DASH_STREAMING (player))
+ {
+ init_buffering_time = (init_buffering_time != 0)?(init_buffering_time):(player->ini.http_buffering_time);
+
+ g_object_set (G_OBJECT(decodebin2), "use-buffering", TRUE,
+ "high-percent", (gint)player->ini.http_buffering_limit,
+ "low-percent", 1, // 1%
+ "max-size-bytes", MAX_DECODEBIN_BUFFER_BYTES,
+ "max-size-time", (guint64)(MAX_DECODEBIN_BUFFER_TIME * GST_SECOND),
+ "max-size-buffers", 0, NULL); // disable or automatic
+ }
+
+ if (GST_STATE_CHANGE_FAILURE == gst_element_sync_state_with_parent(decodebin2))
+ {
+ debug_error("failed to sync decodebin2 state with parent\n");
+ goto ERROR;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
+
+ERROR:
+
+ MMPLAYER_FREEIF( caps_str );
+
+ if (sinkpad)
+ gst_object_unref(GST_OBJECT(sinkpad));
+
+ if (queue2)
+ {
+ /* NOTE : Trying to dispose element queue0, but it is in READY instead of the NULL state.
+ * You need to explicitly set elements to the NULL state before
+ * dropping the final reference, to allow them to clean up.
+ */
+ gst_element_set_state(queue2, GST_STATE_NULL);
+
+ /* And, it still has a parent "player".
+ * You need to let the parent manage the object instead of unreffing the object directly.
+ */
+ gst_bin_remove (GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue2);
+ gst_object_unref (queue2);
+ queue2 = NULL;
+ }
+
+ if (id3demux)
+ {
+ /* NOTE : Trying to dispose element id3demux, but it is in READY instead of the NULL state.
+ * You need to explicitly set elements to the NULL state before
+ * dropping the final reference, to allow them to clean up.
+ */
+ gst_element_set_state(id3demux, GST_STATE_NULL);
+
+ /* And, it still has a parent "player".
+ * You need to let the parent manage the object instead of unreffing the object directly.
+ */
+ gst_bin_remove (GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), id3demux);
+ gst_object_unref (id3demux);
+ id3demux = NULL;
+ }
+
+ if (decodebin2)
+ {
+ /* NOTE : Trying to dispose element queue0, but it is in READY instead of the NULL state.
+ * You need to explicitly set elements to the NULL state before
+ * dropping the final reference, to allow them to clean up.
+ */
+ gst_element_set_state(decodebin2, GST_STATE_NULL);
+
+ /* And, it still has a parent "player".
+ * You need to let the parent manage the object instead of unreffing the object directly.
+ */
+
+ gst_bin_remove (GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), decodebin2);
+ gst_object_unref (decodebin2);
+ decodebin2 = NULL;
+ }
+
+ return FALSE;
+}
+
+/* it will return first created element */
+static gboolean
+__mmplayer_try_to_plug(mm_player_t* player, GstPad *pad, const GstCaps *caps) // @
+{
+ MMPlayerGstElement* mainbin = NULL;
+ const char* mime = NULL;
+ const GList* item = NULL;
+ const gchar* klass = NULL;
+ GstCaps* res = NULL;
+ gboolean skip = FALSE;
+ GstPad* queue_pad = NULL;
+ GstElement* queue = NULL;
+ GstElement *element = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player && player->pipeline && player->pipeline->mainbin, FALSE );
+
+ mainbin = player->pipeline->mainbin;
+
+ mime = gst_structure_get_name(gst_caps_get_structure(caps, 0));
+
+ /* return if we got raw output */
+ if(g_str_has_prefix(mime, "video/x-raw") || g_str_has_prefix(mime, "audio/x-raw")
+ || g_str_has_prefix(mime, "text/plain") ||g_str_has_prefix(mime, "text/x-pango-markup"))
+ {
+
+ element = (GstElement*)gst_pad_get_parent(pad);
+/* NOTE : When no decoder has added during autoplugging. like a simple wave playback.
+ * No queue will be added. I think it can caused breaking sound when playing raw audio
+ * frames but there's no different. Decodebin also doesn't add with those wav fils.
+ * Anyway, currentely raw-queue seems not necessary.
+ */
+#if 1
+ /* NOTE : check if previously linked element is demuxer/depayloader/parse means no decoder
+ * has linked. if so, we need to add queue for quality of output. note that
+ * decodebin also has same problem.
+ */
+ klass = gst_element_factory_get_klass( gst_element_get_factory(element) );
+
+ /* add queue if needed */
+ if( (g_strrstr(klass, "Demux") || g_strrstr(klass, "Depayloader")
+ || g_strrstr(klass, "Parse")) && !g_str_has_prefix(mime, "text"))
+ {
+ debug_log("adding raw queue\n");
+
+ queue = gst_element_factory_make("queue", NULL);
+ if ( ! queue )
+ {
+ debug_warning("failed to create queue\n");
+ goto ERROR;
+ }
+
+ /* warmup */
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(queue, GST_STATE_READY) )
+ {
+ debug_warning("failed to set state READY to queue\n");
+ goto ERROR;
+ }
+
+ /* add to pipeline */
+ if ( ! gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue) )
+ {
+ debug_warning("failed to add queue\n");
+ goto ERROR;
+ }
+
+ /* link queue */
+ queue_pad = gst_element_get_static_pad(queue, "sink");
+
+ if ( GST_PAD_LINK_OK != gst_pad_link(pad, queue_pad) )
+ {
+ debug_warning("failed to link queue\n");
+ goto ERROR;
+ }
+ gst_object_unref ( GST_OBJECT(queue_pad) );
+ queue_pad = NULL;
+
+ /* running */
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(queue, GST_STATE_PAUSED) )
+ {
+ debug_warning("failed to set state READY to queue\n");
+ goto ERROR;
+ }
+
+ /* replace given pad to queue:src */
+ pad = gst_element_get_static_pad(queue, "src");
+ if ( ! pad )
+ {
+ debug_warning("failed to get pad from queue\n");
+ goto ERROR;
+ }
+ }
+#endif
+ /* check if player can do start continually */
+ MMPLAYER_CHECK_CMD_IF_EXIT(player);
+
+ if(__mmplayer_link_sink(player,pad))
+ __mmplayer_gst_decode_callback(element, pad, player);
+
+ gst_object_unref( GST_OBJECT(element));
+ element = NULL;
+
+ return TRUE;
+ }
+
+ item = player->factories;
+ for(; item != NULL ; item = item->next)
+ {
+ GstElementFactory *factory = GST_ELEMENT_FACTORY(item->data);
+ const GList *pads;
+ gint idx = 0;
+
+ skip = FALSE;
+
+ /* filtering exclude keyword */
+ for ( idx = 0; player->ini.exclude_element_keyword[idx][0] != '\0'; idx++ )
+ {
+ if ( g_strrstr(GST_PLUGIN_FEATURE_NAME (factory),
+ player->ini.exclude_element_keyword[idx] ) )
+ {
+ debug_warning("skipping [%s] by exculde keyword [%s]\n",
+ GST_PLUGIN_FEATURE_NAME (factory),
+ player->ini.exclude_element_keyword[idx] );
+
+ skip = TRUE;
+ break;
+ }
+ }
+
+ if ( MMPLAYER_IS_RTSP_STREAMING(player) && g_strrstr(GST_PLUGIN_FEATURE_NAME (factory), "omx_mpeg4dec"))
+ {
+ // omx decoder can not support mpeg4video data partitioned
+ // rtsp streaming didn't know mpeg4video data partitioned format
+ // so, if rtsp playback, player will skip omx_mpeg4dec.
+ debug_warning("skipping [%s] when rtsp streaming \n",
+ GST_PLUGIN_FEATURE_NAME (factory));
+
+ skip = TRUE;
+ }
+
+ if ( skip ) continue;
+
+ /* check factory class for filtering */
+ klass = gst_element_factory_get_klass(GST_ELEMENT_FACTORY(factory));
+
+ /* NOTE : msl don't need to use image plugins.
+ * So, those plugins should be skipped for error handling.
+ */
+ if ( g_strrstr(klass, "Codec/Decoder/Image") )
+ {
+ debug_log("skipping [%s] by not required\n", GST_PLUGIN_FEATURE_NAME (factory));
+ continue;
+ }
+
+ /* check pad compatability */
+ for(pads = gst_element_factory_get_static_pad_templates(factory);
+ pads != NULL; pads=pads->next)
+ {
+ GstStaticPadTemplate *temp1 = pads->data;
+ GstCaps* static_caps = NULL;
+
+ if( temp1->direction != GST_PAD_SINK ||
+ temp1->presence != GST_PAD_ALWAYS)
+ continue;
+
+ if ( GST_IS_CAPS( &temp1->static_caps.caps) )
+ {
+ /* using existing caps */
+ static_caps = gst_caps_ref( &temp1->static_caps.caps );
+ }
+ else
+ {
+ /* create one */
+ static_caps = gst_caps_from_string ( temp1->static_caps.string );
+ }
+
+ res = gst_caps_intersect(caps, static_caps);
+
+ gst_caps_unref( static_caps );
+ static_caps = NULL;
+
+ if( res && !gst_caps_is_empty(res) )
+ {
+ GstElement *new_element;
+ GList *elements = player->parsers;
+ char *name_template = g_strdup(temp1->name_template);
+ gchar *name_to_plug = GST_PLUGIN_FEATURE_NAME(factory);
+ gst_caps_unref(res);
+
+ /* check ALP Codec can be used or not */
+ if ((g_strrstr(klass, "Codec/Decoder/Audio")))
+ {
+ /* consider mp3 audio only */
+ if ( !MMPLAYER_IS_STREAMING(player) && __mmplayer_is_only_mp3_type(player->type) )
+ {
+ /* try to use ALP decoder first instead of selected decoder */
+ GstElement *element = NULL;
+ GstElementFactory * element_facory;
+ gchar *path = NULL;
+ guint64 data_size = 0;
+ #define MIN_THRESHOLD_SIZE 320 * 1024 // 320K
+ struct stat sb;
+
+ mm_attrs_get_string_by_name(player->attrs, "profile_uri", &path);
+
+ if (stat(path, &sb) == 0)
+ {
+ data_size = (guint64)sb.st_size;
+ }
+ debug_log("file size : %u", data_size);
+
+ if (data_size > MIN_THRESHOLD_SIZE)
+ {
+ debug_log("checking if ALP can be used or not");
+ element = gst_element_factory_make("omx_mp3dec", "omx mp3 decoder");
+ if ( element )
+ {
+ /* check availability because multi-instance is not supported */
+ GstStateChangeReturn ret = gst_element_set_state(element, GST_STATE_READY);
+
+ if (ret != GST_STATE_CHANGE_SUCCESS) // use just selected decoder
+ {
+ gst_object_unref (element);
+ }
+ else if (ret == GST_STATE_CHANGE_SUCCESS) // replace facotry to use omx
+ {
+ /* clean */
+ gst_element_set_state(element, GST_STATE_NULL);
+ gst_object_unref (element);
+
+ element_facory = gst_element_factory_find("omx_mp3dec");
+ /* replace, otherwise use selected thing instead */
+ if (element_facory)
+ {
+ factory = element_facory;
+ name_to_plug = GST_PLUGIN_FEATURE_NAME(factory);
+ }
+
+ /* make parser alp mode */
+ #ifdef _MM_PLAYER_ALP_PARSER
+ g_list_foreach (player->parsers, check_name, player);
+ #endif
+ }
+ }
+ }
+ }
+ }
+ else if ((g_strrstr(klass, "Codec/Decoder/Video")))
+ {
+ if ( g_strrstr(GST_PLUGIN_FEATURE_NAME(factory), "omx_") )
+ {
+ char *env = getenv ("MM_PLAYER_HW_CODEC_DISABLE");
+ if (env != NULL)
+ {
+ if (strncasecmp(env, "yes", 3) == 0)
+ {
+ debug_log("skipping [%s] by disabled\n", name_to_plug);
+ MMPLAYER_FREEIF(name_template);
+ continue;
+ }
+ }
+ }
+ }
+
+ debug_log("found %s to plug\n", name_to_plug);
+
+ new_element = gst_element_factory_create(GST_ELEMENT_FACTORY(factory), NULL);
+ if ( ! new_element )
+ {
+ debug_error("failed to create element [%s]. continue with next.\n",
+ GST_PLUGIN_FEATURE_NAME (factory));
+
+ MMPLAYER_FREEIF(name_template);
+
+ continue;
+ }
+
+ /* check and skip it if it was already used. Otherwise, it can be an infinite loop
+ * because parser can accept its own output as input.
+ */
+ if (g_strrstr(klass, "Parser"))
+ {
+ gchar *selected = NULL;
+
+ for ( ; elements; elements = g_list_next(elements))
+ {
+ gchar *element_name = elements->data;
+
+ if (g_strrstr(element_name, name_to_plug))
+ {
+ debug_log("but, %s already linked, so skipping it\n", name_to_plug);
+ skip = TRUE;
+ }
+ }
+
+ if (skip)
+ {
+ MMPLAYER_FREEIF(name_template);
+ continue;
+ }
+
+ selected = g_strdup(name_to_plug);
+ player->parsers = g_list_append(player->parsers, selected);
+ }
+
+ /* store specific handles for futher control */
+ if(g_strrstr(klass, "Demux") || g_strrstr(klass, "Parse"))
+ {
+ /* FIXIT : first value will be overwritten if there's more
+ * than 1 demuxer/parser
+ */
+ debug_log("plugged element is demuxer. take it\n");
+ mainbin[MMPLAYER_M_DEMUX].id = MMPLAYER_M_DEMUX;
+ mainbin[MMPLAYER_M_DEMUX].gst = new_element;
+
+ /*Added for multi audio support */
+ if(g_strrstr(klass, "Demux"))
+ {
+ mainbin[MMPLAYER_M_DEMUX_EX].id = MMPLAYER_M_DEMUX_EX;
+ mainbin[MMPLAYER_M_DEMUX_EX].gst = new_element;
+
+ /* NOTE : workaround for bug in mpegtsdemux since doesn't emit
+ no-more-pad signal. this may cause wrong content attributes at PAUSED state
+ this code should be removed after mpegtsdemux is fixed */
+ if ( g_strrstr(GST_PLUGIN_FEATURE_NAME(factory), "mpegtsdemux") )
+ {
+ debug_warning("force no-more-pad to TRUE since mpegtsdemux os not giving no-more-pad signal. content attributes may wrong");
+ player->no_more_pad = TRUE;
+ }
+ }
+ if (g_strrstr(name_to_plug, "asfdemux")) // to support trust-zone only
+ {
+ g_object_set(mainbin[MMPLAYER_M_DEMUX_EX].gst, "file-location", player->profile.uri,NULL);
+ }
+ }
+ else if(g_strrstr(klass, "Decoder") && __mmplayer_link_decoder(player,pad))
+ {
+ if(mainbin[MMPLAYER_M_DEC1].gst == NULL)
+ {
+ debug_log("plugged element is decoder. take it[MMPLAYER_M_DEC1]\n");
+ mainbin[MMPLAYER_M_DEC1].id = MMPLAYER_M_DEC1;
+ mainbin[MMPLAYER_M_DEC1].gst = new_element;
+ }
+ else if(mainbin[MMPLAYER_M_DEC2].gst == NULL)
+ {
+ debug_log("plugged element is decoder. take it[MMPLAYER_M_DEC2]\n");
+ mainbin[MMPLAYER_M_DEC2].id = MMPLAYER_M_DEC2;
+ mainbin[MMPLAYER_M_DEC2].gst = new_element;
+ }
+ /* NOTE : IF one codec is found, add it to supported_codec and remove from
+ * missing plugin. Both of them are used to check what's supported codec
+ * before returning result of play start. And, missing plugin should be
+ * updated here for multi track files.
+ */
+ if(g_str_has_prefix(mime, "video"))
+ {
+ GstPad *src_pad = NULL;
+ GstPadTemplate *pad_templ = NULL;
+ GstCaps *caps = NULL;
+ gchar *caps_str = NULL;
+
+ debug_log("found VIDEO decoder\n");
+ player->not_supported_codec &= MISSING_PLUGIN_AUDIO;
+ player->can_support_codec |= FOUND_PLUGIN_VIDEO;
+
+ src_pad = gst_element_get_static_pad (new_element, "src");
+ pad_templ = gst_pad_get_pad_template (src_pad);
+ caps = GST_PAD_TEMPLATE_CAPS(pad_templ);
+
+ caps_str = gst_caps_to_string(caps);
+
+ /* clean */
+ MMPLAYER_FREEIF( caps_str );
+ gst_object_unref (src_pad);
+ }
+ else if (g_str_has_prefix(mime, "audio"))
+ {
+ debug_log("found AUDIO decoder\n");
+ player->not_supported_codec &= MISSING_PLUGIN_VIDEO;
+ player->can_support_codec |= FOUND_PLUGIN_AUDIO;
+ }
+ }
+
+ if ( ! __mmplayer_close_link(player, pad, new_element,
+ name_template,gst_element_factory_get_static_pad_templates(factory)) )
+ {
+ MMPLAYER_FREEIF(name_template);
+ if (player->keep_detecting_vcodec)
+ continue;
+
+ /* Link is failed even though a supportable codec is found. */
+ __mmplayer_check_not_supported_codec(player, klass, mime);
+
+ debug_error("failed to call _close_link\n");
+ return FALSE;
+ }
+
+ MMPLAYER_FREEIF(name_template);
+ return TRUE;
+ }
+
+ gst_caps_unref(res);
+ break;
+ }
+ }
+
+ /* There is no available codec. */
+ __mmplayer_check_not_supported_codec(player, klass, mime);
+
+ MMPLAYER_FLEAVE();
+ return FALSE;
+
+ERROR:
+ /* release */
+ if ( queue )
+ gst_object_unref( queue );
+
+ if ( queue_pad )
+ gst_object_unref( queue_pad );
+
+ if ( element )
+ gst_object_unref ( element );
+
+ return FALSE;
+}
+
+
+static int
+__mmplayer_check_not_supported_codec(mm_player_t* player, const gchar* factory_class, const gchar* mime)
+{
+ MMPLAYER_FENTER();
+
+ return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ return_val_if_fail ( mime, MM_ERROR_INVALID_ARGUMENT );
+
+ debug_log("class : %s, mime : %s \n", factory_class, mime );
+
+ /* add missing plugin */
+ /* NOTE : msl should check missing plugin for image mime type.
+ * Some motion jpeg clips can have playable audio track.
+ * So, msl have to play audio after displaying popup written video format not supported.
+ */
+ if ( !( player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst ) )
+ {
+ if ( !( player->can_support_codec | player->videodec_linked | player->audiodec_linked ) )
+ {
+ debug_log("not found demuxer\n");
+ player->not_found_demuxer = TRUE;
+ player->unlinked_demuxer_mime = g_strdup_printf ( "%s", mime );
+
+ goto DONE;
+ }
+ }
+
+ if( !g_strrstr(factory_class, "Demuxer"))
+ {
+ if( ( g_str_has_prefix(mime, "video") ) ||( g_str_has_prefix(mime, "image") ) )
+ {
+ debug_log("can support codec=%d, vdec_linked=%d, adec_linked=%d\n",
+ player->can_support_codec, player->videodec_linked, player->audiodec_linked);
+
+ /* check that clip have multi tracks or not */
+ if ( ( player->can_support_codec & FOUND_PLUGIN_VIDEO ) && ( player->videodec_linked ) )
+ {
+ debug_log("video plugin is already linked\n");
+ }
+ else
+ {
+ debug_warning("add VIDEO to missing plugin\n");
+ player->not_supported_codec |= MISSING_PLUGIN_VIDEO;
+ }
+ }
+ else if ( g_str_has_prefix(mime, "audio") )
+ {
+ if ( ( player->can_support_codec & FOUND_PLUGIN_AUDIO ) && ( player->audiodec_linked ) )
+ {
+ debug_log("audio plugin is already linked\n");
+ }
+ else
+ {
+ debug_warning("add AUDIO to missing plugin\n");
+ player->not_supported_codec |= MISSING_PLUGIN_AUDIO;
+ }
+ }
+ }
+
+DONE:
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
+
+
+static void
+__mmplayer_pipeline_complete(GstElement *decodebin, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+
+ MMPLAYER_FENTER();
+
+ return_if_fail( player );
+
+ /* remove fakesink. */
+ if ( ! __mmplayer_gst_remove_fakesink( player,
+ &player->pipeline->mainbin[MMPLAYER_M_SRC_FAKESINK]) )
+ {
+ /* NOTE : __mmplayer_pipeline_complete() can be called several time. because
+ * signaling mechanism ( pad-added, no-more-pad, new-decoded-pad ) from various
+ * source element are not same. To overcome this situation, this function will called
+ * several places and several times. Therefore, this is not an error case.
+ */
+ return;
+ }
+
+ debug_log("pipeline has completely constructed\n");
+
+ if ( ( player->ini.async_start ) &&
+ ( player->msg_posted == FALSE ) &&
+ ( player->cmd >= MMPLAYER_COMMAND_START ))
+ {
+ __mmplayer_handle_missed_plugin( player );
+ }
+
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-complete" );
+}
+
+static gboolean
+__mmplayer_verify_next_play_path(mm_player_t *player)
+{
+ MMHandleType attrs = 0;
+ MMPlayerParseProfile profile;
+ gint uri_idx = 0, check_cnt = 0;
+ char *uri = NULL;
+ gint mode = MM_PLAYER_PD_MODE_NONE;
+ gint count = 0;
+ guint num_of_list = 0;
+
+ MMPLAYER_FENTER();
+
+ debug_log("checking for next play");
+
+ if (player->pipeline->textbin)
+ {
+ debug_error("subtitle path is enabled. next play is not supported.\n");
+ goto ERROR;
+ }
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("fail to get attributes.\n");
+ goto ERROR;
+ }
+
+ /* seamless playback is supported in case of audio only */
+ mm_attrs_get_int_by_name(attrs, "content_video_found", &mode);
+ if (mode)
+ {
+ debug_log("video found");
+ goto ERROR;
+ }
+
+ if (mm_attrs_get_int_by_name (attrs, "pd_mode", &mode) == MM_ERROR_NONE)
+ {
+ if (mode == TRUE)
+ {
+ debug_warning("pd mode\n");
+ goto ERROR;
+ }
+ }
+
+ if (mm_attrs_get_int_by_name(attrs, "profile_play_count", &count) != MM_ERROR_NONE)
+ {
+ debug_error("can not get play count\n");
+ }
+
+ num_of_list = g_list_length(player->uri_info.uri_list);
+
+ debug_log("repeat count = %d, num_of_list = %d\n", count, num_of_list);
+
+ if ( num_of_list == 0 )
+ {
+ if (mm_attrs_get_string_by_name(player->attrs, "profile_uri", &uri) != MM_ERROR_NONE)
+ {
+ debug_error("can not get profile_uri\n");
+ goto ERROR;
+ }
+
+ if (!uri)
+ {
+ debug_error("uri list is empty.\n");
+ goto ERROR;
+ }
+
+ player->uri_info.uri_list = g_list_append(player->uri_info.uri_list, g_strdup(uri));
+ debug_log("add original path : %s ", uri);
+
+ num_of_list = 1;
+ uri= NULL;
+ }
+
+ uri_idx = player->uri_info.uri_idx;
+
+ while(TRUE)
+ {
+ check_cnt++;
+
+ if (check_cnt > num_of_list)
+ {
+ debug_error("there is no valid uri.");
+ goto ERROR;
+ }
+
+ debug_log("uri idx : %d / %d\n", uri_idx, num_of_list);
+
+ if ( uri_idx < num_of_list-1 )
+ {
+ uri_idx++;
+ }
+ else
+ {
+ if ((count <= 1) && (count != -1))
+ {
+ debug_log("no repeat.");
+ goto ERROR;
+ }
+ else if ( count > 1 ) /* decrease play count */
+ {
+ /* we successeded to rewind. update play count and then wait for next EOS */
+ count--;
+
+ mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
+
+ /* commit attribute */
+ if ( mmf_attrs_commit ( attrs ) )
+ {
+ debug_error("failed to commit attribute\n");
+ }
+ }
+
+ /* count < 0 : repeat continually */
+ uri_idx = 0;
+ }
+
+ uri = g_list_nth_data(player->uri_info.uri_list, uri_idx);
+ debug_log("uri idx : %d, uri = %s\n", uri_idx, uri);
+
+ if (uri == NULL)
+ {
+ debug_warning("next uri does not exist\n");
+ continue;
+ }
+
+ if (__mmfplayer_parse_profile((const char*)uri, NULL, &profile) != MM_ERROR_NONE)
+ {
+ debug_error("failed to parse profile\n");
+ continue;
+ }
+
+ if ((profile.uri_type != MM_PLAYER_URI_TYPE_FILE) &&
+ (profile.uri_type != MM_PLAYER_URI_TYPE_URL_HTTP))
+ {
+ debug_warning("uri type is not supported (%d).", profile.uri_type);
+ continue;
+ }
+
+ break;
+ }
+
+ player->uri_info.uri_idx = uri_idx;
+ mm_attrs_set_string_by_name(player->attrs, "profile_uri", uri);
+
+
+ if (mmf_attrs_commit(player->attrs))
+ {
+ debug_error("failed to commit.\n");
+ goto ERROR;
+ }
+
+ debug_log("next uri %s (%d)\n", uri, uri_idx);
+
+ return TRUE;
+
+ERROR:
+
+ debug_error("unable to play next path. EOS will be posted soon.\n");
+ return FALSE;
+}
+
+static void
+__mmplayer_initialize_next_play(mm_player_t *player)
+{
+ int i;
+
+ MMPLAYER_FENTER();
+
+ player->needed_v_parser = FALSE;
+ player->smooth_streaming = FALSE;
+ player->videodec_linked = 0;
+ player->audiodec_linked = 0;
+ player->videosink_linked = 0;
+ player->audiosink_linked = 0;
+ player->textsink_linked = 0;
+ player->is_external_subtitle_present = FALSE;
+ player->not_supported_codec = MISSING_PLUGIN_NONE;
+ player->can_support_codec = FOUND_PLUGIN_NONE;
+ player->pending_seek.is_pending = FALSE;
+ player->pending_seek.format = MM_PLAYER_POS_FORMAT_TIME;
+ player->pending_seek.pos = 0;
+ player->msg_posted = FALSE;
+ player->has_many_types = FALSE;
+ player->no_more_pad = FALSE;
+ player->is_drm_file = FALSE;
+ player->not_found_demuxer = 0;
+ player->doing_seek = FALSE;
+ player->max_audio_channels = 0;
+ player->is_subtitle_force_drop = FALSE;
+ player->play_subtitle = FALSE;
+ player->use_textoverlay = FALSE;
+ player->adjust_subtitle_pos = 0;
+
+ player->updated_bitrate_count = 0;
+ player->total_bitrate = 0;
+ player->updated_maximum_bitrate_count = 0;
+ player->total_maximum_bitrate = 0;
+
+ for (i = 0; i < MM_PLAYER_STREAM_COUNT_MAX; i++)
+ {
+ player->bitrate[i] = 0;
+ player->maximum_bitrate[i] = 0;
+ }
+
+ if (player->v_stream_caps)
+ {
+ gst_caps_unref(player->v_stream_caps);
+ player->v_stream_caps = NULL;
+ }
+
+ mm_attrs_set_int_by_name(player->attrs, "content_video_found", 0);
+ mm_attrs_set_int_by_name(player->attrs, "content_audio_found", 0);
+
+ /* clean found parsers */
+ if (player->parsers)
+ {
+ GList *parsers = player->parsers;
+ for ( ;parsers ; parsers = g_list_next(parsers))
+ {
+ gchar *name = parsers->data;
+ MMPLAYER_FREEIF(name);
+ }
+ g_list_free(player->parsers);
+ player->parsers = NULL;
+ }
+
+ /* clean found audio decoders */
+ if (player->audio_decoders)
+ {
+ GList *a_dec = player->audio_decoders;
+ for ( ;a_dec ; a_dec = g_list_next(a_dec))
+ {
+ gchar *name = a_dec->data;
+ MMPLAYER_FREEIF(name);
+ }
+ g_list_free(player->audio_decoders);
+ player->audio_decoders = NULL;
+ }
+
+ MMPLAYER_FLEAVE();
+}
+
+static void
+__mmplayer_activate_next_source(mm_player_t *player, GstState target)
+{
+ MMPlayerGstElement *mainbin = NULL;
+ MMMessageParamType msg_param = {0,};
+ GstElement *element = NULL;
+ MMHandleType attrs = 0;
+ char *uri = NULL;
+ enum MainElementID elemId = MMPLAYER_M_NUM;
+
+ MMPLAYER_FENTER();
+
+ if ((player->pipeline == NULL) ||
+ (player->pipeline->mainbin == NULL))
+ {
+ debug_error("player is null.\n");
+ goto ERROR;
+ }
+
+ mainbin = player->pipeline->mainbin;
+ msg_param.code = MM_ERROR_PLAYER_INTERNAL;
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("fail to get attributes.\n");
+ goto ERROR;
+ }
+
+ /* Initialize Player values */
+ __mmplayer_initialize_next_play(player);
+
+ mm_attrs_get_string_by_name(attrs, "profile_uri", &uri);
+
+ if (__mmfplayer_parse_profile((const char*)uri, NULL, &player->profile) != MM_ERROR_NONE)
+ {
+ debug_error("failed to parse profile\n");
+ msg_param.code = MM_ERROR_PLAYER_INVALID_URI;
+ goto ERROR;
+ }
+
+ if ((MMPLAYER_URL_HAS_DASH_SUFFIX(player)) ||
+ (MMPLAYER_URL_HAS_HLS_SUFFIX(player)))
+ {
+ debug_error("it's dash or hls. not support.");
+ msg_param.code = MM_ERROR_PLAYER_INVALID_URI;
+ goto ERROR;
+ }
+
+ /* setup source */
+ switch ( player->profile.uri_type )
+ {
+ /* file source */
+ case MM_PLAYER_URI_TYPE_FILE:
+ {
+
+#ifdef ENABLE_DRMSRC
+
+ char* drmsrc = player->ini.name_of_drmsrc;
+
+ debug_log("using [%s] for 'file://' handler.\n", drmsrc);
+
+ element = gst_element_factory_make(drmsrc, "source");
+ if ( !element )
+ {
+ debug_error("failed to create %s\n", drmsrc);
+ break;
+ }
+#else
+ debug_log("using filesrc for 'file://' handler.\n");
+
+ element = gst_element_factory_make("filesrc", "source");
+
+ if ( !element )
+ {
+ debug_error("failed to create filesrc\n");
+ break;
+ }
+#endif
+ g_object_set(G_OBJECT(element), "location", (player->profile.uri)+7, NULL); /* uri+7 -> remove "file:// */
+ break;
+ }
+ case MM_PLAYER_URI_TYPE_URL_HTTP:
+ {
+ gchar *user_agent, *proxy, *cookies, **cookie_list;
+ gint http_timeout = DEFAULT_HTTP_TIMEOUT;
+ user_agent = proxy = cookies = NULL;
+ cookie_list = NULL;
+
+ element = gst_element_factory_make(player->ini.name_of_httpsrc, "http_streaming_source");
+ if ( !element )
+ {
+ debug_error("failed to create http streaming source element[%s].\n", player->ini.name_of_httpsrc);
+ break;
+ }
+ debug_log("using http streamming source [%s].\n", player->ini.name_of_httpsrc);
+
+ /* get attribute */
+ mm_attrs_get_string_by_name ( attrs, "streaming_cookie", &cookies );
+ mm_attrs_get_string_by_name ( attrs, "streaming_user_agent", &user_agent );
+ mm_attrs_get_string_by_name ( attrs, "streaming_proxy", &proxy );
+ mm_attrs_get_int_by_name ( attrs, "streaming_timeout", &http_timeout );
+
+ if ((http_timeout == DEFAULT_HTTP_TIMEOUT) &&
+ (player->ini.http_timeout != DEFAULT_HTTP_TIMEOUT))
+ {
+ debug_log("get timeout from ini\n");
+ http_timeout = player->ini.http_timeout;
+ }
+
+ /* get attribute */
+ secure_debug_log("location : %s\n", player->profile.uri);
+ secure_debug_log("cookies : %s\n", cookies);
+ secure_debug_log("proxy : %s\n", proxy);
+ secure_debug_log("user_agent : %s\n", user_agent);
+ debug_log("timeout : %d\n", http_timeout);
+
+ /* setting property to streaming source */
+ g_object_set(G_OBJECT(element), "location", player->profile.uri, NULL);
+ g_object_set(G_OBJECT(element), "timeout", http_timeout, NULL);
+ g_object_set(G_OBJECT(element), "blocksize", (unsigned long)(64*1024), NULL);
+
+ /* check if prosy is vailid or not */
+ if ( util_check_valid_url ( proxy ) )
+ g_object_set(G_OBJECT(element), "proxy", proxy, NULL);
+ /* parsing cookies */
+ if ( ( cookie_list = util_get_cookie_list ((const char*)cookies) ) )
+ g_object_set(G_OBJECT(element), "cookies", cookie_list, NULL);
+ if ( user_agent )
+ g_object_set(G_OBJECT(element), "user_agent", user_agent, NULL);
+ break;
+ }
+ default:
+ debug_error("not support uri type %d\n", player->profile.uri_type);
+ break;
+ }
+
+ if ( !element )
+ {
+ debug_error("no source element was created.\n");
+ goto ERROR;
+ }
+
+ if (gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), element) == FALSE)
+ {
+ debug_error("failed to add source element to pipeline\n");
+ gst_object_unref(GST_OBJECT(element));
+ element = NULL;
+ goto ERROR;
+ }
+
+ /* take source element */
+ mainbin[MMPLAYER_M_SRC].id = MMPLAYER_M_SRC;
+ mainbin[MMPLAYER_M_SRC].gst = element;
+
+ element = NULL;
+
+ if (MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ if (player->streamer == NULL)
+ {
+ player->streamer = __mm_player_streaming_create();
+
+ if(player->streamer == NULL)
+ goto ERROR;
+
+ __mm_player_streaming_initialize(player->streamer);
+ }
+
+ elemId = MMPLAYER_M_TYPEFIND;
+ element = gst_element_factory_make("typefind", "typefinder");
+ MMPLAYER_SIGNAL_CONNECT( player, element, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "have-type",
+ G_CALLBACK(__mmplayer_typefind_have_type), (gpointer)player );
+ }
+ else
+ {
+ elemId = MMPLAYER_M_AUTOPLUG;
+ element = __mmplayer_create_decodebin(player);
+ }
+
+ /* check autoplug element is OK */
+ if ( ! element )
+ {
+ debug_error("can not create element (%d)\n", elemId);
+ goto ERROR;
+ }
+
+ if (gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), element) == FALSE)
+ {
+ debug_error("failed to add sinkbin to pipeline\n");
+ gst_object_unref(GST_OBJECT(element));
+ element = NULL;
+ goto ERROR;
+ }
+
+ mainbin[elemId].id = elemId;
+ mainbin[elemId].gst = element;
+
+ if ( gst_element_link (mainbin[MMPLAYER_M_SRC].gst, mainbin[elemId].gst) == FALSE )
+ {
+ debug_error("Failed to link src - autoplug (or typefind)\n");
+ goto ERROR;
+ }
+
+ if (gst_element_set_state (mainbin[MMPLAYER_M_SRC].gst, target) == GST_STATE_CHANGE_FAILURE)
+ {
+ debug_error("Failed to change state of src element\n");
+ goto ERROR;
+ }
+
+ if (!MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ if (gst_element_set_state (mainbin[MMPLAYER_M_AUTOPLUG].gst, target) == GST_STATE_CHANGE_FAILURE)
+ {
+ debug_error("Failed to change state of decodebin2\n");
+ goto ERROR;
+ }
+ }
+ else
+ {
+ if (gst_element_set_state (mainbin[MMPLAYER_M_TYPEFIND].gst, target) == GST_STATE_CHANGE_FAILURE)
+ {
+ debug_error("Failed to change state of src element\n");
+ goto ERROR;
+ }
+ }
+
+ player->src_changed = TRUE;
+ MMPLAYER_FLEAVE();
+ return;
+
+ERROR:
+
+ MMPLAYER_PLAYBACK_UNLOCK(player);
+
+ if (!player->msg_posted)
+ {
+ MMPLAYER_POST_MSG(player, MM_MESSAGE_ERROR, &msg_param);
+ player->msg_posted = TRUE;
+ }
+ return;
+}
+
+static gboolean
+__mmplayer_deactivate_selector(mm_player_t *player, MMPlayerTrackType type)
+{
+ mm_player_selector_t *selector = &player->selector[type];
+ MMPlayerGstElement *sinkbin = NULL;
+ enum MainElementID selectorId = MMPLAYER_M_NUM;
+ enum MainElementID sinkId = MMPLAYER_M_NUM;
+ GstPad *srcpad = NULL;
+ GstPad *sinkpad = NULL;
+
+ MMPLAYER_FENTER();
+ return_val_if_fail (player, FALSE);
+
+ debug_log("type %d", type);
+
+ switch (type)
+ {
+ case MM_PLAYER_TRACK_TYPE_VIDEO:
+ selectorId = MMPLAYER_M_V_INPUT_SELECTOR;
+ sinkId = MMPLAYER_V_BIN;
+ sinkbin = player->pipeline->videobin;
+ break;
+ case MM_PLAYER_TRACK_TYPE_AUDIO:
+ selectorId = MMPLAYER_M_A_INPUT_SELECTOR;
+ sinkId = MMPLAYER_A_BIN;
+ sinkbin = player->pipeline->audiobin;
+ break;
+ case MM_PLAYER_TRACK_TYPE_TEXT:
+ selectorId = MMPLAYER_M_T_INPUT_SELECTOR;
+ sinkId = MMPLAYER_T_BIN;
+ sinkbin = player->pipeline->textbin;
+ break;
+ default:
+ debug_error("requested type is not supportable");
+ return FALSE;
+ break;
+ }
+
+ if (player->pipeline->mainbin[selectorId].gst)
+ {
+ gint n;
+
+ srcpad = gst_element_get_static_pad(player->pipeline->mainbin[selectorId].gst, "src");
+
+ if ((sinkbin) && (sinkbin[sinkId].gst))
+ {
+ sinkpad = gst_element_get_static_pad(sinkbin[sinkId].gst, "sink");
+
+ if (srcpad && sinkpad)
+ {
+ /* after getting drained signal there is no data flows, so no need to do pad_block */
+ debug_log("unlink %s:%s, %s:%s", GST_DEBUG_PAD_NAME(srcpad), GST_DEBUG_PAD_NAME(sinkpad));
+ gst_pad_unlink (srcpad, sinkpad);
+ }
+
+ gst_object_unref (sinkpad);
+ sinkpad = NULL;
+ }
+ gst_object_unref (srcpad);
+ srcpad = NULL;
+
+ debug_log("selector release");
+ if(selector->channels)
+ {
+ debug_log("release and unref request pads");
+
+ /* release and unref requests pad from the selector */
+ for (n = 0; n < selector->channels->len; n++)
+ {
+ GstPad *sinkpad = g_ptr_array_index (selector->channels, n);
+ gst_element_release_request_pad ((player->pipeline->mainbin[selectorId].gst), sinkpad);
+ }
+ g_ptr_array_set_size (selector->channels, 0);
+ }
+ gst_element_set_state (player->pipeline->mainbin[selectorId].gst, GST_STATE_NULL);
+ gst_bin_remove (GST_BIN_CAST (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), player->pipeline->mainbin[selectorId].gst);
+
+ player->pipeline->mainbin[selectorId].gst = NULL;
+ selector = NULL;
+ }
+
+ return TRUE;
+}
+
+static void
+__mmplayer_deactivate_old_path(mm_player_t *player)
+{
+ MMPLAYER_FENTER();
+ return_if_fail ( player );
+
+ g_object_set(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_DEC1].gst), "remove-buffer-signal", TRUE, NULL);
+
+ if ((!__mmplayer_deactivate_selector(player, MM_PLAYER_TRACK_TYPE_VIDEO)) ||
+ (!__mmplayer_deactivate_selector(player, MM_PLAYER_TRACK_TYPE_AUDIO)) ||
+ (!__mmplayer_deactivate_selector(player, MM_PLAYER_TRACK_TYPE_TEXT)))
+ {
+ debug_error("deactivate selector error");
+ goto ERROR;
+ }
+
+ __mmplayer_release_signal_connection( player, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG );
+
+ if (player->streamer)
+ {
+ __mm_player_streaming_deinitialize (player->streamer);
+ __mm_player_streaming_destroy(player->streamer);
+ player->streamer = NULL;
+ }
+
+ MMPLAYER_PLAYBACK_LOCK(player);
+ g_cond_signal( player->next_play_thread_cond );
+
+ MMPLAYER_FLEAVE();
+ return;
+
+ERROR:
+
+ if (!player->msg_posted)
+ {
+ MMMessageParamType msg = {0,};
+
+ /*post error*/
+ msg.code = MM_ERROR_PLAYER_INTERNAL;
+ debug_error("next_uri_play> deactivate error");
+
+ MMPLAYER_POST_MSG(player, MM_MESSAGE_ERROR, &msg);
+ player->msg_posted = TRUE;
+ }
+ return;
+}
+
+int _mmplayer_set_uri(MMHandleType hplayer, const char* uri)
+{
+ int result = MM_ERROR_NONE;
+ mm_player_t* player = (mm_player_t*) hplayer;
+ MMPLAYER_FENTER();
+
+ return_val_if_fail (player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ mm_attrs_set_string_by_name(player->attrs, "profile_uri", uri);
+ if (mmf_attrs_commit(player->attrs))
+ {
+ debug_error("failed to commit the original uri.\n");
+ result = MM_ERROR_PLAYER_INTERNAL;
+ }
+ else
+ {
+ if (_mmplayer_set_next_uri(hplayer, uri, TRUE) != MM_ERROR_NONE)
+ {
+ debug_error("failed to add the original uri in the uri list.\n");
+ }
+ }
+
+ MMPLAYER_FLEAVE();
+ return result;
+}
+
+int _mmplayer_set_next_uri(MMHandleType hplayer, const char* uri, bool is_first_path)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ guint num_of_list = 0;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail (player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ return_val_if_fail (uri, MM_ERROR_INVALID_ARGUMENT);
+
+ if (player->pipeline && player->pipeline->textbin)
+ {
+ debug_error("subtitle path is enabled.\n");
+ return MM_ERROR_PLAYER_INVALID_STATE;
+ }
+
+ num_of_list = g_list_length(player->uri_info.uri_list);
+
+ if (is_first_path == TRUE)
+ {
+ if (num_of_list == 0)
+ {
+ player->uri_info.uri_list = g_list_append(player->uri_info.uri_list, g_strdup(uri));
+ debug_log("add original path : %s", uri);
+ }
+ else
+ {
+ player->uri_info.uri_list = g_list_delete_link(player->uri_info.uri_list, g_list_nth(player->uri_info.uri_list, 0));
+ player->uri_info.uri_list = g_list_insert(player->uri_info.uri_list, g_strdup(uri), 0);
+
+ debug_log("change original path : %s", uri);
+ }
+ }
+ else
+ {
+ if (num_of_list == 0)
+ {
+ MMHandleType attrs = 0;
+ char *original_uri = NULL;
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( attrs )
+ {
+ mm_attrs_get_string_by_name(attrs, "profile_uri", &original_uri);
+
+ if (!original_uri)
+ {
+ debug_error("there is no original uri.");
+ return MM_ERROR_PLAYER_INVALID_STATE;
+ }
+
+ player->uri_info.uri_list = g_list_append(player->uri_info.uri_list, g_strdup(original_uri));
+ player->uri_info.uri_idx = 0;
+
+ debug_log("add original path at first : %s (%d)", original_uri);
+ }
+ }
+
+ player->uri_info.uri_list = g_list_append(player->uri_info.uri_list, g_strdup(uri));
+ debug_log("add new path : %s (total num of list = %d)", uri, g_list_length(player->uri_info.uri_list));
+ }
+
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+}
+
+int _mmplayer_get_next_uri(MMHandleType hplayer, char** uri)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ char *next_uri = NULL;
+ guint num_of_list = 0;
+
+ MMPLAYER_FENTER();
+ return_val_if_fail (player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ num_of_list = g_list_length(player->uri_info.uri_list);
+
+ if (num_of_list > 0)
+ {
+ gint uri_idx = player->uri_info.uri_idx;
+
+ if ( uri_idx < num_of_list-1 )
+ uri_idx++;
+ else
+ uri_idx = 0;
+
+ next_uri = g_list_nth_data(player->uri_info.uri_list, uri_idx);
+ debug_error("next uri idx : %d, uri = %s\n", uri_idx, next_uri);
+
+ *uri = g_strdup(next_uri);
+ }
+
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+}
+
+static void
+__mmplayer_gst_decode_unknown_type(GstElement *elem, GstPad* pad,
+GstCaps *caps, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ const gchar* klass = NULL;
+ const gchar* mime = NULL;
+ gchar* caps_str = NULL;
+
+ klass = gst_element_factory_get_klass (gst_element_get_factory(elem));
+ mime = gst_structure_get_name (gst_caps_get_structure(caps, 0));
+ caps_str = gst_caps_to_string(caps);
+
+ debug_warning("unknown type of caps : %s from %s",
+ caps_str, GST_ELEMENT_NAME (elem));
+
+ MMPLAYER_FREEIF(caps_str);
+
+ /* There is no available codec. */
+ __mmplayer_check_not_supported_codec (player, klass, mime);
+}
+
+static gboolean
+__mmplayer_gst_decode_autoplug_continue(GstElement *bin, GstPad* pad,
+GstCaps * caps, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ const char* mime = NULL;
+ gboolean ret = TRUE;
+
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
+ mime = gst_structure_get_name (gst_caps_get_structure(caps, 0));
+
+ if (g_str_has_prefix(mime, "audio")) {
+ GstStructure* caps_structure = NULL;
+ gint samplerate = 0;
+ gint channels = 0;
+ gchar *caps_str = NULL;
+
+ caps_structure = gst_caps_get_structure(caps, 0);
+ gst_structure_get_int (caps_structure, "rate", &samplerate);
+ gst_structure_get_int (caps_structure, "channels", &channels);
+
+ if ( (channels > 0 && samplerate == 0)) {
+ debug_log("exclude audio...");
+ ret = FALSE;
+ }
+
+ caps_str = gst_caps_to_string(caps);
+ /* set it directly because not sent by TAG */
+ if (g_strrstr(caps_str, "mobile-xmf")) {
+ mm_attrs_set_string_by_name(player->attrs, "content_audio_codec", "mobile-xmf");
+ }
+ MMPLAYER_FREEIF (caps_str);
+ } else if (g_str_has_prefix(mime, "video") && player->videodec_linked) {
+ debug_log("already video linked");
+ ret = FALSE;
+ } else {
+ debug_log("found new stream");
+ }
+
+ return ret;
+}
+
+static gint
+__mmplayer_gst_decode_autoplug_select(GstElement *bin, GstPad* pad,
+GstCaps* caps, GstElementFactory* factory, gpointer data)
+{
+ /* NOTE : GstAutoplugSelectResult is defined in gstplay-enum.h but not exposed
+ We are defining our own and will be removed when it actually exposed */
+ typedef enum {
+ GST_AUTOPLUG_SELECT_TRY,
+ GST_AUTOPLUG_SELECT_EXPOSE,
+ GST_AUTOPLUG_SELECT_SKIP
+ } GstAutoplugSelectResult;
+
+ GstAutoplugSelectResult result = GST_AUTOPLUG_SELECT_TRY;
+ mm_player_t* player = (mm_player_t*)data;
+
+ gchar* factory_name = NULL;
+ gchar* caps_str = NULL;
+ const gchar* klass = NULL;
+ gint idx = 0;
+
+ factory_name = GST_PLUGIN_FEATURE_NAME (factory);
+ klass = gst_element_factory_get_klass(factory);
+ caps_str = gst_caps_to_string(caps);
+
+// debug_log("found new element [%s] to link for caps [%s]", factory_name, caps_str);
+ debug_log("found new element [%s] to link", factory_name);
+
+ /* store type string */
+ if (player->type == NULL)
+ {
+ player->type = gst_caps_to_string(caps);
+
+ /* midi type should be stored because it will be used to set audio gain in avsysaudiosink */
+ if (__mmplayer_is_midi_type(player->type))
+ {
+ player->profile.play_mode = MM_PLAYER_MODE_MIDI;
+ player->bypass_audio_effect = TRUE;
+ }
+ else if (g_strrstr(player->type, "application/x-hls"))
+ {
+ /* If it can't know exact type when it parses uri because of redirection case,
+ * it will be fixed by typefinder here.
+ */
+ player->profile.uri_type = MM_PLAYER_URI_TYPE_HLS;
+ }
+ else if (g_strrstr(player->type, "application/dash+xml"))
+ {
+ player->profile.uri_type = MM_PLAYER_URI_TYPE_DASH;
+ }
+ }
+
+ /* filtering exclude keyword */
+ for ( idx = 0; player->ini.exclude_element_keyword[idx][0] != '\0'; idx++ )
+ {
+ if ( strstr(factory_name, player->ini.exclude_element_keyword[idx] ) )
+ {
+ debug_warning("skipping [%s] by exculde keyword [%s]\n",
+ factory_name, player->ini.exclude_element_keyword[idx] );
+
+ // NOTE : does we need to check n_value against the number of item selected?
+ result = GST_AUTOPLUG_SELECT_SKIP;
+ goto DONE;
+ }
+ }
+
+ /* check factory class for filtering */
+ /* NOTE : msl don't need to use image plugins.
+ * So, those plugins should be skipped for error handling.
+ */
+ if (g_strrstr(klass, "Codec/Decoder/Image"))
+ {
+ debug_log("skipping [%s] by not required\n", factory_name);
+ result = GST_AUTOPLUG_SELECT_SKIP;
+ goto DONE;
+ }
+
+ if (g_strrstr(factory_name, "mpegpsdemux"))
+ {
+ debug_log("skipping PS container - not support\n");
+ result = GST_AUTOPLUG_SELECT_SKIP;
+ goto DONE;
+ }
+
+ if (g_strrstr(factory_name, "mpegtsdemux") || g_strrstr(factory_name, "ssdemux"))
+ {
+ if (g_strrstr(factory_name, "ssdemux"))
+ player->smooth_streaming = TRUE;
+
+ if (g_strrstr(caps_str, "video/x-h264"))
+ {
+ player->needed_v_parser = TRUE;
+ }
+ else if (g_strrstr(caps_str, "video/mpeg"))
+ {
+ player->needed_v_parser = TRUE;
+ }
+ }
+
+ /* check ALP Codec can be used or not */
+ if ((g_strrstr(klass, "Codec/Decoder/Audio")))
+ {
+ GstStructure* str = NULL;
+ gint channels = 0;
+
+ str = gst_caps_get_structure( caps, 0 );
+ if ( str )
+ {
+ gst_structure_get_int (str, "channels", &channels);
+
+ debug_log ("check audio ch : %d %d\n", player->max_audio_channels, channels);
+ if (player->max_audio_channels < channels)
+ {
+ player->max_audio_channels = channels;
+ }
+ }
+
+ if (!player->audiodec_linked)
+ {
+ /* set stream information */
+ __mmplayer_set_audio_attrs (player, caps);
+ }
+ }
+ else if ((g_strrstr(klass, "Codec/Decoder/Video")))
+ {
+ if (g_strrstr(factory_name, "omx_"))
+ {
+ char *env = getenv ("MM_PLAYER_HW_CODEC_DISABLE");
+ if (env != NULL)
+ {
+ if (strncasecmp(env, "yes", 3) == 0)
+ {
+ debug_log ("skipping [%s] by disabled\n", factory_name);
+ result = GST_AUTOPLUG_SELECT_SKIP;
+ goto DONE;
+ }
+ }
+ }
+ }
+
+ if ((g_strrstr(klass, "Codec/Parser/Converter/Video")) ||
+ (g_strrstr(klass, "Codec/Decoder/Video")))
+ {
+ gint stype = 0;
+ gint width = 0;
+ GstStructure *str = NULL;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &stype);
+
+ /* don't make video because of not required */
+ if (stype == MM_DISPLAY_SURFACE_NULL)
+ {
+ debug_log ("no video because it's not required. -> return expose");
+ if (player->set_mode.media_packet_video_stream == FALSE)
+ {
+ result = GST_AUTOPLUG_SELECT_EXPOSE;
+ goto DONE;
+ }
+ }
+
+ /* get w/h for omx state-tune */
+ str = gst_caps_get_structure (caps, 0);
+ gst_structure_get_int (str, "width", &width);
+
+ if (width != 0) {
+ #define UHD_SIZE 3840*2160
+ gint height = 0;
+ gst_structure_get_int (str, "height", &height);
+
+ if (player->v_stream_caps) {
+ gst_caps_unref(player->v_stream_caps);
+ player->v_stream_caps = NULL;
+ }
+
+ player->v_stream_caps = gst_caps_copy(caps);
+ debug_log ("take caps for video state tune");
+ MMPLAYER_LOG_GST_CAPS_TYPE(player->v_stream_caps);
+
+ /* check video resolution */
+ if (width*height == UHD_SIZE) {
+ if ((util_is_miracast_connected()) && (player->msg_posted == FALSE)) {
+ MMMessageParamType msg = {0,};
+ /*post error*/
+ msg.code = MM_ERROR_PLAYER_RESOURCE_LIMIT;
+ debug_error("can't play in miracast");
+ MMPLAYER_POST_MSG(player, MM_MESSAGE_ERROR, &msg);
+ player->msg_posted = TRUE;
+ }
+ }
+ }
+ }
+ if ((g_strrstr(klass, "Codec/Parser/Audio")))
+ {
+ if (player->smooth_streaming && (g_strrstr(caps_str, "audio/mpeg")))
+ {
+ debug_log ("ssdemux can't use audio parse");
+ result = GST_AUTOPLUG_SELECT_SKIP;
+ goto DONE;
+ }
+
+ }
+
+ if (g_strrstr(klass, "Decoder"))
+ {
+ const char* mime = NULL;
+ mime = gst_structure_get_name (gst_caps_get_structure(caps, 0));
+
+ if (g_str_has_prefix(mime, "video"))
+ {
+ // __mmplayer_check_video_zero_cpoy(player, factory);
+
+ player->not_supported_codec &= MISSING_PLUGIN_AUDIO;
+ player->can_support_codec |= FOUND_PLUGIN_VIDEO;
+
+ player->videodec_linked = 1;
+ }
+ else if(g_str_has_prefix(mime, "audio"))
+ {
+ player->not_supported_codec &= MISSING_PLUGIN_VIDEO;
+ player->can_support_codec |= FOUND_PLUGIN_AUDIO;
+
+ player->audiodec_linked = 1;
+ }
+ }
+
+DONE:
+ MMPLAYER_FREEIF(caps_str);
+
+ return result;
+}
+
+
+#if 0
+static GValueArray*
+__mmplayer_gst_decode_autoplug_factories(GstElement *bin, GstPad* pad,
+GstCaps * caps, gpointer data)
+{
+ //mm_player_t* player = (mm_player_t*)data;
+
+ debug_log("decodebin is requesting factories for caps [%s] from element[%s]",
+ gst_caps_to_string(caps),
+ GST_ELEMENT_NAME(GST_PAD_PARENT(pad)));
+
+ return NULL;
+}
+#endif
+
+static GValueArray*
+__mmplayer_gst_decode_autoplug_sort(GstElement *bin, GstPad* pad,
+GstCaps* caps, GValueArray* factories, gpointer data) // @
+{
+#define DEFAULT_IDX 0xFFFF
+
+ guint idx = 0;
+ guint a_omx_idx = DEFAULT_IDX;
+ guint a_dec_idx = DEFAULT_IDX;
+ guint v_dec_idx = DEFAULT_IDX;
+ guint v_parser_idx = DEFAULT_IDX;
+ guint new_pos = DEFAULT_IDX;
+ guint elem_idx = DEFAULT_IDX;
+
+ GValueArray* new_factories = NULL;
+ GValue val = { 0, };
+
+ GstElementFactory *factory = NULL;
+ const gchar* klass = NULL;
+ gchar* factory_name = NULL;
+ gchar* caps_str = NULL;
+
+ mm_player_t* player = (mm_player_t*)data;
+ caps_str = gst_caps_to_string(caps);
+
+ //debug_log("autoplug-sort signal [num: %d]\n", factories->n_values);
+ debug_log("requesting custom sorting for the factories for caps [%s]", caps_str);
+
+ MMPLAYER_FREEIF(caps_str);
+
+ for(idx=0; idx < factories->n_values;idx++)
+ {
+ factory = g_value_get_object (g_value_array_get_nth (factories, idx));
+
+ klass = gst_element_factory_get_klass(factory);
+ factory_name = GST_PLUGIN_FEATURE_NAME (factory);
+
+ //debug_log("Klass [%s] Factory [%s]\n", klass, factory_name);
+
+ if (g_strrstr(klass, "Codec/Decoder/Audio"))
+ {
+ if (a_dec_idx == DEFAULT_IDX)
+ a_dec_idx = idx;
+
+ /* check ALP Codec can be used or not */
+ if ((a_omx_idx == DEFAULT_IDX) &&
+ (g_strrstr(factory_name, "omx_mp3dec")) &&
+ (__mmplayer_is_omx_decoder_type(player)))
+ {
+ debug_log("--> mp3_local : Priority of omx is higher than the others\n");
+ a_omx_idx = idx;
+ break;
+ }
+ }
+ else if ((v_dec_idx == DEFAULT_IDX) && (g_strrstr(klass, "Codec/Decoder/Video")))
+ {
+ v_dec_idx = idx;
+ }
+
+ if ((v_parser_idx == DEFAULT_IDX) &&
+ (((player->smooth_streaming) &&
+ (g_strrstr(factory_name, "legacyh264parse"))) ||
+ // (g_strrstr(factory_name, "h264parse")) ||
+ (g_strrstr(factory_name, "mpeg4videoparse"))))
+ {
+ v_parser_idx = idx;
+ }
+ }
+
+ if (player->needed_v_parser || player->smooth_streaming)
+ {
+ debug_log("needed parser? %s, ss? %s\n",
+ (player->needed_v_parser)?"OO":"XX",
+ (player->smooth_streaming)?"OO":"XX");
+ }
+
+ new_pos = (a_dec_idx!=DEFAULT_IDX)?(a_dec_idx):((player->needed_v_parser)?(v_dec_idx):(v_parser_idx));
+ elem_idx = (a_omx_idx!=DEFAULT_IDX)?(a_omx_idx):((player->needed_v_parser)?(v_parser_idx):(v_dec_idx));
+
+ /* consider file mp3 audio only */
+ if ((elem_idx != DEFAULT_IDX) && (new_pos < elem_idx))
+ {
+ debug_log("[Re-arrange] factories sequence, new_pos : %d, elem_idx : %d\n", new_pos, elem_idx);
+
+ new_factories = g_value_array_copy (factories);
+
+ // insert omx decoder in front of other decoders
+ factory = g_value_get_object (g_value_array_get_nth (factories, elem_idx));
+
+ g_value_init (&val, G_TYPE_OBJECT);
+ g_value_set_object (&val, factory);
+ g_value_array_insert (new_factories, new_pos, &val);
+ g_value_unset (&val);
+
+ // remove duplicated omx element
+ g_value_array_remove(new_factories, elem_idx+1);
+
+ for(idx=0; idx < new_factories->n_values;idx++)
+ {
+ factory = g_value_get_object (g_value_array_get_nth (new_factories, idx));
+
+ debug_log("[Re-arranged] Klass [%s] Factory [%s]\n",
+ gst_element_factory_get_klass(factory), GST_PLUGIN_FEATURE_NAME (factory));
+ }
+ }
+
+ /* returning NULL to keep current order */
+ return (new_factories)?(new_factories):(NULL);
+}
+
+static void
+__mmplayer_gst_decode_pad_removed(GstElement *elem, GstPad* new_pad,
+gpointer data) // @
+{
+ //mm_player_t* player = (mm_player_t*)data;
+ GstCaps* caps = NULL;
+
+ debug_log("[Decodebin2] pad-removed signal\n");
+
+ caps = gst_pad_get_caps(new_pad);
+ if (caps)
+ {
+ gchar* caps_str = NULL;
+ caps_str = gst_caps_to_string(caps);
+
+ debug_log("pad removed caps : %s from %s", caps_str, GST_ELEMENT_NAME(elem) );
+
+ MMPLAYER_FREEIF(caps_str);
+ }
+}
+
+static void
+__mmplayer_gst_decode_drained(GstElement *bin, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+
+ MMPLAYER_FENTER();
+ return_if_fail ( player );
+
+ debug_log("__mmplayer_gst_decode_drained");
+
+ if (player->use_deinterleave == TRUE)
+ {
+ debug_log("group playing mode.");
+ return;
+ }
+
+ if ( !player->cmd_lock )
+ {
+ debug_warning("can't get cmd lock\n");
+ return;
+ }
+
+ if (!g_mutex_trylock(player->cmd_lock))
+ {
+ debug_warning("Fail to get cmd lock");
+ return;
+ }
+
+ if (!__mmplayer_verify_next_play_path(player))
+ {
+ debug_log("decoding is finished.");
+ g_mutex_unlock(player->cmd_lock);
+ return;
+ }
+
+ player->pp_rebuilding = TRUE;
+
+ /* deactivate pipeline except sinkbins to set up the new pipeline of next uri*/
+ __mmplayer_deactivate_old_path(player);
+
+ g_mutex_unlock(player->cmd_lock);
+
+ MMPLAYER_FLEAVE();
+}
+
+static void
+__mmplayer_gst_element_added (GstElement *bin, GstElement *element, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ const gchar* klass = NULL;
+ gchar* factory_name = NULL;
+
+ klass = gst_element_factory_get_klass (gst_element_get_factory(element));
+ factory_name = GST_PLUGIN_FEATURE_NAME (gst_element_get_factory(element));
+
+ debug_log("new elem klass: %s, factory_name: %s, new elem name : %s\n", klass, factory_name, GST_ELEMENT_NAME(element));
+
+ if (__mmplayer_add_dump_buffer_probe(player, element))
+ debug_log("add buffer probe");
+
+ //<-
+ if (g_strrstr(klass, "Codec/Decoder/Audio"))
+ {
+ gchar* selected = NULL;
+ selected = g_strdup( GST_ELEMENT_NAME(element));
+ player->audio_decoders = g_list_append (player->audio_decoders, selected);
+ }
+ //-> temp code
+
+ if (g_strrstr(klass, "Parser"))
+ {
+ gchar* selected = NULL;
+
+ selected = g_strdup (factory_name);
+ player->parsers = g_list_append (player->parsers, selected);
+ }
+
+ if (g_strrstr(klass, "Demux") || g_strrstr(klass, "Parse"))
+ {
+ /* FIXIT : first value will be overwritten if there's more
+ * than 1 demuxer/parser
+ */
+
+ //debug_log ("plugged element is demuxer. take it\n");
+ player->pipeline->mainbin[MMPLAYER_M_DEMUX].id = MMPLAYER_M_DEMUX;
+ player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst = element;
+
+ /*Added for multi audio support */ // Q. del?
+ if (g_strrstr(klass, "Demux"))
+ {
+ player->pipeline->mainbin[MMPLAYER_M_DEMUX_EX].id = MMPLAYER_M_DEMUX_EX;
+ player->pipeline->mainbin[MMPLAYER_M_DEMUX_EX].gst = element;
+ }
+ }
+
+ if (g_strrstr(factory_name, "asfdemux") || g_strrstr(factory_name, "qtdemux") || g_strrstr(factory_name, "avidemux"))
+ {
+ int surface_type = 0;
+
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &surface_type);
+
+ /* playback protection if drm file */
+ if (player->use_video_stream || surface_type == MM_DISPLAY_SURFACE_EVAS || surface_type == MM_DISPLAY_SURFACE_X_EXT)
+ {
+ debug_log("playback can be protected if playready drm");
+ g_object_set (G_OBJECT(element), "playback-protection", TRUE, NULL);
+ }
+ }
+
+ // to support trust-zone only
+ if (g_strrstr(factory_name, "asfdemux"))
+ {
+ debug_log ("set file-location %s\n", player->profile.uri);
+ g_object_set (G_OBJECT(element), "file-location", player->profile.uri, NULL);
+
+ if (player->video_hub_download_mode == TRUE)
+ {
+ g_object_set (G_OBJECT(element), "downloading-mode", player->video_hub_download_mode, NULL);
+ }
+ }
+ else if (g_strrstr(factory_name, "legacyh264parse")) // ssdemux
+ {
+ debug_log ("[ssdemux] output-format to legacyh264parse\n");
+ g_object_set (G_OBJECT(element), "output-format", 1, NULL); /* NALU/Byte Stream format */
+ }
+ else if (g_strrstr(factory_name, "mpegaudioparse"))
+ {
+ if ((MMPLAYER_IS_HTTP_STREAMING(player)) &&
+ (__mmplayer_is_only_mp3_type(player->type)))
+ {
+ debug_log ("[mpegaudioparse] set streaming pull mode.");
+ g_object_set(G_OBJECT(element), "http-pull-mp3dec", TRUE, NULL);
+ }
+ }
+ else if (g_strrstr(factory_name, "omx_"))
+ {
+ if (g_strrstr(klass, "Codec/Decoder/Video"))
+ {
+ gboolean ret = FALSE;
+
+ if (player->v_stream_caps != NULL)
+ {
+ GstPad *pad = gst_element_get_static_pad(element, "sink");
+
+ if (pad)
+ {
+ ret = gst_pad_set_caps(pad, player->v_stream_caps);
+ debug_log("found omx decoder, setting gst_pad_set_caps for omx (ret:%d)", ret);
+ MMPLAYER_LOG_GST_CAPS_TYPE(player->v_stream_caps);
+ gst_object_unref (pad);
+ }
+ }
+ g_object_set (G_OBJECT(element), "state-tuning", TRUE, NULL);
+ }
+#ifdef _MM_PLAYER_ALP_PARSER
+ if (g_strrstr(factory_name, "omx_mp3dec"))
+ {
+ g_list_foreach (player->parsers, check_name, player);
+ }
+#endif
+ player->pipeline->mainbin[MMPLAYER_M_DEC1].gst = element;
+ }
+
+ if (g_strrstr(GST_ELEMENT_NAME(element), "multiqueue"))
+ {
+ debug_log ("plugged element is multiqueue. take it\n");
+ player->pipeline->mainbin[MMPLAYER_M_DEMUXED_S_BUFFER].id = MMPLAYER_M_DEMUXED_S_BUFFER;
+ player->pipeline->mainbin[MMPLAYER_M_DEMUXED_S_BUFFER].gst = element;
+
+ if (MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ __mm_player_streaming_set_multiqueue(player->streamer,
+ element,
+ TRUE,
+ MAX_DECODEBIN_BUFFER_BYTES, // player->ini.http_max_size_bytes,
+ player->ini.http_buffering_time,
+ 1.0,
+ player->ini.http_buffering_limit);
+
+ __mm_player_streaming_sync_property(player->streamer, player->pipeline->mainbin[MMPLAYER_M_AUTOPLUG].gst);
+ }
+ }
+
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-added" );
+ return;
+}
+
+static gboolean __mmplayer_configure_audio_callback(mm_player_t* player)
+{
+ MMPLAYER_FENTER();
+ return_val_if_fail ( player, FALSE );
+
+ if ( MMPLAYER_IS_STREAMING(player) )
+ return FALSE;
+
+ /* This callback can be set to music player only. */
+ if((player->can_support_codec & 0x02) == FOUND_PLUGIN_VIDEO)
+ {
+ debug_warning("audio callback is not supported for video");
+ return FALSE;
+ }
+
+ if (player->audio_stream_cb)
+ {
+ {
+ GstPad *pad = NULL;
+
+ pad = gst_element_get_static_pad (player->pipeline->audiobin[MMPLAYER_A_SINK].gst, "sink");
+
+ if ( !pad )
+ {
+ debug_error("failed to get sink pad from audiosink to probe data\n");
+ return FALSE;
+ }
+
+ player->audio_cb_probe_id = gst_pad_add_buffer_probe (pad,
+ G_CALLBACK (__mmplayer_audio_stream_probe), player);
+
+ gst_object_unref (pad);
+
+ pad = NULL;
+ }
+ }
+ else
+ {
+ debug_error("There is no audio callback to configure.\n");
+ return FALSE;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
+}
+
+static void
+__mmplayer_init_factories(mm_player_t* player) // @
+{
+ return_if_fail ( player );
+
+ player->factories = gst_registry_feature_filter(gst_registry_get_default(),
+ (GstPluginFeatureFilter)__mmplayer_feature_filter, FALSE, NULL);
+ player->factories = g_list_sort(player->factories, (GCompareFunc)util_factory_rank_compare);
+}
+
+static void
+__mmplayer_release_factories(mm_player_t* player) // @
+{
+ MMPLAYER_FENTER();
+ return_if_fail ( player );
+
+ if (player->factories)
+ {
+ gst_plugin_feature_list_free (player->factories);
+ player->factories = NULL;
+ }
+
+ MMPLAYER_FLEAVE();
+}
+
+static void
+__mmplayer_release_misc(mm_player_t* player)
+{
+ int i;
+ gboolean cur_mode = player->set_mode.rich_audio;
+ MMPLAYER_FENTER();
+
+ return_if_fail ( player );
+
+ player->use_video_stream = FALSE;
+ player->video_stream_cb = NULL;
+ player->video_stream_cb_user_param = NULL;
+
+ player->audio_stream_cb = NULL;
+ player->audio_stream_cb_user_param = NULL;
+
+ player->sent_bos = FALSE;
+ player->playback_rate = DEFAULT_PLAYBACK_RATE;
+
+ player->doing_seek = FALSE;
+
+ player->updated_bitrate_count = 0;
+ player->total_bitrate = 0;
+ player->updated_maximum_bitrate_count = 0;
+ player->total_maximum_bitrate = 0;
+
+ player->not_found_demuxer = 0;
+
+ player->last_position = 0;
+ player->duration = 0;
+ player->http_content_size = 0;
+ player->not_supported_codec = MISSING_PLUGIN_NONE;
+ player->can_support_codec = FOUND_PLUGIN_NONE;
+ player->pending_seek.is_pending = FALSE;
+ player->pending_seek.format = MM_PLAYER_POS_FORMAT_TIME;
+ player->pending_seek.pos = 0;
+ player->msg_posted = FALSE;
+ player->has_many_types = FALSE;
+ player->is_drm_file = FALSE;
+ player->max_audio_channels = 0;
+ player->video_share_api_delta = 0;
+ player->video_share_clock_delta = 0;
+ player->sm.keep_last_pos = FALSE;
+ player->display_stat = MMPLAYER_DISPLAY_STATUS_NULL;
+ player->is_subtitle_force_drop = FALSE;
+ player->play_subtitle = FALSE;
+ player->use_textoverlay = FALSE;
+ player->adjust_subtitle_pos = 0;
+ player->last_multiwin_status = FALSE;
+ player->has_closed_caption = FALSE;
+ player->set_mode.media_packet_video_stream = FALSE;
+
+ memset(&player->set_mode, 0, sizeof(MMPlayerSetMode));
+ /* recover mode */
+ player->set_mode.rich_audio = cur_mode;
+
+ for (i = 0; i < MM_PLAYER_STREAM_COUNT_MAX; i++)
+ {
+ player->bitrate[i] = 0;
+ player->maximum_bitrate[i] = 0;
+ }
+
+ /* free memory related to audio effect */
+ MMPLAYER_FREEIF(player->audio_effect_info.custom_ext_level_for_plugin);
+
+ if (player->state_tune_caps)
+ {
+ gst_caps_unref(player->state_tune_caps);
+ player->state_tune_caps = NULL;
+ }
+
+ if (player->video_cb_probe_id)
+ {
+ GstPad *pad = NULL;
+
+ pad = gst_element_get_static_pad (player->video_fakesink, "sink");
+
+ if (pad) {
+ debug_log("release video probe\n");
+
+ /* release audio callback */
+ gst_pad_remove_buffer_probe (pad, player->video_cb_probe_id);
+ player->video_cb_probe_id = 0;
+ player->video_stream_cb = NULL;
+ player->video_stream_cb_user_param = NULL;
+ }
+ }
+
+ MMPLAYER_FLEAVE();
+}
+
+static void
+__mmplayer_release_misc_post(mm_player_t* player)
+{
+ char *original_uri = NULL;
+ MMPLAYER_FENTER();
+
+ /* player->pipeline is already released before. */
+
+ return_if_fail ( player );
+
+ mm_attrs_set_int_by_name(player->attrs, "content_video_found", 0);
+ mm_attrs_set_int_by_name(player->attrs, "content_audio_found", 0);
+
+ /* clean found parsers */
+ if (player->parsers)
+ {
+ GList *parsers = player->parsers;
+ for ( ;parsers ; parsers = g_list_next(parsers))
+ {
+ gchar *name = parsers->data;
+ MMPLAYER_FREEIF(name);
+ }
+ g_list_free(player->parsers);
+ player->parsers = NULL;
+ }
+
+ /* clean found audio decoders */
+ if (player->audio_decoders)
+ {
+ GList *a_dec = player->audio_decoders;
+ for ( ;a_dec ; a_dec = g_list_next(a_dec))
+ {
+ gchar *name = a_dec->data;
+ MMPLAYER_FREEIF(name);
+ }
+ g_list_free(player->audio_decoders);
+ player->audio_decoders = NULL;
+ }
+
+ /* clean the uri list except original uri */
+ if (player->uri_info.uri_list)
+ {
+ original_uri = g_list_nth_data(player->uri_info.uri_list, 0);
+
+ if (player->attrs)
+ {
+ mm_attrs_set_string_by_name(player->attrs, "profile_uri", original_uri);
+ debug_log("restore original uri = %s\n", original_uri);
+
+ if (mmf_attrs_commit(player->attrs))
+ {
+ debug_error("failed to commit the original uri.\n");
+ }
+ }
+
+ GList *uri_list = player->uri_info.uri_list;
+ for ( ;uri_list ; uri_list = g_list_next(uri_list))
+ {
+ gchar *uri = uri_list->data;
+ MMPLAYER_FREEIF(uri);
+ }
+ g_list_free(player->uri_info.uri_list);
+ player->uri_info.uri_list = NULL;
+ }
+
+ player->uri_info.uri_idx = 0;
+ player->src_changed = FALSE;
+
+ MMPLAYER_FLEAVE();
+}
+
+static GstElement *__mmplayer_element_create_and_link(mm_player_t *player, GstPad* pad, const char* name)
+{
+ GstElement *element = NULL;
+ GstPad *sinkpad;
+
+ debug_log("creating %s to plug\n", name);
+
+ element = gst_element_factory_make(name, NULL);
+ if ( ! element )
+ {
+ debug_error("failed to create queue\n");
+ return NULL;
+ }
+
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(element, GST_STATE_READY) )
+ {
+ debug_error("failed to set state READY to %s\n", name);
+ gst_object_unref (element);
+ return NULL;
+ }
+
+ if ( ! gst_bin_add(GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), element))
+ {
+ debug_error("failed to add %s\n", name);
+ gst_object_unref (element);
+ return NULL;
+ }
+
+ sinkpad = gst_element_get_static_pad(element, "sink");
+
+ if ( GST_PAD_LINK_OK != gst_pad_link(pad, sinkpad) )
+ {
+ debug_error("failed to link %s\n", name);
+ gst_object_unref (sinkpad);
+ gst_object_unref (element);
+ return NULL;
+ }
+
+ debug_log("linked %s to pipeline successfully\n", name);
+
+ gst_object_unref (sinkpad);
+
+ return element;
+}
+
+static gboolean
+__mmplayer_close_link(mm_player_t* player, GstPad *srcpad, GstElement *sinkelement,
+const char *padname, const GList *templlist)
+{
+ GstPad *pad = NULL;
+ gboolean has_dynamic_pads = FALSE;
+ gboolean has_many_types = FALSE;
+ const char *klass = NULL;
+ GstStaticPadTemplate *padtemplate = NULL;
+ GstElementFactory *factory = NULL;
+ GstElement* queue = NULL;
+ GstElement* parser = NULL;
+ GstPad *pssrcpad = NULL;
+ GstPad *qsrcpad = NULL, *qsinkpad = NULL;
+ MMPlayerGstElement *mainbin = NULL;
+ GstStructure* str = NULL;
+ GstCaps* srccaps = NULL;
+ GstState target_state = GST_STATE_READY;
+ gboolean isvideo_decoder = FALSE;
+ guint q_max_size_time = 0;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player &&
+ player->pipeline &&
+ player->pipeline->mainbin,
+ FALSE );
+
+ mainbin = player->pipeline->mainbin;
+
+ debug_log("plugging pad %s:%s to newly create %s:%s\n",
+ GST_ELEMENT_NAME( GST_PAD_PARENT ( srcpad ) ),
+ GST_PAD_NAME( srcpad ),
+ GST_ELEMENT_NAME( sinkelement ),
+ padname);
+
+ factory = gst_element_get_factory(sinkelement);
+ klass = gst_element_factory_get_klass(factory);
+
+ /* check if player can do start continually */
+ MMPLAYER_CHECK_CMD_IF_EXIT(player);
+
+ /* need it to warm up omx before linking to pipeline */
+ if (g_strrstr(GST_ELEMENT_NAME( GST_PAD_PARENT ( srcpad ) ), "demux"))
+ {
+ debug_log("get demux caps.\n");
+ if (player->state_tune_caps)
+ {
+ gst_caps_unref(player->state_tune_caps);
+ player->state_tune_caps = NULL;
+ }
+ player->state_tune_caps = gst_caps_copy(GST_PAD_CAPS(srcpad));
+ }
+
+ /* NOTE : OMX Codec can check if resource is available or not at this state. */
+ if (g_strrstr(GST_ELEMENT_NAME(sinkelement), "omx"))
+ {
+ if (player->state_tune_caps != NULL)
+ {
+ debug_log("set demux's caps to omx codec if resource is available");
+ if (gst_pad_set_caps(gst_element_get_static_pad(sinkelement, "sink"), player->state_tune_caps))
+ {
+ target_state = GST_STATE_PAUSED;
+ isvideo_decoder = TRUE;
+ g_object_set(G_OBJECT(sinkelement), "state-tuning", TRUE, NULL);
+ }
+ else
+ {
+ debug_warning("failed to set caps for state tuning");
+ }
+ }
+ gst_caps_unref(player->state_tune_caps);
+ player->state_tune_caps = NULL;
+ }
+
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(sinkelement, target_state) )
+ {
+ debug_error("failed to set %d state to %s\n", target_state, GST_ELEMENT_NAME( sinkelement ));
+ if (isvideo_decoder)
+ {
+ gst_element_set_state(sinkelement, GST_STATE_NULL);
+ gst_object_unref(G_OBJECT(sinkelement));
+ player->keep_detecting_vcodec = TRUE;
+ }
+ goto ERROR;
+ }
+
+ /* add to pipeline */
+ if ( ! gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), sinkelement) )
+ {
+ debug_error("failed to add %s to mainbin\n", GST_ELEMENT_NAME( sinkelement ));
+ goto ERROR;
+ }
+
+ debug_log("element klass : %s\n", klass);
+
+ /* added to support multi track files */
+ /* only decoder case and any of the video/audio still need to link*/
+ if(g_strrstr(klass, "Decoder") && __mmplayer_link_decoder(player,srcpad))
+ {
+ gchar *name = g_strdup(GST_ELEMENT_NAME( GST_PAD_PARENT ( srcpad )));
+
+ if (g_strrstr(name, "mpegtsdemux")|| g_strrstr(name, "ssdemux"))
+ {
+ gchar *src_demux_caps_str = NULL;
+ gchar *needed_parser = NULL;
+ GstCaps *src_demux_caps = NULL;
+ gboolean smooth_streaming = FALSE;
+
+ src_demux_caps = gst_pad_get_caps(srcpad);
+ src_demux_caps_str = gst_caps_to_string(src_demux_caps);
+
+ gst_caps_unref(src_demux_caps);
+
+ if (g_strrstr(src_demux_caps_str, "video/x-h264"))
+ {
+ if (g_strrstr(name, "ssdemux"))
+ {
+ needed_parser = g_strdup("legacyh264parse");
+ smooth_streaming = TRUE;
+ }
+ else
+ {
+ needed_parser = g_strdup("h264parse");
+ }
+ }
+ else if (g_strrstr(src_demux_caps_str, "video/mpeg"))
+ {
+ needed_parser = g_strdup("mpeg4videoparse");
+ }
+ MMPLAYER_FREEIF(src_demux_caps_str);
+
+ if (needed_parser)
+ {
+ parser = __mmplayer_element_create_and_link(player, srcpad, needed_parser);
+ MMPLAYER_FREEIF(needed_parser);
+
+ if ( !parser )
+ {
+ debug_error("failed to create parser\n");
+ }
+ else
+ {
+ if (smooth_streaming)
+ {
+ g_object_set (parser, "output-format", 1, NULL); /* NALU/Byte Stream format */
+ }
+
+ /* update srcpad if parser is created */
+ pssrcpad = gst_element_get_static_pad(parser, "src");
+ srcpad = pssrcpad;
+ }
+ }
+ }
+ MMPLAYER_FREEIF(name);
+
+ queue = __mmplayer_element_create_and_link(player, srcpad, "queue"); // parser - queue or demuxer - queue
+ if ( ! queue )
+ {
+ debug_error("failed to create queue\n");
+ goto ERROR;
+ }
+
+ /* update srcpad to link with decoder */
+ qsrcpad = gst_element_get_static_pad(queue, "src");
+ srcpad = qsrcpad;
+
+ q_max_size_time = GST_QUEUE_DEFAULT_TIME;
+
+ /* assigning queue handle for futher manipulation purpose */
+ /* FIXIT : make it some kind of list so that msl can support more then two stream (text, data, etc...) */
+ if(mainbin[MMPLAYER_M_Q1].gst == NULL)
+ {
+ mainbin[MMPLAYER_M_Q1].id = MMPLAYER_M_Q1;
+ mainbin[MMPLAYER_M_Q1].gst = queue;
+
+ if (player->profile.uri_type == MM_PLAYER_URI_TYPE_SS)
+ {
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q1].gst), "max-size-time", 0 , NULL);
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q1].gst), "max-size-buffers", 2, NULL);
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q1].gst), "max-size-bytes", 0, NULL);
+ }
+ else
+ {
+ if (!MMPLAYER_IS_RTSP_STREAMING(player))
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q1].gst), "max-size-time", q_max_size_time * GST_SECOND, NULL);
+ }
+ }
+ else if(mainbin[MMPLAYER_M_Q2].gst == NULL)
+ {
+ mainbin[MMPLAYER_M_Q2].id = MMPLAYER_M_Q2;
+ mainbin[MMPLAYER_M_Q2].gst = queue;
+
+ if (player->profile.uri_type == MM_PLAYER_URI_TYPE_SS)
+ {
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q2].gst), "max-size-time", 0 , NULL);
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q2].gst), "max-size-buffers", 2, NULL);
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q2].gst), "max-size-bytes", 0, NULL);
+ }
+ else
+ {
+ if (!MMPLAYER_IS_RTSP_STREAMING(player))
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_Q2].gst), "max-size-time", q_max_size_time * GST_SECOND, NULL);
+ }
+ }
+ else
+ {
+ debug_error("Not supporting more then two elementary stream\n");
+ g_assert(1);
+ }
+
+ pad = gst_element_get_static_pad(sinkelement, padname);
+
+ if ( ! pad )
+ {
+ debug_warning("failed to get pad(%s) from %s. retrying with [sink]\n",
+ padname, GST_ELEMENT_NAME(sinkelement) );
+
+ pad = gst_element_get_static_pad(sinkelement, "sink");
+ if ( ! pad )
+ {
+ debug_error("failed to get pad(sink) from %s. \n",
+ GST_ELEMENT_NAME(sinkelement) );
+ goto ERROR;
+ }
+ }
+
+ /* to check the video/audio type set the proper flag*/
+ const gchar *mime_type = NULL;
+ {
+ srccaps = gst_pad_get_caps( srcpad );
+ if ( !srccaps )
+ goto ERROR;
+
+ str = gst_caps_get_structure( srccaps, 0 );
+ if ( ! str )
+ goto ERROR;
+
+ mime_type = gst_structure_get_name(str);
+ if ( ! mime_type )
+ goto ERROR;
+ }
+
+ /* link queue and decoder. so, it will be queue - decoder. */
+ if ( GST_PAD_LINK_OK != gst_pad_link(srcpad, pad) )
+ {
+ gst_object_unref(GST_OBJECT(pad));
+ debug_error("failed to link (%s) to pad(%s)\n", GST_ELEMENT_NAME( sinkelement ), padname );
+
+ /* reconstitute supportable codec */
+ if (strstr(mime_type, "video"))
+ {
+ player->can_support_codec ^= FOUND_PLUGIN_VIDEO;
+ }
+ else if (strstr(mime_type, "audio"))
+ {
+ player->can_support_codec ^= FOUND_PLUGIN_AUDIO;
+ }
+ goto ERROR;
+ }
+
+ if (strstr(mime_type, "video"))
+ {
+ player->videodec_linked = 1;
+ debug_msg("player->videodec_linked set to 1\n");
+
+ }
+ else if (strstr(mime_type, "audio"))
+ {
+ player->audiodec_linked = 1;
+ debug_msg("player->auddiodec_linked set to 1\n");
+ }
+
+ gst_object_unref(GST_OBJECT(pad));
+ gst_caps_unref(GST_CAPS(srccaps));
+ srccaps = NULL;
+ }
+
+ if ( !MMPLAYER_IS_HTTP_PD(player) )
+ {
+ if( (g_strrstr(klass, "Demux") && !g_strrstr(klass, "Metadata")) || (g_strrstr(klass, "Parser") ) )
+ {
+ if (MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ GstFormat fmt = GST_FORMAT_BYTES;
+ gint64 dur_bytes = 0L;
+ gchar *file_buffering_path = NULL;
+ gboolean use_file_buffer = FALSE;
+
+ if ( !mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst)
+ {
+ debug_log("creating http streaming buffering queue\n");
+
+ queue = gst_element_factory_make("queue2", "queue2");
+ if ( ! queue )
+ {
+ debug_error ( "failed to create buffering queue element\n" );
+ goto ERROR;
+ }
+
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(queue, GST_STATE_READY) )
+ {
+ debug_error("failed to set state READY to buffering queue\n");
+ goto ERROR;
+ }
+
+ if ( !gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue) )
+ {
+ debug_error("failed to add buffering queue\n");
+ goto ERROR;
+ }
+
+ qsinkpad = gst_element_get_static_pad(queue, "sink");
+ qsrcpad = gst_element_get_static_pad(queue, "src");
+
+ if ( GST_PAD_LINK_OK != gst_pad_link(srcpad, qsinkpad) )
+ {
+ debug_error("failed to link buffering queue\n");
+ goto ERROR;
+ }
+ srcpad = qsrcpad;
+
+
+ mainbin[MMPLAYER_M_MUXED_S_BUFFER].id = MMPLAYER_M_MUXED_S_BUFFER;
+ mainbin[MMPLAYER_M_MUXED_S_BUFFER].gst = queue;
+
+ if ( !MMPLAYER_IS_HTTP_LIVE_STREAMING(player))
+ {
+ if ( !gst_element_query_duration(player->pipeline->mainbin[MMPLAYER_M_SRC].gst, &fmt, &dur_bytes))
+ debug_error("fail to get duration.\n");
+
+ if (dur_bytes > 0)
+ {
+ use_file_buffer = MMPLAYER_USE_FILE_FOR_BUFFERING(player);
+ file_buffering_path = g_strdup(player->ini.http_file_buffer_path);
+ }
+ else
+ {
+ dur_bytes = 0;
+ }
+ }
+
+ /* NOTE : we cannot get any duration info from ts container in case of streaming */
+ if(!g_strrstr(GST_ELEMENT_NAME(sinkelement), "mpegtsdemux"))
+ {
+ __mm_player_streaming_set_queue2(player->streamer,
+ queue,
+ TRUE,
+ player->ini.http_max_size_bytes,
+ player->ini.http_buffering_time,
+ 1.0,
+ player->ini.http_buffering_limit,
+ use_file_buffer,
+ file_buffering_path,
+ (guint64)dur_bytes);
+ }
+
+ MMPLAYER_FREEIF(file_buffering_path);
+ }
+ }
+ }
+ }
+ /* if it is not decoder or */
+ /* in decoder case any of the video/audio still need to link*/
+ if(!g_strrstr(klass, "Decoder"))
+ {
+
+ pad = gst_element_get_static_pad(sinkelement, padname);
+ if ( ! pad )
+ {
+ debug_warning("failed to get pad(%s) from %s. retrying with [sink]\n",
+ padname, GST_ELEMENT_NAME(sinkelement) );
+
+ pad = gst_element_get_static_pad(sinkelement, "sink");
+
+ if ( ! pad )
+ {
+ debug_error("failed to get pad(sink) from %s. \n",
+ GST_ELEMENT_NAME(sinkelement) );
+ goto ERROR;
+ }
+ }
+
+ if ( GST_PAD_LINK_OK != gst_pad_link(srcpad, pad) )
+ {
+ gst_object_unref(GST_OBJECT(pad));
+ debug_error("failed to link (%s) to pad(%s)\n", GST_ELEMENT_NAME( sinkelement ), padname );
+ goto ERROR;
+ }
+
+ gst_object_unref(GST_OBJECT(pad));
+ }
+
+ for(;templlist != NULL; templlist = templlist->next)
+ {
+ padtemplate = templlist->data;
+
+ debug_log ("director = [%d], presence = [%d]\n", padtemplate->direction, padtemplate->presence);
+
+ if( padtemplate->direction != GST_PAD_SRC ||
+ padtemplate->presence == GST_PAD_REQUEST )
+ continue;
+
+ switch(padtemplate->presence)
+ {
+ case GST_PAD_ALWAYS:
+ {
+ GstPad *srcpad = gst_element_get_static_pad(sinkelement, "src");
+ GstCaps *caps = gst_pad_get_caps(srcpad);
+
+ /* Check whether caps has many types */
+ if ( gst_caps_get_size (caps) > 1 && g_strrstr(klass, "Parser"))
+ {
+ debug_log ("always pad but, caps has many types");
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
+ has_many_types = TRUE;
+ break;
+ }
+
+ if ( ! __mmplayer_try_to_plug(player, srcpad, caps) )
+ {
+ gst_object_unref(GST_OBJECT(srcpad));
+ gst_caps_unref(GST_CAPS(caps));
+
+ debug_error("failed to plug something after %s\n", GST_ELEMENT_NAME( sinkelement ));
+ goto ERROR;
+ }
+
+ gst_caps_unref(GST_CAPS(caps));
+ gst_object_unref(GST_OBJECT(srcpad));
+
+ }
+ break;
+
+
+ case GST_PAD_SOMETIMES:
+ has_dynamic_pads = TRUE;
+ break;
+
+ default:
+ break;
+ }
+ }
+
+ /* check if player can do start continually */
+ MMPLAYER_CHECK_CMD_IF_EXIT(player);
+
+ if( has_dynamic_pads )
+ {
+ player->have_dynamic_pad = TRUE;
+ MMPLAYER_SIGNAL_CONNECT ( player, sinkelement, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added",
+ G_CALLBACK(__mmplayer_add_new_pad), player);
+
+ /* for streaming, more then one typefind will used for each elementary stream
+ * so this doesn't mean the whole pipeline completion
+ */
+ if ( ! MMPLAYER_IS_RTSP_STREAMING( player ) )
+ {
+ MMPLAYER_SIGNAL_CONNECT( player, sinkelement, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "no-more-pads",
+ G_CALLBACK(__mmplayer_pipeline_complete), player);
+ }
+ }
+
+ if (has_many_types)
+ {
+ GstPad *pad = NULL;
+
+ player->has_many_types = has_many_types;
+
+ pad = gst_element_get_static_pad(sinkelement, "src");
+ MMPLAYER_SIGNAL_CONNECT (player, pad, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "notify::caps", G_CALLBACK(__mmplayer_add_new_caps), player);
+ gst_object_unref (GST_OBJECT(pad));
+ }
+
+
+ /* check if player can do start continually */
+ MMPLAYER_CHECK_CMD_IF_EXIT(player);
+
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(sinkelement, GST_STATE_PAUSED) )
+ {
+ debug_error("failed to set state PAUSED to %s\n", GST_ELEMENT_NAME( sinkelement ));
+ goto ERROR;
+ }
+
+ if ( queue )
+ {
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state (queue, GST_STATE_PAUSED) )
+ {
+ debug_error("failed to set state PAUSED to queue\n");
+ goto ERROR;
+ }
+
+ queue = NULL;
+
+ gst_object_unref (GST_OBJECT(qsrcpad));
+ qsrcpad = NULL;
+ }
+
+ if ( parser )
+ {
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state (parser, GST_STATE_PAUSED) )
+ {
+ debug_error("failed to set state PAUSED to queue\n");
+ goto ERROR;
+ }
+
+ parser = NULL;
+
+ gst_object_unref (GST_OBJECT(pssrcpad));
+ pssrcpad = NULL;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
+
+ERROR:
+
+ if ( queue )
+ {
+ gst_object_unref(GST_OBJECT(qsrcpad));
+
+ /* NOTE : Trying to dispose element queue0, but it is in READY instead of the NULL state.
+ * You need to explicitly set elements to the NULL state before
+ * dropping the final reference, to allow them to clean up.
+ */
+ gst_element_set_state(queue, GST_STATE_NULL);
+ /* And, it still has a parent "player".
+ * You need to let the parent manage the object instead of unreffing the object directly.
+ */
+
+ gst_bin_remove (GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), queue);
+ //gst_object_unref( queue );
+ }
+
+ if ( srccaps )
+ gst_caps_unref(GST_CAPS(srccaps));
+
+ return FALSE;
+}
+
+static gboolean __mmplayer_feature_filter(GstPluginFeature *feature, gpointer data) // @
+{
+ const gchar *klass;
+ //const gchar *name;
+
+ /* we only care about element factories */
+ if (!GST_IS_ELEMENT_FACTORY(feature))
+ return FALSE;
+
+ /* only parsers, demuxers and decoders */
+ klass = gst_element_factory_get_klass(GST_ELEMENT_FACTORY(feature));
+ //name = gst_element_factory_get_longname(GST_ELEMENT_FACTORY(feature));
+
+ if( g_strrstr(klass, "Demux") == NULL &&
+ g_strrstr(klass, "Codec/Decoder") == NULL &&
+ g_strrstr(klass, "Depayloader") == NULL &&
+ g_strrstr(klass, "Parse") == NULL)
+ {
+ return FALSE;
+ }
+ return TRUE;
+}
+
+
+static void __mmplayer_add_new_caps(GstPad* pad, GParamSpec* unused, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*) data;
+ GstCaps *caps = NULL;
+ GstStructure *str = NULL;
+ const char *name;
+
+ MMPLAYER_FENTER();
+
+ return_if_fail ( pad )
+ return_if_fail ( unused )
+ return_if_fail ( data )
+
+ caps = gst_pad_get_caps(pad);
+ if ( !caps )
+ return;
+
+ str = gst_caps_get_structure(caps, 0);
+ if ( !str )
+ return;
+
+ name = gst_structure_get_name(str);
+ if ( !name )
+ return;
+ debug_log("name=%s\n", name);
+
+ if ( ! __mmplayer_try_to_plug(player, pad, caps) )
+ {
+ debug_error("failed to autoplug for type (%s)\n", name);
+ gst_caps_unref(caps);
+ return;
+ }
+
+ gst_caps_unref(caps);
+
+ __mmplayer_pipeline_complete( NULL, (gpointer)player );
+
+ MMPLAYER_FLEAVE();
+
+ return;
+}
+
+static void __mmplayer_set_unlinked_mime_type(mm_player_t* player, GstCaps *caps)
+{
+ GstStructure *str;
+ gint version = 0;
+ const char *stream_type;
+ gchar *version_field = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_if_fail ( player );
+ return_if_fail ( caps );
+
+ str = gst_caps_get_structure(caps, 0);
+ if ( !str )
+ return;
+
+ stream_type = gst_structure_get_name(str);
+ if ( !stream_type )
+ return;
+
+
+ /* set unlinked mime type for downloadable codec */
+ if (g_str_has_prefix(stream_type, "video/"))
+ {
+ if (g_str_has_prefix(stream_type, "video/mpeg"))
+ {
+ gst_structure_get_int (str, MM_PLAYER_MPEG_VNAME, &version);
+ version_field = MM_PLAYER_MPEG_VNAME;
+ }
+ else if (g_str_has_prefix(stream_type, "video/x-wmv"))
+ {
+ gst_structure_get_int (str, MM_PLAYER_WMV_VNAME, &version);
+ version_field = MM_PLAYER_WMV_VNAME;
+
+ }
+ else if (g_str_has_prefix(stream_type, "video/x-divx"))
+ {
+ gst_structure_get_int (str, MM_PLAYER_DIVX_VNAME, &version);
+ version_field = MM_PLAYER_DIVX_VNAME;
+ }
+
+ if (version)
+ {
+ player->unlinked_video_mime = g_strdup_printf("%s, %s=%d", stream_type, version_field, version);
+ }
+ else
+ {
+ player->unlinked_video_mime = g_strdup_printf("%s", stream_type);
+ }
+ }
+ else if (g_str_has_prefix(stream_type, "audio/"))
+ {
+ if (g_str_has_prefix(stream_type, "audio/mpeg")) // mp3 or aac
+ {
+ gst_structure_get_int (str, MM_PLAYER_MPEG_VNAME, &version);
+ version_field = MM_PLAYER_MPEG_VNAME;
+ }
+ else if (g_str_has_prefix(stream_type, "audio/x-wma"))
+ {
+ gst_structure_get_int (str, MM_PLAYER_WMA_VNAME, &version);
+ version_field = MM_PLAYER_WMA_VNAME;
+ }
+
+ if (version)
+ {
+ player->unlinked_audio_mime = g_strdup_printf("%s, %s=%d", stream_type, version_field, version);
+ }
+ else
+ {
+ player->unlinked_audio_mime = g_strdup_printf("%s", stream_type);
+ }
+ }
+
+ MMPLAYER_FLEAVE();
+}
+
+static void __mmplayer_add_new_pad(GstElement *element, GstPad *pad, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*) data;
+ GstCaps *caps = NULL;
+ GstStructure *str = NULL;
+ const char *name;
+
+ MMPLAYER_FENTER();
+ return_if_fail ( player );
+ return_if_fail ( pad );
+
+ GST_OBJECT_LOCK (pad);
+ if ((caps = GST_PAD_CAPS(pad)))
+ gst_caps_ref(caps);
+ GST_OBJECT_UNLOCK (pad);
+
+ if ( NULL == caps )
+ {
+ caps = gst_pad_get_caps(pad);
+ if ( !caps ) return;
+ }
+
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
+
+ str = gst_caps_get_structure(caps, 0);
+ if ( !str )
+ return;
+
+ name = gst_structure_get_name(str);
+ if ( !name )
+ return;
+
+ player->num_dynamic_pad++;
+ debug_log("stream count inc : %d\n", player->num_dynamic_pad);
+
+ /* Note : If the stream is the subtitle, we try not to play it. Just close the demuxer subtitle pad.
+ * If want to play it, remove this code.
+ */
+ if (g_strrstr(name, "application"))
+ {
+ if (g_strrstr(name, "x-id3") || g_strrstr(name, "x-apetag"))
+ {
+ /* If id3/ape tag comes, keep going */
+ debug_log("application mime exception : id3/ape tag");
+ }
+ else
+ {
+ /* Otherwise, we assume that this stream is subtile. */
+ debug_log(" application mime type pad is closed.");
+ return;
+ }
+ }
+ else if (g_strrstr(name, "audio"))
+ {
+ gint samplerate = 0, channels = 0;
+
+ if (player->audiodec_linked)
+ {
+ gst_caps_unref(caps);
+ debug_log("multi tracks. skip to plug");
+ return;
+ }
+
+ /* set stream information */
+ /* if possible, set it here because the caps is not distrubed by resampler. */
+ gst_structure_get_int (str, "rate", &samplerate);
+ mm_attrs_set_int_by_name(player->attrs, "content_audio_samplerate", samplerate);
+
+ gst_structure_get_int (str, "channels", &channels);
+ mm_attrs_set_int_by_name(player->attrs, "content_audio_channels", channels);
+
+ debug_log("audio samplerate : %d channels : %d", samplerate, channels);
+ }
+ else if (g_strrstr(name, "video"))
+ {
+ gint stype;
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &stype);
+
+ /* don't make video because of not required */
+ if (stype == MM_DISPLAY_SURFACE_NULL)
+ {
+ debug_log("no video because it's not required");
+ return;
+ }
+
+ player->v_stream_caps = gst_caps_copy(caps); //if needed, video caps is required when videobin is created
+ }
+
+ if ( ! __mmplayer_try_to_plug(player, pad, caps) )
+ {
+ debug_error("failed to autoplug for type (%s)", name);
+
+ __mmplayer_set_unlinked_mime_type(player, caps);
+ }
+
+ gst_caps_unref(caps);
+
+ MMPLAYER_FLEAVE();
+ return;
+}
+
+/* test API for tuning audio gain. this API should be
+ * deprecated before the day of final release
+ */
+int
+_mmplayer_set_volume_tune(MMHandleType hplayer, MMPlayerVolumeType volume)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ gint error = MM_ERROR_NONE;
+ gint vol_max = 0;
+ gboolean isMidi = FALSE;
+ gint i = 0;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED )
+
+ debug_log("clip type=%d(1-midi, 0-others), volume [L]=%d:[R]=%d\n",
+ player->profile.play_mode, volume.level[0], volume.level[1]);
+
+ isMidi = ( player->profile.play_mode == MM_PLAYER_MODE_MIDI ) ? TRUE : FALSE;
+
+ if ( isMidi )
+ vol_max = 1000;
+ else
+ vol_max = 100;
+
+ /* is it proper volume level? */
+ for (i = 0; i < MM_VOLUME_CHANNEL_NUM; ++i)
+ {
+ if (volume.level[i] < 0 || volume.level[i] > vol_max) {
+ debug_log("Invalid Volume level!!!! \n");
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
+ }
+
+ if ( isMidi )
+ {
+ if ( player->pipeline->mainbin )
+ {
+ GstElement *midi_element = player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst;
+
+ if ( midi_element && ( strstr(GST_ELEMENT_NAME(midi_element), "midiparse")) )
+ {
+ debug_log("setting volume (%d) level to midi plugin\n", volume.level[0]);
+
+ g_object_set(midi_element, "volume", volume.level[0], NULL);
+ }
+ }
+ }
+ else
+ {
+ if ( player->pipeline->audiobin )
+ {
+ GstElement *sink_element = player->pipeline->audiobin[MMPLAYER_A_SINK].gst;
+
+ /* Set to Avsysaudiosink element */
+ if ( sink_element )
+ {
+ gint vol_value = 0;
+ gboolean mute = FALSE;
+ vol_value = volume.level[0];
+
+ g_object_set(G_OBJECT(sink_element), "tuningvolume", vol_value, NULL);
+
+ mute = (vol_value == 0)? TRUE:FALSE;
+
+ g_object_set(G_OBJECT(sink_element), "mute", mute, NULL);
+ }
+
+ }
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return error;
+}
+
+gboolean
+__mmplayer_dump_pipeline_state( mm_player_t* player )
+{
+ GstIterator*iter = NULL;
+ gboolean done = FALSE;
+
+ GstElement *item = NULL;
+ GstElementFactory *factory = NULL;
+
+ GstState state = GST_STATE_VOID_PENDING;
+ GstState pending = GST_STATE_VOID_PENDING;
+ GstClockTime time = 200*GST_MSECOND;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player &&
+ player->pipeline &&
+ player->pipeline->mainbin,
+ FALSE );
+
+ iter = gst_bin_iterate_recurse(GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst) );
+
+ if ( iter != NULL )
+ {
+ while (!done) {
+ switch ( gst_iterator_next (iter, (gpointer)&item) )
+ {
+ case GST_ITERATOR_OK:
+ gst_element_get_state(GST_ELEMENT (item),&state, &pending,time);
+
+ factory = gst_element_get_factory (item) ;
+ if (factory)
+ {
+ debug_error("%s:%s : From:%s To:%s refcount : %d\n", GST_OBJECT_NAME(factory) , GST_ELEMENT_NAME(item) ,
+ gst_element_state_get_name(state), gst_element_state_get_name(pending) , GST_OBJECT_REFCOUNT_VALUE(item));
+ }
+ gst_object_unref (item);
+ break;
+ case GST_ITERATOR_RESYNC:
+ gst_iterator_resync (iter);
+ break;
+ case GST_ITERATOR_ERROR:
+ done = TRUE;
+ break;
+ case GST_ITERATOR_DONE:
+ done = TRUE;
+ break;
+ }
+ }
+ }
+
+ item = GST_ELEMENT(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst);
+
+ gst_element_get_state(GST_ELEMENT (item),&state, &pending,time);
+
+ factory = gst_element_get_factory (item) ;
+
+ if (factory)
+ {
+ debug_error("%s:%s : From:%s To:%s refcount : %d\n",
+ GST_OBJECT_NAME(factory),
+ GST_ELEMENT_NAME(item),
+ gst_element_state_get_name(state),
+ gst_element_state_get_name(pending),
+ GST_OBJECT_REFCOUNT_VALUE(item) );
+ }
+
+ if ( iter )
+ gst_iterator_free (iter);
+
+ MMPLAYER_FLEAVE();
+
+ return FALSE;
+}
+
+
+gboolean
+__mmplayer_check_subtitle( mm_player_t* player )
+{
+ MMHandleType attrs = 0;
+ char *subtitle_uri = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, FALSE );
+
+ /* get subtitle attribute */
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ return FALSE;
+
+ mm_attrs_get_string_by_name(attrs, "subtitle_uri", &subtitle_uri);
+ if ( !subtitle_uri || !strlen(subtitle_uri))
+ return FALSE;
+
+ debug_log ("subtite uri is %s[%d]\n", subtitle_uri, strlen(subtitle_uri));
+ player->is_external_subtitle_present = TRUE;
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
+}
+
+static gboolean
+__mmplayer_can_extract_pcm( mm_player_t* player )
+{
+ MMHandleType attrs = 0;
+ gboolean is_drm = FALSE;
+ gboolean sound_extraction = FALSE;
+
+ return_val_if_fail ( player, FALSE );
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("fail to get attributes.");
+ return FALSE;
+ }
+
+ /* check file is drm or not */
+ if (g_object_class_find_property(G_OBJECT_GET_CLASS(player->pipeline->mainbin[MMPLAYER_M_SRC].gst), "is-drm"))
+ g_object_get(G_OBJECT(player->pipeline->mainbin[MMPLAYER_M_SRC].gst), "is-drm", &is_drm, NULL);
+
+ /* get sound_extraction property */
+ mm_attrs_get_int_by_name(attrs, "pcm_extraction", &sound_extraction);
+
+ if ( ! sound_extraction || is_drm )
+ {
+ debug_log("checking pcm extraction mode : %d, drm : %d", sound_extraction, is_drm);
+ return FALSE;
+ }
+
+ return TRUE;
+}
+
+static gboolean
+__mmplayer_handle_gst_error ( mm_player_t* player, GstMessage * message, GError* error )
+{
+ MMMessageParamType msg_param;
+ gchar *msg_src_element;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, FALSE );
+ return_val_if_fail( error, FALSE );
+
+ /* NOTE : do somthing necessary inside of __gst_handle_XXX_error. not here */
+
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
+
+ if ( error->domain == GST_CORE_ERROR )
+ {
+ msg_param.code = __gst_handle_core_error( player, error->code );
+ }
+ else if ( error->domain == GST_LIBRARY_ERROR )
+ {
+ msg_param.code = __gst_handle_library_error( player, error->code );
+ }
+ else if ( error->domain == GST_RESOURCE_ERROR )
+ {
+ msg_param.code = __gst_handle_resource_error( player, error->code );
+ }
+ else if ( error->domain == GST_STREAM_ERROR )
+ {
+ msg_param.code = __gst_handle_stream_error( player, error, message );
+ }
+ else
+ {
+ debug_warning("This error domain is not defined.\n");
+
+ /* we treat system error as an internal error */
+ msg_param.code = MM_ERROR_PLAYER_INVALID_STREAM;
+ }
+
+ if ( message->src )
+ {
+ msg_src_element = GST_ELEMENT_NAME( GST_ELEMENT_CAST( message->src ) );
+
+ msg_param.data = (void *) error->message;
+
+ debug_error("-Msg src : [%s] Domain : [%s] Error : [%s] Code : [%d] is tranlated to error code : [0x%x]\n",
+ msg_src_element, g_quark_to_string (error->domain), error->message, error->code, msg_param.code);
+ }
+
+ /* no error */
+ if (msg_param.code == MM_ERROR_NONE)
+ return TRUE;
+
+ /* post error to application */
+ if ( ! player->msg_posted )
+ {
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+ /* don't post more if one was sent already */
+ player->msg_posted = TRUE;
+ }
+ else
+ {
+ debug_log("skip error post because it's sent already.\n");
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
+}
+
+static gboolean
+__mmplayer_handle_streaming_error ( mm_player_t* player, GstMessage * message )
+{
+ debug_log("\n");
+ MMMessageParamType msg_param;
+ gchar *msg_src_element = NULL;
+ GstStructure *s = NULL;
+ guint error_id = 0;
+ gchar *error_string = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( message, FALSE );
+
+ s = malloc( sizeof(GstStructure) );
+ if ( s == NULL)
+ {
+ debug_error ("malloc fail");
+ return FALSE;
+ }
+ memcpy ( s, gst_message_get_structure ( message ), sizeof(GstStructure));
+
+ if ( !gst_structure_get_uint (s, "error_id", &error_id) )
+ error_id = MMPLAYER_STREAMING_ERROR_NONE;
+
+ switch ( error_id )
+ {
+ case MMPLAYER_STREAMING_ERROR_UNSUPPORTED_AUDIO:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_UNSUPPORTED_AUDIO;
+ break;
+ case MMPLAYER_STREAMING_ERROR_UNSUPPORTED_VIDEO:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_UNSUPPORTED_VIDEO;
+ break;
+ case MMPLAYER_STREAMING_ERROR_CONNECTION_FAIL:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_CONNECTION_FAIL;
+ break;
+ case MMPLAYER_STREAMING_ERROR_DNS_FAIL:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_DNS_FAIL;
+ break;
+ case MMPLAYER_STREAMING_ERROR_SERVER_DISCONNECTED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_SERVER_DISCONNECTED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_BAD_SERVER:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_BAD_SERVER;
+ break;
+ case MMPLAYER_STREAMING_ERROR_INVALID_PROTOCOL:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_INVALID_PROTOCOL;
+ break;
+ case MMPLAYER_STREAMING_ERROR_INVALID_URL:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_INVALID_URL;
+ break;
+ case MMPLAYER_STREAMING_ERROR_UNEXPECTED_MSG:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_UNEXPECTED_MSG;
+ break;
+ case MMPLAYER_STREAMING_ERROR_OUT_OF_MEMORIES:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_OUT_OF_MEMORIES;
+ break;
case MMPLAYER_STREAMING_ERROR_RTSP_TIMEOUT:
msg_param.code = MM_ERROR_PLAYER_STREAMING_RTSP_TIMEOUT;
break;
msg_param.code = MM_ERROR_PLAYER_STREAMING_OPTION_NOT_SUPPORTED;
break;
default:
- return MM_ERROR_PLAYER_STREAMING_FAIL;
+ {
+ MMPLAYER_FREEIF(s);
+ return MM_ERROR_PLAYER_STREAMING_FAIL;
+ }
+ }
+
+ error_string = g_strdup(gst_structure_get_string (s, "error_string"));
+ if ( error_string )
+ msg_param.data = (void *) error_string;
+
+ if ( message->src )
+ {
+ msg_src_element = GST_ELEMENT_NAME( GST_ELEMENT_CAST( message->src ) );
+
+ debug_error("-Msg src : [%s] Code : [%x] Error : [%s] \n",
+ msg_src_element, msg_param.code, (char*)msg_param.data );
+ }
+
+ /* post error to application */
+ if ( ! player->msg_posted )
+ {
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+
+ /* don't post more if one was sent already */
+ player->msg_posted = TRUE;
+ }
+ else
+ {
+ debug_log("skip error post because it's sent already.\n");
+ }
+
+ MMPLAYER_FREEIF(s);
+ MMPLAYER_FLEAVE();
+ g_free(error_string);
+
+ return TRUE;
+
+}
+
+static gint
+__gst_handle_core_error( mm_player_t* player, int code )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ switch ( code )
+ {
+ case GST_CORE_ERROR_MISSING_PLUGIN:
+ return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+ case GST_CORE_ERROR_STATE_CHANGE:
+ case GST_CORE_ERROR_SEEK:
+ case GST_CORE_ERROR_NOT_IMPLEMENTED:
+ case GST_CORE_ERROR_FAILED:
+ case GST_CORE_ERROR_TOO_LAZY:
+ case GST_CORE_ERROR_PAD:
+ case GST_CORE_ERROR_THREAD:
+ case GST_CORE_ERROR_NEGOTIATION:
+ case GST_CORE_ERROR_EVENT:
+ case GST_CORE_ERROR_CAPS:
+ case GST_CORE_ERROR_TAG:
+ case GST_CORE_ERROR_CLOCK:
+ case GST_CORE_ERROR_DISABLED:
+ default:
+ trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
+ break;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return trans_err;
+}
+
+static gint
+__gst_handle_library_error( mm_player_t* player, int code )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ switch ( code )
+ {
+ case GST_LIBRARY_ERROR_FAILED:
+ case GST_LIBRARY_ERROR_TOO_LAZY:
+ case GST_LIBRARY_ERROR_INIT:
+ case GST_LIBRARY_ERROR_SHUTDOWN:
+ case GST_LIBRARY_ERROR_SETTINGS:
+ case GST_LIBRARY_ERROR_ENCODE:
+ default:
+ trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
+ break;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return trans_err;
+}
+
+
+static gint
+__gst_handle_resource_error( mm_player_t* player, int code )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ switch ( code )
+ {
+ case GST_RESOURCE_ERROR_NO_SPACE_LEFT:
+ trans_err = MM_ERROR_PLAYER_NO_FREE_SPACE;
+ break;
+ case GST_RESOURCE_ERROR_NOT_FOUND:
+ case GST_RESOURCE_ERROR_OPEN_READ:
+ if ( MMPLAYER_IS_HTTP_STREAMING(player) || MMPLAYER_IS_HTTP_LIVE_STREAMING ( player )
+ || MMPLAYER_IS_RTSP_STREAMING(player))
+ {
+ trans_err = MM_ERROR_PLAYER_STREAMING_CONNECTION_FAIL;
+ break;
+ }
+ case GST_RESOURCE_ERROR_READ:
+ if ( MMPLAYER_IS_HTTP_STREAMING(player) || MMPLAYER_IS_HTTP_LIVE_STREAMING ( player )
+ || MMPLAYER_IS_RTSP_STREAMING(player))
+ {
+ trans_err = MM_ERROR_PLAYER_STREAMING_FAIL;
+ break;
+ }
+ case GST_RESOURCE_ERROR_WRITE:
+ case GST_RESOURCE_ERROR_FAILED:
+ case GST_RESOURCE_ERROR_SEEK:
+ case GST_RESOURCE_ERROR_TOO_LAZY:
+ case GST_RESOURCE_ERROR_BUSY:
+ case GST_RESOURCE_ERROR_OPEN_WRITE:
+ case GST_RESOURCE_ERROR_OPEN_READ_WRITE:
+ case GST_RESOURCE_ERROR_CLOSE:
+ case GST_RESOURCE_ERROR_SYNC:
+ case GST_RESOURCE_ERROR_SETTINGS:
+ default:
+ trans_err = MM_ERROR_PLAYER_INTERNAL;
+ break;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return trans_err;
+}
+
+
+static gint
+__gst_handle_stream_error( mm_player_t* player, GError* error, GstMessage * message )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( error, MM_ERROR_INVALID_ARGUMENT );
+ return_val_if_fail ( message, MM_ERROR_INVALID_ARGUMENT );
+
+ switch ( error->code )
+ {
+ case GST_STREAM_ERROR_FAILED:
+ case GST_STREAM_ERROR_TYPE_NOT_FOUND:
+ case GST_STREAM_ERROR_DECODE:
+ case GST_STREAM_ERROR_WRONG_TYPE:
+ case GST_STREAM_ERROR_DECRYPT:
+ case GST_STREAM_ERROR_DECRYPT_NOKEY:
+ case GST_STREAM_ERROR_CODEC_NOT_FOUND:
+ trans_err = __gst_transform_gsterror( player, message, error );
+ break;
+
+ case GST_STREAM_ERROR_NOT_IMPLEMENTED:
+ case GST_STREAM_ERROR_TOO_LAZY:
+ case GST_STREAM_ERROR_ENCODE:
+ case GST_STREAM_ERROR_DEMUX:
+ case GST_STREAM_ERROR_MUX:
+ case GST_STREAM_ERROR_FORMAT:
+ default:
+ trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
+ break;
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return trans_err;
+}
+
+/* NOTE : decide gstreamer state whether there is some playable track or not. */
+static gint
+__gst_transform_gsterror( mm_player_t* player, GstMessage * message, GError* error )
+{
+ gchar *src_element_name = NULL;
+ GstElement *src_element = NULL;
+ GstElementFactory *factory = NULL;
+ const gchar* klass = NULL;
+
+ MMPLAYER_FENTER();
+
+ /* FIXIT */
+ return_val_if_fail ( message, MM_ERROR_INVALID_ARGUMENT );
+ return_val_if_fail ( message->src, MM_ERROR_INVALID_ARGUMENT );
+ return_val_if_fail ( error, MM_ERROR_INVALID_ARGUMENT );
+
+ src_element = GST_ELEMENT_CAST(message->src);
+ if ( !src_element )
+ goto INTERNAL_ERROR;
+
+ src_element_name = GST_ELEMENT_NAME(src_element);
+ if ( !src_element_name )
+ goto INTERNAL_ERROR;
+
+ factory = gst_element_get_factory(src_element);
+ if ( !factory )
+ goto INTERNAL_ERROR;
+
+ klass = gst_element_factory_get_klass(factory);
+ if ( !klass )
+ goto INTERNAL_ERROR;
+
+ debug_log("error code=%d, msg=%s, src element=%s, class=%s\n",
+ error->code, error->message, src_element_name, klass);
+
+ //<-
+ {
+ int msg_src_pos = 0;
+ gint active_pad_index = player->selector[MM_PLAYER_TRACK_TYPE_AUDIO].active_pad_index;
+ debug_log ("current active pad index -%d", active_pad_index);
+
+ if (src_element_name) {
+ int idx = 0;
+
+ if (player->audio_decoders) {
+ GList *adec = player->audio_decoders;
+ for ( ;adec ; adec = g_list_next(adec)) {
+ gchar *name = adec->data;
+
+ debug_log("found audio decoder name = %s", name);
+ if (g_strrstr(name, src_element_name)) {
+ msg_src_pos = idx;
+ break;
+ }
+ idx++;
+ }
+ }
+ debug_log("active pad = %d, error src index = %d", active_pad_index, msg_src_pos);
+ }
+
+ if (active_pad_index != msg_src_pos) {
+ debug_log("skip error because error is posted from no activated track");
+ return MM_ERROR_NONE;
+ }
+ }
+ //-> temp code
+
+ switch ( error->code )
+ {
+ case GST_STREAM_ERROR_DECODE:
+ {
+ /* Demuxer can't parse one track because it's corrupted.
+ * So, the decoder for it is not linked.
+ * But, it has one playable track.
+ */
+ if ( g_strrstr(klass, "Demux") )
+ {
+ if ( player->can_support_codec == FOUND_PLUGIN_VIDEO )
+ {
+ return MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
+ }
+ else if ( player->can_support_codec == FOUND_PLUGIN_AUDIO )
+ {
+ return MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
+ }
+ else
+ {
+ if ( player->pipeline->audiobin ) // PCM
+ {
+ return MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
+ }
+ else
+ {
+ goto CODEC_NOT_FOUND;
+ }
+ }
+ }
+ return MM_ERROR_PLAYER_INVALID_STREAM;
+ }
+ break;
+
+ case GST_STREAM_ERROR_CODEC_NOT_FOUND:
+ case GST_STREAM_ERROR_TYPE_NOT_FOUND:
+ case GST_STREAM_ERROR_WRONG_TYPE:
+ return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+
+ case GST_STREAM_ERROR_FAILED:
+ {
+ /* Decoder Custom Message */
+ if ( strstr(error->message, "ongoing") )
+ {
+ if ( strncasecmp(klass, "audio", 5) )
+ {
+ if ( ( player->can_support_codec & FOUND_PLUGIN_VIDEO ) )
+ {
+ debug_log("Video can keep playing.\n");
+ return MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
+ }
+ else
+ {
+ goto CODEC_NOT_FOUND;
+ }
+
+ }
+ else if ( strncasecmp(klass, "video", 5) )
+ {
+ if ( ( player->can_support_codec & FOUND_PLUGIN_AUDIO ) )
+ {
+ debug_log("Audio can keep playing.\n");
+ return MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
+ }
+ else
+ {
+ goto CODEC_NOT_FOUND;
+ }
+ }
+ }
+ return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+ }
+ break;
+
+ case GST_STREAM_ERROR_DECRYPT:
+ case GST_STREAM_ERROR_DECRYPT_NOKEY:
+ {
+ debug_error("decryption error, [%s] failed, reason : [%s]\n", src_element_name, error->message);
+
+ if ( strstr(error->message, "rights expired") )
+ {
+ return MM_ERROR_PLAYER_DRM_EXPIRED;
+ }
+ else if ( strstr(error->message, "no rights") )
+ {
+ return MM_ERROR_PLAYER_DRM_NO_LICENSE;
+ }
+ else if ( strstr(error->message, "has future rights") )
+ {
+ return MM_ERROR_PLAYER_DRM_FUTURE_USE;
+ }
+ else if ( strstr(error->message, "opl violation") )
+ {
+ return MM_ERROR_PLAYER_DRM_OUTPUT_PROTECTION;
+ }
+ return MM_ERROR_PLAYER_DRM_NOT_AUTHORIZED;
+ }
+ break;
+
+ default:
+ break;
}
- error_string = g_strdup(gst_structure_get_string (s, "error_string"));
- if ( error_string )
- msg_param.data = (void *) error_string;
+ MMPLAYER_FLEAVE();
- if ( message->src )
+ return MM_ERROR_PLAYER_INVALID_STREAM;
+
+INTERNAL_ERROR:
+ return MM_ERROR_PLAYER_INTERNAL;
+
+CODEC_NOT_FOUND:
+ debug_log("not found any available codec. Player should be destroyed.\n");
+ return MM_ERROR_PLAYER_CODEC_NOT_FOUND;
+}
+
+static void
+__mmplayer_handle_eos_delay( mm_player_t* player, int delay_in_ms )
+{
+ return_if_fail( player );
+
+
+ /* post now if delay is zero */
+ if ( delay_in_ms == 0 || player->set_mode.pcm_extraction)
{
- msg_src_element = GST_ELEMENT_NAME( GST_ELEMENT_CAST( message->src ) );
+ debug_log("eos delay is zero. posting EOS now\n");
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
- debug_error("-Msg src : [%s] Code : [%x] Error : [%s] \n",
- msg_src_element, msg_param.code, (char*)msg_param.data );
+ if ( player->set_mode.pcm_extraction )
+ __mmplayer_cancel_eos_timer(player);
+
+ return;
+ }
+
+ /* cancel if existing */
+ __mmplayer_cancel_eos_timer( player );
+
+ /* init new timeout */
+ /* NOTE : consider give high priority to this timer */
+ debug_log("posting EOS message after [%d] msec\n", delay_in_ms);
+
+ player->eos_timer = g_timeout_add( delay_in_ms,
+ __mmplayer_eos_timer_cb, player );
+
+ player->context.global_default = g_main_context_default ();
+ debug_log("global default context = %p, eos timer id = %d", player->context.global_default, player->eos_timer);
+
+ /* check timer is valid. if not, send EOS now */
+ if ( player->eos_timer == 0 )
+ {
+ debug_warning("creating timer for delayed EOS has failed. sending EOS now\n");
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
+ }
+}
+
+static void
+__mmplayer_cancel_eos_timer( mm_player_t* player )
+{
+ return_if_fail( player );
+
+ if ( player->eos_timer )
+ {
+ debug_log("cancel eos timer");
+ __mmplayer_remove_g_source_from_context(player->context.global_default, player->eos_timer);
+ player->eos_timer = 0;
+ }
+
+ return;
+}
+
+static gboolean
+__mmplayer_eos_timer_cb(gpointer u_data)
+{
+ mm_player_t* player = NULL;
+ player = (mm_player_t*) u_data;
+
+ return_val_if_fail( player, FALSE );
+
+ if ( player->play_count > 1 )
+ {
+ gint ret_value = 0;
+ ret_value = __gst_set_position( player, MM_PLAYER_POS_FORMAT_TIME, 0, TRUE);
+ if (ret_value == MM_ERROR_NONE)
+ {
+ MMHandleType attrs = 0;
+ attrs = MMPLAYER_GET_ATTRS(player);
+
+ /* we successeded to rewind. update play count and then wait for next EOS */
+ player->play_count--;
+
+ mm_attrs_set_int_by_name(attrs, "profile_play_count", player->play_count);
+ mmf_attrs_commit ( attrs );
+ }
+ else
+ {
+ debug_error("seeking to 0 failed in repeat play");
+ }
+ }
+ else
+ {
+ /* posting eos */
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
+ }
+
+ /* we are returning FALSE as we need only one posting */
+ return FALSE;
+}
+
+static void __mmplayer_set_antishock( mm_player_t* player, gboolean disable_by_force)
+{
+ gint antishock = FALSE;
+ MMHandleType attrs = 0;
+
+ MMPLAYER_FENTER();
+
+ return_if_fail ( player && player->pipeline );
+
+ /* It should be passed for video only clip */
+ if ( ! player->pipeline->audiobin )
+ return;
+
+ if ( ( g_strrstr(player->ini.name_of_audiosink, "avsysaudiosink")) )
+ {
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( ! attrs )
+ {
+ debug_error("fail to get attributes.\n");
+ return;
+ }
+
+ mm_attrs_get_int_by_name(attrs, "sound_fadeup", &antishock);
+
+ if (player->sm.antishock) {
+ antishock = TRUE;
+ player->sm.antishock = 0;
+ }
+
+ debug_log("setting antishock as (%d)\n", antishock);
+
+ if ( disable_by_force )
+ {
+ debug_log("but, antishock is disabled by force when is seeked\n");
+
+ antishock = FALSE;
+ }
+
+ g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "fadeup", antishock, NULL);
+ }
+
+ MMPLAYER_FLEAVE();
+
+ return;
+}
+
+
+static gboolean
+__mmplayer_link_decoder( mm_player_t* player, GstPad *srcpad)
+{
+ const gchar* name = NULL;
+ GstStructure* str = NULL;
+ GstCaps* srccaps = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail( player, FALSE );
+ return_val_if_fail ( srcpad, FALSE );
+
+ /* to check any of the decoder (video/audio) need to be linked to parser*/
+ srccaps = gst_pad_get_caps( srcpad );
+ if ( !srccaps )
+ goto ERROR;
+
+ str = gst_caps_get_structure( srccaps, 0 );
+ if ( ! str )
+ goto ERROR;
+
+ name = gst_structure_get_name(str);
+ if ( ! name )
+ goto ERROR;
+
+ if (strstr(name, "video"))
+ {
+ if(player->videodec_linked)
+ {
+ debug_msg("Video decoder already linked\n");
+ return FALSE;
+ }
+ }
+ if (strstr(name, "audio"))
+ {
+ if(player->audiodec_linked)
+ {
+ debug_msg("Audio decoder already linked\n");
+ return FALSE;
+ }
+ }
+
+ gst_caps_unref( srccaps );
+
+ MMPLAYER_FLEAVE();
+
+ return TRUE;
+
+ERROR:
+ if ( srccaps )
+ gst_caps_unref( srccaps );
+
+ return FALSE;
+}
+
+static gboolean
+__mmplayer_link_sink( mm_player_t* player , GstPad *srcpad)
+{
+ const gchar* name = NULL;
+ GstStructure* str = NULL;
+ GstCaps* srccaps = NULL;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( srcpad, FALSE );
+
+ /* to check any of the decoder (video/audio) need to be linked to parser*/
+ srccaps = gst_pad_get_caps( srcpad );
+ if ( !srccaps )
+ goto ERROR;
+
+ str = gst_caps_get_structure( srccaps, 0 );
+ if ( ! str )
+ goto ERROR;
+
+ name = gst_structure_get_name(str);
+ if ( ! name )
+ goto ERROR;
+
+ if (strstr(name, "video"))
+ {
+ if(player->videosink_linked)
+ {
+ debug_msg("Video Sink already linked\n");
+ return FALSE;
+ }
}
-
- /* post error to application */
- if ( ! player->posted_msg )
+ if (strstr(name, "audio"))
{
- MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
-
- /* don't post more if one was sent already */
- player->posted_msg = TRUE;
+ if(player->audiosink_linked)
+ {
+ debug_msg("Audio Sink already linked\n");
+ return FALSE;
+ }
}
- else
+ if (strstr(name, "text"))
{
- debug_log("skip error post because it's sent already.\n");
+ if(player->textsink_linked)
+ {
+ debug_msg("Text Sink already linked\n");
+ return FALSE;
+ }
}
- debug_fleave();
+ gst_caps_unref( srccaps );
+
+ MMPLAYER_FLEAVE();
return TRUE;
+ //return (!player->videosink_linked || !player->audiosink_linked);
+
+ERROR:
+ if ( srccaps )
+ gst_caps_unref( srccaps );
+ return FALSE;
}
-static gint
-__gst_handle_core_error( mm_player_t* player, int code )
+
+/* sending event to one of sinkelements */
+static gboolean
+__gst_send_event_to_sink( mm_player_t* player, GstEvent* event )
{
- gint trans_err = MM_ERROR_NONE;
+ GstEvent * event2 = NULL;
+ GList *sinks = NULL;
+ gboolean res = FALSE;
+ GstPad *sinkpad = NULL;
+ MMPLAYER_FENTER();
- debug_fenter();
+ return_val_if_fail( player, FALSE );
+ return_val_if_fail ( event, FALSE );
- return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ if ( player->play_subtitle && !player->use_textoverlay)
+ event2 = gst_event_copy((const GstEvent *)event);
- switch ( code )
+ sinks = player->sink_elements;
+ while (sinks)
{
- case GST_CORE_ERROR_STATE_CHANGE:
- case GST_CORE_ERROR_MISSING_PLUGIN:
- case GST_CORE_ERROR_SEEK:
- case GST_CORE_ERROR_NOT_IMPLEMENTED:
- case GST_CORE_ERROR_FAILED:
- case GST_CORE_ERROR_TOO_LAZY:
- case GST_CORE_ERROR_PAD:
- case GST_CORE_ERROR_THREAD:
- case GST_CORE_ERROR_NEGOTIATION:
- case GST_CORE_ERROR_EVENT:
- case GST_CORE_ERROR_CAPS:
- case GST_CORE_ERROR_TAG:
- case GST_CORE_ERROR_CLOCK:
- case GST_CORE_ERROR_DISABLED:
- default:
- trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
- break;
+ GstElement *sink = GST_ELEMENT_CAST (sinks->data);
+
+ if (GST_IS_ELEMENT(sink))
+ {
+ /* keep ref to the event */
+ gst_event_ref (event);
+
+ if ( (res = gst_element_send_event (sink, event)) )
+ {
+ debug_log("sending event[%s] to sink element [%s] success!\n",
+ GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(sink) );
+ break;
+ }
+
+ debug_log("sending event[%s] to sink element [%s] failed. try with next one.\n",
+ GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(sink) );
+ }
+
+ sinks = g_list_next (sinks);
}
- debug_fleave();
+#if 0
+ if (internal_sub)
+ request pad name = sink0;
+ else
+ request pad name = sink1; // external
+#endif
- return trans_err;
+ /* Note : Textbin is not linked to the video or audio bin.
+ * It needs to send the event to the text sink seperatelly.
+ */
+ if ( player->play_subtitle && !player->use_textoverlay)
+ {
+ GstElement *text_sink = NULL;
+ gchar *change_pad_name = NULL;
+
+ debug_log ("external subtitle active idx = %d", player->external_text_idx);
+
+ /* To get the new pad from the selector */
+ change_pad_name = g_strdup_printf ("sink%d", player->external_text_idx);
+ if (change_pad_name == NULL)
+ {
+ debug_warning ("Pad does not exists\n");
+ return FALSE;
+ }
+
+ g_object_get (player->pipeline->mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "active-pad", &sinkpad, NULL);
+ if (!strcmp (GST_PAD_NAME (sinkpad), change_pad_name))
+ {
+ debug_log ("active pad & change pad are same no need to change active pad");
+ }
+ else
+ {
+ sinkpad = gst_element_get_static_pad (player->pipeline->mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, change_pad_name);
+ if (sinkpad == NULL)
+ {
+ debug_error ("failed to get static pad");;
+ return FALSE;
+ }
+
+ debug_log ("Set Active Pad - %s:%s\n", GST_DEBUG_PAD_NAME(sinkpad));
+ g_object_set (player->pipeline->mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, "active-pad", sinkpad, NULL);
+ }
+
+ text_sink = GST_ELEMENT_CAST (player->pipeline->textbin[MMPLAYER_T_FAKE_SINK].gst);
+
+ if (GST_IS_ELEMENT(text_sink))
+ {
+ /* keep ref to the event */
+ gst_event_ref (event2);
+
+ if ( (res != gst_element_send_event (text_sink, event2)) )
+ {
+ debug_error("sending event[%s] to subtitle sink element [%s] failed!\n",
+ GST_EVENT_TYPE_NAME(event2), GST_ELEMENT_NAME(text_sink) );
+ }
+ else
+ {
+ debug_log("sending event[%s] to subtitle sink element [%s] success!\n",
+ GST_EVENT_TYPE_NAME(event2), GST_ELEMENT_NAME(text_sink) );
+ }
+
+ gst_event_unref (event2);
+ }
+ g_free(change_pad_name);
+ }
+
+ gst_event_unref (event);
+
+ MMPLAYER_FLEAVE();
+
+ return res;
}
-static gint
-__gst_handle_library_error( mm_player_t* player, int code )
+static void
+__mmplayer_add_sink( mm_player_t* player, GstElement* sink )
{
- gint trans_err = MM_ERROR_NONE;
-
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_if_fail ( player );
+ return_if_fail ( sink );
- switch ( code )
- {
- case GST_LIBRARY_ERROR_FAILED:
- case GST_LIBRARY_ERROR_TOO_LAZY:
- case GST_LIBRARY_ERROR_INIT:
- case GST_LIBRARY_ERROR_SHUTDOWN:
- case GST_LIBRARY_ERROR_SETTINGS:
- case GST_LIBRARY_ERROR_ENCODE:
- default:
- trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
- break;
- }
-
- debug_fleave();
+ player->sink_elements =
+ g_list_append(player->sink_elements, sink);
- return trans_err;
+ MMPLAYER_FLEAVE();
}
+static void
+__mmplayer_del_sink( mm_player_t* player, GstElement* sink )
+{
+ MMPLAYER_FENTER();
-static gint
-__gst_handle_resource_error( mm_player_t* player, int code )
+ return_if_fail ( player );
+ return_if_fail ( sink );
+
+ player->sink_elements =
+ g_list_remove(player->sink_elements, sink);
+
+ MMPLAYER_FLEAVE();
+}
+
+static gboolean
+__gst_seek(mm_player_t* player, GstElement * element, gdouble rate,
+ GstFormat format, GstSeekFlags flags, GstSeekType cur_type,
+ gint64 cur, GstSeekType stop_type, gint64 stop )
{
- gint trans_err = MM_ERROR_NONE;
+ GstEvent* event = NULL;
+ gboolean result = FALSE;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( player, FALSE );
- switch ( code )
- {
- case GST_RESOURCE_ERROR_NO_SPACE_LEFT:
- trans_err = MM_ERROR_PLAYER_NO_FREE_SPACE;
- break;
- case GST_RESOURCE_ERROR_NOT_FOUND:
- case GST_RESOURCE_ERROR_OPEN_READ:
- if ( MMPLAYER_IS_HTTP_STREAMING(player) || MMPLAYER_IS_HTTP_LIVE_STREAMING ( player ) )
- {
- trans_err = MM_ERROR_PLAYER_STREAMING_CONNECTION_FAIL;
- break;
- }
- case GST_RESOURCE_ERROR_READ:
- if ( MMPLAYER_IS_HTTP_STREAMING(player) || MMPLAYER_IS_HTTP_LIVE_STREAMING ( player ))
- {
- trans_err = MM_ERROR_PLAYER_STREAMING_FAIL;
- break;
- }
- case GST_RESOURCE_ERROR_SEEK:
- case GST_RESOURCE_ERROR_FAILED:
- case GST_RESOURCE_ERROR_TOO_LAZY:
- case GST_RESOURCE_ERROR_BUSY:
- case GST_RESOURCE_ERROR_OPEN_WRITE:
- case GST_RESOURCE_ERROR_OPEN_READ_WRITE:
- case GST_RESOURCE_ERROR_CLOSE:
- case GST_RESOURCE_ERROR_WRITE:
- case GST_RESOURCE_ERROR_SYNC:
- case GST_RESOURCE_ERROR_SETTINGS:
- default:
- trans_err = MM_ERROR_PLAYER_FILE_NOT_FOUND;
- break;
- }
+ __mmplayer_drop_subtitle(player, FALSE);
- debug_fleave();
+ event = gst_event_new_seek (rate, format, flags, cur_type,
+ cur, stop_type, stop);
- return trans_err;
-}
+ result = __gst_send_event_to_sink( player, event );
+ MMPLAYER_FLEAVE();
-static gint
-__gst_handle_stream_error( mm_player_t* player, GError* error, GstMessage * message )
+ return result;
+}
+
+/* NOTE : be careful with calling this api. please refer to below glib comment
+ * glib comment : Note that there is a bug in GObject that makes this function much
+ * less useful than it might seem otherwise. Once gobject is disposed, the callback
+ * will no longer be called, but, the signal handler is not currently disconnected.
+ * If the instance is itself being freed at the same time than this doesn't matter,
+ * since the signal will automatically be removed, but if instance persists,
+ * then the signal handler will leak. You should not remove the signal yourself
+ * because in a future versions of GObject, the handler will automatically be
+ * disconnected.
+ *
+ * It's possible to work around this problem in a way that will continue to work
+ * with future versions of GObject by checking that the signal handler is still
+ * connected before disconnected it:
+ *
+ * if (g_signal_handler_is_connected (instance, id))
+ * g_signal_handler_disconnect (instance, id);
+ */
+static void
+__mmplayer_release_signal_connection(mm_player_t* player, MMPlayerSignalType type)
{
- gint trans_err = MM_ERROR_NONE;
+ GList* sig_list = NULL;
+ MMPlayerSignalItem* item = NULL;
- debug_fenter();
+ MMPLAYER_FENTER();
- return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail( error, MM_ERROR_INVALID_ARGUMENT );
- return_val_if_fail ( message, MM_ERROR_INVALID_ARGUMENT );
+ return_if_fail( player );
- switch ( error->code )
- {
- case GST_STREAM_ERROR_FAILED:
- case GST_STREAM_ERROR_TYPE_NOT_FOUND:
- case GST_STREAM_ERROR_DECODE:
- case GST_STREAM_ERROR_WRONG_TYPE:
- case GST_STREAM_ERROR_DECRYPT:
- trans_err = __gst_transform_gsterror( player, message, error );
- break;
+ debug_log("release signals type : %d", type);
- case GST_STREAM_ERROR_CODEC_NOT_FOUND:
- case GST_STREAM_ERROR_NOT_IMPLEMENTED:
- case GST_STREAM_ERROR_TOO_LAZY:
- case GST_STREAM_ERROR_ENCODE:
- case GST_STREAM_ERROR_DEMUX:
- case GST_STREAM_ERROR_MUX:
- case GST_STREAM_ERROR_FORMAT:
- case GST_STREAM_ERROR_DECRYPT_NOKEY:
- default:
- trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
- break;
+ if ((type < MM_PLAYER_SIGNAL_TYPE_AUTOPLUG) || (type >= MM_PLAYER_SIGNAL_TYPE_ALL))
+ {
+ __mmplayer_release_signal_connection (player, MM_PLAYER_SIGNAL_TYPE_AUTOPLUG);
+ __mmplayer_release_signal_connection (player, MM_PLAYER_SIGNAL_TYPE_VIDEOBIN);
+ __mmplayer_release_signal_connection (player, MM_PLAYER_SIGNAL_TYPE_AUDIOBIN);
+ __mmplayer_release_signal_connection (player, MM_PLAYER_SIGNAL_TYPE_TEXTBIN);
+ __mmplayer_release_signal_connection (player, MM_PLAYER_SIGNAL_TYPE_OTHERS);
+ return;
}
- debug_fleave();
+ sig_list = player->signals[type];
- return trans_err;
-}
+ for ( ; sig_list; sig_list = sig_list->next )
+ {
+ item = sig_list->data;
+ if ( item && item->obj && GST_IS_ELEMENT(item->obj) )
+ {
+ if ( g_signal_handler_is_connected ( item->obj, item->sig ) )
+ {
+ g_signal_handler_disconnect ( item->obj, item->sig );
+ }
+ }
-/* NOTE : decide gstreamer state whether there is some playable track or not. */
-static gint
-__gst_transform_gsterror( mm_player_t* player, GstMessage * message, GError* error )
-{
- gchar *src_element_name = NULL;
- GstElement *src_element = NULL;
- GstElementFactory *factory = NULL;
- const gchar* klass = NULL;
-
- debug_fenter();
+ MMPLAYER_FREEIF( item );
+ }
- /* FIXIT */
- return_val_if_fail ( message, MM_ERROR_INVALID_ARGUMENT );
- return_val_if_fail ( message->src, MM_ERROR_INVALID_ARGUMENT );
- return_val_if_fail ( error, MM_ERROR_INVALID_ARGUMENT );
+ g_list_free ( player->signals[type] );
+ player->signals[type] = NULL;
+
+ MMPLAYER_FLEAVE();
+
+ return;
+}
+
+int _mmplayer_change_videosink(MMHandleType handle, MMDisplaySurfaceType surface_type, void *display_overlay)
+{
+ mm_player_t* player = 0;
+ int prev_display_surface_type = 0;
+ void *prev_display_overlay = NULL;
+ const gchar *klass = NULL;
+ gchar *cur_videosink_name = NULL;
+ int ret = 0;
+ int i = 0;
+ int num_of_dec = 2; /* DEC1, DEC2 */
- src_element = GST_ELEMENT_CAST(message->src);
- if ( !src_element )
- goto INTERNAL_ERROR;
-
- src_element_name = GST_ELEMENT_NAME(src_element);
- if ( !src_element_name )
- goto INTERNAL_ERROR;
+ MMPLAYER_FENTER();
- factory = gst_element_get_factory(src_element);
- if ( !factory )
- goto INTERNAL_ERROR;
-
- klass = gst_element_factory_get_klass(factory);
- if ( !klass )
- goto INTERNAL_ERROR;
+ return_val_if_fail(handle, MM_ERROR_COMMON_INVALID_ARGUMENT);
+ return_val_if_fail(display_overlay, MM_ERROR_COMMON_INVALID_ARGUMENT);
- debug_log("error code=%d, msg=%s, src element=%s, class=%s\n",
- error->code, error->message, src_element_name, klass);
+ player = MM_PLAYER_CAST(handle);
+ if (surface_type < MM_DISPLAY_SURFACE_X && surface_type > MM_DISPLAY_SURFACE_EVAS)
+ {
+ debug_error("Not support this surface type(%d) for changing vidoesink", surface_type);
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
- switch ( error->code )
+ /* load previous attributes */
+ if (player->attrs)
{
- case GST_STREAM_ERROR_DECODE:
+ mm_attrs_get_int_by_name (player->attrs, "display_surface_type", &prev_display_surface_type);
+ mm_attrs_get_data_by_name (player->attrs, "display_overlay", &prev_display_overlay);
+ debug_log("[0: X surface, 1: EVAS surface] previous surface type(%d), new surface type(%d)", prev_display_surface_type, surface_type);
+ if (prev_display_surface_type == surface_type)
{
- /* NOTE : Delay is needed because gst callback is sometime sent
- * before completing autoplugging.
- * Timer is more better than usleep.
- * But, transformed msg value should be stored in player handle
- * for function to call by timer.
- */
- if ( PLAYER_INI()->async_start )
- usleep(500000);
-
- /* Demuxer can't parse one track because it's corrupted.
- * So, the decoder for it is not linked.
- * But, it has one playable track.
- */
- if ( g_strrstr(klass, "Demux") )
- {
- if ( player->can_support_codec == FOUND_PLUGIN_VIDEO )
- {
- return MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
- }
- else if ( player->can_support_codec == FOUND_PLUGIN_AUDIO )
- {
- return MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
- }
- else
- {
- if ( player->pipeline->audiobin ) // PCM
- {
- return MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
- }
- else
- {
- goto CODEC_NOT_FOUND;
- }
- }
- }
- return MM_ERROR_PLAYER_INVALID_STREAM;
+ debug_log("incoming display surface type is same as previous one, do nothing..");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
}
- break;
+ }
+ else
+ {
+ debug_error("failed to load attributes");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- case GST_STREAM_ERROR_WRONG_TYPE:
+ /* check videosink element is created */
+ if (!player->pipeline || !player->pipeline->videobin ||
+ !player->pipeline->videobin[MMPLAYER_V_SINK].gst )
+ {
+ debug_log("videosink element is not yet ready");
+
+ /* videobin is not created yet, so we just set attributes related to display surface */
+ debug_log("store display attribute for given surface type(%d)", surface_type);
+ mm_attrs_set_int_by_name (player->attrs, "display_surface_type", surface_type);
+ mm_attrs_set_data_by_name (player->attrs, "display_overlay", display_overlay, sizeof(display_overlay));
+ if ( mmf_attrs_commit ( player->attrs ) )
{
- return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+ debug_error("failed to commit attribute");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+ }
+ else
+ {
+ /* get player command status */
+ if ( !(player->cmd == MMPLAYER_COMMAND_START || player->cmd == MMPLAYER_COMMAND_RESUME || player->cmd == MMPLAYER_COMMAND_PAUSE) )
+ {
+ debug_error("invalid player command status(%d), __mmplayer_do_change_videosink() is only available with START/RESUME/PAUSE command",player->cmd);
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INVALID_STATE;
}
- break;
- case GST_STREAM_ERROR_FAILED:
+ /* get a current videosink name */
+ cur_videosink_name = GST_ELEMENT_NAME(player->pipeline->videobin[MMPLAYER_V_SINK].gst);
+
+ /* surface change */
+ for ( i = 0 ; i < num_of_dec ; i++)
{
- /* Decoder Custom Message */
- if ( strstr(error->message, "ongoing") )
+ if ( player->pipeline->mainbin &&
+ player->pipeline->mainbin[MMPLAYER_M_DEC1+i].gst )
{
- if ( strcasestr(klass, "audio") )
+ klass = gst_element_factory_get_klass( gst_element_get_factory(player->pipeline->mainbin[MMPLAYER_M_DEC1+i].gst) );
+ if ((g_strrstr(klass, "Codec/Decoder/Video")))
{
- if ( ( player->can_support_codec & FOUND_PLUGIN_VIDEO ) )
- {
- debug_log("Video can keep playing.\n");
- return MM_ERROR_PLAYER_AUDIO_CODEC_NOT_FOUND;
- }
- else
+ if ( !strncmp(cur_videosink_name, "x", 1) && (surface_type == MM_DISPLAY_SURFACE_EVAS) )
{
- goto CODEC_NOT_FOUND;
+ ret = __mmplayer_do_change_videosink(player, MMPLAYER_M_DEC1+i, player->ini.videosink_element_evas, surface_type, display_overlay);
+ if (ret)
+ {
+ goto ERROR_CASE;
+ }
+ else
+ {
+ debug_warning("success to changing display surface(%d)",surface_type);
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+ }
}
-
- }
- else if ( strcasestr(klass, "video") )
- {
- if ( ( player->can_support_codec & FOUND_PLUGIN_AUDIO ) )
+ else if (!strncmp(cur_videosink_name, "evas", 4) && (surface_type == MM_DISPLAY_SURFACE_X) )
{
- debug_log("Audio can keep playing.\n");
- return MM_ERROR_PLAYER_VIDEO_CODEC_NOT_FOUND;
+ ret = __mmplayer_do_change_videosink(player, MMPLAYER_M_DEC1+i, player->ini.videosink_element_x, surface_type, display_overlay);
+ if (ret)
+ {
+ goto ERROR_CASE;
+ }
+ else
+ {
+ debug_warning("success to changing display surface(%d)",surface_type);
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+ }
}
else
{
- goto CODEC_NOT_FOUND;
+ debug_error("invalid incoming surface type(%d) and current videosink_name(%s) for changing display surface",surface_type, cur_videosink_name);
+ ret = MM_ERROR_PLAYER_INTERNAL;
+ goto ERROR_CASE;
}
}
}
- return MM_ERROR_PLAYER_INVALID_STREAM;
- }
- break;
-
- case GST_STREAM_ERROR_TYPE_NOT_FOUND:
- {
- goto CODEC_NOT_FOUND;
- }
- break;
-
- case GST_STREAM_ERROR_DECRYPT:
- {
- debug_log("%s failed reason : %s\n", src_element_name, error->message);
- return MM_MESSAGE_DRM_NOT_AUTHORIZED;
}
- break;
-
- default:
- break;
}
- debug_fleave();
-
- return MM_ERROR_PLAYER_INVALID_STREAM;
-
-INTERNAL_ERROR:
- return MM_ERROR_PLAYER_INTERNAL;
-
-CODEC_NOT_FOUND:
- debug_log("not found any available codec. Player should be destroyed.\n");
- return MM_ERROR_PLAYER_CODEC_NOT_FOUND;
+ERROR_CASE:
+ /* rollback to previous attributes */
+ mm_attrs_set_int_by_name (player->attrs, "display_surface_type", prev_display_surface_type);
+ mm_attrs_set_data_by_name(player->attrs, "display_overlay", prev_display_overlay, sizeof(prev_display_overlay));
+ if ( mmf_attrs_commit ( player->attrs ) )
+ {
+ debug_error("failed to commit attributes to rollback");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ MMPLAYER_FLEAVE();
+ return ret;
}
-static void
-__mmplayer_post_delayed_eos( mm_player_t* player, int delay_in_ms )
+/* NOTE : It does not support some use cases, eg using colorspace converter */
+int
+__mmplayer_do_change_videosink(mm_player_t* player, const int dec_index, const char *videosink_element, MMDisplaySurfaceType surface_type, void *display_overlay)
{
- debug_fenter();
+ GstPad *src_pad_dec = NULL;
+ GstPad *sink_pad_videosink = NULL;
+ GstPad *sink_pad_videobin = NULL;
+ GstClock *clock = NULL;
+ MMPlayerStateType previous_state = MM_PLAYER_STATE_NUM;
+ int ret = MM_ERROR_NONE;
+ gboolean is_audiobin_created = TRUE;
- return_if_fail( player );
+ MMPLAYER_FENTER();
- /* cancel if existing */
- __mmplayer_cancel_delayed_eos( player );
+ return_val_if_fail(player, MM_ERROR_COMMON_INVALID_ARGUMENT);
+ return_val_if_fail(videosink_element, MM_ERROR_COMMON_INVALID_ARGUMENT);
+ return_val_if_fail(display_overlay, MM_ERROR_COMMON_INVALID_ARGUMENT);
+ debug_log("video dec is found(idx:%d), we are going to change videosink to %s", dec_index, videosink_element);
+ debug_log("surface type(%d), display overlay(%x)", surface_type, display_overlay);
- /* post now if delay is zero */
- if ( delay_in_ms == 0 || player->is_sound_extraction)
+ /* get information whether if audiobin is created */
+ if ( !player->pipeline->audiobin ||
+ !player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
{
- debug_log("eos delay is zero. posting EOS now\n");
- MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
-
- if ( player->is_sound_extraction )
- __mmplayer_cancel_delayed_eos(player);
-
- return;
+ debug_warning("audiobin is null, this video content may not have audio data");
+ is_audiobin_created = FALSE;
}
- /* init new timeout */
- /* NOTE : consider give high priority to this timer */
-
- debug_log("posting EOS message after [%d] msec\n", delay_in_ms);
- player->eos_timer = g_timeout_add( delay_in_ms,
- __mmplayer_eos_timer_cb, player );
+ /* get current state of player */
+ previous_state = MMPLAYER_CURRENT_STATE(player);
+ debug_log("previous state(%d)", previous_state);
- /* check timer is valid. if not, send EOS now */
- if ( player->eos_timer == 0 )
+ /* get src pad of decoder and block it */
+ src_pad_dec = gst_element_get_static_pad (GST_ELEMENT(player->pipeline->mainbin[dec_index].gst), "src");
+ if (!src_pad_dec)
{
- debug_warning("creating timer for delayed EOS has failed. sending EOS now\n");
- MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
+ debug_error("failed to get src pad from decode in mainbin");
+ return MM_ERROR_PLAYER_INTERNAL;
}
- debug_fleave();
-}
-
-static void
-__mmplayer_cancel_delayed_eos( mm_player_t* player )
-{
- debug_fenter();
-
- return_if_fail( player );
-
- if ( player->eos_timer )
+ if (!player->doing_seek && previous_state == MM_PLAYER_STATE_PLAYING)
{
- g_source_remove( player->eos_timer );
+ debug_warning("trying to block pad(video)");
+ if (!gst_pad_set_blocked (src_pad_dec, TRUE))
+ {
+ debug_error("failed to set block pad(video)");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ debug_warning("pad is blocked(video)");
+ }
+ else
+ {
+ /* no data flows, so no need to do pad_block */
+ if (player->doing_seek) {
+ debug_warning("not completed seek(%d), do nothing", player->doing_seek);
+ }
+ debug_log("MM_PLAYER_STATE is not PLAYING now, skip pad-block(TRUE)");
}
- player->eos_timer = 0;
-
- debug_fleave();
-
- return;
-}
+ /* remove pad */
+ if (!gst_element_remove_pad(player->pipeline->videobin[MMPLAYER_V_BIN].gst,
+ GST_PAD_CAST(GST_GHOST_PAD(player->ghost_pad_for_videobin))))
+ {
+ debug_error("failed to remove previous ghost_pad for videobin");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
-static gboolean
-__mmplayer_eos_timer_cb(gpointer u_data)
-{
- mm_player_t* player = NULL;
- player = (mm_player_t*) u_data;
+ /* change state of videobin to NULL */
+ debug_log("setting [%s] state to : %d", GST_ELEMENT_NAME(player->pipeline->videobin[MMPLAYER_V_BIN].gst), GST_STATE_NULL);
+ ret = gst_element_set_state(player->pipeline->videobin[MMPLAYER_V_BIN].gst, GST_STATE_NULL);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ {
+ debug_error("failed to change state of videobin to NULL");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- debug_fenter();
+ /* unlink between decoder and videobin and remove previous videosink from videobin */
+ GST_ELEMENT_UNLINK(GST_ELEMENT(player->pipeline->mainbin[dec_index].gst),GST_ELEMENT(player->pipeline->videobin[MMPLAYER_V_BIN].gst));
+ if ( !gst_bin_remove (GST_BIN(player->pipeline->videobin[MMPLAYER_V_BIN].gst), GST_ELEMENT(player->pipeline->videobin[MMPLAYER_V_SINK].gst)) )
+ {
+ debug_error("failed to remove former videosink from videobin");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- return_val_if_fail( player, FALSE );
+ __mmplayer_del_sink( player, player->pipeline->videobin[MMPLAYER_V_SINK].gst );
- /* posting eos */
- MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
+ /* create a new videosink and add it to videobin */
+ player->pipeline->videobin[MMPLAYER_V_SINK].gst = gst_element_factory_make(videosink_element, videosink_element);
+ gst_bin_add (GST_BIN(player->pipeline->videobin[MMPLAYER_V_BIN].gst), GST_ELEMENT(player->pipeline->videobin[MMPLAYER_V_SINK].gst));
+ __mmplayer_add_sink( player, player->pipeline->videobin[MMPLAYER_V_SINK].gst );
+ g_object_set (G_OBJECT (player->pipeline->videobin[MMPLAYER_V_SINK].gst), "qos", TRUE, NULL);
- /* cleare timer id */
- player->eos_timer = 0;
+ /* save attributes */
+ if (player->attrs)
+ {
+ /* set a new display surface type */
+ mm_attrs_set_int_by_name (player->attrs, "display_surface_type", surface_type);
+ /* set a new diplay overlay */
+ switch (surface_type)
+ {
+ case MM_DISPLAY_SURFACE_X:
+ debug_log("save attributes related to display surface to X : xid = %d", *(int*)display_overlay);
+ mm_attrs_set_data_by_name (player->attrs, "display_overlay", display_overlay, sizeof(display_overlay));
+ break;
+ case MM_DISPLAY_SURFACE_EVAS:
+ debug_log("save attributes related to display surface to EVAS : evas image object = %x", display_overlay);
+ mm_attrs_set_data_by_name (player->attrs, "display_overlay", display_overlay, sizeof(display_overlay));
+ break;
+ default:
+ debug_error("invalid type(%d) for changing display surface",surface_type);
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
+ if ( mmf_attrs_commit ( player->attrs ) )
+ {
+ debug_error("failed to commit");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ }
+ else
+ {
+ debug_error("player->attrs is null, failed to save attributes");
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- debug_fleave();
+ /* update video param */
+ if ( MM_ERROR_NONE != _mmplayer_update_video_param( player ) )
+ {
+ debug_error("failed to update video param");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- /* we are returning FALSE as we need only one posting */
- return FALSE;
-}
+ /* change state of videobin to READY */
+ debug_log("setting [%s] state to : %d", GST_ELEMENT_NAME(player->pipeline->videobin[MMPLAYER_V_BIN].gst), GST_STATE_READY);
+ ret = gst_element_set_state(player->pipeline->videobin[MMPLAYER_V_BIN].gst, GST_STATE_READY);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ {
+ debug_error("failed to change state of videobin to READY");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
-static void __mmplayer_set_antishock( mm_player_t* player, gboolean disable_by_force)
-{
- gint antishock = FALSE;
- MMHandleType attrs = 0;
+ /* change ghostpad */
+ sink_pad_videosink = gst_element_get_static_pad(GST_ELEMENT(player->pipeline->videobin[MMPLAYER_V_SINK].gst), "sink");
+ if ( !sink_pad_videosink )
+ {
+ debug_error("failed to get sink pad from videosink element");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ player->ghost_pad_for_videobin = gst_ghost_pad_new("sink", sink_pad_videosink);
+ if (!gst_pad_set_active(player->ghost_pad_for_videobin, TRUE))
+ {
+ debug_error("failed to set active to ghost_pad");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ if ( FALSE == gst_element_add_pad(player->pipeline->videobin[MMPLAYER_V_BIN].gst, player->ghost_pad_for_videobin) )
+ {
+ debug_error("failed to change ghostpad for videobin");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ gst_object_unref(sink_pad_videosink);
- debug_fenter();
+ /* link decoder with videobin */
+ sink_pad_videobin = gst_element_get_static_pad( GST_ELEMENT(player->pipeline->videobin[MMPLAYER_V_BIN].gst), "sink");
+ if ( !sink_pad_videobin )
+ {
+ debug_error("failed to get sink pad from videobin");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(src_pad_dec, sink_pad_videobin) )
+ {
+ debug_error("failed to link");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ gst_object_unref(sink_pad_videobin);
- return_if_fail ( player && player->pipeline );
+ /* clock setting for a new videosink plugin */
+ /* NOTE : Below operation is needed, because a new videosink plugin doesn't have clock for basesink,
+ so we set it from audiosink plugin or pipeline(system clock) */
+ if (!is_audiobin_created)
+ {
+ debug_warning("audiobin is not created, get clock from pipeline..");
+ clock = GST_ELEMENT_CLOCK (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst);
+ }
+ else
+ {
+ clock = GST_ELEMENT_CLOCK (player->pipeline->audiobin[MMPLAYER_A_SINK].gst);
+ }
+ if (clock)
+ {
+ GstClockTime now;
+ GstClockTime base_time;
+ debug_log("set the clock to videosink");
+ gst_element_set_clock (GST_ELEMENT_CAST(player->pipeline->videobin[MMPLAYER_V_SINK].gst), clock);
+ clock = GST_ELEMENT_CLOCK (player->pipeline->videobin[MMPLAYER_V_SINK].gst);
+ if (clock)
+ {
+ debug_log("got clock of videosink");
+ now = gst_clock_get_time ( clock );
+ base_time = GST_ELEMENT_CAST (player->pipeline->videobin[MMPLAYER_V_SINK].gst)->base_time;
+ debug_log ("at time %" GST_TIME_FORMAT ", base %"
+ GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
+ }
+ else
+ {
+ debug_error("failed to get clock of videosink after setting clock");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ }
+ else
+ {
+ debug_warning("failed to get clock, maybe it is the time before first playing");
+ }
- /* It should be passed for video only clip */
- if ( ! player->pipeline->audiobin )
- return;
+ if (!player->doing_seek && previous_state == MM_PLAYER_STATE_PLAYING)
+ {
+ /* change state of videobin to PAUSED */
+ debug_log("setting [%s] state to : %d", GST_ELEMENT_NAME(player->pipeline->videobin[MMPLAYER_V_BIN].gst), GST_STATE_PLAYING);
+ ret = gst_element_set_state(player->pipeline->videobin[MMPLAYER_V_BIN].gst, GST_STATE_PLAYING);
+ if (ret != GST_STATE_CHANGE_FAILURE)
+ {
+ debug_warning("change state of videobin to PLAYING, ret(%d)", ret);
+ }
+ else
+ {
+ debug_error("failed to change state of videobin to PLAYING");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
- if ( ( g_strrstr(PLAYER_INI()->name_of_audiosink, "avsysaudiosink")) )
+ /* release blocked and unref src pad of video decoder */
+ if (!gst_pad_set_blocked (src_pad_dec, FALSE))
+ {
+ debug_error("failed to set pad blocked FALSE(video)");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ debug_warning("pad is unblocked(video)");
+ }
+ else
{
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( ! attrs )
+ if (player->doing_seek) {
+ debug_warning("not completed seek(%d)", player->doing_seek);
+ }
+ /* change state of videobin to PAUSED */
+ debug_log("setting [%s] state to : %d", GST_ELEMENT_NAME(player->pipeline->videobin[MMPLAYER_V_BIN].gst), GST_STATE_PAUSED);
+ ret = gst_element_set_state(player->pipeline->videobin[MMPLAYER_V_BIN].gst, GST_STATE_PAUSED);
+ if (ret != GST_STATE_CHANGE_FAILURE)
{
- debug_error("fail to get attributes.\n");
- return;
+ debug_warning("change state of videobin to PAUSED, ret(%d)", ret);
+ }
+ else
+ {
+ debug_error("failed to change state of videobin to PLAYING");
+ return MM_ERROR_PLAYER_INTERNAL;
}
- mm_attrs_get_int_by_name(attrs, "sound_fadeup", &antishock);
+ /* already skipped pad block */
+ debug_log("previous MM_PLAYER_STATE is not PLAYING, skip pad-block(FALSE)");
+ }
- debug_log("setting antishock as (%d)\n", antishock);
+ /* do get/set position for new videosink plugin */
+ {
+ unsigned long position = 0;
+ gint64 pos_msec = 0;
- if ( disable_by_force )
+ debug_log("do get/set position for new videosink plugin");
+ if (__gst_get_position(player, MM_PLAYER_POS_FORMAT_TIME, &position ))
{
- debug_log("but, antishock is disabled by force when is seeked\n");
-
- antishock = FALSE;
+ debug_error("failed to get position");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+#ifdef SINKCHANGE_WITH_ACCURATE_SEEK
+ /* accurate seek */
+ if (__gst_set_position(player, MM_PLAYER_POS_FORMAT_TIME, position, TRUE ))
+ {
+ debug_error("failed to set position");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+#else
+ /* key unit seek */
+ pos_msec = position * G_GINT64_CONSTANT(1000000);
+ ret = __gst_seek ( player, player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, 1.0,
+ GST_FORMAT_TIME, ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT ),
+ GST_SEEK_TYPE_SET, pos_msec,
+ GST_SEEK_TYPE_NONE, GST_CLOCK_TIME_NONE );
+ if ( !ret )
+ {
+ debug_error("failed to set position");
+ return MM_ERROR_PLAYER_INTERNAL;
}
+#endif
+ }
- g_object_set(G_OBJECT(player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "fadeup", antishock, NULL);
+ if (src_pad_dec)
+ {
+ gst_object_unref (src_pad_dec);
}
+ debug_log("success to change sink");
- debug_fleave();
+ MMPLAYER_FLEAVE();
- return;
+ return MM_ERROR_NONE;
}
-
-static gboolean
-__mmplayer_link_decoder( mm_player_t* player, GstPad *srcpad)
+int _mmplayer_sync_subtitle_pipeline(mm_player_t* player)
{
- const gchar* name = NULL;
- GstStructure* str = NULL;
- GstCaps* srccaps = NULL;
-
- debug_fenter();
+ MMPlayerGstElement* mainbin = NULL;
+ MMPlayerGstElement* textbin = NULL;
+ GstStateChangeReturn ret = GST_STATE_CHANGE_FAILURE;
+ GstState current_state = GST_STATE_VOID_PENDING;
+ GstState element_state = GST_STATE_VOID_PENDING;
+ GstState element_pending_state = GST_STATE_VOID_PENDING;
+ gint64 time = 0;
+ GstFormat format = GST_FORMAT_TIME;
+ GstEvent *event = NULL;
+ int result = MM_ERROR_NONE;
- return_val_if_fail( player, FALSE );
- return_val_if_fail ( srcpad, FALSE );
+ GstClock *curr_clock = NULL;
+ GstClockTime base_time, start_time, curr_time;
- /* to check any of the decoder (video/audio) need to be linked to parser*/
- srccaps = gst_pad_get_caps( srcpad );
- if ( !srccaps )
- goto ERROR;
- str = gst_caps_get_structure( srccaps, 0 );
- if ( ! str )
- goto ERROR;
+ MMPLAYER_FENTER();
- name = gst_structure_get_name(str);
- if ( ! name )
- goto ERROR;
+ /* check player handle */
+ return_val_if_fail ( player && player->pipeline , MM_ERROR_PLAYER_NOT_INITIALIZED);
- if (strstr(name, "video"))
- {
- if(player->videodec_linked)
- {
- debug_msg("Video decoder already linked\n");
- return FALSE;
- }
- }
- if (strstr(name, "audio"))
+ if (!(player->pipeline->mainbin) || !(player->pipeline->textbin))
{
- if(player->audiodec_linked)
- {
- debug_msg("Audio decoder already linked\n");
- return FALSE;
- }
+ debug_error("Pipeline is not in proper state\n");
+ result = MM_ERROR_PLAYER_NOT_INITIALIZED;
+ goto EXIT;
}
- gst_caps_unref( srccaps );
-
- debug_fleave();
-
- return TRUE;
+ mainbin = player->pipeline->mainbin;
+ textbin = player->pipeline->textbin;
-ERROR:
- if ( srccaps )
- gst_caps_unref( srccaps );
+ current_state = GST_STATE (mainbin[MMPLAYER_M_PIPE].gst);
- return FALSE;
-}
+ // sync clock with current pipeline
+ curr_clock = GST_ELEMENT_CLOCK (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst);
+ curr_time = gst_clock_get_time (curr_clock);
-static gboolean
-__mmplayer_link_sink( mm_player_t* player , GstPad *srcpad)
-{
- const gchar* name = NULL;
- GstStructure* str = NULL;
- GstCaps* srccaps = NULL;
+ base_time = gst_element_get_base_time (GST_ELEMENT_CAST (player->pipeline->mainbin[MMPLAYER_M_PIPE].gst));
+ start_time = gst_element_get_start_time (GST_ELEMENT_CAST(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst));
- debug_fenter();
+ debug_log ("base_time=%" GST_TIME_FORMAT " start_time=%" GST_TIME_FORMAT " curr_time=%" GST_TIME_FORMAT,
+ GST_TIME_ARGS (base_time), GST_TIME_ARGS (start_time), GST_TIME_ARGS (curr_time));
- return_val_if_fail ( player, FALSE );
- return_val_if_fail ( srcpad, FALSE );
+ if (current_state > GST_STATE_READY)
+ {
+ // sync state with current pipeline
+ gst_element_set_state(textbin[MMPLAYER_T_BIN].gst, GST_STATE_PAUSED);
+ gst_element_set_state(mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, GST_STATE_PAUSED); // Naveen : enable if required only
+ gst_element_set_state(mainbin[MMPLAYER_M_T_SUBMUX_EXTERNAL].gst, GST_STATE_PAUSED);
+ gst_element_set_state(mainbin[MMPLAYER_M_SUBSRC].gst, GST_STATE_PAUSED);
- /* to check any of the decoder (video/audio) need to be linked to parser*/
- srccaps = gst_pad_get_caps( srcpad );
- if ( !srccaps )
- goto ERROR;
+ ret = gst_element_get_state (mainbin[MMPLAYER_M_SUBSRC].gst, &element_state, &element_pending_state, 5 * GST_SECOND);
+ if ( GST_STATE_CHANGE_FAILURE == ret )
+ {
+ debug_error("fail to state change.\n");
+ }
+ }
- str = gst_caps_get_structure( srccaps, 0 );
- if ( ! str )
- goto ERROR;
+ gst_element_set_base_time (textbin[MMPLAYER_T_BIN].gst, base_time);
+ gst_element_set_start_time(textbin[MMPLAYER_T_BIN].gst, start_time);
- name = gst_structure_get_name(str);
- if ( ! name )
- goto ERROR;
+ if (curr_clock)
+ {
+ gst_element_set_clock (textbin[MMPLAYER_T_BIN].gst, curr_clock);
+ gst_object_unref (curr_clock);
+ }
- if (strstr(name, "video"))
+ // seek to current position
+ if (!gst_element_query_position (mainbin[MMPLAYER_M_PIPE].gst, &format, &time))
{
- if(player->videosink_linked)
- {
- debug_msg("Video Sink already linked\n");
- return FALSE;
- }
+ result = MM_ERROR_PLAYER_INVALID_STATE;
+ debug_error("gst_element_query_position failed, invalid state\n");
+ goto EXIT;
}
- if (strstr(name, "audio"))
+
+ debug_log("seek time = %lld\n", time);
+
+ event = gst_event_new_seek (1.0, GST_FORMAT_TIME, (GstSeekFlags) (GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_SKIP), GST_SEEK_TYPE_SET, time, GST_SEEK_TYPE_NONE, -1);
+ if (event)
{
- if(player->audiosink_linked)
- {
- debug_msg("Audio Sink already linked\n");
- return FALSE;
- }
+ __gst_send_event_to_sink(player, event);
}
- if (strstr(name, "text"))
+ else
{
- if(player->textsink_linked)
- {
- debug_msg("Text Sink already linked\n");
- return FALSE;
- }
+ result = MM_ERROR_PLAYER_INTERNAL;
+ debug_error("gst_event_new_seek failed\n");
+ goto EXIT;
}
- gst_caps_unref( srccaps );
-
- debug_fleave();
+ // sync state with current pipeline
+ gst_element_sync_state_with_parent(textbin[MMPLAYER_T_BIN].gst);
+ gst_element_sync_state_with_parent(mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst);
+ gst_element_sync_state_with_parent(mainbin[MMPLAYER_M_T_SUBMUX_EXTERNAL].gst);
+ gst_element_sync_state_with_parent(mainbin[MMPLAYER_M_SUBSRC].gst);
- return TRUE;
- //return (!player->videosink_linked || !player->audiosink_linked);
+EXIT:
+ return result;
+}
-ERROR:
- if ( srccaps )
- gst_caps_unref( srccaps );
+static int
+__mmplayer_change_external_subtitle_language(mm_player_t* player, const char* filepath)
+{
+ GstStateChangeReturn ret = GST_STATE_CHANGE_FAILURE;
+ GstState current_state = GST_STATE_VOID_PENDING;
- return FALSE;
-}
+ MMHandleType attrs = 0;
+ MMPlayerGstElement* mainbin = NULL;
+ MMPlayerGstElement* textbin = NULL;
+ gchar* subtitle_uri = NULL;
+ int result = MM_ERROR_NONE;
+ const gchar *charset = NULL;
-/* sending event to one of sinkelements */
-static gboolean
-__gst_send_event_to_sink( mm_player_t* player, GstEvent* event )
-{
- GList *sinks = NULL;
- gboolean res = FALSE;
+ MMPLAYER_FENTER();
- debug_fenter();
-
- return_val_if_fail( player, FALSE );
- return_val_if_fail ( event, FALSE );
+ /* check player handle */
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( filepath, MM_ERROR_COMMON_INVALID_ARGUMENT );
- sinks = player->sink_elements;
- while (sinks)
+ if (!(player->pipeline) || !(player->pipeline->mainbin))
{
- GstElement *sink = GST_ELEMENT_CAST (sinks->data);
-
- if ( sink )
- {
- /* keep ref to the event */
- gst_event_ref (event);
+ result = MM_ERROR_PLAYER_INVALID_STATE;
+ debug_error("Pipeline is not in proper state\n");
+ goto EXIT;
+ }
- debug_log("sending event[%s] to sink element [%s]\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(sink) );
+ mainbin = player->pipeline->mainbin;
+ textbin = player->pipeline->textbin;
- if ( (res = gst_element_send_event (sink, event)) )
- {
- debug_log("sending event[%s] to sink element [%s] success!\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(sink) );
- break;
- }
+ current_state = GST_STATE (mainbin[MMPLAYER_M_PIPE].gst);
+ if (current_state < GST_STATE_READY)
+ {
+ result = MM_ERROR_PLAYER_INVALID_STATE;
+ debug_error("Pipeline is not in proper state\n");
+ goto EXIT;
+ }
- debug_log("sending event[%s] to sink element [%s] failed. try with next one.\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(sink) );
- }
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if (!attrs)
+ {
+ debug_error("cannot get content attribute\n");
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
- sinks = g_list_next (sinks);
+ mm_attrs_get_string_by_name (attrs, "subtitle_uri", &subtitle_uri);
+ if (!subtitle_uri || strlen(subtitle_uri) < 1)
+ {
+ debug_error("subtitle uri is not proper filepath\n");
+ result = MM_ERROR_PLAYER_INVALID_URI;
+ goto EXIT;
}
- /* Note : Textbin is not linked to the video or audio bin.
- * It needs to send the event to the text sink seperatelly.
- */
- if ( MMPLAYER_PLAY_SUBTITLE(player) )
- {
- GstElement *subtitle_sink = GST_ELEMENT_CAST (player->pipeline->subtitlebin[MMPLAYER_SUB_SINK].gst);
+ debug_log("old subtitle file path is [%s]\n", subtitle_uri);
+ debug_log("new subtitle file path is [%s]\n", filepath);
- if ( (res != gst_element_send_event (subtitle_sink, event)) )
- {
- debug_error("sending event[%s] to subtitle sink element [%s] failed!\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(subtitle_sink) );
- }
- else
+ if (!strcmp (filepath, subtitle_uri))
+ {
+ debug_log("No need to swtich subtitle, as input filepath is same as current filepath\n");
+ goto EXIT;
+ }
+ else
+ {
+ mm_attrs_set_string_by_name(player->attrs, "subtitle_uri", filepath);
+ if (mmf_attrs_commit(player->attrs))
{
- debug_log("sending event[%s] to subtitle sink element [%s] success!\n",
- GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(subtitle_sink) );
+ debug_error("failed to commit.\n");
+ goto EXIT;
}
- }
-
- gst_event_unref (event);
-
- debug_fleave();
-
- return res;
-}
-
-static void
-__mmplayer_add_sink( mm_player_t* player, GstElement* sink )
-{
- debug_fenter();
-
- return_if_fail ( player );
- return_if_fail ( sink );
+ }
- player->sink_elements =
- g_list_append(player->sink_elements, sink);
+ //gst_pad_set_blocked_async(src-srcpad, TRUE)
- debug_fleave();
-}
+ ret = gst_element_set_state(textbin[MMPLAYER_T_BIN].gst, GST_STATE_READY);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ {
+ debug_error("failed to change state of textbin to READY");
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
-static gboolean
-__gst_seek(mm_player_t* player, GstElement * element, gdouble rate,
- GstFormat format, GstSeekFlags flags, GstSeekType cur_type,
- gint64 cur, GstSeekType stop_type, gint64 stop )
-{
- GstEvent* event = NULL;
- gboolean result = FALSE;
+ ret = gst_element_set_state(mainbin[MMPLAYER_M_T_INPUT_SELECTOR].gst, GST_STATE_READY);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ {
+ debug_error("failed to change state of input-selector to READY");
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
- debug_fenter();
-
- return_val_if_fail( player, FALSE );
+ ret = gst_element_set_state(mainbin[MMPLAYER_M_T_SUBMUX_EXTERNAL].gst, GST_STATE_READY);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ {
+ debug_error("failed to change state of subparse to READY");
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
- event = gst_event_new_seek (rate, format, flags, cur_type,
- cur, stop_type, stop);
+ ret = gst_element_set_state(mainbin[MMPLAYER_M_SUBSRC].gst, GST_STATE_READY);
+ if (ret != GST_STATE_CHANGE_SUCCESS)
+ {
+ debug_error("failed to change state of filesrc to READY");
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
- result = __gst_send_event_to_sink( player, event );
+ g_object_set(G_OBJECT(mainbin[MMPLAYER_M_SUBSRC].gst), "location", filepath, NULL);
- /* Note : After sending seek event, the sink elements receive flush event
- * Not only buffering state but after seeking, state_lost needs to be set to TRUE.
- */
- if ( MMPLAYER_IS_RTSP_STREAMING ( player ) && result )
- player->state_lost = TRUE;
+ charset = util_get_charset(filepath);
+ if (charset)
+ {
+ debug_log ("detected charset is %s\n", charset );
+ g_object_set (G_OBJECT (mainbin[MMPLAYER_M_T_SUBMUX_EXTERNAL].gst), "subtitle-encoding", charset, NULL);
+ }
- debug_fleave();
+ result = _mmplayer_sync_subtitle_pipeline(player);
+EXIT:
+ MMPLAYER_FLEAVE();
return result;
}
-/* NOTE : be careful with calling this api. please refer to below glib comment
- * glib comment : Note that there is a bug in GObject that makes this function much
- * less useful than it might seem otherwise. Once gobject is disposed, the callback
- * will no longer be called, but, the signal handler is not currently disconnected.
- * If the instance is itself being freed at the same time than this doesn't matter,
- * since the signal will automatically be removed, but if instance persists,
- * then the signal handler will leak. You should not remove the signal yourself
- * because in a future versions of GObject, the handler will automatically be
- * disconnected.
- *
- * It's possible to work around this problem in a way that will continue to work
- * with future versions of GObject by checking that the signal handler is still
- * connected before disconnected it:
- *
- * if (g_signal_handler_is_connected (instance, id))
- * g_signal_handler_disconnect (instance, id);
- */
-static void
-__mmplayer_release_signal_connection(mm_player_t* player)
+/* API to switch between external subtitles */
+int _mmplayer_set_external_subtitle_path(MMHandleType hplayer, const char* filepath)
{
- GList* sig_list = player->signals;
- MMPlayerSignalItem* item = NULL;
+ int result = MM_ERROR_NONE;
+ mm_player_t* player = (mm_player_t*)hplayer;
- debug_fenter();
-
- return_if_fail( player );
+ MMPLAYER_FENTER();
+
+ /* check player handle */
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- for ( ; sig_list; sig_list = sig_list->next )
+ if (!player->pipeline) // IDLE state
{
- item = sig_list->data;
-
- if ( item && item->obj && GST_IS_ELEMENT(item->obj) )
+ mm_attrs_set_string_by_name(player->attrs, "subtitle_uri", filepath);
+ if (mmf_attrs_commit(player->attrs))
{
- debug_log("checking signal connection : [%lud] from [%s]\n", item->sig, GST_OBJECT_NAME( item->obj ));
+ debug_error("failed to commit.\n");
+ result= MM_ERROR_PLAYER_INTERNAL;
+ }
+ }
+ else // curr state <> IDLE (READY, PAUSE, PLAYING..)
+ {
+ if ( filepath == NULL )
+ return MM_ERROR_COMMON_INVALID_ARGUMENT;
- if ( g_signal_handler_is_connected ( item->obj, item->sig ) )
+ if (!__mmplayer_check_subtitle(player))
+ {
+ mm_attrs_set_string_by_name(player->attrs, "subtitle_uri", filepath);
+ if (mmf_attrs_commit(player->attrs))
{
- debug_log("signal disconnecting : [%lud] from [%s]\n", item->sig, GST_OBJECT_NAME( item->obj ));
- g_signal_handler_disconnect ( item->obj, item->sig );
+ debug_error("failed to commit.\n");
+ result = MM_ERROR_PLAYER_INTERNAL;
}
- }
- MMPLAYER_FREEIF( item );
+ if ( MM_ERROR_NONE != __mmplayer_gst_create_subtitle_src(player) )
+ debug_error("fail to create subtitle src\n");
+ result = _mmplayer_sync_subtitle_pipeline(player);
+ }
+ else
+ {
+ result = __mmplayer_change_external_subtitle_language(player, filepath);
+ }
}
- g_list_free ( player->signals );
- player->signals = NULL;
-
- debug_fleave();
- return;
+ MMPLAYER_FLEAVE();
+ return result;
}
-
-/* Note : if silent is true, then subtitle would not be displayed. :*/
-int _mmplayer_set_subtitle_silent (MMHandleType hplayer, int silent)
+static int
+__mmplayer_change_selector_pad (mm_player_t* player, MMPlayerTrackType type, int index)
{
- mm_player_t* player = (mm_player_t*) hplayer;
+ int result = MM_ERROR_NONE;
+ gchar* change_pad_name = NULL;
+ GstPad* sinkpad = NULL;
+ MMPlayerGstElement* mainbin = NULL;
+ enum MainElementID elemId = MMPLAYER_M_NUM;
+ GstCaps* caps = NULL;
+ gint total_track_num = 0;
- debug_fenter();
+ MMPLAYER_FENTER();
- /* check player handle */
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( MMPLAYER_PLAY_SUBTITLE(player), MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail (player && player->pipeline && player->pipeline->mainbin,
+ MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ debug_log ("Change Track(%d) to %d\n", type, index);
+
+ mainbin = player->pipeline->mainbin;
+
+ if (type == MM_PLAYER_TRACK_TYPE_AUDIO)
+ {
+ elemId = MMPLAYER_M_A_INPUT_SELECTOR;
+ }
+ else if (type == MM_PLAYER_TRACK_TYPE_TEXT)
+ {
+ elemId = MMPLAYER_M_T_INPUT_SELECTOR;
+ }
+ else
+ {
+ debug_error ("Track Type Error\n");
+ goto EXIT;
+ }
- player->is_subtitle_off = silent;
+ if (mainbin[elemId].gst == NULL)
+ {
+ result = MM_ERROR_PLAYER_NO_OP;
+ debug_log ("Req track doesn't exist\n");
+ goto EXIT;
+ }
- debug_log("subtitle is %s.\n", player->is_subtitle_off ? "ON" : "OFF");
+ total_track_num = player->selector[type].total_track_num;
+ if (total_track_num <= 0)
+ {
+ result = MM_ERROR_PLAYER_NO_OP;
+ debug_log ("Language list is not available \n");
+ goto EXIT;
+ }
- debug_fleave();
+ if ((index < 0) || (index >= total_track_num))
+ {
+ result = MM_ERROR_INVALID_ARGUMENT;
+ debug_log ("Not a proper index : %d \n", index);
+ goto EXIT;
+ }
- return MM_ERROR_NONE;
-}
+ /*To get the new pad from the selector*/
+ change_pad_name = g_strdup_printf ("sink%d", index);
+ if (change_pad_name == NULL)
+ {
+ result = MM_ERROR_PLAYER_INTERNAL;
+ debug_log ("Pad does not exists\n");
+ goto EXIT;
+ }
+ debug_log ("new active pad name: %s\n", change_pad_name);
-int _mmplayer_get_subtitle_silent (MMHandleType hplayer, int* silent)
-{
- mm_player_t* player = (mm_player_t*) hplayer;
+ sinkpad = gst_element_get_static_pad (mainbin[elemId].gst, change_pad_name);
+ if (sinkpad == NULL)
+ {
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
- debug_fenter();
+ debug_log ("Set Active Pad - %s:%s\n", GST_DEBUG_PAD_NAME(sinkpad));
+ g_object_set (mainbin[elemId].gst, "active-pad", sinkpad, NULL);
- /* check player handle */
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED );
- return_val_if_fail ( MMPLAYER_PLAY_SUBTITLE(player), MM_ERROR_PLAYER_NOT_INITIALIZED );
+ caps = GST_PAD_CAPS(sinkpad);
+ MMPLAYER_LOG_GST_CAPS_TYPE(caps);
- *silent = player->is_subtitle_off;
+ if (sinkpad)
+ gst_object_unref (sinkpad);
- debug_log("subtitle is %s.\n", silent ? "ON" : "OFF");
+ if (type == MM_PLAYER_TRACK_TYPE_AUDIO)
+ {
+ __mmplayer_set_audio_attrs (player, caps);
+ }
- debug_fleave();
+EXIT:
- return MM_ERROR_NONE;
+ MMPLAYER_FREEIF(change_pad_name);
+ return result;
}
-int _mmplayer_get_track_count(MMHandleType hplayer, MMPlayerTrackType track_type, int *count)
+int _mmplayer_change_track_language (MMHandleType hplayer, MMPlayerTrackType type, int index)
{
- mm_player_t* player = (mm_player_t*) hplayer;
- MMHandleType attrs = 0;
- int ret = MM_ERROR_NONE;
+ int result = MM_ERROR_NONE;
+ mm_player_t* player = NULL;
+ MMPlayerGstElement* mainbin = NULL;
- debug_fenter();
+ gint current_active_index = 0;
- /* check player handle */
- return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
- return_val_if_fail(count, MM_ERROR_COMMON_INVALID_ARGUMENT);
- return_val_if_fail((MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PAUSED)
- ||(MMPLAYER_CURRENT_STATE(player) != MM_PLAYER_STATE_PLAYING),
- MM_ERROR_PLAYER_INVALID_STATE);
+ GstState current_state = GST_STATE_VOID_PENDING;
+ GstEvent* event = NULL;
+ GstFormat format = GST_FORMAT_TIME;
+ gint64 time = 0;
- attrs = MMPLAYER_GET_ATTRS(player);
- if ( !attrs )
+ MMPLAYER_FENTER();
+
+ player = (mm_player_t*)hplayer;
+ return_val_if_fail (player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ if (!player->pipeline)
{
- debug_error("cannot get content attribute");
- return MM_ERROR_PLAYER_INTERNAL;
+ debug_error ("Track %d pre setting -> %d\n", type, index);
+
+ player->selector[type].active_pad_index = index;
+ goto EXIT;
}
- switch (track_type)
+ mainbin = player->pipeline->mainbin;
+
+ current_active_index = player->selector[type].active_pad_index;
+
+ /*If index is same as running index no need to change the pad*/
+ if (current_active_index == index)
{
- case MM_PLAYER_TRACK_TYPE_AUDIO:
- ret = mm_attrs_get_int_by_name(attrs, "content_audio_track_num", count);
- break;
- case MM_PLAYER_TRACK_TYPE_VIDEO:
- ret = mm_attrs_get_int_by_name(attrs, "content_video_track_num", count);
- break;
- case MM_PLAYER_TRACK_TYPE_TEXT:
- ret = mm_attrs_get_int_by_name(attrs, "content_text_track_num", count);
- break;
- default:
- ret = MM_ERROR_COMMON_INVALID_ARGUMENT;
- break;
+ goto EXIT;
+ }
+
+ if (!gst_element_query_position(mainbin[MMPLAYER_M_PIPE].gst, &format, &time))
+ {
+ result = MM_ERROR_PLAYER_INVALID_STATE;
+ goto EXIT;
}
- debug_log ("%d track num is %d\n", track_type, *count);
+ current_state = GST_STATE(mainbin[MMPLAYER_M_PIPE].gst);
+ if (current_state < GST_STATE_PAUSED)
+ {
+ result = MM_ERROR_PLAYER_INVALID_STATE;
+ debug_warning ("Pipeline not in porper state\n");
+ goto EXIT;
+ }
- debug_fleave();
+ result = __mmplayer_change_selector_pad(player, type, index);
+ if (result != MM_ERROR_NONE)
+ {
+ debug_error ("change selector pad error\n");
+ goto EXIT;
+ }
- return ret;
-}
+ player->selector[type].active_pad_index = index;
+ if (current_state == GST_STATE_PLAYING)
+ {
+ event = gst_event_new_seek (1.0, GST_FORMAT_TIME,(GstSeekFlags) (GST_SEEK_FLAG_SEGMENT | GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_SKIP),GST_SEEK_TYPE_SET, time, GST_SEEK_TYPE_NONE, -1);
+ if (event)
+ {
+ __gst_send_event_to_sink (player, event);
+ }
+ else
+ {
+ result = MM_ERROR_PLAYER_INTERNAL;
+ goto EXIT;
+ }
+ }
+EXIT:
+ return result;
+}
-const gchar *
+const gchar *
__get_state_name ( int state )
{
switch ( state )
return "INVAID";
}
}
+
gboolean
__is_rtsp_streaming ( mm_player_t* player )
{
return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_URL_RTSP ) ? TRUE : FALSE;
}
+gboolean
+__is_wfd_streaming ( mm_player_t* player )
+{
+ return_val_if_fail ( player, FALSE );
+
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_URL_WFD ) ? TRUE : FALSE;
+}
+
static gboolean
__is_http_streaming ( mm_player_t* player )
{
{
return_val_if_fail ( player, FALSE );
- return ( __is_rtsp_streaming ( player ) || __is_http_streaming ( player ) || __is_http_live_streaming ( player )) ? TRUE : FALSE;
+ return ( __is_rtsp_streaming ( player ) || __is_wfd_streaming ( player ) || __is_http_streaming ( player )
+ || __is_http_live_streaming ( player ) || __is_dash_streaming ( player ) || __is_smooth_streaming(player) ) ? TRUE : FALSE;
}
gboolean
{
return_val_if_fail( player, FALSE );
- return ( (player->ahs_player) && (player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS) ) ? TRUE : FALSE;
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS ) ? TRUE : FALSE;
+}
+
+static gboolean
+__is_dash_streaming ( mm_player_t* player )
+{
+ return_val_if_fail ( player, FALSE );
+
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_DASH ) ? TRUE : FALSE;
+}
+
+static gboolean
+__is_smooth_streaming ( mm_player_t* player )
+{
+ return_val_if_fail ( player, FALSE );
+
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_SS ) ? TRUE : FALSE;
}
+
static gboolean
__is_http_progressive_down(mm_player_t* player)
{
return ((player->pd_mode) ? TRUE:FALSE);
}
+
+gboolean
+__has_suffix(mm_player_t* player, const gchar* suffix)
+{
+ return_val_if_fail( player, FALSE );
+ return_val_if_fail( suffix, FALSE );
+
+ gboolean ret = FALSE;
+ gchar* t_url = g_ascii_strdown(player->profile.uri, -1);
+ gchar* t_suffix = g_ascii_strdown(suffix, -1);
+
+ if ( g_str_has_suffix(player->profile.uri, suffix) )
+ {
+ ret = TRUE;
+ }
+
+ MMPLAYER_FREEIF(t_url);
+ MMPLAYER_FREEIF(t_suffix);
+
+ return ret;
+}
+
+int
+_mmplayer_set_display_zoom(MMHandleType hplayer, float level, int x, int y)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ MMPLAYER_VIDEO_SINK_CHECK(player);
+
+ debug_log("setting display zoom level = %f, offset = %d, %d", level, x, y);
+
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst, "zoom", level, "zoom-pos-x", x, "zoom-pos-y", y, NULL);
+
+ return MM_ERROR_NONE;
+}
+int
+_mmplayer_get_display_zoom(MMHandleType hplayer, float *level, int *x, int *y)
+{
+
+ mm_player_t* player = (mm_player_t*) hplayer;
+ float _level = 0.0;
+ int _x = 0;
+ int _y = 0;
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ MMPLAYER_VIDEO_SINK_CHECK(player);
+
+ g_object_get(player->pipeline->videobin[MMPLAYER_V_SINK].gst, "zoom", &_level, "zoom-pos-x", &_x, "zoom-pos-y", &_y, NULL);
+
+ debug_log("display zoom level = %f, start off x = %d, y = %d", _level, _x, _y);
+
+ *level = _level;
+ *x = _x;
+ *y = _y;
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_get_video_rotate_angle(MMHandleType hplayer, int *angle)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ int org_angle = 0;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( angle, MM_ERROR_COMMON_INVALID_ARGUMENT );
+
+ if (player->v_stream_caps)
+ {
+ GstStructure *str = NULL;
+
+ str = gst_caps_get_structure (player->v_stream_caps, 0);
+ if ( !gst_structure_get_int (str, "orientation", &org_angle))
+ {
+ debug_log ("missing 'orientation' field in video caps");
+ }
+ }
+
+ debug_log("orientation: %d", org_angle);
+ *angle = org_angle;
+
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+}
+
+gboolean
+__mmplayer_is_streaming(mm_player_t* player)
+{
+ gboolean result = FALSE;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail (player, FALSE);
+ result = __is_streaming (player) ;
+
+ MMPLAYER_FLEAVE();
+ return result;
+}
+
+static gboolean
+__mmplayer_add_dump_buffer_probe(mm_player_t *player, GstElement *element)
+{
+ return_val_if_fail (player, FALSE);
+ return_val_if_fail (element, FALSE);
+
+ gchar *factory_name = GST_PLUGIN_FEATURE_NAME (gst_element_get_factory(element));
+ gchar dump_file_name[PLAYER_INI_MAX_STRLEN*2];
+ int len = PLAYER_INI_MAX_STRLEN*2;
+
+ int idx = 0;
+
+ for ( idx = 0; player->ini.dump_element_keyword[idx][0] != '\0'; idx++ )
+ {
+ if (g_strrstr(factory_name, player->ini.dump_element_keyword[idx]))
+ {
+ debug_log("dump [%s] sink pad", player->ini.dump_element_keyword[idx]);
+ mm_player_dump_t *dump_s;
+ dump_s = g_malloc (sizeof(mm_player_dump_t));
+
+ if (dump_s == NULL)
+ {
+ debug_error ("malloc fail");
+ return FALSE;
+ }
+
+ dump_s->dump_element_file = NULL;
+ dump_s->dump_pad = NULL;
+ dump_s->dump_pad = gst_element_get_static_pad (element, "sink");
+
+ if (dump_s->dump_pad)
+ {
+ memset (dump_file_name, 0x00, PLAYER_INI_MAX_STRLEN*2);
+ snprintf (dump_file_name, len, "%s/%s_sink_pad.dump", player->ini.dump_element_path, player->ini.dump_element_keyword[idx]);
+ dump_s->dump_element_file = fopen(dump_file_name,"w+");
+ dump_s->probe_handle_id = gst_pad_add_buffer_probe (dump_s->dump_pad, G_CALLBACK(__mmplayer_dump_buffer_probe_cb), dump_s->dump_element_file);
+ /* add list for removed buffer probe and close FILE */
+ player->dump_list = g_list_append (player->dump_list, dump_s);
+ debug_log ("%s sink pad added buffer probe for dump", factory_name);
+ return TRUE;
+ }
+ else
+ {
+ g_free(dump_s);
+ dump_s = NULL;
+ debug_error ("failed to get %s sink pad added", factory_name);
+ }
+
+
+ }
+ }
+ return FALSE;
+}
+
+static gboolean
+__mmplayer_dump_buffer_probe_cb(GstPad *pad, GstBuffer *buffer, gpointer u_data)
+{
+ FILE *dump_data = (FILE *) u_data;
+// int written = 0;
+
+ return_val_if_fail ( dump_data, FALSE );
+
+// debug_log ("buffer timestamp = %" GST_TIME_FORMAT, GST_TIME_ARGS( GST_BUFFER_TIMESTAMP(buffer)));
+
+ fwrite ( GST_BUFFER_DATA(buffer), 1, GST_BUFFER_SIZE(buffer), dump_data);
+
+ return TRUE;
+}
+
+static void
+__mmplayer_release_dump_list (GList *dump_list)
+{
+ if (dump_list)
+ {
+ GList *d_list = dump_list;
+ for ( ;d_list ; d_list = g_list_next(d_list))
+ {
+ mm_player_dump_t *dump_s = d_list->data;
+ if (dump_s->dump_pad)
+ {
+ if (dump_s->probe_handle_id)
+ {
+ gst_pad_remove_buffer_probe (dump_s->dump_pad, dump_s->probe_handle_id);
+ }
+
+ }
+ if (dump_s->dump_element_file)
+ {
+ fclose(dump_s->dump_element_file);
+ dump_s->dump_element_file = NULL;
+ }
+ MMPLAYER_FREEIF(dump_s);
+ }
+ g_list_free(dump_list);
+ dump_list = NULL;
+ }
+}
+
+static int
+__mmplayer_ignore_current_external_display_mode(mm_player_t* player)
+{
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ MMPLAYER_VIDEO_SINK_CHECK(player);
+
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst, "keep-external-fullscreen-prev", TRUE, NULL);
+ debug_log("set keep-external-fullscreen-prev property to TRUE");
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_enable_media_packet_video_stream(MMHandleType hplayer, bool enable)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ MMPLAYER_FENTER();
+
+ return_val_if_fail (player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+ return_val_if_fail (enable == TRUE || enable == FALSE, MM_ERROR_INVALID_ARGUMENT);
+
+ player->set_mode.media_packet_video_stream = enable;
+
+ MMPLAYER_FLEAVE();
+
+ return MM_ERROR_NONE;
+}
projects / framework / multimedia / libmm-player.git / blobdiff