--- /dev/null
+/*
+ * libmm-player
+ *
+ * Copyright (c) 2000 - 2011 Samsung Electronics Co., Ltd. All rights reserved.
+ *
+ * Contact: JongHyuk Choi <jhchoi.choi@samsung.com>, YeJin Cho <cho.yejin@samsung.com>, YoungHwan An <younghwan_.an@samsung.com>
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ *
+ */
+
+/*===========================================================================================
+| |
+| INCLUDE FILES |
+| |
+========================================================================================== */
+#include <glib.h>
+#include <gst/gst.h>
+#include <gst/app/gstappsrc.h>
+#include <gst/interfaces/xoverlay.h>
+#include <unistd.h>
+#include <string.h>
+#include <sys/time.h>
+#include <stdlib.h>
+
+#include <mm_error.h>
+#include <mm_attrs.h>
+#include <mm_attrs_private.h>
+#include <mm_debug.h>
+
+#include "mm_player_priv.h"
+#include "mm_player_ini.h"
+#include "mm_player_attrs.h"
+#include "mm_player_capture.h"
+
+/*===========================================================================================
+| |
+| LOCAL DEFINITIONS AND DECLARATIONS FOR MODULE |
+| |
+========================================================================================== */
+
+/*---------------------------------------------------------------------------
+| GLOBAL CONSTANT DEFINITIONS: |
+---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------
+| IMPORTED VARIABLE DECLARATIONS: |
+---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------
+| IMPORTED FUNCTION DECLARATIONS: |
+---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------
+| LOCAL #defines: |
+---------------------------------------------------------------------------*/
+#define TRICK_PLAY_MUTE_THRESHOLD_MAX 2.0
+#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_PLAYER_FADEOUT_TIME_DEFAULT 700000 // 700 msec
+
+#define MM_PLAYER_MPEG_VNAME "mpegversion"
+#define MM_PLAYER_DIVX_VNAME "divxversion"
+#define MM_PLAYER_WMV_VNAME "wmvversion"
+#define MM_PLAYER_WMA_VNAME "wmaversion"
+
+#define DEFAULT_PLAYBACK_RATE 1.0
+
+#define GST_QUEUE_DEFAULT_TIME 8
+#define GST_QUEUE_HLS_TIME 8
+
+/* 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 LAZY_PAUSE_TIMEOUT_MSEC 700
+
+/*---------------------------------------------------------------------------
+| LOCAL CONSTANT DEFINITIONS: |
+---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------
+| LOCAL DATA TYPE DEFINITIONS: |
+---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------
+| GLOBAL VARIABLE DEFINITIONS: |
+---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------
+| LOCAL VARIABLE 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, 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_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_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_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_rtp_dynamic_pad (GstElement *element, GstPad *pad, gpointer data);
+static gboolean __mmplayer_update_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 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);
+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_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 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 gboolean __mmplayer_configure_audio_callback(mm_player_t* player);
+static void __mmplayer_add_sink( mm_player_t* player, GstElement* sink);
+static void __mmplayer_del_sink( mm_player_t* player, GstElement* sink);
+static void __mmplayer_release_signal_connection(mm_player_t* player);
+static void __mmplayer_set_antishock( mm_player_t* player, gboolean disable_by_force);
+static gpointer __mmplayer_repeat_thread(gpointer data);
+int _mmplayer_get_track_count(MMHandleType hplayer, MMPlayerTrackType track_type, int *count);
+
+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);
+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, 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_handle_resource_error( mm_player_t* player, int code );
+static gint __gst_handle_stream_error( mm_player_t* player, GError* error, GstMessage * message );
+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_set_pcm_extraction(mm_player_t* player);
+static gboolean __mmplayer_can_extract_pcm( mm_player_t* player );
+
+/*fadeout */
+static void __mmplayer_do_sound_fadedown(mm_player_t* player, unsigned int time);
+static void __mmplayer_undo_sound_fadedown(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);
+
+/* 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_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_http_progressive_down(mm_player_t* player);
+
+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);
+
+
+/*===========================================================================================
+| |
+| FUNCTION DEFINITIONS |
+| |
+========================================================================================== */
+
+/* implementing player FSM */
+/* FIXIT : We need to handle state transition also at here since start api is no more sync */
+static int
+__mmplayer_check_state(mm_player_t* player, enum PlayerCommandState command)
+{
+ 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();
+
+ 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);
+
+ MMPLAYER_PRINT_STATE(player);
+
+ 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 ||
+ current_state == MM_PLAYER_STATE_PLAYING )
+ goto NO_OP;
+ }
+ break;
+
+ case MMPLAYER_COMMAND_DESTROY:
+ {
+ /* destroy can called anytime */
+
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_NONE;
+ }
+ break;
+
+ case MMPLAYER_COMMAND_REALIZE:
+ {
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_READY;
+
+ if ( pending_state != MM_PLAYER_STATE_NONE )
+ {
+ goto INVALID_STATE;
+ }
+ else
+ {
+ /* need ready state to realize */
+ if ( current_state == MM_PLAYER_STATE_READY )
+ goto NO_OP;
+
+ if ( current_state != MM_PLAYER_STATE_NULL )
+ goto INVALID_STATE;
+ }
+ }
+ break;
+
+ case MMPLAYER_COMMAND_UNREALIZE:
+ {
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_NULL;
+
+ if ( current_state == MM_PLAYER_STATE_NULL )
+ goto NO_OP;
+ }
+ break;
+
+ case MMPLAYER_COMMAND_START:
+ {
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_PLAYING;
+
+ if ( pending_state == MM_PLAYER_STATE_NONE )
+ {
+ if ( current_state == MM_PLAYER_STATE_PLAYING )
+ goto NO_OP;
+ else if ( current_state != MM_PLAYER_STATE_READY &&
+ current_state != MM_PLAYER_STATE_PAUSED )
+ goto INVALID_STATE;
+ }
+ else if ( pending_state == MM_PLAYER_STATE_PLAYING )
+ {
+ goto ALREADY_GOING;
+ }
+ else if ( pending_state == MM_PLAYER_STATE_PAUSED )
+ {
+ debug_log("player is going to paused state, just change the pending state as playing");
+ }
+ else
+ {
+ goto INVALID_STATE;
+ }
+ }
+ break;
+
+ case MMPLAYER_COMMAND_STOP:
+ {
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_READY;
+
+ 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 )
+ goto INVALID_STATE;
+ }
+ break;
+
+ case MMPLAYER_COMMAND_PAUSE:
+ {
+ if ( MMPLAYER_IS_LIVE_STREAMING( player ) )
+ goto NO_OP;
+
+ if (player->doing_seek)
+ goto NOT_COMPLETED_SEEK;
+
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_PAUSED;
+
+ if ( pending_state == MM_PLAYER_STATE_NONE )
+ {
+ if ( current_state == MM_PLAYER_STATE_PAUSED )
+ goto NO_OP;
+ else if ( current_state != MM_PLAYER_STATE_PLAYING && current_state != MM_PLAYER_STATE_READY ) // support loading state of broswer
+ goto INVALID_STATE;
+ }
+ else if ( pending_state == MM_PLAYER_STATE_PAUSED )
+ {
+ goto ALREADY_GOING;
+ }
+ else if ( pending_state == MM_PLAYER_STATE_PLAYING )
+ {
+ if ( current_state == MM_PLAYER_STATE_PAUSED ) {
+ debug_log("player is PAUSED going to PLAYING, just change the pending state as PAUSED");
+ } else {
+ goto INVALID_STATE;
+ }
+ }
+ }
+ break;
+
+ case MMPLAYER_COMMAND_RESUME:
+ {
+ if ( MMPLAYER_IS_LIVE_STREAMING(player) )
+ goto NO_OP;
+
+ if (player->doing_seek)
+ goto NOT_COMPLETED_SEEK;
+
+ MMPLAYER_TARGET_STATE(player) = MM_PLAYER_STATE_PLAYING;
+
+ if ( pending_state == MM_PLAYER_STATE_NONE )
+ {
+ if ( current_state == MM_PLAYER_STATE_PLAYING )
+ goto NO_OP;
+ else if ( current_state != MM_PLAYER_STATE_PAUSED )
+ goto INVALID_STATE;
+ }
+ else if ( pending_state == MM_PLAYER_STATE_PLAYING )
+ {
+ goto ALREADY_GOING;
+ }
+ else if ( pending_state == MM_PLAYER_STATE_PAUSED )
+ {
+ debug_log("player is going to paused state, just change the pending state as playing");
+ }
+ else
+ {
+ goto INVALID_STATE;
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+ player->cmd = command;
+
+ debug_fleave();
+ 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:
+ debug_warning("player is in the desired state(%s). doing noting", MMPLAYER_STATE_GET_NAME(current_state));
+ return MM_ERROR_PLAYER_NO_OP;
+
+ALREADY_GOING:
+ debug_warning("player is already going to %s, doing nothing", MMPLAYER_STATE_GET_NAME(pending_state));
+ return MM_ERROR_PLAYER_NO_OP;
+}
+
+int
+__mmplayer_gst_set_state (mm_player_t* player, GstElement * element, GstState state, gboolean async, gint timeout) // @
+{
+ GstState element_state = GST_STATE_VOID_PENDING;
+ GstState element_pending_state = GST_STATE_VOID_PENDING;
+ GstStateChangeReturn ret = GST_STATE_CHANGE_FAILURE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( element, MM_ERROR_INVALID_ARGUMENT );
+
+ debug_log("setting [%s] element state to : %d\n", GST_ELEMENT_NAME(element), 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);
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* return here so state transition to be done in async mode */
+ if ( async )
+ {
+ debug_log("async state transition. not waiting for state complete.\n");
+ return MM_ERROR_NONE;
+ }
+
+ /* wait for state transition */
+ ret = gst_element_get_state( element, &element_state, &element_pending_state, timeout * GST_SECOND );
+
+ if ( ret == GST_STATE_CHANGE_FAILURE || ( state != element_state ) )
+ {
+ 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),
+ gst_element_state_get_name(element_pending_state) );
+
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ debug_log("[%s] element state has changed to %s \n",
+ GST_ELEMENT_NAME(element),
+ gst_element_state_get_name(element_state));
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+static void
+__mmplayer_videostream_cb(GstElement *element, void *stream,
+int width, int height, gpointer data) // @
+{
+ mm_player_t* player = (mm_player_t*)data;
+ int length = 0;
+
+ return_if_fail ( player );
+
+ debug_fenter();
+
+ 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);
+ }
+
+ debug_fleave();
+}
+
+static void
+__mmplayer_videoframe_render_error_cb(GstElement *element, void *error_id, gpointer data)
+{
+ mm_player_t* player = (mm_player_t*)data;
+
+ return_if_fail ( player );
+
+ debug_fenter();
+
+ if (player->video_frame_render_error_cb )
+ {
+ if (player->attrs)
+ {
+ 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
+ {
+ debug_error("could not get surface type");
+ }
+ }
+ else
+ {
+ debug_warning("video_frame_render_error_cb was not set");
+ }
+
+ debug_fleave();
+}
+
+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 );
+
+ if ( ! player->need_update_content_attrs )
+ {
+ debug_log("content attributes are already updated");
+ return TRUE;
+ }
+
+ /* get content attribute first */
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute");
+ return FALSE;
+ }
+
+ /* 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;
+ }
+
+ 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;
+ }
+
+ break;
+ }
+
+ player->duration = dur_nsec;
+ debug_log("duration : %lld msec", GST_TIME_AS_MSECONDS(dur_nsec));
+
+ /* try to get streaming service type */
+ __mmplayer_update_stream_service_type( player );
+
+ /* check duration is OK */
+ if ( dur_nsec == 0 && !MMPLAYER_IS_LIVE_STREAMING( player ) )
+ {
+ /* FIXIT : find another way to get duration here. */
+ debug_error("finally it's failed to get duration from pipeline. progressbar will not work correctely!");
+ }
+ else
+ {
+ player->need_update_content_dur = FALSE;
+ }
+
+ /*update duration */
+ mm_attrs_set_int_by_name(attrs, "content_duration", GST_TIME_AS_MSECONDS(dur_nsec));
+ }
+ else
+ {
+ debug_log("not ready to get duration or already updated");
+ }
+
+ /* 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;
+
+ pad = gst_element_get_static_pad(
+ player->pipeline->audiobin[MMPLAYER_A_CONV].gst, "sink" );
+
+ if ( pad )
+ {
+ caps_a = gst_pad_get_negotiated_caps( pad );
+
+ if ( caps_a )
+ {
+ p = gst_caps_get_structure (caps_a, 0);
+
+ mm_attrs_get_int_by_name(attrs, "content_audio_samplerate", &samplerate);
+ if ( ! samplerate ) // check if update already or not
+ {
+ gst_structure_get_int (p, "rate", &samplerate);
+ mm_attrs_set_int_by_name(attrs, "content_audio_samplerate", samplerate);
+
+ gst_structure_get_int (p, "channels", &channels);
+ mm_attrs_set_int_by_name(attrs, "content_audio_channels", channels);
+
+ debug_log("samplerate : %d channels : %d", samplerate, channels);
+ }
+ gst_caps_unref( caps_a );
+ caps_a = NULL;
+ }
+ else
+ {
+ debug_warning("not ready to get audio caps");
+ }
+
+ gst_object_unref( pad );
+ }
+ else
+ {
+ debug_warning("failed to get pad from audiosink");
+ }
+ }
+
+ /* update width, height, framerate */
+ if ( player->pipeline->videobin &&
+ player->pipeline->videobin[MMPLAYER_V_SINK].gst )
+ {
+ GstCaps *caps_v = NULL;
+ GstPad* pad = NULL;
+ gint tmpNu, tmpDe;
+ gint width, height;
+
+ 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);
+
+ gst_structure_get_int (p, "height", &height);
+ mm_attrs_set_int_by_name(attrs, "content_video_height", height);
+
+ gst_structure_get_fraction (p, "framerate", &tmpNu, &tmpDe);
+
+ debug_log("width : %d height : %d", width, height );
+
+ gst_caps_unref( caps_v );
+ caps_v = NULL;
+
+ 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");
+ }
+ gst_object_unref( pad );
+ pad = NULL;
+ }
+ else
+ {
+ debug_warning("failed to get pad from videosink");
+ }
+ }
+
+ if (player->duration)
+ {
+ guint64 data_size = 0;
+
+ if (!MMPLAYER_IS_STREAMING(player) && (player->can_support_codec & FOUND_PLUGIN_VIDEO))
+ {
+ mm_attrs_get_string_by_name(attrs, "profile_uri", &path);
+
+ if (stat(path, &sb) == 0)
+ {
+ data_size = (guint64)sb.st_size;
+ }
+ }
+ else if (MMPLAYER_IS_HTTP_STREAMING(player))
+ {
+ data_size = player->http_content_size;
+ }
+
+ if (data_size)
+ {
+ guint64 bitrate = 0;
+ guint64 msec_dur = 0;
+
+ 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);
+ }
+ }
+
+
+ /* validate all */
+ if ( mmf_attrs_commit ( attrs ) )
+ {
+ debug_error("failed to update attributes\n");
+ return FALSE;
+ }
+
+ player->need_update_content_attrs = FALSE;
+
+ return TRUE;
+}
+
+gboolean __mmplayer_update_stream_service_type( mm_player_t* player )
+{
+ MMHandleType attrs = 0;
+ gint streaming_type = STREAMING_SERVICE_NONE;
+
+ debug_fenter();
+
+ 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 FALSE;
+
+ if (MMPLAYER_IS_RTSP_STREAMING(player))
+ {
+ /* 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))
+ {
+ if ( player->duration == 0)
+ streaming_type = STREAMING_SERVICE_LIVE;
+ else
+ streaming_type = STREAMING_SERVICE_VOD;
+ }
+
+ player->streaming_type = streaming_type;
+
+ if ( player->streaming_type == STREAMING_SERVICE_LIVE)
+ {
+ debug_log("It's live streaming. pause/resume/seek are not working.\n");
+ }
+ else if (player->streaming_type == STREAMING_SERVICE_LIVE)
+ {
+ debug_log("It's vod streaming. pause/resume/seek are working.\n");
+ }
+ else
+ {
+ debug_warning("fail to determine streaming type. pause/resume/seek may not working properly if stream is live stream\n");
+ }
+
+ /* get profile attribute */
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute\n");
+ return FALSE;
+ }
+
+ 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;
+ }
+
+ debug_fleave();
+
+ return TRUE;
+}
+
+
+/* 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) // @
+{
+ MMMessageParamType msg = {0, };
+ int asm_result = MM_ERROR_NONE;
+
+ debug_fenter();
+ 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 TRUE;
+ }
+
+ /* post message to application */
+ if (MMPLAYER_TARGET_STATE(player) == state)
+ {
+ /* fill the message with state of player */
+ msg.state.previous = MMPLAYER_CURRENT_STATE(player);
+ msg.state.current = state;
+
+ /* 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 );
+ }
+
+ debug_log ("player reach the target state, then do something in each state(%s).\n",
+ MMPLAYER_STATE_GET_NAME(MMPLAYER_TARGET_STATE(player)));
+ }
+ else
+ {
+ debug_log ("intermediate state, do nothing.\n");
+ MMPLAYER_PRINT_STATE(player);
+
+ return TRUE;
+ }
+
+ /* update player states */
+ MMPLAYER_PREV_STATE(player) = MMPLAYER_CURRENT_STATE(player);
+ MMPLAYER_CURRENT_STATE(player) = state;
+ if ( MMPLAYER_CURRENT_STATE(player) == MMPLAYER_PENDING_STATE(player) )
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NONE;
+
+ /* print state */
+ MMPLAYER_PRINT_STATE(player);
+
+ 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.
+ */
+ if ( ! player->sent_bos )
+ {
+ player->need_update_content_attrs = TRUE;
+ player->need_update_content_dur = TRUE;
+ _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:
+ {
+ /* non-managed prepare case, should be updated */
+ if ( ! player->need_update_content_dur)
+ {
+ player->need_update_content_dur = TRUE;
+ _mmplayer_update_content_attrs ( player );
+ }
+ if (MMPLAYER_IS_STREAMING(player))
+ {
+ /* force setting value to TRUE for streaming */
+ 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");
+ }
+
+ MMTA_ACUM_ITEM_END("[KPI] start media player service", FALSE);
+ MMTA_ACUM_ITEM_END("[KPI] media player service create->playing", FALSE);
+
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_BEGIN_OF_STREAM, NULL );
+ player->sent_bos = TRUE;
+ }
+ }
+ break;
+
+ case MM_PLAYER_STATE_NONE:
+ default:
+ debug_warning("invalid target state, there is nothing to do.\n");
+ break;
+ }
+
+ debug_fleave();
+
+ return TRUE;
+}
+
+
+gboolean
+__mmplayer_post_message(mm_player_t* player, enum MMMessageType msgtype, MMMessageParamType* param) // @
+{
+ return_val_if_fail( player, FALSE );
+
+ debug_fenter();
+
+ if ( !player->msg_cb )
+ {
+ debug_warning("no msg callback. can't post\n");
+ return FALSE;
+ }
+
+ //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) // @
+{
+ 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));
+
+ debug_fleave();
+
+ return state;
+}
+
+static void
+__gst_set_async_state_change(mm_player_t* player, gboolean async)
+{
+ //debug_fenter();
+ return_if_fail( player && player->pipeline && player->pipeline->mainbin );
+
+ /* need only when we are using decodebin */
+ if ( ! PLAYER_INI()->use_decodebin )
+ return;
+
+ /* audio sink */
+ if ( player->pipeline->audiobin &&
+ player->pipeline->audiobin[MMPLAYER_A_SINK].gst )
+ {
+ debug_log("audiosink async : %d\n", async);
+ g_object_set (G_OBJECT (player->pipeline->audiobin[MMPLAYER_A_SINK].gst), "async", async, NULL);
+ }
+
+ /* video sink */
+ if ( player->pipeline->videobin &&
+ player->pipeline->videobin[MMPLAYER_V_SINK].gst )
+ {
+ debug_log("videosink async : %d\n", async);
+ g_object_set (G_OBJECT (player->pipeline->videobin[MMPLAYER_V_SINK].gst), "async", async, NULL);
+ }
+
+ /* decodebin if enabled */
+ if ( PLAYER_INI()->use_decodebin )
+ {
+ debug_log("decodebin async : %d\n", async);
+ g_object_set (G_OBJECT (player->pipeline->mainbin[MMPLAYER_M_AUTOPLUG].gst), "async-handling", async, NULL);
+ }
+
+ //debug_fleave();
+}
+
+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;
+
+ return_val_if_fail ( player, 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 );
+
+ if ( player->repeat_thread_exit )
+ {
+ debug_log("exiting repeat thread\n");
+ break;
+ }
+
+ if ( !player->cmd_lock )
+ {
+ debug_log("can't get cmd lock\n");
+ return NULL;
+ }
+
+ /* lock */
+ g_mutex_lock(player->cmd_lock);
+
+ 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;
+ }
+
+ 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 );
+ }
+
+ 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);
+
+ /* initialize */
+ player->sent_bos = FALSE;
+ }
+
+ if ( ! ret_value )
+ {
+ debug_error("failed to set position to zero for rewind\n");
+ continue;
+ }
+
+ /* decrease play count */
+ if ( count > 1 )
+ {
+ /* 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");
+ }
+ }
+
+ /* unlock */
+ g_mutex_unlock(player->cmd_lock);
+ }
+
+ 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 ( MMPLAYER_IS_RTSP_STREAMING(player) )
+ return;
+
+ 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));
+
+ /* 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;
+
+ 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;
+
+ 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;
+
+ 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;
+
+ case MM_PLAYER_STATE_PLAYING:
+ {
+ debug_log("player is already going to playing state, there is nothing to do.\n");
+ }
+ 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;
+ }
+ }
+ break;
+
+ 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)
+ __gst_pause ( player, TRUE );
+ }
+ break;
+
+ 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_NULL:
+ case MM_PLAYER_STATE_READY:
+ default :
+ {
+ debug_warning("invalid pending state [%s].\n", MMPLAYER_STATE_GET_NAME(pending_state) );
+ }
+ break;
+ }
+ }
+}
+
+static gboolean
+__mmplayer_gst_callback(GstBus *bus, GstMessage *msg, gpointer data) // @
+{
+ mm_player_t* player = (mm_player_t*) data;
+ gboolean ret = TRUE;
+ static gboolean async_done = FALSE;
+
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( msg && GST_IS_MESSAGE(msg), FALSE );
+
+ switch ( GST_MESSAGE_TYPE( msg ) )
+ {
+ case GST_MESSAGE_UNKNOWN:
+ debug_warning("unknown message received\n");
+ break;
+
+ case GST_MESSAGE_EOS:
+ {
+ MMHandleType attrs = 0;
+ gint count = 0;
+
+ debug_log("GST_MESSAGE_EOS received\n");
+
+ /* 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");
+
+ 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;
+
+ }
+
+ /* rewind if repeat count is greater then zero */
+ /* get play count */
+ attrs = MMPLAYER_GET_ATTRS(player);
+
+ if ( attrs )
+ {
+ gboolean smooth_repeat = FALSE;
+
+ mm_attrs_get_int_by_name(attrs, "profile_play_count", &count);
+ mm_attrs_get_int_by_name(attrs, "profile_smooth_repeat", &smooth_repeat);
+
+ debug_log("remaining play count: %d, playback rate: %f\n", count, player->playback_rate);
+
+ 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");
+
+ 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
+ {
+
+ 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 );
+ }
+
+ ret_value = __gst_set_position( player, MM_PLAYER_POS_FORMAT_TIME, 0, TRUE);
+
+ /* initialize */
+ player->sent_bos = FALSE;
+ }
+
+ 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--;
+
+ mm_attrs_set_int_by_name(attrs, "profile_play_count", count);
+
+ if ( mmf_attrs_commit ( attrs ) )
+ debug_error("failed to commit attrs\n");
+ }
+ }
+
+ break;
+ }
+ }
+ }
+
+ 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 );
+
+ msg_src_element = GST_ELEMENT_NAME( GST_ELEMENT_CAST( msg->src ) );
+ 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 );
+
+ /* 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 );
+
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+
+ }
+
+ if (MMPLAYER_IS_HTTP_PD(player))
+ {
+ _mmplayer_unrealize_pd_downloader ((MMHandleType)player);
+ }
+
+ 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);
+
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_WARNING, NULL );
+
+ MMPLAYER_FREEIF( debug );
+ g_error_free( error );
+ }
+ break;
+
+ case GST_MESSAGE_INFO: debug_log("GST_MESSAGE_STATE_DIRTY\n"); break;
+
+ case GST_MESSAGE_TAG:
+ {
+ 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;
+
+ case GST_MESSAGE_BUFFERING:
+ {
+ 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) );
+ if (update_buffering_percent)
+ {
+ msg_param.connection.buffering = player->streamer->buffering_percent;
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_BUFFERING, &msg_param );
+ }
+ }
+ break;
+
+ case GST_MESSAGE_STATE_CHANGED:
+ {
+ MMPlayerGstElement *mainbin;
+ const GValue *voldstate, *vnewstate, *vpending;
+ GstState oldstate, newstate, pending;
+
+ if ( ! ( player->pipeline && player->pipeline->mainbin ) )
+ {
+ debug_error("player pipeline handle is null");
+
+ break;
+ }
+
+ mainbin = player->pipeline->mainbin;
+
+ /* 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");
+
+ oldstate = (GstState)voldstate->data[0].v_int;
+ newstate = (GstState)vnewstate->data[0].v_int;
+ pending = (GstState)vpending->data[0].v_int;
+
+ if (oldstate == newstate)
+ break;
+
+ 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 ) );
+
+ /* we only handle messages from pipeline */
+ if( msg->src != (GstObject *)mainbin[MMPLAYER_M_PIPE].gst )
+ break;
+
+ switch(newstate)
+ {
+ case GST_STATE_VOID_PENDING:
+ break;
+
+ case GST_STATE_NULL:
+ break;
+
+ case GST_STATE_READY:
+ break;
+
+ case GST_STATE_PAUSED:
+ {
+ gboolean prepare_async = FALSE;
+
+ player->need_update_content_dur = TRUE;
+
+ if ( ! player->audio_cb_probe_id && player->is_sound_extraction )
+ __mmplayer_configure_audio_callback(player);
+
+ 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);
+ }
+
+ if ( MMPLAYER_IS_STREAMING(player) || prepare_async )
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PAUSED );
+
+ if (player->streamer)
+ __mm_player_streaming_set_content_bitrate(player->streamer, player->total_maximum_bitrate, player->total_bitrate);
+ }
+ }
+ break;
+
+ case GST_STATE_PLAYING:
+ {
+ if (player->doing_seek && async_done)
+ {
+ player->doing_seek = FALSE;
+ async_done = FALSE;
+ MMPLAYER_POST_MSG ( player, MM_MESSAGE_SEEK_COMPLETED, NULL );
+ }
+
+ 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.
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PLAYING);
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+ }
+ 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"));
+
+ if (PLAYER_INI()->provide_clock)
+ {
+ debug_log ("Provide clock is TRUE, do pause->resume\n");
+ __gst_pause(player, FALSE);
+ __gst_resume(player, FALSE);
+ }
+ }
+ break;
+
+ case GST_MESSAGE_NEW_CLOCK:
+ {
+ GstClock *clock = NULL;
+ gst_message_parse_new_clock (msg, &clock);
+ debug_log("GST_MESSAGE_NEW_CLOCK : %s\n", (clock ? GST_OBJECT_NAME (clock) : "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;
+
+ case GST_MESSAGE_ELEMENT:
+ {
+ debug_log("GST_MESSAGE_ELEMENT\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_DURATION:
+ {
+ debug_log("GST_MESSAGE_DURATION\n");
+
+ if (MMPLAYER_IS_STREAMING(player))
+ {
+ GstFormat format;
+ gint64 bytes = 0;
+
+ gst_message_parse_duration (msg, &format, &bytes);
+ if (format == GST_FORMAT_BYTES)
+ {
+ debug_log("data total size of http content: %lld", bytes);
+ player->http_content_size = bytes;
+ }
+ }
+
+ player->need_update_content_attrs = TRUE;
+ player->need_update_content_dur = TRUE;
+ _mmplayer_update_content_attrs(player);
+ }
+ 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:
+ {
+ debug_log("GST_MESSAGE_ASYNC_DONE : %s\n", gst_element_get_name(GST_MESSAGE_SRC(msg)));
+
+ 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;
+
+ 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;
+ }
+
+ /* 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;
+}
+
+static gboolean
+__mmplayer_gst_extract_tag_from_msg(mm_player_t* player, GstMessage* msg) // @
+{
+
+/* 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)); \
+ } \
+}
+
+#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;\
+ }\
+}
+
+#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);\
+ }\
+}
+
+#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;\
+ }\
+}
+
+#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;\
+ }\
+}
+
+ /* function start */
+ GstTagList* tag_list = NULL;
+
+ MMHandleType attrs = 0;
+
+ char *string = NULL;
+ guint v_uint = 0;
+ GDate *date = NULL;
+ /* album cover */
+ GstBuffer *buffer = NULL;
+ gint index = 0;
+ const GValue *value;
+
+ /* 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 );
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+
+ return_val_if_fail( attrs, FALSE );
+
+ /* get tag list from gst message */
+ gst_message_parse_tag(msg, &tag_list);
+
+ /* 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 ( mmf_attrs_commit ( attrs ) )
+ debug_error("failed to commit.\n");
+
+ gst_tag_list_free(tag_list);
+
+ return TRUE;
+}
+
+static void
+__mmplayer_gst_rtp_no_more_pads (GstElement *element, gpointer data) // @
+{
+ mm_player_t* player = (mm_player_t*) data;
+
+ debug_fenter();
+
+ /* 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
+
+ * [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 ( ! __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;
+ }
+ }
+
+ /* create dot before error-return. for debugging */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-no-more-pad" );
+
+ /* 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 );
+
+ player->no_more_pad = TRUE;
+
+ debug_fleave();
+}
+
+static gboolean
+__mmplayer_gst_remove_fakesink(mm_player_t* player, MMPlayerGstElement* fakesink) // @
+{
+ GstElement* parent = NULL;
+
+ return_val_if_fail(player && player->pipeline && fakesink, FALSE);
+
+ /* lock */
+ g_mutex_lock( player->fsink_lock );
+
+ if ( ! fakesink->gst )
+ {
+ goto ERROR;
+ }
+
+ /* get parent of fakesink */
+ parent = (GstElement*)gst_object_get_parent( (GstObject*)fakesink->gst );
+ if ( ! parent )
+ {
+ debug_log("fakesink already removed\n");
+ goto ERROR;
+ }
+
+ gst_element_set_locked_state( fakesink->gst, TRUE );
+
+ /* 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");
+
+ /* FIXIT : should I return here? or try to proceed to next? */
+ /* return FALSE; */
+ }
+
+ /* remove fakesink from it's parent */
+ if ( ! gst_bin_remove( GST_BIN( parent ), fakesink->gst ) )
+ {
+ debug_error("failed to remove fakesink\n");
+
+ gst_object_unref( parent );
+
+ goto ERROR;
+ }
+
+ 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;
+
+ debug_log("state-holder removed\n");
+
+ gst_element_set_locked_state( fakesink->gst, FALSE );
+
+ g_mutex_unlock( player->fsink_lock );
+ return TRUE;
+
+ERROR:
+ if ( fakesink->gst )
+ {
+ gst_element_set_locked_state( fakesink->gst, FALSE );
+ }
+
+ g_mutex_unlock( player->fsink_lock );
+ return FALSE;
+}
+
+
+static void
+__mmplayer_gst_rtp_dynamic_pad (GstElement *element, GstPad *pad, gpointer data) // @
+{
+ GstPad *sinkpad = NULL;
+ GstCaps* caps = NULL;
+ GstElement* new_element = NULL;
+
+ mm_player_t* player = (mm_player_t*) data;
+
+ debug_fenter();
+
+ return_if_fail( element && pad );
+ return_if_fail( player &&
+ player->pipeline &&
+ player->pipeline->mainbin );
+
+
+ /* 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);
+
+ /* perform autoplugging if dump is disabled */
+ if ( PLAYER_INI()->rtsp_do_typefinding )
+ {
+ /* create typefind */
+ new_element = gst_element_factory_make( "typefind", NULL );
+ if ( ! new_element )
+ {
+ debug_error("failed to create typefind\n");
+ goto ERROR;
+ }
+
+ MMPLAYER_SIGNAL_CONNECT( player,
+ G_OBJECT(new_element),
+ "have-type",
+ G_CALLBACK(__mmplayer_typefind_have_type),
+ (gpointer)player);
+
+ /* 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");
+
+ caps = gst_pad_get_caps( pad );
+
+ MMPLAYER_CHECK_NULL( caps );
+
+ /* clear previous result*/
+ player->have_dynamic_pad = FALSE;
+
+ if ( ! __mmplayer_try_to_plug( player, pad, caps ) )
+ {
+ debug_error("failed to autoplug for caps : %s\n", gst_caps_to_string( caps ) );
+ goto ERROR;
+ }
+
+ /* 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;
+ }
+
+ gst_caps_unref( caps );
+ caps = NULL;
+ }
+
+ /* 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_error("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_error("failed to get sinkpad from autoplug element\n");
+ goto ERROR;
+ }
+
+ /* link it */
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
+ {
+ debug_error("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 );
+ }
+
+ debug_fleave();
+
+ return;
+
+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));
+
+ if ( sinkpad )
+ gst_object_unref(GST_OBJECT(sinkpad));
+
+ if ( caps )
+ gst_object_unref(GST_OBJECT(caps));
+
+ /* 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_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;
+
+ /* check handles */
+ player = (mm_player_t*) data;
+
+ return_if_fail( decodebin && pad );
+ return_if_fail(player && player->pipeline && player->pipeline->mainbin);
+
+ pipeline = player->pipeline->mainbin[MMPLAYER_M_PIPE].gst;
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute\n");
+ goto ERROR;
+ }
+
+ /* get mimetype from caps */
+ caps = gst_pad_get_caps( pad );
+ if ( !caps )
+ {
+ debug_error("cannot get caps from pad.\n");
+ goto ERROR;
+ }
+
+ str = gst_caps_get_structure( caps, 0 );
+ if ( ! str )
+ {
+ 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;
+ }
+
+ debug_log("detected mimetype : %s\n", name);
+
+ 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;
+ }
+ )
+
+ sinkbin = player->pipeline->audiobin[MMPLAYER_A_BIN].gst;
+ debug_log("creating audiosink bin success\n");
+ }
+ else
+ {
+ sinkbin = player->pipeline->audiobin[MMPLAYER_A_BIN].gst;
+ debug_log("re-using audiobin\n");
+ }
+
+ /* FIXIT : track number shouldn't be hardcoded */
+ mm_attrs_set_int_by_name(attrs, "content_audio_track_num", 1);
+
+ player->audiosink_linked = 1;
+ debug_msg("player->audsink_linked set to 1\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 (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);
+
+ if (surface_type == MM_DISPLAY_SURFACE_NULL)
+ {
+ debug_log("not make videobin because it dose not want\n");
+ goto ERROR;
+ }
+
+ __ta__("__mmplayer_gst_create_video_pipeline",
+ 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;
+ }
+ )
+
+ 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");
+
+ 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, "text"))
+ {
+ if (player->pipeline->textbin == NULL)
+ {
+ __ta__("__mmplayer_gst_create_text_pipeline",
+ if (MM_ERROR_NONE != __mmplayer_gst_create_text_pipeline(player))
+ {
+ debug_error("failed to create textbin. continuing without text\n");
+ goto ERROR;
+ }
+ )
+
+ 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);
+
+ player->textsink_linked = 1;
+ debug_msg("player->textsink_linked set to 1\n");
+
+ sinkpad = gst_element_get_static_pad( GST_ELEMENT(sinkbin), "text_sink" );
+ if ( !sinkpad )
+ {
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
+ }
+ }
+ else
+ {
+ debug_warning("unknown type of elementary stream! ignoring it...\n");
+ goto ERROR;
+ }
+
+ 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;
+ }
+
+ /* add */
+ if ( FALSE == gst_bin_add( GST_BIN(pipeline), sinkbin ) )
+ {
+ debug_error("failed to add sinkbin to pipeline\n");
+ goto ERROR;
+ }
+
+ /* link */
+ if ( GST_PAD_LINK_OK != GST_PAD_LINK(pad, sinkpad) )
+ {
+ debug_error("failed to get pad from sinkbin\n");
+ goto ERROR;
+ }
+
+ /* 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;
+ }
+
+ gst_object_unref( sinkpad );
+ sinkpad = NULL;
+ }
+
+ /* update track number attributes */
+ if ( mmf_attrs_commit ( attrs ) )
+ debug_error("failed to commit attrs\n");
+
+ debug_log("linking sink bin success\n");
+
+ /* 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--;
+
+ debug_log("stream count dec : %d (num of dynamic pad)\n", player->num_dynamic_pad);
+
+ if ( ( player->no_more_pad ) && ( player->num_dynamic_pad == 0 ) )
+ {
+ __mmplayer_pipeline_complete( NULL, player );
+ }
+
+ERROR:
+ if ( caps )
+ gst_caps_unref( caps );
+
+ if ( sinkpad )
+ gst_object_unref(GST_OBJECT(sinkpad));
+
+ return;
+}
+
+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;
+ char *element_name = NULL;
+ int rotation_using_type = -1;
+ #define ROTATION_USING_X 0
+ #define ROTATION_USING_FLIP 1
+
+ return_val_if_fail(player, FALSE);
+ return_val_if_fail(value, FALSE);
+ return_val_if_fail(rotation_angle >= 0, FALSE);
+
+ if (rotation_angle >= 360)
+ {
+ dest_angle = rotation_angle - 360;
+ }
+
+ /* chech if supported or not */
+ if ( dest_angle % 90 )
+ {
+ debug_log("not supported rotation angle = %d", rotation_angle);
+ return FALSE;
+ }
+
+ if (player->use_video_stream)
+ {
+ rotation_using_type = ROTATION_USING_FLIP;
+ }
+ else
+ {
+ int surface_type = 0;
+ mm_attrs_get_int_by_name(player->attrs, "display_surface_type", &surface_type);
+ debug_log("check display surface type for rotation: %d", surface_type);
+
+ switch (surface_type)
+ {
+ case MM_DISPLAY_SURFACE_X:
+ rotation_using_type = ROTATION_USING_X;
+ break;
+ case MM_DISPLAY_SURFACE_EVAS:
+ default:
+ rotation_using_type = ROTATION_USING_FLIP;
+ break;
+ }
+ }
+
+ debug_log("using %d type for rotation", rotation_using_type);
+
+ /* get property value for setting */
+ switch(rotation_using_type)
+ {
+ case ROTATION_USING_X: // xvimagesink
+ {
+ 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;
+ }
+ }
+ break;
+ case ROTATION_USING_FLIP: // videoflip
+ {
+ switch (dest_angle)
+ {
+
+ 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;
+ }
+
+ debug_log("setting rotation property value : %d", pro_value);
+
+ *value = pro_value;
+
+ return TRUE;
+}
+
+int
+_mmplayer_update_video_param(mm_player_t* player) // @
+{
+ 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();
+
+ /* 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 )
+ {
+ debug_error("cannot get content attribute");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* update user roation */
+ mm_attrs_get_int_by_name(attrs, "display_rotation", &user_angle_type);
+
+ /* get angle with user type */
+ switch(user_angle_type)
+ {
+ 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;
+ }
+
+ /* get original orientation */
+ 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");
+ }
+ else
+ {
+ debug_log("origianl video orientation = %d", org_angle);
+ }
+ }
+
+ debug_log("check user angle: %d, org angle: %d", user_angle, org_angle);
+
+ /* get rotation value to set */
+ __mmplayer_get_property_value_for_rotation(player, org_angle+user_angle, &rotation_value);
+
+ /* check video stream callback is used */
+ if( player->use_video_stream )
+ {
+ debug_log("using video stream callback with memsink. player handle : [%p]", player);
+
+ /* apply roate */
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_FLIP].gst, "method", rotation_value, NULL);
+
+ return 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);
+
+ /* configuring display */
+ switch ( surface_type )
+ {
+ case MM_DISPLAY_SURFACE_X:
+ {
+ /* ximagesink or xvimagesink */
+ void *xid = NULL;
+ int zoom = 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 x surface */
+ 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.");
+ }
+
+ /* 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_int_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_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);
+
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].gst,
+ "force-aspect-ratio", force_aspect_ratio,
+ "zoom", zoom,
+ "rotate", rotation_value,
+ "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 );
+
+ debug_log("set video param : zoom %d", zoom);
+ debug_log("set video param : rotate %d", rotation_value);
+ 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);
+ }
+ }
+ break;
+ case MM_DISPLAY_SURFACE_EVAS:
+ {
+ void *object = NULL;
+ int scaling = 0;
+ gboolean visible = TRUE;
+
+ /* 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);
+ if (object)
+ {
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].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;
+ }
+
+ /* 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;
+
+ 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);
+
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_SINK].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 );
+
+ 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);
+ }
+ g_object_set(player->pipeline->videobin[MMPLAYER_V_FLIP].gst, "method", rotation_value, NULL);
+ }
+ 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;
+
+ /* if xvimagesink */
+ if (strcmp(PLAYER_INI()->videosink_element_x,"xvimagesink"))
+ {
+ debug_error("videosink is not xvimagesink");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* 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);
+
+ 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;
+ }
+ debug_log("set video param : method %d", display_method);
+ debug_log("set video param : visible %d", 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,
+ "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();
+
+ return MM_ERROR_NONE;
+}
+
+static int
+__mmplayer_gst_element_link_bucket(GList* element_bucket) // @
+{
+ GList* bucket = element_bucket;
+ MMPlayerGstElement* element = NULL;
+ MMPlayerGstElement* prv_element = NULL;
+ gint successful_link_count = 0;
+
+ debug_fenter();
+
+ return_val_if_fail(element_bucket, -1);
+
+ prv_element = (MMPlayerGstElement*)bucket->data;
+ bucket = bucket->next;
+
+ for ( ; bucket; bucket = bucket->next )
+ {
+ 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_fleave();
+
+ return successful_link_count;
+}
+
+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 ++;
+ }
+ }
+
+ debug_fleave();
+
+ return successful_add_count;
+}
+
+
+
+/**
+ * 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_critical("failed to create %s \n", x_factory);\
+ goto ERROR;\
+}\
+
+#define MMPLAYER_CREATE_ELEMENT_ADD_BIN(x_bin, x_id, x_factory, x_name, y_bin) \
+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_critical("failed to create %s \n", x_factory);\
+ goto ERROR;\
+}\
+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;\
+}\
+
+/* 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 )\
+ {\
+ debug_critical("failed to create %s \n", x_factory);\
+ goto ERROR;\
+ }\
+ if ( x_add_bucket )\
+ element_bucket = g_list_append(element_bucket, &x_bin[x_id]);\
+} while(0);
+
+
+/**
+ * AUDIO PIPELINE
+ * - Local playback : audioconvert !volume ! capsfilter ! audioeq ! 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();
+
+ return_val_if_fail( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* alloc handles */
+ 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;
+ }
+
+ 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);
+
+ /* converter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV, "audioconvert", "audioconverter", TRUE);
+
+ if ( ! player->is_sound_extraction )
+ {
+ GstCaps* caps = NULL;
+ guint channels = 0;
+
+ /* 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);
+
+ if (player->sound.mute)
+ {
+ debug_log("mute enabled\n");
+ g_object_set(G_OBJECT (audiobin[MMPLAYER_A_VOL].gst), "mute", player->sound.mute, NULL);
+ }
+
+ /*capsfilter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_DEFAULT, "capsfilter", "audiocapsfilter", TRUE);
+
+ 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 );
+
+ /* chech if multi-chennels */
+ if (player->pipeline->mainbin && player->pipeline->mainbin[MMPLAYER_M_DEMUX].gst)
+ {
+ GstPad *srcpad = NULL;
+ GstCaps *caps = NULL;
+
+ 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);
+ }
+ }
+
+ /* audio effect element. if audio effect is enabled */
+ if ( channels <= 2 && (PLAYER_INI()->use_audio_effect_preset || PLAYER_INI()->use_audio_effect_custom) )
+ {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER, PLAYER_INI()->name_of_audio_effect, "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);
+ }
+
+ if ( g_strrstr(PLAYER_INI()->name_of_audiosink, "avsysaudiosink") )
+ {
+ gint volume_type = 0;
+ gint audio_route = 0;
+ gint sound_priority = FALSE;
+ gint is_spk_out_only = 0;
+ gint latency_mode = 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", &is_spk_out_only);
+ mm_attrs_get_int_by_name(attrs, "audio_latency_mode", &latency_mode);
+
+ /* hook sound_type if emergency case */
+ if ( player->sm.event == ASM_EVENT_EMERGENCY)
+ {
+ debug_log ("This is emergency session, hook sound_type from [%d] to [%d]\n", volume_type, MM_SOUND_VOLUME_TYPE_EMERGENCY);
+ volume_type = MM_SOUND_VOLUME_TYPE_EMERGENCY;
+ }
+
+ g_object_set(audiobin[MMPLAYER_A_SINK].gst,
+ "volumetype", volume_type,
+ "audio-route", audio_route,
+ "priority", sound_priority,
+ "user-route", is_spk_out_only,
+ "latency", latency_mode,
+ NULL);
+
+ debug_log("audiosink property status...volume type:%d, route:%d, priority=%d, user-route=%d, latency=%d\n",
+ volume_type, audio_route, sound_priority, is_spk_out_only, latency_mode);
+ }
+
+ /* 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);
+
+ caps_type = gst_caps_to_string(caps);
+ debug_log("resampler new caps : %s\n", caps_type);
+
+ g_object_set (GST_ELEMENT(audiobin[MMPLAYER_A_CAPS_DEFAULT].gst), "caps", caps, NULL );
+
+ /* clean */
+ gst_caps_unref( caps );
+ MMPLAYER_FREEIF( caps_type );
+
+ /* fake sink */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, "fakesink", "fakesink", TRUE);
+
+ /* set sync */
+ g_object_set (G_OBJECT (audiobin[MMPLAYER_A_SINK].gst), "sync", FALSE, NULL);
+
+ __mmplayer_add_sink( player, audiobin[MMPLAYER_A_SINK].gst );
+ }
+
+ /* 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;
+ }
+
+ /* 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;
+ }
+
+ /* get first element's sinkpad for creating ghostpad */
+ first_element = (MMPlayerGstElement *)element_bucket->data;
+
+ 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;
+ }
+
+ ghostpad = gst_ghost_pad_new("sink", pad);
+ if ( ! ghostpad )
+ {
+ debug_error("failed to create ghostpad\n");
+ goto ERROR;
+ }
+
+ 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);
+
+ if ( !player->bypass_audio_effect && (PLAYER_INI()->use_audio_effect_preset || PLAYER_INI()->use_audio_effect_custom) )
+ {
+ if ( player->audio_effect_info.effect_type == MM_AUDIO_EFFECT_TYPE_PRESET )
+ {
+ if (!_mmplayer_audio_effect_preset_apply(player, player->audio_effect_info.preset))
+ {
+ debug_msg("apply audio effect(preset:%d) setting success\n",player->audio_effect_info.preset);
+ }
+ }
+ else if ( player->audio_effect_info.effect_type == MM_AUDIO_EFFECT_TYPE_CUSTOM )
+ {
+ if (!_mmplayer_audio_effect_custom_apply(player))
+ {
+ debug_msg("apply audio effect(custom) setting success\n");
+ }
+ }
+ }
+
+ /* done. free allocated variables */
+ MMPLAYER_FREEIF( device_name );
+ g_list_free(element_bucket);
+
+ 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_fleave();
+
+ return MM_ERROR_NONE;
+
+ERROR:
+
+ debug_log("ERROR : releasing audiobin\n");
+
+ MMPLAYER_FREEIF( device_name );
+
+ 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_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));
+ }
+ }
+ }
+
+ /* release audiobin with it's childs */
+ if ( audiobin[MMPLAYER_A_BIN].gst )
+ {
+ gst_object_unref(GST_OBJECT(audiobin[MMPLAYER_A_BIN].gst));
+ }
+
+ MMPLAYER_FREEIF( audiobin );
+
+ player->pipeline->audiobin = NULL;
+
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+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;
+
+ data = GST_BUFFER_DATA(buffer);
+ size = GST_BUFFER_SIZE(buffer);
+
+ if (player->audio_stream_cb && size && data)
+ player->audio_stream_cb((void *)data, size, player->audio_stream_cb_user_param);
+
+ return TRUE;
+}
+
+/**
+ * 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) : videoflip ! xvimagesink
+ * - evas surface (arm) : ffmpegcolorspace ! videoflip ! evasimagesink
+ * - evas surface (x86) : videoconvertor ! videoflip ! evasimagesink
+ */
+static int
+__mmplayer_gst_create_video_pipeline(mm_player_t* player, GstCaps* caps, MMDisplaySurfaceType surface_type)
+{
+ GstPad *pad = NULL;
+ MMHandleType attrs;
+ GList*element_bucket = NULL;
+ MMPlayerGstElement* first_element = NULL;
+ MMPlayerGstElement* videobin = NULL;
+ gchar* vconv_factory = NULL;
+ gchar *videosink_element = NULL;
+
+ debug_fenter();
+
+ return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ /* alloc handles */
+ videobin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_V_NUM);
+ if ( !videobin )
+ {
+ return MM_ERROR_PLAYER_NO_FREE_SPACE;
+ }
+
+ player->pipeline->videobin = videobin;
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* 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_critical("failed to create videobin");
+ goto ERROR;
+ }
+
+ if( player->use_video_stream ) // video stream callback, so send raw video data to application
+ {
+ GstStructure *str = NULL;
+ guint32 fourcc = 0;
+ gint ret = 0;
+ gint width = 0; //width of video
+ gint height = 0; //height of video
+ GstCaps* video_caps = NULL;
+
+ debug_log("using memsink\n");
+
+ /* first, create colorspace convert */
+ if (strlen(PLAYER_INI()->name_of_video_converter) > 0)
+ {
+ vconv_factory = PLAYER_INI()->name_of_video_converter;
+ }
+
+ if (vconv_factory)
+ {
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CONV, vconv_factory, "video converter", TRUE);
+ }
+
+ /* rotator, scaler and capsfilter */
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_FLIP, "videoflip", "video rotator", TRUE);
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SCALE, "videoscale", "video scaler", TRUE);
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CAPS, "capsfilter", "videocapsfilter", TRUE);
+
+ /* 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;
+ }
+
+ mm_attrs_get_int_by_name(attrs, "display_width", &width);
+ mm_attrs_get_int_by_name(attrs, "display_height", &height);
+ if (!width || !height)
+ {
+ /* we set width/height of original media's size to capsfilter for scaling video */
+ ret = gst_structure_get_int (str, "width", &width);
+ if ( !ret )
+ {
+ debug_error("cannot get width");
+ goto ERROR;
+ }
+
+ ret = gst_structure_get_int(str, "height", &height);
+ if ( !ret )
+ {
+ debug_error("cannot get height");
+ goto ERROR;
+ }
+ }
+
+ video_caps = gst_caps_new_simple( "video/x-raw-rgb",
+ "width", G_TYPE_INT, width,
+ "height", G_TYPE_INT, height,
+ NULL);
+
+ g_object_set (GST_ELEMENT(videobin[MMPLAYER_V_CAPS].gst), "caps", video_caps, NULL );
+
+ gst_caps_unref( video_caps );
+
+ /* finally, create video sink. output will be BGRA8888. */
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, "avsysmemsink", "videosink", TRUE);
+
+ MMPLAYER_SIGNAL_CONNECT( player,
+ videobin[MMPLAYER_V_SINK].gst,
+ "video-stream",
+ G_CALLBACK(__mmplayer_videostream_cb),
+ player );
+ }
+ else // render video data using sink plugin like xvimagesink
+ {
+ debug_log("using videosink");
+
+ /* set video converter */
+ if (strlen(PLAYER_INI()->name_of_video_converter) > 0)
+ {
+ vconv_factory = PLAYER_INI()->name_of_video_converter;
+ if (vconv_factory)
+ {
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_CONV, vconv_factory, "video converter", TRUE);
+ debug_log("using video converter: %s", vconv_factory);
+ }
+ }
+
+ /* set video rotator */
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_FLIP, "videoflip", "video rotator", TRUE);
+
+ /* videoscaler */
+ #if !defined(__arm__)
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SCALE, "videoscale", "videoscaler", TRUE);
+ #endif
+
+ /* 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;
+ debug_warning("video texture usage");
+ }
+ 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;
+ }
+
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, videosink_element, videosink_element, TRUE);
+ debug_log("selected videosink name: %s", videosink_element);
+
+ /* 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,
+ "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;
+ }
+ }
+
+ if ( _mmplayer_update_video_param(player) != MM_ERROR_NONE)
+ goto ERROR;
+
+ /* qos on */
+ g_object_set (G_OBJECT (videobin[MMPLAYER_V_SINK].gst), "qos", TRUE, NULL);
+
+ /* store it as it's sink element */
+ __mmplayer_add_sink( player, videobin[MMPLAYER_V_SINK].gst );
+
+ /* 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;
+ }
+
+ /* 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;
+ }
+
+ /* 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;
+ }
+
+ 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;
+ }
+
+ /* create ghostpad */
+ if (FALSE == gst_element_add_pad(videobin[MMPLAYER_V_BIN].gst, gst_ghost_pad_new("sink", pad)))
+ {
+ debug_error("failed to add ghostpad to videobin\n");
+ goto ERROR;
+ }
+ 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");
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+
+ERROR:
+ debug_error("ERROR : releasing videobin\n");
+
+ g_list_free( element_bucket );
+
+ if (pad)
+ gst_object_unref(GST_OBJECT(pad));
+
+ /* release videobin with it's childs */
+ if ( videobin[MMPLAYER_V_BIN].gst )
+ {
+ gst_object_unref(GST_OBJECT(videobin[MMPLAYER_V_BIN].gst));
+ }
+
+
+ MMPLAYER_FREEIF( videobin );
+
+ player->pipeline->videobin = NULL;
+
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+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_fenter();
+
+ return_val_if_fail( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* alloc handles */
+ textbin = (MMPlayerGstElement*)g_malloc0(sizeof(MMPlayerGstElement) * MMPLAYER_T_NUM);
+ if ( ! textbin )
+ {
+ debug_error("failed to allocate memory for textbin\n");
+ return MM_ERROR_PLAYER_NO_FREE_SPACE;
+ }
+
+ /* 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_critical("failed to create textbin\n");
+ goto ERROR;
+ }
+
+ /* take it */
+ player->pipeline->textbin = textbin;
+
+ /* fakesink */
+ if (player->use_textoverlay)
+ {
+ debug_log ("use textoverlay for displaying \n");
+
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_TEXT_QUEUE, "queue", "text_t_queue", textbin[MMPLAYER_T_BIN].gst);
+
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_VIDEO_QUEUE, "queue", "text_v_queue", textbin[MMPLAYER_T_BIN].gst);
+
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_VIDEO_CONVERTER, "fimcconvert", "text_v_converter", textbin[MMPLAYER_T_BIN].gst);
+
+ MMPLAYER_CREATE_ELEMENT_ADD_BIN(textbin, MMPLAYER_T_OVERLAY, "textoverlay", "text_overlay", textbin[MMPLAYER_T_BIN].gst);
+
+ 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 (!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;
+ }
+
+ 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;
+ }
+
+ }
+ else
+ {
+ debug_log ("use subtitle message for displaying \n");
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_TEXT_QUEUE, "queue", "text_queue", TRUE);
+
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_SINK, "fakesink", "text_sink", TRUE);
+
+ 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_SIGNAL_CONNECT( player,
+ G_OBJECT(textbin[MMPLAYER_T_SINK].gst),
+ "handoff",
+ G_CALLBACK(__mmplayer_update_subtitle),
+ (gpointer)player );
+
+ 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_SINK].gst));
+ }
+
+ /* 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;
+ }
+
+ /* 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;
+ }
+
+ /* done. free allocated variables */
+ g_list_free(element_bucket);
+ }
+
+ if (textbin[MMPLAYER_T_TEXT_QUEUE].gst)
+ {
+ pad = gst_element_get_static_pad(GST_ELEMENT(textbin[MMPLAYER_T_TEXT_QUEUE].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;
+ }
+ }
+
+ if (textbin[MMPLAYER_T_VIDEO_QUEUE].gst)
+ {
+ pad = gst_element_get_static_pad(GST_ELEMENT(textbin[MMPLAYER_T_VIDEO_QUEUE].gst), "sink");
+ if (!pad)
+ {
+ debug_error("failed to get video pad of textbin\n");
+ goto ERROR;
+ }
+
+ ghostpad = gst_ghost_pad_new("video_sink", pad);
+ if (!ghostpad)
+ {
+ 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;
+ }
+ }
+
+ if (textbin[MMPLAYER_T_OVERLAY].gst)
+ {
+ pad = gst_element_get_static_pad(GST_ELEMENT(textbin[MMPLAYER_T_OVERLAY].gst), "src");
+ if (!pad)
+ {
+ debug_error("failed to get src pad of textbin\n");
+ goto ERROR;
+ }
+
+ ghostpad = gst_ghost_pad_new("src", pad);
+ if (!ghostpad)
+ {
+ 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;
+ }
+ }
+
+ gst_object_unref(pad);
+
+ debug_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 );
+
+ if ( !parent )
+ {
+ gst_object_unref(GST_OBJECT(textbin[i].gst));
+ textbin[i].gst = NULL;
+ }
+ else
+ {
+ gst_object_unref(GST_OBJECT(parent));
+ }
+ }
+ }
+
+ /* release textbin with it's childs */
+ if ( textbin[MMPLAYER_T_BIN].gst )
+ {
+ gst_object_unref(GST_OBJECT(textbin[MMPLAYER_T_BIN].gst));
+ }
+
+ MMPLAYER_FREEIF( textbin );
+
+ player->pipeline->textbin = NULL;
+
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+
+static int
+__mmplayer_gst_create_subtitle_src(mm_player_t* player)
+{
+ MMPlayerGstElement* mainbin = NULL;
+ MMHandleType attrs = 0;
+ GstElement * pipeline = NULL;
+ GstElement *subsrc = NULL;
+ GstElement *subparse = NULL;
+ GstPad *sinkpad = NULL;
+ gchar *subtitle_uri =NULL;
+ gchar *charset = NULL;
+
+ debug_fenter();
+
+ /* get mainbin */
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ pipeline = player->pipeline->mainbin[MMPLAYER_M_PIPE].gst;
+ mainbin = player->pipeline->mainbin;
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ 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");
+ return MM_ERROR_PLAYER_INVALID_URI;
+ }
+ debug_log("subtitle file path is [%s].\n", subtitle_uri);
+
+
+ /* create the subtitle source */
+ subsrc = gst_element_factory_make("filesrc", "subtitle_source");
+ if ( !subsrc )
+ {
+ debug_error ( "failed to create filesrc element\n" );
+ goto ERROR;
+ }
+ g_object_set(G_OBJECT (subsrc), "location", subtitle_uri, NULL);
+
+ 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_warning("failed to add queue\n");
+ goto ERROR;
+ }
+
+ /* subparse */
+ subparse = gst_element_factory_make("subparse", "subtitle_parser");
+ if ( !subparse )
+ {
+ debug_error ( "failed to create subparse element\n" );
+ goto ERROR;
+ }
+
+ charset = util_get_charset(subtitle_uri);
+ if (charset)
+ {
+ debug_log ("detected charset is %s\n", charset );
+ g_object_set (G_OBJECT (subparse), "subtitle-encoding", charset, NULL);
+ }
+
+ mainbin[MMPLAYER_M_SUBPARSE].id = MMPLAYER_M_SUBPARSE;
+ mainbin[MMPLAYER_M_SUBPARSE].gst = subparse;
+
+ if (!gst_bin_add(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), subparse))
+ {
+ debug_warning("failed to add subparse\n");
+ goto ERROR;
+ }
+
+ if (!gst_element_link_pads (subsrc, "src", subparse, "sink"))
+ {
+ debug_warning("failed to link subsrc and subparse\n");
+ goto ERROR;
+ }
+
+ player->play_subtitle = TRUE;
+ debug_log ("play subtitle using subtitle file\n");
+
+ if (MM_ERROR_NONE != __mmplayer_gst_create_text_pipeline(player))
+ {
+ debug_error("failed to create textbin. continuing without text\n");
+ goto ERROR;
+ }
+
+ 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;
+ }
+
+ if (!gst_element_link_pads (subparse, "src", player->pipeline->textbin[MMPLAYER_T_BIN].gst, "text_sink"))
+ {
+ debug_warning("failed to link subparse and textbin\n");
+ goto ERROR;
+ }
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+
+
+ERROR:
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( buffer, FALSE );
+
+ text = GST_BUFFER_DATA(buffer);
+ duration = GST_BUFFER_DURATION(buffer);
+
+ if ( player->is_subtitle_off )
+ {
+ debug_log("subtitle is OFF.\n" );
+ return TRUE;
+ }
+
+ if ( !text )
+ {
+ debug_log("There is no subtitle to be displayed.\n" );
+ return TRUE;
+ }
+
+ 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;
+}
+
+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;
+
+ debug_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)
+ {
+ debug_log ("nothing to do\n");
+ return MM_ERROR_NONE;
+ }
+
+ switch (format)
+ {
+ case MM_PLAYER_POS_FORMAT_TIME:
+ {
+ /* check current postion */
+ if (__gst_get_position(player, MM_PLAYER_POS_FORMAT_TIME, ¤t_pos ))
+ {
+ debug_error("failed to get position");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ adusted_pos = (gint64)current_pos + ((gint64)position * G_GINT64_CONSTANT(1000000));
+ if (adusted_pos < 0)
+ adusted_pos = G_GUINT64_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;
+
+ default:
+ {
+ debug_warning("invalid format.\n");
+ return MM_ERROR_INVALID_ARGUMENT;
+ }
+ }
+
+ /* keep ref to the event */
+ gst_event_ref (event);
+
+ debug_log("sending event[%s] to subparse element [%s]\n",
+ GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(player->pipeline->mainbin[MMPLAYER_M_SUBPARSE].gst) );
+
+ if (gst_element_send_event (player->pipeline->mainbin[MMPLAYER_M_SUBPARSE].gst, event))
+ {
+ debug_log("sending event[%s] to subparse element [%s] success!\n",
+ GST_EVENT_TYPE_NAME(event), GST_ELEMENT_NAME(player->pipeline->mainbin[MMPLAYER_M_SUBPARSE].gst) );
+ }
+
+ /* unref to the event */
+ gst_event_unref (event);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+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;
+
+ return_if_fail ( element );
+ return_if_fail ( buf );
+
+ buffer = gst_buffer_new ();
+
+ if (buf->offset >= buf->len)
+ {
+ debug_log("call eos appsrc\n");
+ g_signal_emit_by_name (appsrc, "end-of-stream", &ret);
+ return;
+ }
+
+ if ( buf->len - buf->offset < size)
+ {
+ len = buf->len - buf->offset + buf->offset;
+ }
+
+ 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 );
+
+ buf->offset = (int)size;
+
+ return TRUE;
+}
+
+static void
+__gst_appsrc_feed_data(GstElement *element, guint size, gpointer user_data) // @
+{
+ mm_player_t *player = (mm_player_t*)user_data;
+
+ return_if_fail ( player );
+
+ debug_msg("app-src: feed data\n");
+
+ if(player->need_data_cb)
+ player->need_data_cb(size, player->buffer_cb_user_param);
+}
+
+static gboolean
+__gst_appsrc_seek_data(GstElement *element, guint64 offset, gpointer user_data) // @
+{
+ mm_player_t *player = (mm_player_t*)user_data;
+
+ return_val_if_fail ( player, FALSE );
+
+ debug_msg("app-src: seek data\n");
+
+ 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_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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* check current state */
+// MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_START );
+
+
+ /* NOTE : we should check and create pipeline again if not created as we destroy
+ * whole pipeline when stopping in streamming playback
+ */
+ if ( ! player->pipeline )
+ {
+ if ( MM_ERROR_NONE != __gst_realize( player ) )
+ {
+ debug_error("failed to realize before starting. only in streamming\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+ }
+
+ debug_msg("app-src: pushing data\n");
+
+ if ( buf == NULL )
+ {
+ debug_error("buf is null\n");
+ return MM_ERROR_NONE;
+ }
+
+ buffer = gst_buffer_new ();
+
+ 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;
+ }
+
+ GST_BUFFER_DATA(buffer) = (guint8*)(buf);
+ GST_BUFFER_SIZE(buffer) = size;
+
+ 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_fleave();
+
+ return ret;
+}
+
+static GstBusSyncReply
+__mmplayer_bus_sync_callback (GstBus * bus, GstMessage * message, gpointer data)
+{
+ mm_player_t *player = (mm_player_t *)data;
+
+ switch (GST_MESSAGE_TYPE (message))
+ {
+ case GST_MESSAGE_TAG:
+ __mmplayer_gst_extract_tag_from_msg(player, message);
+ break;
+
+ default:
+ return GST_BUS_PASS;
+ }
+ gst_message_unref (message);
+
+ return GST_BUS_DROP;
+}
+
+/**
+ * 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) // @
+{
+ GstBus *bus = NULL;
+ MMPlayerGstElement *mainbin = NULL;
+ MMHandleType attrs = 0;
+ GstElement* element = NULL;
+ GList* element_bucket = NULL;
+ gboolean need_state_holder = TRUE;
+ gint i = 0;
+
+ debug_fenter();
+
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ /* get profile attribute */
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute\n");
+ goto INIT_ERROR;
+ }
+
+ /* create pipeline handles */
+ if ( player->pipeline )
+ {
+ debug_warning("pipeline should be released before create new one\n");
+ goto INIT_ERROR;
+ }
+
+ player->pipeline = (MMPlayerGstPipelineInfo*) g_malloc0( sizeof(MMPlayerGstPipelineInfo) );
+ if (player->pipeline == NULL)
+ goto INIT_ERROR;
+
+ memset( player->pipeline, 0, sizeof(MMPlayerGstPipelineInfo) );
+
+
+ /* 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 )
+ {
+ debug_error("failed to create pipeline\n");
+ goto INIT_ERROR;
+ }
+
+
+ /* create source element */
+ switch ( player->profile.uri_type )
+ {
+ /* rtsp streamming */
+ case MM_PLAYER_URI_TYPE_URL_RTSP:
+ {
+ gint network_bandwidth;
+ gchar *user_agent, *wap_profile;
+
+ element = gst_element_factory_make(PLAYER_INI()->name_of_rtspsrc, "streaming_source");
+
+ if ( !element )
+ {
+ debug_critical("failed to create streaming source element\n");
+ break;
+ }
+
+ debug_log("using streamming source [%s].\n", PLAYER_INI()->name_of_rtspsrc);
+
+ /* make it zero */
+ network_bandwidth = 0;
+ user_agent = wap_profile = NULL;
+
+ /* 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 );
+
+ debug_log("setting streaming source ----------------\n");
+ 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");
+
+ /* 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);
+
+ 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 );
+
+ player->no_more_pad = FALSE;
+ player->num_dynamic_pad = 0;
+
+ /* NOTE : we cannot determine it yet. this filed will be filled by
+ * _mmplayer_update_content_attrs() after START.
+ */
+ player->streaming_type = STREAMING_SERVICE_NONE;
+ }
+ break;
+
+ /* http streaming*/
+ case MM_PLAYER_URI_TYPE_URL_HTTP:
+ {
+ 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;
+
+ debug_log("http playback, PD mode : %d\n", player->pd_mode);
+
+ if ( ! MMPLAYER_IS_HTTP_PD(player) )
+ {
+ element = gst_element_factory_make(PLAYER_INI()->name_of_httpsrc, "http_streaming_source");
+ if ( !element )
+ {
+ debug_critical("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 );
+
+ /* 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)
+ {
+ gchar *path = NULL;
+
+ mm_attrs_get_string_by_name ( attrs, "pd_location", &path );
+
+ MMPLAYER_FREEIF(player->pd_file_save_path);
+
+ debug_log("PD Location : %s\n", path);
+
+ 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;
+ }
+ }
+
+ element = gst_element_factory_make("pdpushsrc", "PD pushsrc");
+ if ( !element )
+ {
+ debug_critical("failed to create PD push source element[%s].\n", "pdpushsrc");
+ break;
+ }
+
+ g_object_set(G_OBJECT(element), "location", player->pd_file_save_path, NULL);
+ }
+
+ player->streaming_type = STREAMING_SERVICE_NONE;
+ }
+ break;
+
+ /* file source */
+ case MM_PLAYER_URI_TYPE_FILE:
+ {
+ 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_critical("failed to create %s\n", drmsrc);
+ break;
+ }
+
+ 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;
+
+ /* appsrc */
+ case MM_PLAYER_URI_TYPE_BUFF:
+ {
+ guint64 stream_type = GST_APP_STREAM_TYPE_STREAM;
+
+ debug_log("mem src is selected\n");
+
+ element = gst_element_factory_make("appsrc", "buff-source");
+ if ( !element )
+ {
+ debug_critical("failed to create appsrc element\n");
+ break;
+ }
+
+ 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), 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);
+ }
+ break;
+
+ /* appsrc */
+ case MM_PLAYER_URI_TYPE_MEM:
+ {
+ guint64 stream_type = GST_APP_STREAM_TYPE_RANDOM_ACCESS;
+
+ debug_log("mem src is selected\n");
+
+ element = gst_element_factory_make("appsrc", "mem-source");
+ if ( !element )
+ {
+ debug_critical("failed to create appsrc element\n");
+ break;
+ }
+
+ 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 );
+ }
+ break;
+ case MM_PLAYER_URI_TYPE_URL:
+ break;
+
+ case MM_PLAYER_URI_TYPE_TEMP:
+ break;
+
+ case MM_PLAYER_URI_TYPE_NONE:
+ default:
+ break;
+ }
+
+ /* check source element is OK */
+ if ( ! element )
+ {
+ debug_critical("no source element was created.\n");
+ goto INIT_ERROR;
+ }
+
+ /* 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]);
+
+ if (MMPLAYER_IS_STREAMING(player))
+ {
+ player->streamer = __mm_player_streaming_create();
+ __mm_player_streaming_initialize(player->streamer);
+ }
+
+ 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);
+ }
+
+ /* create autoplugging element if src element is not a streamming src */
+ if ( player->profile.uri_type != MM_PLAYER_URI_TYPE_URL_RTSP )
+ {
+ element = NULL;
+
+ if( PLAYER_INI()->use_decodebin )
+ {
+ /* create decodebin */
+ element = gst_element_factory_make("decodebin", "decodebin");
+
+ g_object_set(G_OBJECT(element), "async-handling", TRUE, NULL);
+
+ /* set signal handler */
+ MMPLAYER_SIGNAL_CONNECT( player, G_OBJECT(element), "new-decoded-pad",
+ G_CALLBACK(__mmplayer_gst_decode_callback), player);
+
+ /* 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 );
+ }
+
+ /* check autoplug element is OK */
+ if ( ! element )
+ {
+ debug_critical("can not create autoplug element\n");
+ goto INIT_ERROR;
+ }
+
+ mainbin[MMPLAYER_M_AUTOPLUG].id = MMPLAYER_M_AUTOPLUG;
+ mainbin[MMPLAYER_M_AUTOPLUG].gst = element;
+
+ element_bucket = g_list_append(element_bucket, &mainbin[MMPLAYER_M_AUTOPLUG]);
+ }
+
+
+ /* add elements to pipeline */
+ if( !__mmplayer_gst_element_add_bucket_to_bin(GST_BIN(mainbin[MMPLAYER_M_PIPE].gst), element_bucket))
+ {
+ debug_error("Failed to add elements to pipeline\n");
+ goto INIT_ERROR;
+ }
+
+
+ /* linking elements in the bucket by added order. */
+ if ( __mmplayer_gst_element_link_bucket(element_bucket) == -1 )
+ {
+ debug_error("Failed to link some elements\n");
+ goto INIT_ERROR;
+ }
+
+
+ /* 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");
+
+ 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);
+
+ /* 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) )
+ {
+ debug_error("failed to add fakesink to bin\n");
+ goto INIT_ERROR;
+ }
+ }
+
+ /* 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 )
+ {
+ debug_error ("cannot get bus from pipeline.\n");
+ goto INIT_ERROR;
+ }
+ player->bus_watcher = gst_bus_add_watch(bus, (GstBusFunc)__mmplayer_gst_callback, player);
+
+ /* Note : check whether subtitle atrribute uri is set. If uri is set, then tyr 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")
+ }
+
+ /* 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:
+
+ __mmplayer_gst_destroy_pipeline(player);
+ g_list_free(element_bucket);
+
+ /* 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));
+ }
+ }
+ }
+
+ /* release pipeline with it's childs */
+ if ( mainbin[MMPLAYER_M_PIPE].gst )
+ {
+ gst_object_unref(GST_OBJECT(mainbin[MMPLAYER_M_PIPE].gst));
+ }
+
+ MMPLAYER_FREEIF( player->pipeline );
+ MMPLAYER_FREEIF( mainbin );
+
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+
+static int
+__mmplayer_gst_destroy_pipeline(mm_player_t* player) // @
+{
+ gint timeout = 0;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_INVALID_HANDLE );
+
+ /* cleanup stuffs */
+ MMPLAYER_FREEIF(player->type);
+ player->have_dynamic_pad = FALSE;
+ player->no_more_pad = FALSE;
+ player->num_dynamic_pad = 0;
+
+ if (player->v_stream_caps)
+ {
+ gst_caps_unref(player->v_stream_caps);
+ player->v_stream_caps = NULL;
+ }
+
+ if (ahs_appsrc_cb_probe_id )
+ {
+ 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;
+ }
+
+ if ( player->sink_elements )
+ g_list_free ( player->sink_elements );
+ player->sink_elements = NULL;
+
+ /* cleanup unlinked mime type */
+ MMPLAYER_FREEIF(player->unlinked_audio_mime);
+ MMPLAYER_FREEIF(player->unlinked_video_mime);
+ MMPLAYER_FREEIF(player->unlinked_demuxer_mime);
+
+ /* cleanup running stuffs */
+ __mmplayer_cancel_delayed_eos( player );
+
+ /* cleanup gst stuffs */
+ if ( player->pipeline )
+ {
+ MMPlayerGstElement* mainbin = player->pipeline->mainbin;
+ GstTagList* tag_list = player->pipeline->tag_list;
+
+ /* first we need to disconnect all signal hander */
+ __mmplayer_release_signal_connection( player );
+
+ /* disconnecting bus watch */
+ if ( player->bus_watcher )
+ g_source_remove( player->bus_watcher );
+ player->bus_watcher = 0;
+
+ if ( mainbin )
+ {
+ 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);
+
+ debug_log("pipeline status before set state to NULL\n");
+ __mmplayer_dump_pipeline_state( player );
+
+ 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("pipeline status before unrefering pipeline\n");
+ __mmplayer_dump_pipeline_state( player );
+
+ 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 );
+ }
+
+ if ( tag_list )
+ gst_tag_list_free(tag_list);
+
+ MMPLAYER_FREEIF( player->pipeline );
+ }
+
+ 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;
+
+ debug_fenter();
+
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_READY;
+
+ __ta__("__mmplayer_gst_create_pipeline",
+ ret = __mmplayer_gst_create_pipeline(player);
+ if ( ret )
+ {
+ debug_critical("failed to create pipeline\n");
+ return ret;
+ }
+ )
+
+ /* 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);
+
+ if ( ret != MM_ERROR_NONE )
+ {
+ /* return error if failed to set state */
+ debug_error("failed to set state PAUSED (live : READY).\n");
+
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+
+ return ret;
+ }
+ else
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_READY );
+ }
+
+ /* create dot before error-return. for debugging */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-realize" );
+
+ debug_fleave();
+
+ return ret;
+}
+
+static int __gst_unrealize(mm_player_t* player) // @
+{
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_NULL;
+ MMPLAYER_PRINT_STATE(player);
+
+ /* 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;
+ }
+
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_NULL );
+
+ debug_fleave();
+
+ return ret;
+}
+
+static int __gst_pending_seek ( mm_player_t* player )
+{
+ MMPlayerStateType current_state = MM_PLAYER_STATE_NONE;
+ MMPlayerStateType pending_state = MM_PLAYER_STATE_NONE;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ 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);
+ pending_state = MMPLAYER_PENDING_STATE(player);
+
+ if ( current_state != MM_PLAYER_STATE_PAUSED && current_state != MM_PLAYER_STATE_PLAYING )
+ {
+ 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, FALSE );
+
+ if ( MM_ERROR_NONE != ret )
+ debug_error("failed to seek pending postion. just keep staying current position.\n");
+
+ player->pending_seek.is_pending = FALSE;
+
+ debug_fleave();
+
+ return ret;
+}
+
+static int __gst_start(mm_player_t* player) // @
+{
+ gboolean sound_extraction = 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))
+ {
+ 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");
+
+ 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");
+ }
+ }
+ }
+
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PLAYING;
+ MMPLAYER_PRINT_STATE(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)
+ {
+ MMPLAYER_SET_STATE(player, MM_PLAYER_STATE_PLAYING);
+ }
+ 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();
+
+ return ret;
+}
+
+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();
+}
+
+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);
+
+ debug_fleave();
+}
+
+static int __gst_stop(mm_player_t* player) // @
+{
+ GstStateChangeReturn change_ret = GST_STATE_CHANGE_SUCCESS;
+ MMHandleType attrs = 0;
+ gboolean fadewown = FALSE;
+ gboolean rewind = FALSE;
+ gint timeout = 0;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ 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("cannot get content attribute\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ mm_attrs_get_int_by_name(attrs,"sound_fadedown", &fadewown);
+
+ /* enable fadedown */
+ if (fadewown)
+ __mmplayer_do_sound_fadedown(player, MM_PLAYER_FADEOUT_TIME_DEFAULT);
+
+ /* 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 || player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS)
+ {
+ ret = __mmplayer_gst_set_state(player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_READY, FALSE, timeout );
+ }
+ else
+ {
+ ret = __mmplayer_gst_set_state( player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PAUSED, FALSE, timeout );
+
+ if ( !MMPLAYER_IS_STREAMING(player))
+ rewind = TRUE;
+ }
+
+ /* disable fadeout */
+ if (fadewown)
+ __mmplayer_undo_sound_fadedown(player);
+
+
+ /* return if set_state has failed */
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to set state.\n");
+
+ /* dump state of all element. don't care it success or not */
+ __mmplayer_dump_pipeline_state( player );
+
+ return ret;
+ }
+
+ /* 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;
+ }
+ }
+
+ /* 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;
+ }
+
+ /* generate dot file if enabled */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-stop" );
+
+ debug_fleave();
+
+ return ret;
+}
+
+int __gst_pause(mm_player_t* player, gboolean async) // @
+{
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail(player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PAUSED;
+ MMPLAYER_PRINT_STATE(player);
+
+ /* set pipeline status to PAUSED */
+ ret = __mmplayer_gst_set_state(player,
+ player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, GST_STATE_PAUSED, async, MMPLAYER_STATE_CHANGE_TIMEOUT(player));
+
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to set state to PAUSED\n");
+
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+
+ return ret;
+ }
+ else
+ {
+ if ( async == FALSE )
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PAUSED );
+ }
+ }
+
+ /* FIXIT : analyze so called "async problem" */
+ /* set async off */
+ __gst_set_async_state_change( player, TRUE);
+
+ /* generate dot file before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-pause" );
+
+ debug_fleave();
+
+ return ret;
+}
+
+int __gst_resume(mm_player_t* player, gboolean async) // @
+{
+ int ret = MM_ERROR_NONE;
+ gint timeout = 0;
+
+ debug_fenter();
+
+ return_val_if_fail(player && player->pipeline,
+ MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ debug_log("current state before doing transition");
+ MMPLAYER_PENDING_STATE(player) = MM_PLAYER_STATE_PLAYING;
+ MMPLAYER_PRINT_STATE(player);
+
+ /* generate dot file before returning error */
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-resume" );
+
+ __mmplayer_set_antishock( player , FALSE );
+
+ if ( async )
+ debug_log("do async state transition to PLAYING.\n");
+
+ /* 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");
+
+ /* dump state of all element */
+ __mmplayer_dump_pipeline_state( player );
+
+ return ret;
+ }
+ else
+ {
+ if (async == FALSE)
+ {
+ MMPLAYER_SET_STATE ( player, MM_PLAYER_STATE_PLAYING );
+ }
+ }
+
+ /* 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" );
+
+ debug_fleave();
+
+ return ret;
+}
+
+static int
+__gst_set_position(mm_player_t* player, int format, unsigned long position, gboolean internal_called) // @
+{
+ GstFormat fmt = GST_FORMAT_TIME;
+ unsigned long dur_msec = 0;
+ gint64 dur_nsec = 0;
+ gint64 pos_nsec = 0;
+ gboolean ret = TRUE;
+
+ debug_fenter();
+ 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 );
+
+ 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 ( !gst_element_query_duration( player->pipeline->mainbin[MMPLAYER_M_PIPE].gst, &fmt, &dur_nsec ))
+ {
+ goto SEEK_ERROR;
+ }
+ player->duration = dur_nsec;
+ }
+
+ 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("playback rate: %f\n", player->playback_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_log("seeking to (%lu) msec, duration is %d msec\n", position, dur_msec);
+
+ if (player->doing_seek)
+ {
+ debug_log("not completed seek");
+ return MM_ERROR_PLAYER_DOING_SEEK;
+ }
+
+ if ( !internal_called)
+ player->doing_seek = TRUE;
+
+ pos_nsec = 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_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;
+
+ case MM_PLAYER_POS_FORMAT_PERCENT:
+ {
+ debug_log("seeking to (%lu)%% \n", position);
+
+ if (player->doing_seek)
+ {
+ debug_log("not completed seek");
+ return MM_ERROR_PLAYER_DOING_SEEK;
+ }
+
+ if ( !internal_called)
+ player->doing_seek = TRUE;
+
+ /* 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, 1.0,
+ GST_FORMAT_TIME, ( GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE ),
+ 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;
+
+ default:
+ goto INVALID_ARGS;
+
+ }
+
+ /* NOTE : store last seeking point to overcome some bad operation
+ * ( returning zero when getting current position ) of some elements
+ */
+ player->last_position = pos_nsec;
+
+ /* 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 MM_ERROR_NONE;
+
+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(MMPLAYER_CURRENT_STATE(player)), MMPLAYER_STATE_GET_NAME(MMPLAYER_PENDING_STATE(player)), player->pending_seek.pos);
+
+ return MM_ERROR_NONE;
+
+INVALID_ARGS:
+ debug_error("invalid arguments, position : %ld dur : %ld format : %d \n", position, dur_msec, format);
+ return MM_ERROR_INVALID_ARGUMENT;
+
+SEEK_ERROR:
+ player->doing_seek = FALSE;
+ return MM_ERROR_PLAYER_SEEK;
+}
+
+#define TRICKPLAY_OFFSET GST_MSECOND
+
+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;
+
+ return_val_if_fail( player && position && player->pipeline && player->pipeline->mainbin,
+ MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ current_state = MMPLAYER_CURRENT_STATE(player);
+
+ /* 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);
+ }
+
+ /* 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 ("pos_msec = %"GST_TIME_FORMAT" and ret = %d and state = %d", GST_TIME_ARGS (pos_msec), ret, current_state);
+
+ if(player->playback_rate < 0.0)
+ pos_msec = player->last_position - TRICKPLAY_OFFSET;
+ else
+ pos_msec = player->last_position;
+
+ if (!ret)
+ pos_msec = player->last_position;
+ else
+ player->last_position = pos_msec;
+
+ debug_warning("returning last point : %"GST_TIME_FORMAT, GST_TIME_ARGS(pos_msec));
+
+ }
+ 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:
+ {
+ int dur = 0;
+ int pos = 0;
+
+ dur = player->duration / GST_SECOND;
+ if (dur <= 0)
+ {
+ debug_log ("duration is [%d], so returning position 0\n",dur);
+ *position = 0;
+ }
+ else
+ {
+ pos = pos_msec / GST_SECOND;
+ *position = pos * 100 / dur;
+ }
+ break;
+ }
+ default:
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ debug_log("current position : %lu\n", *position);
+
+
+ return MM_ERROR_NONE;
+}
+
+
+static int __gst_get_buffer_position(mm_player_t* player, int format, unsigned long* start_pos, unsigned long* stop_pos)
+{
+ GstElement *element = NULL;
+ GstQuery *query = NULL;
+
+ return_val_if_fail( player &&
+ player->pipeline &&
+ player->pipeline->mainbin,
+ MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ return_val_if_fail( start_pos && stop_pos, MM_ERROR_INVALID_ARGUMENT );
+
+ if ( MMPLAYER_IS_HTTP_STREAMING ( player ))
+ {
+ /* 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 );
+ }
+ else if ( MMPLAYER_IS_RTSP_STREAMING ( player ) )
+ {
+ debug_warning ( "it's not supported yet.\n" );
+ return MM_ERROR_NONE;
+ }
+ else
+ {
+ debug_warning ( "it's only used for streaming case.\n" );
+ return MM_ERROR_NONE;
+ }
+
+ *start_pos = 0;
+ *stop_pos = 0;
+
+ switch ( format )
+ {
+ 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;
+
+ gst_query_parse_buffering_percent ( query, &busy, &percent);
+ gst_query_parse_buffering_range ( query, &format, &start, &stop, NULL );
+
+ debug_log ( "buffering start %" G_GINT64_FORMAT ", stop %" G_GINT64_FORMAT "\n", start, stop);
+
+ if ( start != -1)
+ *start_pos = 100 * start / GST_FORMAT_PERCENT_MAX;
+ else
+ *start_pos = 0;
+
+ if ( stop != -1)
+ *stop_pos = 100 * stop / GST_FORMAT_PERCENT_MAX;
+ else
+ *stop_pos = 0;
+ }
+ gst_query_unref (query);
+ }
+ break;
+
+ case MM_PLAYER_POS_FORMAT_TIME :
+ debug_warning ( "Time format is not supported yet.\n" );
+ break;
+
+ default :
+ break;
+ }
+
+ debug_log("current buffer position : %lu~%lu \n", *start_pos, *stop_pos );
+
+ return MM_ERROR_NONE;
+}
+
+static int
+__gst_set_message_callback(mm_player_t* player, MMMessageCallback callback, gpointer user_param) // @
+{
+ debug_fenter();
+
+ if ( !player )
+ {
+ debug_warning("set_message_callback is called with invalid player handle\n");
+ return MM_ERROR_PLAYER_NOT_INITIALIZED;
+ }
+
+ 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);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+static gboolean __mmfplayer_parse_profile(const char *uri, void *param, MMPlayerParseProfile* data) // @
+{
+ gboolean ret = FALSE;
+ char *path = NULL;
+
+ debug_fenter();
+
+ 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));
+
+ if ((path = strstr(uri, "file://")))
+ {
+ if (util_exist_file_path(path + 7)) {
+ strncpy(data->uri, path, MM_MAX_URL_LEN-1);
+
+ 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);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_HTTP;
+
+ ret = TRUE;
+ }
+ }
+ else if ((path = strstr(uri, "https://")))
+ {
+ if (strlen(path)) {
+ strcpy(data->uri, uri);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_HTTP;
+
+ ret = TRUE;
+ }
+ }
+ else if ((path = strstr(uri, "rtspu://")))
+ {
+ if (strlen(path)) {
+ strcpy(data->uri, uri);
+ data->uri_type = MM_PLAYER_URI_TYPE_URL_RTSP;
+ ret = TRUE;
+ }
+ }
+ 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,};
+
+ if (seperator) {
+ if ((buffer = strstr(path, "ext="))) {
+ buffer += strlen("ext=");
+
+ if (strlen(buffer)) {
+ strcpy(ext, buffer);
+
+ if ((seperator = strchr(ext, ','))
+ || (seperator = strchr(ext, ' '))
+ || (seperator = strchr(ext, '\0'))) {
+ seperator[0] = '\0';
+ }
+ }
+ }
+
+ if ((buffer = strstr(path, "size="))) {
+ buffer += strlen("size=");
+
+ if (strlen(buffer) > 0) {
+ strcpy(size, buffer);
+
+ if ((seperator = strchr(size, ','))
+ || (seperator = strchr(size, ' '))
+ || (seperator = strchr(size, '\0'))) {
+ seperator[0] = '\0';
+ }
+
+ mem_size = atoi(size);
+ }
+ }
+ }
+
+ 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;
+ }
+ }
+ }
+ else
+ {
+ /* if no protocol prefix exist. check file existence and then give file:// as it's prefix */
+ if (util_exist_file_path(uri))
+ {
+ debug_warning("uri has no protocol-prefix. giving 'file://' by default.\n");
+ g_snprintf(data->uri, MM_MAX_URL_LEN, "file://%s", uri);
+
+ 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;
+ }
+ else
+ {
+ debug_error ("invalid uri, could not play..\n");
+ data->uri_type = MM_PLAYER_URI_TYPE_NONE;
+ }
+ }
+
+ if (data->uri_type == MM_PLAYER_URI_TYPE_NONE) {
+ ret = FALSE;
+ }
+
+ /* dump parse result */
+ debug_log("profile parsing result ---\n");
+ debug_warning("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();
+
+ return ret;
+}
+
+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)
+{
+ mm_player_t* player = (mm_player_t*)data;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, FALSE );
+
+ 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);
+
+ player->sm.by_asm_cb = 0; //should be reset here
+
+ debug_fleave();
+
+ 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;
+ int result = MM_ERROR_NONE;
+ gboolean lazy_pause = FALSE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player && player->pipeline, ASM_CB_RES_IGNORE );
+ return_val_if_fail ( player->attrs, MM_ERROR_PLAYER_INTERNAL );
+
+ 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(event_src == ASM_EVENT_SOURCE_EARJACK_UNPLUG )
+ {
+ int stop_by_asm = 0;
+
+ 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 (PLAYER_INI()->multiple_codec_supported)
+ {
+ debug_log("video conflict but, can support to use video overlay simultaneously");
+ result = _mmplayer_pause((MMHandleType)player);
+ cb_res = ASM_CB_RES_PAUSE;
+ }
+ else
+ {
+ debug_log("video conflict, can't support for multiple codec instance");
+ result = _mmplayer_unrealize((MMHandleType)player);
+ cb_res = ASM_CB_RES_STOP;
+ }
+ }
+ return cb_res;
+ }
+
+ switch(command)
+ {
+ case ASM_COMMAND_PLAY:
+ debug_warning ("Got unexpected asm command (%d)", command);
+ break;
+
+ case ASM_COMMAND_STOP: // notification case
+ {
+ debug_log("Got msg from asm to stop");
+
+ 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 = 0; // reset because no message any more from asm
+ }
+ break;
+
+ case ASM_COMMAND_PAUSE:
+ {
+ 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_MEDIA)
+ {
+ //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)
+ {
+ debug_warning("fail to set Pause state by asm");
+ cb_res = ASM_CB_RES_IGNORE;
+ break;
+ }
+ __mmplayer_undo_sound_fadedown(player);
+ }
+ 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_log ("set lazy pause timer (id=[%d], timeout=[%d ms])", player->lazy_pause_event_id, LAZY_PAUSE_TIMEOUT_MSEC);
+ }
+ else
+ {
+ //immediate pause
+ debug_log ("immediate pause");
+ result = _mmplayer_pause((MMHandleType)player);
+ }
+ 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;
+ }
+
+ if (!lazy_pause)
+ player->sm.by_asm_cb = 0;
+
+ debug_fleave();
+
+ return cb_res;
+}
+
+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);
+
+ /* 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;
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL ( player, MMPLAYER_COMMAND_CREATE );
+
+ /* construct attributes */
+ player->attrs = _mmplayer_construct_attribute(handle);
+
+ if ( !player->attrs )
+ {
+ debug_critical("Failed to construct attributes\n");
+ goto ERROR;
+ }
+
+ /* initialize gstreamer with configured parameter */
+ if ( ! __mmplayer_gstreamer_init() )
+ {
+ debug_critical("Initializing gstreamer failed\n");
+ goto ERROR;
+ }
+
+ /* 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 )
+ {
+ debug_critical("Cannot create mutex for command lock\n");
+ goto ERROR;
+ }
+
+ /* create repeat mutex */
+ player->repeat_thread_mutex = g_mutex_new();
+ if ( ! player->repeat_thread_mutex )
+ {
+ debug_critical("Cannot create repeat mutex\n");
+ goto ERROR;
+ }
+
+ /* create repeat cond */
+ player->repeat_thread_cond = g_cond_new();
+ if ( ! player->repeat_thread_cond )
+ {
+ debug_critical("Cannot create repeat cond\n");
+ goto ERROR;
+ }
+
+ /* create repeat thread */
+ player->repeat_thread =
+ g_thread_create (__mmplayer_repeat_thread, (gpointer)player, TRUE, NULL);
+ if ( ! player->repeat_thread )
+ {
+ goto ERROR;
+ }
+
+ if ( MM_ERROR_NONE != _mmplayer_initialize_video_capture(player))
+ {
+ debug_error("failed to initialize video capture\n");
+ goto ERROR;
+ }
+
+ /* 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 (MMPLAYER_IS_HTTP_PD(player))
+ {
+ player->pd_downloader = NULL;
+ player->pd_file_save_path = NULL;
+ }
+
+ /* give default value of audio effect setting */
+ player->bypass_audio_effect = TRUE;
+ player->sound.volume = MM_VOLUME_FACTOR_DEFAULT;
+ player->playback_rate = DEFAULT_PLAYBACK_RATE;
+ player->no_more_pad = TRUE;
+
+ player->play_subtitle = FALSE;
+ player->use_textoverlay = FALSE;
+
+ /* 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();
+
+ return MM_ERROR_NONE;
+
+ERROR:
+ /* free lock */
+ if ( player->fsink_lock )
+ g_mutex_free( player->fsink_lock );
+ player->fsink_lock = NULL;
+
+ /* 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 );
+
+ g_thread_join( player->repeat_thread );
+ player->repeat_thread = NULL;
+
+ g_mutex_free ( player->repeat_thread_mutex );
+ player->repeat_thread_mutex = NULL;
+
+ 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 );
+
+ if ( player->repeat_thread_cond )
+ g_cond_free ( player->repeat_thread_cond );
+
+ /* release attributes */
+ _mmplayer_deconstruct_attribute(handle);
+
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+static gboolean
+__mmplayer_gstreamer_init(void) // @
+{
+ static gboolean initialized = FALSE;
+ static const int max_argc = 50;
+ gint* argc = NULL;
+ gchar** argv = NULL;
+ GError *err = NULL;
+ int i = 0;
+
+ debug_fenter();
+
+ if ( initialized )
+ {
+ debug_log("gstreamer already initialized.\n");
+ return TRUE;
+ }
+
+ /* alloc */
+ argc = malloc( sizeof(int) );
+ argv = malloc( sizeof(gchar*) * max_argc );
+
+ if ( !argc || !argv )
+ goto ERROR;
+
+ memset( argv, 0, sizeof(gchar*) * max_argc );
+
+ /* add initial */
+ *argc = 1;
+ argv[0] = g_strdup( "mmplayer" );
+
+ /* 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 )
+ {
+ argv[*argc] = g_strdup( PLAYER_INI()->gst_param[i] );
+ (*argc)++;
+ }
+ }
+
+ /* we would not do fork for scanning plugins */
+ argv[*argc] = g_strdup("--gst-disable-registry-fork");
+ (*argc)++;
+
+ /* check disable registry scan */
+ if ( PLAYER_INI()->skip_rescan )
+ {
+ argv[*argc] = g_strdup("--gst-disable-registry-update");
+ (*argc)++;
+ }
+
+ /* check disable segtrap */
+ if ( PLAYER_INI()->disable_segtrap )
+ {
+ argv[*argc] = g_strdup("--gst-disable-segtrap");
+ (*argc)++;
+ }
+
+ debug_log("initializing gstreamer with following parameter\n");
+ debug_log("argc : %d\n", *argc);
+
+ for ( i = 0; i < *argc; i++ )
+ {
+ debug_log("argv[%d] : %s\n", i, argv[i]);
+ }
+
+
+ /* initializing gstreamer */
+ __ta__("gst_init time",
+
+ 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);
+ }
+
+ goto ERROR;
+ }
+ );
+
+ /* release */
+ for ( i = 0; i < *argc; i++ )
+ {
+ MMPLAYER_FREEIF( argv[i] );
+ }
+
+ MMPLAYER_FREEIF( argv );
+ MMPLAYER_FREEIF( argc );
+
+ /* done */
+ initialized = TRUE;
+
+ debug_fleave();
+
+ return TRUE;
+
+ERROR:
+
+ MMPLAYER_FREEIF( argv );
+ MMPLAYER_FREEIF( argc );
+
+ return FALSE;
+}
+
+int
+__mmplayer_destroy_streaming_ext(mm_player_t* player)
+{
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ if (player->pd_downloader)
+ _mmplayer_unrealize_pd_downloader((MMHandleType)player);
+
+ if (MMPLAYER_IS_HTTP_PD(player))
+ _mmplayer_destroy_pd_downloader((MMHandleType)player);
+
+ if (MMPLAYER_IS_STREAMING(player))
+ {
+ if (player->streamer)
+ {
+ __mm_player_streaming_deinitialize (player->streamer);
+ __mm_player_streaming_destroy(player->streamer);
+ player->streamer = NULL;
+ }
+ }
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_destroy(MMHandleType handle) // @
+{
+ mm_player_t* player = MM_PLAYER_CAST(handle);
+
+ debug_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 );
+
+ __mmplayer_destroy_streaming_ext(player);
+
+ /* 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 (MM_ERROR_NONE != _mmplayer_release_video_capture(player))
+ {
+ debug_error("failed to release video capture\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* withdraw asm */
+ if ( MM_ERROR_NONE != _mmplayer_asm_deregister(&player->sm) )
+ {
+ debug_error("failed to deregister asm server\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* release pipeline */
+ if ( MM_ERROR_NONE != __mmplayer_gst_destroy_pipeline( player ) )
+ {
+ debug_error("failed to destory pipeline\n");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ /* release attributes */
+ _mmplayer_deconstruct_attribute( handle );
+
+ /* release factories */
+ __mmplayer_release_factories( player );
+
+ /* release lock */
+ if ( player->fsink_lock )
+ g_mutex_free( player->fsink_lock );
+
+ if ( player->msg_cb_lock )
+ g_mutex_free( player->msg_cb_lock );
+
+ if (player->lazy_pause_event_id)
+ {
+ g_source_remove (player->lazy_pause_event_id);
+ player->lazy_pause_event_id = 0;
+ }
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+__mmplayer_realize_streaming_ext(mm_player_t* player)
+{
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ if (MMPLAYER_IS_HTTP_PD(player))
+ {
+ gboolean bret = FALSE;
+
+ player->pd_downloader = _mmplayer_create_pd_downloader();
+ if ( !player->pd_downloader )
+ {
+ debug_error ("Unable to create PD Downloader...");
+ ret = MM_ERROR_PLAYER_NO_FREE_SPACE;
+ }
+
+ bret = _mmplayer_realize_pd_downloader((MMHandleType)player, player->profile.uri, player->pd_file_save_path, player->pipeline->mainbin[MMPLAYER_M_SRC].gst);
+
+ if (FALSE == bret)
+ {
+ debug_error ("Unable to create PD Downloader...");
+ ret = MM_ERROR_PLAYER_NOT_INITIALIZED;
+ }
+ }
+
+ debug_fleave();
+ return ret;
+}
+
+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;
+
+ debug_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("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;
+ }
+
+ 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;
+ }
+
+ 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;
+
+ /* we need to update content attrs only the content has changed */
+ player->need_update_content_attrs = TRUE;
+ player->need_update_content_dur = FALSE;
+
+ /* 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();
+
+ /* then we have to rebuild factories */
+ __mmplayer_release_factories( player );
+ __mmplayer_init_factories(player);
+ }
+
+ /* realize pipeline */
+ ret = __gst_realize( player );
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("fail to realize the player.\n");
+ }
+ else
+ {
+ __mmplayer_realize_streaming_ext(player);
+ }
+
+ debug_fleave();
+
+ return ret;
+}
+
+int
+__mmplayer_unrealize_streaming_ext(mm_player_t *player)
+{
+ debug_fenter();
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* destroy can called at anytime */
+ if (player->pd_downloader && MMPLAYER_IS_HTTP_PD(player))
+ {
+ _mmplayer_unrealize_pd_downloader ((MMHandleType)player);
+ player->pd_downloader = NULL;
+ }
+
+ debug_fleave();
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_unrealize(MMHandleType hplayer) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED )
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_UNREALIZE );
+
+ __mmplayer_unrealize_streaming_ext(player);
+
+ /* unrealize pipeline */
+ ret = __gst_unrealize( 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;
+ }
+ }
+
+ debug_fleave();
+
+ return ret;
+}
+
+int
+_mmplayer_set_message_callback(MMHandleType hplayer, MMMessageCallback callback, gpointer user_param) // @
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ return __gst_set_message_callback(player, callback, user_param);
+}
+
+int
+_mmplayer_get_state(MMHandleType hplayer, int* state) // @
+{
+ mm_player_t *player = (mm_player_t*)hplayer;
+
+ return_val_if_fail(state, MM_ERROR_INVALID_ARGUMENT);
+
+ *state = MMPLAYER_CURRENT_STATE(player);
+
+ return MM_ERROR_NONE;
+}
+
+
+int
+_mmplayer_set_volume(MMHandleType hplayer, MMPlayerVolumeType volume) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ GstElement* vol_element = NULL;
+ int i = 0;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ debug_log("volume [L]=%f:[R]=%f\n",
+ volume.level[MM_VOLUME_CHANNEL_LEFT], volume.level[MM_VOLUME_CHANNEL_RIGHT]);
+
+ /* 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;
+ }
+ }
+
+ /* Save volume to handle. Currently the first array element will be saved. */
+ player->sound.volume = volume.level[0];
+
+ /* 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");
+
+ /* 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;
+ }
+
+ /* setting volume to volume element */
+ vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+
+ if ( vol_element )
+ {
+ debug_log("volume is set [%f]\n", player->sound.volume);
+ g_object_set(vol_element, "volume", player->sound.volume, NULL);
+ }
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+
+int
+_mmplayer_get_volume(MMHandleType hplayer, MMPlayerVolumeType* volume)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ int i = 0;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail( volume, MM_ERROR_INVALID_ARGUMENT );
+
+ /* returning stored volume */
+ for (i = 0; i < MM_VOLUME_CHANNEL_NUM; i++)
+ volume->level[i] = player->sound.volume;
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+
+
+int
+_mmplayer_set_mute(MMHandleType hplayer, int mute) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ GstElement* vol_element = NULL;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ debug_log("mute : %d\n", mute);
+
+ /* mute value shoud 0 or 1 */
+ if ( mute != 0 && mute != 1 )
+ {
+ debug_error("bad mute value\n");
+
+ /* FIXIT : definitly, we need _BAD_PARAM error code */
+ return 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. holding mute value\n");
+ player->sound.mute = mute;
+ return MM_ERROR_NONE;
+ }
+
+
+ vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+
+ /* NOTE : volume will only created when the bt is enabled */
+ if ( vol_element )
+ {
+ g_object_set(vol_element, "mute", mute, NULL);
+ }
+ else
+ {
+ debug_log("volume elemnet is not created. using volume in audiosink\n");
+ }
+
+ player->sound.mute = mute;
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_get_mute(MMHandleType hplayer, int* pmute) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ GstElement* vol_element = NULL;
+
+ debug_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;
+ }
+
+
+ vol_element = player->pipeline->audiobin[MMPLAYER_A_VOL].gst;
+
+ if ( vol_element )
+ {
+ g_object_get(vol_element, "mute", pmute, NULL);
+ debug_log("mute=%d\n\n", *pmute);
+ }
+ else
+ {
+ *pmute = player->sound.mute;
+ }
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_set_videostream_cb(MMHandleType hplayer, mm_player_video_stream_callback callback, void *user_param) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( callback, MM_ERROR_INVALID_ARGUMENT );
+
+ 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);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_set_audiostream_cb(MMHandleType hplayer, mm_player_audio_stream_callback callback, void *user_param) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+
+ 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);
+
+ debug_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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+
+ 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);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+
+ player->need_data_cb = callback;
+ player->buffer_cb_user_param = user_param;
+
+ debug_log("buffer need dataHandle value is %p : %p\n", player, player->need_data_cb);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_set_buffer_enough_data_cb(MMHandleType hplayer, mm_player_buffer_enough_data_callback callback, void *user_param) // @
+{
+ 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);
+
+ player->enough_data_cb = callback;
+ player->buffer_cb_user_param = user_param;
+
+ debug_log("buffer enough data cb Handle value is %p : %p\n", player, player->enough_data_cb);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player && player->pipeline, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail(callback, MM_ERROR_INVALID_ARGUMENT);
+
+ player->seek_data_cb = callback;
+ player->buffer_cb_user_param = user_param;
+
+ debug_log("buffer seek data cb Handle value is %p : %p\n", player, player->seek_data_cb);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_set_videoframe_render_error_cb(MMHandleType hplayer, mm_player_video_frame_render_error_callback callback, void *user_param) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+ return_val_if_fail ( callback, MM_ERROR_INVALID_ARGUMENT );
+
+ player->video_frame_render_error_cb = callback;
+ player->video_frame_render_error_cb_user_param = user_param;
+
+ debug_log("Video frame render error cb Handle value is %p : %p\n", player, player->video_frame_render_error_cb);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int
+__mmplayer_start_streaming_ext(mm_player_t *player)
+{
+ gint ret = MM_ERROR_NONE;
+
+ debug_fenter();
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ if (MMPLAYER_IS_HTTP_PD(player))
+ {
+ if ( !player->pd_downloader )
+ {
+ ret = __mmplayer_realize_streaming_ext(player);
+
+ if ( ret != MM_ERROR_NONE)
+ {
+ debug_error ("failed to realize streaming ext\n");
+ return ret;
+ }
+ }
+
+ 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;
+ }
+ }
+
+ debug_fleave();
+ return ret;
+}
+
+int
+_mmplayer_start(MMHandleType hplayer) // @
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ gint ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_START );
+
+ 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;
+ }
+ }
+
+ ret = __mmplayer_start_streaming_ext(player);
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to start streaming ext \n");
+ }
+
+ /* start pipeline */
+ ret = __gst_start( player );
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to start player.\n");
+ }
+
+ debug_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_post_missed_plugin(mm_player_t* player)
+{
+ MMMessageParamType msg_param;
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
+ gboolean post_msg_direct = FALSE;
+
+ debug_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 )
+ {
+ msg_param.code = MM_ERROR_PLAYER_CODEC_NOT_FOUND;
+ msg_param.data = g_strdup_printf("%s", player->unlinked_demuxer_mime);
+
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+ MMPLAYER_FREEIF(msg_param.data);
+
+ return MM_ERROR_NONE;
+ }
+
+ if (player->not_supported_codec)
+ {
+ 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;
+ }
+
+ if ( post_msg_direct )
+ {
+ MMMessageParamType msg_param;
+ memset (&msg_param, 0, sizeof(MMMessageParamType));
+
+ 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);
+ }
+ }
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+/* 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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_STOP );
+
+ /* NOTE : application should not wait for EOS after calling STOP */
+ __mmplayer_cancel_delayed_eos( player );
+
+ __mmplayer_unrealize_streaming_ext(player);
+
+ /* stop pipeline */
+ ret = __gst_stop( player );
+
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to stop player.\n");
+ }
+
+ debug_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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ /* check current state */
+ MMPLAYER_CHECK_STATE_RETURN_IF_FAIL( player, MMPLAYER_COMMAND_PAUSE );
+
+ switch (MMPLAYER_CURRENT_STATE(player))
+ {
+ case MM_PLAYER_STATE_READY:
+ {
+ /* 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 mode : %s", (async ? "async" : "sync"));
+ }
+ break;
+
+ case MM_PLAYER_STATE_PLAYING:
+ {
+ /* NOTE : store current point to overcome some bad operation
+ * ( returning zero when getting current position in paused state) of some
+ * elements
+ */
+ 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");
+
+ player->last_position = pos_msec;
+ }
+ break;
+ }
+
+ /* pause pipeline */
+ ret = __gst_pause( player, async );
+
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to pause player.\n");
+ }
+
+ debug_fleave();
+
+ return ret;
+}
+
+int
+_mmplayer_resume(MMHandleType hplayer)
+{
+ mm_player_t* player = (mm_player_t*)hplayer;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ 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 );
+
+ /* resume pipeline */
+ ret = __gst_resume( player, FALSE );
+
+ if ( ret != MM_ERROR_NONE )
+ {
+ debug_error("failed to resume player.\n");
+ }
+
+
+ debug_fleave();
+
+ return ret;
+}
+
+int
+__mmplayer_set_play_count(mm_player_t* player, gint count)
+{
+ MMHandleType attrs = 0;
+
+ debug_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");
+
+ debug_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;
+
+ debug_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,
+ 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)))
+ {
+ 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);
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+static int
+__mmplayer_set_pcm_extraction(mm_player_t* player)
+{
+ 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 );
+
+ 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);
+
+ debug_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;
+
+ debug_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;
+ }
+
+ debug_fenter();
+
+ return MM_ERROR_NONE;
+}
+
+int
+_mmplayer_set_playspeed(MMHandleType hplayer, gdouble 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;
+ debug_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 %fl\n", rate);
+
+ debug_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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ ret = __gst_set_position ( player, format, (unsigned long)position, FALSE );
+
+ debug_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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ ret = __gst_adjust_subtitle_position(player, format, position);
+
+ debug_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;
+ }
+
+ debug_log("not midi.\n");
+
+ return FALSE;
+}
+
+static gboolean
+__mmplayer_is_amr_type (gchar *str_caps)
+{
+ if ((g_strrstr(str_caps, "AMR")) ||
+ (g_strrstr(str_caps, "amr")))
+ {
+ 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;
+}
+
+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_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));
+
+ /* 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;
+ }
+ else if (__mmplayer_is_amr_type(player->type))
+ {
+ player->bypass_audio_effect = FALSE;
+ if ( (PLAYER_INI()->use_audio_effect_preset || PLAYER_INI()->use_audio_effect_custom) )
+ {
+ if ( player->audio_effect_info.effect_type == MM_AUDIO_EFFECT_TYPE_PRESET )
+ {
+ if (!_mmplayer_audio_effect_preset_apply(player, player->audio_effect_info.preset))
+ {
+ debug_msg("apply audio effect(preset:%d) setting success\n",player->audio_effect_info.preset);
+ }
+ }
+ else if ( player->audio_effect_info.effect_type == MM_AUDIO_EFFECT_TYPE_CUSTOM )
+ {
+ if (!_mmplayer_audio_effect_custom_apply(player))
+ {
+ debug_msg("apply audio 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 ) &&
+ ( 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 );
+ }
+ }
+
+DONE:
+ gst_object_unref( GST_OBJECT(pad) );
+
+ debug_fleave();
+
+ return;
+}
+
+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) // @
+{
+ 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;
+
+ debug_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, FALSE, 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 ( 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("player doesn't need [%s] so, skipping it\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);
+
+ 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) 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;
+ }
+ 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_type = 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_type = gst_caps_to_string(caps);
+
+ if ( g_strrstr( caps_type, "ST12") )
+ player->is_nv12_tiled = TRUE;
+
+ /* clean */
+ MMPLAYER_FREEIF( caps_type );
+ 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)) )
+ {
+ if (player->keep_detecting_vcodec)
+ continue;
+
+ /* Link is failed even though a supportable codec is found. */
+ __mmplayer_check_not_supported_codec(player, (gchar *)mime);
+
+ MMPLAYER_FREEIF(name_template);
+ debug_error("failed to call _close_link\n");
+ return FALSE;
+ }
+
+ MMPLAYER_FREEIF(name_template);
+ return TRUE;
+ }
+
+ gst_caps_unref(res);
+
+ break;
+ }
+ }
+
+ /* There is no any found codec. */
+ __mmplayer_check_not_supported_codec(player,(gchar *)mime);
+
+ debug_error("failed to autoplug\n");
+
+ debug_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, gchar* mime)
+{
+ 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 );
+
+ /* 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_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:
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+
+static void __mmplayer_pipeline_complete(GstElement *decodebin, gpointer data) // @
+{
+ mm_player_t* player = (mm_player_t*)data;
+
+ debug_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");
+
+ player->pipeline_is_constructed = TRUE;
+
+ if ( ( PLAYER_INI()->async_start ) &&
+ ( player->posted_msg == FALSE ) &&
+ ( player->cmd >= MMPLAYER_COMMAND_START ))
+ {
+ __mmplayer_post_missed_plugin( player );
+ }
+
+ MMPLAYER_GENERATE_DOT_IF_ENABLED ( player, "pipeline-status-complete" );
+}
+
+static gboolean __mmplayer_configure_audio_callback(mm_player_t* player)
+{
+ debug_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;
+ }
+
+ debug_fleave();
+
+ return TRUE;
+}
+
+static void
+__mmplayer_init_factories(mm_player_t* player) // @
+{
+ debug_fenter();
+
+ 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);
+
+ debug_fleave();
+}
+
+static void
+__mmplayer_release_factories(mm_player_t* player) // @
+{
+ debug_fenter();
+
+ return_if_fail ( player );
+
+ if (player->factories)
+ {
+ gst_plugin_feature_list_free (player->factories);
+ player->factories = NULL;
+ }
+
+ debug_fleave();
+}
+
+static void
+__mmplayer_release_misc(mm_player_t* player)
+{
+ int i;
+ debug_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->audio_buffer_cb = NULL;
+ player->audio_buffer_cb_user_param = NULL;
+
+ player->sent_bos = FALSE;
+ player->playback_rate = DEFAULT_PLAYBACK_RATE;
+
+ player->doing_seek = FALSE;
+
+ player->streamer = NULL;
+ 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->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;
+ player->posted_msg = FALSE;
+ player->has_many_types = FALSE;
+
+ for (i = 0; i < MM_PLAYER_STREAM_COUNT_MAX; i++)
+ {
+ player->bitrate[i] = 0;
+ player->maximum_bitrate[i] = 0;
+ }
+
+ /* clean found parsers */
+ if (player->parsers)
+ {
+ g_list_free(player->parsers);
+ player->parsers = NULL;
+ }
+
+ MMPLAYER_FREEIF(player->album_art);
+
+ /* free memory related to audio effect */
+ if(player->audio_effect_info.custom_ext_level_for_plugin)
+ {
+ free(player->audio_effect_info.custom_ext_level_for_plugin);
+ }
+
+ debug_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);
+ return NULL;
+ }
+
+ if ( ! gst_bin_add(GST_BIN(player->pipeline->mainbin[MMPLAYER_M_PIPE].gst), element))
+ {
+ debug_error("failed to add %s\n", name);
+ 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);
+
+ 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 warmup = GST_STATE_READY;
+ gboolean isvideo_decoder = FALSE;
+ guint q_max_size_time = 0;
+
+ debug_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);
+
+ if ( GST_STATE_CHANGE_FAILURE == gst_element_set_state(sinkelement, warmup) )
+ {
+ if (isvideo_decoder)
+ player->keep_detecting_vcodec = TRUE;
+
+ debug_error("failed to set %d state to %s\n", warmup, GST_ELEMENT_NAME( sinkelement ));
+ 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 = 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;
+
+ 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 (parser_name)
+ {
+ parser = __mmplayer_element_create_and_link(player, srcpad, parser_name);
+
+ 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);
+
+ 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;
+
+ 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;
+
+ 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);
+ }
+
+ 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*/
+ {
+ 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;
+ }
+
+ /* 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(name, "video"))
+ {
+ player->can_support_codec ^= FOUND_PLUGIN_VIDEO;
+ }
+ else if (strstr(name, "audio"))
+ {
+ player->can_support_codec ^= FOUND_PLUGIN_AUDIO;
+ }
+ goto ERROR;
+ }
+
+ if (strstr(name, "video"))
+ {
+ player->videodec_linked = 1;
+ debug_msg("player->videodec_linked set to 1\n");
+
+ }
+ else if (strstr(name, "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_S_BUFFER].gst)
+ {
+ debug_log("creating http streaming buffering queue\n");
+
+ queue = gst_element_factory_make("queue2", "http_streaming_buffer");
+ if ( ! queue )
+ {
+ debug_critical ( "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_S_BUFFER].id = MMPLAYER_M_S_BUFFER;
+ mainbin[MMPLAYER_M_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);
+ }
+ }
+
+ __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_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 ("has_many_types for this caps [%s]\n", gst_caps_to_string(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, "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, "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, "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;
+ }
+
+ debug_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;
+
+ debug_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 );
+
+ debug_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;
+
+ debug_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);
+ }
+ }
+
+ debug_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;
+
+ debug_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;
+
+ /* 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);
+
+ /* validate all */
+ if ( mmf_attrs_commit ( player->attrs ) )
+ {
+ debug_error("failed to update attributes");
+ return;
+ }
+ }
+ 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);
+
+ debug_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;
+
+ debug_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);
+ }
+
+ }
+ }
+
+ debug_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;
+
+ debug_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) ;
+ 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) ;
+
+ 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);
+
+ debug_fleave();
+
+ return FALSE;
+}
+
+
+gboolean
+__mmplayer_check_subtitle( mm_player_t* player )
+{
+ MMHandleType attrs = 0;
+ char *subtitle_uri = NULL;
+
+ debug_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));
+
+ debug_fleave();
+
+ return TRUE;
+}
+
+static gboolean
+__mmplayer_can_extract_pcm( mm_player_t* player )
+{
+ MMHandleType attrs = 0;
+ gboolean is_drm = FALSE;
+ gboolean sound_extraction = FALSE;
+
+ debug_fenter();
+
+ 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 */
+ 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("pcm extraction param.. is drm = %d, extraction mode = %d", is_drm, sound_extraction);
+ return FALSE;
+ }
+
+ debug_fleave();
+
+ return TRUE;
+}
+
+static gboolean
+__mmplayer_handle_gst_error ( mm_player_t* player, GstMessage * message, GError* error )
+{
+ MMMessageParamType msg_param;
+ gchar *msg_src_element;
+
+ debug_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);
+ }
+
+ /* post error to application */
+ if ( ! player->posted_msg )
+ {
+ if (msg_param.code == MM_MESSAGE_DRM_NOT_AUTHORIZED )
+ {
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_DRM_NOT_AUTHORIZED, NULL );
+ }
+ else
+ {
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+ }
+
+ /* don't post more if one was sent already */
+ player->posted_msg = TRUE;
+ }
+ else
+ {
+ debug_log("skip error post because it's sent already.\n");
+ }
+
+ debug_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;
+
+ debug_fenter();
+
+ return_val_if_fail ( player, FALSE );
+ return_val_if_fail ( message, FALSE );
+
+ s = malloc( sizeof(GstStructure) );
+ 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;
+ case MMPLAYER_STREAMING_ERROR_BAD_REQUEST:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_BAD_REQUEST;
+ break;
+ case MMPLAYER_STREAMING_ERROR_NOT_AUTHORIZED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_NOT_AUTHORIZED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_PAYMENT_REQUIRED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_PAYMENT_REQUIRED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_FORBIDDEN:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_FORBIDDEN;
+ break;
+ case MMPLAYER_STREAMING_ERROR_CONTENT_NOT_FOUND:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_CONTENT_NOT_FOUND;
+ break;
+ case MMPLAYER_STREAMING_ERROR_METHOD_NOT_ALLOWED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_METHOD_NOT_ALLOWED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_NOT_ACCEPTABLE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_NOT_ACCEPTABLE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_PROXY_AUTHENTICATION_REQUIRED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_PROXY_AUTHENTICATION_REQUIRED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_SERVER_TIMEOUT:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_SERVER_TIMEOUT;
+ break;
+ case MMPLAYER_STREAMING_ERROR_GONE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_GONE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_LENGTH_REQUIRED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_LENGTH_REQUIRED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_PRECONDITION_FAILED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_PRECONDITION_FAILED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_REQUEST_ENTITY_TOO_LARGE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_REQUEST_ENTITY_TOO_LARGE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_REQUEST_URI_TOO_LARGE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_REQUEST_URI_TOO_LARGE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_UNSUPPORTED_MEDIA_TYPE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_UNSUPPORTED_MEDIA_TYPE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_PARAMETER_NOT_UNDERSTOOD:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_PARAMETER_NOT_UNDERSTOOD;
+ break;
+ case MMPLAYER_STREAMING_ERROR_CONFERENCE_NOT_FOUND:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_CONFERENCE_NOT_FOUND;
+ break;
+ case MMPLAYER_STREAMING_ERROR_NOT_ENOUGH_BANDWIDTH:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_NOT_ENOUGH_BANDWIDTH;
+ break;
+ case MMPLAYER_STREAMING_ERROR_NO_SESSION_ID:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_NO_SESSION_ID;
+ break;
+ case MMPLAYER_STREAMING_ERROR_METHOD_NOT_VALID_IN_THIS_STATE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_METHOD_NOT_VALID_IN_THIS_STATE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_HEADER_FIELD_NOT_VALID_FOR_SOURCE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_HEADER_FIELD_NOT_VALID_FOR_SOURCE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_INVALID_RANGE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_INVALID_RANGE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_PARAMETER_IS_READONLY:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_PARAMETER_IS_READONLY;
+ break;
+ case MMPLAYER_STREAMING_ERROR_AGGREGATE_OP_NOT_ALLOWED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_AGGREGATE_OP_NOT_ALLOWED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_ONLY_AGGREGATE_OP_ALLOWED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_ONLY_AGGREGATE_OP_ALLOWED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_BAD_TRANSPORT:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_BAD_TRANSPORT;
+ break;
+ case MMPLAYER_STREAMING_ERROR_DESTINATION_UNREACHABLE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_DESTINATION_UNREACHABLE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_INTERNAL_SERVER_ERROR:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_INTERNAL_SERVER_ERROR;
+ break;
+ case MMPLAYER_STREAMING_ERROR_NOT_IMPLEMENTED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_NOT_IMPLEMENTED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_BAD_GATEWAY:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_BAD_GATEWAY;
+ break;
+ case MMPLAYER_STREAMING_ERROR_SERVICE_UNAVAILABLE:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_SERVICE_UNAVAILABLE;
+ break;
+ case MMPLAYER_STREAMING_ERROR_GATEWAY_TIME_OUT:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_GATEWAY_TIME_OUT;
+ break;
+ case MMPLAYER_STREAMING_ERROR_RTSP_VERSION_NOT_SUPPORTED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_RTSP_VERSION_NOT_SUPPORTED;
+ break;
+ case MMPLAYER_STREAMING_ERROR_OPTION_NOT_SUPPORTED:
+ msg_param.code = MM_ERROR_PLAYER_STREAMING_OPTION_NOT_SUPPORTED;
+ break;
+ default:
+ 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->posted_msg )
+ {
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_ERROR, &msg_param );
+
+ /* don't post more if one was sent already */
+ player->posted_msg = TRUE;
+ }
+ else
+ {
+ debug_log("skip error post because it's sent already.\n");
+ }
+
+ debug_fleave();
+
+ return TRUE;
+
+}
+
+static gint
+__gst_handle_core_error( mm_player_t* player, int code )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ return_val_if_fail( player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ switch ( code )
+ {
+ 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;
+ }
+
+ debug_fleave();
+
+ return trans_err;
+}
+
+static gint
+__gst_handle_library_error( mm_player_t* player, int code )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ debug_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;
+ }
+
+ debug_fleave();
+
+ return trans_err;
+}
+
+
+static gint
+__gst_handle_resource_error( mm_player_t* player, int code )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ debug_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 ) )
+ {
+ 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;
+ }
+
+ debug_fleave();
+
+ return trans_err;
+}
+
+
+static gint
+__gst_handle_stream_error( mm_player_t* player, GError* error, GstMessage * message )
+{
+ gint trans_err = MM_ERROR_NONE;
+
+ debug_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:
+ trans_err = __gst_transform_gsterror( player, message, error );
+ break;
+
+ 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:
+ default:
+ trans_err = MM_ERROR_PLAYER_INVALID_STREAM;
+ break;
+ }
+
+ debug_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;
+
+ debug_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);
+
+
+ switch ( error->code )
+ {
+ case GST_STREAM_ERROR_DECODE:
+ {
+ /* 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;
+ }
+ break;
+
+ case GST_STREAM_ERROR_WRONG_TYPE:
+ {
+ return MM_ERROR_PLAYER_NOT_SUPPORTED_FORMAT;
+ }
+ break;
+
+ case GST_STREAM_ERROR_FAILED:
+ {
+ /* Decoder Custom Message */
+ if ( strstr(error->message, "ongoing") )
+ {
+ if ( strcasestr(klass, "audio") )
+ {
+ 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 ( strcasestr(klass, "video") )
+ {
+ 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_INVALID_STREAM;
+ }
+ break;
+
+ case GST_STREAM_ERROR_TYPE_NOT_FOUND:
+ {
+ goto CODEC_NOT_FOUND;
+ }
+ 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);
+ 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;
+}
+
+static void
+__mmplayer_post_delayed_eos( mm_player_t* player, int delay_in_ms )
+{
+ debug_fenter();
+
+ return_if_fail( player );
+
+ /* cancel if existing */
+ __mmplayer_cancel_delayed_eos( player );
+
+
+ /* post now if delay is zero */
+ if ( delay_in_ms == 0 || player->is_sound_extraction)
+ {
+ 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;
+ }
+
+ /* 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 );
+
+
+ /* 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 );
+ }
+
+ debug_fleave();
+}
+
+static void
+__mmplayer_cancel_delayed_eos( mm_player_t* player )
+{
+ debug_fenter();
+
+ return_if_fail( player );
+
+ if ( player->eos_timer )
+ {
+ g_source_remove( player->eos_timer );
+ }
+
+ player->eos_timer = 0;
+
+ debug_fleave();
+
+ return;
+}
+
+static gboolean
+__mmplayer_eos_timer_cb(gpointer u_data)
+{
+ mm_player_t* player = NULL;
+ player = (mm_player_t*) u_data;
+
+ debug_fenter();
+
+ return_val_if_fail( player, FALSE );
+
+ /* posting eos */
+ MMPLAYER_POST_MSG( player, MM_MESSAGE_END_OF_STREAM, NULL );
+
+ /* cleare timer id */
+ player->eos_timer = 0;
+
+ debug_fleave();
+
+ /* 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;
+
+ debug_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);
+
+ 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);
+ }
+
+ debug_fleave();
+
+ return;
+}
+
+
+static gboolean
+__mmplayer_link_decoder( mm_player_t* player, GstPad *srcpad)
+{
+ const gchar* name = NULL;
+ GstStructure* str = NULL;
+ GstCaps* srccaps = NULL;
+
+ debug_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 );
+
+ debug_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;
+
+ debug_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;
+ }
+ }
+ if (strstr(name, "audio"))
+ {
+ if(player->audiosink_linked)
+ {
+ debug_msg("Audio Sink already linked\n");
+ return FALSE;
+ }
+ }
+ if (strstr(name, "text"))
+ {
+ if(player->textsink_linked)
+ {
+ debug_msg("Text Sink already linked\n");
+ return FALSE;
+ }
+ }
+
+ gst_caps_unref( srccaps );
+
+ debug_fleave();
+
+ return TRUE;
+ //return (!player->videosink_linked || !player->audiosink_linked);
+
+ERROR:
+ if ( srccaps )
+ gst_caps_unref( srccaps );
+
+ return FALSE;
+}
+
+
+/* sending event to one of sinkelements */
+static gboolean
+__gst_send_event_to_sink( mm_player_t* player, GstEvent* event )
+{
+ GstEvent * event2 = NULL;
+ GList *sinks = NULL;
+ gboolean res = FALSE;
+
+ debug_fenter();
+
+ return_val_if_fail( player, FALSE );
+ return_val_if_fail ( event, FALSE );
+
+ if ( player->play_subtitle && !player->use_textoverlay)
+ event2 = gst_event_copy((const GstEvent *)event);
+
+ sinks = player->sink_elements;
+ while (sinks)
+ {
+ 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);
+ }
+
+ /* 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 = GST_ELEMENT_CAST (player->pipeline->textbin[MMPLAYER_T_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);
+ }
+ }
+
+ 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);
+
+ debug_fleave();
+}
+
+static void
+__mmplayer_del_sink( mm_player_t* player, GstElement* sink )
+{
+ debug_fenter();
+
+ return_if_fail ( player );
+ return_if_fail ( sink );
+
+ player->sink_elements =
+ g_list_remove(player->sink_elements, sink);
+
+ debug_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 )
+{
+ GstEvent* event = NULL;
+ gboolean result = FALSE;
+
+ debug_fenter();
+
+ return_val_if_fail( player, FALSE );
+
+ event = gst_event_new_seek (rate, format, flags, cur_type,
+ cur, stop_type, stop);
+
+ result = __gst_send_event_to_sink( player, event );
+
+ debug_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)
+{
+ GList* sig_list = player->signals;
+ MMPlayerSignalItem* item = NULL;
+
+ debug_fenter();
+
+ return_if_fail( player );
+
+ for ( ; sig_list; sig_list = sig_list->next )
+ {
+ item = sig_list->data;
+
+ if ( item && item->obj && GST_IS_ELEMENT(item->obj) )
+ {
+ debug_log("checking signal connection : [%lud] from [%s]\n", item->sig, GST_OBJECT_NAME( item->obj ));
+
+ if ( g_signal_handler_is_connected ( item->obj, item->sig ) )
+ {
+ debug_log("signal disconnecting : [%lud] from [%s]\n", item->sig, GST_OBJECT_NAME( item->obj ));
+ g_signal_handler_disconnect ( item->obj, item->sig );
+ }
+ }
+
+ MMPLAYER_FREEIF( item );
+
+ }
+ g_list_free ( player->signals );
+ player->signals = NULL;
+
+ debug_fleave();
+
+ return;
+}
+
+
+/* Note : if silent is true, then subtitle would not be displayed. :*/
+int _mmplayer_set_subtitle_silent (MMHandleType hplayer, int silent)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ debug_fenter();
+
+ /* check player handle */
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ player->is_subtitle_off = silent;
+
+ debug_log("subtitle is %s.\n", player->is_subtitle_off ? "ON" : "OFF");
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+
+int _mmplayer_get_subtitle_silent (MMHandleType hplayer, int* silent)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+
+ debug_fenter();
+
+ /* check player handle */
+ return_val_if_fail(player, MM_ERROR_PLAYER_NOT_INITIALIZED );
+
+ *silent = player->is_subtitle_off;
+
+ debug_log("subtitle is %s.\n", silent ? "ON" : "OFF");
+
+ debug_fleave();
+
+ return MM_ERROR_NONE;
+}
+
+int _mmplayer_get_track_count(MMHandleType hplayer, MMPlayerTrackType track_type, int *count)
+{
+ mm_player_t* player = (mm_player_t*) hplayer;
+ MMHandleType attrs = 0;
+ int ret = MM_ERROR_NONE;
+
+ debug_fenter();
+
+ /* 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);
+
+ attrs = MMPLAYER_GET_ATTRS(player);
+ if ( !attrs )
+ {
+ debug_error("cannot get content attribute");
+ return MM_ERROR_PLAYER_INTERNAL;
+ }
+
+ switch (track_type)
+ {
+ 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;
+ }
+
+ debug_log ("%d track num is %d\n", track_type, *count);
+
+ debug_fleave();
+
+ return ret;
+}
+
+
+
+const gchar *
+__get_state_name ( int state )
+{
+ switch ( state )
+ {
+ case MM_PLAYER_STATE_NULL:
+ return "NULL";
+ case MM_PLAYER_STATE_READY:
+ return "READY";
+ case MM_PLAYER_STATE_PAUSED:
+ return "PAUSED";
+ case MM_PLAYER_STATE_PLAYING:
+ return "PLAYING";
+ case MM_PLAYER_STATE_NONE:
+ return "NONE";
+ default:
+ return "INVAID";
+ }
+}
+gboolean
+__is_rtsp_streaming ( mm_player_t* player )
+{
+ return_val_if_fail ( player, FALSE );
+
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_URL_RTSP ) ? TRUE : FALSE;
+}
+
+static gboolean
+__is_http_streaming ( mm_player_t* player )
+{
+ return_val_if_fail ( player, FALSE );
+
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_URL_HTTP ) ? TRUE : FALSE;
+}
+
+static gboolean
+__is_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;
+}
+
+gboolean
+__is_live_streaming ( mm_player_t* player )
+{
+ return_val_if_fail ( player, FALSE );
+
+ return ( __is_rtsp_streaming ( player ) && player->streaming_type == STREAMING_SERVICE_LIVE ) ? TRUE : FALSE;
+}
+
+static gboolean
+__is_http_live_streaming( mm_player_t* player )
+{
+ return_val_if_fail( player, FALSE );
+
+ return ( player->profile.uri_type == MM_PLAYER_URI_TYPE_HLS ) ? TRUE : FALSE;
+}
+
+static gboolean
+__is_http_progressive_down(mm_player_t* player)
+{
+ return_val_if_fail( player, FALSE );
+
+ return ((player->pd_mode) ? TRUE:FALSE);
+}
projects / platform / core / multimedia / libmm-player.git / commitdiff