GstElement *parent = NULL;
MMPLAYER_RETURN_VAL_IF_FAIL(player && player->pipeline, FALSE);
-
- /* if we have no fakesink. this meas we are using decodebin which doesn'
- t need to add extra fakesink */
- MMPLAYER_RETURN_VAL_IF_FAIL(fakesink, TRUE);
+ MMPLAYER_RETURN_VAL_IF_FAIL(fakesink && fakesink->gst, TRUE);
/* lock */
MMPLAYER_FSINK_LOCK(player);
- if (!fakesink->gst)
- goto ERROR;
-
/* get parent of fakesink */
parent = (GstElement *)gst_object_get_parent((GstObject *)fakesink->gst);
if (!parent) {
GList *bucket = element_bucket;
mmplayer_gst_element_t *element = NULL;
mmplayer_gst_element_t *prv_element = NULL;
+ GstElement *tee_element = NULL;
gint successful_link_count = 0;
MMPLAYER_FENTER();
if (element && element->gst) {
if (prv_element && prv_element->gst) {
+ if (strstr(GST_ELEMENT_NAME(element->gst), "audio-tee-queue") && strcmp(GST_ELEMENT_NAME(prv_element->gst), "audio-tee")) {
+ if (tee_element) {
+ prv_element->gst = tee_element;
+ } else {
+ LOGD("failed to make new audio branch - linking [%s] to [%s] is not supported",
+ GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
+ GST_ELEMENT_NAME(GST_ELEMENT(element->gst)));
+ return -1;
+ }
+ }
if (gst_element_link(GST_ELEMENT(prv_element->gst), GST_ELEMENT(element->gst))) {
LOGD("linking [%s] to [%s] success",
GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
GST_ELEMENT_NAME(GST_ELEMENT(element->gst)));
successful_link_count++;
+ if (!strcmp(GST_ELEMENT_NAME(prv_element->gst), "audio-tee")) {
+ LOGD("keep audio-tee element for next audio pipeline branch");
+ tee_element = prv_element->gst;
+ }
} else {
LOGD("linking [%s] to [%s] failed",
GST_ELEMENT_NAME(GST_ELEMENT(prv_element->gst)),
return;
}
-/**
- * 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_sink_bin
- */
-/* macro for code readability. just for sinkbin-creation functions */
-#define MMPLAYER_CREATE_ELEMENT(x_bin, x_id, x_factory, x_name, x_add_bucket, x_player) \
- do {\
- x_bin[x_id].id = x_id;\
- x_bin[x_id].gst = gst_element_factory_make(x_factory, x_name);\
- if (!x_bin[x_id].gst) {\
- LOGE("failed to create %s", x_factory);\
- goto ERROR;\
- } else {\
- if (x_player->ini.set_dump_element_flag)\
- __mmplayer_add_dump_buffer_probe(x_player, x_bin[x_id].gst);\
- } \
- if (x_add_bucket)\
- element_bucket = g_list_append(element_bucket, &x_bin[x_id]);\
- } while (0);
-
void
__mmplayer_audio_stream_clear_buffer(mmplayer_t *player, gboolean send_all)
{
g_object_set(sink, "sync", TRUE, NULL);
g_object_set(sink, "signal-handoffs", TRUE, NULL);
+ /* keep the first sink reference only */
+ if (!audiobin[MMPLAYER_A_SINK].gst) {
+ audiobin[MMPLAYER_A_SINK].id = MMPLAYER_A_SINK;
+ audiobin[MMPLAYER_A_SINK].gst = sink;
+ }
+
gst_element_set_state(sink, GST_STATE_PAUSED);
gst_element_set_state(queue, GST_STATE_PAUSED);
G_CALLBACK(__mmplayer_audio_stream_decoded_render_cb),
(gpointer)player);
+ __mmplayer_add_sink(player, sink);
+
MMPLAYER_FLEAVE();
return;
}
void
-__mmplayer_gst_set_pulsesink_property(mmplayer_t *player, MMHandleType attrs)
+__mmplayer_gst_set_pulsesink_property(mmplayer_t *player)
{
#define MAX_PROPS_LEN 128
gint latency_mode = 0;
MMPLAYER_FENTER();
MMPLAYER_RETURN_IF_FAIL(player && player->pipeline && player->pipeline->audiobin);
- mm_attrs_get_int_by_name(attrs, "sound_stream_index", &stream_id);
- mm_attrs_get_string_by_name(attrs, "sound_stream_type", &stream_type);
+ mm_attrs_get_int_by_name(player->attrs, "sound_stream_index", &stream_id);
+ mm_attrs_get_string_by_name(player->attrs, "sound_stream_type", &stream_type);
if (!stream_type) {
LOGE("stream_type is null.");
gst_structure_free(props);
}
- mm_attrs_get_int_by_name(attrs, "sound_latency_mode", &latency_mode);
+ mm_attrs_get_int_by_name(player->attrs, "sound_latency_mode", &latency_mode);
switch (latency_mode) {
case AUDIO_LATENCY_MODE_LOW:
}
static int
-__mmplayer_gst_fill_audio_bucket(mmplayer_t *player, GList **bucket)
+__mmplayer_gst_make_audio_playback_sink(mmplayer_t *player, GList **bucket)
{
mmplayer_gst_element_t *audiobin = NULL;
- MMHandleType attrs = 0;
- GList *element_bucket = NULL;
- GstCaps *acaps = NULL;
GstPad *sink_pad = NULL;
+ GstCaps *acaps = NULL;
+ gint channels = 0;
int pitch_control = 0;
double pitch_value = 1.0;
player->pipeline->audiobin, MM_ERROR_PLAYER_NOT_INITIALIZED);
audiobin = player->pipeline->audiobin;
- attrs = MMPLAYER_GET_ATTRS(player);
- if (player->build_audio_offload) { /* skip all the audio filters */
- LOGD("create audio offload sink : %s", player->ini.audio_offload_sink_element);
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, player->ini.audio_offload_sink_element, "audiosink", TRUE, player);
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "sync", TRUE, NULL);
- __mmplayer_add_sink(player, audiobin[MMPLAYER_A_SINK].gst);
- goto DONE;
- }
+ LOGD("make element for normal audio playback");
- /* pitch */
+ /* audio bin structure for playback. {} means optional.
+ optional : pitch, audioeq, custom audioeq, openalsink for 360 audio content
+
+ * src - ... - {aconv - pitch} - aconv - rgvolume - resample - volume -
+ {audioeq} - {custom audioeq} - pulsesink or {aconv - capsfilter - openalsink}
+ */
+
+ /* for pitch control */
mm_attrs_multiple_get(player->attrs, NULL,
MM_PLAYER_PITCH_CONTROL, &pitch_control,
MM_PLAYER_PITCH_VALUE, &pitch_value,
gst_object_unref(factory);
/* converter */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV_PITCH, "audioconvert", "audio convert pitch", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV_PITCH, "audioconvert", "audio convert pitch", *bucket, player);
/* pitch */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_PITCH, "pitch", "audio pitch", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_PITCH, "pitch", "audio pitch", *bucket, player);
g_object_set(G_OBJECT(audiobin[MMPLAYER_A_PITCH].gst), "pitch", (gdouble)pitch_value, NULL);
} else {
LOGW("there is no pitch element");
}
/* converter */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV, "audioconvert", "audio converter", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV, "audioconvert", "audio converter", *bucket, player);
/* replaygain volume */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_RGVOL, "rgvolume", "audio rgvolume", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_RGVOL, "rgvolume", "audio rgvolume", *bucket, player);
if (player->sound.rg_enable)
g_object_set(G_OBJECT(audiobin[MMPLAYER_A_RGVOL].gst), "enable-rgvolume", TRUE, NULL);
else
g_object_set(G_OBJECT(audiobin[MMPLAYER_A_RGVOL].gst), "enable-rgvolume", FALSE, NULL);
/* resampler */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_RESAMPLER, player->ini.audioresampler_element, "audio resampler", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_RESAMPLER, player->ini.audioresampler_element, "audio resampler", *bucket, player);
- if (player->audio_decoded_cb) { /* pcm extraction only, no sound output */
- gchar *dst_format = NULL;
- int dst_len = 0;
- int dst_samplerate = 0;
- int dst_channels = 0;
- GstCaps *caps = NULL;
- char *caps_str = NULL;
+ /* for logical volume control */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_VOL, "volume", "volume", *bucket, player);
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_VOL].gst), "volume", player->sound.volume, NULL);
- /* get conf. values */
- mm_attrs_multiple_get(player->attrs, NULL,
- "pcm_audioformat", &dst_format, &dst_len,
- "pcm_extraction_samplerate", &dst_samplerate,
- "pcm_extraction_channels", &dst_channels,
- NULL);
+ if (player->sound.mute) {
+ LOGD("mute enabled");
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_VOL].gst), "mute", player->sound.mute, NULL);
+ }
- LOGD("required pcm format - format: %s(%d), samplerate : %d, channel: %d", dst_format, dst_len, dst_samplerate, dst_channels);
- if (dst_format == NULL || dst_len == 0 || dst_samplerate == 0 || dst_channels == 0) {
- mm_attrs_multiple_get(player->attrs, NULL,
- "content_audio_format", &dst_format, &dst_len, /* get string and len */
- "content_audio_samplerate", &dst_samplerate,
- "content_audio_channels", &dst_channels,
- NULL);
+ mm_attrs_get_int_by_name(player->attrs, "content_audio_channels", &channels);
- LOGD("decoded pcm format - format: %s(%d), samplerate : %d, channel: %d", dst_format, dst_len, dst_samplerate, dst_channels);
+ /* audio effect element. if audio effect is enabled */
+ if ((strcmp(player->ini.audioeffect_element, ""))
+ && (channels <= 2)
+ && (player->ini.use_audio_effect_preset || player->ini.use_audio_effect_custom)) {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER, player->ini.audioeffect_element, "audio effect filter", *bucket, player);
- /* If there is no enough information, set it to platform default value. */
- if (dst_format == NULL || util_convert_audio_pcm_str_to_media_format_mime(dst_format) == MEDIA_FORMAT_MAX) {
- LOGD("set platform default format");
- dst_format = DEFAULT_PCM_OUT_FORMAT;
+ LOGD("audio effect config. bypass = %d, effect type = %d", player->bypass_audio_effect, player->audio_effect_info.effect_type);
+
+ 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_CUSTOM) {
+ if (!_mmplayer_audio_effect_custom_apply(player))
+ LOGI("apply audio effect(custom) setting success");
}
- if (dst_samplerate <= 0) dst_samplerate = DEFAULT_PCM_OUT_SAMPLERATE;
- if (dst_channels <= 0) dst_channels = DEFAULT_PCM_OUT_CHANNEL;
}
- /* capsfilter */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_DEFAULT, "capsfilter", "audio capsfilter", TRUE, player);
- caps = gst_caps_new_simple("audio/x-raw",
- "format", G_TYPE_STRING, dst_format,
- "rate", G_TYPE_INT, dst_samplerate,
- "channels", G_TYPE_INT, dst_channels,
- NULL);
+ if ((strcmp(player->ini.audioeffect_element_custom, ""))
+ && (player->set_mode.rich_audio)) {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER_SEC, player->ini.audioeffect_element_custom, "audio effect filter custom", *bucket, player);
+ }
+ }
- caps_str = gst_caps_to_string(caps);
- LOGD("new caps : %s", caps_str);
+ /* create audio sink */
+ LOGD("spherical %d, channels %d, ambisonic type %d, format %d, order %d",
+ player->is_content_spherical, channels, player->video360_metadata.ambisonic_type,
+ player->video360_metadata.ambisonic_format, player->video360_metadata.ambisonic_order);
- g_object_set(GST_ELEMENT(audiobin[MMPLAYER_A_CAPS_DEFAULT].gst), "caps", caps, NULL);
+ /* Note: qtdemux converts audio metadata defaults to openalsink defaults. */
+ if (player->is_360_feature_enabled &&
+ player->is_content_spherical &&
+ channels == 4 &&
+ player->video360_metadata.ambisonic_type == MMFILE_AMBISONIC_TYPE_PERIPHONIC &&
+ player->video360_metadata.ambisonic_format == MMFILE_AMBISONIC_FORMAT_AMB &&
+ player->video360_metadata.ambisonic_order == MMFILE_AMBISONIC_ORDER_FOA) {
- /* clean */
- gst_caps_unref(caps);
- MMPLAYER_FREEIF(caps_str);
+ strncpy(player->ini.audiosink_element, "openalsink", PLAYER_INI_MAX_STRLEN - 1);
- if (player->audio_extract_opt & MM_PLAYER_AUDIO_EXTRACT_DEINTERLEAVE) {
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_DEINTERLEAVE, "deinterleave", "deinterleave", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV_BFORMAT, "audioconvert", "audio-converter-bformat", *bucket, player);
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_DEINTERLEAVE].gst), "keep-positions", TRUE, NULL);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_360, "capsfilter", "audio-caps-filter", *bucket, player);
+ acaps = gst_caps_from_string(SPATIAL_AUDIO_CAPS);
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_CAPS_360].gst), "caps", acaps, NULL);
+ gst_caps_unref(acaps);
- /* raw pad handling signal, audiosink will be added after getting signal */
- __mmplayer_add_signal_connection(player, G_OBJECT(audiobin[MMPLAYER_A_DEINTERLEAVE].gst),
- MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added", G_CALLBACK(__mmplayer_gst_audio_deinterleave_pad_added), (gpointer)player);
- } else {
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, "fakesink", "audiosink", TRUE, player);
- if (!(player->audio_extract_opt & MM_PLAYER_AUDIO_EXTRACT_NO_SYNC_WITH_CLOCK))
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "sync", TRUE, NULL);
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "signal-handoffs", TRUE, NULL);
-
- __mmplayer_add_signal_connection(player,
- G_OBJECT(audiobin[MMPLAYER_A_SINK].gst),
- MM_PLAYER_SIGNAL_TYPE_AUDIOBIN,
- "handoff",
- G_CALLBACK(__mmplayer_audio_stream_decoded_render_cb),
- (gpointer)player);
- }
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, "openalsink", "audiosink", *bucket, player);
+
+ player->is_openal_plugin_used = TRUE;
} else {
- /* normal playback */
- gint channels = 0;
+ if (player->is_360_feature_enabled && player->is_content_spherical)
+ LOGW("Audio track isn't of the ambisonic type and can't be played back as a spatial sound.");
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, player->ini.audiosink_element, "audiosink", *bucket, player);
+ }
- /* for logical volume control */
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_VOL, "volume", "volume", TRUE, player);
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_VOL].gst), "volume", player->sound.volume, NULL);
+ if ((MMPLAYER_IS_RTSP_STREAMING(player)) ||
+ (player->videodec_linked && player->ini.use_system_clock)) {
+ LOGD("system clock will be used.");
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "provide-clock", FALSE, NULL);
+ }
- if (player->sound.mute) {
- LOGD("mute enabled");
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_VOL].gst), "mute", player->sound.mute, NULL);
- }
+ if (g_strrstr(player->ini.audiosink_element, "pulsesink"))
+ __mmplayer_gst_set_pulsesink_property(player);
+ else if (g_strrstr(player->ini.audiosink_element, "openalsink"))
+ __mmplayer_gst_set_openalsink_property(player);
- mm_attrs_get_int_by_name(player->attrs, "content_audio_channels", &channels);
+ /* qos on */
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "qos", TRUE, NULL); /* qos on */
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "slave-method", GST_AUDIO_BASE_SINK_SLAVE_NONE, NULL);
- /* audio effect element. if audio effect is enabled */
- if ((strcmp(player->ini.audioeffect_element, ""))
- && (channels <= 2)
- && (player->ini.use_audio_effect_preset || player->ini.use_audio_effect_custom)) {
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER, player->ini.audioeffect_element, "audio effect filter", TRUE, player);
+ sink_pad = gst_element_get_static_pad(audiobin[MMPLAYER_A_SINK].gst, "sink");
+ __mmplayer_add_signal_connection(player, G_OBJECT(sink_pad), MM_PLAYER_SIGNAL_TYPE_AUDIOBIN,
+ "notify::caps", G_CALLBACK(__mmplayer_gst_caps_notify_cb), (gpointer)player);
+ gst_object_unref(GST_OBJECT(sink_pad));
- LOGD("audio effect config. bypass = %d, effect type = %d", player->bypass_audio_effect, player->audio_effect_info.effect_type);
+ __mmplayer_add_sink(player, audiobin[MMPLAYER_A_SINK].gst);
- 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_CUSTOM) {
- if (!_mmplayer_audio_effect_custom_apply(player))
- LOGI("apply audio effect(custom) setting success");
- }
- }
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
- if ((strcmp(player->ini.audioeffect_element_custom, ""))
- && (player->set_mode.rich_audio))
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_FILTER_SEC, player->ini.audioeffect_element_custom, "audio effect filter custom", TRUE, player);
- }
+ERROR: /* MMPLAYER_CREATE_ELEMENT */
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+}
- /* create audio sink */
- LOGD("360 spherical %d, channels %d, ambisonic type %d, format %d, order %d",
- player->is_content_spherical, channels, player->video360_metadata.ambisonic_type,
- player->video360_metadata.ambisonic_format, player->video360_metadata.ambisonic_order);
+static int
+__mmplayer_gst_make_audio_extract_sink(mmplayer_t *player, GList **bucket)
+{
+ mmplayer_gst_element_t *audiobin = NULL;
+ enum audio_element_id extract_sink_id = MMPLAYER_A_SINK;
- /* Note: qtdemux converts audio metadata defaults to openalsink defaults. */
- if (player->is_360_feature_enabled &&
- player->is_content_spherical &&
- channels == 4 &&
- player->video360_metadata.ambisonic_type == MMFILE_AMBISONIC_TYPE_PERIPHONIC &&
- player->video360_metadata.ambisonic_format == MMFILE_AMBISONIC_FORMAT_AMB &&
- player->video360_metadata.ambisonic_order == MMFILE_AMBISONIC_ORDER_FOA) {
+ gchar *dst_format = NULL;
+ int dst_len = 0;
+ int dst_samplerate = 0;
+ int dst_channels = 0;
+ GstCaps *caps = NULL;
+ char *caps_str = NULL;
+
+ MMPLAYER_FENTER();
+ MMPLAYER_RETURN_VAL_IF_FAIL(player && player->pipeline &&
+ player->pipeline->audiobin, MM_ERROR_PLAYER_NOT_INITIALIZED);
- strncpy(player->ini.audiosink_element, "openalsink", PLAYER_INI_MAX_STRLEN - 1);
+ audiobin = player->pipeline->audiobin;
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CONV_BFORMAT, "audioconvert", "audio-converter-bformat", TRUE, player);
+ LOGD("make element for audio extract, option = 0x%X", player->audio_extract_opt);
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_CAPS_360, "capsfilter", "audio-caps-filter", TRUE, player);
- acaps = gst_caps_from_string(SPATIAL_AUDIO_CAPS);
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_CAPS_360].gst), "caps", acaps, NULL);
- gst_caps_unref(acaps);
+ /* audio bin structure according to the mmplayer_audio_extract_opt_e.
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, "openalsink", "audiosink", TRUE, player);
+ [case 1] extract interleave audio pcm without playback
+ : MM_PLAYER_AUDIO_EXTRACT_DEFAULT (sync)
+ MM_PLAYER_AUDIO_EXTRACT_NO_SYNC_WITH_CLOCK (non sync)
- player->is_openal_plugin_used = TRUE;
- } else {
- if (player->is_360_feature_enabled && player->is_content_spherical)
- LOGW("Audio track isn't of the ambisonic type and can't be played back as a spatial sound.");
- MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, player->ini.audiosink_element, "audiosink", TRUE, player);
- }
+ * src - ... - aconv - resample - capsfilter - fakesink (sync or not)
- if ((MMPLAYER_IS_RTSP_STREAMING(player)) ||
- (player->videodec_linked && player->ini.use_system_clock)) {
- LOGD("system clock will be used.");
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "provide-clock", FALSE, NULL);
+ [case 2] deinterleave for each channel without playback
+ : MM_PLAYER_AUDIO_EXTRACT_DEINTERLEAVE (sync)
+ MM_PLAYER_AUDIO_EXTRACT_NO_SYNC_AND_DEINTERLEAVE (non sync)
+
+ * src - ... - aconv - resample - capsfilter - deinterleave - fakesink (sync or not)
+ - fakesink (sync or not)
+ - ... (sync or not)
+
+ [case 3] [case 1(sync only)] + playback
+ : MM_PLAYER_AUDIO_EXTRACT_WITH_PLAYBACK
+
+ * src - ... - tee - queue1 - playback path
+ - queue2 - [case1 pipeline with sync]
+
+ [case 4] [case 2(sync only)] + playback
+ : MM_PLAYER_AUDIO_EXTRACT_DEINTERLEAVE_WITH_PLAYBACK
+
+ * src - ... - tee - queue1 - playback path
+ - queue2 - [case2 pipeline with sync]
+
+ */
+
+ /* 1. create tee and playback path
+ 'tee' should be added at first to copy the decoded stream
+ */
+ if (player->audio_extract_opt & MM_PLAYER_AUDIO_EXTRACT_WITH_PLAYBACK) {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_TEE, "tee", "audio-tee", *bucket, player);
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_TEE].gst), "num-src-pads", 2, NULL);
+
+ /* tee - path 1 : for playback path */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_TEE_Q1, "queue", "audio-tee-queue1", *bucket, player);
+ __mmplayer_gst_make_audio_playback_sink(player, bucket);
+
+ /* tee - path 2 : for extract path */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_TEE_Q2, "queue", "audio-tee-queue2", *bucket, player);
+ extract_sink_id = MMPLAYER_A_EXTRACT_SINK; /* there is another playback sink */
+ }
+
+ /* if there is tee, 'tee - path 2' is linked here */
+ /* converter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_EXTRACT_CONV, "audioconvert", "audio-ext-conv", *bucket, player);
+
+ /* resampler */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_EXTRACT_RESAMPLER, player->ini.audioresampler_element, "audio-ext-resampler", *bucket, player);
+
+ /* 2. decide the extract pcm format */
+ mm_attrs_multiple_get(player->attrs, NULL,
+ "pcm_audioformat", &dst_format, &dst_len,
+ "pcm_extraction_samplerate", &dst_samplerate,
+ "pcm_extraction_channels", &dst_channels,
+ NULL);
+
+ LOGD("required extract pcm format - format: %s(%d), samplerate : %d, channel: %d",
+ dst_format, dst_len, dst_samplerate, dst_channels);
+
+ if (dst_format == NULL || dst_len == 0 || dst_samplerate == 0 || dst_channels == 0) {
+ mm_attrs_multiple_get(player->attrs, NULL,
+ "content_audio_format", &dst_format, &dst_len, /* get string and len */
+ "content_audio_samplerate", &dst_samplerate,
+ "content_audio_channels", &dst_channels,
+ NULL);
+
+ LOGD("apply the decoded pcm format - format: %s(%d), samplerate : %d, channel: %d",
+ dst_format, dst_len, dst_samplerate, dst_channels);
+
+ /* If there is no enough information, set it to platform default value. */
+ if (dst_format == NULL || util_convert_audio_pcm_str_to_media_format_mime(dst_format) == MEDIA_FORMAT_MAX) {
+ LOGD("set platform default format");
+ dst_format = DEFAULT_PCM_OUT_FORMAT;
}
+ if (dst_samplerate <= 0) dst_samplerate = DEFAULT_PCM_OUT_SAMPLERATE;
+ if (dst_channels <= 0) dst_channels = DEFAULT_PCM_OUT_CHANNEL;
+ }
- if (g_strrstr(player->ini.audiosink_element, "pulsesink"))
- __mmplayer_gst_set_pulsesink_property(player, attrs);
- else if (g_strrstr(player->ini.audiosink_element, "openalsink"))
- __mmplayer_gst_set_openalsink_property(player);
+ /* 3. create capsfilter */
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_EXTRACT_CAPS, "capsfilter", "audio-ext-caps", *bucket, player);
+ caps = gst_caps_new_simple("audio/x-raw",
+ "format", G_TYPE_STRING, dst_format,
+ "rate", G_TYPE_INT, dst_samplerate,
+ "channels", G_TYPE_INT, dst_channels,
+ NULL);
- /* qos on */
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "qos", TRUE, NULL); /* qos on */
- g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "slave-method", GST_AUDIO_BASE_SINK_SLAVE_NONE, NULL);
+ caps_str = gst_caps_to_string(caps);
+ LOGD("new caps : %s", caps_str);
- sink_pad = gst_element_get_static_pad(audiobin[MMPLAYER_A_SINK].gst, "sink");
- __mmplayer_add_signal_connection(player, G_OBJECT(sink_pad), MM_PLAYER_SIGNAL_TYPE_AUDIOBIN,
- "notify::caps", G_CALLBACK(__mmplayer_gst_caps_notify_cb), (gpointer)player);
- gst_object_unref(GST_OBJECT(sink_pad));
+ g_object_set(GST_ELEMENT(audiobin[MMPLAYER_A_EXTRACT_CAPS].gst), "caps", caps, NULL);
+
+ /* clean */
+ gst_caps_unref(caps);
+ MMPLAYER_FREEIF(caps_str);
+
+ /* 4-1. create deinterleave to extract pcm for each channel */
+ if (player->audio_extract_opt & MM_PLAYER_AUDIO_EXTRACT_DEINTERLEAVE) {
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_EXTRACT_DEINTERLEAVE, "deinterleave", "deinterleave", *bucket, player);
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_EXTRACT_DEINTERLEAVE].gst), "keep-positions", TRUE, NULL);
+
+ /* audiosink will be added after getting signal for each channel */
+ __mmplayer_add_signal_connection(player, G_OBJECT(audiobin[MMPLAYER_A_EXTRACT_DEINTERLEAVE].gst),
+ MM_PLAYER_SIGNAL_TYPE_AUTOPLUG, "pad-added", G_CALLBACK(__mmplayer_gst_audio_deinterleave_pad_added), (gpointer)player);
+ } else {
+ /* 4-2. create fakesink to extract interlevaed pcm */
+ LOGD("add audio fakesink for interleaved audio");
+ MMPLAYER_CREATE_ELEMENT(audiobin, extract_sink_id, "fakesink", "fakeaudiosink", *bucket, player);
+ if (!(player->audio_extract_opt & MM_PLAYER_AUDIO_EXTRACT_NO_SYNC_WITH_CLOCK))
+ g_object_set(G_OBJECT(audiobin[extract_sink_id].gst), "sync", TRUE, NULL);
+ g_object_set(G_OBJECT(audiobin[extract_sink_id].gst), "signal-handoffs", TRUE, NULL);
+ __mmplayer_add_signal_connection(player,
+ G_OBJECT(audiobin[extract_sink_id].gst),
+ MM_PLAYER_SIGNAL_TYPE_AUDIOBIN,
+ "handoff",
+ G_CALLBACK(__mmplayer_audio_stream_decoded_render_cb),
+ (gpointer)player);
+
+ __mmplayer_add_sink(player, audiobin[extract_sink_id].gst);
+ }
+
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_NONE;
+
+ERROR: /* MMPLAYER_CREATE_ELEMENT */
+ MMPLAYER_FLEAVE();
+ return MM_ERROR_PLAYER_INTERNAL;
+}
+
+static int
+__mmplayer_gst_make_audio_bin_element(mmplayer_t *player, GList **bucket)
+{
+ int ret = MM_ERROR_NONE;
+ mmplayer_gst_element_t *audiobin = NULL;
+ GList *element_bucket = NULL;
+
+ MMPLAYER_FENTER();
+ MMPLAYER_RETURN_VAL_IF_FAIL(player && player->pipeline &&
+ player->pipeline->audiobin, MM_ERROR_PLAYER_NOT_INITIALIZED);
+
+ audiobin = player->pipeline->audiobin;
+
+ if (player->build_audio_offload) { /* skip all the audio filters */
+ LOGD("create audio offload sink : %s", player->ini.audio_offload_sink_element);
+ MMPLAYER_CREATE_ELEMENT(audiobin, MMPLAYER_A_SINK, player->ini.audio_offload_sink_element, "audiosink", element_bucket, player);
+ g_object_set(G_OBJECT(audiobin[MMPLAYER_A_SINK].gst), "sync", TRUE, NULL);
__mmplayer_add_sink(player, audiobin[MMPLAYER_A_SINK].gst);
+ goto DONE;
}
+ /* FIXME: need to mention the supportable condition at API reference */
+ if (player->audio_decoded_cb && (!MMPLAYER_IS_RTSP_STREAMING(player)))
+ ret = __mmplayer_gst_make_audio_extract_sink(player, &element_bucket);
+ else
+ ret = __mmplayer_gst_make_audio_playback_sink(player, &element_bucket);
+
+ if (ret != MM_ERROR_NONE)
+ goto ERROR;
DONE:
+ LOGD("success to make audio bin element");
*bucket = element_bucket;
MMPLAYER_FLEAVE();
return MM_ERROR_NONE;
ERROR:
+ LOGE("failed to make audio bin element");
g_list_free(element_bucket);
*bucket = NULL;
player->pipeline->audiobin = audiobin;
/* create audio filters and audiosink */
- if (__mmplayer_gst_fill_audio_bucket(player, &element_bucket) != MM_ERROR_NONE)
+ if (__mmplayer_gst_make_audio_bin_element(player, &element_bucket) != MM_ERROR_NONE)
goto ERROR;
/* adding created elements to bin */
/* create video360 filter */
if (player->is_360_feature_enabled && player->is_content_spherical) {
LOGD("create video360 element");
- MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_360, "video360", "video-360", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_360, "video360", "video-360", element_bucket, player);
__mmplayer_gst_set_video360_property(player);
goto EXIT;
}
/* in case of sw codec & overlay surface type, except 360 playback.
* if libav video decoder is selected, videoconvert is required to render the shm wl-buffer which support RGB only via tizenwlsink. */
LOGD("create video converter: %s", video_csc);
- MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_CONV, video_csc, "video converter", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(player->pipeline->videobin, MMPLAYER_V_CONV, video_csc, "video converter", element_bucket, player);
EXIT:
*bucket = element_bucket;
goto ERROR;
videosink_factory_name = __mmplayer_get_videosink_factory_name(player, surface_type);
- MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, videosink_factory_name, "videosink", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(videobin, MMPLAYER_V_SINK, videosink_factory_name, "videosink", element_bucket, player);
/* additional setting for sink plug-in */
if (__mmplayer_gst_set_videosink_property(player, surface_type) != MM_ERROR_NONE) {
GList *element_bucket = NULL;
mmplayer_gst_element_t *textbin = player->pipeline->textbin;
- MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_QUEUE, "queue", "text_queue", TRUE, player);
- MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_IDENTITY, "identity", "text_identity", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_QUEUE, "queue", "text_queue", element_bucket, player);
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_IDENTITY, "identity", "text_identity", element_bucket, player);
g_object_set(G_OBJECT(textbin[MMPLAYER_T_IDENTITY].gst),
"signal-handoffs", FALSE,
NULL);
- MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_FAKE_SINK, "fakesink", "text_fakesink", TRUE, player);
+ MMPLAYER_CREATE_ELEMENT(textbin, MMPLAYER_T_FAKE_SINK, "fakesink", "text_fakesink", element_bucket, player);
__mmplayer_add_signal_connection(player,
G_OBJECT(textbin[MMPLAYER_T_FAKE_SINK].gst),
MM_PLAYER_SIGNAL_TYPE_TEXTBIN,
GstPad *pad = NULL;
gint samplerate = 0, channels = 0;
GstStructure *p = NULL;
+ GstElement *aconv = NULL;
LOGD("try to update audio attrs");
MMPLAYER_RETURN_VAL_IF_FAIL(player->pipeline->audiobin, FALSE);
- MMPLAYER_RETURN_VAL_IF_FAIL(player->pipeline->audiobin[MMPLAYER_A_SINK].gst, FALSE);
- pad = gst_element_get_static_pad(
- player->pipeline->audiobin[MMPLAYER_A_CONV].gst, "sink");
+ if (player->pipeline->audiobin[MMPLAYER_A_CONV].gst) {
+ aconv = player->pipeline->audiobin[MMPLAYER_A_CONV].gst;
+ } else if (player->pipeline->audiobin[MMPLAYER_A_EXTRACT_CONV].gst) {
+ aconv = player->pipeline->audiobin[MMPLAYER_A_EXTRACT_CONV].gst;
+ } else {
+ LOGE("there is no audio converter");
+ return FALSE;
+ }
+
+ pad = gst_element_get_static_pad(aconv, "sink");
if (!pad) {
- LOGW("failed to get pad from audiosink");
+ LOGW("failed to get pad from audio converter");
return FALSE;
}
projects / platform / core / multimedia / libmm-player.git / commitdiff