void PlaybackPipeline::reattachTrack(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate, RefPtr<TrackPrivateBase> trackPrivate)
{
GST_DEBUG("Re-attaching track");
// FIXME: Maybe remove this method. Now the caps change is managed by gst_appsrc_push_sample() in enqueueSample()
// and flushAndEnqueueNonDisplayingSamples().
WebKitMediaSrc* webKitMediaSrc = m_webKitMediaSrc.get();
GST_OBJECT_LOCK(webKitMediaSrc);
Stream* stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get());
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream && stream->type != Invalid);
// The caps change is managed by gst_appsrc_push_sample() in enqueueSample() and
// flushAndEnqueueNonDisplayingSamples(), so the caps aren't set from here.
GRefPtr<GstCaps> appsrcCaps = adoptGRef(gst_app_src_get_caps(GST_APP_SRC(stream->appsrc)));
const gchar* mediaType = gst_structure_get_name(gst_caps_get_structure(appsrcCaps.get(), 0));
int signal = -1;
GST_OBJECT_LOCK(webKitMediaSrc);
if (g_str_has_prefix(mediaType, "audio")) {
ASSERT(stream->type == Audio);
signal = SIGNAL_AUDIO_CHANGED;
stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "video")) {
ASSERT(stream->type == Video);
signal = SIGNAL_VIDEO_CHANGED;
stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "text")) {
ASSERT(stream->type == Text);
signal = SIGNAL_TEXT_CHANGED;
// FIXME: Support text tracks.
}
GST_OBJECT_UNLOCK(webKitMediaSrc);
if (signal != -1)
g_signal_emit(G_OBJECT(stream->parent), webKitMediaSrcSignals[signal], 0, nullptr);
}
void PlaybackPipeline::attachTrack(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate, RefPtr<TrackPrivateBase> trackPrivate, GstCaps* caps)
{
WebKitMediaSrc* webKitMediaSrc = m_webKitMediaSrc.get();
Stream* stream = 0;
//GstCaps* appsrccaps = 0;
GstStructure* s = 0;
const gchar* appsrctypename = 0;
const gchar* mediaType = 0;
gchar *parserBinName;
bool capsNotifyHandlerConnected = false;
unsigned padId = 0;
GST_OBJECT_LOCK(webKitMediaSrc);
stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get());
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream != 0);
//gst_app_src_set_caps(GST_APP_SRC(stream->appsrc), caps);
//appsrccaps = gst_app_src_get_caps(GST_APP_SRC(stream->appsrc));
s = gst_caps_get_structure(caps, 0);
appsrctypename = gst_structure_get_name(s);
mediaType = appsrctypename;
GST_OBJECT_LOCK(webKitMediaSrc);
padId = stream->parent->priv->numberOfPads;
stream->parent->priv->numberOfPads++;
GST_OBJECT_UNLOCK(webKitMediaSrc);
parserBinName = g_strdup_printf("streamparser%u", padId);
ASSERT(caps != 0);
stream->parser = gst_bin_new(parserBinName);
g_free(parserBinName);
GST_DEBUG_OBJECT(webKitMediaSrc, "Configured track %s: appsrc=%s, padId=%u, mediaType=%s, caps=%" GST_PTR_FORMAT, trackPrivate->id().string().utf8().data(), GST_ELEMENT_NAME(stream->appsrc), padId, mediaType, caps);
if (!g_strcmp0(mediaType, "video/x-h264")) {
GstElement* parser;
GstElement* capsfilter;
GstPad* pad = nullptr;
GstCaps* filtercaps;
filtercaps = gst_caps_new_simple("video/x-h264", "alignment", G_TYPE_STRING, "au", NULL);
parser = gst_element_factory_make("h264parse", 0);
capsfilter = gst_element_factory_make("capsfilter", 0);
g_object_set(capsfilter, "caps", filtercaps, NULL);
gst_caps_unref(filtercaps);
gst_bin_add_many(GST_BIN(stream->parser), parser, capsfilter, NULL);
gst_element_link_pads(parser, "src", capsfilter, "sink");
if (!pad)
pad = gst_element_get_static_pad(parser, "sink");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
pad = gst_element_get_static_pad(capsfilter, "src");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad));
gst_object_unref(pad);
} else if (!g_strcmp0(mediaType, "video/x-h265")) {
GstElement* parser;
GstElement* capsfilter;
GstPad* pad = nullptr;
GstCaps* filtercaps;
filtercaps = gst_caps_new_simple("video/x-h265", "alignment", G_TYPE_STRING, "au", NULL);
parser = gst_element_factory_make("h265parse", 0);
capsfilter = gst_element_factory_make("capsfilter", 0);
g_object_set(capsfilter, "caps", filtercaps, NULL);
gst_caps_unref(filtercaps);
gst_bin_add_many(GST_BIN(stream->parser), parser, capsfilter, NULL);
gst_element_link_pads(parser, "src", capsfilter, "sink");
if (!pad)
pad = gst_element_get_static_pad(parser, "sink");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
pad = gst_element_get_static_pad(capsfilter, "src");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad));
gst_object_unref(pad);
} else if (!g_strcmp0(mediaType, "audio/mpeg")) {
gint mpegversion = -1;
GstElement* parser;
GstPad* pad = nullptr;
gst_structure_get_int(s, "mpegversion", &mpegversion);
if (mpegversion == 1) {
parser = gst_element_factory_make("mpegaudioparse", 0);
} else if (mpegversion == 2 || mpegversion == 4) {
parser = gst_element_factory_make("aacparse", 0);
} else {
ASSERT_NOT_REACHED();
}
gst_bin_add(GST_BIN(stream->parser), parser);
if (!pad)
pad = gst_element_get_static_pad(parser, "sink");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad));
gst_object_unref(pad);
pad = gst_element_get_static_pad(parser, "src");
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad));
gst_object_unref(pad);
} else {
GST_ERROR_OBJECT(stream->parent, "Unsupported caps: %" GST_PTR_FORMAT, caps);
gst_object_unref(GST_OBJECT(stream->parser));
return;
}
//gst_caps_unref(appsrccaps);
GST_OBJECT_LOCK(webKitMediaSrc);
stream->type = Unknown;
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream->parser);
gst_bin_add(GST_BIN(stream->parent), stream->parser);
gst_element_sync_state_with_parent(stream->parser);
GstPad* sinkpad = gst_element_get_static_pad(stream->parser, "sink");
GstPad* srcpad = gst_element_get_static_pad(stream->appsrc, "src");
gst_pad_link(srcpad, sinkpad);
gst_object_unref(srcpad);
srcpad = 0;
gst_object_unref(sinkpad);
sinkpad = 0;
srcpad = gst_element_get_static_pad(stream->parser, "src");
// TODO: Is padId the best way to identify the Stream? What about trackId?
g_object_set_data(G_OBJECT(srcpad), "id", GINT_TO_POINTER(padId));
if (!capsNotifyHandlerConnected)
g_signal_connect(srcpad, "notify::caps", G_CALLBACK(webKitMediaSrcParserNotifyCaps), stream);
webKitMediaSrcLinkStreamToSrcPad(srcpad, stream);
ASSERT(stream->parent->priv->mediaPlayerPrivate);
int signal = -1;
if (g_str_has_prefix(mediaType, "audio")) {
GST_OBJECT_LOCK(webKitMediaSrc);
stream->type = Audio;
stream->parent->priv->nAudio++;
GST_OBJECT_UNLOCK(webKitMediaSrc);
signal = SIGNAL_AUDIO_CHANGED;
stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "video")) {
GST_OBJECT_LOCK(webKitMediaSrc);
stream->type = Video;
stream->parent->priv->nVideo++;
GST_OBJECT_UNLOCK(webKitMediaSrc);
signal = SIGNAL_VIDEO_CHANGED;
stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "text")) {
GST_OBJECT_LOCK(webKitMediaSrc);
stream->type = Text;
stream->parent->priv->nText++;
GST_OBJECT_UNLOCK(webKitMediaSrc);
signal = SIGNAL_TEXT_CHANGED;
// TODO: Support text tracks.
}
if (signal != -1)
g_signal_emit(G_OBJECT(stream->parent), webkit_media_src_signals[signal], 0, NULL);
gst_object_unref(srcpad);
srcpad = 0;
}
void PlaybackPipeline::attachTrack(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate, RefPtr<TrackPrivateBase> trackPrivate, GstStructure* structure, GstCaps* caps)
{
WebKitMediaSrc* webKitMediaSrc = m_webKitMediaSrc.get();
GST_OBJECT_LOCK(webKitMediaSrc);
Stream* stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get());
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream);
GST_OBJECT_LOCK(webKitMediaSrc);
unsigned padId = stream->parent->priv->numberOfPads;
stream->parent->priv->numberOfPads++;
GST_OBJECT_UNLOCK(webKitMediaSrc);
const gchar* mediaType = gst_structure_get_name(structure);
GST_DEBUG_OBJECT(webKitMediaSrc, "Configured track %s: appsrc=%s, padId=%u, mediaType=%s", trackPrivate->id().string().utf8().data(), GST_ELEMENT_NAME(stream->appsrc), padId, mediaType);
GUniquePtr<gchar> parserBinName(g_strdup_printf("streamparser%u", padId));
if (!g_strcmp0(mediaType, "video/x-h264")) {
GRefPtr<GstCaps> filterCaps = adoptGRef(gst_caps_new_simple("video/x-h264", "alignment", G_TYPE_STRING, "au", nullptr));
GstElement* capsfilter = gst_element_factory_make("capsfilter", nullptr);
g_object_set(capsfilter, "caps", filterCaps.get(), nullptr);
stream->parser = gst_bin_new(parserBinName.get());
GstElement* parser = gst_element_factory_make("h264parse", nullptr);
gst_bin_add_many(GST_BIN(stream->parser), parser, capsfilter, nullptr);
gst_element_link_pads(parser, "src", capsfilter, "sink");
GRefPtr<GstPad> pad = adoptGRef(gst_element_get_static_pad(parser, "sink"));
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad.get()));
pad = adoptGRef(gst_element_get_static_pad(capsfilter, "src"));
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad.get()));
} else if (!g_strcmp0(mediaType, "video/x-h265")) {
GRefPtr<GstCaps> filterCaps = adoptGRef(gst_caps_new_simple("video/x-h265", "alignment", G_TYPE_STRING, "au", nullptr));
GstElement* capsfilter = gst_element_factory_make("capsfilter", nullptr);
g_object_set(capsfilter, "caps", filterCaps.get(), nullptr);
stream->parser = gst_bin_new(parserBinName.get());
GstElement* parser = gst_element_factory_make("h265parse", nullptr);
gst_bin_add_many(GST_BIN(stream->parser), parser, capsfilter, nullptr);
gst_element_link_pads(parser, "src", capsfilter, "sink");
GRefPtr<GstPad> pad = adoptGRef(gst_element_get_static_pad(parser, "sink"));
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad.get()));
pad = adoptGRef(gst_element_get_static_pad(capsfilter, "src"));
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad.get()));
} else if (!g_strcmp0(mediaType, "audio/mpeg")) {
gint mpegversion = -1;
gst_structure_get_int(structure, "mpegversion", &mpegversion);
GstElement* parser = nullptr;
if (mpegversion == 1)
parser = gst_element_factory_make("mpegaudioparse", nullptr);
else if (mpegversion == 2 || mpegversion == 4)
parser = gst_element_factory_make("aacparse", nullptr);
else
ASSERT_NOT_REACHED();
stream->parser = gst_bin_new(parserBinName.get());
gst_bin_add(GST_BIN(stream->parser), parser);
GRefPtr<GstPad> pad = adoptGRef(gst_element_get_static_pad(parser, "sink"));
gst_element_add_pad(stream->parser, gst_ghost_pad_new("sink", pad.get()));
pad = adoptGRef(gst_element_get_static_pad(parser, "src"));
gst_element_add_pad(stream->parser, gst_ghost_pad_new("src", pad.get()));
} else if (!g_strcmp0(mediaType, "video/x-vp9"))
stream->parser = nullptr;
else {
GST_ERROR_OBJECT(stream->parent, "Unsupported media format: %s", mediaType);
return;
}
GST_OBJECT_LOCK(webKitMediaSrc);
stream->type = Unknown;
GST_OBJECT_UNLOCK(webKitMediaSrc);
GRefPtr<GstPad> sourcePad;
if (stream->parser) {
gst_bin_add(GST_BIN(stream->parent), stream->parser);
gst_element_sync_state_with_parent(stream->parser);
GRefPtr<GstPad> sinkPad = adoptGRef(gst_element_get_static_pad(stream->parser, "sink"));
sourcePad = adoptGRef(gst_element_get_static_pad(stream->appsrc, "src"));
gst_pad_link(sourcePad.get(), sinkPad.get());
sourcePad = adoptGRef(gst_element_get_static_pad(stream->parser, "src"));
} else {
GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "Stream of type %s doesn't require a parser bin", mediaType);
sourcePad = adoptGRef(gst_element_get_static_pad(stream->appsrc, "src"));
}
ASSERT(sourcePad);
// FIXME: Is padId the best way to identify the Stream? What about trackId?
g_object_set_data(G_OBJECT(sourcePad.get()), "padId", GINT_TO_POINTER(padId));
webKitMediaSrcLinkParser(sourcePad.get(), caps, stream);
ASSERT(stream->parent->priv->mediaPlayerPrivate);
int signal = -1;
GST_OBJECT_LOCK(webKitMediaSrc);
if (g_str_has_prefix(mediaType, "audio")) {
stream->type = Audio;
stream->parent->priv->numberOfAudioStreams++;
signal = SIGNAL_AUDIO_CHANGED;
stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "video")) {
stream->type = Video;
stream->parent->priv->numberOfVideoStreams++;
signal = SIGNAL_VIDEO_CHANGED;
stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "text")) {
stream->type = Text;
stream->parent->priv->numberOfTextStreams++;
signal = SIGNAL_TEXT_CHANGED;
// FIXME: Support text tracks.
}
GST_OBJECT_UNLOCK(webKitMediaSrc);
if (signal != -1)
g_signal_emit(G_OBJECT(stream->parent), webKitMediaSrcSignals[signal], 0, nullptr);
}
void PlaybackPipeline::reattachTrack(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate, RefPtr<TrackPrivateBase> trackPrivate, GstCaps* caps)
{
LOG_MEDIA_MESSAGE("Re-attaching track");
UNUSED_PARAM(caps);
// TODO: Maybe remove this method.
// Now the caps change is managed by gst_appsrc_push_sample()
// in enqueueSample() and flushAndEnqueueNonDisplayingSamples().
WebKitMediaSrc* webKitMediaSrc = m_webKitMediaSrc.get();
GST_OBJECT_LOCK(webKitMediaSrc);
Stream* stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get());
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream != 0);
ASSERT(stream->type != Invalid);
GstCaps* oldAppsrccaps = gst_app_src_get_caps(GST_APP_SRC(stream->appsrc));
// Now the caps change is managed by gst_appsrc_push_sample()
// in enqueueSample() and flushAndEnqueueNonDisplayingSamples().
// gst_app_src_set_caps(GST_APP_SRC(stream->appsrc), caps);
GstCaps* appsrccaps = gst_app_src_get_caps(GST_APP_SRC(stream->appsrc));
const gchar* mediaType = gst_structure_get_name(gst_caps_get_structure(appsrccaps, 0));
if (!gst_caps_is_equal(oldAppsrccaps, appsrccaps)) {
LOG_MEDIA_MESSAGE("Caps have changed, but reconstructing the sequence of elements is not supported yet");
gchar* stroldcaps = gst_caps_to_string(oldAppsrccaps);
gchar* strnewcaps = gst_caps_to_string(appsrccaps);
LOG_MEDIA_MESSAGE("oldcaps: %s", stroldcaps);
LOG_MEDIA_MESSAGE("newcaps: %s", strnewcaps);
g_free(stroldcaps);
g_free(strnewcaps);
}
int signal = -1;
GST_OBJECT_LOCK(webKitMediaSrc);
if (g_str_has_prefix(mediaType, "audio")) {
ASSERT(stream->type == Audio);
signal = SIGNAL_AUDIO_CHANGED;
stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "video")) {
ASSERT(stream->type == Video);
signal = SIGNAL_VIDEO_CHANGED;
stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "text")) {
ASSERT(stream->type == Text);
signal = SIGNAL_TEXT_CHANGED;
// TODO: Support text tracks and mediaTypes related to EME
}
GST_OBJECT_UNLOCK(webKitMediaSrc);
gst_caps_unref(appsrccaps);
gst_caps_unref(oldAppsrccaps);
if (signal != -1)
g_signal_emit(G_OBJECT(stream->parent), webkit_media_src_signals[signal], 0, NULL);
}
