PlaybackPipeline.cpp [plain text]
#include "config.h"
#include "PlaybackPipeline.h"
#if ENABLE(VIDEO) && USE(GSTREAMER) && ENABLE(MEDIA_SOURCE)
#include "AudioTrackPrivateGStreamer.h"
#include "GStreamerMediaSample.h"
#include "GStreamerUtilities.h"
#include "MediaSample.h"
#include "SourceBufferPrivateGStreamer.h"
#include "VideoTrackPrivateGStreamer.h"
#include <gst/app/gstappsrc.h>
#include <gst/gst.h>
#include <wtf/MainThread.h>
#include <wtf/RefCounted.h>
#include <wtf/glib/GMutexLocker.h>
#include <wtf/glib/GRefPtr.h>
#include <wtf/glib/GUniquePtr.h>
#include <wtf/text/AtomicString.h>
static Stream* getStreamByTrackId(WebKitMediaSrc*, AtomicString);
static Stream* getStreamBySourceBufferPrivate(WebKitMediaSrc*, WebCore::SourceBufferPrivateGStreamer*);
static Stream* getStreamByTrackId(WebKitMediaSrc* source, AtomicString trackIdString)
{
for (Stream* stream : source->priv->streams) {
if (stream->type != WebCore::Invalid
&& ((stream->audioTrack && stream->audioTrack->id() == trackIdString)
|| (stream->videoTrack && stream->videoTrack->id() == trackIdString) ) ) {
return stream;
}
}
return nullptr;
}
static Stream* getStreamBySourceBufferPrivate(WebKitMediaSrc* source, WebCore::SourceBufferPrivateGStreamer* sourceBufferPrivate)
{
for (Stream* stream : source->priv->streams) {
if (stream->sourceBuffer == sourceBufferPrivate)
return stream;
}
return nullptr;
}
static GstFlowReturn pushSample(GstAppSrc* appsrc, GstSample* sample)
{
g_return_val_if_fail(GST_IS_SAMPLE(sample), GST_FLOW_ERROR);
GstCaps* caps = gst_sample_get_caps(sample);
if (caps)
gst_app_src_set_caps(appsrc, caps);
else
GST_WARNING_OBJECT(appsrc, "received sample without caps");
GstBuffer* buffer = gst_sample_get_buffer(sample);
if (UNLIKELY(!buffer)) {
GST_WARNING_OBJECT(appsrc, "received sample without buffer");
return GST_FLOW_OK;
}
return gst_app_src_push_buffer(appsrc, gst_buffer_ref(buffer));
}
namespace WebCore {
void PlaybackPipeline::setWebKitMediaSrc(WebKitMediaSrc* webKitMediaSrc)
{
GST_DEBUG("webKitMediaSrc=%p", webKitMediaSrc);
m_webKitMediaSrc = webKitMediaSrc;
}
WebKitMediaSrc* PlaybackPipeline::webKitMediaSrc()
{
return m_webKitMediaSrc.get();
}
MediaSourcePrivate::AddStatus PlaybackPipeline::addSourceBuffer(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate)
{
WebKitMediaSrcPrivate* priv = m_webKitMediaSrc->priv;
if (priv->allTracksConfigured) {
GST_ERROR_OBJECT(m_webKitMediaSrc.get(), "Adding new source buffers after first data not supported yet");
return MediaSourcePrivate::NotSupported;
}
GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "State %d", int(GST_STATE(m_webKitMediaSrc.get())));
Stream* stream = new Stream{ };
stream->parent = m_webKitMediaSrc.get();
stream->appsrc = gst_element_factory_make("appsrc", nullptr);
stream->appsrcNeedDataFlag = false;
stream->sourceBuffer = sourceBufferPrivate.get();
stream->type = Invalid;
stream->parser = nullptr;
stream->caps = nullptr;
stream->audioTrack = nullptr;
stream->videoTrack = nullptr;
stream->presentationSize = WebCore::FloatSize();
stream->lastEnqueuedTime = MediaTime::invalidTime();
gst_app_src_set_callbacks(GST_APP_SRC(stream->appsrc), &enabledAppsrcCallbacks, stream->parent, nullptr);
gst_app_src_set_emit_signals(GST_APP_SRC(stream->appsrc), FALSE);
gst_app_src_set_stream_type(GST_APP_SRC(stream->appsrc), GST_APP_STREAM_TYPE_SEEKABLE);
gst_app_src_set_max_bytes(GST_APP_SRC(stream->appsrc), 2 * WTF::MB);
g_object_set(G_OBJECT(stream->appsrc), "block", FALSE, "min-percent", 20, nullptr);
GST_OBJECT_LOCK(m_webKitMediaSrc.get());
priv->streams.prepend(stream);
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
gst_bin_add(GST_BIN(m_webKitMediaSrc.get()), stream->appsrc);
gst_element_sync_state_with_parent(stream->appsrc);
return MediaSourcePrivate::Ok;
}
void PlaybackPipeline::removeSourceBuffer(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate)
{
ASSERT(WTF::isMainThread());
GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "Element removed from MediaSource");
GST_OBJECT_LOCK(m_webKitMediaSrc.get());
WebKitMediaSrcPrivate* priv = m_webKitMediaSrc->priv;
Stream* stream = nullptr;
Deque<Stream*>::iterator streamPosition = priv->streams.begin();
for (; streamPosition != priv->streams.end(); ++streamPosition) {
if ((*streamPosition)->sourceBuffer == sourceBufferPrivate.get()) {
stream = *streamPosition;
break;
}
}
if (stream)
priv->streams.remove(streamPosition);
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
if (stream)
webKitMediaSrcFreeStream(m_webKitMediaSrc.get(), stream);
}
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);
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;
}
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)
{
GST_DEBUG("Re-attaching track");
WebKitMediaSrc* webKitMediaSrc = m_webKitMediaSrc.get();
GST_OBJECT_LOCK(webKitMediaSrc);
Stream* stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get());
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream && stream->type != Invalid);
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;
}
GST_OBJECT_UNLOCK(webKitMediaSrc);
if (signal != -1)
g_signal_emit(G_OBJECT(stream->parent), webKitMediaSrcSignals[signal], 0, nullptr);
}
void PlaybackPipeline::notifyDurationChanged()
{
gst_element_post_message(GST_ELEMENT(m_webKitMediaSrc.get()), gst_message_new_duration_changed(GST_OBJECT(m_webKitMediaSrc.get())));
}
void PlaybackPipeline::markEndOfStream(MediaSourcePrivate::EndOfStreamStatus)
{
WebKitMediaSrcPrivate* priv = m_webKitMediaSrc->priv;
GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "Have EOS");
GST_OBJECT_LOCK(m_webKitMediaSrc.get());
bool allTracksConfigured = priv->allTracksConfigured;
if (!allTracksConfigured)
priv->allTracksConfigured = true;
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
if (!allTracksConfigured) {
gst_element_no_more_pads(GST_ELEMENT(m_webKitMediaSrc.get()));
webKitMediaSrcDoAsyncDone(m_webKitMediaSrc.get());
}
Vector<GstAppSrc*> appsrcs;
GST_OBJECT_LOCK(m_webKitMediaSrc.get());
for (Stream* stream : priv->streams) {
if (stream->appsrc)
appsrcs.append(GST_APP_SRC(stream->appsrc));
}
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
for (GstAppSrc* appsrc : appsrcs)
gst_app_src_end_of_stream(appsrc);
}
void PlaybackPipeline::flush(AtomicString trackId)
{
ASSERT(WTF::isMainThread());
GST_DEBUG("flush: trackId=%s", trackId.string().utf8().data());
GST_OBJECT_LOCK(m_webKitMediaSrc.get());
Stream* stream = getStreamByTrackId(m_webKitMediaSrc.get(), trackId);
if (!stream) {
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
return;
}
stream->lastEnqueuedTime = MediaTime::invalidTime();
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
}
void PlaybackPipeline::enqueueSample(RefPtr<MediaSample> mediaSample)
{
ASSERT(WTF::isMainThread());
AtomicString trackId = mediaSample->trackID();
GST_TRACE("enqueing sample trackId=%s PTS=%f presentationSize=%.0fx%.0f at %" GST_TIME_FORMAT " duration: %" GST_TIME_FORMAT,
trackId.string().utf8().data(), mediaSample->presentationTime().toFloat(),
mediaSample->presentationSize().width(), mediaSample->presentationSize().height(),
GST_TIME_ARGS(WebCore::toGstClockTime(mediaSample->presentationTime().toDouble())),
GST_TIME_ARGS(WebCore::toGstClockTime(mediaSample->duration().toDouble())));
Stream* stream = getStreamByTrackId(m_webKitMediaSrc.get(), trackId);
if (!stream) {
GST_WARNING("No stream!");
return;
}
if (!stream->sourceBuffer->isReadyForMoreSamples(trackId)) {
GST_DEBUG("enqueueSample: skip adding new sample for trackId=%s, SB is not ready yet", trackId.string().utf8().data());
return;
}
GstElement* appsrc = stream->appsrc;
MediaTime lastEnqueuedTime = stream->lastEnqueuedTime;
GStreamerMediaSample* sample = static_cast<GStreamerMediaSample*>(mediaSample.get());
if (sample->sample() && gst_sample_get_buffer(sample->sample())) {
GRefPtr<GstSample> gstSample = sample->sample();
GstBuffer* buffer = gst_sample_get_buffer(gstSample.get());
lastEnqueuedTime = sample->presentationTime();
GST_BUFFER_FLAG_UNSET(buffer, GST_BUFFER_FLAG_DECODE_ONLY);
pushSample(GST_APP_SRC(appsrc), gstSample.get());
stream->lastEnqueuedTime = lastEnqueuedTime;
}
}
GstElement* PlaybackPipeline::pipeline()
{
if (!m_webKitMediaSrc || !GST_ELEMENT_PARENT(GST_ELEMENT(m_webKitMediaSrc.get())))
return nullptr;
return GST_ELEMENT_PARENT(GST_ELEMENT_PARENT(GST_ELEMENT(m_webKitMediaSrc.get())));
}
}
#endif // USE(GSTREAMER)