Deleted: trunk/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp (277262 => 277263)
--- trunk/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp 2021-05-10 12:17:46 UTC (rev 277262)
+++ trunk/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp 2021-05-10 15:41:25 UTC (rev 277263)
@@ -1,405 +0,0 @@
-/*
- * Copyright (C) 2014, 2015 Sebastian Dröge <[email protected]>
- * Copyright (C) 2016 Metrological Group B.V.
- * Copyright (C) 2016 Igalia S.L
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public License
- * aint with this library; see the file COPYING.LIB. If not, write to
- * the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
- * Boston, MA 02110-1301, USA.
- */
-
-#include "config.h"
-#include "PlaybackPipeline.h"
-
-#if ENABLE(VIDEO) && USE(GSTREAMER) && ENABLE(MEDIA_SOURCE)
-
-#include "AudioTrackPrivateGStreamer.h"
-#include "GStreamerCommon.h"
-#include "MediaSampleGStreamer.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/GRefPtr.h>
-#include <wtf/glib/GUniquePtr.h>
-#include <wtf/text/AtomString.h>
-
-GST_DEBUG_CATEGORY_EXTERN(webkit_mse_debug);
-#define GST_CAT_DEFAULT webkit_mse_debug
-
-static Stream* getStreamByTrackId(WebKitMediaSrc*, AtomString);
-static Stream* getStreamBySourceBufferPrivate(WebKitMediaSrc*, WebCore::SourceBufferPrivateGStreamer*);
-
-static Stream* getStreamByTrackId(WebKitMediaSrc* source, AtomString trackIdString)
-{
- // WebKitMediaSrc should be locked at this point.
- 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;
-}
-
-// FIXME: Use gst_app_src_push_sample() instead when we switch to the appropriate GStreamer version.
-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;
- }
-
- // gst_app_src_push_buffer() steals the reference, we need an additional one.
- 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)
-{
- ASSERT(m_webKitMediaSrc);
- 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::AddStatus::NotSupported;
- }
-
- GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "State %d", int(GST_STATE(m_webKitMediaSrc.get())));
-
- Stream* stream = new Stream{ };
- stream->parent = m_webKitMediaSrc.get();
-
- // Ensure ownership is not transfered to the bin. The appsrc element is managed by its parent Stream.
- stream->appsrc = GST_ELEMENT_CAST(gst_object_ref_sink(gst_element_factory_make("appsrc", nullptr)));
-
- stream->appsrcNeedDataFlag = false;
- stream->sourceBuffer = sourceBufferPrivate.get();
-
- // No track has been attached yet.
- stream->type = Invalid;
- 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, "format", GST_FORMAT_TIME, nullptr);
-
- GST_OBJECT_LOCK(m_webKitMediaSrc.get());
- priv->streams.append(stream);
- GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
-
- gst_bin_add(GST_BIN_CAST(m_webKitMediaSrc.get()), stream->appsrc);
- gst_element_sync_state_with_parent(stream->appsrc);
-
- return MediaSourcePrivate::AddStatus::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 = getStreamBySourceBufferPrivate(m_webKitMediaSrc.get(), sourceBufferPrivate.get());
- if (stream)
- priv->streams.removeFirst(stream);
- GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
-
- if (stream)
- webKitMediaSrcFreeStream(m_webKitMediaSrc.get(), stream);
-}
-
-void PlaybackPipeline::attachTrack(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate, RefPtr<TrackPrivateBase> trackPrivate, 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 char* mediaType = capsMediaType(caps);
- 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);
-
- GST_OBJECT_LOCK(webKitMediaSrc);
- stream->type = Unknown;
- GST_OBJECT_UNLOCK(webKitMediaSrc);
-
- GRefPtr<GstPad> 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));
- webKitMediaSrcLinkSourcePad(sourcePad.get(), caps, stream);
-
- ASSERT(stream->parent->priv->mediaPlayerPrivate);
- int signal = -1;
-
- GST_OBJECT_LOCK(webKitMediaSrc);
- if (doCapsHaveType(caps, GST_AUDIO_CAPS_TYPE_PREFIX)) {
- stream->type = Audio;
- stream->parent->priv->numberOfAudioStreams++;
- signal = SIGNAL_AUDIO_CHANGED;
- stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
- } else if (doCapsHaveType(caps, GST_VIDEO_CAPS_TYPE_PREFIX)) {
- stream->type = Video;
- stream->parent->priv->numberOfVideoStreams++;
- signal = SIGNAL_VIDEO_CHANGED;
- stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
- } else if (doCapsHaveType(caps, GST_TEXT_CAPS_TYPE_PREFIX)) {
- 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)
-{
- 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);
-
- int signal = -1;
-
- GST_OBJECT_LOCK(webKitMediaSrc);
- if (doCapsHaveType(caps, GST_AUDIO_CAPS_TYPE_PREFIX)) {
- ASSERT(stream->type == Audio);
- signal = SIGNAL_AUDIO_CHANGED;
- stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
- } else if (doCapsHaveType(caps, GST_VIDEO_CAPS_TYPE_PREFIX)) {
- ASSERT(stream->type == Video);
- signal = SIGNAL_VIDEO_CHANGED;
- stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
- } else if (doCapsHaveType(caps, GST_TEXT_CAPS_TYPE_PREFIX)) {
- 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::notifyDurationChanged()
-{
- ASSERT(m_webKitMediaSrc);
- gst_element_post_message(GST_ELEMENT(m_webKitMediaSrc.get()), gst_message_new_duration_changed(GST_OBJECT(m_webKitMediaSrc.get())));
- // WebKitMediaSrc will ask MediaPlayerPrivateGStreamerMSE for the new duration later, when somebody asks for it.
-}
-
-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(AtomString 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();
- GstElement* appsrc = stream->appsrc;
- GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
-
- if (!appsrc)
- return;
-
- gint64 position = GST_CLOCK_TIME_NONE;
- GRefPtr<GstQuery> query = adoptGRef(gst_query_new_position(GST_FORMAT_TIME));
- if (gst_element_query(pipeline(), query.get()))
- gst_query_parse_position(query.get(), 0, &position);
-
- GST_TRACE("Position: %" GST_TIME_FORMAT, GST_TIME_ARGS(position));
-
- if (static_cast<guint64>(position) == GST_CLOCK_TIME_NONE) {
- GST_DEBUG("Can't determine position, avoiding flush");
- return;
- }
-
- if (!gst_element_send_event(GST_ELEMENT(appsrc), gst_event_new_flush_start())) {
- GST_WARNING("Failed to send flush-start event for trackId=%s", trackId.string().utf8().data());
- }
-
- if (!gst_element_send_event(GST_ELEMENT(appsrc), gst_event_new_flush_stop(false))) {
- GST_WARNING("Failed to send flush-stop event for trackId=%s", trackId.string().utf8().data());
- }
-
- GST_DEBUG("trackId=%s flushed", trackId.string().utf8().data());
-}
-
-void PlaybackPipeline::enqueueSample(Ref<MediaSample>&& mediaSample)
-{
- ASSERT(WTF::isMainThread());
-
- AtomString 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())),
- GST_TIME_ARGS(WebCore::toGstClockTime(mediaSample->duration())));
-
- // No need to lock to access the Stream here because the only chance of conflict with this read and with the usage
- // of the sample fields done in this method would be the deletion of the stream. However, that operation can only
- // happen in the main thread, but we're already there. Therefore there's no conflict and locking would only cause
- // a performance penalty on the readers working in other threads.
- 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;
- }
-
- // This field doesn't change after creation, no need to lock.
- GstElement* appsrc = stream->appsrc;
-
- // Only modified by the main thread, no need to lock.
- MediaTime lastEnqueuedTime = stream->lastEnqueuedTime;
-
- ASSERT(mediaSample->platformSample().type == PlatformSample::GStreamerSampleType);
- GRefPtr<GstSample> gstSample = mediaSample->platformSample().sample.gstSample;
- if (gstSample && gst_sample_get_buffer(gstSample.get())) {
- GstBuffer* buffer = gst_sample_get_buffer(gstSample.get());
- lastEnqueuedTime = mediaSample->presentationTime();
-
- GST_BUFFER_FLAG_UNSET(buffer, GST_BUFFER_FLAG_DECODE_ONLY);
- pushSample(GST_APP_SRC(appsrc), gstSample.get());
- // gst_app_src_push_sample() uses transfer-none for gstSample.
-
- stream->lastEnqueuedTime = lastEnqueuedTime;
- }
-}
-
-void PlaybackPipeline::allSamplesInTrackEnqueued(const AtomString& trackId)
-{
- Stream* stream = getStreamByTrackId(m_webKitMediaSrc.get(), trackId);
- gst_app_src_end_of_stream(GST_APP_SRC(stream->appsrc));
-}
-
-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())));
-}
-
-} // namespace WebCore.
-
-#endif // USE(GSTREAMER)
