Attachment 292415 Details for Bug 162901 – Patch

[patch] Patch

bug-162901-20161021213538.patch (text/plain), 24.99 KB, created by Enrique Ocaña on 2016-10-21 14:39:18 PDT

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Creator: Enrique Ocaña

Created: 2016-10-21 14:39:18 PDT

Size: 24.99 KB

patch

obsolete

>Subversion Revision: 207639
>diff --git a/Source/WebCore/ChangeLog b/Source/WebCore/ChangeLog
>index e8b88bb343b550bddfde5b6666e1b205f6c9bd56..6bee1096a47ba75e2d1db27f0b029a3961f8a134 100644
>--- a/Source/WebCore/ChangeLog
>+++ b/Source/WebCore/ChangeLog
>@@ -1,5 +1,35 @@
> 2016-10-04  Enrique OcaÃ±a GonzÃ¡lez  <eocanha@igalia.com>
> 
>+        [GStreamer][MSE] Playback pipeline
>+        https://bugs.webkit.org/show_bug.cgi?id=162901
>+
>+        Reviewed by Xabier Rodriguez-Calvar.
>+
>+        Encapsulate the responsibility to interact with the GStreamer playback pipeline and the
>+        different streams managed by the WebKitMediaSrc element into a new PlaybackPipeline class.
>+
>+        * platform/graphics/gstreamer/mse/PlaybackPipeline.cpp: Added.
>+        (getStreamByTrackId):
>+        (getStreamBySourceBufferPrivate):
>+        (pushSample):
>+        (WebCore::PlaybackPipeline::create):
>+        (WebCore::PlaybackPipeline::PlaybackPipeline):
>+        (WebCore::PlaybackPipeline::~PlaybackPipeline):
>+        (WebCore::PlaybackPipeline::setWebKitMediaSrc):
>+        (WebCore::PlaybackPipeline::webKitMediaSrc):
>+        (WebCore::PlaybackPipeline::addSourceBuffer):
>+        (WebCore::PlaybackPipeline::removeSourceBuffer):
>+        (WebCore::PlaybackPipeline::attachTrack):
>+        (WebCore::PlaybackPipeline::reattachTrack):
>+        (WebCore::PlaybackPipeline::notifyDurationChanged):
>+        (WebCore::PlaybackPipeline::markEndOfStream):
>+        (WebCore::PlaybackPipeline::flushAndEnqueueNonDisplayingSamples):
>+        (WebCore::PlaybackPipeline::enqueueSample):
>+        (WebCore::PlaybackPipeline::pipeline):
>+        * platform/graphics/gstreamer/mse/PlaybackPipeline.h: Added.
>+
>+2016-10-04  Enrique OcaÃ±a GonzÃ¡lez  <eocanha@igalia.com>
>+
>         [GStreamer][MSE][EME] MSE specialization of Media Player Private for GStreamer
>         https://bugs.webkit.org/show_bug.cgi?id=162900
> 
>diff --git a/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp b/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp
>new file mode 100644
>index 0000000000000000000000000000000000000000..f1c8aba50e9fa326358553a4ff5b1973b7ed53ed
>--- /dev/null
>+++ b/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.cpp
>@@ -0,0 +1,484 @@
>+/*
>+ * Copyright (C) 2014, 2015 Sebastian DrÃ¶ge <sebastian@centricular.com>
>+ * 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 "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)
>+{
>+    // 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)
>+{
>+    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 { 0 };
>+    stream->parent = m_webKitMediaSrc.get();
>+    stream->appsrc = gst_element_factory_make("appsrc", nullptr);
>+    stream->appsrcNeedDataFlag = false;
>+    stream->sourceBuffer = sourceBufferPrivate.get();
>+
>+    // No track has been attached yet.
>+    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;
>+        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)
>+{
>+    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::notifyDurationChanged()
>+{
>+    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::flushAndEnqueueNonDisplayingSamples(Vector<RefPtr<MediaSample>> samples)
>+{
>+    ASSERT(WTF::isMainThread());
>+
>+    if (!samples.size()) {
>+        GST_DEBUG("No samples, trackId unknown");
>+        return;
>+    }
>+
>+    AtomicString trackId = samples[0]->trackID();
>+    GST_DEBUG("flushAndEnqueueNonDisplayingSamples: trackId=%s PTS[0]=%f ... PTS[n]=%f", trackId.string().utf8().data(), samples[0]->presentationTime().toFloat(), samples[samples.size()-1]->presentationTime().toFloat());
>+
>+    GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "Flushing and re-enqueing %zu samples for stream %s", samples.size(), 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;
>+    }
>+
>+    if (!stream->sourceBuffer->isReadyForMoreSamples(trackId)) {
>+        GST_DEBUG("flushAndEnqueueNonDisplayingSamples: skip adding new sample for trackId=%s, SB is not ready yet", trackId.string().utf8().data());
>+        GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
>+        return;
>+    }
>+
>+    GstElement* appsrc = stream->appsrc;
>+    MediaTime lastEnqueuedTime = stream->lastEnqueuedTime;
>+    GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
>+
>+    if (!m_webKitMediaSrc->priv->mediaPlayerPrivate->seeking())
>+        GST_DEBUG("flushAndEnqueueNonDisplayingSamples: trackId=%s pipeline needs flushing.", trackId.string().utf8().data());
>+
>+    for (const RefPtr<MediaSample>& sample : samples) {
>+        GStreamerMediaSample* gStreamerMediaSample = static_cast<GStreamerMediaSample*>(sample.get());
>+        GstBuffer* buffer = nullptr;
>+        if (gStreamerMediaSample->sample())
>+            buffer = gst_sample_get_buffer(gStreamerMediaSample->sample());
>+        if (buffer) {
>+            GstSample* gstSample = gst_sample_ref(gStreamerMediaSample->sample());
>+            lastEnqueuedTime = gStreamerMediaSample->presentationTime();
>+
>+            GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_FLAG_DECODE_ONLY);
>+            pushSample(GST_APP_SRC(appsrc), gstSample);
>+            // gst_app_src_push_sample() uses transfer-none for gstSample.
>+
>+            gst_sample_unref(gstSample);
>+        }
>+    }
>+    GST_OBJECT_LOCK(m_webKitMediaSrc.get());
>+    stream->lastEnqueuedTime = lastEnqueuedTime;
>+    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());
>+        // gst_app_src_push_sample() uses transfer-none for gstSample.
>+
>+        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())));
>+}
>+
>+} // namespace WebCore.
>+
>+#endif // USE(GSTREAMER)
>diff --git a/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.h b/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.h
>new file mode 100644
>index 0000000000000000000000000000000000000000..810fb13cfb67d303f33664bb0526d7b65f9ec1fa
>--- /dev/null
>+++ b/Source/WebCore/platform/graphics/gstreamer/mse/PlaybackPipeline.h
>@@ -0,0 +1,79 @@
>+/*
>+ * 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.
>+ */
>+
>+#pragma once
>+
>+#if ENABLE(VIDEO) && USE(GSTREAMER) && ENABLE(MEDIA_SOURCE)
>+
>+// PlaybackPipeline is (sort of) a friend class of WebKitMediaSourceGStreamer.
>+
>+#include "WebKitMediaSourceGStreamer.h"
>+#include "WebKitMediaSourceGStreamerPrivate.h"
>+
>+#include <gst/gst.h>
>+#include <wtf/Condition.h>
>+#include <wtf/glib/GRefPtr.h>
>+
>+namespace WTF {
>+template<> GRefPtr<WebKitMediaSrc> adoptGRef(WebKitMediaSrc*);
>+template<> WebKitMediaSrc* refGPtr<WebKitMediaSrc>(WebKitMediaSrc*);
>+template<> void derefGPtr<WebKitMediaSrc>(WebKitMediaSrc*);
>+};
>+
>+namespace WebCore {
>+
>+class ContentType;
>+class SourceBufferPrivateGStreamer;
>+class MediaSourceGStreamer;
>+
>+class PlaybackPipeline: public RefCounted<PlaybackPipeline> {
>+public:
>+    static Ref<PlaybackPipeline> create()
>+    {
>+        return adoptRef(*new PlaybackPipeline());
>+    }
>+
>+    virtual ~PlaybackPipeline() = default;
>+
>+    void setWebKitMediaSrc(WebKitMediaSrc*);
>+    WebKitMediaSrc* webKitMediaSrc();
>+
>+    MediaSourcePrivate::AddStatus addSourceBuffer(RefPtr<SourceBufferPrivateGStreamer>);
>+    void removeSourceBuffer(RefPtr<SourceBufferPrivateGStreamer>);
>+    void attachTrack(RefPtr<SourceBufferPrivateGStreamer>, RefPtr<TrackPrivateBase>, GstStructure*, GstCaps*);
>+    void reattachTrack(RefPtr<SourceBufferPrivateGStreamer>, RefPtr<TrackPrivateBase>);
>+    void notifyDurationChanged();
>+
>+    // From MediaSourceGStreamer.
>+    void markEndOfStream(MediaSourcePrivate::EndOfStreamStatus);
>+
>+    // From SourceBufferPrivateGStreamer.
>+    void flushAndEnqueueNonDisplayingSamples(Vector<RefPtr<MediaSample>>);
>+    void enqueueSample(RefPtr<MediaSample>);
>+
>+    GstElement* pipeline();
>+private:
>+    PlaybackPipeline() = default;
>+    GRefPtr<WebKitMediaSrc> m_webKitMediaSrc;
>+};
>+
>+} // namespace WebCore.
>+
>+#endif // USE(GSTREAMER)

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Attachments on bug 162901: 290595 | 291764 | 292410 | 292415 | 292782 | 292895