Google Git
Sign in
webkit / WebKit / 68af9eb4847afe6fa034ef8468a4fa2ee76c05b3 / . / Source / WebCore / platform / graphics / gstreamer / MediaPlayerPrivateGStreamer.cpp
blob: 36a95262ef4277a9acb5f6eeabb63e85c4188645 [file] [log] [blame]
/*
* Copyright (C) 2007, 2009 Apple Inc. All rights reserved.
* Copyright (C) 2007 Collabora Ltd. All rights reserved.
* Copyright (C) 2007 Alp Toker <alp@atoker.com>
* Copyright (C) 2009 Gustavo Noronha Silva <gns@gnome.org>
* Copyright (C) 2009, 2010, 2011, 2012, 2013, 2015, 2016 Igalia S.L
* Copyright (C) 2014 Cable Television Laboratories, Inc.
* Copyright (C) 2015, 2016 Metrological Group B.V.
*
* 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 "MediaPlayerPrivateGStreamer.h"
#if ENABLE(VIDEO) && USE(GSTREAMER)
#include "FileSystem.h"
#include "GStreamerCommon.h"
#include "HTTPHeaderNames.h"
#include "MIMETypeRegistry.h"
#include "MediaPlayer.h"
#include "MediaPlayerRequestInstallMissingPluginsCallback.h"
#include "NotImplemented.h"
#include "SecurityOrigin.h"
#include "TimeRanges.h"
#include "URL.h"
#include "WebKitWebSourceGStreamer.h"
#include <glib.h>
#include <gst/gst.h>
#include <gst/pbutils/missing-plugins.h>
#include <limits>
#include <wtf/HexNumber.h>
#include <wtf/MediaTime.h>
#include <wtf/NeverDestroyed.h>
#include <wtf/StringPrintStream.h>
#include <wtf/glib/GUniquePtr.h>
#include <wtf/glib/RunLoopSourcePriority.h>
#include <wtf/text/CString.h>
#if ENABLE(VIDEO_TRACK)
#include "AudioTrackPrivateGStreamer.h"
#include "InbandMetadataTextTrackPrivateGStreamer.h"
#include "InbandTextTrackPrivateGStreamer.h"
#include "TextCombinerGStreamer.h"
#include "TextSinkGStreamer.h"
#include "VideoTrackPrivateGStreamer.h"
#endif
#if ENABLE(VIDEO_TRACK) && USE(GSTREAMER_MPEGTS)
#define GST_USE_UNSTABLE_API
#include <gst/mpegts/mpegts.h>
#undef GST_USE_UNSTABLE_API
#endif
#include <gst/audio/streamvolume.h>
#if ENABLE(MEDIA_SOURCE)
#include "MediaSource.h"
#include "WebKitMediaSourceGStreamer.h"
#endif
#if ENABLE(WEB_AUDIO)
#include "AudioSourceProviderGStreamer.h"
#endif
GST_DEBUG_CATEGORY_EXTERN(webkit_media_player_debug);
#define GST_CAT_DEFAULT webkit_media_player_debug
namespace WebCore {
using namespace std;
static void busMessageCallback(GstBus*, GstMessage* message, MediaPlayerPrivateGStreamer* player)
{
player->handleMessage(message);
}
void MediaPlayerPrivateGStreamer::setAudioStreamPropertiesCallback(MediaPlayerPrivateGStreamer* player, GObject* object)
{
player->setAudioStreamProperties(object);
}
void MediaPlayerPrivateGStreamer::setAudioStreamProperties(GObject* object)
{
if (g_strcmp0(G_OBJECT_TYPE_NAME(object), "GstPulseSink"))
return;
const char* role = m_player->client().mediaPlayerIsVideo() ? "video" : "music";
GstStructure* structure = gst_structure_new("stream-properties", "media.role", G_TYPE_STRING, role, nullptr);
g_object_set(object, "stream-properties", structure, nullptr);
gst_structure_free(structure);
GUniquePtr<gchar> elementName(gst_element_get_name(GST_ELEMENT(object)));
GST_DEBUG("Set media.role as %s at %s", role, elementName.get());
}
void MediaPlayerPrivateGStreamer::registerMediaEngine(MediaEngineRegistrar registrar)
{
if (isAvailable())
registrar([](MediaPlayer* player) { return std::make_unique<MediaPlayerPrivateGStreamer>(player); },
getSupportedTypes, supportsType, nullptr, nullptr, nullptr, supportsKeySystem);
}
bool MediaPlayerPrivateGStreamer::isAvailable()
{
if (!MediaPlayerPrivateGStreamerBase::initializeGStreamerAndRegisterWebKitElements())
return false;
GRefPtr<GstElementFactory> factory = adoptGRef(gst_element_factory_find("playbin"));
return factory;
}
MediaPlayerPrivateGStreamer::MediaPlayerPrivateGStreamer(MediaPlayer* player)
: MediaPlayerPrivateGStreamerBase(player)
, m_buffering(false)
, m_bufferingPercentage(0)
, m_canFallBackToLastFinishedSeekPosition(false)
, m_changingRate(false)
, m_downloadFinished(false)
, m_errorOccured(false)
, m_isEndReached(false)
, m_isStreaming(false)
, m_durationAtEOS(MediaTime::invalidTime())
, m_paused(true)
, m_playbackRate(1)
, m_requestedState(GST_STATE_VOID_PENDING)
, m_resetPipeline(false)
, m_seeking(false)
, m_seekIsPending(false)
, m_seekTime(MediaTime::invalidTime())
, m_source(nullptr)
, m_volumeAndMuteInitialized(false)
, m_mediaLocations(nullptr)
, m_mediaLocationCurrentIndex(0)
, m_playbackRatePause(false)
, m_timeOfOverlappingSeek(MediaTime::invalidTime())
, m_lastPlaybackRate(1)
, m_fillTimer(*this, &MediaPlayerPrivateGStreamer::fillTimerFired)
, m_maxTimeLoaded(MediaTime::zeroTime())
, m_preload(player->preload())
, m_delayingLoad(false)
, m_maxTimeLoadedAtLastDidLoadingProgress(MediaTime::zeroTime())
, m_hasVideo(false)
, m_hasAudio(false)
, m_readyTimerHandler(RunLoop::main(), this, &MediaPlayerPrivateGStreamer::readyTimerFired)
, m_totalBytes(0)
, m_preservesPitch(false)
{
#if USE(GLIB)
m_readyTimerHandler.setPriority(G_PRIORITY_DEFAULT_IDLE);
#endif
}
MediaPlayerPrivateGStreamer::~MediaPlayerPrivateGStreamer()
{
GST_DEBUG("Disposing player");
#if ENABLE(VIDEO_TRACK)
for (auto& track : m_audioTracks.values())
track->disconnect();
for (auto& track : m_textTracks.values())
track->disconnect();
for (auto& track : m_videoTracks.values())
track->disconnect();
#endif
if (m_fillTimer.isActive())
m_fillTimer.stop();
if (m_mediaLocations) {
gst_structure_free(m_mediaLocations);
m_mediaLocations = nullptr;
}
if (WEBKIT_IS_WEB_SRC(m_source.get()) && GST_OBJECT_PARENT(m_source.get()))
g_signal_handlers_disconnect_by_func(GST_ELEMENT_PARENT(m_source.get()), reinterpret_cast<gpointer>(uriDecodeBinElementAddedCallback), this);
if (m_autoAudioSink)
g_signal_handlers_disconnect_by_func(G_OBJECT(m_autoAudioSink.get()),
reinterpret_cast<gpointer>(setAudioStreamPropertiesCallback), this);
m_readyTimerHandler.stop();
for (auto& missingPluginCallback : m_missingPluginCallbacks) {
if (missingPluginCallback)
missingPluginCallback->invalidate();
}
m_missingPluginCallbacks.clear();
if (m_videoSink) {
GRefPtr<GstPad> videoSinkPad = adoptGRef(gst_element_get_static_pad(m_videoSink.get(), "sink"));
g_signal_handlers_disconnect_matched(videoSinkPad.get(), G_SIGNAL_MATCH_DATA, 0, 0, nullptr, nullptr, this);
}
if (m_pipeline) {
GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline.get())));
ASSERT(bus);
g_signal_handlers_disconnect_by_func(bus.get(), gpointer(busMessageCallback), this);
gst_bus_remove_signal_watch(bus.get());
gst_bus_set_sync_handler(bus.get(), nullptr, nullptr, nullptr);
g_signal_handlers_disconnect_matched(m_pipeline.get(), G_SIGNAL_MATCH_DATA, 0, 0, nullptr, nullptr, this);
}
}
static void convertToInternalProtocol(URL& url)
{
if (url.protocolIsInHTTPFamily() || url.protocolIsBlob())
url.setProtocol("webkit+" + url.protocol());
}
void MediaPlayerPrivateGStreamer::setPlaybinURL(const URL& url)
{
// Clean out everything after file:// url path.
String cleanURLString(url.string());
if (url.isLocalFile())
cleanURLString = cleanURLString.substring(0, url.pathEnd());
m_url = URL(URL(), cleanURLString);
convertToInternalProtocol(m_url);
GST_INFO("Load %s", m_url.string().utf8().data());
g_object_set(m_pipeline.get(), "uri", m_url.string().utf8().data(), nullptr);
}
void MediaPlayerPrivateGStreamer::load(const String& urlString)
{
// FIXME: This method is still called even if supportsType() returned
// IsNotSupported. This would deserve more investigation but meanwhile make
// sure we don't ever try to play animated gif assets.
if (m_player->contentMIMEType() == "image/gif") {
loadingFailed(MediaPlayer::FormatError);
return;
}
if (!MediaPlayerPrivateGStreamerBase::initializeGStreamerAndRegisterWebKitElements())
return;
URL url(URL(), urlString);
if (url.isBlankURL())
return;
if (!m_pipeline)
createGSTPlayBin(isMediaSource() ? "playbin" : nullptr);
if (m_fillTimer.isActive())
m_fillTimer.stop();
ASSERT(m_pipeline);
setPlaybinURL(url);
GST_DEBUG("preload: %s", convertEnumerationToString(m_preload).utf8().data());
if (m_preload == MediaPlayer::None) {
GST_INFO("Delaying load.");
m_delayingLoad = true;
}
// Reset network and ready states. Those will be set properly once
// the pipeline pre-rolled.
m_networkState = MediaPlayer::Loading;
m_player->networkStateChanged();
m_readyState = MediaPlayer::HaveNothing;
m_player->readyStateChanged();
m_volumeAndMuteInitialized = false;
m_durationAtEOS = MediaTime::invalidTime();
if (!m_delayingLoad)
commitLoad();
}
#if ENABLE(MEDIA_SOURCE)
void MediaPlayerPrivateGStreamer::load(const String&, MediaSourcePrivateClient*)
{
// Properly fail so the global MediaPlayer tries to fallback to the next MediaPlayerPrivate.
m_networkState = MediaPlayer::FormatError;
m_player->networkStateChanged();
}
#endif
#if ENABLE(MEDIA_STREAM)
void MediaPlayerPrivateGStreamer::load(MediaStreamPrivate&)
{
// Properly fail so the global MediaPlayer tries to fallback to the next MediaPlayerPrivate.
m_networkState = MediaPlayer::FormatError;
m_player->networkStateChanged();
notImplemented();
}
#endif
void MediaPlayerPrivateGStreamer::commitLoad()
{
ASSERT(!m_delayingLoad);
GST_DEBUG("Committing load.");
// GStreamer needs to have the pipeline set to a paused state to
// start providing anything useful.
changePipelineState(GST_STATE_PAUSED);
setDownloadBuffering();
updateStates();
}
MediaTime MediaPlayerPrivateGStreamer::playbackPosition() const
{
if (m_isEndReached) {
// Position queries on a null pipeline return 0. If we're at
// the end of the stream the pipeline is null but we want to
// report either the seek time or the duration because this is
// what the Media element spec expects us to do.
if (m_seeking)
return m_seekTime;
MediaTime duration = durationMediaTime();
return duration.isInvalid() ? MediaTime::zeroTime() : duration;
}
// Position is only available if no async state change is going on and the state is either paused or playing.
gint64 position = GST_CLOCK_TIME_NONE;
GstQuery* query= gst_query_new_position(GST_FORMAT_TIME);
if (gst_element_query(m_pipeline.get(), query))
gst_query_parse_position(query, 0, &position);
gst_query_unref(query);
GST_LOG("Position %" GST_TIME_FORMAT, GST_TIME_ARGS(position));
MediaTime playbackPosition = MediaTime::zeroTime();
GstClockTime gstreamerPosition = static_cast<GstClockTime>(position);
if (GST_CLOCK_TIME_IS_VALID(gstreamerPosition))
playbackPosition = MediaTime(gstreamerPosition, GST_SECOND);
else if (m_canFallBackToLastFinishedSeekPosition)
playbackPosition = m_seekTime;
return playbackPosition;
}
void MediaPlayerPrivateGStreamer::readyTimerFired()
{
GST_DEBUG("In READY for too long. Releasing pipeline resources.");
changePipelineState(GST_STATE_NULL);
}
bool MediaPlayerPrivateGStreamer::changePipelineState(GstState newState)
{
ASSERT(m_pipeline);
GstState currentState;
GstState pending;
gst_element_get_state(m_pipeline.get(), &currentState, &pending, 0);
if (currentState == newState || pending == newState) {
GST_DEBUG("Rejected state change to %s from %s with %s pending", gst_element_state_get_name(newState),
gst_element_state_get_name(currentState), gst_element_state_get_name(pending));
return true;
}
GST_DEBUG("Changing state change to %s from %s with %s pending", gst_element_state_get_name(newState),
gst_element_state_get_name(currentState), gst_element_state_get_name(pending));
#if USE(GSTREAMER_GL)
if (currentState == GST_STATE_READY && newState == GST_STATE_PAUSED)
ensureGLVideoSinkContext();
#endif
GstStateChangeReturn setStateResult = gst_element_set_state(m_pipeline.get(), newState);
GstState pausedOrPlaying = newState == GST_STATE_PLAYING ? GST_STATE_PAUSED : GST_STATE_PLAYING;
if (currentState != pausedOrPlaying && setStateResult == GST_STATE_CHANGE_FAILURE) {
return false;
}
// Create a timer when entering the READY state so that we can free resources
// if we stay for too long on READY.
// Also lets remove the timer if we request a state change for any state other than READY.
// See also https://bugs.webkit.org/show_bug.cgi?id=117354
if (newState == GST_STATE_READY && !m_readyTimerHandler.isActive()) {
// Max interval in seconds to stay in the READY state on manual
// state change requests.
static const Seconds readyStateTimerDelay { 1_min };
m_readyTimerHandler.startOneShot(readyStateTimerDelay);
} else if (newState != GST_STATE_READY)
m_readyTimerHandler.stop();
return true;
}
void MediaPlayerPrivateGStreamer::prepareToPlay()
{
GST_DEBUG("Prepare to play");
m_preload = MediaPlayer::Auto;
if (m_delayingLoad) {
m_delayingLoad = false;
commitLoad();
}
}
void MediaPlayerPrivateGStreamer::play()
{
if (!m_playbackRate) {
m_playbackRatePause = true;
return;
}
if (changePipelineState(GST_STATE_PLAYING)) {
m_isEndReached = false;
m_delayingLoad = false;
m_preload = MediaPlayer::Auto;
setDownloadBuffering();
GST_INFO("Play");
} else {
loadingFailed(MediaPlayer::Empty);
}
}
void MediaPlayerPrivateGStreamer::pause()
{
m_playbackRatePause = false;
GstState currentState, pendingState;
gst_element_get_state(m_pipeline.get(), &currentState, &pendingState, 0);
if (currentState < GST_STATE_PAUSED && pendingState <= GST_STATE_PAUSED)
return;
if (changePipelineState(GST_STATE_PAUSED))
GST_INFO("Pause");
else
loadingFailed(MediaPlayer::Empty);
}
MediaTime MediaPlayerPrivateGStreamer::durationMediaTime() const
{
if (!m_pipeline || m_errorOccured)
return MediaTime::invalidTime();
if (m_durationAtEOS.isValid())
return m_durationAtEOS;
// The duration query would fail on a not-prerolled pipeline.
if (GST_STATE(m_pipeline.get()) < GST_STATE_PAUSED)
return MediaTime::invalidTime();
gint64 timeLength = 0;
if (!gst_element_query_duration(m_pipeline.get(), GST_FORMAT_TIME, &timeLength) || !GST_CLOCK_TIME_IS_VALID(timeLength)) {
GST_DEBUG("Time duration query failed for %s", m_url.string().utf8().data());
return MediaTime::positiveInfiniteTime();
}
GST_LOG("Duration: %" GST_TIME_FORMAT, GST_TIME_ARGS(timeLength));
return MediaTime(timeLength, GST_SECOND);
// FIXME: handle 3.14.9.5 properly
}
MediaTime MediaPlayerPrivateGStreamer::currentMediaTime() const
{
if (!m_pipeline || m_errorOccured)
return MediaTime::invalidTime();
if (m_seeking)
return m_seekTime;
// Workaround for
// https://bugzilla.gnome.org/show_bug.cgi?id=639941 In GStreamer
// 0.10.35 basesink reports wrong duration in case of EOS and
// negative playback rate. There's no upstream accepted patch for
// this bug yet, hence this temporary workaround.
if (m_isEndReached && m_playbackRate < 0)
return MediaTime::invalidTime();
return playbackPosition();
}
void MediaPlayerPrivateGStreamer::seek(const MediaTime& mediaTime)
{
if (!m_pipeline)
return;
if (m_errorOccured)
return;
GST_INFO("[Seek] seek attempt to %s", toString(mediaTime).utf8().data());
// Avoid useless seeking.
if (mediaTime == currentMediaTime())
return;
MediaTime time = std::min(mediaTime, durationMediaTime());
if (isLiveStream())
return;
GST_INFO("[Seek] seeking to %s", toString(time).utf8().data());
if (m_seeking) {
m_timeOfOverlappingSeek = time;
if (m_seekIsPending) {
m_seekTime = time;
return;
}
}
GstState state;
GstStateChangeReturn getStateResult = gst_element_get_state(m_pipeline.get(), &state, nullptr, 0);
if (getStateResult == GST_STATE_CHANGE_FAILURE || getStateResult == GST_STATE_CHANGE_NO_PREROLL) {
GST_DEBUG("[Seek] cannot seek, current state change is %s", gst_element_state_change_return_get_name(getStateResult));
return;
}
if (getStateResult == GST_STATE_CHANGE_ASYNC || state < GST_STATE_PAUSED || m_isEndReached) {
m_seekIsPending = true;
if (m_isEndReached) {
GST_DEBUG("[Seek] reset pipeline");
m_resetPipeline = true;
if (!changePipelineState(GST_STATE_PAUSED))
loadingFailed(MediaPlayer::Empty);
}
} else {
// We can seek now.
if (!doSeek(time, m_player->rate(), static_cast<GstSeekFlags>(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE))) {
GST_DEBUG("[Seek] seeking to %s failed", toString(time).utf8().data());
return;
}
}
m_seeking = true;
m_seekTime = time;
m_isEndReached = false;
}
bool MediaPlayerPrivateGStreamer::doSeek(const MediaTime& position, float rate, GstSeekFlags seekType)
{
// Default values for rate >= 0.
MediaTime startTime = position, endTime = MediaTime::invalidTime();
// TODO: Should do more than that, need to notify the media source
// and probably flush the pipeline at least.
if (isMediaSource())
return true;
if (rate < 0) {
startTime = MediaTime::zeroTime();
// If we are at beginning of media, start from the end to
// avoid immediate EOS.
if (position < MediaTime::zeroTime())
endTime = durationMediaTime();
else
endTime = position;
}
if (!rate)
rate = 1.0;
return gst_element_seek(m_pipeline.get(), rate, GST_FORMAT_TIME, seekType,
GST_SEEK_TYPE_SET, toGstClockTime(startTime), GST_SEEK_TYPE_SET, toGstClockTime(endTime));
}
void MediaPlayerPrivateGStreamer::updatePlaybackRate()
{
if (!m_changingRate)
return;
GST_INFO("Set Rate to %f", m_playbackRate);
// Mute the sound if the playback rate is negative or too extreme and audio pitch is not adjusted.
bool mute = m_playbackRate <= 0 || (!m_preservesPitch && (m_playbackRate < 0.8 || m_playbackRate > 2));
GST_INFO(mute ? "Need to mute audio" : "Do not need to mute audio");
if (doSeek(playbackPosition(), m_playbackRate, static_cast<GstSeekFlags>(GST_SEEK_FLAG_FLUSH))) {
g_object_set(m_pipeline.get(), "mute", mute, nullptr);
m_lastPlaybackRate = m_playbackRate;
} else {
m_playbackRate = m_lastPlaybackRate;
GST_ERROR("Set rate to %f failed", m_playbackRate);
}
if (m_playbackRatePause) {
GstState state;
GstState pending;
gst_element_get_state(m_pipeline.get(), &state, &pending, 0);
if (state != GST_STATE_PLAYING && pending != GST_STATE_PLAYING)
changePipelineState(GST_STATE_PLAYING);
m_playbackRatePause = false;
}
m_changingRate = false;
m_player->rateChanged();
}
bool MediaPlayerPrivateGStreamer::paused() const
{
if (m_isEndReached) {
GST_DEBUG("Ignoring pause at EOS");
return true;
}
if (m_playbackRatePause) {
GST_DEBUG("Playback rate is 0, simulating PAUSED state");
return false;
}
GstState state;
gst_element_get_state(m_pipeline.get(), &state, nullptr, 0);
bool paused = state <= GST_STATE_PAUSED;
GST_DEBUG("Paused: %s", toString(paused).utf8().data());
return paused;
}
bool MediaPlayerPrivateGStreamer::seeking() const
{
return m_seeking;
}
#if GST_CHECK_VERSION(1, 10, 0)
void MediaPlayerPrivateGStreamer::updateTracks()
{
ASSERT(!m_isLegacyPlaybin);
bool useMediaSource = isMediaSource();
unsigned length = gst_stream_collection_get_size(m_streamCollection.get());
Vector<String> validAudioStreams;
Vector<String> validVideoStreams;
Vector<String> validTextStreams;
for (unsigned i = 0; i < length; i++) {
GRefPtr<GstStream> stream = gst_stream_collection_get_stream(m_streamCollection.get(), i);
String streamId(gst_stream_get_stream_id(stream.get()));
GstStreamType type = gst_stream_get_stream_type(stream.get());
GST_DEBUG("Inspecting %s track with ID %s", gst_stream_type_get_name(type), streamId.utf8().data());
if (type & GST_STREAM_TYPE_AUDIO) {
validAudioStreams.append(streamId);
#if ENABLE(VIDEO_TRACK)
if (!useMediaSource) {
unsigned localIndex = i - validVideoStreams.size() - validTextStreams.size();
if (localIndex < m_audioTracks.size()) {
if (m_audioTracks.contains(streamId))
continue;
}
RefPtr<AudioTrackPrivateGStreamer> track = AudioTrackPrivateGStreamer::create(createWeakPtr(), i, stream);
m_audioTracks.add(track->id(), track);
m_player->addAudioTrack(*track);
}
#endif
} else if (type & GST_STREAM_TYPE_VIDEO) {
validVideoStreams.append(streamId);
#if ENABLE(VIDEO_TRACK)
if (!useMediaSource) {
unsigned localIndex = i - validAudioStreams.size() - validTextStreams.size();
if (localIndex < m_videoTracks.size()) {
if (m_videoTracks.contains(streamId))
continue;
}
RefPtr<VideoTrackPrivateGStreamer> track = VideoTrackPrivateGStreamer::create(createWeakPtr(), i, stream);
m_videoTracks.add(track->id(), track);
m_player->addVideoTrack(*track);
}
#endif
} else if (type & GST_STREAM_TYPE_TEXT) {
validTextStreams.append(streamId);
#if ENABLE(VIDEO_TRACK)
if (!useMediaSource) {
unsigned localIndex = i - validVideoStreams.size() - validAudioStreams.size();
if (localIndex < m_textTracks.size()) {
if (m_textTracks.contains(streamId))
continue;
}
RefPtr<InbandTextTrackPrivateGStreamer> track = InbandTextTrackPrivateGStreamer::create(localIndex, stream);
m_textTracks.add(streamId, track);
m_player->addTextTrack(*track);
}
#endif
} else
GST_WARNING("Unknown track type found for stream %s", streamId.utf8().data());
}
GST_INFO("Media has %u video tracks, %u audio tracks and %u text tracks", validVideoStreams.size(), validAudioStreams.size(), validTextStreams.size());
bool oldHasAudio = m_hasAudio;
bool oldHasVideo = m_hasVideo;
m_hasAudio = !validAudioStreams.isEmpty();
m_hasVideo = !validVideoStreams.isEmpty();
if ((oldHasVideo != m_hasVideo) || (oldHasAudio != m_hasAudio))
m_player->characteristicChanged();
if (m_hasVideo)
m_player->sizeChanged();
if (useMediaSource) {
GST_DEBUG("Tracks managed by source element. Bailing out now.");
m_player->client().mediaPlayerEngineUpdated(m_player);
return;
}
#if ENABLE(VIDEO_TRACK)
purgeInvalidAudioTracks(validAudioStreams);
purgeInvalidVideoTracks(validVideoStreams);
purgeInvalidTextTracks(validTextStreams);
#endif
m_player->client().mediaPlayerEngineUpdated(m_player);
}
#endif
void MediaPlayerPrivateGStreamer::enableTrack(TrackPrivateBaseGStreamer::TrackType trackType, unsigned index)
{
const char* propertyName;
const char* trackTypeAsString;
GList* selectedStreams = nullptr;
switch (trackType) {
case TrackPrivateBaseGStreamer::TrackType::Audio:
propertyName = "current-audio";
trackTypeAsString = "audio";
if (!m_currentTextStreamId.isEmpty())
selectedStreams = g_list_append(selectedStreams, g_strdup(m_currentTextStreamId.utf8().data()));
if (!m_currentVideoStreamId.isEmpty())
selectedStreams = g_list_append(selectedStreams, g_strdup(m_currentVideoStreamId.utf8().data()));
break;
case TrackPrivateBaseGStreamer::TrackType::Video:
propertyName = "current-video";
trackTypeAsString = "video";
if (!m_currentAudioStreamId.isEmpty())
selectedStreams = g_list_append(selectedStreams, g_strdup(m_currentAudioStreamId.utf8().data()));
if (!m_currentTextStreamId.isEmpty())
selectedStreams = g_list_append(selectedStreams, g_strdup(m_currentTextStreamId.utf8().data()));
break;
case TrackPrivateBaseGStreamer::TrackType::Text:
propertyName = "current-text";
trackTypeAsString = "text";
if (!m_currentAudioStreamId.isEmpty())
selectedStreams = g_list_append(selectedStreams, g_strdup(m_currentAudioStreamId.utf8().data()));
if (!m_currentVideoStreamId.isEmpty())
selectedStreams = g_list_append(selectedStreams, g_strdup(m_currentVideoStreamId.utf8().data()));
break;
case TrackPrivateBaseGStreamer::TrackType::Unknown:
default:
ASSERT_NOT_REACHED();
}
GST_INFO("Enabling %s track with index: %lu", trackTypeAsString, index);
// FIXME: Remove isMediaSource() test below when fixing https://bugs.webkit.org/show_bug.cgi?id=182531
if (m_isLegacyPlaybin || isMediaSource()) {
GstElement* element = isMediaSource() ? m_source.get() : m_pipeline.get();
g_object_set(element, propertyName, index, nullptr);
}
#if GST_CHECK_VERSION(1, 10, 0)
else {
GstStream* stream = gst_stream_collection_get_stream(m_streamCollection.get(), index);
if (stream) {
String streamId = gst_stream_get_stream_id(stream);
selectedStreams = g_list_append(selectedStreams, g_strdup(streamId.utf8().data()));
} else
GST_WARNING("%s stream %lu not found", trackTypeAsString, index);
// TODO: MSE GstStream API support: https://bugs.webkit.org/show_bug.cgi?id=182531
gst_element_send_event(m_pipeline.get(), gst_event_new_select_streams(selectedStreams));
}
#endif
if (selectedStreams)
g_list_free_full(selectedStreams, reinterpret_cast<GDestroyNotify>(g_free));
}
void MediaPlayerPrivateGStreamer::videoChangedCallback(MediaPlayerPrivateGStreamer* player)
{
player->m_notifier->notify(MainThreadNotification::VideoChanged, [player] { player->notifyPlayerOfVideo(); });
}
void MediaPlayerPrivateGStreamer::notifyPlayerOfVideo()
{
if (UNLIKELY(!m_pipeline || !m_source))
return;
ASSERT(m_isLegacyPlaybin || isMediaSource());
gint numTracks = 0;
bool useMediaSource = isMediaSource();
GstElement* element = useMediaSource ? m_source.get() : m_pipeline.get();
g_object_get(element, "n-video", &numTracks, nullptr);
GST_INFO("Media has %d video tracks", numTracks);
bool oldHasVideo = m_hasVideo;
m_hasVideo = numTracks > 0;
if (oldHasVideo != m_hasVideo)
m_player->characteristicChanged();
if (m_hasVideo)
m_player->sizeChanged();
if (useMediaSource) {
GST_DEBUG("Tracks managed by source element. Bailing out now.");
m_player->client().mediaPlayerEngineUpdated(m_player);
return;
}
#if ENABLE(VIDEO_TRACK)
Vector<String> validVideoStreams;
for (gint i = 0; i < numTracks; ++i) {
GRefPtr<GstPad> pad;
g_signal_emit_by_name(m_pipeline.get(), "get-video-pad", i, &pad.outPtr(), nullptr);
ASSERT(pad);
String streamId = "V" + String::number(i);
validVideoStreams.append(streamId);
if (i < static_cast<gint>(m_videoTracks.size())) {
RefPtr<VideoTrackPrivateGStreamer> existingTrack = m_videoTracks.get(streamId);
if (existingTrack) {
existingTrack->setIndex(i);
if (existingTrack->pad() == pad)
continue;
}
}
RefPtr<VideoTrackPrivateGStreamer> track = VideoTrackPrivateGStreamer::create(createWeakPtr(), i, pad);
ASSERT(streamId == track->id());
m_videoTracks.add(streamId, track);
m_player->addVideoTrack(*track);
}
purgeInvalidVideoTracks(validVideoStreams);
#endif
m_player->client().mediaPlayerEngineUpdated(m_player);
}
void MediaPlayerPrivateGStreamer::videoSinkCapsChangedCallback(MediaPlayerPrivateGStreamer* player)
{
player->m_notifier->notify(MainThreadNotification::VideoCapsChanged, [player] { player->notifyPlayerOfVideoCaps(); });
}
void MediaPlayerPrivateGStreamer::notifyPlayerOfVideoCaps()
{
m_videoSize = IntSize();
m_player->client().mediaPlayerEngineUpdated(m_player);
}
void MediaPlayerPrivateGStreamer::audioChangedCallback(MediaPlayerPrivateGStreamer* player)
{
player->m_notifier->notify(MainThreadNotification::AudioChanged, [player] { player->notifyPlayerOfAudio(); });
}
void MediaPlayerPrivateGStreamer::notifyPlayerOfAudio()
{
if (UNLIKELY(!m_pipeline || !m_source))
return;
ASSERT(m_isLegacyPlaybin || isMediaSource());
gint numTracks = 0;
bool useMediaSource = isMediaSource();
GstElement* element = useMediaSource ? m_source.get() : m_pipeline.get();
g_object_get(element, "n-audio", &numTracks, nullptr);
GST_INFO("Media has %d audio tracks", numTracks);
bool oldHasAudio = m_hasAudio;
m_hasAudio = numTracks > 0;
if (oldHasAudio != m_hasAudio)
m_player->characteristicChanged();
if (useMediaSource) {
GST_DEBUG("Tracks managed by source element. Bailing out now.");
m_player->client().mediaPlayerEngineUpdated(m_player);
return;
}
#if ENABLE(VIDEO_TRACK)
Vector<String> validAudioStreams;
for (gint i = 0; i < numTracks; ++i) {
GRefPtr<GstPad> pad;
g_signal_emit_by_name(m_pipeline.get(), "get-audio-pad", i, &pad.outPtr(), nullptr);
ASSERT(pad);
String streamId = "A" + String::number(i);
validAudioStreams.append(streamId);
if (i < static_cast<gint>(m_audioTracks.size())) {
RefPtr<AudioTrackPrivateGStreamer> existingTrack = m_audioTracks.get(streamId);
if (existingTrack) {
existingTrack->setIndex(i);
if (existingTrack->pad() == pad)
continue;
}
}
RefPtr<AudioTrackPrivateGStreamer> track = AudioTrackPrivateGStreamer::create(createWeakPtr(), i, pad);
ASSERT(streamId == track->id());
m_audioTracks.add(streamId, track);
m_player->addAudioTrack(*track);
}
purgeInvalidAudioTracks(validAudioStreams);
#endif
m_player->client().mediaPlayerEngineUpdated(m_player);
}
#if ENABLE(VIDEO_TRACK)
void MediaPlayerPrivateGStreamer::textChangedCallback(MediaPlayerPrivateGStreamer* player)
{
player->m_notifier->notify(MainThreadNotification::TextChanged, [player] { player->notifyPlayerOfText(); });
}
void MediaPlayerPrivateGStreamer::notifyPlayerOfText()
{
if (UNLIKELY(!m_pipeline || !m_source))
return;
ASSERT(m_isLegacyPlaybin || isMediaSource());
gint numTracks = 0;
bool useMediaSource = isMediaSource();
GstElement* element = useMediaSource ? m_source.get() : m_pipeline.get();
g_object_get(element, "n-text", &numTracks, nullptr);
GST_INFO("Media has %d text tracks", numTracks);
if (useMediaSource) {
GST_DEBUG("Tracks managed by source element. Bailing out now.");
return;
}
Vector<String> validTextStreams;
for (gint i = 0; i < numTracks; ++i) {
GRefPtr<GstPad> pad;
g_signal_emit_by_name(m_pipeline.get(), "get-text-pad", i, &pad.outPtr(), nullptr);
ASSERT(pad);
// We can't assume the pad has a sticky event here like implemented in
// InbandTextTrackPrivateGStreamer because it might be emitted after the
// track was created. So fallback to a dummy stream ID like in the Audio
// and Video tracks.
String streamId = "T" + String::number(i);
validTextStreams.append(streamId);
if (i < static_cast<gint>(m_textTracks.size())) {
RefPtr<InbandTextTrackPrivateGStreamer> existingTrack = m_textTracks.get(streamId);
if (existingTrack) {
existingTrack->setIndex(i);
if (existingTrack->pad() == pad)
continue;
}
}
RefPtr<InbandTextTrackPrivateGStreamer> track = InbandTextTrackPrivateGStreamer::create(i, pad);
m_textTracks.add(streamId, track);
m_player->addTextTrack(*track);
}
purgeInvalidTextTracks(validTextStreams);
}
GstFlowReturn MediaPlayerPrivateGStreamer::newTextSampleCallback(MediaPlayerPrivateGStreamer* player)
{
player->newTextSample();
return GST_FLOW_OK;
}
void MediaPlayerPrivateGStreamer::newTextSample()
{
if (!m_textAppSink)
return;
GRefPtr<GstEvent> streamStartEvent = adoptGRef(
gst_pad_get_sticky_event(m_textAppSinkPad.get(), GST_EVENT_STREAM_START, 0));
GRefPtr<GstSample> sample;
g_signal_emit_by_name(m_textAppSink.get(), "pull-sample", &sample.outPtr(), nullptr);
ASSERT(sample);
if (streamStartEvent) {
bool found = FALSE;
const gchar* id;
gst_event_parse_stream_start(streamStartEvent.get(), &id);
for (auto& track : m_textTracks.values()) {
if (!strcmp(track->streamId().utf8().data(), id)) {
track->handleSample(sample);
found = true;
break;
}
}
if (!found)
GST_WARNING("Got sample with unknown stream ID %s.", id);
} else
GST_WARNING("Unable to handle sample with no stream start event.");
}
#endif
void MediaPlayerPrivateGStreamer::setRate(float rate)
{
// Higher rate causes crash.
rate = clampTo(rate, -20.0, 20.0);
// Avoid useless playback rate update.
if (m_playbackRate == rate) {
// and make sure that upper layers were notified if rate was set
if (!m_changingRate && m_player->rate() != m_playbackRate)
m_player->rateChanged();
return;
}
if (isLiveStream()) {
// notify upper layers that we cannot handle passed rate.
m_changingRate = false;
m_player->rateChanged();
return;
}
GstState state;
GstState pending;
m_playbackRate = rate;
m_changingRate = true;
gst_element_get_state(m_pipeline.get(), &state, &pending, 0);
if (!rate) {
m_changingRate = false;
m_playbackRatePause = true;
if (state != GST_STATE_PAUSED && pending != GST_STATE_PAUSED)
changePipelineState(GST_STATE_PAUSED);
return;
}
if ((state != GST_STATE_PLAYING && state != GST_STATE_PAUSED)
|| (pending == GST_STATE_PAUSED))
return;
updatePlaybackRate();
}
double MediaPlayerPrivateGStreamer::rate() const
{
return m_playbackRate;
}
void MediaPlayerPrivateGStreamer::setPreservesPitch(bool preservesPitch)
{
m_preservesPitch = preservesPitch;
}
std::unique_ptr<PlatformTimeRanges> MediaPlayerPrivateGStreamer::buffered() const
{
auto timeRanges = std::make_unique<PlatformTimeRanges>();
if (m_errorOccured || isLiveStream())
return timeRanges;
MediaTime mediaDuration = durationMediaTime();
if (!mediaDuration || mediaDuration.isPositiveInfinite())
return timeRanges;
GstQuery* query = gst_query_new_buffering(GST_FORMAT_PERCENT);
if (!gst_element_query(m_pipeline.get(), query)) {
gst_query_unref(query);
return timeRanges;
}
guint numBufferingRanges = gst_query_get_n_buffering_ranges(query);
for (guint index = 0; index < numBufferingRanges; index++) {
gint64 rangeStart = 0, rangeStop = 0;
if (gst_query_parse_nth_buffering_range(query, index, &rangeStart, &rangeStop))
timeRanges->add(MediaTime(rangeStart * toGstUnsigned64Time(mediaDuration) / GST_FORMAT_PERCENT_MAX, GST_SECOND),
MediaTime(rangeStop * toGstUnsigned64Time(mediaDuration) / GST_FORMAT_PERCENT_MAX, GST_SECOND));
}
// Fallback to the more general maxTimeLoaded() if no range has
// been found.
if (!timeRanges->length())
if (MediaTime loaded = maxTimeLoaded())
timeRanges->add(MediaTime::zeroTime(), loaded);
gst_query_unref(query);
return timeRanges;
}
void MediaPlayerPrivateGStreamer::handleMessage(GstMessage* message)
{
GUniqueOutPtr<GError> err;
GUniqueOutPtr<gchar> debug;
MediaPlayer::NetworkState error;
bool issueError = true;
bool attemptNextLocation = false;
const GstStructure* structure = gst_message_get_structure(message);
GstState requestedState, currentState;
m_canFallBackToLastFinishedSeekPosition = false;
if (structure) {
const gchar* messageTypeName = gst_structure_get_name(structure);
// Redirect messages are sent from elements, like qtdemux, to
// notify of the new location(s) of the media.
if (!g_strcmp0(messageTypeName, "redirect")) {
mediaLocationChanged(message);
return;
}
}
// We ignore state changes from internal elements. They are forwarded to playbin2 anyway.
bool messageSourceIsPlaybin = GST_MESSAGE_SRC(message) == reinterpret_cast<GstObject*>(m_pipeline.get());
GST_LOG("Message %s received from element %s", GST_MESSAGE_TYPE_NAME(message), GST_MESSAGE_SRC_NAME(message));
switch (GST_MESSAGE_TYPE(message)) {
case GST_MESSAGE_ERROR:
if (m_resetPipeline || !m_missingPluginCallbacks.isEmpty() || m_errorOccured)
break;
gst_message_parse_error(message, &err.outPtr(), &debug.outPtr());
GST_ERROR("Error %d: %s (url=%s)", err->code, err->message, m_url.string().utf8().data());
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_ALL, "webkit-video.error");
error = MediaPlayer::Empty;
if (g_error_matches(err.get(), GST_STREAM_ERROR, GST_STREAM_ERROR_CODEC_NOT_FOUND)
|| g_error_matches(err.get(), GST_STREAM_ERROR, GST_STREAM_ERROR_WRONG_TYPE)
|| g_error_matches(err.get(), GST_STREAM_ERROR, GST_STREAM_ERROR_FAILED)
|| g_error_matches(err.get(), GST_CORE_ERROR, GST_CORE_ERROR_MISSING_PLUGIN)
|| g_error_matches(err.get(), GST_RESOURCE_ERROR, GST_RESOURCE_ERROR_NOT_FOUND))
error = MediaPlayer::FormatError;
else if (g_error_matches(err.get(), GST_STREAM_ERROR, GST_STREAM_ERROR_TYPE_NOT_FOUND)) {
// Let the mediaPlayerClient handle the stream error, in
// this case the HTMLMediaElement will emit a stalled
// event.
GST_ERROR("Decode error, let the Media element emit a stalled event.");
m_loadingStalled = true;
break;
} else if (err->domain == GST_STREAM_ERROR) {
error = MediaPlayer::DecodeError;
attemptNextLocation = true;
} else if (err->domain == GST_RESOURCE_ERROR)
error = MediaPlayer::NetworkError;
if (attemptNextLocation)
issueError = !loadNextLocation();
if (issueError)
loadingFailed(error);
break;
case GST_MESSAGE_EOS:
didEnd();
break;
case GST_MESSAGE_ASYNC_DONE:
if (!messageSourceIsPlaybin || m_delayingLoad)
break;
asyncStateChangeDone();
break;
case GST_MESSAGE_STATE_CHANGED: {
if (!messageSourceIsPlaybin || m_delayingLoad)
break;
updateStates();
// Construct a filename for the graphviz dot file output.
GstState newState;
gst_message_parse_state_changed(message, &currentState, &newState, nullptr);
CString dotFileName = String::format("webkit-video.%s_%s", gst_element_state_get_name(currentState), gst_element_state_get_name(newState)).utf8();
GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(GST_BIN(m_pipeline.get()), GST_DEBUG_GRAPH_SHOW_ALL, dotFileName.data());
break;
}
case GST_MESSAGE_BUFFERING:
processBufferingStats(message);
break;
case GST_MESSAGE_DURATION_CHANGED:
// Duration in MSE is managed by MediaSource, SourceBuffer and AppendPipeline.
if (messageSourceIsPlaybin && !isMediaSource())
durationChanged();
break;
case GST_MESSAGE_REQUEST_STATE:
gst_message_parse_request_state(message, &requestedState);
gst_element_get_state(m_pipeline.get(), &currentState, nullptr, 250 * GST_NSECOND);
if (requestedState < currentState) {
GST_INFO("Element %s requested state change to %s", GST_MESSAGE_SRC_NAME(message),
gst_element_state_get_name(requestedState));
m_requestedState = requestedState;
if (!changePipelineState(requestedState))
loadingFailed(MediaPlayer::Empty);
}
break;
case GST_MESSAGE_CLOCK_LOST:
// This can only happen in PLAYING state and we should just
// get a new clock by moving back to PAUSED and then to
// PLAYING again.
// This can happen if the stream that ends in a sink that
// provides the current clock disappears, for example if
// the audio sink provides the clock and the audio stream
// is disabled. It also happens relatively often with
// HTTP adaptive streams when switching between different
// variants of a stream.
gst_element_set_state(m_pipeline.get(), GST_STATE_PAUSED);
gst_element_set_state(m_pipeline.get(), GST_STATE_PLAYING);
break;
case GST_MESSAGE_LATENCY:
// Recalculate the latency, we don't need any special handling
// here other than the GStreamer default.
// This can happen if the latency of live elements changes, or
// for one reason or another a new live element is added or
// removed from the pipeline.
gst_bin_recalculate_latency(GST_BIN(m_pipeline.get()));
break;
case GST_MESSAGE_ELEMENT:
if (gst_is_missing_plugin_message(message)) {
if (gst_install_plugins_supported()) {
RefPtr<MediaPlayerRequestInstallMissingPluginsCallback> missingPluginCallback = MediaPlayerRequestInstallMissingPluginsCallback::create([weakThis = createWeakPtr()](uint32_t result, MediaPlayerRequestInstallMissingPluginsCallback& missingPluginCallback) {
if (!weakThis) {
GST_INFO("got missing pluging installation callback in destroyed player with result %u", result);
return;
}
GST_DEBUG("got missing plugin installation callback with result %u", result);
RefPtr<MediaPlayerRequestInstallMissingPluginsCallback> protectedMissingPluginCallback = &missingPluginCallback;
weakThis->m_missingPluginCallbacks.removeFirst(protectedMissingPluginCallback);
if (result != GST_INSTALL_PLUGINS_SUCCESS)
return;
weakThis->changePipelineState(GST_STATE_READY);
weakThis->changePipelineState(GST_STATE_PAUSED);
});
m_missingPluginCallbacks.append(missingPluginCallback);
GUniquePtr<char> detail(gst_missing_plugin_message_get_installer_detail(message));
GUniquePtr<char> description(gst_missing_plugin_message_get_description(message));
m_player->client().requestInstallMissingPlugins(String::fromUTF8(detail.get()), String::fromUTF8(description.get()), *missingPluginCallback);
}
}
#if ENABLE(VIDEO_TRACK) && USE(GSTREAMER_MPEGTS)
else if (GstMpegtsSection* section = gst_message_parse_mpegts_section(message)) {
processMpegTsSection(section);
gst_mpegts_section_unref(section);
}
#endif
#if ENABLE(ENCRYPTED_MEDIA)
else if (gst_structure_has_name(structure, "drm-key-needed")) {
GST_DEBUG("drm-key-needed message from %s", GST_MESSAGE_SRC_NAME(message));
GRefPtr<GstEvent> event;
gst_structure_get(structure, "event", GST_TYPE_EVENT, &event.outPtr(), nullptr);
handleProtectionEvent(event.get());
} else if (gst_structure_has_name(structure, "decrypt-key-needed")) {
GST_DEBUG("decrypt-key-needed message from %s", GST_MESSAGE_SRC_NAME(message));
MediaPlayerPrivateGStreamerBase::dispatchCDMInstance();
}
#endif
else if (gst_structure_has_name(structure, "http-headers")) {
GstStructure* responseHeaders;
if (gst_structure_get(structure, "response-headers", GST_TYPE_STRUCTURE, &responseHeaders, nullptr)) {
if (!gst_structure_has_field(responseHeaders, httpHeaderNameString(HTTPHeaderName::ContentLength).utf8().data())) {
GST_INFO("Live stream detected. Disabling on-disk buffering");
m_isStreaming = true;
setDownloadBuffering();
}
gst_structure_free(responseHeaders);
}
} else
GST_DEBUG("Unhandled element message: %" GST_PTR_FORMAT, structure);
break;
#if ENABLE(VIDEO_TRACK)
case GST_MESSAGE_TOC:
processTableOfContents(message);
break;
#endif
case GST_MESSAGE_TAG: {
GstTagList* tags = nullptr;
GUniqueOutPtr<gchar> tag;
gst_message_parse_tag(message, &tags);
if (gst_tag_list_get_string(tags, GST_TAG_IMAGE_ORIENTATION, &tag.outPtr())) {
if (!g_strcmp0(tag.get(), "rotate-90"))
setVideoSourceOrientation(ImageOrientation(OriginRightTop));
else if (!g_strcmp0(tag.get(), "rotate-180"))
setVideoSourceOrientation(ImageOrientation(OriginBottomRight));
else if (!g_strcmp0(tag.get(), "rotate-270"))
setVideoSourceOrientation(ImageOrientation(OriginLeftBottom));
}
gst_tag_list_unref(tags);
break;
}
#if GST_CHECK_VERSION(1, 10, 0)
case GST_MESSAGE_STREAM_COLLECTION: {
GRefPtr<GstStreamCollection> collection;
gst_message_parse_stream_collection(message, &collection.outPtr());
if (collection) {
m_streamCollection.swap(collection);
m_notifier->notify(MainThreadNotification::StreamCollectionChanged, [this] {
this->updateTracks();
});
}
break;
}
case GST_MESSAGE_STREAMS_SELECTED: {
GRefPtr<GstStreamCollection> collection;
gst_message_parse_streams_selected(message, &collection.outPtr());
if (!collection)
break;
m_streamCollection.swap(collection);
m_currentAudioStreamId = "";
m_currentVideoStreamId = "";
m_currentTextStreamId = "";
unsigned length = gst_message_streams_selected_get_size(message);
for (unsigned i = 0; i < length; i++) {
GRefPtr<GstStream> stream = adoptGRef(gst_message_streams_selected_get_stream(message, i));
if (!stream)
continue;
GstStreamType type = gst_stream_get_stream_type(stream.get());
String streamId(gst_stream_get_stream_id(stream.get()));
GST_DEBUG("Selecting %s track with ID: %s", gst_stream_type_get_name(type), streamId.utf8().data());
// Playbin3 can send more than one selected stream of the same type
// but there's no priority or ordering system in place, so we assume
// the selected stream is the last one as reported by playbin3.
if (type & GST_STREAM_TYPE_AUDIO) {
m_currentAudioStreamId = streamId;
auto track = m_audioTracks.get(m_currentAudioStreamId);
ASSERT(track);
track->markAsActive();
} else if (type & GST_STREAM_TYPE_VIDEO) {
m_currentVideoStreamId = streamId;
auto track = m_videoTracks.get(m_currentVideoStreamId);
ASSERT(track);
track->markAsActive();
} else if (type & GST_STREAM_TYPE_TEXT)
m_currentTextStreamId = streamId;
else
GST_WARNING("Unknown stream type with stream-id %s", streamId.utf8().data());
}
break;
}
#endif
default:
GST_DEBUG("Unhandled GStreamer message type: %s", GST_MESSAGE_TYPE_NAME(message));
break;
}
}
void MediaPlayerPrivateGStreamer::processBufferingStats(GstMessage* message)
{
m_buffering = true;
gst_message_parse_buffering(message, &m_bufferingPercentage);
GST_DEBUG("[Buffering] Buffering: %d%%.", m_bufferingPercentage);
if (m_bufferingPercentage == 100)
updateStates();
}
#if ENABLE(VIDEO_TRACK) && USE(GSTREAMER_MPEGTS)
void MediaPlayerPrivateGStreamer::processMpegTsSection(GstMpegtsSection* section)
{
ASSERT(section);
if (section->section_type == GST_MPEGTS_SECTION_PMT) {
const GstMpegtsPMT* pmt = gst_mpegts_section_get_pmt(section);
m_metadataTracks.clear();
for (guint i = 0; i < pmt->streams->len; ++i) {
const GstMpegtsPMTStream* stream = static_cast<const GstMpegtsPMTStream*>(g_ptr_array_index(pmt->streams, i));
if (stream->stream_type == 0x05 || stream->stream_type >= 0x80) {
AtomicString pid = String::number(stream->pid);
RefPtr<InbandMetadataTextTrackPrivateGStreamer> track = InbandMetadataTextTrackPrivateGStreamer::create(
InbandTextTrackPrivate::Metadata, InbandTextTrackPrivate::Data, pid);
// 4.7.10.12.2 Sourcing in-band text tracks
// If the new text track's kind is metadata, then set the text track in-band metadata track dispatch
// type as follows, based on the type of the media resource:
// Let stream type be the value of the "stream_type" field describing the text track's type in the
// file's program map section, interpreted as an 8-bit unsigned integer. Let length be the value of
// the "ES_info_length" field for the track in the same part of the program map section, interpreted
// as an integer as defined by the MPEG-2 specification. Let descriptor bytes be the length bytes
// following the "ES_info_length" field. The text track in-band metadata track dispatch type must be
// set to the concatenation of the stream type byte and the zero or more descriptor bytes bytes,
// expressed in hexadecimal using uppercase ASCII hex digits.
String inbandMetadataTrackDispatchType;
appendUnsignedAsHexFixedSize(stream->stream_type, inbandMetadataTrackDispatchType, 2);
for (guint j = 0; j < stream->descriptors->len; ++j) {
const GstMpegtsDescriptor* descriptor = static_cast<const GstMpegtsDescriptor*>(g_ptr_array_index(stream->descriptors, j));
for (guint k = 0; k < descriptor->length; ++k)
appendByteAsHex(descriptor->data[k], inbandMetadataTrackDispatchType);
}
track->setInBandMetadataTrackDispatchType(inbandMetadataTrackDispatchType);
m_metadataTracks.add(pid, track);
m_player->addTextTrack(*track);
}
}
} else {
AtomicString pid = String::number(section->pid);
RefPtr<InbandMetadataTextTrackPrivateGStreamer> track = m_metadataTracks.get(pid);
if (!track)
return;
GRefPtr<GBytes> data = gst_mpegts_section_get_data(section);
gsize size;
const void* bytes = g_bytes_get_data(data.get(), &size);
track->addDataCue(currentMediaTime(), currentMediaTime(), bytes, size);
}
}
#endif
#if ENABLE(VIDEO_TRACK)
void MediaPlayerPrivateGStreamer::processTableOfContents(GstMessage* message)
{
if (m_chaptersTrack)
m_player->removeTextTrack(*m_chaptersTrack);
m_chaptersTrack = InbandMetadataTextTrackPrivateGStreamer::create(InbandTextTrackPrivate::Chapters, InbandTextTrackPrivate::Generic);
m_player->addTextTrack(*m_chaptersTrack);
GRefPtr<GstToc> toc;
gboolean updated;
gst_message_parse_toc(message, &toc.outPtr(), &updated);
ASSERT(toc);
for (GList* i = gst_toc_get_entries(toc.get()); i; i = i->next)
processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data));
}
void MediaPlayerPrivateGStreamer::processTableOfContentsEntry(GstTocEntry* entry)
{
ASSERT(entry);
auto cue = GenericCueData::create();
gint64 start = -1, stop = -1;
gst_toc_entry_get_start_stop_times(entry, &start, &stop);
if (start != -1)
cue->setStartTime(MediaTime(start, GST_SECOND));
if (stop != -1)
cue->setEndTime(MediaTime(stop, GST_SECOND));
GstTagList* tags = gst_toc_entry_get_tags(entry);
if (tags) {
gchar* title = nullptr;
gst_tag_list_get_string(tags, GST_TAG_TITLE, &title);
if (title) {
cue->setContent(title);
g_free(title);
}
}
m_chaptersTrack->addGenericCue(cue);
for (GList* i = gst_toc_entry_get_sub_entries(entry); i; i = i->next)
processTableOfContentsEntry(static_cast<GstTocEntry*>(i->data));
}
void MediaPlayerPrivateGStreamer::purgeInvalidAudioTracks(Vector<String> validTrackIds)
{
m_audioTracks.removeIf([validTrackIds](auto& keyAndValue) {
return !validTrackIds.contains(keyAndValue.key);
});
}
void MediaPlayerPrivateGStreamer::purgeInvalidVideoTracks(Vector<String> validTrackIds)
{
m_videoTracks.removeIf([validTrackIds](auto& keyAndValue) {
return !validTrackIds.contains(keyAndValue.key);
});
}
void MediaPlayerPrivateGStreamer::purgeInvalidTextTracks(Vector<String> validTrackIds)
{
m_textTracks.removeIf([validTrackIds](auto& keyAndValue) {
return !validTrackIds.contains(keyAndValue.key);
});
}
#endif
static int findHLSQueue(const GValue* item)
{
GstElement* element = GST_ELEMENT(g_value_get_object(item));
if (g_str_has_prefix(GST_ELEMENT_NAME(element), "queue")) {
GstElement* parent = GST_ELEMENT(GST_ELEMENT_PARENT(element));
if (!GST_IS_OBJECT(parent))
return 1;
if (g_str_has_prefix(GST_ELEMENT_NAME(GST_ELEMENT_PARENT(parent)), "hlsdemux"))
return 0;
}
return 1;
}
static bool isHLSProgressing(GstElement* playbin, GstQuery* query)
{
GValue item = { };
GstIterator* binIterator = gst_bin_iterate_recurse(GST_BIN(playbin));
bool foundHLSQueue = gst_iterator_find_custom(binIterator, reinterpret_cast<GCompareFunc>(findHLSQueue), &item, nullptr);
gst_iterator_free(binIterator);
if (!foundHLSQueue)
return false;
GstElement* queueElement = GST_ELEMENT(g_value_get_object(&item));
bool queryResult = gst_element_query(queueElement, query);
g_value_unset(&item);
return queryResult;
}
void MediaPlayerPrivateGStreamer::fillTimerFired()
{
GstQuery* query = gst_query_new_buffering(GST_FORMAT_PERCENT);
if (G_UNLIKELY(!gst_element_query(m_pipeline.get(), query))) {
// This query always fails for live pipelines. In the case of HLS, try and find
// the queue inside the HLS element to get a proxy measure of progress. Note
// that the percentage value is rather meaningless as used below.
// This is a hack, see https://bugs.webkit.org/show_bug.cgi?id=141469.
if (!isHLSProgressing(m_pipeline.get(), query)) {
gst_query_unref(query);
return;
}
}
gint64 start, stop;
gdouble fillStatus = 100.0;
gst_query_parse_buffering_range(query, nullptr, &start, &stop, nullptr);
gst_query_unref(query);
if (stop != -1)
fillStatus = 100.0 * stop / GST_FORMAT_PERCENT_MAX;
GST_DEBUG("[Buffering] Download buffer filled up to %f%%", fillStatus);
MediaTime mediaDuration = durationMediaTime();
// Update maxTimeLoaded only if the media duration is
// available. Otherwise we can't compute it.
if (mediaDuration) {
if (fillStatus == 100.0)
m_maxTimeLoaded = mediaDuration;
else
m_maxTimeLoaded = MediaTime(fillStatus * static_cast<double>(toGstUnsigned64Time(mediaDuration)) / 100, GST_SECOND);
GST_DEBUG("[Buffering] Updated maxTimeLoaded: %s", toString(m_maxTimeLoaded).utf8().data());
}
m_downloadFinished = fillStatus == 100.0;
if (!m_downloadFinished) {
updateStates();
return;
}
// Media is now fully loaded. It will play even if network
// connection is cut. Buffering is done, remove the fill source
// from the main loop.
m_fillTimer.stop();
updateStates();
}
MediaTime MediaPlayerPrivateGStreamer::maxMediaTimeSeekable() const
{
if (m_errorOccured)
return MediaTime::zeroTime();
MediaTime duration = durationMediaTime();
GST_DEBUG("maxMediaTimeSeekable, duration: %s", toString(duration).utf8().data());
// infinite duration means live stream
if (duration.isPositiveInfinite())
return MediaTime::zeroTime();
return duration;
}
MediaTime MediaPlayerPrivateGStreamer::maxTimeLoaded() const
{
if (m_errorOccured)
return MediaTime::zeroTime();
MediaTime loaded = m_maxTimeLoaded;
if (m_isEndReached)
loaded = durationMediaTime();
GST_LOG("maxTimeLoaded: %s", toString(loaded).utf8().data());
return loaded;
}
bool MediaPlayerPrivateGStreamer::didLoadingProgress() const
{
if (m_errorOccured || m_loadingStalled)
return false;
if (isLiveStream())
return true;
if (UNLIKELY(!m_pipeline || !durationMediaTime() || (!isMediaSource() && !totalBytes())))
return false;
MediaTime currentMaxTimeLoaded = maxTimeLoaded();
bool didLoadingProgress = currentMaxTimeLoaded != m_maxTimeLoadedAtLastDidLoadingProgress;
m_maxTimeLoadedAtLastDidLoadingProgress = currentMaxTimeLoaded;
GST_LOG("didLoadingProgress: %s", toString(didLoadingProgress).utf8().data());
return didLoadingProgress;
}
unsigned long long MediaPlayerPrivateGStreamer::totalBytes() const
{
if (m_errorOccured)
return 0;
if (m_totalBytes)
return m_totalBytes;
if (!m_source)
return 0;
if (isLiveStream())
return 0;
GstFormat fmt = GST_FORMAT_BYTES;
gint64 length = 0;
if (gst_element_query_duration(m_source.get(), fmt, &length)) {
GST_INFO("totalBytes %" G_GINT64_FORMAT, length);
m_totalBytes = static_cast<unsigned long long>(length);
m_isStreaming = !length;
return m_totalBytes;
}
// Fall back to querying the source pads manually.
// See also https://bugzilla.gnome.org/show_bug.cgi?id=638749
GstIterator* iter = gst_element_iterate_src_pads(m_source.get());
bool done = false;
while (!done) {
GValue item = G_VALUE_INIT;
switch (gst_iterator_next(iter, &item)) {
case GST_ITERATOR_OK: {
GstPad* pad = static_cast<GstPad*>(g_value_get_object(&item));
gint64 padLength = 0;
if (gst_pad_query_duration(pad, fmt, &padLength) && padLength > length)
length = padLength;
break;
}
case GST_ITERATOR_RESYNC:
gst_iterator_resync(iter);
break;
case GST_ITERATOR_ERROR:
FALLTHROUGH;
case GST_ITERATOR_DONE:
done = true;
break;
}
g_value_unset(&item);
}
gst_iterator_free(iter);
GST_INFO("totalBytes %" G_GINT64_FORMAT, length);
m_totalBytes = static_cast<unsigned long long>(length);
m_isStreaming = !length;
return m_totalBytes;
}
void MediaPlayerPrivateGStreamer::sourceSetupCallback(MediaPlayerPrivateGStreamer* player, GstElement* sourceElement)
{
player->sourceSetup(sourceElement);
}
void MediaPlayerPrivateGStreamer::uriDecodeBinElementAddedCallback(GstBin* bin, GstElement* element, MediaPlayerPrivateGStreamer* player)
{
if (g_strcmp0(G_OBJECT_CLASS_NAME(G_OBJECT_GET_CLASS(G_OBJECT(element))), "GstDownloadBuffer"))
return;
player->m_downloadBuffer = element;
g_signal_handlers_disconnect_by_func(bin, reinterpret_cast<gpointer>(uriDecodeBinElementAddedCallback), player);
g_signal_connect_swapped(element, "notify::temp-location", G_CALLBACK(downloadBufferFileCreatedCallback), player);
GUniqueOutPtr<char> oldDownloadTemplate;
g_object_get(element, "temp-template", &oldDownloadTemplate.outPtr(), nullptr);
GUniquePtr<char> newDownloadTemplate(g_build_filename(G_DIR_SEPARATOR_S, "var", "tmp", "WebKit-Media-XXXXXX", nullptr));
g_object_set(element, "temp-template", newDownloadTemplate.get(), nullptr);
GST_DEBUG("Reconfigured file download template from '%s' to '%s'", oldDownloadTemplate.get(), newDownloadTemplate.get());
player->purgeOldDownloadFiles(oldDownloadTemplate.get());
}
void MediaPlayerPrivateGStreamer::downloadBufferFileCreatedCallback(MediaPlayerPrivateGStreamer* player)
{
ASSERT(player->m_downloadBuffer);
g_signal_handlers_disconnect_by_func(player->m_downloadBuffer.get(), reinterpret_cast<gpointer>(downloadBufferFileCreatedCallback), player);
GUniqueOutPtr<char> downloadFile;
g_object_get(player->m_downloadBuffer.get(), "temp-location", &downloadFile.outPtr(), nullptr);
player->m_downloadBuffer = nullptr;
if (UNLIKELY(!FileSystem::deleteFile(downloadFile.get()))) {
GST_WARNING("Couldn't unlink media temporary file %s after creation", downloadFile.get());
return;
}
GST_DEBUG("Unlinked media temporary file %s after creation", downloadFile.get());
}
void MediaPlayerPrivateGStreamer::purgeOldDownloadFiles(const char* downloadFileTemplate)
{
if (!downloadFileTemplate)
return;
GUniquePtr<char> templatePath(g_path_get_dirname(downloadFileTemplate));
GUniquePtr<char> templateFile(g_path_get_basename(downloadFileTemplate));
String templatePattern = String(templateFile.get()).replace("X", "?");
for (auto& filePath : FileSystem::listDirectory(templatePath.get(), templatePattern)) {
if (UNLIKELY(!FileSystem::deleteFile(filePath))) {
GST_WARNING("Couldn't unlink legacy media temporary file: %s", filePath.utf8().data());
continue;
}
GST_TRACE("Unlinked legacy media temporary file: %s", filePath.utf8().data());
}
}
void MediaPlayerPrivateGStreamer::sourceSetup(GstElement* sourceElement)
{
GST_DEBUG("Source element set-up for %s", GST_ELEMENT_NAME(sourceElement));
if (WEBKIT_IS_WEB_SRC(m_source.get()) && GST_OBJECT_PARENT(m_source.get()))
g_signal_handlers_disconnect_by_func(GST_ELEMENT_PARENT(m_source.get()), reinterpret_cast<gpointer>(uriDecodeBinElementAddedCallback), this);
m_source = sourceElement;
if (WEBKIT_IS_WEB_SRC(m_source.get())) {
webKitWebSrcSetMediaPlayer(WEBKIT_WEB_SRC(m_source.get()), m_player);
g_signal_connect(GST_ELEMENT_PARENT(m_source.get()), "element-added", G_CALLBACK(uriDecodeBinElementAddedCallback), this);
}
}
bool MediaPlayerPrivateGStreamer::hasSingleSecurityOrigin() const
{
if (!m_source)
return false;
if (!WEBKIT_IS_WEB_SRC(m_source.get()))
return true;
GUniqueOutPtr<char> originalURI, resolvedURI;
g_object_get(m_source.get(), "location", &originalURI.outPtr(), "resolved-location", &resolvedURI.outPtr(), nullptr);
if (!originalURI || !resolvedURI)
return false;
if (!g_strcmp0(originalURI.get(), resolvedURI.get()))
return true;
Ref<SecurityOrigin> resolvedOrigin(SecurityOrigin::createFromString(String::fromUTF8(resolvedURI.get())));
Ref<SecurityOrigin> requestedOrigin(SecurityOrigin::createFromString(String::fromUTF8(originalURI.get())));
return resolvedOrigin->isSameSchemeHostPort(requestedOrigin.get());
}
void MediaPlayerPrivateGStreamer::cancelLoad()
{
if (m_networkState < MediaPlayer::Loading || m_networkState == MediaPlayer::Loaded)
return;
if (m_pipeline)
changePipelineState(GST_STATE_READY);
}
void MediaPlayerPrivateGStreamer::asyncStateChangeDone()
{
if (!m_pipeline || m_errorOccured)
return;
if (m_seeking) {
if (m_seekIsPending)
updateStates();
else {
GST_DEBUG("[Seek] seeked to %s", toString(m_seekTime).utf8().data());
m_seeking = false;
if (m_timeOfOverlappingSeek != m_seekTime && m_timeOfOverlappingSeek.isValid()) {
seek(m_timeOfOverlappingSeek);
m_timeOfOverlappingSeek = MediaTime::invalidTime();
return;
}
m_timeOfOverlappingSeek = MediaTime::invalidTime();
// The pipeline can still have a pending state. In this case a position query will fail.
// Right now we can use m_seekTime as a fallback.
m_canFallBackToLastFinishedSeekPosition = true;
timeChanged();
}
} else
updateStates();
}
void MediaPlayerPrivateGStreamer::updateStates()
{
if (!m_pipeline)
return;
if (m_errorOccured)
return;
MediaPlayer::NetworkState oldNetworkState = m_networkState;
MediaPlayer::ReadyState oldReadyState = m_readyState;
GstState pending;
GstState state;
bool stateReallyChanged = false;
GstStateChangeReturn getStateResult = gst_element_get_state(m_pipeline.get(), &state, &pending, 250 * GST_NSECOND);
if (state != m_currentState) {
m_oldState = m_currentState;
m_currentState = state;
stateReallyChanged = true;
}
bool shouldUpdatePlaybackState = false;
switch (getStateResult) {
case GST_STATE_CHANGE_SUCCESS: {
GST_DEBUG("State: %s, pending: %s", gst_element_state_get_name(m_currentState), gst_element_state_get_name(pending));
// Do nothing if on EOS and state changed to READY to avoid recreating the player
// on HTMLMediaElement and properly generate the video 'ended' event.
if (m_isEndReached && m_currentState == GST_STATE_READY)
break;
m_resetPipeline = m_currentState <= GST_STATE_READY;
bool didBuffering = m_buffering;
// Update ready and network states.
switch (m_currentState) {
case GST_STATE_NULL:
m_readyState = MediaPlayer::HaveNothing;
m_networkState = MediaPlayer::Empty;
break;
case GST_STATE_READY:
m_readyState = MediaPlayer::HaveMetadata;
m_networkState = MediaPlayer::Empty;
break;
case GST_STATE_PAUSED:
case GST_STATE_PLAYING:
if (m_buffering) {
if (m_bufferingPercentage == 100) {
GST_DEBUG("[Buffering] Complete.");
m_buffering = false;
m_readyState = MediaPlayer::HaveEnoughData;
m_networkState = m_downloadFinished ? MediaPlayer::Idle : MediaPlayer::Loading;
} else {
m_readyState = MediaPlayer::HaveCurrentData;
m_networkState = MediaPlayer::Loading;
}
} else if (m_downloadFinished) {
m_readyState = MediaPlayer::HaveEnoughData;
m_networkState = MediaPlayer::Loaded;
} else {
m_readyState = MediaPlayer::HaveFutureData;
m_networkState = MediaPlayer::Loading;
}
break;
default:
ASSERT_NOT_REACHED();
break;
}
// Sync states where needed.
if (m_currentState == GST_STATE_PAUSED) {
if (!m_volumeAndMuteInitialized) {
notifyPlayerOfVolumeChange();
notifyPlayerOfMute();
m_volumeAndMuteInitialized = true;
}
if (didBuffering && !m_buffering && !m_paused && m_playbackRate) {
GST_DEBUG("[Buffering] Restarting playback.");
changePipelineState(GST_STATE_PLAYING);
}
} else if (m_currentState == GST_STATE_PLAYING) {
m_paused = false;
if ((m_buffering && !isLiveStream()) || !m_playbackRate) {
GST_DEBUG("[Buffering] Pausing stream for buffering.");
changePipelineState(GST_STATE_PAUSED);
}
} else
m_paused = true;
GST_DEBUG("Old state: %s, new state: %s (requested: %s)", gst_element_state_get_name(m_oldState), gst_element_state_get_name(m_currentState), gst_element_state_get_name(m_requestedState));
if (m_requestedState == GST_STATE_PAUSED && m_currentState == GST_STATE_PAUSED) {
shouldUpdatePlaybackState = true;
GST_INFO("Requested state change to %s was completed", gst_element_state_get_name(m_currentState));
}
// Emit play state change notification only when going to PLAYING so that
// the media element gets a chance to enable its page sleep disabler.
// Emitting this notification in more cases triggers unwanted code paths
// and test timeouts.
if (stateReallyChanged && (m_oldState != m_currentState) && (m_oldState == GST_STATE_PAUSED && m_currentState == GST_STATE_PLAYING)) {
GST_INFO("Playback state changed from %s to %s. Notifying the media player client", gst_element_state_get_name(m_oldState), gst_element_state_get_name(m_currentState));
shouldUpdatePlaybackState = true;
}
break;
}
case GST_STATE_CHANGE_ASYNC:
GST_DEBUG("Async: State: %s, pending: %s", gst_element_state_get_name(m_currentState), gst_element_state_get_name(pending));
// Change in progress.
break;
case GST_STATE_CHANGE_FAILURE:
GST_DEBUG("Failure: State: %s, pending: %s", gst_element_state_get_name(m_currentState), gst_element_state_get_name(pending));
// Change failed
return;
case GST_STATE_CHANGE_NO_PREROLL:
GST_DEBUG("No preroll: State: %s, pending: %s", gst_element_state_get_name(m_currentState), gst_element_state_get_name(pending));
// Live pipelines go in PAUSED without prerolling.
m_isStreaming = true;
setDownloadBuffering();
if (m_currentState == GST_STATE_READY)
m_readyState = MediaPlayer::HaveNothing;
else if (m_currentState == GST_STATE_PAUSED) {
m_readyState = MediaPlayer::HaveEnoughData;
m_paused = true;
} else if (m_currentState == GST_STATE_PLAYING)
m_paused = false;
if (!m_paused && m_playbackRate)
changePipelineState(GST_STATE_PLAYING);
m_networkState = MediaPlayer::Loading;
break;
default:
GST_DEBUG("Else : %d", getStateResult);
break;
}
m_requestedState = GST_STATE_VOID_PENDING;
if (shouldUpdatePlaybackState)
m_player->playbackStateChanged();
if (m_networkState != oldNetworkState) {
GST_DEBUG("Network State Changed from %s to %s", convertEnumerationToString(oldNetworkState).utf8().data(), convertEnumerationToString(m_networkState).utf8().data());
m_player->networkStateChanged();
}
if (m_readyState != oldReadyState) {
GST_DEBUG("Ready State Changed from %s to %s", convertEnumerationToString(oldReadyState).utf8().data(), convertEnumerationToString(m_readyState).utf8().data());
m_player->readyStateChanged();
}
if (getStateResult == GST_STATE_CHANGE_SUCCESS && m_currentState >= GST_STATE_PAUSED) {
updatePlaybackRate();
if (m_seekIsPending) {
GST_DEBUG("[Seek] committing pending seek to %s", toString(m_seekTime).utf8().data());
m_seekIsPending = false;
m_seeking = doSeek(m_seekTime, m_player->rate(), static_cast<GstSeekFlags>(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE));
if (!m_seeking)
GST_DEBUG("[Seek] seeking to %s failed", toString(m_seekTime).utf8().data());
}
}
}
void MediaPlayerPrivateGStreamer::mediaLocationChanged(GstMessage* message)
{
if (m_mediaLocations)
gst_structure_free(m_mediaLocations);
const GstStructure* structure = gst_message_get_structure(message);
if (structure) {
// This structure can contain:
// - both a new-location string and embedded locations structure
// - or only a new-location string.
m_mediaLocations = gst_structure_copy(structure);
const GValue* locations = gst_structure_get_value(m_mediaLocations, "locations");
if (locations)
m_mediaLocationCurrentIndex = static_cast<int>(gst_value_list_get_size(locations)) -1;
loadNextLocation();
}
}
bool MediaPlayerPrivateGStreamer::loadNextLocation()
{
if (!m_mediaLocations)
return false;
const GValue* locations = gst_structure_get_value(m_mediaLocations, "locations");
const gchar* newLocation = nullptr;
if (!locations) {
// Fallback on new-location string.
newLocation = gst_structure_get_string(m_mediaLocations, "new-location");
if (!newLocation)
return false;
}
if (!newLocation) {
if (m_mediaLocationCurrentIndex < 0) {
m_mediaLocations = nullptr;
return false;
}
const GValue* location = gst_value_list_get_value(locations,
m_mediaLocationCurrentIndex);
const GstStructure* structure = gst_value_get_structure(location);
if (!structure) {
m_mediaLocationCurrentIndex--;
return false;
}
newLocation = gst_structure_get_string(structure, "new-location");
}
if (newLocation) {
// Found a candidate. new-location is not always an absolute url
// though. We need to take the base of the current url and
// append the value of new-location to it.
URL baseUrl = gst_uri_is_valid(newLocation) ? URL() : m_url;
URL newUrl = URL(baseUrl, newLocation);
convertToInternalProtocol(newUrl);
RefPtr<SecurityOrigin> securityOrigin = SecurityOrigin::create(m_url);
if (securityOrigin->canRequest(newUrl)) {
GST_INFO("New media url: %s", newUrl.string().utf8().data());
// Reset player states.
m_networkState = MediaPlayer::Loading;
m_player->networkStateChanged();
m_readyState = MediaPlayer::HaveNothing;
m_player->readyStateChanged();
// Reset pipeline state.
m_resetPipeline = true;
changePipelineState(GST_STATE_READY);
GstState state;
gst_element_get_state(m_pipeline.get(), &state, nullptr, 0);
if (state <= GST_STATE_READY) {
// Set the new uri and start playing.
setPlaybinURL(newUrl);
changePipelineState(GST_STATE_PLAYING);
return true;
}
} else
GST_INFO("Not allowed to load new media location: %s", newUrl.string().utf8().data());
}
m_mediaLocationCurrentIndex--;
return false;
}
void MediaPlayerPrivateGStreamer::loadStateChanged()
{
updateStates();
}
void MediaPlayerPrivateGStreamer::timeChanged()
{
updateStates();
m_player->timeChanged();
}
void MediaPlayerPrivateGStreamer::didEnd()
{
GST_INFO("Playback ended");
// Synchronize position and duration values to not confuse the
// HTMLMediaElement. In some cases like reverse playback the
// position is not always reported as 0 for instance.
MediaTime now = currentMediaTime();
if (now > MediaTime { } && now <= durationMediaTime())
m_player->durationChanged();
m_isEndReached = true;
timeChanged();
if (!m_player->client().mediaPlayerIsLooping()) {
m_paused = true;
m_durationAtEOS = durationMediaTime();
changePipelineState(GST_STATE_READY);
m_downloadFinished = false;
}
}
void MediaPlayerPrivateGStreamer::durationChanged()
{
MediaTime previousDuration = durationMediaTime();
// FIXME: Check if this method is still useful, because it's not doing its work at all
// since bug #159458 removed a cacheDuration() call here.
// Avoid emiting durationchanged in the case where the previous
// duration was 0 because that case is already handled by the
// HTMLMediaElement.
if (previousDuration && durationMediaTime() != previousDuration)
m_player->durationChanged();
}
void MediaPlayerPrivateGStreamer::loadingFailed(MediaPlayer::NetworkState error)
{
GST_WARNING("Loading failed, error: %d", error);
m_errorOccured = true;
if (m_networkState != error) {
m_networkState = error;
m_player->networkStateChanged();
}
if (m_readyState != MediaPlayer::HaveNothing) {
m_readyState = MediaPlayer::HaveNothing;
m_player->readyStateChanged();
}
// Loading failed, remove ready timer.
m_readyTimerHandler.stop();
}
static HashSet<String, ASCIICaseInsensitiveHash>& mimeTypeSet()
{
static NeverDestroyed<HashSet<String, ASCIICaseInsensitiveHash>> mimeTypes = []()
{
MediaPlayerPrivateGStreamerBase::initializeGStreamerAndRegisterWebKitElements();
HashSet<String, ASCIICaseInsensitiveHash> set;
GList* audioDecoderFactories = gst_element_factory_list_get_elements(GST_ELEMENT_FACTORY_TYPE_DECODER | GST_ELEMENT_FACTORY_TYPE_MEDIA_AUDIO, GST_RANK_MARGINAL);
GList* videoDecoderFactories = gst_element_factory_list_get_elements(GST_ELEMENT_FACTORY_TYPE_DECODER | GST_ELEMENT_FACTORY_TYPE_MEDIA_VIDEO, GST_RANK_MARGINAL);
GList* demuxerFactories = gst_element_factory_list_get_elements(GST_ELEMENT_FACTORY_TYPE_DEMUXER, GST_RANK_MARGINAL);
enum ElementType {
AudioDecoder = 0,
VideoDecoder,
Demuxer
};
struct GstCapsWebKitMapping {
ElementType elementType;
const char* capsString;
Vector<AtomicString> webkitMimeTypes;
};
Vector<GstCapsWebKitMapping> mapping = {
{AudioDecoder, "audio/midi", {"audio/midi", "audio/riff-midi"}},
{AudioDecoder, "audio/x-sbc", { }},
{AudioDecoder, "audio/x-sid", { }},
{AudioDecoder, "audio/x-flac", {"audio/x-flac", "audio/flac"}},
{AudioDecoder, "audio/x-wav", {"audio/x-wav", "audio/wav", "audio/vnd.wave"}},
{AudioDecoder, "audio/x-wavpack", {"audio/x-wavpack"}},
{AudioDecoder, "audio/x-speex", {"audio/speex", "audio/x-speex"}},
{AudioDecoder, "audio/x-ac3", { }},
{AudioDecoder, "audio/x-eac3", {"audio/x-ac3"}},
{AudioDecoder, "audio/x-dts", { }},
{VideoDecoder, "video/x-h264, profile=(string)high", {"video/mp4", "video/x-m4v"}},
{VideoDecoder, "video/x-msvideocodec", {"video/x-msvideo"}},
{VideoDecoder, "video/x-h263", { }},
{VideoDecoder, "video/mpegts", { }},
{VideoDecoder, "video/mpeg, mpegversion=(int){1,2}, systemstream=(boolean)false", {"video/mpeg"}},
{VideoDecoder, "video/x-dirac", { }},
{VideoDecoder, "video/x-flash-video", {"video/flv", "video/x-flv"}},
{Demuxer, "video/quicktime", { }},
{Demuxer, "video/quicktime, variant=(string)3gpp", {"video/3gpp"}},
{Demuxer, "application/x-3gp", { }},
{Demuxer, "video/x-ms-asf", { }},
{Demuxer, "audio/x-aiff", { }},
{Demuxer, "application/x-pn-realaudio", { }},
{Demuxer, "application/vnd.rn-realmedia", { }},
{Demuxer, "audio/x-wav", {"audio/x-wav", "audio/wav", "audio/vnd.wave"}},
{Demuxer, "application/x-hls", {"application/vnd.apple.mpegurl", "application/x-mpegurl"}}
};
for (auto& current : mapping) {
GList* factories = demuxerFactories;
if (current.elementType == AudioDecoder)
factories = audioDecoderFactories;
else if (current.elementType == VideoDecoder)
factories = videoDecoderFactories;
if (gstRegistryHasElementForMediaType(factories, current.capsString)) {
if (!current.webkitMimeTypes.isEmpty()) {
for (const auto& mimeType : current.webkitMimeTypes)
set.add(mimeType);
} else
set.add(AtomicString(current.capsString));
}
}
bool opusSupported = false;
if (gstRegistryHasElementForMediaType(audioDecoderFactories, "audio/x-opus")) {
opusSupported = true;
set.add(AtomicString("audio/opus"));
}
bool vorbisSupported = false;
if (gstRegistryHasElementForMediaType(demuxerFactories, "application/ogg")) {
set.add(AtomicString("application/ogg"));
vorbisSupported = gstRegistryHasElementForMediaType(audioDecoderFactories, "audio/x-vorbis");
if (vorbisSupported) {
set.add(AtomicString("audio/ogg"));
set.add(AtomicString("audio/x-vorbis+ogg"));
}
if (gstRegistryHasElementForMediaType(videoDecoderFactories, "video/x-theora"))
set.add(AtomicString("video/ogg"));
}
bool audioMpegSupported = false;
if (gstRegistryHasElementForMediaType(audioDecoderFactories, "audio/mpeg, mpegversion=(int)1, layer=(int)[1, 3]")) {
audioMpegSupported = true;
set.add(AtomicString("audio/mp1"));
set.add(AtomicString("audio/mp3"));
set.add(AtomicString("audio/x-mp3"));
}
if (gstRegistryHasElementForMediaType(audioDecoderFactories, "audio/mpeg, mpegversion=(int){2, 4}")) {
audioMpegSupported = true;
set.add(AtomicString("audio/aac"));
set.add(AtomicString("audio/mp2"));
set.add(AtomicString("audio/mp4"));
set.add(AtomicString("audio/x-m4a"));
}
if (audioMpegSupported) {
set.add(AtomicString("audio/mpeg"));
set.add(AtomicString("audio/x-mpeg"));
}
if (gstRegistryHasElementForMediaType(demuxerFactories, "video/x-matroska")) {
set.add(AtomicString("video/x-matroska"));
if (gstRegistryHasElementForMediaType(videoDecoderFactories, "video/x-vp8")
|| gstRegistryHasElementForMediaType(videoDecoderFactories, "video/x-vp9")
|| gstRegistryHasElementForMediaType(videoDecoderFactories, "video/x-vp10"))
set.add(AtomicString("video/webm"));
if (vorbisSupported || opusSupported)
set.add(AtomicString("audio/webm"));
}
gst_plugin_feature_list_free(audioDecoderFactories);
gst_plugin_feature_list_free(videoDecoderFactories);
gst_plugin_feature_list_free(demuxerFactories);
return set;
}();
return mimeTypes;
}
void MediaPlayerPrivateGStreamer::getSupportedTypes(HashSet<String, ASCIICaseInsensitiveHash>& types)
{
types = mimeTypeSet();
}
MediaPlayer::SupportsType MediaPlayerPrivateGStreamer::supportsType(const MediaEngineSupportParameters& parameters)
{
MediaPlayer::SupportsType result = MediaPlayer::IsNotSupported;
#if ENABLE(MEDIA_SOURCE)
// MediaPlayerPrivateGStreamerMSE is in charge of mediasource playback, not us.
if (parameters.isMediaSource)
return result;
#endif
if (parameters.isMediaStream)
return result;
if (parameters.type.isEmpty())
return result;
// spec says we should not return "probably" if the codecs string is empty
if (mimeTypeSet().contains(parameters.type.containerType()))
result = parameters.type.codecs().isEmpty() ? MediaPlayer::MayBeSupported : MediaPlayer::IsSupported;
return extendedSupportsType(parameters, result);
}
void MediaPlayerPrivateGStreamer::setDownloadBuffering()
{
if (!m_pipeline)
return;
unsigned flags;
g_object_get(m_pipeline.get(), "flags", &flags, nullptr);
unsigned flagDownload = getGstPlayFlag("download");
// We don't want to stop downloading if we already started it.
if (flags & flagDownload && m_readyState > MediaPlayer::HaveNothing && !m_resetPipeline) {
GST_DEBUG("Download already started, not starting again");
return;
}
bool shouldDownload = !isLiveStream() && m_preload == MediaPlayer::Auto;
if (shouldDownload) {
GST_INFO("Enabling on-disk buffering");
g_object_set(m_pipeline.get(), "flags", flags | flagDownload, nullptr);
m_fillTimer.startRepeating(200_ms);
} else {
GST_INFO("Disabling on-disk buffering");
g_object_set(m_pipeline.get(), "flags", flags & ~flagDownload, nullptr);
m_fillTimer.stop();
}
}
void MediaPlayerPrivateGStreamer::setPreload(MediaPlayer::Preload preload)
{
GST_DEBUG("Setting preload to %s", convertEnumerationToString(preload).utf8().data());
if (preload == MediaPlayer::Auto && isLiveStream())
return;
m_preload = preload;
setDownloadBuffering();
if (m_delayingLoad && m_preload != MediaPlayer::None) {
m_delayingLoad = false;
commitLoad();
}
}
GstElement* MediaPlayerPrivateGStreamer::createAudioSink()
{
m_autoAudioSink = gst_element_factory_make("autoaudiosink", nullptr);
if (!m_autoAudioSink) {
GST_WARNING("GStreamer's autoaudiosink not found. Please check your gst-plugins-good installation");
return nullptr;
}
g_signal_connect_swapped(m_autoAudioSink.get(), "child-added", G_CALLBACK(setAudioStreamPropertiesCallback), this);
GstElement* audioSinkBin;
if (webkitGstCheckVersion(1, 4, 2)) {
#if ENABLE(WEB_AUDIO)
audioSinkBin = gst_bin_new("audio-sink");
ensureAudioSourceProvider();
m_audioSourceProvider->configureAudioBin(audioSinkBin, nullptr);
return audioSinkBin;
#else
return m_autoAudioSink.get();
#endif
}
// Construct audio sink only if pitch preserving is enabled.
// If GStreamer 1.4.2 is used the audio-filter playbin property is used instead.
if (m_preservesPitch) {
GstElement* scale = gst_element_factory_make("scaletempo", nullptr);
if (!scale) {
GST_WARNING("Failed to create scaletempo");
return m_autoAudioSink.get();
}
audioSinkBin = gst_bin_new("audio-sink");
gst_bin_add(GST_BIN(audioSinkBin), scale);
GRefPtr<GstPad> pad = adoptGRef(gst_element_get_static_pad(scale, "sink"));
gst_element_add_pad(audioSinkBin, gst_ghost_pad_new("sink", pad.get()));
#if ENABLE(WEB_AUDIO)
ensureAudioSourceProvider();
m_audioSourceProvider->configureAudioBin(audioSinkBin, scale);
#else
GstElement* convert = gst_element_factory_make("audioconvert", nullptr);
GstElement* resample = gst_element_factory_make("audioresample", nullptr);
gst_bin_add_many(GST_BIN(audioSinkBin), convert, resample, m_autoAudioSink.get(), nullptr);
if (!gst_element_link_many(scale, convert, resample, m_autoAudioSink.get(), nullptr)) {
GST_WARNING("Failed to link audio sink elements");
gst_object_unref(audioSinkBin);
return m_autoAudioSink.get();
}
#endif
return audioSinkBin;
}
#if ENABLE(WEB_AUDIO)
audioSinkBin = gst_bin_new("audio-sink");
ensureAudioSourceProvider();
m_audioSourceProvider->configureAudioBin(audioSinkBin, nullptr);
return audioSinkBin;
#endif
ASSERT_NOT_REACHED();
return nullptr;
}
GstElement* MediaPlayerPrivateGStreamer::audioSink() const
{
GstElement* sink;
g_object_get(m_pipeline.get(), "audio-sink", &sink, nullptr);
return sink;
}
#if ENABLE(WEB_AUDIO)
void MediaPlayerPrivateGStreamer::ensureAudioSourceProvider()
{
if (!m_audioSourceProvider)
m_audioSourceProvider = std::make_unique<AudioSourceProviderGStreamer>();
}
AudioSourceProvider* MediaPlayerPrivateGStreamer::audioSourceProvider()
{
ensureAudioSourceProvider();
return m_audioSourceProvider.get();
}
#endif
void MediaPlayerPrivateGStreamer::createGSTPlayBin(const gchar* playbinName)
{
if (m_pipeline) {
if (!playbinName) {
GST_INFO_OBJECT(pipeline(), "Keeping same playbin as nothing forced");
return;
}
if (!g_strcmp0(GST_OBJECT_NAME(gst_element_get_factory(m_pipeline.get())), playbinName)) {
GST_INFO_OBJECT(pipeline(), "Already using %s", playbinName);
return;
}
GST_INFO_OBJECT(pipeline(), "Tearing down as we need to use %s now.",
playbinName);
changePipelineState(GST_STATE_NULL);
m_pipeline = nullptr;
}
ASSERT(!m_pipeline);
#if GST_CHECK_VERSION(1, 10, 0)
m_isLegacyPlaybin = !g_getenv("USE_PLAYBIN3");
if (!m_isLegacyPlaybin)
playbinName = "playbin3";
#endif
if (!playbinName)
playbinName = "playbin";
// gst_element_factory_make() returns a floating reference so
// we should not adopt.
setPipeline(gst_element_factory_make(playbinName, "play"));
setStreamVolumeElement(GST_STREAM_VOLUME(m_pipeline.get()));
GST_INFO("Using legacy playbin element: %s", boolForPrinting(m_isLegacyPlaybin));
// Let also other listeners subscribe to (application) messages in this bus.
GRefPtr<GstBus> bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline.get())));
gst_bus_add_signal_watch_full(bus.get(), RunLoopSourcePriority::RunLoopDispatcher);
g_signal_connect(bus.get(), "message", G_CALLBACK(busMessageCallback), this);
g_object_set(m_pipeline.get(), "mute", m_player->muted(), nullptr);
g_signal_connect_swapped(m_pipeline.get(), "source-setup", G_CALLBACK(sourceSetupCallback), this);
if (m_isLegacyPlaybin) {
g_signal_connect_swapped(m_pipeline.get(), "video-changed", G_CALLBACK(videoChangedCallback), this);
g_signal_connect_swapped(m_pipeline.get(), "audio-changed", G_CALLBACK(audioChangedCallback), this);
}
#if ENABLE(VIDEO_TRACK)
if (m_isLegacyPlaybin)
g_signal_connect_swapped(m_pipeline.get(), "text-changed", G_CALLBACK(textChangedCallback), this);
GstElement* textCombiner = webkitTextCombinerNew();
ASSERT(textCombiner);
g_object_set(m_pipeline.get(), "text-stream-combiner", textCombiner, nullptr);
m_textAppSink = webkitTextSinkNew();
ASSERT(m_textAppSink);
m_textAppSinkPad = adoptGRef(gst_element_get_static_pad(m_textAppSink.get(), "sink"));
ASSERT(m_textAppSinkPad);
GRefPtr<GstCaps> textCaps;
if (webkitGstCheckVersion(1, 13, 0))
textCaps = adoptGRef(gst_caps_new_empty_simple("application/x-subtitle-vtt"));
else
textCaps = adoptGRef(gst_caps_new_empty_simple("text/vtt"));
g_object_set(m_textAppSink.get(), "emit-signals", TRUE, "enable-last-sample", FALSE, "caps", textCaps.get(), nullptr);
g_signal_connect_swapped(m_textAppSink.get(), "new-sample", G_CALLBACK(newTextSampleCallback), this);
g_object_set(m_pipeline.get(), "text-sink", m_textAppSink.get(), nullptr);
#endif
g_object_set(m_pipeline.get(), "video-sink", createVideoSink(), "audio-sink", createAudioSink(), nullptr);
configurePlaySink();
// On 1.4.2 and newer we use the audio-filter property instead.
// See https://bugzilla.gnome.org/show_bug.cgi?id=735748 for
// the reason for using >= 1.4.2 instead of >= 1.4.0.
if (m_preservesPitch && webkitGstCheckVersion(1, 4, 2)) {
GstElement* scale = gst_element_factory_make("scaletempo", nullptr);
if (!scale)
GST_WARNING("Failed to create scaletempo");
else
g_object_set(m_pipeline.get(), "audio-filter", scale, nullptr);
}
if (!m_renderingCanBeAccelerated) {
// If not using accelerated compositing, let GStreamer handle
// the image-orientation tag.
GstElement* videoFlip = gst_element_factory_make("videoflip", nullptr);
if (videoFlip) {
g_object_set(videoFlip, "method", 8, nullptr);
g_object_set(m_pipeline.get(), "video-filter", videoFlip, nullptr);
} else
GST_WARNING("The videoflip element is missing, video rotation support is now disabled. Please check your gst-plugins-good installation.");
}
GRefPtr<GstPad> videoSinkPad = adoptGRef(gst_element_get_static_pad(m_videoSink.get(), "sink"));
if (videoSinkPad)
g_signal_connect_swapped(videoSinkPad.get(), "notify::caps", G_CALLBACK(videoSinkCapsChangedCallback), this);
}
void MediaPlayerPrivateGStreamer::simulateAudioInterruption()
{
GstMessage* message = gst_message_new_request_state(GST_OBJECT(m_pipeline.get()), GST_STATE_PAUSED);
gst_element_post_message(m_pipeline.get(), message);
}
bool MediaPlayerPrivateGStreamer::didPassCORSAccessCheck() const
{
if (WEBKIT_IS_WEB_SRC(m_source.get()))
return webKitSrcPassedCORSAccessCheck(WEBKIT_WEB_SRC(m_source.get()));
return false;
}
bool MediaPlayerPrivateGStreamer::canSaveMediaData() const
{
if (isLiveStream())
return false;
if (m_url.isLocalFile())
return true;
if (m_url.protocolIsInHTTPFamily())
return true;
return false;
}
}
#endif // USE(GSTREAMER)