Source/WebCore/platform/graphics/gstreamer/MediaPlayerPrivateGStreamer.cpp

/*
 * 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) 2014 Cable Television Laboratories, Inc.
 * Copyright (C) 2009, 2019 Igalia S.L
 * Copyright (C) 2015, 2019 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 "GraphicsContext.h"
#include "GStreamerAudioMixer.h"
#include "GStreamerCommon.h"
#include "GStreamerRegistryScanner.h"
#include "HTTPHeaderNames.h"
#include "ImageGStreamer.h"
#include "ImageOrientation.h"
#include "IntRect.h"
#include "Logging.h"
#include "MediaPlayer.h"
#include "MediaPlayerRequestInstallMissingPluginsCallback.h"
#include "MIMETypeRegistry.h"
#include "NotImplemented.h"
#include "SecurityOrigin.h"
#include "TimeRanges.h"
#include "VideoSinkGStreamer.h"
#include "WebKitAudioSinkGStreamer.h"
#include "WebKitWebSourceGStreamer.h"
#include "AudioTrackPrivateGStreamer.h"
#include "InbandMetadataTextTrackPrivateGStreamer.h"
#include "InbandTextTrackPrivateGStreamer.h"
#include "TextCombinerGStreamer.h"
#include "TextSinkGStreamer.h"
#include "VideoTrackPrivateGStreamer.h"

#if ENABLE(MEDIA_STREAM)
#include "GStreamerMediaStreamSource.h"
#include "MediaStreamPrivate.h"
#endif

#if ENABLE(MEDIA_SOURCE)
#include "MediaSource.h"
#include "WebKitMediaSourceGStreamer.h"
#endif

#if ENABLE(ENCRYPTED_MEDIA)
#include "CDMInstance.h"
#include "CDMProxyClearKey.h"
#include "GStreamerEMEUtilities.h"
#include "SharedBuffer.h"
#include "WebKitCommonEncryptionDecryptorGStreamer.h"
#endif

#if ENABLE(WEB_AUDIO)
#include "AudioSourceProviderGStreamer.h"
#endif

#include <glib.h>
#include <gst/audio/streamvolume.h>
#include <gst/gst.h>
#include <gst/pbutils/missing-plugins.h>
#include <gst/video/gstvideometa.h>
#include <limits>
#include <wtf/FileSystem.h>
#include <wtf/glib/GUniquePtr.h>
#include <wtf/glib/RunLoopSourcePriority.h>
#include <wtf/MathExtras.h>
#include <wtf/MediaTime.h>
#include <wtf/NeverDestroyed.h>
#include <wtf/StdLibExtras.h>
#include <wtf/StringPrintStream.h>
#include <wtf/text/AtomString.h>
#include <wtf/text/CString.h>
#include <wtf/text/StringConcatenateNumbers.h>
#include <wtf/URL.h>
#include <wtf/WallTime.h>

#if USE(GSTREAMER_MPEGTS)
#define GST_USE_UNSTABLE_API
#include <gst/mpegts/mpegts.h>
#undef GST_USE_UNSTABLE_API
#endif // ENABLE(VIDEO) && USE(GSTREAMER_MPEGTS)

#if USE(GSTREAMER_GL)
#include "GLVideoSinkGStreamer.h"
#include "VideoTextureCopierGStreamer.h"

#define TEXTURE_COPIER_COLOR_CONVERT_FLAG VideoTextureCopierGStreamer::ColorConversion::NoConvert
#endif // USE(GSTREAMER_GL)

#if USE(TEXTURE_MAPPER_GL)
#include "BitmapTextureGL.h"
#include "BitmapTexturePool.h"
#include "GStreamerVideoFrameHolder.h"
#include "TextureMapperContextAttributes.h"
#include "TextureMapperPlatformLayerBuffer.h"
#include "TextureMapperPlatformLayerProxy.h"
#endif // USE(TEXTURE_MAPPER_GL)

#if USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)
#include "PlatformDisplayLibWPE.h"
#include <gst/gl/egl/gsteglimage.h>
#include <gst/gl/egl/gstglmemoryegl.h>
#include <wpe/extensions/video-plane-display-dmabuf.h>
#endif

GST_DEBUG_CATEGORY(webkit_media_player_debug);
#define GST_CAT_DEFAULT webkit_media_player_debug

namespace WebCore {
using namespace std;

#if USE(GSTREAMER_HOLEPUNCH)
static const FloatSize s_holePunchDefaultFrameSize(1280, 720);
#endif

static void initializeDebugCategory()
{
    static std::once_flag onceFlag;
    std::call_once(onceFlag, [] {
        GST_DEBUG_CATEGORY_INIT(webkit_media_player_debug, "webkitmediaplayer", 0, "WebKit media player");
    });
}

MediaPlayerPrivateGStreamer::MediaPlayerPrivateGStreamer(MediaPlayer* player)
    : m_notifier(MainThreadNotifier<MainThreadNotification>::create())
    , m_player(player)
    , m_referrer(player->referrer())
    , m_cachedDuration(MediaTime::invalidTime())
    , m_seekTime(MediaTime::invalidTime())
    , m_timeOfOverlappingSeek(MediaTime::invalidTime())
    , m_fillTimer(*this, &MediaPlayerPrivateGStreamer::fillTimerFired)
    , m_maxTimeLoaded(MediaTime::zeroTime())
    , m_preload(player->preload())
    , m_maxTimeLoadedAtLastDidLoadingProgress(MediaTime::zeroTime())
    , m_drawTimer(RunLoop::main(), this, &MediaPlayerPrivateGStreamer::repaint)
    , m_readyTimerHandler(RunLoop::main(), this, &MediaPlayerPrivateGStreamer::readyTimerFired)
#if USE(TEXTURE_MAPPER_GL)
#if USE(NICOSIA)
    , m_nicosiaLayer(Nicosia::ContentLayer::create(Nicosia::ContentLayerTextureMapperImpl::createFactory(*this)))
#else
    , m_platformLayerProxy(adoptRef(new TextureMapperPlatformLayerProxy()))
#endif
#endif
#if !RELEASE_LOG_DISABLED
    , m_logger(player->mediaPlayerLogger())
    , m_logIdentifier(player->mediaPlayerLogIdentifier())
#endif
{
#if USE(GLIB)
    m_readyTimerHandler.setPriority(G_PRIORITY_DEFAULT_IDLE);
#endif
    m_isPlayerShuttingDown.store(false);

    ensureGStreamerInitialized();
    m_audioSink = createAudioSink();

#if USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)
    auto& sharedDisplay = PlatformDisplay::sharedDisplay();
    if (is<PlatformDisplayLibWPE>(sharedDisplay))
        m_wpeVideoPlaneDisplayDmaBuf.reset(wpe_video_plane_display_dmabuf_source_create(downcast<PlatformDisplayLibWPE>(sharedDisplay).backend()));
#endif
}

MediaPlayerPrivateGStreamer::~MediaPlayerPrivateGStreamer()
{
    GST_DEBUG_OBJECT(pipeline(), "Disposing player");
    m_isPlayerShuttingDown.store(true);

    for (auto& track : m_audioTracks.values())
        track->disconnect();

    for (auto& track : m_textTracks.values())
        track->disconnect();

    for (auto& track : m_videoTracks.values())
        track->disconnect();

    if (m_fillTimer.isActive())
        m_fillTimer.stop();

    auto* sink = audioSink();
    if (sink && !WEBKIT_IS_AUDIO_SINK(sink))
        g_signal_handlers_disconnect_by_func(G_OBJECT(sink), 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) {
        auto bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline.get())));
        ASSERT(bus);
        g_signal_handlers_disconnect_matched(bus.get(), G_SIGNAL_MATCH_DATA, 0, 0, nullptr, nullptr, this);
        gst_bus_remove_signal_watch(bus.get());
        g_signal_handlers_disconnect_matched(m_pipeline.get(), G_SIGNAL_MATCH_DATA, 0, 0, nullptr, nullptr, this);
    }

#if USE(GSTREAMER_GL)
    if (m_videoDecoderPlatform == GstVideoDecoderPlatform::Video4Linux)
        flushCurrentBuffer();
#endif
#if USE(TEXTURE_MAPPER_GL) && USE(NICOSIA)
    downcast<Nicosia::ContentLayerTextureMapperImpl>(m_nicosiaLayer->impl()).invalidateClient();
#endif

    if (m_videoSink)
        g_signal_handlers_disconnect_matched(m_videoSink.get(), G_SIGNAL_MATCH_DATA, 0, 0, nullptr, nullptr, this);

    if (m_volumeElement)
        g_signal_handlers_disconnect_matched(m_volumeElement.get(), G_SIGNAL_MATCH_DATA, 0, 0, nullptr, nullptr, this);

    // This will release the GStreamer thread from m_drawCondition in non AC mode in case there's an ongoing triggerRepaint call
    // waiting there, and ensure that any triggerRepaint call reaching the lock won't wait on m_drawCondition.
    cancelRepaint(true);

#if ENABLE(ENCRYPTED_MEDIA)
    {
        Locker cdmAttachmentLocker { m_cdmAttachmentLock };
        m_cdmAttachmentCondition.notifyAll();
    }
#endif

    // The change to GST_STATE_NULL state is always synchronous. So after this gets executed we don't need to worry
    // about handlers running in the GStreamer thread.
    if (m_pipeline)
        gst_element_set_state(m_pipeline.get(), GST_STATE_NULL);

    m_player = nullptr;
    m_notifier->invalidate();
}

bool MediaPlayerPrivateGStreamer::isAvailable()
{
    return true;
}

class MediaPlayerFactoryGStreamer final : public MediaPlayerFactory {
private:
    MediaPlayerEnums::MediaEngineIdentifier identifier() const final { return MediaPlayerEnums::MediaEngineIdentifier::GStreamer; };

    std::unique_ptr<MediaPlayerPrivateInterface> createMediaEnginePlayer(MediaPlayer* player) const final
    {
        return makeUnique<MediaPlayerPrivateGStreamer>(player);
    }

    void getSupportedTypes(HashSet<String, ASCIICaseInsensitiveHash>& types) const final
    {
        return MediaPlayerPrivateGStreamer::getSupportedTypes(types);
    }

    MediaPlayer::SupportsType supportsTypeAndCodecs(const MediaEngineSupportParameters& parameters) const final
    {
        return MediaPlayerPrivateGStreamer::supportsType(parameters);
    }

    bool supportsKeySystem(const String& keySystem, const String& mimeType) const final
    {
        return MediaPlayerPrivateGStreamer::supportsKeySystem(keySystem, mimeType);
    }
};

void MediaPlayerPrivateGStreamer::registerMediaEngine(MediaEngineRegistrar registrar)
{
    initializeDebugCategory();
    registrar(makeUnique<MediaPlayerFactoryGStreamer>());
}

void MediaPlayerPrivateGStreamer::load(const String& urlString)
{
    URL url(URL(), urlString);
    if (url.protocolIsAbout()) {
        loadingFailed(MediaPlayer::NetworkState::FormatError, MediaPlayer::ReadyState::HaveNothing, true);
        return;
    }

    if (!ensureGStreamerInitialized()) {
        loadingFailed(MediaPlayer::NetworkState::FormatError, MediaPlayer::ReadyState::HaveNothing, true);
        return;
    }

    registerWebKitGStreamerElements();

    if (!m_pipeline)
        createGSTPlayBin(url);
    syncOnClock(true);
    if (m_fillTimer.isActive())
        m_fillTimer.stop();

    ASSERT(m_pipeline);

    setPlaybinURL(url);

    GST_DEBUG_OBJECT(pipeline(), "preload: %s", convertEnumerationToString(m_preload).utf8().data());
    if (m_preload == MediaPlayer::Preload::None && !isMediaSource()) {
        GST_INFO_OBJECT(pipeline(), "Delaying load.");
        m_isDelayingLoad = true;
    }

    // Reset network and ready states. Those will be set properly once
    // the pipeline pre-rolled.
    m_networkState = MediaPlayer::NetworkState::Loading;
    m_player->networkStateChanged();
    m_readyState = MediaPlayer::ReadyState::HaveNothing;
    m_player->readyStateChanged();
    m_areVolumeAndMuteInitialized = false;
    m_hasTaintedOrigin = std::nullopt;

    if (!m_isDelayingLoad)
        commitLoad();
}

#if ENABLE(MEDIA_SOURCE)
void MediaPlayerPrivateGStreamer::load(const URL&, const ContentType&, MediaSourcePrivateClient*)
{
    // Properly fail so the global MediaPlayer tries to fallback to the next MediaPlayerPrivate.
    m_networkState = MediaPlayer::NetworkState::FormatError;
    m_player->networkStateChanged();
}
#endif

#if ENABLE(MEDIA_STREAM)
void MediaPlayerPrivateGStreamer::load(MediaStreamPrivate& stream)
{
    m_streamPrivate = &stream;
    load(String("mediastream://") + stream.id());
    syncOnClock(false);

    m_player->play();
}
#endif

void MediaPlayerPrivateGStreamer::cancelLoad()
{
    if (m_networkState < MediaPlayer::NetworkState::Loading || m_networkState == MediaPlayer::NetworkState::Loaded)
        return;

    if (m_pipeline)
        changePipelineState(GST_STATE_READY);
}

void MediaPlayerPrivateGStreamer::prepareToPlay()
{
    GST_DEBUG_OBJECT(pipeline(), "Prepare to play");
    m_preload = MediaPlayer::Preload::Auto;
    if (m_isDelayingLoad) {
        m_isDelayingLoad = false;
        commitLoad();
    }
}

void MediaPlayerPrivateGStreamer::play()
{
    if (!m_playbackRate) {
        m_isPlaybackRatePaused = true;
        return;
    }

    if (changePipelineState(GST_STATE_PLAYING)) {
        m_isEndReached = false;
        m_isDelayingLoad = false;
        m_preload = MediaPlayer::Preload::Auto;
        updateDownloadBufferingFlag();
        GST_INFO_OBJECT(pipeline(), "Play");
    } else
        loadingFailed(MediaPlayer::NetworkState::Empty);
}

void MediaPlayerPrivateGStreamer::pause()
{
    m_isPlaybackRatePaused = 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_OBJECT(pipeline(), "Pause");
    else
        loadingFailed(MediaPlayer::NetworkState::Empty);
}

bool MediaPlayerPrivateGStreamer::paused() const
{
    if (!m_pipeline)
        return true;

    if (m_isEndReached) {
        GST_DEBUG_OBJECT(pipeline(), "Ignoring pause at EOS");
        return true;
    }

    if (m_isPlaybackRatePaused) {
        GST_DEBUG_OBJECT(pipeline(), "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_LOG_OBJECT(pipeline(), "Paused: %s", toString(paused).utf8().data());
    return paused;
}

bool MediaPlayerPrivateGStreamer::doSeek(const MediaTime& position, float rate, GstSeekFlags seekFlags)
{
    // Default values for rate >= 0.
    MediaTime startTime = position, endTime = MediaTime::invalidTime();

    if (rate < 0) {
        startTime = MediaTime::zeroTime();
        // If we are at beginning of media, start from the end to avoid immediate EOS.
        endTime = position < MediaTime::zeroTime() ? durationMediaTime() : position;
    }

    if (!rate)
        rate = 1.0;

    if (m_hasWebKitWebSrcSentEOS && m_downloadBuffer) {
        GST_DEBUG_OBJECT(pipeline(), "Setting high-percent=0 on GstDownloadBuffer to force 100%% buffered reporting");
        g_object_set(m_downloadBuffer.get(), "high-percent", 0, nullptr);
    }

    GST_DEBUG_OBJECT(pipeline(), "[Seek] Performing actual seek to %" GST_TIME_FORMAT " (endTime: %" GST_TIME_FORMAT ") at rate %f", GST_TIME_ARGS(toGstClockTime(startTime)), GST_TIME_ARGS(toGstClockTime(endTime)), rate);
    return gst_element_seek(m_pipeline.get(), rate, GST_FORMAT_TIME, seekFlags,
        GST_SEEK_TYPE_SET, toGstClockTime(startTime), GST_SEEK_TYPE_SET, toGstClockTime(endTime));
}

void MediaPlayerPrivateGStreamer::seek(const MediaTime& mediaTime)
{
    if (!m_pipeline || m_didErrorOccur)
        return;

#if ENABLE(MEDIA_STREAM)
    if (WEBKIT_IS_MEDIA_STREAM_SRC(m_source.get()))
        return;
#endif

    GST_INFO_OBJECT(pipeline(), "[Seek] seek attempt to %s", toString(mediaTime).utf8().data());

    // Avoid useless seeking.
    if (mediaTime == currentMediaTime()) {
        GST_DEBUG_OBJECT(pipeline(), "[Seek] Already at requested position. Aborting.");
        return;
    }

    if (m_isLiveStream) {
        GST_DEBUG_OBJECT(pipeline(), "[Seek] Live stream seek unhandled");
        return;
    }

    MediaTime time = std::min(mediaTime, durationMediaTime());
    GST_INFO_OBJECT(pipeline(), "[Seek] seeking to %s", toString(time).utf8().data());

    if (m_isSeeking) {
        m_timeOfOverlappingSeek = time;
        if (m_isSeekPending) {
            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_OBJECT(pipeline(), "[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_isSeekPending = true;
        if (m_isEndReached) {
            GST_DEBUG_OBJECT(pipeline(), "[Seek] reset pipeline");
            m_shouldResetPipeline = true;
            if (!changePipelineState(GST_STATE_PAUSED))
                loadingFailed(MediaPlayer::NetworkState::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_OBJECT(pipeline(), "[Seek] seeking to %s failed", toString(time).utf8().data());
            return;
        }
    }

    m_isSeeking = true;
    m_seekTime = time;
    m_isEndReached = false;
}

void MediaPlayerPrivateGStreamer::updatePlaybackRate()
{
#if ENABLE(MEDIA_STREAM)
    if (WEBKIT_IS_MEDIA_STREAM_SRC(m_source.get()))
        return;
#endif

    if (!m_isChangingRate)
        return;

    GST_INFO_OBJECT(pipeline(), "Set playback 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_shouldPreservePitch && (m_playbackRate < 0.8 || m_playbackRate > 2));

    GST_INFO_OBJECT(pipeline(), mute ? "Need to mute audio" : "Do not need to mute audio");

    if (m_lastPlaybackRate != m_playbackRate) {
        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 {
            GST_ERROR_OBJECT(pipeline(), "Set rate to %f failed", m_playbackRate);
            m_playbackRate = m_lastPlaybackRate;
        }
    }

    if (m_isPlaybackRatePaused) {
        GstState state, 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_isPlaybackRatePaused = false;
    }

    m_isChangingRate = false;
    m_player->rateChanged();
}

MediaTime MediaPlayerPrivateGStreamer::durationMediaTime() const
{
    GST_TRACE_OBJECT(pipeline(), "Cached duration: %s", m_cachedDuration.toString().utf8().data());
    if (m_cachedDuration.isValid())
        return m_cachedDuration;

    MediaTime duration = platformDuration();
    if (!duration || duration.isInvalid())
        return MediaTime::zeroTime();

    m_cachedDuration = duration;

    return m_cachedDuration;
}

MediaTime MediaPlayerPrivateGStreamer::currentMediaTime() const
{
    if (!m_pipeline || m_didErrorOccur)
        return MediaTime::invalidTime();

    GST_TRACE_OBJECT(pipeline(), "seeking: %s, seekTime: %s", boolForPrinting(m_isSeeking), m_seekTime.toString().utf8().data());
    if (m_isSeeking)
        return m_seekTime;

    return playbackPosition();
}

void MediaPlayerPrivateGStreamer::setRate(float rate)
{
    float rateClamped = clampTo(rate, -20.0, 20.0);
    if (rateClamped != rate)
        GST_WARNING_OBJECT(pipeline(), "Clamping original rate (%f) to [-20, 20] (%f), higher rates cause crashes", rate, rateClamped);

    GST_DEBUG_OBJECT(pipeline(), "Setting playback rate to %f", rateClamped);
    // Avoid useless playback rate update.
    if (m_playbackRate == rateClamped) {
        // And make sure that upper layers were notified if rate was set.

        if (!m_isChangingRate && m_player->rate() != m_playbackRate)
            m_player->rateChanged();
        return;
    }

    if (m_isLiveStream) {
        // Notify upper layers that we cannot handle passed rate.
        m_isChangingRate = false;
        m_player->rateChanged();
        return;
    }

    GstState state, pending;

    m_playbackRate = rateClamped;
    m_isChangingRate = true;

    gst_element_get_state(m_pipeline.get(), &state, &pending, 0);

    if (!rateClamped) {
        m_isChangingRate = false;
        m_isPlaybackRatePaused = 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)
{
    GST_DEBUG_OBJECT(pipeline(), "Preserving audio pitch: %s", boolForPrinting(preservesPitch));
    m_shouldPreservePitch = preservesPitch;
}

void MediaPlayerPrivateGStreamer::setPreload(MediaPlayer::Preload preload)
{
#if ENABLE(MEDIA_STREAM)
    if (WEBKIT_IS_MEDIA_STREAM_SRC(m_source.get()))
        return;
#endif

    GST_DEBUG_OBJECT(pipeline(), "Setting preload to %s", convertEnumerationToString(preload).utf8().data());
    if (preload == MediaPlayer::Preload::Auto && m_isLiveStream)
        return;

    m_preload = preload;
    updateDownloadBufferingFlag();

    if (m_isDelayingLoad && m_preload != MediaPlayer::Preload::None) {
        m_isDelayingLoad = false;
        commitLoad();
    }
}

std::unique_ptr<PlatformTimeRanges> MediaPlayerPrivateGStreamer::buffered() const
{
    auto timeRanges = makeUnique<PlatformTimeRanges>();
    if (m_didErrorOccur || m_isLiveStream)
        return timeRanges;

    MediaTime mediaDuration = durationMediaTime();
    if (!mediaDuration || mediaDuration.isPositiveInfinite())
        return timeRanges;

    GRefPtr<GstQuery> query = adoptGRef(gst_query_new_buffering(GST_FORMAT_PERCENT));

    if (!gst_element_query(m_pipeline.get(), query.get()))
        return timeRanges;

    unsigned numBufferingRanges = gst_query_get_n_buffering_ranges(query.get());
    for (unsigned index = 0; index < numBufferingRanges; index++) {
        gint64 rangeStart = 0, rangeStop = 0;
        if (gst_query_parse_nth_buffering_range(query.get(), index, &rangeStart, &rangeStop)) {
            uint64_t startTime = gst_util_uint64_scale_int_round(toGstUnsigned64Time(mediaDuration), rangeStart, GST_FORMAT_PERCENT_MAX);
            uint64_t stopTime = gst_util_uint64_scale_int_round(toGstUnsigned64Time(mediaDuration), rangeStop, GST_FORMAT_PERCENT_MAX);
            timeRanges->add(MediaTime(startTime, GST_SECOND), MediaTime(stopTime, GST_SECOND));
        }
    }

    // Fallback to the more general maxTimeLoaded() if no range has been found.
    if (!timeRanges->length()) {
        MediaTime loaded = maxTimeLoaded();
        if (loaded.isValid() && loaded)
            timeRanges->add(MediaTime::zeroTime(), loaded);
    }

    return timeRanges;
}

MediaTime MediaPlayerPrivateGStreamer::maxMediaTimeSeekable() const
{
    GST_TRACE_OBJECT(pipeline(), "errorOccured: %s, isLiveStream: %s", boolForPrinting(m_didErrorOccur), boolForPrinting(m_isLiveStream));
    if (m_didErrorOccur)
        return MediaTime::zeroTime();

    if (m_isLiveStream)
        return MediaTime::zeroTime();

#if ENABLE(MEDIA_STREAM)
    if (WEBKIT_IS_MEDIA_STREAM_SRC(m_source.get()))
        return MediaTime::zeroTime();
#endif

    MediaTime duration = durationMediaTime();
    GST_DEBUG_OBJECT(pipeline(), "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_didErrorOccur)
        return MediaTime::zeroTime();

    MediaTime loaded = m_maxTimeLoaded;
    if (m_isEndReached)
        loaded = durationMediaTime();
    GST_LOG_OBJECT(pipeline(), "maxTimeLoaded: %s", toString(loaded).utf8().data());
    return loaded;
}

bool MediaPlayerPrivateGStreamer::didLoadingProgress() const
{
    if (m_didErrorOccur || m_loadingStalled)
        return false;

    if (WEBKIT_IS_WEB_SRC(m_source.get())) {
        GST_LOG_OBJECT(pipeline(), "Last network read position: %" G_GUINT64_FORMAT ", current: %" G_GUINT64_FORMAT, m_readPositionAtLastDidLoadingProgress, m_networkReadPosition);
        bool didLoadingProgress = m_readPositionAtLastDidLoadingProgress < m_networkReadPosition;
        m_readPositionAtLastDidLoadingProgress = m_networkReadPosition;
        GST_LOG_OBJECT(pipeline(), "didLoadingProgress: %s", boolForPrinting(didLoadingProgress));
        return didLoadingProgress;
    }

    if (UNLIKELY(!m_pipeline || !durationMediaTime() || (!isMediaSource() && !totalBytes())))
        return false;

    MediaTime currentMaxTimeLoaded = maxTimeLoaded();
    bool didLoadingProgress = currentMaxTimeLoaded != m_maxTimeLoadedAtLastDidLoadingProgress;
    m_maxTimeLoadedAtLastDidLoadingProgress = currentMaxTimeLoaded;
    GST_LOG_OBJECT(pipeline(), "didLoadingProgress: %s", boolForPrinting(didLoadingProgress));
    return didLoadingProgress;
}

unsigned long long MediaPlayerPrivateGStreamer::totalBytes() const
{
    if (m_didErrorOccur || !m_source || m_isLiveStream)
        return 0;

    if (m_totalBytes)
        return m_totalBytes;

    GstFormat fmt = GST_FORMAT_BYTES;
    gint64 length = 0;
    if (gst_element_query_duration(m_source.get(), fmt, &length)) {
        GST_INFO_OBJECT(pipeline(), "totalBytes %" G_GINT64_FORMAT, length);
        m_totalBytes = static_cast<unsigned long long>(length);
        m_isLiveStream = !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_OBJECT(pipeline(), "totalBytes %" G_GINT64_FORMAT, length);
    m_totalBytes = static_cast<unsigned long long>(length);
    m_isLiveStream = !length;
    return m_totalBytes;
}

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());
}

std::optional<bool> MediaPlayerPrivateGStreamer::wouldTaintOrigin(const SecurityOrigin& origin) const
{
    GST_TRACE_OBJECT(pipeline(), "Checking %u origins", m_origins.size());
    for (auto& responseOrigin : m_origins) {
        if (!origin.isSameOriginDomain(*responseOrigin)) {
            GST_DEBUG_OBJECT(pipeline(), "Found reachable response origin");
            return true;
        }
    }
    GST_DEBUG_OBJECT(pipeline(), "No valid response origin found");
    return false;
}

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);
}

#if ENABLE(WEB_AUDIO)
void MediaPlayerPrivateGStreamer::ensureAudioSourceProvider()
{
    if (!m_audioSourceProvider)
        m_audioSourceProvider = makeUnique<AudioSourceProviderGStreamer>();
}

AudioSourceProvider* MediaPlayerPrivateGStreamer::audioSourceProvider()
{
    ensureAudioSourceProvider();
    return m_audioSourceProvider.get();
}
#endif

void MediaPlayerPrivateGStreamer::durationChanged()
{
    MediaTime previousDuration = durationMediaTime();
    m_cachedDuration = MediaTime::invalidTime();

    // Avoid emitting 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::sourceSetup(GstElement* sourceElement)
{
    GST_DEBUG_OBJECT(pipeline(), "Source element set-up for %s", GST_ELEMENT_NAME(sourceElement));

    m_source = sourceElement;

    if (WEBKIT_IS_WEB_SRC(m_source.get())) {
        webKitWebSrcSetMediaPlayer(WEBKIT_WEB_SRC_CAST(m_source.get()), m_player, m_referrer);
#if ENABLE(MEDIA_STREAM)
    } else if (WEBKIT_IS_MEDIA_STREAM_SRC(sourceElement)) {
        auto stream = m_streamPrivate.get();
        ASSERT(stream);
        webkitMediaStreamSrcSetStream(WEBKIT_MEDIA_STREAM_SRC(sourceElement), stream, m_player->isVideoPlayer());
#endif
    }
}

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->isVideoPlayer() ? "video" : "music";
    GUniquePtr<GstStructure> properties(gst_structure_new("stream-properties", "media.role", G_TYPE_STRING, role, nullptr));
    g_object_set(object, "stream-properties", properties.get(), nullptr);
    GUniquePtr<gchar> elementName(gst_element_get_name(GST_ELEMENT(object)));
    GST_DEBUG_OBJECT(pipeline(), "Set media.role as %s at %s", role, elementName.get());
}

void MediaPlayerPrivateGStreamer::sourceSetupCallback(MediaPlayerPrivateGStreamer* player, GstElement* sourceElement)
{
    player->sourceSetup(sourceElement);
}

bool MediaPlayerPrivateGStreamer::changePipelineState(GstState newState)
{
    ASSERT(m_pipeline);

    GstState currentState, pending;

    gst_element_get_state(m_pipeline.get(), &currentState, &pending, 0);
    if (currentState == newState || pending == newState) {
        GST_DEBUG_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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));

    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::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);
    GST_INFO_OBJECT(pipeline(), "Load %s", m_url.string().utf8().data());
    g_object_set(m_pipeline.get(), "uri", m_url.string().utf8().data(), nullptr);
}

static void setSyncOnClock(GstElement *element, bool sync)
{
    if (!element)
        return;

    if (!GST_IS_BIN(element)) {
        g_object_set(element, "sync", sync, nullptr);
        return;
    }

    GUniquePtr<GstIterator> iterator(gst_bin_iterate_sinks(GST_BIN_CAST(element)));
    while (gst_iterator_foreach(iterator.get(), static_cast<GstIteratorForeachFunction>([](const GValue* item, void* syncPtr) {
        bool* sync = static_cast<bool*>(syncPtr);
        setSyncOnClock(GST_ELEMENT_CAST(g_value_get_object(item)), *sync);
    }), &sync) == GST_ITERATOR_RESYNC)
        gst_iterator_resync(iterator.get());
}

void MediaPlayerPrivateGStreamer::syncOnClock(bool sync)
{
    setSyncOnClock(videoSink(), sync);
    setSyncOnClock(audioSink(), sync);
}

template <typename TrackPrivateType>
void MediaPlayerPrivateGStreamer::notifyPlayerOfTrack()
{
    if (UNLIKELY(!m_pipeline || !m_source))
        return;

    ASSERT(m_isLegacyPlaybin);

    using TrackType = TrackPrivateBaseGStreamer::TrackType;
    std::variant<HashMap<AtomString, RefPtr<AudioTrackPrivateGStreamer>>*, HashMap<AtomString, RefPtr<VideoTrackPrivateGStreamer>>*, HashMap<AtomString, RefPtr<InbandTextTrackPrivateGStreamer>>*> variantTracks = static_cast<HashMap<AtomString, RefPtr<TrackPrivateType>>*>(0);
    auto type(static_cast<TrackType>(variantTracks.index()));
    const char* typeName;
    bool* hasType;
    switch (type) {
    case TrackType::Audio:
        typeName = "audio";
        hasType = &m_hasAudio;
        variantTracks = &m_audioTracks;
        break;
    case TrackType::Video:
        typeName = "video";
        hasType = &m_hasVideo;
        variantTracks = &m_videoTracks;
        break;
    case TrackType::Text:
        typeName = "text";
        hasType = nullptr;
        variantTracks = &m_textTracks;
        break;
    default:
        ASSERT_NOT_REACHED();
    }
    HashMap<AtomString, RefPtr<TrackPrivateType>>& tracks = *std::get<HashMap<AtomString, RefPtr<TrackPrivateType>>*>(variantTracks);

    // Ignore notifications after a EOS. We don't want the tracks to disappear when the video is finished.
    if (m_isEndReached && (type == TrackType::Audio || type == TrackType::Video))
        return;

    unsigned numberOfTracks = 0;
    StringPrintStream numberOfTracksProperty;
    numberOfTracksProperty.printf("n-%s", typeName);
    g_object_get(m_pipeline.get(), numberOfTracksProperty.toCString().data(), &numberOfTracks, nullptr);

    GST_INFO_OBJECT(pipeline(), "Media has %d %s tracks", numberOfTracks, typeName);

    if (hasType) {
        bool oldHasType = *hasType;
        *hasType = numberOfTracks > 0;
        if (oldHasType != *hasType)
            m_player->characteristicChanged();

        if (*hasType && type == TrackType::Video)
            m_player->sizeChanged();
    }

    Vector<AtomString> validStreams;
    StringPrintStream getPadProperty;
    getPadProperty.printf("get-%s-pad", typeName);

    for (unsigned i = 0; i < numberOfTracks; ++i) {
        GRefPtr<GstPad> pad;
        g_signal_emit_by_name(m_pipeline.get(), getPadProperty.toCString().data(), i, &pad.outPtr(), nullptr);
        ASSERT(pad);

        // The pad might not have a sticky stream-start event yet, so we can't use
        // gst_pad_get_stream_id() here.
        auto streamId = TrackPrivateBaseGStreamer::generateUniquePlaybin2StreamID(type, i);
        validStreams.append(streamId);

        if (i < tracks.size()) {
            RefPtr<TrackPrivateType> existingTrack = tracks.get(streamId);
            if (existingTrack) {
                existingTrack->setIndex(i);
                // If the video has been played twice, the track is still there, but we need
                // to update the pad pointer.
                if (existingTrack->pad() != pad)
                    existingTrack->setPad(GRefPtr(pad));
                continue;
            }
        }

        auto track = TrackPrivateType::create(*this, i, WTFMove(pad));
        if (!track->trackIndex() && (type == TrackType::Audio || type == TrackType::Video))
            track->setActive(true);
        ASSERT(streamId == track->id());
        tracks.add(streamId, track.copyRef());

        std::variant<AudioTrackPrivate*, VideoTrackPrivate*, InbandTextTrackPrivate*> variantTrack(&track.get());
        switch (variantTrack.index()) {
        case TrackType::Audio:
            m_player->addAudioTrack(*std::get<AudioTrackPrivate*>(variantTrack));
            break;
        case TrackType::Video:
            m_player->addVideoTrack(*std::get<VideoTrackPrivate*>(variantTrack));
            break;
        case TrackType::Text:
            m_player->addTextTrack(*std::get<InbandTextTrackPrivate*>(variantTrack));
            break;
        }
    }

    // Purge invalid tracks
    tracks.removeIf([validStreams](auto& keyAndValue) {
        return !validStreams.contains(keyAndValue.key);
    });

    m_player->mediaEngineUpdated();
}

bool MediaPlayerPrivateGStreamer::hasFirstVideoSampleReachedSink() const
{
    Locker sampleLocker { m_sampleMutex };
    return !!m_sample;
}

void MediaPlayerPrivateGStreamer::videoSinkCapsChanged(GstPad* videoSinkPad)
{
    GRefPtr<GstCaps> caps = adoptGRef(gst_pad_get_current_caps(videoSinkPad));
    if (!caps) {
        // This can happen when downgrading the state of the pipeline, which causes the caps to be unset.
        return;
    }
    // We're in videoSinkPad streaming thread.
    ASSERT(!isMainThread());
    GST_DEBUG_OBJECT(videoSinkPad, "Received new caps: %" GST_PTR_FORMAT, caps.get());

    if (!hasFirstVideoSampleReachedSink()) {
        // We want to wait for the sink to receive the first buffer before emitting dimensions, since only by then we
        // are guaranteed that any potential tag event with a rotation has been handled.
        GST_DEBUG_OBJECT(videoSinkPad, "Ignoring notify::caps until the first buffer reaches the sink.");
        return;
    }

    RunLoop::main().dispatch([weakThis = WeakPtr { *this }, this, caps = WTFMove(caps)] {
        if (!weakThis)
            return;
        updateVideoSizeAndOrientationFromCaps(caps.get());
    });
}

void MediaPlayerPrivateGStreamer::audioChangedCallback(MediaPlayerPrivateGStreamer* player)
{
    player->m_notifier->notify(MainThreadNotification::AudioChanged, [player] {
        player->notifyPlayerOfTrack<AudioTrackPrivateGStreamer>();
    });
}

void MediaPlayerPrivateGStreamer::textChangedCallback(MediaPlayerPrivateGStreamer* player)
{
    player->m_notifier->notify(MainThreadNotification::TextChanged, [player] {
        player->notifyPlayerOfTrack<InbandTextTrackPrivateGStreamer>();
    });
}

void MediaPlayerPrivateGStreamer::handleTextSample(GstSample* sample, const char* streamId)
{
    for (auto& track : m_textTracks.values()) {
        if (!strcmp(track->id().string().utf8().data(), streamId)) {
            track->handleSample(sample);
            return;
        }
    }

    GST_WARNING_OBJECT(m_pipeline.get(), "Got sample with unknown stream ID %s.", streamId);
}

MediaTime MediaPlayerPrivateGStreamer::platformDuration() const
{
    if (!m_pipeline)
        return MediaTime::invalidTime();

    GST_TRACE_OBJECT(pipeline(), "errorOccured: %s, pipeline state: %s", boolForPrinting(m_didErrorOccur), gst_element_state_get_name(GST_STATE(m_pipeline.get())));
    if (m_didErrorOccur)
        return MediaTime::invalidTime();

    // The duration query would fail on a not-prerolled pipeline.
    if (GST_STATE(m_pipeline.get()) < GST_STATE_PAUSED)
        return MediaTime::invalidTime();

    int64_t duration = 0;
    if (!gst_element_query_duration(m_pipeline.get(), GST_FORMAT_TIME, &duration) || !GST_CLOCK_TIME_IS_VALID(duration)) {
        GST_DEBUG_OBJECT(pipeline(), "Time duration query failed for %s", m_url.string().utf8().data());
        return MediaTime::positiveInfiniteTime();
    }

    GST_LOG_OBJECT(pipeline(), "Duration: %" GST_TIME_FORMAT, GST_TIME_ARGS(duration));
    return MediaTime(duration, GST_SECOND);
}

bool MediaPlayerPrivateGStreamer::isMuted() const
{
    if (!m_volumeElement)
        return false;

    gboolean isMuted;
    g_object_get(m_volumeElement.get(), "mute", &isMuted, nullptr);
    GST_INFO_OBJECT(pipeline(), "Player is muted: %s", boolForPrinting(isMuted));
    return isMuted;
}

void MediaPlayerPrivateGStreamer::commitLoad()
{
    ASSERT(!m_isDelayingLoad);
    GST_DEBUG_OBJECT(pipeline(), "Committing load.");

    // GStreamer needs to have the pipeline set to a paused state to
    // start providing anything useful.
    changePipelineState(GST_STATE_PAUSED);

    updateDownloadBufferingFlag();
    updateStates();
}

void MediaPlayerPrivateGStreamer::fillTimerFired()
{
    if (m_didErrorOccur) {
        GST_DEBUG_OBJECT(pipeline(), "[Buffering] An error occurred, disabling the fill timer");
        m_fillTimer.stop();
        return;
    }

    GRefPtr<GstQuery> query = adoptGRef(gst_query_new_buffering(GST_FORMAT_PERCENT));
    double fillStatus = 100.0;
    GstBufferingMode mode = GST_BUFFERING_DOWNLOAD;

    if (gst_element_query(pipeline(), query.get())) {
        gst_query_parse_buffering_stats(query.get(), &mode, nullptr, nullptr, nullptr);

        int percentage;
        gst_query_parse_buffering_percent(query.get(), nullptr, &percentage);
        fillStatus = percentage;
    } else if (m_httpResponseTotalSize) {
        GST_DEBUG_OBJECT(pipeline(), "[Buffering] Query failed, falling back to network read position estimation");
        fillStatus = 100.0 * (static_cast<double>(m_networkReadPosition) / static_cast<double>(m_httpResponseTotalSize));
    } else {
        GST_DEBUG_OBJECT(pipeline(), "[Buffering] Unable to determine on-disk buffering status");
        return;
    }

    updateBufferingStatus(mode, fillStatus);
}

void MediaPlayerPrivateGStreamer::loadStateChanged()
{
    updateStates();
}

void MediaPlayerPrivateGStreamer::timeChanged()
{
    updateStates();
    GST_DEBUG_OBJECT(pipeline(), "Emitting timeChanged notification");
    m_player->timeChanged();
}

void MediaPlayerPrivateGStreamer::loadingFailed(MediaPlayer::NetworkState networkError, MediaPlayer::ReadyState readyState, bool forceNotifications)
{
    GST_WARNING("Loading failed, error: %s", convertEnumerationToString(networkError).utf8().data());

    m_didErrorOccur = true;
    if (forceNotifications || m_networkState != networkError) {
        m_networkState = networkError;
        m_player->networkStateChanged();
    }
    if (forceNotifications || m_readyState != readyState) {
        m_readyState = readyState;
        m_player->readyStateChanged();
    }

    // Loading failed, remove ready timer.
    m_readyTimerHandler.stop();
}

GstElement* MediaPlayerPrivateGStreamer::createAudioSink()
{
    GstElement* audioSink = createPlatformAudioSink();
    RELEASE_ASSERT(audioSink);
    if (!audioSink)
        return nullptr;

    if (!WEBKIT_IS_AUDIO_SINK(audioSink))
        g_signal_connect_swapped(audioSink, "child-added", G_CALLBACK(setAudioStreamPropertiesCallback), this);

#if ENABLE(WEB_AUDIO)
    GstElement* audioSinkBin = gst_bin_new("audio-sink");
    ensureAudioSourceProvider();
    m_audioSourceProvider->configureAudioBin(audioSinkBin, audioSink);
    return audioSinkBin;
#else
    return audioSink;
#endif
}

GstElement* MediaPlayerPrivateGStreamer::audioSink() const
{
    if (!m_pipeline)
        return nullptr;

    GstElement* sink;
    g_object_get(m_pipeline.get(), "audio-sink", &sink, nullptr);
    return sink;
}

GstClockTime MediaPlayerPrivateGStreamer::gstreamerPositionFromSinks() const
{
    gint64 gstreamerPosition = GST_CLOCK_TIME_NONE;
    // Asking directly to the sinks and choosing the highest value is faster than asking to the pipeline.
    GRefPtr<GstQuery> query = adoptGRef(gst_query_new_position(GST_FORMAT_TIME));
    if (m_audioSink && gst_element_query(m_audioSink.get(), query.get())) {
        gint64 audioPosition = GST_CLOCK_TIME_NONE;
        gst_query_parse_position(query.get(), 0, &audioPosition);
        if (GST_CLOCK_TIME_IS_VALID(audioPosition))
            gstreamerPosition = audioPosition;
        GST_TRACE_OBJECT(pipeline(), "Audio position %" GST_TIME_FORMAT, GST_TIME_ARGS(audioPosition));
        query = adoptGRef(gst_query_new_position(GST_FORMAT_TIME));
    }
    if (m_videoSink && gst_element_query(m_videoSink.get(), query.get())) {
        gint64 videoPosition = GST_CLOCK_TIME_NONE;
        gst_query_parse_position(query.get(), 0, &videoPosition);
        GST_TRACE_OBJECT(pipeline(), "Video position %" GST_TIME_FORMAT, GST_TIME_ARGS(videoPosition));
        if (GST_CLOCK_TIME_IS_VALID(videoPosition) && (!GST_CLOCK_TIME_IS_VALID(gstreamerPosition)
            || (m_playbackRate >= 0 && videoPosition > gstreamerPosition)
            || (m_playbackRate < 0 && videoPosition < gstreamerPosition)))
            gstreamerPosition = videoPosition;
    }
    return static_cast<GstClockTime>(gstreamerPosition);
}

MediaTime MediaPlayerPrivateGStreamer::playbackPosition() const
{
    GST_TRACE_OBJECT(pipeline(), "isEndReached: %s, seeking: %s, seekTime: %s", boolForPrinting(m_isEndReached), boolForPrinting(m_isSeeking), m_seekTime.toString().utf8().data());

#if ENABLE(MEDIA_STREAM)
    if (m_streamPrivate && m_player->isVideoPlayer() && !hasFirstVideoSampleReachedSink())
        return MediaTime::zeroTime();
#endif

    if (m_isSeeking)
        return m_seekTime;
    if (m_isEndReached)
        return m_playbackRate > 0 ? durationMediaTime() : MediaTime::zeroTime();

    if (m_cachedPosition) {
        GST_TRACE_OBJECT(pipeline(), "Returning cached position: %s", m_cachedPosition.value().toString().utf8().data());
        return m_cachedPosition.value();
    }

    GstClockTime gstreamerPosition = gstreamerPositionFromSinks();
    GST_TRACE_OBJECT(pipeline(), "Position %" GST_TIME_FORMAT ", canFallBackToLastFinishedSeekPosition: %s", GST_TIME_ARGS(gstreamerPosition), boolForPrinting(m_canFallBackToLastFinishedSeekPosition));

    MediaTime playbackPosition = MediaTime::zeroTime();

    if (GST_CLOCK_TIME_IS_VALID(gstreamerPosition))
        playbackPosition = MediaTime(gstreamerPosition, GST_SECOND);
    else if (m_canFallBackToLastFinishedSeekPosition)
        playbackPosition = m_seekTime;

    m_cachedPosition = playbackPosition;
    invalidateCachedPositionOnNextIteration();
    return playbackPosition;
}

void MediaPlayerPrivateGStreamer::updateEnabledVideoTrack()
{
    VideoTrackPrivateGStreamer* wantedTrack = nullptr;
    for (auto& pair : m_videoTracks) {
        VideoTrackPrivateGStreamer* track = pair.value.get();
        if (track->selected()) {
            wantedTrack = track;
            break;
        }
    }

    // No active track, no changes.
    if (!wantedTrack)
        return;

    if (m_isLegacyPlaybin) {
        GST_DEBUG_OBJECT(m_pipeline.get(), "Setting playbin2 current-video=%d", wantedTrack->trackIndex());
        g_object_set(m_pipeline.get(), "current-video", wantedTrack->trackIndex(), nullptr);
    } else {
        m_wantedVideoStreamId = wantedTrack->id();
        playbin3SendSelectStreamsIfAppropriate();
    }
}

void MediaPlayerPrivateGStreamer::updateEnabledAudioTrack()
{
    AudioTrackPrivateGStreamer* wantedTrack = nullptr;
    for (auto& pair : m_audioTracks) {
        AudioTrackPrivateGStreamer* track = pair.value.get();
        if (track->enabled()) {
            wantedTrack = track;
            break;
        }
    }

    // No active track, no changes.
    if (!wantedTrack)
        return;

    if (m_isLegacyPlaybin) {
        GST_DEBUG_OBJECT(m_pipeline.get(), "Setting playbin2 current-audio=%d", wantedTrack->trackIndex());
        g_object_set(m_pipeline.get(), "current-audio", wantedTrack->trackIndex(), nullptr);
    } else {
        m_wantedAudioStreamId = wantedTrack->id();
        playbin3SendSelectStreamsIfAppropriate();
    }
}

void MediaPlayerPrivateGStreamer::playbin3SendSelectStreamsIfAppropriate()
{
    ASSERT(!m_isLegacyPlaybin);

    bool haveDifferentStreamIds = (m_wantedAudioStreamId != m_currentAudioStreamId || m_wantedVideoStreamId != m_currentVideoStreamId);
    bool shouldSendSelectStreams = !m_waitingForStreamsSelectedEvent && haveDifferentStreamIds && m_currentState == GST_STATE_PLAYING;
    GST_DEBUG_OBJECT(m_pipeline.get(), "Checking if to send SELECT_STREAMS, m_waitingForStreamsSelectedEvent = %s, haveDifferentStreamIds = %s, m_currentState = %s... shouldSendSelectStreams = %s",
        boolForPrinting(m_waitingForStreamsSelectedEvent), boolForPrinting(haveDifferentStreamIds), gst_element_state_get_name(m_currentState), boolForPrinting(shouldSendSelectStreams));
    if (!shouldSendSelectStreams)
        return;

    GList* streams = nullptr;
    if (!m_wantedVideoStreamId.isNull()) {
        m_requestedVideoStreamId = m_wantedVideoStreamId;
        streams = g_list_append(streams, g_strdup(m_wantedVideoStreamId.string().utf8().data()));
    }
    if (!m_wantedAudioStreamId.isNull()) {
        m_requestedAudioStreamId = m_wantedAudioStreamId;
        streams = g_list_append(streams, g_strdup(m_wantedAudioStreamId.string().utf8().data()));
    }

    if (!streams)
        return;

    m_waitingForStreamsSelectedEvent = true;
    gst_element_send_event(m_pipeline.get(), gst_event_new_select_streams(streams));
    g_list_free_full(streams, reinterpret_cast<GDestroyNotify>(g_free));
}

void MediaPlayerPrivateGStreamer::updateTracks(const GRefPtr<GstStreamCollection>& streamCollection)
{
    ASSERT(!m_isLegacyPlaybin);

    bool useMediaSource = isMediaSource();
    unsigned length = gst_stream_collection_get_size(streamCollection.get());
    GST_DEBUG_OBJECT(pipeline(), "Processing a stream collection with %u streams", length);

    bool oldHasAudio = m_hasAudio;
    bool oldHasVideo = m_hasVideo;

    // New stream collections override previous ones so in order to keep our internal tracks
    // consistent with the collection contents, we can't reuse our old tracks.
#define REMOVE_OLD_TRACKS(type, Type) G_STMT_START {             \
        for (const auto& trackId : m_##type##Tracks.keys()) {    \
            auto track = m_##type##Tracks.get(trackId);          \
            m_player->remove##Type##Track(*track);               \
        }                                                        \
        m_##type##Tracks.clear();                                \
    } G_STMT_END

    REMOVE_OLD_TRACKS(audio, Audio);
    REMOVE_OLD_TRACKS(video, Video);
    REMOVE_OLD_TRACKS(text, Text);

    unsigned audioTrackIndex = 0;
    unsigned videoTrackIndex = 0;
    unsigned textTrackIndex = 0;

#define CREATE_TRACK(type, Type) G_STMT_START {                     \
        RefPtr<Type##TrackPrivateGStreamer> track = Type##TrackPrivateGStreamer::create(*this, type##TrackIndex, WTFMove(stream)); \
        auto trackId = track->id();                                 \
        if (!type##TrackIndex) {                                    \
            m_wanted##Type##StreamId = trackId;                     \
            m_requested##Type##StreamId = trackId;                  \
            track->setActive(true);                                 \
        }                                                           \
        type##TrackIndex++;                                         \
        m_##type##Tracks.add(trackId, track);                       \
        m_player->add##Type##Track(*track);                         \
    } G_STMT_END

    for (unsigned i = 0; i < length; i++) {
        GRefPtr<GstStream> stream = gst_stream_collection_get_stream(streamCollection.get(), i);
        const char* streamId = gst_stream_get_stream_id(stream.get());
        auto type = gst_stream_get_stream_type(stream.get());

        GST_DEBUG_OBJECT(pipeline(), "Inspecting %s track with ID %s", gst_stream_type_get_name(type), streamId);

        if (type & GST_STREAM_TYPE_AUDIO) {
            CREATE_TRACK(audio, Audio);
            configureMediaStreamAudioTracks();
        } else if (type & GST_STREAM_TYPE_VIDEO && m_player->isVideoPlayer())
            CREATE_TRACK(video, Video);
        else if (type & GST_STREAM_TYPE_TEXT && !useMediaSource) {
            auto track = InbandTextTrackPrivateGStreamer::create(textTrackIndex++, WTFMove(stream));
            m_textTracks.add(streamId, track.copyRef());
            m_player->addTextTrack(track.get());
        } else
            GST_WARNING("Unknown track type found for stream %s", streamId);
    }

    m_hasAudio = !m_audioTracks.isEmpty();
    m_hasVideo = !m_videoTracks.isEmpty();

    if (oldHasVideo != m_hasVideo || oldHasAudio != m_hasAudio)
        m_player->characteristicChanged();

    if (!oldHasVideo && m_hasVideo)
        m_player->sizeChanged();

    m_player->mediaEngineUpdated();
}

void MediaPlayerPrivateGStreamer::videoChangedCallback(MediaPlayerPrivateGStreamer* player)
{
    player->m_notifier->notify(MainThreadNotification::VideoChanged, [player] {
        player->notifyPlayerOfTrack<VideoTrackPrivateGStreamer>();
    });
}

void MediaPlayerPrivateGStreamer::handleStreamCollectionMessage(GstMessage* message)
{
    if (m_isLegacyPlaybin)
        return;

    // GStreamer workaround: Unfortunately, when we have a stream-collection aware source (like
    // WebKitMediaSrc) parsebin and decodebin3 emit their own stream-collection messages, but late,
    // and sometimes with duplicated streams. Let's only listen for stream-collection messages from
    // the source to avoid these issues.
    if (isMediaSource() && GST_MESSAGE_SRC(message) != GST_OBJECT(m_source.get())) {
        GST_DEBUG_OBJECT(pipeline(), "Ignoring redundant STREAM_COLLECTION from %" GST_PTR_FORMAT, message->src);
        return;
    }

    ASSERT(GST_MESSAGE_TYPE(message) == GST_MESSAGE_STREAM_COLLECTION);
    GRefPtr<GstStreamCollection> collection;
    gst_message_parse_stream_collection(message, &collection.outPtr());
    if (!collection)
        return;

#ifndef GST_DISABLE_DEBUG
    GST_DEBUG_OBJECT(pipeline(), "Received STREAM_COLLECTION message with upstream id \"%s\" from %" GST_PTR_FORMAT " defining the following streams:", gst_stream_collection_get_upstream_id(collection.get()), GST_MESSAGE_SRC(message));
    unsigned numStreams = gst_stream_collection_get_size(collection.get());
    for (unsigned i = 0; i < numStreams; i++) {
        GstStream* stream = gst_stream_collection_get_stream(collection.get(), i);
        GST_DEBUG_OBJECT(pipeline(), "#%u %s %s", i, gst_stream_type_get_name(gst_stream_get_stream_type(stream)), gst_stream_get_stream_id(stream));
    }
#endif

    auto callback = [player = WeakPtr { *this }, collection = WTFMove(collection)] {
        if (player)
            player->updateTracks(collection);
    };

    if (isMediaSource())
        callOnMainThreadAndWait(WTFMove(callback));
    else
        callOnMainThread(WTFMove(callback));
}

bool MediaPlayerPrivateGStreamer::handleNeedContextMessage(GstMessage* message)
{
    ASSERT(GST_MESSAGE_TYPE(message) == GST_MESSAGE_NEED_CONTEXT);

    const gchar* contextType;
    if (!gst_message_parse_context_type(message, &contextType))
        return false;

    GST_DEBUG_OBJECT(pipeline(), "Handling %s need-context message for %s", contextType, GST_MESSAGE_SRC_NAME(message));

    if (!g_strcmp0(contextType, WEBKIT_WEB_SRC_PLAYER_CONTEXT_TYPE_NAME)) {
        GRefPtr<GstContext> context = adoptGRef(gst_context_new(WEBKIT_WEB_SRC_PLAYER_CONTEXT_TYPE_NAME, FALSE));
        GstStructure* contextStructure = gst_context_writable_structure(context.get());

        ASSERT(m_player);
        gst_structure_set(contextStructure, "player", G_TYPE_POINTER, m_player, nullptr);
        gst_element_set_context(GST_ELEMENT(GST_MESSAGE_SRC(message)), context.get());
        return true;
    }

#if ENABLE(ENCRYPTED_MEDIA)
    if (!g_strcmp0(contextType, "drm-preferred-decryption-system-id")) {
        initializationDataEncountered(parseInitDataFromProtectionMessage(message));
        bool isCDMAttached = waitForCDMAttachment();
        if (isCDMAttached && !isPlayerShuttingDown() && !m_cdmInstance->keySystem().isEmpty()) {
            const char* preferredKeySystemUuid = GStreamerEMEUtilities::keySystemToUuid(m_cdmInstance->keySystem());
            GST_INFO_OBJECT(pipeline(), "working with key system %s, continuing with key system %s on %s", m_cdmInstance->keySystem().utf8().data(), preferredKeySystemUuid, GST_MESSAGE_SRC_NAME(message));

            GRefPtr<GstContext> context = adoptGRef(gst_context_new("drm-preferred-decryption-system-id", FALSE));
            GstStructure* contextStructure = gst_context_writable_structure(context.get());
            gst_structure_set(contextStructure, "decryption-system-id", G_TYPE_STRING, preferredKeySystemUuid, nullptr);
            gst_element_set_context(GST_ELEMENT(GST_MESSAGE_SRC(message)), context.get());
            return true;
        }

        GST_WARNING_OBJECT(pipeline(), "waiting for a CDM failed, no CDM available");
        return false;
    }
#endif // ENABLE(ENCRYPTED_MEDIA)

    GST_DEBUG_OBJECT(pipeline(), "Unhandled %s need-context message for %s", contextType, GST_MESSAGE_SRC_NAME(message));
    return false;
}

// Returns the size of the video.
FloatSize MediaPlayerPrivateGStreamer::naturalSize() const
{
#if USE(GSTREAMER_HOLEPUNCH)
    // When using the holepuch we may not be able to get the video frames size, so we can't use
    // it. But we need to report some non empty naturalSize for the player's GraphicsLayer
    // to be properly created.
    return s_holePunchDefaultFrameSize;
#endif

#if ENABLE(MEDIA_STREAM)
    if (!m_isLegacyPlaybin && !m_wantedVideoStreamId.isEmpty()) {
        RefPtr<VideoTrackPrivateGStreamer> videoTrack = m_videoTracks.get(m_wantedVideoStreamId);

        if (videoTrack) {
            auto tags = adoptGRef(gst_stream_get_tags(videoTrack->stream()));
            gint width, height;

            if (tags && gst_tag_list_get_int(tags.get(), WEBKIT_MEDIA_TRACK_TAG_WIDTH, &width) && gst_tag_list_get_int(tags.get(), WEBKIT_MEDIA_TRACK_TAG_HEIGHT, &height))
                return FloatSize(width, height);
        }
    }
#endif // ENABLE(MEDIA_STREAM)

    if (!hasVideo())
        return FloatSize();

    return m_videoSize;
}

void MediaPlayerPrivateGStreamer::configureMediaStreamAudioTracks()
{
#if ENABLE(MEDIA_STREAM)
    if (WEBKIT_IS_MEDIA_STREAM_SRC(m_source.get()))
        webkitMediaStreamSrcConfigureAudioTracks(WEBKIT_MEDIA_STREAM_SRC(m_source.get()), volume(), isMuted(), !m_isPaused);
#endif
}

void MediaPlayerPrivateGStreamer::setVolume(float volume)
{
    if (!m_volumeElement)
        return;

    GST_DEBUG_OBJECT(pipeline(), "Setting volume: %f", volume);
    gst_stream_volume_set_volume(m_volumeElement.get(), GST_STREAM_VOLUME_FORMAT_LINEAR, static_cast<double>(volume));
    configureMediaStreamAudioTracks();
}

float MediaPlayerPrivateGStreamer::volume() const
{
    if (!m_volumeElement)
        return 0;

    auto volume = gst_stream_volume_get_volume(m_volumeElement.get(), GST_STREAM_VOLUME_FORMAT_LINEAR);
    GST_DEBUG_OBJECT(pipeline(), "Volume: %f", volume);
    return volume;
}

void MediaPlayerPrivateGStreamer::notifyPlayerOfVolumeChange()
{
    if (!m_player || !m_volumeElement)
        return;

    // get_volume() can return values superior to 1.0 if the user applies software user gain via
    // third party application (GNOME volume control for instance).
    auto oldVolume = this->volume();
    auto volume = CLAMP(oldVolume, 0.0, 1.0);

    if (volume != oldVolume)
        GST_DEBUG_OBJECT(pipeline(), "Volume value (%f) was not in [0,1] range. Clamped to %f", oldVolume, volume);
    m_player->volumeChanged(volume);
}

void MediaPlayerPrivateGStreamer::volumeChangedCallback(MediaPlayerPrivateGStreamer* player)
{
    if (player->isPlayerShuttingDown())
        return;

    // This is called when m_volumeElement receives the notify::volume signal.
    GST_DEBUG_OBJECT(player->pipeline(), "Volume changed to: %f", player->volume());

    player->m_notifier->notify(MainThreadNotification::VolumeChanged, [player] {
        player->notifyPlayerOfVolumeChange();
    });
}

MediaPlayer::NetworkState MediaPlayerPrivateGStreamer::networkState() const
{
    return m_networkState;
}

MediaPlayer::ReadyState MediaPlayerPrivateGStreamer::readyState() const
{
    return m_readyState;
}

void MediaPlayerPrivateGStreamer::setMuted(bool shouldMute)
{
    if (!m_volumeElement || shouldMute == isMuted())
        return;

    GST_INFO_OBJECT(pipeline(), "Setting muted state to %s", boolForPrinting(shouldMute));
    g_object_set(m_volumeElement.get(), "mute", shouldMute, nullptr);
    configureMediaStreamAudioTracks();
}

void MediaPlayerPrivateGStreamer::notifyPlayerOfMute()
{
    if (!m_player || !m_volumeElement)
        return;

    gboolean muted;
    g_object_get(m_volumeElement.get(), "mute", &muted, nullptr);
    GST_DEBUG_OBJECT(pipeline(), "Notifying player of new mute value: %s", boolForPrinting(muted));
    m_player->muteChanged(static_cast<bool>(muted));
}

void MediaPlayerPrivateGStreamer::muteChangedCallback(MediaPlayerPrivateGStreamer* player)
{
    // This is called when m_volumeElement receives the notify::mute signal.
    player->m_notifier->notify(MainThreadNotification::MuteChanged, [player] {
        player->notifyPlayerOfMute();
    });
}

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_OBJECT(pipeline(), "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_shouldResetPipeline || !m_missingPluginCallbacks.isEmpty() || m_didErrorOccur)
            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::NetworkState::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::NetworkState::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::NetworkState::DecodeError;
            attemptNextLocation = true;
        } else if (err->domain == GST_RESOURCE_ERROR)
            error = MediaPlayer::NetworkState::NetworkError;

        if (attemptNextLocation)
            issueError = !loadNextLocation();
        if (issueError) {
            m_didErrorOccur = true;
            if (m_networkState != error) {
                m_networkState = error;
                m_player->networkStateChanged();
            }
        }
        break;
    case GST_MESSAGE_EOS: {
        // In some specific cases, an EOS GstEvent can happen right before a seek. The event is translated
        // by playbin as an EOS GstMessage and posted to the bus, waiting to be forwarded to the main thread.
        // The EOS message (now irrelevant after the seek) is received and processed right after the seek,
        // causing the termination of the media at the player private and upper levels. This can even happen
        // after the seek has completed (m_isSeeking already false).
        // The code below detects that condition by ensuring that the playback is coherent with the EOS message,
        // that is, if we're still playing somewhere inside the playable ranges, there should be no EOS at
        // all. If that's the case, it's considered to be one of those spureous EOS and is ignored.
        // Live streams (infinite duration) are special and we still have to detect legitimate EOS there, so
        // this message bailout isn't done in those cases.
        MediaTime playbackPosition = MediaTime::invalidTime();
        MediaTime duration = durationMediaTime();
        GstClockTime gstreamerPosition = gstreamerPositionFromSinks();
        if (GST_CLOCK_TIME_IS_VALID(gstreamerPosition))
            playbackPosition = MediaTime(gstreamerPosition, GST_SECOND);
        if (playbackPosition.isValid() && duration.isValid()
            && ((m_playbackRate >= 0 && playbackPosition < duration && duration.isFinite())
            || (m_playbackRate < 0 && playbackPosition > MediaTime::zeroTime()))) {
            GST_DEBUG_OBJECT(pipeline(), "EOS received but position %s is still in the finite playable limits [%s, %s], ignoring it",
                playbackPosition.toString().utf8().data(), MediaTime::zeroTime().toString().utf8().data(), duration.toString().utf8().data());
            break;
        }
        didEnd();
        break;
    }
    case GST_MESSAGE_ASYNC_DONE:
        if (!messageSourceIsPlaybin || m_isDelayingLoad)
            break;
        asyncStateChangeDone();
        break;
    case GST_MESSAGE_STATE_CHANGED: {
        if (!messageSourceIsPlaybin || m_isDelayingLoad)
            break;

        // Construct a filename for the graphviz dot file output.
        GstState newState;
        gst_message_parse_state_changed(message, &currentState, &newState, nullptr);
        CString dotFileName = makeString(GST_OBJECT_NAME(m_pipeline.get()), '.',
            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());

        GST_DEBUG_OBJECT(pipeline(), "Changed state from %s to %s", gst_element_state_get_name(currentState), gst_element_state_get_name(newState));

        if (!m_isLegacyPlaybin && currentState == GST_STATE_PAUSED && newState == GST_STATE_PLAYING)
            playbin3SendSelectStreamsIfAppropriate();
        updateStates();

        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_OBJECT(pipeline(), "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::NetworkState::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 !USE(GSTREAMER_FULL)
            if (gst_install_plugins_supported()) {
                auto missingPluginCallback = MediaPlayerRequestInstallMissingPluginsCallback::create([weakThis = WeakPtr { *this }](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.copyRef());
                GUniquePtr<char> detail(gst_missing_plugin_message_get_installer_detail(message));
                GUniquePtr<char> description(gst_missing_plugin_message_get_description(message));
                m_player->requestInstallMissingPlugins(String::fromUTF8(detail.get()), String::fromUTF8(description.get()), missingPluginCallback.get());
            }
#endif
        }
#if USE(GSTREAMER_MPEGTS)
        else if (GstMpegtsSection* section = gst_message_parse_mpegts_section(message)) {
            processMpegTsSection(section);
            gst_mpegts_section_unref(section);
        }
#endif
        else if (gst_structure_has_name(structure, "http-headers")) {
            GST_DEBUG_OBJECT(pipeline(), "Processing HTTP headers: %" GST_PTR_FORMAT, structure);
            if (const char* uri = gst_structure_get_string(structure, "uri")) {
                URL url(URL(), uri);
                m_origins.add(SecurityOrigin::create(url));

                if (url != m_url) {
                    GST_DEBUG_OBJECT(pipeline(), "Ignoring HTTP response headers for non-main URI.");
                    break;
                }
            }

            bool isRangeRequest = false;
            GUniqueOutPtr<GstStructure> requestHeaders;
            if (gst_structure_get(structure, "request-headers", GST_TYPE_STRUCTURE, &requestHeaders.outPtr(), nullptr))
                isRangeRequest = gst_structure_has_field(requestHeaders.get(), "Range");

            GST_DEBUG_OBJECT(pipeline(), "Is range request: %s", boolForPrinting(isRangeRequest));

            GUniqueOutPtr<GstStructure> responseHeaders;
            if (gst_structure_get(structure, "response-headers", GST_TYPE_STRUCTURE, &responseHeaders.outPtr(), nullptr)) {
                CString contentLengthHeaderName = httpHeaderNameString(HTTPHeaderName::ContentLength).utf8();
                uint64_t contentLength = 0;
                if (!gst_structure_get_uint64(responseHeaders.get(), contentLengthHeaderName.data(), &contentLength)) {
                    // souphttpsrc sets a string for Content-Length, so
                    // handle it here, until we remove the webkit+ protocol
                    // prefix from webkitwebsrc.
                    if (const char* contentLengthAsString = gst_structure_get_string(responseHeaders.get(), contentLengthHeaderName.data())) {
                        contentLength = g_ascii_strtoull(contentLengthAsString, nullptr, 10);
                        if (contentLength == G_MAXUINT64)
                            contentLength = 0;
                    }
                }
                if (!isRangeRequest) {
                    m_isLiveStream = !contentLength;
                    GST_INFO_OBJECT(pipeline(), "%s stream detected", m_isLiveStream ? "Live" : "Non-live");
                    updateDownloadBufferingFlag();
                }
            }
        } else if (gst_structure_has_name(structure, "webkit-network-statistics")) {
            if (gst_structure_get(structure, "read-position", G_TYPE_UINT64, &m_networkReadPosition, "size", G_TYPE_UINT64, &m_httpResponseTotalSize, nullptr))
                GST_DEBUG_OBJECT(pipeline(), "Updated network read position %" G_GUINT64_FORMAT ", size: %" G_GUINT64_FORMAT, m_networkReadPosition, m_httpResponseTotalSize);
        } else if (gst_structure_has_name(structure, "GstCacheDownloadComplete")) {
            GST_INFO_OBJECT(pipeline(), "Stream is fully downloaded, stopping monitoring downloading progress.");
            m_fillTimer.stop();
            m_bufferingPercentage = 100;
            updateStates();
        } else if (gst_structure_has_name(structure, "webkit-web-src-has-eos")) {
            GST_DEBUG_OBJECT(pipeline(), "WebKitWebSrc has EOS");
            m_hasWebKitWebSrcSentEOS = true;
        } else
            GST_DEBUG_OBJECT(pipeline(), "Unhandled element message: %" GST_PTR_FORMAT, structure);
        break;
    case GST_MESSAGE_TOC:
        processTableOfContents(message);
        break;
    case GST_MESSAGE_STREAMS_SELECTED: {
        if (m_isLegacyPlaybin)
            break;

#ifndef GST_DISABLE_DEBUG
        GST_DEBUG_OBJECT(m_pipeline.get(), "Received STREAMS_SELECTED message selecting the following streams:");
        unsigned numStreams = gst_message_streams_selected_get_size(message);
        for (unsigned i = 0; i < numStreams; i++) {
            GstStream* stream = gst_message_streams_selected_get_stream(message, i);
            GST_DEBUG_OBJECT(pipeline(), "#%u %s %s", i, gst_stream_type_get_name(gst_stream_get_stream_type(stream)), gst_stream_get_stream_id(stream));
        }
#endif
        GST_DEBUG_OBJECT(m_pipeline.get(), "Setting m_waitingForStreamsSelectedEvent to false.");
        m_waitingForStreamsSelectedEvent = false;

        // Unfortunately, STREAMS_SELECTED messages from playbin3 are highly unreliable, often only including the audio
        // stream or only the video stream when both are present and going to be played.
        // Therefore, instead of reading the event data, we will just assume our previously requested selection was honored.
        m_currentAudioStreamId = m_requestedAudioStreamId;
        m_currentVideoStreamId = m_requestedVideoStreamId;

        // It's possible the user made a track switch before the initial STREAMS_SELECED. Now it's a good moment to
        // request it being attended. Note that it's not possible to send a SELECT_STREAMS before the first
        // STREAMS_SELECTED message because at that point the pipeline is not compeletely constructed.
        playbin3SendSelectStreamsIfAppropriate();
        break;
    }
    default:
        GST_DEBUG_OBJECT(pipeline(), "Unhandled GStreamer message type: %s", GST_MESSAGE_TYPE_NAME(message));
        break;
    }
}

void MediaPlayerPrivateGStreamer::processBufferingStats(GstMessage* message)
{
    GstBufferingMode mode;
    gst_message_parse_buffering_stats(message, &mode, nullptr, nullptr, nullptr);

    int percentage;
    gst_message_parse_buffering(message, &percentage);

    updateBufferingStatus(mode, percentage);
}

void MediaPlayerPrivateGStreamer::updateMaxTimeLoaded(double percentage)
{
    MediaTime mediaDuration = durationMediaTime();
    if (!mediaDuration)
        return;

    m_maxTimeLoaded = MediaTime(percentage * static_cast<double>(toGstUnsigned64Time(mediaDuration)) / 100, GST_SECOND);
    GST_DEBUG_OBJECT(pipeline(), "[Buffering] Updated maxTimeLoaded: %s", toString(m_maxTimeLoaded).utf8().data());
}

void MediaPlayerPrivateGStreamer::updateBufferingStatus(GstBufferingMode mode, double percentage)
{
    bool wasBuffering = m_isBuffering;

#ifndef GST_DISABLE_GST_DEBUG
    GUniquePtr<char> modeString(g_enum_to_string(GST_TYPE_BUFFERING_MODE, mode));
    GST_DEBUG_OBJECT(pipeline(), "[Buffering] mode: %s, status: %f%%", modeString.get(), percentage);
#endif

    m_didDownloadFinish = percentage == 100;
    m_isBuffering = !m_didDownloadFinish;

    if (!m_didDownloadFinish)
        m_isBuffering = true;
    else
        m_fillTimer.stop();

    m_bufferingPercentage = percentage;
    switch (mode) {
    case GST_BUFFERING_STREAM: {
        updateMaxTimeLoaded(percentage);

        m_bufferingPercentage = percentage;
        if (m_didDownloadFinish || (!wasBuffering && m_isBuffering))
            updateStates();

        break;
    }
    case GST_BUFFERING_DOWNLOAD: {
        updateMaxTimeLoaded(percentage);
        updateStates();
        break;
    }
    default:
#ifndef GST_DISABLE_GST_DEBUG
        GST_DEBUG_OBJECT(pipeline(), "Unhandled buffering mode: %s", modeString.get());
#endif
        break;
    }
}

#if 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 (unsigned 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) {
                AtomString pid = String::number(stream->pid);
                auto track = InbandMetadataTextTrackPrivateGStreamer::create(
                    InbandTextTrackPrivate::Kind::Metadata, InbandTextTrackPrivate::CueFormat::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.
                StringBuilder inbandMetadataTrackDispatchType;
                inbandMetadataTrackDispatchType.append(hex(stream->stream_type, 2));
                for (unsigned j = 0; j < stream->descriptors->len; ++j) {
                    const GstMpegtsDescriptor* descriptor = static_cast<const GstMpegtsDescriptor*>(g_ptr_array_index(stream->descriptors, j));
                    for (unsigned k = 0; k < descriptor->length; ++k)
                        inbandMetadataTrackDispatchType.append(hex(descriptor->data[k], 2));
                }
                track->setInBandMetadataTrackDispatchType(inbandMetadataTrackDispatchType.toString());

                m_metadataTracks.add(pid, track);
                m_player->addTextTrack(*track);
            }
        }
    } else {
        AtomString 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

void MediaPlayerPrivateGStreamer::processTableOfContents(GstMessage* message)
{
    if (m_chaptersTrack)
        m_player->removeTextTrack(*m_chaptersTrack);

    m_chaptersTrack = InbandMetadataTextTrackPrivateGStreamer::create(InbandTextTrackPrivate::Kind::Chapters, InbandTextTrackPrivate::CueFormat::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 = InbandGenericCue::create();

    gint64 start = -1, stop = -1;
    gst_toc_entry_get_start_stop_times(entry, &start, &stop);

    uint32_t truncatedGstSecond = static_cast<uint32_t>(GST_SECOND);
    if (start != -1)
        cue->setStartTime(MediaTime(static_cast<int64_t>(start), truncatedGstSecond));
    if (stop != -1)
        cue->setEndTime(MediaTime(static_cast<int64_t>(stop), truncatedGstSecond));

    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::configureDownloadBuffer(GstElement* element)
{
    GUniquePtr<char> elementName(gst_element_get_name(element));
    RELEASE_ASSERT(g_str_has_prefix(elementName.get(), "downloadbuffer"));

    m_downloadBuffer = element;
    g_signal_connect_swapped(element, "notify::temp-location", G_CALLBACK(downloadBufferFileCreatedCallback), this);

    // Set the GstDownloadBuffer size to our preferred value controls the thresholds for buffering events.
    g_object_set(element, "max-size-bytes", 100 * KB, nullptr);

    GUniqueOutPtr<char> oldDownloadTemplate;
    g_object_get(element, "temp-template", &oldDownloadTemplate.outPtr(), nullptr);

#if PLATFORM(WPE)
    GUniquePtr<char> mediaDiskCachePath(g_strdup(std::getenv("WPE_SHELL_MEDIA_DISK_CACHE_PATH")));
    if (!mediaDiskCachePath || !*mediaDiskCachePath) {
        GUniquePtr<char> defaultValue(g_build_filename(G_DIR_SEPARATOR_S, "var", "tmp", nullptr));
        mediaDiskCachePath.swap(defaultValue);
    }
#else
    GUniquePtr<char> mediaDiskCachePath(g_build_filename(G_DIR_SEPARATOR_S, "var", "tmp", nullptr));
#endif

    GUniquePtr<char> newDownloadTemplate(g_build_filename(G_DIR_SEPARATOR_S, mediaDiskCachePath.get(), "WebKit-Media-XXXXXX", nullptr));
    g_object_set(element, "temp-template", newDownloadTemplate.get(), nullptr);
    GST_DEBUG_OBJECT(pipeline(), "Reconfigured file download template from '%s' to '%s'", oldDownloadTemplate.get(), newDownloadTemplate.get());

    String newDownloadPrefixPath = newDownloadTemplate.get();
    purgeOldDownloadFiles(newDownloadPrefixPath.replace("XXXXXX", ""));
}

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);

    if (UNLIKELY(!FileSystem::deleteFile(downloadFile.get()))) {
        GST_WARNING("Couldn't unlink media temporary file %s after creation", downloadFile.get());
        return;
    }

    GST_DEBUG_OBJECT(player->pipeline(), "Unlinked media temporary file %s after creation", downloadFile.get());
}

void MediaPlayerPrivateGStreamer::purgeOldDownloadFiles(const String& downloadFilePrefixPath)
{
    if (downloadFilePrefixPath.isEmpty())
        return;

    auto templateDirectory = FileSystem::parentPath(downloadFilePrefixPath);
    auto templatePrefix = FileSystem::pathFileName(downloadFilePrefixPath);
    for (auto& fileName : FileSystem::listDirectory(templateDirectory)) {
        if (!fileName.startsWith(templatePrefix))
            continue;

        auto filePath = FileSystem::pathByAppendingComponent(templateDirectory, fileName);
        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::asyncStateChangeDone()
{
    if (!m_pipeline || m_didErrorOccur)
        return;

    if (m_isSeeking) {
        if (m_isSeekPending)
            updateStates();
        else {
            GST_DEBUG_OBJECT(pipeline(), "[Seek] seeked to %s", toString(m_seekTime).utf8().data());
            m_isSeeking = false;
            invalidateCachedPosition();
            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 || m_didErrorOccur)
        return;

    MediaPlayer::NetworkState oldNetworkState = m_networkState;
    MediaPlayer::ReadyState oldReadyState = m_readyState;
    bool oldIsPaused = m_isPaused;
    GstState pending, 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_OBJECT(pipeline(), "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_shouldResetPipeline = m_currentState <= GST_STATE_READY;

        bool didBuffering = m_isBuffering;

        // Update ready and network states.
        switch (m_currentState) {
        case GST_STATE_NULL:
            m_readyState = MediaPlayer::ReadyState::HaveNothing;
            m_networkState = MediaPlayer::NetworkState::Empty;
            break;
        case GST_STATE_READY:
            m_readyState = MediaPlayer::ReadyState::HaveMetadata;
            m_networkState = MediaPlayer::NetworkState::Empty;
            break;
        case GST_STATE_PAUSED:
            FALLTHROUGH;
        case GST_STATE_PLAYING:
            if (m_isBuffering) {
                GRefPtr<GstQuery> query = adoptGRef(gst_query_new_buffering(GST_FORMAT_PERCENT));

                m_isBuffering = m_bufferingPercentage == 100;
                if (gst_element_query(m_pipeline.get(), query.get())) {
                    gboolean isBuffering = m_isBuffering;
                    gst_query_parse_buffering_percent(query.get(), &isBuffering, nullptr);
                    m_isBuffering = isBuffering;
                }

                if (!m_isBuffering) {
                    GST_INFO_OBJECT(pipeline(), "[Buffering] Complete.");
                    m_readyState = MediaPlayer::ReadyState::HaveEnoughData;
                    m_networkState = m_didDownloadFinish ? MediaPlayer::NetworkState::Idle : MediaPlayer::NetworkState::Loading;
                } else {
                    m_readyState = MediaPlayer::ReadyState::HaveCurrentData;
                    m_networkState = MediaPlayer::NetworkState::Loading;
                }
            } else if (m_didDownloadFinish) {
                m_readyState = MediaPlayer::ReadyState::HaveEnoughData;
                m_networkState = MediaPlayer::NetworkState::Loaded;
            } else {
                m_readyState = MediaPlayer::ReadyState::HaveFutureData;
                m_networkState = MediaPlayer::NetworkState::Loading;
            }

            break;
        default:
            ASSERT_NOT_REACHED();
            break;
        }

        // Sync states where needed.
        if (m_currentState == GST_STATE_PAUSED) {
            if (!m_areVolumeAndMuteInitialized) {
                notifyPlayerOfVolumeChange();
                notifyPlayerOfMute();
                m_areVolumeAndMuteInitialized = true;
            }

            if (didBuffering && !m_isBuffering && !m_isPaused && m_playbackRate) {
                GST_INFO_OBJECT(pipeline(), "[Buffering] Restarting playback.");
                changePipelineState(GST_STATE_PLAYING);
            }
        } else if (m_currentState == GST_STATE_PLAYING) {
            m_isPaused = false;

            if ((m_isBuffering && !m_isLiveStream) || !m_playbackRate) {
                GST_INFO_OBJECT(pipeline(), "[Buffering] Pausing stream for buffering.");
                changePipelineState(GST_STATE_PAUSED);
            }
        } else
            m_isPaused = true;

        GST_DEBUG_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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_isLiveStream = true;
        updateDownloadBufferingFlag();

        if (m_currentState == GST_STATE_READY)
            m_readyState = MediaPlayer::ReadyState::HaveNothing;
        else if (m_currentState == GST_STATE_PAUSED)
            m_isPaused = true;
        else if (m_currentState == GST_STATE_PLAYING)
            m_isPaused = false;

        if (!m_isPaused && m_playbackRate)
            changePipelineState(GST_STATE_PLAYING);

        m_networkState = MediaPlayer::NetworkState::Loading;
        break;
    default:
        GST_DEBUG_OBJECT(pipeline(), "Else : %d", getStateResult);
        break;
    }

    m_requestedState = GST_STATE_VOID_PENDING;

    if (shouldUpdatePlaybackState)
        m_player->playbackStateChanged();

    if (m_networkState != oldNetworkState) {
        GST_DEBUG_OBJECT(pipeline(), "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_OBJECT(pipeline(), "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_isSeekPending) {
            GST_DEBUG_OBJECT(pipeline(), "[Seek] committing pending seek to %s", toString(m_seekTime).utf8().data());
            m_isSeekPending = false;
            m_isSeeking = doSeek(m_seekTime, m_player->rate(), static_cast<GstSeekFlags>(GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_ACCURATE));
            if (!m_isSeeking) {
                invalidateCachedPosition();
                GST_DEBUG_OBJECT(pipeline(), "[Seek] seeking to %s failed", toString(m_seekTime).utf8().data());
            }
        }
    }

    if (oldIsPaused != m_isPaused)
        configureMediaStreamAudioTracks();
}

void MediaPlayerPrivateGStreamer::mediaLocationChanged(GstMessage* message)
{
    const GstStructure* structure = gst_message_get_structure(message);
    if (!structure)
        return;

    // This structure can contain:
    // - both a new-location string and embedded locations structure
    // - or only a new-location string.
    m_mediaLocations.reset(gst_structure_copy(structure));
    const GValue* locations = gst_structure_get_value(m_mediaLocations.get(), "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.get(), "locations");
    const char* newLocation = nullptr;

    if (!locations) {
        // Fallback on new-location string.
        newLocation = gst_structure_get_string(m_mediaLocations.get(), "new-location");
        if (!newLocation)
            return false;
    }

    if (!newLocation) {
        if (m_mediaLocationCurrentIndex < 0) {
            m_mediaLocations.reset();
            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);

        GUniqueOutPtr<gchar> playbinUrlStr;
        g_object_get(m_pipeline.get(), "current-uri", &playbinUrlStr.outPtr(), nullptr);
        URL playbinUrl(URL(), playbinUrlStr.get());

        if (playbinUrl == newUrl) {
            GST_DEBUG_OBJECT(pipeline(), "Playbin already handled redirection.");

            m_url = playbinUrl;

            return true;
        }

        changePipelineState(GST_STATE_READY);
        auto securityOrigin = SecurityOrigin::create(m_url);
        if (securityOrigin->canRequest(newUrl)) {
            GST_INFO_OBJECT(pipeline(), "New media url: %s", newUrl.string().utf8().data());

            // Reset player states.
            m_networkState = MediaPlayer::NetworkState::Loading;
            m_player->networkStateChanged();
            m_readyState = MediaPlayer::ReadyState::HaveNothing;
            m_player->readyStateChanged();

            // Reset pipeline state.
            m_shouldResetPipeline = true;

            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_OBJECT(pipeline(), "Not allowed to load new media location: %s", newUrl.string().utf8().data());
    }
    m_mediaLocationCurrentIndex--;
    return false;
}

void MediaPlayerPrivateGStreamer::didEnd()
{
    invalidateCachedPosition();
    MediaTime now = currentMediaTime();
    GST_INFO_OBJECT(pipeline(), "Playback ended, currentMediaTime = %s, duration = %s", now.toString().utf8().data(), durationMediaTime().toString().utf8().data());
    m_isEndReached = true;

    if (!durationMediaTime().isFinite()) {
        // From the HTMLMediaElement spec.
        // If an "infinite" stream ends for some reason, then the duration would change from positive Infinity to the
        // time of the last frame or sample in the stream, and the durationchange event would be fired.
        GST_DEBUG_OBJECT(pipeline(), "HTMLMediaElement duration previously infinite or unknown (e.g. live stream), setting it to current position.");
        m_cachedDuration = now;
        m_player->durationChanged();
    }

    // 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.
    if (!m_isSeeking) {
        m_cachedPosition = m_playbackRate > 0 ? durationMediaTime() : MediaTime::zeroTime();
        GST_DEBUG("Position adjusted: %s", currentMediaTime().toString().utf8().data());
    }

    // Now that playback has ended it's NOT a safe time to send a SELECT_STREAMS event. In fact, as of GStreamer 1.16,
    // playbin3 will crash on a GStreamer assertion (combine->sinkpad being unexpectedly null) if we try. Instead, wait
    // until we get the initial STREAMS_SELECTED message one more time.
    m_waitingForStreamsSelectedEvent = true;

    if (!m_player->isLooping() && !isMediaSource()) {
        m_isPaused = true;
        changePipelineState(GST_STATE_READY);
        m_didDownloadFinish = false;
        configureMediaStreamAudioTracks();

#if USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)
        wpe_video_plane_display_dmabuf_source_end_of_stream(m_wpeVideoPlaneDisplayDmaBuf.get());
#endif
    }
    timeChanged();
}

void MediaPlayerPrivateGStreamer::getSupportedTypes(HashSet<String, ASCIICaseInsensitiveHash>& types)
{
    GStreamerRegistryScanner::getSupportedDecodingTypes(types);
}

MediaPlayer::SupportsType MediaPlayerPrivateGStreamer::supportsType(const MediaEngineSupportParameters& parameters)
{
    MediaPlayer::SupportsType result = MediaPlayer::SupportsType::IsNotSupported;
#if ENABLE(MEDIA_SOURCE)
    // MediaPlayerPrivateGStreamerMSE is in charge of mediasource playback, not us.
    if (parameters.isMediaSource)
        return result;
#endif

    if (parameters.isMediaStream) {
#if ENABLE(MEDIA_STREAM)
        return MediaPlayer::SupportsType::IsSupported;
#else
        return result;
#endif
    }

    GST_DEBUG("Checking mime-type \"%s\"", parameters.type.raw().utf8().data());
    if (parameters.type.isEmpty())
        return result;

    // This player doesn't support pictures rendering.
    if (parameters.type.raw().startsWith("image"_s))
        return result;

    auto& gstRegistryScanner = GStreamerRegistryScanner::singleton();
    result = gstRegistryScanner.isContentTypeSupported(GStreamerRegistryScanner::Configuration::Decoding, parameters.type, parameters.contentTypesRequiringHardwareSupport);

    auto finalResult = extendedSupportsType(parameters, result);
    GST_DEBUG("Supported: %s", convertEnumerationToString(finalResult).utf8().data());
    return finalResult;
}

void MediaPlayerPrivateGStreamer::updateDownloadBufferingFlag()
{
    if (!m_pipeline)
        return;

    unsigned flags;
    g_object_get(m_pipeline.get(), "flags", &flags, nullptr);

    unsigned flagDownload = getGstPlayFlag("download");

    if (m_url.protocolIsBlob()) {
        GST_DEBUG_OBJECT(pipeline(), "Blob URI detected. Disabling on-disk buffering");
        g_object_set(m_pipeline.get(), "flags", flags & ~flagDownload, nullptr);
        m_fillTimer.stop();
        return;
    }

    // We don't want to stop downloading if we already started it.
    if (flags & flagDownload && m_readyState > MediaPlayer::ReadyState::HaveNothing && !m_shouldResetPipeline) {
        GST_DEBUG_OBJECT(pipeline(), "Download already started, not starting again");
        return;
    }

    bool shouldDownload = !m_isLiveStream && m_preload == MediaPlayer::Preload::Auto;
    if (shouldDownload) {
        GST_INFO_OBJECT(pipeline(), "Enabling on-disk buffering");
        g_object_set(m_pipeline.get(), "flags", flags | flagDownload, nullptr);
        m_fillTimer.startRepeating(200_ms);
    } else {
        GST_INFO_OBJECT(pipeline(), "Disabling on-disk buffering");
        g_object_set(m_pipeline.get(), "flags", flags & ~flagDownload, nullptr);
        m_fillTimer.stop();
    }
}

static void setPlaybackFlags(GstElement* pipeline)
{
    unsigned hasAudio = getGstPlayFlag("audio");
    unsigned hasVideo = getGstPlayFlag("video");
    unsigned hasText = getGstPlayFlag("text");
    unsigned hasNativeVideo = getGstPlayFlag("native-video");
    unsigned hasNativeAudio = getGstPlayFlag("native-audio");
    unsigned hasSoftwareColorBalance = getGstPlayFlag("soft-colorbalance");

    unsigned flags = 0;
    g_object_get(pipeline, "flags", &flags, nullptr);
    GST_TRACE_OBJECT(pipeline, "default flags %x", flags);
    flags = flags & ~hasText;
    flags = flags & ~hasNativeAudio;
    flags = flags & ~hasNativeVideo;
    flags = flags & ~hasSoftwareColorBalance;

#if !USE(GSTREAMER_TEXT_SINK)
    hasText = 0x0;
#endif

#if USE(GSTREAMER_NATIVE_VIDEO)
    hasSoftwareColorBalance = 0x0;
#else
    hasNativeVideo = 0x0;
#endif

#if !USE(GSTREAMER_NATIVE_AUDIO)
    hasNativeAudio = 0x0;
#endif

    GST_INFO_OBJECT(pipeline, "text %s, audio %s (native %s), video %s (native %s, software color balance %s)", boolForPrinting(hasText),
        boolForPrinting(hasAudio), boolForPrinting(hasNativeAudio), boolForPrinting(hasVideo), boolForPrinting(hasNativeVideo),
        boolForPrinting(hasSoftwareColorBalance));
    flags |= hasText | hasAudio | hasVideo | hasNativeVideo | hasNativeAudio | hasSoftwareColorBalance;
    g_object_set(pipeline, "flags", flags, nullptr);
    GST_DEBUG_OBJECT(pipeline, "current pipeline flags %x", flags);
}

void MediaPlayerPrivateGStreamer::createGSTPlayBin(const URL& url)
{
    GST_INFO("Creating pipeline for %s player", m_player->isVideoPlayer() ? "video" : "audio");
    const char* playbinName = "playbin";

    // MSE and Mediastream require playbin3. Regular playback can use playbin3 on-demand with the
    // WEBKIT_GST_USE_PLAYBIN3 environment variable.
    const char* usePlaybin3 = g_getenv("WEBKIT_GST_USE_PLAYBIN3");
    if ((isMediaSource() || url.protocolIs("mediastream") || (usePlaybin3 && equal(usePlaybin3, "1"))))
        playbinName = "playbin3";

    ASSERT(!m_pipeline);

    auto elementId = m_player->elementId();
    if (elementId.isEmpty())
        elementId = "media-player";

    const char* type = isMediaSource() ? "MSE-" : url.protocolIs("mediastream") ? "mediastream-" : "";

    m_isLegacyPlaybin = !g_strcmp0(playbinName, "playbin");

    static Atomic<uint32_t> pipelineId;

    m_pipeline = makeGStreamerElement(playbinName, makeString(type, elementId, '-', pipelineId.exchangeAdd(1)).ascii().data());
    if (!m_pipeline) {
        GST_WARNING("%s not found, make sure to install gst-plugins-base", playbinName);
        loadingFailed(MediaPlayer::NetworkState::FormatError, MediaPlayer::ReadyState::HaveNothing, true);
        return;
    }

    setStreamVolumeElement(GST_STREAM_VOLUME(m_pipeline.get()));

    GST_INFO_OBJECT(pipeline(), "Using legacy playbin element: %s", boolForPrinting(m_isLegacyPlaybin));

    setPlaybackFlags(pipeline());

    // Let also other listeners subscribe to (application) messages in this bus.
    auto bus = adoptGRef(gst_pipeline_get_bus(GST_PIPELINE(m_pipeline.get())));
    gst_bus_enable_sync_message_emission(bus.get());
    gst_bus_add_signal_watch_full(bus.get(), RunLoopSourcePriority::RunLoopDispatcher);
    g_signal_connect_swapped(bus.get(), "message", G_CALLBACK(+[](MediaPlayerPrivateGStreamer* player, GstMessage* message) {
        player->handleMessage(message);
    }), this);
    g_signal_connect_swapped(bus.get(), "sync-message::need-context", G_CALLBACK(+[](MediaPlayerPrivateGStreamer* player, GstMessage* message) {
        player->handleNeedContextMessage(message);
    }), this);

    // In the MSE case stream collection messages are emitted from the main thread right before the
    // initilization segment is parsed and "updateend" is fired. We need therefore to handle these
    // synchronously in the same main thread tick to make the tracks information available to JS no
    // later than "updateend". There is no such limitation otherwise (if playbin3 is enabled or in
    // MediaStream cases).
    auto streamCollectionSignalName = makeString(isMediaSource() ? "sync-" : "", "message::stream-collection");
    g_signal_connect_swapped(bus.get(), streamCollectionSignalName.ascii().data(), G_CALLBACK(+[](MediaPlayerPrivateGStreamer* player, GstMessage* message) {
        player->handleStreamCollectionMessage(message);
    }), this);

    g_object_set(m_pipeline.get(), "mute", m_player->muted(), nullptr);

    g_signal_connect(GST_BIN_CAST(m_pipeline.get()), "deep-element-added", G_CALLBACK(+[](GstBin*, GstBin* subBin, GstElement* element, MediaPlayerPrivateGStreamer* player) {
        GUniquePtr<char> binName(gst_element_get_name(GST_ELEMENT_CAST(subBin)));
        GUniquePtr<char> elementName(gst_element_get_name(element));

        if (g_str_has_prefix(elementName.get(), "downloadbuffer")) {
            player->configureDownloadBuffer(element);
            return;
        }

        if (g_str_has_prefix(elementName.get(), "uridecodebin")) {
            // This will set the multiqueue size to the default value.
            g_object_set(element, "buffer-size", 2 * MB, nullptr);
            return;
        }

        if (!g_str_has_prefix(binName.get(), "decodebin"))
            return;

        if (g_str_has_prefix(elementName.get(), "v4l2"))
            player->m_videoDecoderPlatform = GstVideoDecoderPlatform::Video4Linux;
        else if (g_str_has_prefix(elementName.get(), "imxvpudec"))
            player->m_videoDecoderPlatform = GstVideoDecoderPlatform::ImxVPU;
        else if (g_str_has_prefix(elementName.get(), "omx"))
            player->m_videoDecoderPlatform = GstVideoDecoderPlatform::OpenMAX;

#if USE(TEXTURE_MAPPER_GL)
        player->updateTextureMapperFlags();
#endif
    }), this);

    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 (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_textSink = webkitTextSinkNew(*this);
    ASSERT(m_textSink);

    g_object_set(m_pipeline.get(), "text-sink", m_textSink.get(), nullptr);

    if (!m_audioSink)
        m_audioSink = createAudioSink();

    g_object_set(m_pipeline.get(), "audio-sink", m_audioSink.get(), "video-sink", createVideoSink(), nullptr);

    if (m_shouldPreservePitch) {
        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_player->isVideoPlayer())
        return;

    GRefPtr<GstPad> videoSinkPad = adoptGRef(gst_element_get_static_pad(m_videoSink.get(), "sink"));
    if (videoSinkPad)
        g_signal_connect(videoSinkPad.get(), "notify::caps", G_CALLBACK(+[](GstPad* videoSinkPad, GParamSpec*, MediaPlayerPrivateGStreamer* player) {
            player->videoSinkCapsChanged(videoSinkPad);
        }), this);
}

bool MediaPlayerPrivateGStreamer::didPassCORSAccessCheck() const
{
    if (WEBKIT_IS_WEB_SRC(m_source.get()))
        return webKitSrcPassedCORSAccessCheck(WEBKIT_WEB_SRC_CAST(m_source.get()));
    return false;
}

bool MediaPlayerPrivateGStreamer::canSaveMediaData() const
{
    if (m_isLiveStream)
        return false;

    if (m_url.isLocalFile())
        return true;

    if (m_url.protocolIsInHTTPFamily())
        return true;

    return false;
}

void MediaPlayerPrivateGStreamer::readyTimerFired()
{
    GST_DEBUG_OBJECT(pipeline(), "In READY for too long. Releasing pipeline resources.");
    changePipelineState(GST_STATE_NULL);
}

void MediaPlayerPrivateGStreamer::acceleratedRenderingStateChanged()
{
    m_canRenderingBeAccelerated = m_player && m_player->acceleratedCompositingEnabled();
}

bool MediaPlayerPrivateGStreamer::performTaskAtMediaTime(Function<void()>&& task, const MediaTime& time)
{
    ASSERT(isMainThread());

    // Ignore the cases when the time isn't marching on or the position is unknown.
    MediaTime currentTime = playbackPosition();
    if (!m_pipeline || m_didErrorOccur || m_isSeeking || m_isPaused || !m_playbackRate || !currentTime.isValid())
        return false;

    std::optional<Function<void()>> taskToSchedule;
    {
        DataMutexLocker taskAtMediaTimeScheduler { m_TaskAtMediaTimeSchedulerDataMutex };
        taskAtMediaTimeScheduler->setTask(WTFMove(task), time,
            m_playbackRate >= 0 ? TaskAtMediaTimeScheduler::Forward : TaskAtMediaTimeScheduler::Backward);
        taskToSchedule = taskAtMediaTimeScheduler->checkTaskForScheduling(currentTime);
    }

    // Dispatch the task if the time is already reached. Dispatching instead of directly running the
    // task prevents infinite recursion in case the task calls performTaskAtMediaTime() internally.
    if (taskToSchedule)
        RunLoop::main().dispatch(WTFMove(taskToSchedule.value()));

    return true;
}

#if USE(TEXTURE_MAPPER_GL)
PlatformLayer* MediaPlayerPrivateGStreamer::platformLayer() const
{
#if USE(NICOSIA)
    return m_nicosiaLayer.ptr();
#else
    return const_cast<MediaPlayerPrivateGStreamer*>(this);
#endif
}

#if USE(NICOSIA)
void MediaPlayerPrivateGStreamer::swapBuffersIfNeeded()
{
#if USE(GSTREAMER_HOLEPUNCH)
    pushNextHolePunchBuffer();
#endif
}
#else
RefPtr<TextureMapperPlatformLayerProxy> MediaPlayerPrivateGStreamer::proxy() const
{
    return m_platformLayerProxy.copyRef();
}

void MediaPlayerPrivateGStreamer::swapBuffersIfNeeded()
{
#if USE(GSTREAMER_HOLEPUNCH)
    pushNextHolePunchBuffer();
#endif
}
#endif

#if USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)
class GStreamerDMABufHolePunchClient : public TextureMapperPlatformLayerBuffer::HolePunchClient {
public:
    GStreamerDMABufHolePunchClient(std::unique_ptr<GstVideoFrameHolder>&& frameHolder, struct wpe_video_plane_display_dmabuf_source* videoPlaneDisplayDmaBufSource)
        : m_frameHolder(WTFMove(frameHolder))
        , m_wpeVideoPlaneDisplayDmaBuf(videoPlaneDisplayDmaBufSource) { };
    void setVideoRectangle(const IntRect& rect) final
    {
        if (m_wpeVideoPlaneDisplayDmaBuf)
            m_frameHolder->handoffVideoDmaBuf(m_wpeVideoPlaneDisplayDmaBuf, rect);
    }
private:
    std::unique_ptr<GstVideoFrameHolder> m_frameHolder;
    struct wpe_video_plane_display_dmabuf_source* m_wpeVideoPlaneDisplayDmaBuf;
};
#endif // USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)

void MediaPlayerPrivateGStreamer::pushTextureToCompositor()
{
    Locker sampleLocker { m_sampleMutex };
    if (!GST_IS_SAMPLE(m_sample.get()))
        return;

    auto internalCompositingOperation = [this](TextureMapperPlatformLayerProxy& proxy, std::unique_ptr<GstVideoFrameHolder>&& frameHolder) {
        std::unique_ptr<TextureMapperPlatformLayerBuffer> layerBuffer;
        if (frameHolder->hasMappedTextures()) {
            layerBuffer = frameHolder->platformLayerBuffer();
            if (!layerBuffer)
                return;
            layerBuffer->setUnmanagedBufferDataHolder(WTFMove(frameHolder));
        } else {
            layerBuffer = proxy.getAvailableBuffer(frameHolder->size(), GL_DONT_CARE);
            if (UNLIKELY(!layerBuffer)) {
                auto texture = BitmapTextureGL::create(TextureMapperContextAttributes::get());
                texture->reset(frameHolder->size(), frameHolder->hasAlphaChannel() ? BitmapTexture::SupportsAlpha : BitmapTexture::NoFlag);
                layerBuffer = makeUnique<TextureMapperPlatformLayerBuffer>(WTFMove(texture));
            }
            frameHolder->updateTexture(layerBuffer->textureGL());
            layerBuffer->setExtraFlags(m_textureMapperFlags | (frameHolder->hasAlphaChannel() ? TextureMapperGL::ShouldBlend | TextureMapperGL::ShouldPremultiply : 0));
        }
        proxy.pushNextBuffer(WTFMove(layerBuffer));
    };

#if USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)
    auto proxyOperation =
        [this, internalCompositingOperation](TextureMapperPlatformLayerProxy& proxy)
        {
            Locker locker { proxy.lock() };

            if (!proxy.isActive())
                return;

            auto frameHolder = makeUnique<GstVideoFrameHolder>(m_sample.get(), m_videoDecoderPlatform, m_textureMapperFlags, !m_isUsingFallbackVideoSink);
            if (frameHolder->hasDMABuf()) {
                auto layerBuffer = makeUnique<TextureMapperPlatformLayerBuffer>(0, m_size, TextureMapperGL::ShouldNotBlend, GL_DONT_CARE);
                auto holePunchClient = makeUnique<GStreamerDMABufHolePunchClient>(WTFMove(frameHolder), m_wpeVideoPlaneDisplayDmaBuf.get());
                layerBuffer->setHolePunchClient(WTFMove(holePunchClient));
                proxy.pushNextBuffer(WTFMove(layerBuffer));
            } else
                internalCompositingOperation(proxy, WTFMove(frameHolder));
        };
#else
    auto proxyOperation =
        [this, internalCompositingOperation](TextureMapperPlatformLayerProxy& proxy)
        {
            Locker locker { proxy.lock() };

            if (!proxy.isActive())
                return;

            auto frameHolder = makeUnique<GstVideoFrameHolder>(m_sample.get(), m_videoDecoderPlatform, m_textureMapperFlags, !m_isUsingFallbackVideoSink);
            internalCompositingOperation(proxy, WTFMove(frameHolder));
        };
#endif // USE(WPE_VIDEO_PLANE_DISPLAY_DMABUF)

#if USE(NICOSIA)
    proxyOperation(downcast<Nicosia::ContentLayerTextureMapperImpl>(m_nicosiaLayer->impl()).proxy());
#else
    proxyOperation(*m_platformLayerProxy);
#endif
}
#endif // USE(TEXTURE_MAPPER_GL)

void MediaPlayerPrivateGStreamer::repaint()
{
    ASSERT(m_sample);
    ASSERT(isMainThread());

    m_player->repaint();

    Locker locker { m_drawLock };
    m_drawCondition.notifyOne();
}

static ImageOrientation getVideoOrientation(const GstTagList* tagList)
{
    ASSERT(tagList);
    GUniqueOutPtr<gchar> tag;
    if (!gst_tag_list_get_string(tagList, GST_TAG_IMAGE_ORIENTATION, &tag.outPtr())) {
        GST_DEBUG("No image_orientation tag, applying no rotation.");
        return ImageOrientation::None;
    }

    GST_DEBUG("Found image_orientation tag: %s", tag.get());
    if (!g_strcmp0(tag.get(), "flip-rotate-0"))
        return ImageOrientation::OriginTopRight;
    if (!g_strcmp0(tag.get(), "rotate-180"))
        return ImageOrientation::OriginBottomRight;
    if (!g_strcmp0(tag.get(), "flip-rotate-180"))
        return ImageOrientation::OriginBottomLeft;
    if (!g_strcmp0(tag.get(), "flip-rotate-270"))
        return ImageOrientation::OriginLeftTop;
    if (!g_strcmp0(tag.get(), "rotate-90"))
        return ImageOrientation::OriginRightTop;
    if (!g_strcmp0(tag.get(), "flip-rotate-90"))
        return ImageOrientation::OriginRightBottom;
    if (!g_strcmp0(tag.get(), "rotate-270"))
        return ImageOrientation::OriginLeftBottom;

    // Default rotation.
    return ImageOrientation::None;
}

void MediaPlayerPrivateGStreamer::updateVideoOrientation(const GstTagList* tagList)
{
    GST_DEBUG_OBJECT(pipeline(), "Updating orientation from %" GST_PTR_FORMAT, tagList);
    setVideoSourceOrientation(getVideoOrientation(tagList));

    // If the video is tagged as rotated 90 or 270 degrees, swap width and height.
    if (m_videoSourceOrientation.usesWidthAsHeight())
        m_videoSize = m_videoSize.transposedSize();

    callOnMainThreadAndWait([this] {
        m_player->sizeChanged();
    });
}

void MediaPlayerPrivateGStreamer::updateVideoSizeAndOrientationFromCaps(const GstCaps* caps)
{
    ASSERT(isMainThread());

    // TODO: handle possible clean aperture data. See https://bugzilla.gnome.org/show_bug.cgi?id=596571
    // TODO: handle possible transformation matrix. See https://bugzilla.gnome.org/show_bug.cgi?id=596326

    // Get the video PAR and original size, if this fails the
    // video-sink has likely not yet negotiated its caps.
    int pixelAspectRatioNumerator, pixelAspectRatioDenominator, stride;
    IntSize originalSize;
    GstVideoFormat format;
    if (!getVideoSizeAndFormatFromCaps(caps, originalSize, format, pixelAspectRatioNumerator, pixelAspectRatioDenominator, stride)) {
        GST_WARNING("Failed to get size and format from caps: %" GST_PTR_FORMAT, caps);
        return;
    }

    auto pad = adoptGRef(gst_element_get_static_pad(m_videoSink.get(), "sink"));
    ASSERT(pad);
    auto tagsEvent = adoptGRef(gst_pad_get_sticky_event(pad.get(), GST_EVENT_TAG, 0));
    auto orientation = ImageOrientation::None;
    if (tagsEvent) {
        GstTagList* tagList;
        gst_event_parse_tag(tagsEvent.get(), &tagList);
        orientation = getVideoOrientation(tagList);
    }

    setVideoSourceOrientation(orientation);
    // If the video is tagged as rotated 90 or 270 degrees, swap width and height.
    if (m_videoSourceOrientation.usesWidthAsHeight())
        originalSize = originalSize.transposedSize();

    GST_DEBUG_OBJECT(pipeline(), "Original video size: %dx%d", originalSize.width(), originalSize.height());
    GST_DEBUG_OBJECT(pipeline(), "Pixel aspect ratio: %d/%d", pixelAspectRatioNumerator, pixelAspectRatioDenominator);

    // Calculate DAR based on PAR and video size.
    int displayWidth = originalSize.width() * pixelAspectRatioNumerator;
    int displayHeight = originalSize.height() * pixelAspectRatioDenominator;

    // Divide display width and height by their GCD to avoid possible overflows.
    int displayAspectRatioGCD = gst_util_greatest_common_divisor(displayWidth, displayHeight);
    displayWidth /= displayAspectRatioGCD;
    displayHeight /= displayAspectRatioGCD;

    // Apply DAR to original video size. This is the same behavior as in xvimagesink's setcaps function.
    uint64_t width = 0, height = 0;
    if (!(originalSize.height() % displayHeight)) {
        GST_DEBUG_OBJECT(pipeline(), "Keeping video original height");
        width = gst_util_uint64_scale_int(originalSize.height(), displayWidth, displayHeight);
        height = originalSize.height();
    } else if (!(originalSize.width() % displayWidth)) {
        GST_DEBUG_OBJECT(pipeline(), "Keeping video original width");
        height = gst_util_uint64_scale_int(originalSize.width(), displayHeight, displayWidth);
        width = originalSize.width();
    } else {
        GST_DEBUG_OBJECT(pipeline(), "Approximating while keeping original video height");
        width = gst_util_uint64_scale_int(originalSize.height(), displayWidth, displayHeight);
        height = originalSize.height();
    }

    GST_DEBUG_OBJECT(pipeline(), "Saving natural size: %" G_GUINT64_FORMAT "x%" G_GUINT64_FORMAT, width, height);
    m_videoSize = FloatSize(static_cast<int>(width), static_cast<int>(height));
    m_player->sizeChanged();
}

void MediaPlayerPrivateGStreamer::invalidateCachedPosition() const
{
    m_cachedPosition.reset();
}

void MediaPlayerPrivateGStreamer::invalidateCachedPositionOnNextIteration() const
{
    RunLoop::main().dispatch([weakThis = WeakPtr { *this }, this] {
        if (!weakThis)
            return;
        invalidateCachedPosition();
    });
}

void MediaPlayerPrivateGStreamer::triggerRepaint(GstSample* sample)
{
    ASSERT(!isMainThread());

    GstBuffer* buffer = gst_sample_get_buffer(sample);
    if (buffer && GST_BUFFER_PTS_IS_VALID(buffer)) {
        // Heuristic to avoid asking for playbackPosition() from a non-main thread.
        MediaTime currentTime = MediaTime(gst_segment_to_stream_time(gst_sample_get_segment(sample), GST_FORMAT_TIME, GST_BUFFER_PTS(buffer)), GST_SECOND);
        DataMutexLocker taskAtMediaTimeScheduler { m_TaskAtMediaTimeSchedulerDataMutex };
        if (auto task = taskAtMediaTimeScheduler->checkTaskForScheduling(currentTime))
            RunLoop::main().dispatch(WTFMove(task.value()));
    }

    bool shouldTriggerResize;
    {
        Locker sampleLocker { m_sampleMutex };
        shouldTriggerResize = !m_sample;
        m_sample = sample;
    }

    if (shouldTriggerResize) {
        GST_DEBUG_OBJECT(pipeline(), "First sample reached the sink, triggering video dimensions update");
        GRefPtr<GstCaps> caps(gst_sample_get_caps(sample));
        if (!caps) {
            GST_ERROR_OBJECT(pipeline(), "Received sample without caps: %" GST_PTR_FORMAT, sample);
            return;
        }
        RunLoop::main().dispatch([weakThis = WeakPtr { *this }, this, caps = WTFMove(caps)] {
            if (!weakThis)
                return;
            updateVideoSizeAndOrientationFromCaps(caps.get());

            // Live streams start without pre-rolling, that means they can reach PAUSED while sinks
            // still haven't received a sample to render. So we need to notify the media element in
            // such cases only after pre-rolling has completed. Otherwise the media element might
            // emit a play event too early, before pre-rolling has been completed.
            if (m_isLiveStream && m_readyState < MediaPlayer::ReadyState::HaveEnoughData) {
                m_readyState = MediaPlayer::ReadyState::HaveEnoughData;
                m_player->readyStateChanged();
            }
        });
    }

    if (!m_canRenderingBeAccelerated) {
        Locker locker { m_drawLock };
        if (m_isBeingDestroyed)
            return;
        m_drawTimer.startOneShot(0_s);
        m_drawCondition.wait(m_drawLock);
        return;
    }

#if USE(TEXTURE_MAPPER_GL)
    if (m_isUsingFallbackVideoSink) {
        Locker locker { m_drawLock };
        auto proxyOperation =
            [this](TextureMapperPlatformLayerProxy& proxy)
            {
                return proxy.scheduleUpdateOnCompositorThread([this] { this->pushTextureToCompositor(); });
            };
#if USE(NICOSIA)
        if (!proxyOperation(downcast<Nicosia::ContentLayerTextureMapperImpl>(m_nicosiaLayer->impl()).proxy()))
            return;
#else
        if (!proxyOperation(*m_platformLayerProxy))
            return;
#endif
        m_drawTimer.startOneShot(0_s);
        m_drawCondition.wait(m_drawLock);
    } else
        pushTextureToCompositor();
#endif // USE(TEXTURE_MAPPER_GL)
}

void MediaPlayerPrivateGStreamer::repaintCallback(MediaPlayerPrivateGStreamer* player, GstSample* sample)
{
    player->triggerRepaint(sample);
}

void MediaPlayerPrivateGStreamer::cancelRepaint(bool destroying)
{
    // The goal of this function is to release the GStreamer thread from m_drawCondition in triggerRepaint() in non-AC case,
    // to avoid a deadlock if the player gets paused while waiting for drawing (see https://bugs.webkit.org/show_bug.cgi?id=170003):
    // the main thread is waiting for the GStreamer thread to pause, but the GStreamer thread is locked waiting for the
    // main thread to draw. This deadlock doesn't happen when using AC because the sample is processed (not painted) in the compositor
    // thread, so the main thread can request the pause and wait if the GStreamer thread is waiting for the compositor thread.
    //
    // This function is also used when destroying the player (destroying parameter is true), to release the gstreamer thread from
    // m_drawCondition and to ensure that new triggerRepaint calls won't wait on m_drawCondition.
    if (m_isUsingFallbackVideoSink) {
        Locker locker { m_drawLock };
        m_drawTimer.stop();
        m_isBeingDestroyed = destroying;
        m_drawCondition.notifyOne();
    }
}

void MediaPlayerPrivateGStreamer::repaintCancelledCallback(MediaPlayerPrivateGStreamer* player)
{
    player->cancelRepaint();
}

#if USE(GSTREAMER_GL)
void MediaPlayerPrivateGStreamer::flushCurrentBuffer()
{
    Locker sampleLocker { m_sampleMutex };

    if (m_sample) {
        // Allocate a new copy of the sample which has to be released. The copy is necessary so that
        // the video dimensions can still be fetched and also for canvas rendering. The release is
        // necessary because the sample might have been allocated by a hardware decoder and memory
        // might have to be reclaimed by a non-sysmem buffer pool.
        const GstStructure* info = gst_sample_get_info(m_sample.get());
        auto buffer = adoptGRef(gst_buffer_copy_deep(gst_sample_get_buffer(m_sample.get())));
        m_sample = adoptGRef(gst_sample_new(buffer.get(), gst_sample_get_caps(m_sample.get()),
            gst_sample_get_segment(m_sample.get()), info ? gst_structure_copy(info) : nullptr));
    }

    bool shouldWait = m_videoDecoderPlatform == GstVideoDecoderPlatform::Video4Linux;
    auto proxyOperation = [shouldWait, pipeline = pipeline()](TextureMapperPlatformLayerProxy& proxy) {
        GST_DEBUG_OBJECT(pipeline, "Flushing video sample %s", shouldWait ? "synchronously" : "");
        if (shouldWait) {
            if (proxy.isActive())
                proxy.dropCurrentBufferWhilePreservingTexture(shouldWait);
        } else {
            Locker locker { proxy.lock() };
            if (proxy.isActive())
                proxy.dropCurrentBufferWhilePreservingTexture(shouldWait);
        }
    };

#if USE(NICOSIA)
    proxyOperation(downcast<Nicosia::ContentLayerTextureMapperImpl>(m_nicosiaLayer->impl()).proxy());
#else
    proxyOperation(*m_platformLayerProxy);
#endif
}
#endif

void MediaPlayerPrivateGStreamer::setSize(const IntSize& size)
{
    m_size = size;
}

void MediaPlayerPrivateGStreamer::paint(GraphicsContext& context, const FloatRect& rect)
{
    if (context.paintingDisabled())
        return;

    if (!m_visible)
        return;

    Locker sampleLocker { m_sampleMutex };
    if (!GST_IS_SAMPLE(m_sample.get()))
        return;

#if USE(GSTREAMER_GL)
    // Ensure the input is RGBA. We handle YUV video natively, so we need to do
    // this conversion on-demand here.
    GstBuffer* buffer = gst_sample_get_buffer(m_sample.get());
    if (UNLIKELY(!GST_IS_BUFFER(buffer)))
        return;

    GstCaps* caps = gst_sample_get_caps(m_sample.get());

    GstVideoInfo videoInfo;
    gst_video_info_init(&videoInfo);
    if (!gst_video_info_from_caps(&videoInfo, caps))
        return;

    GstMemory* memory = gst_buffer_peek_memory(buffer, 0);
    bool hasExternalOESTexture = false;
    if (gst_is_gl_memory(memory))
        hasExternalOESTexture = gst_gl_memory_get_texture_target(GST_GL_MEMORY_CAST(memory)) == GST_GL_TEXTURE_TARGET_EXTERNAL_OES;

    if (!GST_VIDEO_INFO_IS_RGB(&videoInfo) || hasExternalOESTexture) {
        if (!m_colorConvert)
            m_colorConvert = adoptGRef(gst_gl_color_convert_new(GST_GL_BASE_MEMORY_CAST(memory)->context));

        if (!m_colorConvertInputCaps || !gst_caps_is_equal(m_colorConvertInputCaps.get(), caps)) {
            m_colorConvertInputCaps = caps;
            m_colorConvertOutputCaps = adoptGRef(gst_caps_copy(caps));

            // These caps must match the internal format of a cairo surface with CAIRO_FORMAT_ARGB32,
            // so we don't need to perform color conversions when painting the video frame.
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
            const char* formatString = GST_VIDEO_INFO_HAS_ALPHA(&videoInfo) ? "BGRA" : "BGRx";
#else
            const char* formatString = GST_VIDEO_INFO_HAS_ALPHA(&videoInfo) ? "ARGB" : "xRGB";
#endif
            gst_caps_set_simple(m_colorConvertOutputCaps.get(), "format", G_TYPE_STRING, formatString,
                "texture-target", G_TYPE_STRING, GST_GL_TEXTURE_TARGET_2D_STR, nullptr);
            if (!gst_gl_color_convert_set_caps(m_colorConvert.get(), caps, m_colorConvertOutputCaps.get()))
                return;
        }

        GRefPtr<GstBuffer> rgbBuffer = adoptGRef(gst_gl_color_convert_perform(m_colorConvert.get(), buffer));
        if (UNLIKELY(!GST_IS_BUFFER(rgbBuffer.get())))
            return;

        const GstStructure* info = gst_sample_get_info(m_sample.get());
        m_sample = adoptGRef(gst_sample_new(rgbBuffer.get(), m_colorConvertOutputCaps.get(),
            gst_sample_get_segment(m_sample.get()), info ? gst_structure_copy(info) : nullptr));
    }
#endif

    auto gstImage = ImageGStreamer::createImage(m_sample.get());
    if (!gstImage)
        return;

    FloatRect imageRect = m_videoSourceOrientation.usesWidthAsHeight() ? FloatRect(gstImage->rect().location(), gstImage->rect().size().transposedSize()) : gstImage->rect();

    context.drawImage(gstImage->image(), rect, imageRect, { gstImage->hasAlpha() ? CompositeOperator::SourceOver : CompositeOperator::Copy, m_videoSourceOrientation });
}

DestinationColorSpace MediaPlayerPrivateGStreamer::colorSpace()
{
    return DestinationColorSpace::SRGB();
}

#if USE(GSTREAMER_GL)
bool MediaPlayerPrivateGStreamer::copyVideoTextureToPlatformTexture(GraphicsContextGL* context, PlatformGLObject outputTexture, GCGLenum outputTarget, GCGLint level, GCGLenum internalFormat, GCGLenum format, GCGLenum type, bool premultiplyAlpha, bool flipY)
{
    UNUSED_PARAM(context);

    if (m_isUsingFallbackVideoSink)
        return false;

    Locker sampleLocker { m_sampleMutex };

    if (!GST_IS_SAMPLE(m_sample.get()))
        return false;

    std::unique_ptr<GstVideoFrameHolder> frameHolder = makeUnique<GstVideoFrameHolder>(m_sample.get(), m_videoDecoderPlatform, m_textureMapperFlags, true);

    std::unique_ptr<TextureMapperPlatformLayerBuffer> layerBuffer = frameHolder->platformLayerBuffer();
    if (!layerBuffer)
        return false;

    auto size = frameHolder->size();
    if (m_videoSourceOrientation.usesWidthAsHeight())
        size = size.transposedSize();

    if (!m_videoTextureCopier)
        m_videoTextureCopier = makeUnique<VideoTextureCopierGStreamer>(TEXTURE_COPIER_COLOR_CONVERT_FLAG);

    frameHolder->waitForCPUSync();

    return m_videoTextureCopier->copyVideoTextureToPlatformTexture(*layerBuffer.get(), size, outputTexture, outputTarget, level, internalFormat, format, type, flipY, m_videoSourceOrientation, premultiplyAlpha);
}

RefPtr<NativeImage> MediaPlayerPrivateGStreamer::nativeImageForCurrentTime()
{
    return nullptr;
}
#endif // USE(GSTREAMER_GL)

void MediaPlayerPrivateGStreamer::setVideoSourceOrientation(ImageOrientation orientation)
{
    if (m_videoSourceOrientation == orientation)
        return;

    m_videoSourceOrientation = orientation;
#if USE(TEXTURE_MAPPER_GL)
    updateTextureMapperFlags();
#endif
}

#if USE(TEXTURE_MAPPER_GL)
void MediaPlayerPrivateGStreamer::updateTextureMapperFlags()
{
    switch (m_videoSourceOrientation) {
    case ImageOrientation::OriginTopLeft:
        m_textureMapperFlags = 0;
        break;
    case ImageOrientation::OriginRightTop:
        m_textureMapperFlags = TextureMapperGL::ShouldRotateTexture90;
        break;
    case ImageOrientation::OriginBottomRight:
        m_textureMapperFlags = TextureMapperGL::ShouldRotateTexture180;
        break;
    case ImageOrientation::OriginLeftBottom:
        m_textureMapperFlags = TextureMapperGL::ShouldRotateTexture270;
        break;
    default:
        // FIXME: Handle OriginTopRight, OriginBottomLeft, OriginLeftTop and OriginRightBottom?
        m_textureMapperFlags = 0;
        break;
    }
}
#endif

bool MediaPlayerPrivateGStreamer::supportsFullscreen() const
{
    return true;
}

MediaPlayer::MovieLoadType MediaPlayerPrivateGStreamer::movieLoadType() const
{
    if (m_readyState == MediaPlayer::ReadyState::HaveNothing)
        return MediaPlayer::MovieLoadType::Unknown;

    if (m_isLiveStream)
        return MediaPlayer::MovieLoadType::LiveStream;

    return MediaPlayer::MovieLoadType::Download;
}

#if USE(GSTREAMER_GL)
GstElement* MediaPlayerPrivateGStreamer::createVideoSinkGL()
{
    const char* disableGLSink = g_getenv("WEBKIT_GST_DISABLE_GL_SINK");
    if (disableGLSink && equal(disableGLSink, "1")) {
        GST_INFO("Disabling hardware-accelerated rendering per user request.");
        return nullptr;
    }

    if (!webKitGLVideoSinkProbePlatform()) {
        g_warning("WebKit wasn't able to find the GL video sink dependencies. Hardware-accelerated zero-copy video rendering can't be enabled without this plugin.");
        return nullptr;
    }

    GstElement* sink = gst_element_factory_make("webkitglvideosink", nullptr);
    ASSERT(sink);
    webKitGLVideoSinkSetMediaPlayerPrivate(WEBKIT_GL_VIDEO_SINK(sink), this);

    return sink;
}
#endif // USE(GSTREAMER_GL)

#if USE(GSTREAMER_HOLEPUNCH)
static void setRectangleToVideoSink(GstElement* videoSink, const IntRect& rect)
{
    // Here goes the platform-dependant code to set to the videoSink the size
    // and position of the video rendering window. Mark them unused as default.
    UNUSED_PARAM(videoSink);
    UNUSED_PARAM(rect);
}

class GStreamerHolePunchClient : public TextureMapperPlatformLayerBuffer::HolePunchClient {
public:
    GStreamerHolePunchClient(GRefPtr<GstElement>&& videoSink) : m_videoSink(WTFMove(videoSink)) { };
    void setVideoRectangle(const IntRect& rect) final { setRectangleToVideoSink(m_videoSink.get(), rect); }
private:
    GRefPtr<GstElement> m_videoSink;
};

GstElement* MediaPlayerPrivateGStreamer::createHolePunchVideoSink()
{
    // Here goes the platform-dependant code to create the videoSink. As a default
    // we use a fakeVideoSink so nothing is drawn to the page.
    GstElement* videoSink =  makeGStreamerElement("fakevideosink", nullptr);

    return videoSink;
}

void MediaPlayerPrivateGStreamer::pushNextHolePunchBuffer()
{
    auto proxyOperation =
        [this](TextureMapperPlatformLayerProxy& proxy)
        {
            Locker locker { proxy.lock() };
            std::unique_ptr<TextureMapperPlatformLayerBuffer> layerBuffer = makeUnique<TextureMapperPlatformLayerBuffer>(0, m_size, TextureMapperGL::ShouldNotBlend, GL_DONT_CARE);
            std::unique_ptr<GStreamerHolePunchClient> holePunchClient = makeUnique<GStreamerHolePunchClient>(m_videoSink.get());
            layerBuffer->setHolePunchClient(WTFMove(holePunchClient));
            proxy.pushNextBuffer(WTFMove(layerBuffer));
        };

#if USE(NICOSIA)
    proxyOperation(downcast<Nicosia::ContentLayerTextureMapperImpl>(m_nicosiaLayer->impl()).proxy());
#else
    proxyOperation(*m_platformLayerProxy);
#endif
}
#endif

GstElement* MediaPlayerPrivateGStreamer::createVideoSink()
{
    acceleratedRenderingStateChanged();

    if (!m_player->isVideoPlayer()) {
        m_videoSink = makeGStreamerElement("fakevideosink", nullptr);
        return m_videoSink.get();
    }

#if USE(GSTREAMER_HOLEPUNCH)
    m_videoSink = createHolePunchVideoSink();
    pushNextHolePunchBuffer();
    return m_videoSink.get();
#endif

#if USE(GSTREAMER_GL)
    if (m_canRenderingBeAccelerated)
        m_videoSink = createVideoSinkGL();
#endif

    if (!m_videoSink) {
        m_isUsingFallbackVideoSink = true;
        m_videoSink = webkitVideoSinkNew();
        g_signal_connect_swapped(m_videoSink.get(), "repaint-requested", G_CALLBACK(repaintCallback), this);
        g_signal_connect_swapped(m_videoSink.get(), "repaint-cancelled", G_CALLBACK(repaintCancelledCallback), this);
    }

    GstElement* videoSink = nullptr;
    if (!webkitGstCheckVersion(1, 18, 0)) {
        m_fpsSink = gst_element_factory_make("fpsdisplaysink", "sink");
        if (m_fpsSink) {
            g_object_set(m_fpsSink.get(), "silent", TRUE , nullptr);

            // Turn off text overlay unless tracing is enabled.
            if (gst_debug_category_get_threshold(webkit_media_player_debug) < GST_LEVEL_TRACE)
                g_object_set(m_fpsSink.get(), "text-overlay", FALSE , nullptr);

            if (g_object_class_find_property(G_OBJECT_GET_CLASS(m_fpsSink.get()), "video-sink")) {
                g_object_set(m_fpsSink.get(), "video-sink", m_videoSink.get(), nullptr);
                videoSink = m_fpsSink.get();
            } else
                m_fpsSink = nullptr;
        }
    }

    if (!m_fpsSink)
        videoSink = m_videoSink.get();

    ASSERT(videoSink);
    return videoSink;
}

void MediaPlayerPrivateGStreamer::setStreamVolumeElement(GstStreamVolume* volume)
{
    ASSERT(!m_volumeElement);
    m_volumeElement = volume;

    // We don't set the initial volume because we trust the sink to keep it for us. See
    // https://bugs.webkit.org/show_bug.cgi?id=118974 for more information.
    if (!m_player->platformVolumeConfigurationRequired()) {
        GST_DEBUG_OBJECT(pipeline(), "Setting stream volume to %f", m_player->volume());
        gst_stream_volume_set_volume(m_volumeElement.get(), GST_STREAM_VOLUME_FORMAT_LINEAR, static_cast<double>(m_player->volume()));
    } else
        GST_DEBUG_OBJECT(pipeline(), "Not setting stream volume, trusting system one");

    GST_DEBUG_OBJECT(pipeline(), "Setting stream muted %s", toString(m_player->muted()).utf8().data());
    g_object_set(m_volumeElement.get(), "mute", m_player->muted(), nullptr);

    g_signal_connect_swapped(m_volumeElement.get(), "notify::volume", G_CALLBACK(volumeChangedCallback), this);
    g_signal_connect_swapped(m_volumeElement.get(), "notify::mute", G_CALLBACK(muteChangedCallback), this);
}

std::optional<VideoPlaybackQualityMetrics> MediaPlayerPrivateGStreamer::videoPlaybackQualityMetrics()
{
    if (!webkitGstCheckVersion(1, 18, 0) && !m_fpsSink)
        return std::nullopt;

    uint64_t totalVideoFrames = 0;
    uint64_t droppedVideoFrames = 0;
    if (webkitGstCheckVersion(1, 18, 0)) {
        GUniqueOutPtr<GstStructure> stats;
        g_object_get(m_videoSink.get(), "stats", &stats.outPtr(), nullptr);

        if (!gst_structure_get_uint64(stats.get(), "rendered", &totalVideoFrames))
            return std::nullopt;

        if (!gst_structure_get_uint64(stats.get(), "dropped", &droppedVideoFrames))
            return std::nullopt;
    } else if (m_fpsSink) {
        unsigned renderedFrames, droppedFrames;
        g_object_get(m_fpsSink.get(), "frames-rendered", &renderedFrames, "frames-dropped", &droppedFrames, nullptr);
        totalVideoFrames = renderedFrames;
        droppedVideoFrames = droppedFrames;
    }

    // Cache or reuse cached statistics. Caching is required so that metrics queries performed
    // after EOS still return valid values.
    if (totalVideoFrames)
        m_totalVideoFrames = totalVideoFrames;
    else if (m_totalVideoFrames)
        totalVideoFrames = m_totalVideoFrames;
    if (droppedVideoFrames)
        m_droppedVideoFrames = droppedVideoFrames;
    else if (m_droppedVideoFrames)
        droppedVideoFrames = m_droppedVideoFrames;

    uint32_t corruptedVideoFrames = 0;
    double totalFrameDelay = 0;
    uint32_t displayCompositedVideoFrames = 0;
    return VideoPlaybackQualityMetrics {
        static_cast<uint32_t>(totalVideoFrames),
        static_cast<uint32_t>(droppedVideoFrames),
        corruptedVideoFrames,
        totalFrameDelay,
        displayCompositedVideoFrames,
    };
}

#if ENABLE(ENCRYPTED_MEDIA)
InitData MediaPlayerPrivateGStreamer::parseInitDataFromProtectionMessage(GstMessage* message)
{
    ASSERT(!isMainThread());

    InitData initData;
    {
        Locker locker { m_protectionMutex };
        ProtectionSystemEvents protectionSystemEvents(message);
        GST_TRACE_OBJECT(pipeline(), "found %zu protection events, %zu decryptors available", protectionSystemEvents.events().size(), protectionSystemEvents.availableSystems().size());

        for (auto& event : protectionSystemEvents.events()) {
            const char* eventKeySystemId = nullptr;
            GstBuffer* data = nullptr;
            gst_event_parse_protection(event.get(), &eventKeySystemId, &data, nullptr);

            initData.append({eventKeySystemId, data});
            m_handledProtectionEvents.add(GST_EVENT_SEQNUM(event.get()));
        }
    }

    return initData;
}

bool MediaPlayerPrivateGStreamer::waitForCDMAttachment()
{
    if (isMainThread()) {
        GST_ERROR_OBJECT(pipeline(), "can't block the main thread waiting for a CDM instance");
        ASSERT_NOT_REACHED();
        return false;
    }

    GST_INFO_OBJECT(pipeline(), "waiting for a CDM instance");

    bool didCDMAttach = false;
    {
        Locker cdmAttachmentLocker { m_cdmAttachmentLock };
        didCDMAttach = m_cdmAttachmentCondition.waitFor(m_cdmAttachmentLock, 4_s, [this]() {
            assertIsHeld(m_cdmAttachmentLock);
            return isCDMAttached();
        });
    }

    return didCDMAttach;
}

void MediaPlayerPrivateGStreamer::initializationDataEncountered(InitData&& initData)
{
    ASSERT(!isMainThread());

    RunLoop::main().dispatch([weakThis = WeakPtr { *this }, initData = WTFMove(initData)] {
        if (!weakThis)
            return;

        GST_DEBUG("scheduling initializationDataEncountered %s event of size %zu", initData.payloadContainerType().utf8().data(),
            initData.payload()->size());
        GST_MEMDUMP("init datas", reinterpret_cast<const uint8_t*>(initData.payload()->data()), initData.payload()->size());
        weakThis->m_player->initializationDataEncountered(initData.payloadContainerType(), initData.payload()->tryCreateArrayBuffer());
    });
}

void MediaPlayerPrivateGStreamer::cdmInstanceAttached(CDMInstance& instance)
{
    ASSERT(isMainThread());

    if (m_cdmInstance == &instance)
        return;

    if (!m_pipeline) {
        GST_ERROR("no pipeline yet");
        ASSERT_NOT_REACHED();
        return;
    }

    m_cdmInstance = reinterpret_cast<CDMInstanceProxy*>(&instance);
    RELEASE_ASSERT(m_cdmInstance);
    m_cdmInstance->setPlayer(m_player);

    GRefPtr<GstContext> context = adoptGRef(gst_context_new("drm-cdm-proxy", FALSE));
    GstStructure* contextStructure = gst_context_writable_structure(context.get());
    gst_structure_set(contextStructure, "cdm-proxy", G_TYPE_POINTER, m_cdmInstance->proxy().get(), nullptr);
    gst_element_set_context(GST_ELEMENT(m_pipeline.get()), context.get());

    GST_DEBUG_OBJECT(m_pipeline.get(), "CDM proxy instance %p dispatched as context", m_cdmInstance->proxy().get());

    Locker cdmAttachmentLocker { m_cdmAttachmentLock };
    // We must notify all waiters, since several demuxers can be simultaneously waiting for a CDM.
    m_cdmAttachmentCondition.notifyAll();
}

void MediaPlayerPrivateGStreamer::cdmInstanceDetached(CDMInstance& instance)
{
    UNUSED_PARAM(instance);
    ASSERT(isMainThread());
    ASSERT(m_pipeline);

    if (!m_cdmInstance)
        return;

    ASSERT(m_cdmInstance == &instance);
    GST_DEBUG_OBJECT(m_pipeline.get(), "detaching CDM instance %p, setting empty context", m_cdmInstance.get());
    m_cdmInstance = nullptr;
    GRefPtr<GstContext> context = adoptGRef(gst_context_new("drm-cdm-proxy", FALSE));
    gst_element_set_context(GST_ELEMENT(m_pipeline.get()), context.get());
}

void MediaPlayerPrivateGStreamer::attemptToDecryptWithInstance(CDMInstance& instance)
{
    ASSERT(m_cdmInstance == &instance);
    GST_TRACE("instance %p, current stored %p", &instance, m_cdmInstance.get());
    attemptToDecryptWithLocalInstance();
}

void MediaPlayerPrivateGStreamer::attemptToDecryptWithLocalInstance()
{
    bool wasEventHandled = gst_element_send_event(pipeline(), gst_event_new_custom(GST_EVENT_CUSTOM_DOWNSTREAM_OOB, gst_structure_new_empty("attempt-to-decrypt")));
    GST_DEBUG("attempting to decrypt, event handled %s", boolForPrinting(wasEventHandled));
}

void MediaPlayerPrivateGStreamer::handleProtectionEvent(GstEvent* event)
{
    {
        Locker locker { m_protectionMutex };
        if (m_handledProtectionEvents.contains(GST_EVENT_SEQNUM(event))) {
            GST_DEBUG_OBJECT(pipeline(), "event %u already handled", GST_EVENT_SEQNUM(event));
            return;
        }
    }
    GST_DEBUG_OBJECT(pipeline(), "handling event %u from MSE", GST_EVENT_SEQNUM(event));
    const char* eventKeySystemUUID = nullptr;
    GstBuffer* initData = nullptr;
    gst_event_parse_protection(event, &eventKeySystemUUID, &initData, nullptr);
    initializationDataEncountered({eventKeySystemUUID, initData});
}

bool MediaPlayerPrivateGStreamer::waitingForKey() const
{
    if (!m_pipeline || !m_cdmInstance)
        return false;

    return m_cdmInstance->isWaitingForKey();
}
#endif

bool MediaPlayerPrivateGStreamer::supportsKeySystem(const String& keySystem, const String& mimeType)
{
    bool result = false;

#if ENABLE(ENCRYPTED_MEDIA)
    result = GStreamerEMEUtilities::isClearKeyKeySystem(keySystem);
#endif

    GST_DEBUG("checking for KeySystem support with %s and type %s: %s", keySystem.utf8().data(), mimeType.utf8().data(), boolForPrinting(result));
    return result;
}

MediaPlayer::SupportsType MediaPlayerPrivateGStreamer::extendedSupportsType(const MediaEngineSupportParameters& parameters, MediaPlayer::SupportsType result)
{
    UNUSED_PARAM(parameters);
    return result;
}

#if !RELEASE_LOG_DISABLED
WTFLogChannel& MediaPlayerPrivateGStreamer::logChannel() const
{
    return WebCore::LogMedia;
}
#endif

}

#endif // USE(GSTREAMER)