void MediaPlayerPrivateGStreamerBase::setVolume(float volume)
{
if (!m_volumeElement)
return;
LOG_MEDIA_MESSAGE("Setting volume: %f", volume);
gst_stream_volume_set_volume(m_volumeElement.get(), GST_STREAM_VOLUME_FORMAT_CUBIC, static_cast<double>(volume));
}
void MediaPlayerPrivateGStreamerBase::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()) {
LOG_MEDIA_MESSAGE("Setting stream volume to %f", m_player->volume());
g_object_set(m_volumeElement.get(), "volume", m_player->volume(), NULL);
} else
LOG_MEDIA_MESSAGE("Not setting stream volume, trusting system one");
LOG_MEDIA_MESSAGE("Setting stream muted %d", m_player->muted());
g_object_set(m_volumeElement.get(), "mute", m_player->muted(), NULL);
m_volumeSignalHandler = g_signal_connect(m_volumeElement.get(), "notify::volume", G_CALLBACK(mediaPlayerPrivateVolumeChangedCallback), this);
m_muteSignalHandler = g_signal_connect(m_volumeElement.get(), "notify::mute", G_CALLBACK(mediaPlayerPrivateMuteChangedCallback), this);
}
// Returns the size of the video
IntSize MediaPlayerPrivateGStreamerBase::naturalSize() const
{
if (!hasVideo())
return IntSize();
if (!m_videoSize.isEmpty())
return m_videoSize;
GRefPtr<GstCaps> caps = currentVideoSinkCaps();
if (!caps)
return IntSize();
// 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.get(), originalSize, format, pixelAspectRatioNumerator, pixelAspectRatioDenominator, stride))
return IntSize();
LOG_MEDIA_MESSAGE("Original video size: %dx%d", originalSize.width(), originalSize.height());
LOG_MEDIA_MESSAGE("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 = greatestCommonDivisor(displayWidth, displayHeight);
displayWidth /= displayAspectRatioGCD;
displayHeight /= displayAspectRatioGCD;
// Apply DAR to original video size. This is the same behavior as in xvimagesink's setcaps function.
guint64 width = 0, height = 0;
if (!(originalSize.height() % displayHeight)) {
LOG_MEDIA_MESSAGE("Keeping video original height");
width = gst_util_uint64_scale_int(originalSize.height(), displayWidth, displayHeight);
height = static_cast<guint64>(originalSize.height());
} else if (!(originalSize.width() % displayWidth)) {
LOG_MEDIA_MESSAGE("Keeping video original width");
height = gst_util_uint64_scale_int(originalSize.width(), displayHeight, displayWidth);
width = static_cast<guint64>(originalSize.width());
} else {
LOG_MEDIA_MESSAGE("Approximating while keeping original video height");
width = gst_util_uint64_scale_int(originalSize.height(), displayWidth, displayHeight);
height = static_cast<guint64>(originalSize.height());
}
LOG_MEDIA_MESSAGE("Natural size: %" G_GUINT64_FORMAT "x%" G_GUINT64_FORMAT, width, height);
m_videoSize = IntSize(static_cast<int>(width), static_cast<int>(height));
return m_videoSize;
}
void PlaybackPipeline::flushAndEnqueueNonDisplayingSamples(Vector<RefPtr<MediaSample> > samples)
{
ASSERT(WTF::isMainThread());
if (samples.size() == 0) {
LOG_MEDIA_MESSAGE("No samples, trackId unknown");
return;
}
AtomicString trackId = samples[0]->trackID();
LOG_MEDIA_MESSAGE("flushAndEnqueueNonDisplayingSamples: trackId=%s PTS[0]=%f ... PTS[n]=%f", trackId.string().utf8().data(), samples[0]->presentationTime().toFloat(), samples[samples.size()-1]->presentationTime().toFloat());
GST_DEBUG_OBJECT(m_webKitMediaSrc.get(), "Flushing and re-enqueing %d samples for stream %s", samples.size(), trackId.string().utf8().data());
GST_OBJECT_LOCK(m_webKitMediaSrc.get());
Stream* stream = getStreamByTrackId(m_webKitMediaSrc.get(), trackId);
if (!stream) {
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
return;
}
GstElement* appsrc = stream->appsrc;
GST_OBJECT_UNLOCK(m_webKitMediaSrc.get());
// Actually no need to flush. The seek preparations have done it for us.
for (Vector<RefPtr<MediaSample> >::iterator it = samples.begin(); it != samples.end(); ++it) {
GStreamerMediaSample* sample = static_cast<GStreamerMediaSample*>(it->get());
if (sample->sample() && gst_sample_get_buffer(sample->sample())) {
GstSample* gstsample = gst_sample_ref(sample->sample());
GST_BUFFER_FLAG_SET(gst_sample_get_buffer(gstsample), GST_BUFFER_FLAG_DECODE_ONLY);
push_sample(GST_APP_SRC(appsrc), gstsample);
// gst_app_src_push_sample() uses transfer-none for gstsample
gst_sample_unref(gstsample);
}
}
}
static void mediaPlayerPrivateVolumeChangedCallback(GObject*, GParamSpec*, MediaPlayerPrivateGStreamerBase* player)
{
// This is called when m_volumeElement receives the notify::volume signal.
LOG_MEDIA_MESSAGE("Volume changed to: %f", player->volume());
player->volumeChanged();
}
void PlaybackPipeline::setWebKitMediaSrc(WebKitMediaSrc* webKitMediaSrc)
{
LOG_MEDIA_MESSAGE("webKitMediaSrc=%p", webKitMediaSrc);
m_webKitMediaSrc = adoptGRef(static_cast<WebKitMediaSrc*>(gst_object_ref(webKitMediaSrc)));
}
void PlaybackPipeline::reattachTrack(RefPtr<SourceBufferPrivateGStreamer> sourceBufferPrivate, RefPtr<TrackPrivateBase> trackPrivate, GstCaps* caps)
{
LOG_MEDIA_MESSAGE("Re-attaching track");
UNUSED_PARAM(caps);
// TODO: Maybe remove this method.
// Now the caps change is managed by gst_appsrc_push_sample()
// in enqueueSample() and flushAndEnqueueNonDisplayingSamples().
WebKitMediaSrc* webKitMediaSrc = m_webKitMediaSrc.get();
GST_OBJECT_LOCK(webKitMediaSrc);
Stream* stream = getStreamBySourceBufferPrivate(webKitMediaSrc, sourceBufferPrivate.get());
GST_OBJECT_UNLOCK(webKitMediaSrc);
ASSERT(stream != 0);
ASSERT(stream->type != Invalid);
GstCaps* oldAppsrccaps = gst_app_src_get_caps(GST_APP_SRC(stream->appsrc));
// Now the caps change is managed by gst_appsrc_push_sample()
// in enqueueSample() and flushAndEnqueueNonDisplayingSamples().
// gst_app_src_set_caps(GST_APP_SRC(stream->appsrc), caps);
GstCaps* appsrccaps = gst_app_src_get_caps(GST_APP_SRC(stream->appsrc));
const gchar* mediaType = gst_structure_get_name(gst_caps_get_structure(appsrccaps, 0));
if (!gst_caps_is_equal(oldAppsrccaps, appsrccaps)) {
LOG_MEDIA_MESSAGE("Caps have changed, but reconstructing the sequence of elements is not supported yet");
gchar* stroldcaps = gst_caps_to_string(oldAppsrccaps);
gchar* strnewcaps = gst_caps_to_string(appsrccaps);
LOG_MEDIA_MESSAGE("oldcaps: %s", stroldcaps);
LOG_MEDIA_MESSAGE("newcaps: %s", strnewcaps);
g_free(stroldcaps);
g_free(strnewcaps);
}
int signal = -1;
GST_OBJECT_LOCK(webKitMediaSrc);
if (g_str_has_prefix(mediaType, "audio")) {
ASSERT(stream->type == Audio);
signal = SIGNAL_AUDIO_CHANGED;
stream->audioTrack = RefPtr<WebCore::AudioTrackPrivateGStreamer>(static_cast<WebCore::AudioTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "video")) {
ASSERT(stream->type == Video);
signal = SIGNAL_VIDEO_CHANGED;
stream->videoTrack = RefPtr<WebCore::VideoTrackPrivateGStreamer>(static_cast<WebCore::VideoTrackPrivateGStreamer*>(trackPrivate.get()));
} else if (g_str_has_prefix(mediaType, "text")) {
ASSERT(stream->type == Text);
signal = SIGNAL_TEXT_CHANGED;
// TODO: Support text tracks and mediaTypes related to EME
}
GST_OBJECT_UNLOCK(webKitMediaSrc);
gst_caps_unref(appsrccaps);
gst_caps_unref(oldAppsrccaps);
if (signal != -1)
g_signal_emit(G_OBJECT(stream->parent), webkit_media_src_signals[signal], 0, NULL);
}
