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/*
* Copyright (C) 2016 Metrological Group B.V.
* Copyright (C) 2016, 2017, 2018 Igalia S.L
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public License
* aint with this library; see the file COPYING.LIB. If not, write to
* the Free Software Foundation, Inc., 51 Franklin Street, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#include "config.h"
#include "MediaSampleGStreamer.h"
#include "GStreamerCommon.h"
#include <algorithm>
#if ENABLE(VIDEO) && USE(GSTREAMER)
namespace WebCore {
MediaSampleGStreamer::MediaSampleGStreamer(GRefPtr<GstSample>&& sample, const FloatSize& presentationSize, const AtomString& trackId)
: m_pts(MediaTime::zeroTime())
, m_dts(MediaTime::zeroTime())
, m_duration(MediaTime::zeroTime())
, m_trackId(trackId)
, m_presentationSize(presentationSize)
{
const GstClockTime minimumDuration = 1000; // 1 us
ASSERT(sample);
GstBuffer* buffer = gst_sample_get_buffer(sample.get());
RELEASE_ASSERT(buffer);
auto createMediaTime =
[](GstClockTime time) -> MediaTime {
return MediaTime(GST_TIME_AS_USECONDS(time), G_USEC_PER_SEC);
};
if (GST_BUFFER_PTS_IS_VALID(buffer))
m_pts = createMediaTime(GST_BUFFER_PTS(buffer));
if (GST_BUFFER_DTS_IS_VALID(buffer) || GST_BUFFER_PTS_IS_VALID(buffer))
m_dts = createMediaTime(GST_BUFFER_DTS_OR_PTS(buffer));
if (GST_BUFFER_DURATION_IS_VALID(buffer)) {
// Sometimes (albeit rarely, so far seen only at the end of a track)
// frames have very small durations, so small that may be under the
// precision we are working with and be truncated to zero.
// SourceBuffer algorithms are not expecting frames with zero-duration,
// so let's use something very small instead in those fringe cases.
m_duration = createMediaTime(std::max(GST_BUFFER_DURATION(buffer), minimumDuration));
} else {
// Unfortunately, sometimes samples don't provide a duration. This can never happen in MP4 because of the way
// the format is laid out, but it's pretty common in WebM.
// The good part is that durations don't matter for playback, just for buffered ranges and coded frame deletion.
// We want to pick something small enough to not cause unwanted frame deletion, but big enough to never be
// mistaken for a rounding artifact.
m_duration = createMediaTime(16666667); // 1/60 seconds
}
m_size = gst_buffer_get_size(buffer);
m_sample = sample;
if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DELTA_UNIT))
m_flags = MediaSample::None;
if (GST_BUFFER_FLAG_IS_SET(buffer, GST_BUFFER_FLAG_DECODE_ONLY))
m_flags = static_cast<MediaSample::SampleFlags>(m_flags | MediaSample::IsNonDisplaying);
}
MediaSampleGStreamer::MediaSampleGStreamer(const FloatSize& presentationSize, const AtomString& trackId)
: m_pts(MediaTime::zeroTime())
, m_dts(MediaTime::zeroTime())
, m_duration(MediaTime::zeroTime())
, m_trackId(trackId)
, m_presentationSize(presentationSize)
{
}
Ref<MediaSampleGStreamer> MediaSampleGStreamer::createFakeSample(GstCaps*, MediaTime pts, MediaTime dts, MediaTime duration, const FloatSize& presentationSize, const AtomString& trackId)
{
MediaSampleGStreamer* gstreamerMediaSample = new MediaSampleGStreamer(presentationSize, trackId);
gstreamerMediaSample->m_pts = pts;
gstreamerMediaSample->m_dts = dts;
gstreamerMediaSample->m_duration = duration;
gstreamerMediaSample->m_flags = MediaSample::IsNonDisplaying;
return adoptRef(*gstreamerMediaSample);
}
void MediaSampleGStreamer::applyPtsOffset(MediaTime timestampOffset)
{
if (m_pts > timestampOffset) {
m_duration = m_duration + (m_pts - timestampOffset);
m_pts = timestampOffset;
}
}
void MediaSampleGStreamer::offsetTimestampsBy(const MediaTime& timestampOffset)
{
if (!timestampOffset)
return;
m_pts += timestampOffset;
m_dts += timestampOffset;
if (auto* buffer = gst_sample_get_buffer(m_sample.get())) {
GST_BUFFER_PTS(buffer) = toGstClockTime(m_pts);
GST_BUFFER_DTS(buffer) = toGstClockTime(m_dts);
}
}
PlatformSample MediaSampleGStreamer::platformSample()
{
PlatformSample sample = { PlatformSample::GStreamerSampleType, { .gstSample = m_sample.get() } };
return sample;
}
Ref<MediaSample> MediaSampleGStreamer::createNonDisplayingCopy() const
{
if (!m_sample)
return createFakeSample(nullptr, m_pts, m_dts, m_duration, m_presentationSize, m_trackId);
GstBuffer* buffer = gst_sample_get_buffer(m_sample.get());
GST_BUFFER_FLAG_SET(buffer, GST_BUFFER_FLAG_DECODE_ONLY);
GstCaps* caps = gst_sample_get_caps(m_sample.get());
GstSegment* segment = gst_sample_get_segment(m_sample.get());
const GstStructure* originalInfo = gst_sample_get_info(m_sample.get());
GstStructure* info = originalInfo ? gst_structure_copy(originalInfo) : nullptr;
GRefPtr<GstSample> sample = adoptGRef(gst_sample_new(buffer, caps, segment, info));
return adoptRef(*new MediaSampleGStreamer(sample.get(), m_presentationSize, m_trackId));
}
void MediaSampleGStreamer::dump(PrintStream& out) const
{
out.print("{PTS(", presentationTime(), "), DTS(", decodeTime(), "), duration(", duration(), "), flags(");
bool anyFlags = false;
auto appendFlag = [&out, &anyFlags](const char* flagName) {
if (anyFlags)
out.print(",");
out.print(flagName);
anyFlags = true;
};
if (flags() & MediaSample::IsSync)
appendFlag("sync");
if (flags() & MediaSample::IsNonDisplaying)
appendFlag("non-displaying");
if (flags() & MediaSample::HasAlpha)
appendFlag("has-alpha");
if (flags() & ~(MediaSample::IsSync | MediaSample::IsNonDisplaying | MediaSample::HasAlpha))
appendFlag("unknown-flag");
out.print("), trackId(", trackID().string(), "), presentationSize(", presentationSize().width(), "x", presentationSize().height(), ")}");
}
} // namespace WebCore.
#endif // ENABLE(VIDEO) && USE(GSTREAMER)