void PlaybackPipeline::enqueueSample(RefPtr<MediaSample> mediaSample)
{
ASSERT(WTF::isMainThread());
AtomicString trackId = mediaSample->trackID();
GST_TRACE("enqueing sample trackId=%s PTS=%f presentationSize=%.0fx%.0f at %" GST_TIME_FORMAT " duration: %" GST_TIME_FORMAT,
trackId.string().utf8().data(), mediaSample->presentationTime().toFloat(),
mediaSample->presentationSize().width(), mediaSample->presentationSize().height(),
GST_TIME_ARGS(WebCore::toGstClockTime(mediaSample->presentationTime().toDouble())),
GST_TIME_ARGS(WebCore::toGstClockTime(mediaSample->duration().toDouble())));
Stream* stream = getStreamByTrackId(m_webKitMediaSrc.get(), trackId);
if (!stream) {
GST_WARNING("No stream!");
return;
}
if (!stream->sourceBuffer->isReadyForMoreSamples(trackId)) {
GST_DEBUG("enqueueSample: skip adding new sample for trackId=%s, SB is not ready yet", trackId.string().utf8().data());
return;
}
GstElement* appsrc = stream->appsrc;
MediaTime lastEnqueuedTime = stream->lastEnqueuedTime;
GStreamerMediaSample* sample = static_cast<GStreamerMediaSample*>(mediaSample.get());
if (sample->sample() && gst_sample_get_buffer(sample->sample())) {
GRefPtr<GstSample> gstSample = sample->sample();
GstBuffer* buffer = gst_sample_get_buffer(gstSample.get());
lastEnqueuedTime = sample->presentationTime();
GST_BUFFER_FLAG_UNSET(buffer, GST_BUFFER_FLAG_DECODE_ONLY);
pushSample(GST_APP_SRC(appsrc), gstSample.get());
// gst_app_src_push_sample() uses transfer-none for gstSample.
stream->lastEnqueuedTime = lastEnqueuedTime;
}
}
void pushSamples (const float* inputSamples, const int num) noexcept
{
for (int i = 0; i < num; ++i)
pushSample (inputSamples[i]);
}
