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(Auto)update libjingle 68701339-> 68703656

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6352 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-06-06 16:29:00 +00:00
parent 910473b31a
commit 91c910469f
1 changed files with 18 additions and 19 deletions
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37
talk/media/webrtc/webrtcvideoengine.cc
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@ -260,11 +260,11 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
const int kVideoCodecClockratekHz = cricket::kVideoCodecClockrate / 1000;
int64 elapsed_time_ms =
(rtp_ts_wraparound_handler_.Unwrap(rtp_time_stamp) -
capture_start_rtp_time_stamp_) / kVideoCodecClockratekHz;
#ifdef USE_WEBRTC_DEV_BRANCH
if (ntp_time_ms > 0) {
int64 elapsed_time_ms =
(rtp_ts_wraparound_handler_.Unwrap(rtp_time_stamp) -
capture_start_rtp_time_stamp_) / kVideoCodecClockratekHz;
capture_start_ntp_time_ms_ = ntp_time_ms - elapsed_time_ms;
}
#endif
@ -272,30 +272,31 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
if (renderer_ == NULL) {
return 0;
}
// Convert 90K rtp timestamp to ns timestamp.
int64 rtp_time_stamp_in_ns = (rtp_time_stamp / kVideoCodecClockratekHz) *
talk_base::kNumNanosecsPerMillisec;
// Convert elapsed_time_ms to ns timestamp.
int64 elapsed_time_ns =
elapsed_time_ms * talk_base::kNumNanosecsPerMillisec;
// Convert milisecond render time to ns timestamp.
int64 render_time_stamp_in_ns = render_time *
int64 render_time_ns = render_time *
talk_base::kNumNanosecsPerMillisec;
// Send the rtp timestamp to renderer as the VideoFrame timestamp.
// and the render timestamp as the VideoFrame elapsed_time.
// Note that here we send the |elapsed_time_ns| to renderer as the
// cricket::VideoFrame's elapsed_time_ and the |render_time_ns| as the
// cricket::VideoFrame's time_stamp_.
if (handle == NULL) {
return DeliverBufferFrame(buffer, buffer_size, render_time_stamp_in_ns,
rtp_time_stamp_in_ns);
return DeliverBufferFrame(buffer, buffer_size, render_time_ns,
elapsed_time_ns);
} else {
return DeliverTextureFrame(handle, render_time_stamp_in_ns,
rtp_time_stamp_in_ns);
return DeliverTextureFrame(handle, render_time_ns,
elapsed_time_ns);
}
}
virtual bool IsTextureSupported() { return true; }
int DeliverBufferFrame(unsigned char* buffer, int buffer_size,
int64 elapsed_time, int64 rtp_time_stamp_in_ns) {
int64 time_stamp, int64 elapsed_time) {
WebRtcVideoFrame video_frame;
video_frame.Alias(buffer, buffer_size, width_, height_,
1, 1, elapsed_time, rtp_time_stamp_in_ns, 0);
1, 1, elapsed_time, time_stamp, 0);
// Sanity check on decoded frame size.
if (buffer_size != static_cast<int>(VideoFrame::SizeOf(width_, height_))) {
@ -308,12 +309,10 @@ class WebRtcRenderAdapter : public webrtc::ExternalRenderer {
return ret;
}
int DeliverTextureFrame(void* handle,
int64 elapsed_time,
int64 rtp_time_stamp_in_ns) {
int DeliverTextureFrame(void* handle, int64 time_stamp, int64 elapsed_time) {
WebRtcTextureVideoFrame video_frame(
static_cast<webrtc::NativeHandle*>(handle), width_, height_,
elapsed_time, rtp_time_stamp_in_ns);
elapsed_time, time_stamp);
return renderer_->RenderFrame(&video_frame);
}