generic-library/webrtc
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(Auto)update libjingle 69588608-> 69588980

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git-svn-id: http://webrtc.googlecode.com/svn/trunk@6502 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
buildbot@webrtc.org 2014-06-20 00:18:36 +00:00
parent 8563ef448a
commit 0970dd8767
2 changed files with 1 additions and 4 deletions
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3
talk/media/webrtc/webrtcvideoengine.cc
View File

@ -3261,7 +3261,7 @@ bool WebRtcVideoMediaChannel::SendFrame(
}
const VideoFrame* frame_out = frame;
talk_base::scoped_ptr<VideoFrame> processed_frame;
// Disable muting for screencast.
// TODO(hellner): Remove the need for disabling mute when screencasting.
const bool mute = (send_channel->muted() && !is_screencast);
send_channel->ProcessFrame(*frame_out, mute, processed_frame.use());
if (processed_frame) {
@ -3767,7 +3767,6 @@ bool WebRtcVideoMediaChannel::SetSendCodec(
return true;
}
static std::string ToString(webrtc::VideoCodecComplexity complexity) {
switch (complexity) {
case webrtc::kComplexityNormal:

2
talk/media/webrtc/webrtcvideoengine.h
View File

@ -45,7 +45,6 @@
#error "Bogus include."
#endif
namespace webrtc {
class VideoCaptureModule;
class VideoDecoder;
@ -390,7 +389,6 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler,
}
bool RemoveCapturer(uint32 ssrc);
talk_base::MessageQueue* worker_thread() { return engine_->worker_thread(); }
void QueueBlackFrame(uint32 ssrc, int64 timestamp, int framerate);
void FlushBlackFrame(uint32 ssrc, int64 timestamp);