(Auto)update libjingle 69588608-> 69588980
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6502 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -3261,7 +3261,7 @@ bool WebRtcVideoMediaChannel::SendFrame(
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}
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}
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const VideoFrame* frame_out = frame;
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const VideoFrame* frame_out = frame;
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talk_base::scoped_ptr<VideoFrame> processed_frame;
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talk_base::scoped_ptr<VideoFrame> processed_frame;
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// Disable muting for screencast.
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// TODO(hellner): Remove the need for disabling mute when screencasting.
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const bool mute = (send_channel->muted() && !is_screencast);
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const bool mute = (send_channel->muted() && !is_screencast);
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send_channel->ProcessFrame(*frame_out, mute, processed_frame.use());
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send_channel->ProcessFrame(*frame_out, mute, processed_frame.use());
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if (processed_frame) {
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if (processed_frame) {
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@ -3767,7 +3767,6 @@ bool WebRtcVideoMediaChannel::SetSendCodec(
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return true;
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return true;
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}
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}
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static std::string ToString(webrtc::VideoCodecComplexity complexity) {
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static std::string ToString(webrtc::VideoCodecComplexity complexity) {
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switch (complexity) {
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switch (complexity) {
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case webrtc::kComplexityNormal:
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case webrtc::kComplexityNormal:
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@ -45,7 +45,6 @@
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#error "Bogus include."
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#error "Bogus include."
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#endif
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#endif
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namespace webrtc {
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namespace webrtc {
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class VideoCaptureModule;
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class VideoCaptureModule;
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class VideoDecoder;
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class VideoDecoder;
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@ -390,7 +389,6 @@ class WebRtcVideoMediaChannel : public talk_base::MessageHandler,
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}
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}
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bool RemoveCapturer(uint32 ssrc);
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bool RemoveCapturer(uint32 ssrc);
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talk_base::MessageQueue* worker_thread() { return engine_->worker_thread(); }
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talk_base::MessageQueue* worker_thread() { return engine_->worker_thread(); }
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void QueueBlackFrame(uint32 ssrc, int64 timestamp, int framerate);
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void QueueBlackFrame(uint32 ssrc, int64 timestamp, int framerate);
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void FlushBlackFrame(uint32 ssrc, int64 timestamp);
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void FlushBlackFrame(uint32 ssrc, int64 timestamp);
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