| Index: webrtc/media/engine/webrtcvideoengine2.h
|
| diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h
|
| index ba2dbd8a02b1d71faec77e49aa357ce426bcad56..b9ab0b71ed6f41dcd277c1e83dfcd211cc3c1b28 100644
|
| --- a/webrtc/media/engine/webrtcvideoengine2.h
|
| +++ b/webrtc/media/engine/webrtcvideoengine2.h
|
| @@ -161,7 +161,9 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| bool SetSink(uint32_t ssrc,
|
| rtc::VideoSinkInterface<VideoFrame>* sink) override;
|
| bool GetStats(VideoMediaInfo* info) override;
|
| - bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) override;
|
| + void SetSource(
|
| + uint32_t ssrc,
|
| + rtc::VideoSourceInterface<cricket::VideoFrame>* source) override;
|
|
|
| void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
|
| const rtc::PacketTime& packet_time) override;
|
| @@ -217,8 +219,6 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| bool GetChangedRecvParameters(const VideoRecvParameters& params,
|
| ChangedRecvParameters* changed_params) const;
|
|
|
| - bool MuteStream(uint32_t ssrc, bool mute);
|
| -
|
| void SetMaxSendBandwidth(int bps);
|
| void SetOptions(uint32_t ssrc, const VideoOptions& options);
|
|
|
| @@ -235,7 +235,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| static std::string CodecSettingsVectorToString(
|
| const std::vector<VideoCodecSettings>& codecs);
|
|
|
| - // Wrapper for the sender part, this is where the capturer is connected and
|
| + // Wrapper for the sender part, this is where the source is connected and
|
| // frames are then converted from cricket frames to webrtc frames.
|
| class WebRtcVideoSendStream
|
| : public rtc::VideoSinkInterface<cricket::VideoFrame>,
|
| @@ -261,9 +261,8 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| webrtc::RtpParameters GetRtpParameters() const;
|
|
|
| void OnFrame(const cricket::VideoFrame& frame) override;
|
| - bool SetCapturer(VideoCapturer* capturer);
|
| - void MuteStream(bool mute);
|
| - bool DisconnectCapturer();
|
| + void SetSource(rtc::VideoSourceInterface<cricket::VideoFrame>* source);
|
| + void DisconnectSource();
|
|
|
| void SetSend(bool send);
|
|
|
| @@ -365,12 +364,12 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| webrtc::Call* const call_;
|
| rtc::VideoSinkWants sink_wants_;
|
| // Counter used for deciding if the video resolution is currently
|
| - // restricted by CPU usage. It is reset if |capturer_| is changed.
|
| + // restricted by CPU usage. It is reset if |source_| is changed.
|
| int cpu_restricted_counter_;
|
| // Total number of times resolution as been requested to be changed due to
|
| // CPU adaptation.
|
| int number_of_cpu_adapt_changes_;
|
| - VideoCapturer* capturer_;
|
| + rtc::VideoSourceInterface<cricket::VideoFrame>* source_;
|
| WebRtcVideoEncoderFactory* const external_encoder_factory_
|
| GUARDED_BY(lock_);
|
|
|
| @@ -393,14 +392,13 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport {
|
| webrtc::kVideoRotation_0;
|
|
|
| bool sending_ GUARDED_BY(lock_);
|
| - bool muted_ GUARDED_BY(lock_);
|
|
|
| // The timestamp of the first frame received
|
| // Used to generate the timestamps of subsequent frames
|
| int64_t first_frame_timestamp_ms_ GUARDED_BY(lock_);
|
|
|
| // The timestamp of the last frame received
|
| - // Used to generate timestamp for the black frame when capturer is removed
|
| + // Used to generate timestamp for the black frame when source is removed
|
| int64_t last_frame_timestamp_ms_ GUARDED_BY(lock_);
|
| };
|
|
|
|
|