Index: webrtc/media/engine/webrtcvideoengine2.h |
diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h |
index 27eadb299ae69160ba7d9a41ca38c74463f4b691..5ae0c71bf135a84305b647b7a6df96c75946e012 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; |
@@ -207,8 +209,6 @@ 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); |
@@ -225,7 +225,7 @@ 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>, |
@@ -250,9 +250,8 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport { |
bool SetRtpParameters(const webrtc::RtpParameters& parameters); |
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 Start(); |
void Stop(); |
@@ -353,12 +352,12 @@ 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_); |
@@ -376,19 +375,21 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport { |
bool pending_encoder_reconfiguration_ GUARDED_BY(lock_); |
VideoEncoderSettings encoder_settings_ GUARDED_BY(lock_); |
AllocatedEncoder allocated_encoder_ GUARDED_BY(lock_); |
+ // Original frame dimension, before adaptation due to cpu or |
+ // network bottlenecks. |
+ Dimensions input_dimensions_ GUARDED_BY(lock_); |
Dimensions last_dimensions_ GUARDED_BY(lock_); |
webrtc::VideoRotation last_rotation_ GUARDED_BY(lock_) = |
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_); |
}; |