Chromium Code Reviews| Index: talk/media/webrtc/webrtcvideoengine2.cc |
| diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc |
| index f237b8fde66787c2653cac6fac1d3bf4384eea63..e4e80e2722b4ada6efee5c6249d29996d9198fd0 100644 |
| --- a/talk/media/webrtc/webrtcvideoengine2.cc |
| +++ b/talk/media/webrtc/webrtcvideoengine2.cc |
| @@ -929,6 +929,15 @@ bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable, |
| } |
| } |
| +rtc::VideoSinkInterface<cricket::VideoFrame>* WebRtcVideoChannel2::GetSink( |
| + uint32_t ssrc) { |
| + rtc::CritScope stream_lock(&stream_crit_); |
| + if (send_streams_.find(ssrc) == send_streams_.end()) { |
| + return NULL; |
| + } |
| + return send_streams_[ssrc]; |
| +} |
| + |
| bool WebRtcVideoChannel2::ValidateSendSsrcAvailability( |
| const StreamParams& sp) const { |
| for (uint32_t ssrc: sp.ssrcs) { |
| @@ -1640,12 +1649,11 @@ static void CreateBlackFrame(webrtc::VideoFrame* video_frame, |
| video_frame->allocated_size(webrtc::kVPlane)); |
| } |
| -void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( |
| - VideoCapturer* capturer, |
| - const VideoFrame* frame) { |
| - TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::InputFrame"); |
| - webrtc::VideoFrame video_frame(frame->GetVideoFrameBuffer(), 0, 0, |
| - frame->GetVideoRotation()); |
| +void WebRtcVideoChannel2::WebRtcVideoSendStream::OnFrame( |
| + const cricket::VideoFrame& frame) { |
| + TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::OnFrame"); |
| + webrtc::VideoFrame video_frame(frame.GetVideoFrameBuffer(), 0, 0, |
| + frame.GetVideoRotation()); |
| rtc::CritScope cs(&lock_); |
| if (stream_ == NULL) { |
| // Frame input before send codecs are configured, dropping frame. |
| @@ -1662,15 +1670,9 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( |
| LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame."; |
| return; |
| } |
| - if (muted_) { |
| - // Create a black frame to transmit instead. |
| - CreateBlackFrame(&video_frame, |
| - static_cast<int>(frame->GetWidth()), |
| - static_cast<int>(frame->GetHeight())); |
| - } |
| - int64_t frame_delta_ms = frame->GetTimeStamp() / rtc::kNumNanosecsPerMillisec; |
| - // frame->GetTimeStamp() is essentially a delta, align to webrtc time |
| + int64_t frame_delta_ms = frame.GetTimeStamp() / rtc::kNumNanosecsPerMillisec; |
| + // frame.GetTimeStamp() is essentially a delta, align to webrtc time |
| if (first_frame_timestamp_ms_ == 0) { |
| first_frame_timestamp_ms_ = rtc::Time() - frame_delta_ms; |
| } |
| @@ -1679,7 +1681,8 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( |
| video_frame.set_render_time_ms(last_frame_timestamp_ms_); |
| // Reconfigure codec if necessary. |
| SetDimensions( |
| - video_frame.width(), video_frame.height(), capturer->IsScreencast()); |
| + // TODO(nisse): Get screencast flag from video options. |
|
perkj_webrtc
2016/01/26 15:40:56
capturer_->IsScreencast() for the time beeing. cas
nisse-webrtc
2016/01/28 08:24:54
I'm adding an is_screencast_ member, set by SetCap
|
| + video_frame.width(), video_frame.height(), false /* is_screencast */); |
| stream_->Input()->IncomingCapturedFrame(video_frame); |
| } |
| @@ -1723,9 +1726,6 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer( |
| capturer_ = capturer; |
| } |
| - // Lock cannot be held while connecting the capturer to prevent lock-order |
| - // violations. |
| - capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame); |
| return true; |
| } |
| @@ -1772,7 +1772,6 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() { |
| capturer = capturer_; |
| capturer_ = NULL; |
| } |
| - capturer->SignalVideoFrame.disconnect(this); |
| return true; |
| } |