Chromium Code Reviews| Index: talk/media/webrtc/webrtcvideoengine2.cc |
| diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc |
| index 2d5ec5383b3f30ff6ae6e2bb7c9b8ddfd655b102..3b5cb41513c7b5aa021f7f1533287424f076bc63 100644 |
| --- a/talk/media/webrtc/webrtcvideoengine2.cc |
| +++ b/talk/media/webrtc/webrtcvideoengine2.cc |
| @@ -1018,7 +1018,7 @@ bool WebRtcVideoChannel2::SetVideoSend(uint32_t ssrc, bool enable, |
| bool WebRtcVideoChannel2::ValidateSendSsrcAvailability( |
| const StreamParams& sp) const { |
| - for (uint32_t ssrc: sp.ssrcs) { |
| + for (uint32_t ssrc : sp.ssrcs) { |
|
nisse-webrtc
2016/02/03 09:16:34
Unrelated change.
perkj_webrtc
2016/02/08 14:32:01
yes. git cl format, git cl lint.
|
| if (send_ssrcs_.find(ssrc) != send_ssrcs_.end()) { |
| LOG(LS_ERROR) << "Send stream with SSRC '" << ssrc << "' already exists."; |
| return false; |
| @@ -1029,7 +1029,7 @@ bool WebRtcVideoChannel2::ValidateSendSsrcAvailability( |
| bool WebRtcVideoChannel2::ValidateReceiveSsrcAvailability( |
| const StreamParams& sp) const { |
| - for (uint32_t ssrc: sp.ssrcs) { |
| + for (uint32_t ssrc : sp.ssrcs) { |
| if (receive_ssrcs_.find(ssrc) != receive_ssrcs_.end()) { |
| LOG(LS_ERROR) << "Receive stream with SSRC '" << ssrc |
| << "' already exists."; |
| @@ -1335,11 +1335,6 @@ bool WebRtcVideoChannel2::SetCapturer(uint32_t ssrc, VideoCapturer* capturer) { |
| return false; |
| } |
| } |
| - |
| - if (capturer) { |
| - capturer->SetApplyRotation(!ContainsHeaderExtension( |
| - send_rtp_extensions_, kRtpVideoRotationHeaderExtension)); |
| - } |
| { |
| rtc::CritScope lock(&capturer_crit_); |
| capturers_[ssrc] = capturer; |
| @@ -1623,12 +1618,11 @@ 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 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. |
| @@ -1647,12 +1641,11 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( |
| } |
| if (muted_) { |
| // Create a black frame to transmit instead. |
| - CreateBlackFrame(&video_frame, |
| - static_cast<int>(frame->GetWidth()), |
| - static_cast<int>(frame->GetHeight())); |
| + CreateBlackFrame(&video_frame, static_cast<int>(frame.GetWidth()), |
| + static_cast<int>(frame.GetHeight())); |
| } |
| - int64_t frame_delta_ms = frame->GetTimeStamp() / rtc::kNumNanosecsPerMillisec; |
| + 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; |
| @@ -1661,8 +1654,8 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame( |
| last_frame_timestamp_ms_ = first_frame_timestamp_ms_ + frame_delta_ms; |
| video_frame.set_render_time_ms(last_frame_timestamp_ms_); |
| // Reconfigure codec if necessary. |
| - SetDimensions( |
| - video_frame.width(), video_frame.height(), capturer->IsScreencast()); |
| + SetDimensions(video_frame.width(), video_frame.height(), |
| + capturer_->IsScreencast()); |
| stream_->Input()->IncomingCapturedFrame(video_frame); |
| } |
| @@ -1705,10 +1698,8 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer( |
| } |
| capturer_ = capturer; |
| + capturer_->AddSink(this, sink_capabilities_); |
|
pthatcher1
2016/02/03 15:38:36
I think it would be better to add capabilities() t
perkj_webrtc
2016/02/08 14:32:01
PTAL. I think I have now changed this to be as we
|
| } |
| - // Lock cannot be held while connecting the capturer to prevent lock-order |
| - // violations. |
| - capturer->SignalVideoFrame.connect(this, &WebRtcVideoSendStream::InputFrame); |
| return true; |
| } |
| @@ -1756,7 +1747,7 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::DisconnectCapturer() { |
| capturer = capturer_; |
| capturer_ = NULL; |
| } |
| - capturer->SignalVideoFrame.disconnect(this); |
| + capturer->RemoveSink(this); |
| return true; |
| } |
| @@ -1898,9 +1889,11 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters( |
| } |
| if (params.rtp_header_extensions) { |
| parameters_.config.rtp.extensions = *params.rtp_header_extensions; |
| + sink_capabilities_.can_apply_rotation = ContainsHeaderExtension( |
| + *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension); |
| if (capturer_) { |
| - capturer_->SetApplyRotation(!ContainsHeaderExtension( |
| - *params.rtp_header_extensions, kRtpVideoRotationHeaderExtension)); |
| + capturer_->RemoveSink(this); |
| + capturer_->AddSink(this, sink_capabilities_); |
|
pthatcher1
2016/02/03 15:38:36
Why not just fire SignalCapabilitiesChanged()?
perkj_webrtc
2016/02/08 14:32:01
PTAL
|
| } |
| recreate_stream = true; |
| } |