Chromium Code Reviews| Index: webrtc/media/engine/webrtcvideoengine2.h |
| diff --git a/webrtc/media/engine/webrtcvideoengine2.h b/webrtc/media/engine/webrtcvideoengine2.h |
| index ccf753e9fc663bb6cec50bc7ea364b08df80adbf..6f1ddfbcf74be56fe6550e9b36aa89358006e929 100644 |
| --- a/webrtc/media/engine/webrtcvideoengine2.h |
| +++ b/webrtc/media/engine/webrtcvideoengine2.h |
| @@ -190,7 +190,6 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport { |
| rtc::Optional<std::vector<webrtc::RtpExtension>> rtp_header_extensions; |
| rtc::Optional<int> max_bandwidth_bps; |
| rtc::Optional<bool> conference_mode; |
| - rtc::Optional<VideoOptions> options; |
| rtc::Optional<webrtc::RtcpMode> rtcp_mode; |
| }; |
| @@ -234,6 +233,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport { |
| webrtc::Call* call, |
| const StreamParams& sp, |
| const webrtc::VideoSendStream::Config& config, |
| + const VideoOptions& options, |
| WebRtcVideoEncoderFactory* external_encoder_factory, |
| bool enable_cpu_overuse_detection, |
| int max_bitrate_bps, |
| @@ -516,6 +516,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel, public webrtc::Transport { |
| // TODO(deadbeef): Don't duplicate information between |
| // send_params/recv_params, rtp_extensions, options, etc. |
| VideoSendParameters send_params_; |
| + VideoOptions send_default_options_; |
|
pthatcher1
2016/03/05 01:49:14
default_send_options_
|
| VideoRecvParameters recv_params_; |
| }; |