| Index: webrtc/media/base/mediachannel.h
|
| diff --git a/webrtc/media/base/mediachannel.h b/webrtc/media/base/mediachannel.h
|
| index 3f6c8dda63943fa92aebcd948a7ce4d240aab4fd..8f15878ecd80efcff2fe1c4a438bf8f30a52003e 100644
|
| --- a/webrtc/media/base/mediachannel.h
|
| +++ b/webrtc/media/base/mediachannel.h
|
| @@ -129,7 +129,6 @@ struct AudioOptions {
|
| change.audio_jitter_buffer_fast_accelerate);
|
| SetFrom(&typing_detection, change.typing_detection);
|
| SetFrom(&aecm_generate_comfort_noise, change.aecm_generate_comfort_noise);
|
| - SetFrom(&conference_mode, change.conference_mode);
|
| SetFrom(&adjust_agc_delta, change.adjust_agc_delta);
|
| SetFrom(&experimental_agc, change.experimental_agc);
|
| SetFrom(&extended_filter_aec, change.extended_filter_aec);
|
| @@ -156,7 +155,6 @@ struct AudioOptions {
|
| o.audio_jitter_buffer_fast_accelerate &&
|
| typing_detection == o.typing_detection &&
|
| aecm_generate_comfort_noise == o.aecm_generate_comfort_noise &&
|
| - conference_mode == o.conference_mode &&
|
| experimental_agc == o.experimental_agc &&
|
| extended_filter_aec == o.extended_filter_aec &&
|
| delay_agnostic_aec == o.delay_agnostic_aec &&
|
| @@ -185,7 +183,6 @@ struct AudioOptions {
|
| audio_jitter_buffer_fast_accelerate);
|
| ost << ToStringIfSet("typing", typing_detection);
|
| ost << ToStringIfSet("comfort_noise", aecm_generate_comfort_noise);
|
| - ost << ToStringIfSet("conference", conference_mode);
|
| ost << ToStringIfSet("agc_delta", adjust_agc_delta);
|
| ost << ToStringIfSet("experimental_agc", experimental_agc);
|
| ost << ToStringIfSet("extended_filter_aec", extended_filter_aec);
|
| @@ -221,7 +218,6 @@ struct AudioOptions {
|
| // Audio processing to detect typing.
|
| rtc::Optional<bool> typing_detection;
|
| rtc::Optional<bool> aecm_generate_comfort_noise;
|
| - rtc::Optional<bool> conference_mode;
|
| rtc::Optional<int> adjust_agc_delta;
|
| rtc::Optional<bool> experimental_agc;
|
| rtc::Optional<bool> extended_filter_aec;
|
| @@ -256,14 +252,12 @@ struct AudioOptions {
|
| struct VideoOptions {
|
| void SetAll(const VideoOptions& change) {
|
| SetFrom(&video_noise_reduction, change.video_noise_reduction);
|
| - SetFrom(&conference_mode, change.conference_mode);
|
| SetFrom(&suspend_below_min_bitrate, change.suspend_below_min_bitrate);
|
| SetFrom(&screencast_min_bitrate_kbps, change.screencast_min_bitrate_kbps);
|
| }
|
|
|
| bool operator==(const VideoOptions& o) const {
|
| return video_noise_reduction == o.video_noise_reduction &&
|
| - conference_mode == o.conference_mode &&
|
| suspend_below_min_bitrate == o.suspend_below_min_bitrate &&
|
| screencast_min_bitrate_kbps == o.screencast_min_bitrate_kbps;
|
| }
|
| @@ -272,7 +266,6 @@ struct VideoOptions {
|
| std::ostringstream ost;
|
| ost << "VideoOptions {";
|
| ost << ToStringIfSet("noise reduction", video_noise_reduction);
|
| - ost << ToStringIfSet("conference mode", conference_mode);
|
| ost << ToStringIfSet("suspend below min bitrate",
|
| suspend_below_min_bitrate);
|
| ost << ToStringIfSet("screencast min bitrate kbps",
|
| @@ -285,13 +278,6 @@ struct VideoOptions {
|
| // constraint 'googNoiseReduction', and WebRtcVideoEngine2 passes it
|
| // on to the codec options. Disabled by default.
|
| rtc::Optional<bool> video_noise_reduction;
|
| - // Use conference mode? This flag comes from the remote
|
| - // description's SDP line 'a=x-google-flag:conference', copied over
|
| - // by VideoChannel::SetRemoteContent_w, and ultimately used by
|
| - // conference mode screencast logic in
|
| - // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
|
| - // The special screencast behaviour is disabled by default.
|
| - rtc::Optional<bool> conference_mode;
|
| // Enable WebRTC suspension of video. No video frames will be sent
|
| // when the bitrate is below the configured minimum bitrate. This
|
| // flag comes from the PeerConnection constraint
|
| @@ -946,6 +932,13 @@ class VoiceMediaChannel : public MediaChannel {
|
| };
|
|
|
| struct VideoSendParameters : RtpSendParameters<VideoCodec, VideoOptions> {
|
| + // Use conference mode? This flag comes from the remote
|
| + // description's SDP line 'a=x-google-flag:conference', copied over
|
| + // by VideoChannel::SetRemoteContent_w, and ultimately used by
|
| + // conference mode screencast logic in
|
| + // WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig.
|
| + // The special screencast behaviour is disabled by default.
|
| + bool conference_mode = false;
|
| };
|
|
|
| struct VideoRecvParameters : RtpParameters<VideoCodec> {
|
|
|