| 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> { | 
|  |