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> { |