Remove conference_mode flag from AudioOptions and VideoOptions.

For audio, the flag is apparently unused. For video, the flag is moved to
VideoSendParameters, with the intention to keep only per-stream flags in
VideoOptions. The flag is used for the webrtcvideoengine2 logic commented like

  // Conference mode screencast uses 2 temporal layers split at 100kbit.

  // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked
  // on the VideoCodec struct as target and max bitrates, respectively.
  // See eg. webrtc::VP8EncoderImpl::SetRates().

BUG=webrtc:5426

Review URL: https://codereview.webrtc.org/1697163002

Cr-Commit-Position: refs/heads/master@{#11651}

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> {

webrtc/media/base/videoengine_unittest.h

@@ -494,7 +494,7 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_TRUE(SetOneCodec(DefaultCodec()));
     cricket::VideoSendParameters parameters;
     parameters.codecs.push_back(DefaultCodec());
-    parameters.options.conference_mode = rtc::Optional<bool>(true);
+    parameters.conference_mode = true;
     EXPECT_TRUE(channel_->SetSendParameters(parameters));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(channel_->AddRecvStream(
@@ -545,7 +545,7 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_TRUE(SetOneCodec(DefaultCodec()));
     cricket::VideoSendParameters parameters;
     parameters.codecs.push_back(DefaultCodec());
-    parameters.options.conference_mode = rtc::Optional<bool>(true);
+    parameters.conference_mode = true;
     EXPECT_TRUE(channel_->SetSendParameters(parameters));
     EXPECT_TRUE(channel_->AddRecvStream(
         cricket::StreamParams::CreateLegacy(kSsrc)));
@@ -725,7 +725,7 @@ class VideoMediaChannelTest : public testing::Test,
     EXPECT_TRUE(SetDefaultCodec());
     cricket::VideoSendParameters parameters;
     parameters.codecs.push_back(DefaultCodec());
-    parameters.options.conference_mode = rtc::Optional<bool>(true);
+    parameters.conference_mode = true;
     EXPECT_TRUE(channel_->SetSendParameters(parameters));
     EXPECT_TRUE(SetSend(true));
     EXPECT_TRUE(channel_->AddRecvStream(

webrtc/media/engine/webrtcvideoengine2.cc

@@ -733,6 +733,12 @@ bool WebRtcVideoChannel2::GetChangedSendParameters(
         params.max_bandwidth_bps == 0 ? -1 : params.max_bandwidth_bps);
   }
 
+  // Handle conference mode.
+  if (params.conference_mode != send_params_.conference_mode) {
+    changed_params->conference_mode =
+        rtc::Optional<bool>(params.conference_mode);
+  }
+
   // Handle options.
   // TODO(pbos): Require VideoSendParameters to contain a full set of options
   // and check if params.options != options_ instead of applying a delta.
@@ -1503,6 +1509,7 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
       first_frame_timestamp_ms_(0),
       last_frame_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
+  parameters_.conference_mode = send_params.conference_mode;
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
@@ -1792,6 +1799,9 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
     parameters_.max_bitrate_bps = *params.max_bandwidth_bps;
     pending_encoder_reconfiguration_ = true;
   }
+  if (params.conference_mode) {
+    parameters_.conference_mode = *params.conference_mode;
+  }
   // Set codecs and options.
   if (params.codec) {
     SetCodecAndOptions(*params.codec,
@@ -1806,6 +1816,10 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::SetSendParameters(
       parameters_.options = *params.options;
     }
   }
+  else if (params.conference_mode && parameters_.codec_settings) {
+    SetCodecAndOptions(*parameters_.codec_settings, parameters_.options);
+    return;
+  }
   if (recreate_stream) {
     LOG(LS_INFO) << "RecreateWebRtcStream (send) because of SetSendParameters";
     RecreateWebRtcStream();
@@ -1855,8 +1869,8 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::CreateVideoEncoderConfig(
                          parameters_.max_bitrate_bps, stream_count);
 
   // Conference mode screencast uses 2 temporal layers split at 100kbit.
-  if (parameters_.options.conference_mode.value_or(false) &&
-      dimensions.is_screencast && encoder_config.streams.size() == 1) {
+  if (parameters_.conference_mode && dimensions.is_screencast &&
+      encoder_config.streams.size() == 1) {
     ScreenshareLayerConfig config = ScreenshareLayerConfig::GetDefault();
 
     // For screenshare in conference mode, tl0 and tl1 bitrates are piggybacked

webrtc/media/engine/webrtcvideoengine2.h

@@ -192,6 +192,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel,
     rtc::Optional<VideoCodecSettings> codec;
     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;
   };
@@ -273,6 +274,7 @@ class WebRtcVideoChannel2 : public VideoMediaChannel,
       webrtc::VideoSendStream::Config config;
       VideoOptions options;
       int max_bitrate_bps;
+      bool conference_mode;
       rtc::Optional<VideoCodecSettings> codec_settings;
       // Sent resolutions + bitrates etc. by the underlying VideoSendStream,
       // typically changes when setting a new resolution or reconfiguring

webrtc/media/engine/webrtcvideoengine2_unittest.cc

@@ -1069,7 +1069,7 @@ TEST_F(WebRtcVideoChannel2Test, RecvStreamWithSimAndRtx) {
   parameters.codecs = engine_.codecs();
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
   EXPECT_TRUE(channel_->SetSend(true));
-  parameters.options.conference_mode = rtc::Optional<bool>(true);
+  parameters.conference_mode = true;
   EXPECT_TRUE(channel_->SetSendParameters(parameters));
 
   // Send side.
@@ -1513,7 +1513,7 @@ TEST_F(WebRtcVideoChannel2Test,
        ConferenceModeScreencastConfiguresTemporalLayer) {
   static const int kConferenceScreencastTemporalBitrateBps =
       ScreenshareLayerConfig::GetDefault().tl0_bitrate_kbps * 1000;
-  send_parameters_.options.conference_mode = rtc::Optional<bool>(true);
+  send_parameters_.conference_mode = true;
   channel_->SetSendParameters(send_parameters_);
 
   AddSendStream();

webrtc/pc/channel.cc

@@ -1798,7 +1798,7 @@ bool VideoChannel::SetRemoteContent_w(const MediaContentDescription* content,
   VideoSendParameters send_params = last_send_params_;
   RtpSendParametersFromMediaDescription(video, &send_params);
   if (video->conference_mode()) {
-    send_params.options.conference_mode = rtc::Optional<bool>(true);
+    send_params.conference_mode = true;
   }
   if (!media_channel()->SetSendParameters(send_params)) {
     SafeSetError("Failed to set remote video description send parameters.",