Hooked up Opus CBR support when configured manually or through an SdpAudioFormat.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 238 matching lines @@
 AudioEncoderOpus::Config AudioEncoderOpus::CreateConfig(
     int payload_type,
     const SdpAudioFormat& format) {
   AudioEncoderOpus::Config config;
 
   config.num_channels = GetChannelCount(format);
   config.frame_size_ms = GetFrameSizeMs(format);
   config.max_playback_rate_hz = GetMaxPlaybackRate(format);
   config.fec_enabled = (GetFormatParameter(format, "useinbandfec") == "1");
   config.dtx_enabled = (GetFormatParameter(format, "usedtx") == "1");
+  config.cbr_enabled = (GetFormatParameter(format, "cbr") == "1");
   config.bitrate_bps = rtc::Optional<int>(
       CalculateBitrate(config.max_playback_rate_hz, config.num_channels,
                        GetFormatParameter(format, "maxaveragebitrate")));
   config.payload_type = payload_type;
   config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
                                                 : AudioEncoderOpus::kAudio;
 #if WEBRTC_OPUS_VARIABLE_COMPLEXITY
   config.low_rate_complexity = 9;
 #endif
 
@@ skipping 377 matching lines @@
   complexity_ = config.GetNewComplexity().value_or(config.complexity);
   RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
   if (config.dtx_enabled) {
     RTC_CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
   } else {
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
-
+  if (config.cbr_enabled) {
+    RTC_CHECK_EQ(0, WebRtcOpus_EnableCbr(inst_));
+  } else {
+    RTC_CHECK_EQ(0, WebRtcOpus_DisableCbr(inst_));
+  }
   num_channels_to_encode_ = NumChannels();
   next_frame_length_ms_ = config_.frame_size_ms;
   return true;
 }
 
 void AudioEncoderOpus::SetFrameLength(int frame_length_ms) {
   next_frame_length_ms_ = frame_length_ms;
 }
 
 void AudioEncoderOpus::SetNumChannelsToEncode(size_t num_channels_to_encode) {
@@ skipping 70 matching lines @@
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc