Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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 1954 matching lines @@
         dtmf_payload_freq_ = codec.clockrate;
       }
     }
   }
 
   // Scan through the list to figure out the codec to use for sending, along
   // with the proper configuration for VAD, CNG, NACK and Opus-specific
   // parameters.
   // TODO(solenberg): Refactor this logic once we create AudioEncoders here.
   webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec;
-  {
+  do {
     send_codec_spec.nack_enabled = send_codec_spec_.nack_enabled;
 
     // Find send codec (the first non-telephone-event/CN codec).
     const AudioCodec* codec = WebRtcVoiceCodecs::GetPreferredCodec(
         codecs, &send_codec_spec.codec_inst);
     if (!codec) {
       LOG(LS_WARNING) << "Received empty list of codecs.";
-      return false;
+      break;
     }
 
     send_codec_spec.transport_cc_enabled = HasTransportCc(*codec);
     send_codec_spec.nack_enabled = HasNack(*codec);
     bitrate_config_ = GetBitrateConfigForCodec(*codec);
 
     // For Opus as the send codec, we are to determine inband FEC, maximum
     // playback rate, and opus internal dtx.
     if (IsCodec(*codec, kOpusCodecName)) {
       GetOpusConfig(*codec, &send_codec_spec.codec_inst,
@@ skipping 49 matching lines @@
     }
 
     // Find the telephone-event PT exactly matching the preferred send codec.
     for (const AudioCodec& dtmf_codec : dtmf_codecs) {
       if (dtmf_codec.clockrate == codec->clockrate) {
         dtmf_payload_type_ = rtc::Optional<int>(dtmf_codec.id);
         dtmf_payload_freq_ = dtmf_codec.clockrate;
         break;
       }
     }
-  }
+  } while (0);
 
   if (send_codec_spec_ != send_codec_spec) {
     send_codec_spec_ = std::move(send_codec_spec);
     // Apply new settings to all streams.
     for (const auto& kv : send_streams_) {
       kv.second->RecreateAudioSendStream(send_codec_spec_);
     }
   } else {
     // If the codec isn't changing, set the start bitrate to -1 which means
     // "unchanged" so that BWE isn't affected.
@@ skipping 645 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE