Pass SdpAudioFormat through Channel, without converting to CodecInst

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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 1354 matching lines @@
                               ACMVADMode& mode,
                               bool& disabledDTX) {
   const auto* params = codec_manager_.GetStackParams();
   enabledVAD = params->use_cng;
   mode = params->vad_mode;
   disabledDTX = !params->use_cng;
   return 0;
 }
 
 int32_t Channel::SetRecPayloadType(const CodecInst& codec) {
+  return SetRecPayloadType(codec.pltype, CodecInstToSdp(codec));
+}
+
+int32_t Channel::SetRecPayloadType(int payload_type,
+                                   const SdpAudioFormat& format) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRecPayloadType()");
 
   if (channel_state_.Get().playing) {
     _engineStatisticsPtr->SetLastError(
         VE_ALREADY_PLAYING, kTraceError,
         "SetRecPayloadType() unable to set PT while playing");
     return -1;
   }
 
-  if (codec.pltype == -1) {
+  const CodecInst codec = [&] {
+    CodecInst c = SdpToCodecInst(payload_type, format);
+
+    // Bug 6986: Emulate an old bug that caused us to always choose to decode
+    // Opus in stereo. To be able to remove this, we first need to fix the
+    // other half of bug 6986, which is about losing the Opus "stereo"
+    // parameter.
+    // TODO(kwiberg): Remove this special case, a.k.a. fix bug 6986.
+    if (STR_CASE_CMP(codec.plname, "opus") == 0) {
+      c.channels = 2;
+    }
+
+    return c;
+  }();
+
+  if (payload_type == -1) {
     // De-register the selected codec (RTP/RTCP module and ACM)
 
     int8_t pltype(-1);
     CodecInst rxCodec = codec;
 
     // Get payload type for the given codec
     rtp_payload_registry_->ReceivePayloadType(rxCodec, &pltype);
     rxCodec.pltype = pltype;
 
     if (rtp_receiver_->DeRegisterReceivePayload(pltype) != 0) {
@@ skipping 17 matching lines @@
     // TODO(kwiberg): Retrying is probably not necessary, since
     // AcmReceiver::AddCodec also retries.
     rtp_receiver_->DeRegisterReceivePayload(codec.pltype);
     if (rtp_receiver_->RegisterReceivePayload(codec) != 0) {
       _engineStatisticsPtr->SetLastError(
           VE_RTP_RTCP_MODULE_ERROR, kTraceError,
           "SetRecPayloadType() RTP/RTCP-module registration failed");
       return -1;
     }
   }
-  if (!audio_coding_->RegisterReceiveCodec(codec.pltype,
-                                           CodecInstToSdp(codec))) {
-    audio_coding_->UnregisterReceiveCodec(codec.pltype);
-    if (!audio_coding_->RegisterReceiveCodec(codec.pltype,
-                                             CodecInstToSdp(codec))) {
+  if (!audio_coding_->RegisterReceiveCodec(payload_type, format)) {
+    audio_coding_->UnregisterReceiveCodec(payload_type);
+    if (!audio_coding_->RegisterReceiveCodec(payload_type, format)) {
       _engineStatisticsPtr->SetLastError(
           VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
           "SetRecPayloadType() ACM registration failed - 1");
       return -1;
     }
   }
   return 0;
 }
 
 int32_t Channel::GetRecPayloadType(CodecInst& codec) {
@@ skipping 1839 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc