WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

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 969 matching lines @@
   // RTCP is enabled by default.
   _rtpRtcpModule->SetRTCPStatus(RtcpMode::kCompound);
   // --- Register all permanent callbacks
   if (audio_coding_->RegisterTransportCallback(this) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_INIT_CHANNEL, kTraceError,
         "Channel::Init() callbacks not registered");
     return -1;
   }
 
-  // --- Register all supported codecs to the receiving side of the
-  // RTP/RTCP module
+  return 0;
+}
 
+void Channel::RegisterLegacyCodecs() {
   CodecInst codec;
   const uint8_t nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
 
   for (int idx = 0; idx < nSupportedCodecs; idx++) {
     // Open up the RTP/RTCP receiver for all supported codecs
     if ((audio_coding_->Codec(idx, &codec) == -1) ||
         (rtp_receiver_->RegisterReceivePayload(codec) == -1)) {
       WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                    "Channel::Init() unable to register %s "
                    "(%d/%d/%" PRIuS "/%d) to RTP/RTCP receiver",
@@ skipping 31 matching lines @@
           !audio_coding_->RegisterReceiveCodec(codec.pltype,
                                                CodecInstToSdp(codec)) ||
           _rtpRtcpModule->RegisterSendPayload(codec) == -1) {
         WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                      "Channel::Init() failed to register CN (%d/%d) "
                      "correctly - 1",
                      codec.pltype, codec.plfreq);
       }
     }
   }
-
-  return 0;
 }
 
 void Channel::Terminate() {
   RTC_DCHECK(construction_thread_.CalledOnValidThread());
   // Must be called on the same thread as Init().
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Terminate");
 
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(NULL);
 
@@ skipping 297 matching lines @@
 int32_t Channel::GetVADStatus(bool& enabledVAD,
                               ACMVADMode& mode,
                               bool& disabledDTX) {
   const auto* params = codec_manager_.GetStackParams();
   enabledVAD = params->use_cng;
   mode = params->vad_mode;
   disabledDTX = !params->use_cng;
   return 0;
 }
 
+void Channel::SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) {
+  rtp_payload_registry_->SetAudioReceivePayloads(codecs);
+  audio_coding_->SetReceiveCodecs(codecs);
+}
+
 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) {
@@ skipping 1656 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