WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/modules/audio_coding/acm2/audio_coding_module.cc

Index: webrtc/modules/audio_coding/acm2/audio_coding_module.cc
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
index daeea3577e23a854d26f9b702c6d84f1c5ec989d..e30174b74b66134b597757e9da1e34a8745ff2e8 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module.cc
@@ -121,6 +121,8 @@ class AudioCodingModuleImpl final : public AudioCodingModule {
   // Get current playout frequency.
   int PlayoutFrequency() const override;
 
+  void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
+
   bool RegisterReceiveCodec(int rtp_payload_type,
                             const SdpAudioFormat& audio_format) override;
 
@@ -987,6 +989,12 @@ int AudioCodingModuleImpl::PlayoutFrequency() const {
   return receiver_.last_output_sample_rate_hz();
 }
 
+void AudioCodingModuleImpl::SetReceiveCodecs(
+    const std::map<int, SdpAudioFormat>& codecs) {
+  rtc::CritScope lock(&acm_crit_sect_);
+  receiver_.SetCodecs(codecs);
+}
+
 bool AudioCodingModuleImpl::RegisterReceiveCodec(
     int rtp_payload_type,
     const SdpAudioFormat& audio_format) {