WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType

webrtc/voice_engine/channel_proxy.h

 /*
  *  Copyright (c) 2015 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 64 matching lines @@
   virtual AudioDecodingCallStats GetDecodingCallStatistics() const;
   virtual int GetSpeechOutputLevel() const;
   virtual int GetSpeechOutputLevelFullRange() const;
   virtual uint32_t GetDelayEstimate() const;
   virtual bool SetSendTelephoneEventPayloadType(int payload_type,
                                                 int payload_frequency);
   virtual bool SendTelephoneEventOutband(int event, int duration_ms);
   virtual void SetBitrate(int bitrate_bps, int64_t probing_interval_ms);
   virtual void SetRecPayloadType(int payload_type,
                                  const SdpAudioFormat& format);
+  virtual void SetReceiveCodecs(const std::map<int, SdpAudioFormat>& codecs);
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
   virtual void SetInputMute(bool muted);
   virtual void RegisterExternalTransport(Transport* transport);
   virtual void DeRegisterExternalTransport();
   virtual void OnRtpPacket(const RtpPacketReceived& packet);
   virtual bool ReceivedRTCPPacket(const uint8_t* packet, size_t length);
   virtual const rtc::scoped_refptr<AudioDecoderFactory>&
       GetAudioDecoderFactory() const;
   virtual void SetChannelOutputVolumeScaling(float scaling);
   virtual void SetRtcEventLog(RtcEventLog* event_log);
@@ skipping 17 matching lines @@
   virtual bool GetSendCodec(CodecInst* codec_inst) const;
   virtual bool SetVADStatus(bool enable);
   virtual bool SetCodecFECStatus(bool enable);
   virtual bool SetOpusDtx(bool enable);
   virtual bool SetOpusMaxPlaybackRate(int frequency_hz);
   virtual bool SetSendCodec(const CodecInst& codec_inst);
   virtual bool SetSendCNPayloadType(int type, PayloadFrequencies frequency);
   virtual void OnTwccBasedUplinkPacketLossRate(float packet_loss_rate);
   virtual void OnRecoverableUplinkPacketLossRate(
       float recoverable_packet_loss_rate);
+  virtual void RegisterLegacyCodecs();
 
  private:
   Channel* channel() const;
 
   // Thread checkers document and lock usage of some methods on voe::Channel to
   // specific threads we know about. The goal is to eventually split up
   // voe::Channel into parts with single-threaded semantics, and thereby reduce
   // the need for locks.
   rtc::ThreadChecker worker_thread_checker_;
   rtc::ThreadChecker module_process_thread_checker_;
   // Methods accessed from audio and video threads are checked for sequential-
   // only access. We don't necessarily own and control these threads, so thread
   // checkers cannot be used. E.g. Chromium may transfer "ownership" from one
   // audio thread to another, but access is still sequential.
   rtc::RaceChecker audio_thread_race_checker_;
   rtc::RaceChecker video_capture_thread_race_checker_;
   ChannelOwner channel_owner_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(ChannelProxy);
 };
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_PROXY_H_