Pass SdpAudioFormat through Channel, without converting to CodecInst

webrtc/voice_engine/channel.h

 /*
  *  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 177 matching lines @@
   int32_t DeRegisterVoiceEngineObserver();
 
   // VoECodec
   int32_t GetSendCodec(CodecInst& codec);
   int32_t GetRecCodec(CodecInst& codec);
   int32_t SetSendCodec(const CodecInst& codec);
   void SetBitRate(int bitrate_bps);
   int32_t SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX);
   int32_t GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX);
   int32_t SetRecPayloadType(const CodecInst& codec);
+  int32_t SetRecPayloadType(int payload_type, const SdpAudioFormat& format);
   int32_t GetRecPayloadType(CodecInst& codec);
   int32_t SetSendCNPayloadType(int type, PayloadFrequencies frequency);
   int SetOpusMaxPlaybackRate(int frequency_hz);
   int SetOpusDtx(bool enable_dtx);
   int GetOpusDtx(bool* enabled);
   bool EnableAudioNetworkAdaptor(const std::string& config_string);
   void DisableAudioNetworkAdaptor();
   void SetReceiverFrameLengthRange(int min_frame_length_ms,
                                    int max_frame_length_ms);
 
@@ skipping 334 matching lines @@
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_