AudioCodingModule: Specify decoders using SdpAudioFormat

webrtc/modules/audio_coding/include/audio_coding_module.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 493 matching lines @@
 
   // Registers an external decoder. The name is only used to provide information
   // back to the caller about the decoder. Hence, the name is arbitrary, and may
   // be empty.
   virtual int RegisterExternalReceiveCodec(int rtp_payload_type,
                                            AudioDecoder* external_decoder,
                                            int sample_rate_hz,
                                            int num_channels,
                                            const std::string& name) = 0;
 
+  // Registers a decoder for the given payload type. Returns true iff

[ossu, 2016/09/28 14:12:28]

Place this one directly under the other RegisterReceiveCodec.

[kwiberg-webrtc, 2016/09/29 12:38:34]

Done.

+  // successful.
+  virtual bool RegisterReceiveCodec(int rtp_payload_type,
+                                    const SdpAudioFormat& audio_format) = 0;
+
   ///////////////////////////////////////////////////////////////////////////
   // int32_t UnregisterReceiveCodec()
   // Unregister the codec currently registered with a specific payload type
   // from the list of possible receive codecs.
   //
   // Input:
   //   -payload_type        : The number representing the payload type to
   //                         unregister.
   //
   // Output:
@@ skipping 270 matching lines @@
   virtual std::vector<uint16_t> GetNackList(
       int64_t round_trip_time_ms) const = 0;
 
   virtual void GetDecodingCallStatistics(
       AudioDecodingCallStats* call_stats) const = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_H_