Pass SdpAudioFormat through Channel, without converting to CodecInst

webrtc/call/audio_receive_stream.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 84 matching lines @@
     // level components.
     // TODO(solenberg): Remove when VoiceEngine channels are created outside
     // of Call.
     int voe_channel_id = -1;
 
     // Identifier for an A/V synchronization group. Empty string to disable.
     // TODO(pbos): Synchronize streams in a sync group, not just one video
     // stream to one audio stream. Tracked by issue webrtc:4762.
     std::string sync_group;
 
-    // Decoders for every payload that we can receive. Call owns the
-    // AudioDecoder instances once the Config is submitted to
-    // Call::CreateReceiveStream().
-    // TODO(solenberg): Use unique_ptr<> once our std lib fully supports C++11.
-    std::map<uint8_t, AudioDecoder*> decoder_map;
+    // Decoder specifications for every payload type that we can receive.
+    std::map<int, SdpAudioFormat> decoder_map;

[the sun, 2016/12/14 08:58:21]

nit: is decoder_map really the right name nowadays? payload_type_map?

[kwiberg-webrtc, 2016/12/14 13:09:36]

It's a payload type -> decoder specification map, so either of those names could
work. In my mind, it's more important that its a map *to* decoders (or decoder
specs) than that it's a map *from* payload types, so I'm inclined to leave the
name as is.

If you feel strongly about it, I can change it.

 
     rtc::scoped_refptr<AudioDecoderFactory> decoder_factory;
   };
 
   // Starts stream activity.
   // When a stream is active, it can receive, process and deliver packets.
   virtual void Start() = 0;
   // Stops stream activity.
   // When a stream is stopped, it can't receive, process or deliver packets.
   virtual void Stop() = 0;
@@ skipping 13 matching lines @@
   // Sets playback gain of the stream, applied when mixing, and thus after it
   // is potentially forwarded to any attached AudioSinkInterface implementation.
   virtual void SetGain(float gain) = 0;
 
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_