NetEq: Add codec name and RTP timestamp rate to DecoderInfo

webrtc/modules/audio_coding/neteq/include/neteq.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 163 matching lines @@
                        size_t* samples_per_channel, int* num_channels,
                        NetEqOutputType* type) = 0;
 
   // Associates |rtp_payload_type| with |codec| and stores the information in
   // the codec database. Returns 0 on success, -1 on failure.
   virtual int RegisterPayloadType(NetEqDecoder codec,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
   // decoder database. The decoder implements a decoder of type |codec| and
-  // associates it with |rtp_payload_type|. The decoder will produce samples
-  // at the rate |sample_rate_hz|. Returns kOK on success, kFail on failure.
+  // associates it with |rtp_payload_type| and |codec_name|. The decoder will
+  // produce samples at the rate |sample_rate_hz|. Returns kOK on success, kFail
+  // on failure.

[kwiberg-webrtc, 2015/12/02 09:50:42]

You don't explain what the name is used for. Will that be in a future CL?

[hlundin-webrtc, 2015/12/02 16:19:24]

Done.

   virtual int RegisterExternalDecoder(AudioDecoder* decoder,
                                       NetEqDecoder codec,
+                                      const std::string& codec_name,
                                       uint8_t rtp_payload_type,
                                       int sample_rate_hz) = 0;
 
   // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
   // -1 on failure.
   virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
 
   // Sets a minimum delay in millisecond for packet buffer. The minimum is
   // maintained unless a higher latency is dictated by channel condition.
   // Returns true if the minimum is successfully applied, otherwise false is
@@ skipping 94 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_