AcmReceiver: Look up last decoder in NetEq's table of decoders

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 234 matching lines @@
 
   // Returns the RTP timestamp for the last sample delivered by GetAudio().
   // The return value will be empty if no valid timestamp is available.
   virtual rtc::Optional<uint32_t> GetPlayoutTimestamp() const = 0;
 
   // Returns the sample rate in Hz of the audio produced in the last GetAudio
   // call. If GetAudio has not been called yet, the configured sample rate
   // (Config::sample_rate_hz) is returned.
   virtual int last_output_sample_rate_hz() const = 0;
 
+  virtual rtc::Optional<CodecInst> GetDecoder(int payload_type) const = 0;

[hlundin-webrtc, 2016/09/19 11:37:36]

Short comment, please. What if not found?

[kwiberg-webrtc, 2016/09/19 12:18:34]

Done.

+
   // Not implemented.
   virtual int SetTargetNumberOfChannels() = 0;
 
   // Not implemented.
   virtual int SetTargetSampleRate() = 0;
 
   // Returns the error code for the last occurred error. If no error has
   // occurred, 0 is returned.
   virtual int LastError() const = 0;
 
@@ skipping 23 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_