Added a ParsePayload method to AudioDecoder.

webrtc/modules/audio_coding/neteq/packet_buffer.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 97 matching lines @@
 
   // Discards all packets that are (strictly) older than timestamp_limit.
   virtual int DiscardAllOldPackets(uint32_t timestamp_limit);
 
   // Returns the number of packets in the buffer, including duplicates and
   // redundant packets.
   virtual size_t NumPacketsInBuffer() const;
 
   // Returns the number of samples in the buffer, including samples carried in
   // duplicate and redundant packets.
-  virtual size_t NumSamplesInBuffer(DecoderDatabase* decoder_database,
-                                    size_t last_decoded_length) const;
+  virtual size_t NumSamplesInBuffer(size_t last_decoded_length) const;
 
   virtual void BufferStat(int* num_packets, int* max_num_packets) const;
 
   // Static method that properly deletes the first packet, and its payload
   // array, in |packet_list|. Returns false if |packet_list| already was empty,
   // otherwise true.
   static bool DeleteFirstPacket(PacketList* packet_list);
 
   // Static method that properly deletes all packets, and their payload arrays,
   // in |packet_list|.
@@ skipping 15 matching lines @@
 
  private:
   size_t max_number_of_packets_;
   PacketList buffer_;
   const TickTimer* tick_timer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(PacketBuffer);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_