NetEq: Use TickTimer in PacketBuffer

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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_PACKET_BUFFER_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/packet.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
-// Forward declaration.
 class DecoderDatabase;
+class TickTimer;
 
 // This is the actual buffer holding the packets before decoding.
 class PacketBuffer {
  public:
   enum BufferReturnCodes {
     kOK = 0,
     kFlushed,
     kNotFound,
     kBufferEmpty,
     kInvalidPacket,
     kInvalidPointer
   };
 
   // Constructor creates a buffer which can hold a maximum of
   // |max_number_of_packets| packets.
-  PacketBuffer(size_t max_number_of_packets);
+  PacketBuffer(size_t max_number_of_packets, const TickTimer* tick_timer);
 
   // Deletes all packets in the buffer before destroying the buffer.
   virtual ~PacketBuffer();
 
   // Flushes the buffer and deletes all packets in it.
   virtual void Flush();
 
   // Returns true for an empty buffer.
   virtual bool Empty() const;
 
@@ skipping 61 matching lines @@
 
   // 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;
 
-  // Increase the waiting time counter for every packet in the buffer by |inc|.
-  // The default value for |inc| is 1.
-  virtual void IncrementWaitingTimes(int inc = 1);
-
   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|.
   static void DeleteAllPackets(PacketList* packet_list);
 
   // Static method returning true if |timestamp| is older than |timestamp_limit|
   // but less than |horizon_samples| behind |timestamp_limit|. For instance,
   // with timestamp_limit = 100 and horizon_samples = 10, a timestamp in the
   // range (90, 100) is considered obsolete, and will yield true.
   // Setting |horizon_samples| to 0 is the same as setting it to 2^31, i.e.,
   // half the 32-bit timestamp range.
   static bool IsObsoleteTimestamp(uint32_t timestamp,
                                   uint32_t timestamp_limit,
                                   uint32_t horizon_samples) {
     return IsNewerTimestamp(timestamp_limit, timestamp) &&
            (horizon_samples == 0 ||
             IsNewerTimestamp(timestamp, timestamp_limit - horizon_samples));
   }
 
  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_