Adding 120 ms frame length support in NetEq.

webrtc/modules/audio_coding/neteq/merge.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 19 matching lines @@
 // end of the expanded data (thanks to how the Expand class operates). However,
 // if a later packet arrives instead, the loss is a fact, and the new data must
 // be stitched together with the end of the expanded data. This stitching is
 // what the Merge class does.
 class Merge {
  public:
   Merge(int fs_hz,
         size_t num_channels,
         Expand* expand,
         SyncBuffer* sync_buffer);
-  virtual ~Merge() {}
+  virtual ~Merge();
 
   // The main method to produce the audio data. The decoded data is supplied in
   // |input|, having |input_length| samples in total for all channels
   // (interleaved). The result is written to |output|. The number of channels
   // allocated in |output| defines the number of channels that will be used when
   // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
   // will be used to scale the audio, and is updated in the process. The array
   // must have |num_channels_| elements.
   virtual size_t Process(int16_t* input, size_t input_length,
                          int16_t* external_mute_factor_array,
@@ skipping 35 matching lines @@
   size_t CorrelateAndPeakSearch(size_t start_position, size_t input_length,
                                 size_t expand_period) const;
 
   const int fs_mult_;  // fs_hz_ / 8000.
   const size_t timestamps_per_call_;
   Expand* expand_;
   SyncBuffer* sync_buffer_;
   int16_t expanded_downsampled_[kExpandDownsampLength];
   int16_t input_downsampled_[kInputDownsampLength];
   AudioMultiVector expanded_;
+  std::vector<int16_t> temp_data_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_