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diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/merge.h
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+/*
+ *  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_MERGE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
+
+#include <assert.h>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Expand;
+class SyncBuffer;
+
+// This class handles the transition from expansion to normal operation.
+// When a packet is not available for decoding when needed, the expand operation
+// is called to generate extrapolation data. If the missing packet arrives,
+// i.e., it was just delayed, it can be decoded and appended directly to the
+// 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)
+      : fs_hz_(fs_hz),
+        num_channels_(num_channels),
+        fs_mult_(fs_hz_ / 8000),
+        timestamps_per_call_(fs_hz_ / 100),
+        expand_(expand),
+        sync_buffer_(sync_buffer),
+        expanded_(num_channels_) {
+    assert(num_channels_ > 0);
+  }
+
+  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 int Process(int16_t* input, size_t input_length,
+                      int16_t* external_mute_factor_array,
+                      AudioMultiVector* output);
+
+  virtual int RequiredFutureSamples();
+
+ protected:
+  const int fs_hz_;
+  const size_t num_channels_;
+
+ private:
+  static const int kMaxSampleRate = 48000;
+  static const int kExpandDownsampLength = 100;
+  static const int kInputDownsampLength = 40;
+  static const int kMaxCorrelationLength = 60;
+
+  // Calls |expand_| to get more expansion data to merge with. The data is
+  // written to |expanded_signal_|. Returns the length of the expanded data,
+  // while |expand_period| will be the number of samples in one expansion period
+  // (typically one pitch period). The value of |old_length| will be the number
+  // of samples that were taken from the |sync_buffer_|.
+  int GetExpandedSignal(int* old_length, int* expand_period);
+
+  // Analyzes |input| and |expanded_signal| to find maximum values. Returns
+  // a muting factor (Q14) to be used on the new data.
+  int16_t SignalScaling(const int16_t* input, int input_length,
+                        const int16_t* expanded_signal,
+                        int16_t* expanded_max, int16_t* input_max) const;
+
+  // Downsamples |input| (|input_length| samples) and |expanded_signal| to
+  // 4 kHz sample rate. The downsampled signals are written to
+  // |input_downsampled_| and |expanded_downsampled_|, respectively.
+  void Downsample(const int16_t* input, int input_length,
+                  const int16_t* expanded_signal, int expanded_length);
+
+  // Calculates cross-correlation between |input_downsampled_| and
+  // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
+  // lag is returned.
+  int16_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
+                                 int start_position, int input_length,
+                                 int expand_period) const;
+
+  const int fs_mult_;  // fs_hz_ / 8000.
+  const int timestamps_per_call_;
+  Expand* expand_;
+  SyncBuffer* sync_buffer_;
+  int16_t expanded_downsampled_[kExpandDownsampLength];
+  int16_t input_downsampled_[kInputDownsampLength];
+  AudioMultiVector expanded_;
+
+  DISALLOW_COPY_AND_ASSIGN(Merge);
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_