NetEq: Add functionality to assist with delay analysis and tooling

webrtc/modules/audio_coding/neteq/sync_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_SYNC_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class SyncBuffer : public AudioMultiVector {
  public:
   SyncBuffer(size_t channels, size_t length)
       : AudioMultiVector(channels, length),
         next_index_(length),
         end_timestamp_(0),
         dtmf_index_(0) {}
 
-  // Returns the number of samples yet to play out form the buffer.
+  // Returns the number of samples yet to play out from the buffer.

[hlundin-webrtc, 2017/04/26 11:41:31]

Totally unrelated typo fix.

   size_t FutureLength() const;
 
   // Adds the contents of |append_this| to the back of the SyncBuffer. Removes
   // the same number of samples from the beginning of the SyncBuffer, to
   // maintain a constant buffer size. The |next_index_| is updated to reflect
   // the move of the beginning of "future" data.
   void PushBack(const AudioMultiVector& append_this) override;
 
   // Adds |length| zeros to the beginning of each channel. Removes
   // the same number of samples from the end of the SyncBuffer, to
@@ skipping 52 matching lines @@
  private:
   size_t next_index_;
   uint32_t end_timestamp_;  // The timestamp of the last sample in the buffer.
   size_t dtmf_index_;  // Index to the first non-DTMF sample in the buffer.
 
   RTC_DISALLOW_COPY_AND_ASSIGN(SyncBuffer);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_SYNC_BUFFER_H_