Unit test for AudioFrame output from AcmReceiver::GetAudio

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.
  */
 
@@ skipping 49 matching lines @@
                               size_t position);
 
   // Same as the above method, but where all of |insert_this| is written (with
   // the same constraints as above, that the SyncBuffer is not extended).
   virtual void ReplaceAtIndex(const AudioMultiVector& insert_this,
                               size_t position);
 
   // Reads |requested_len| samples from each channel and writes them interleaved
   // into |output|. The |next_index_| is updated to point to the sample to read
   // next time. The AudioFrame |output| is first reset, and the |data_|,
-  // |interleaved_|, |num_channels_|, and |samples_per_channel_| fields are
-  // updated.
+  // |num_channels_|, and |samples_per_channel_| fields are updated.
   void GetNextAudioInterleaved(size_t requested_len, AudioFrame* output);
 
   // Adds |increment| to |end_timestamp_|.
   void IncreaseEndTimestamp(uint32_t increment);
 
   // Flushes the buffer. The buffer will contain only zeros after the flush, and
   // |next_index_| will point to the end, like when the buffer was first
   // created.
   void Flush();
 
@@ skipping 11 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_