Unit test for AudioFrame output from AcmReceiver::GetAudio

webrtc/modules/audio_coding/neteq/sync_buffer.h

Index: webrtc/modules/audio_coding/neteq/sync_buffer.h
diff --git a/webrtc/modules/audio_coding/neteq/sync_buffer.h b/webrtc/modules/audio_coding/neteq/sync_buffer.h
index c3bb4a90690b36ae379320a8a776e92deb78dd63..5eae4bfc464b59d56f1df5a2e44f7e83d4b66241 100644
--- a/webrtc/modules/audio_coding/neteq/sync_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -67,8 +67,7 @@ class SyncBuffer : public AudioMultiVector {
   // 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_|.