Unit test for AudioFrame output from AcmReceiver::GetAudio
This new unit test verifies the parameter fields (not the audio data
itself) written to the AudioFrame output by AcmReceiver::GetAudio.
Also corrected a few comments reflecting recent changes in the code.
BUG=webrtc:5669
Review URL: https://codereview.webrtc.org/1859953002
Cr-Commit-Position: refs/heads/master@{#12253}
diff --git a/webrtc/modules/audio_coding/neteq/sync_buffer.h b/webrtc/modules/audio_coding/neteq/sync_buffer.h
index c3bb4a9..5eae4bf 100644
--- a/webrtc/modules/audio_coding/neteq/sync_buffer.h
+++ b/webrtc/modules/audio_coding/neteq/sync_buffer.h
@@ -67,8 +67,7 @@
// 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_|.