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/include/neteq.h b/webrtc/modules/audio_coding/neteq/include/neteq.h
index d53551f..5e06d48 100644
--- a/webrtc/modules/audio_coding/neteq/include/neteq.h
+++ b/webrtc/modules/audio_coding/neteq/include/neteq.h
@@ -156,8 +156,8 @@
uint32_t receive_timestamp) = 0;
// Instructs NetEq to deliver 10 ms of audio data. The data is written to
- // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |interleaved_|,
- // |num_channels_|, |samples_per_channel_|, |speech_type_|, and
+ // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |speech_type_|,
+ // |num_channels_|, |sample_rate_hz_|, |samples_per_channel_|, and
// |vad_activity_| are updated upon success. If an error is returned, some
// fields may not have been updated.
// Returns kOK on success, or kFail in case of an error.
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_|.