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Unit test for AudioFrame output from AcmReceiver::GetAudio

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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}
This commit is contained in:
henrik.lundin
2016-04-06 01:03:02 -07:00
committed by Commit bot
parent 6172401972
commit 7dc68897d3
3 changed files with 83 additions and 4 deletions
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3
webrtc/modules/audio_coding/neteq/sync_buffer.h
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@ -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_|.