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Fix the number of frames used when interleaving in AudioBuffer::InterleaveTo()

Browse source 
R=henrik.lundin@webrtc.org, peah@webrtc.org
TBR=tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1862553002 .

Cr-Commit-Position: refs/heads/master@{#12249}
This commit is contained in:
Alejandro Luebs 2016-04-05 17:29:19 -07:00
parent faed4ab24b
commit 40cbec5415
3 changed files with 2 additions and 7 deletions
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data/audio_processing/output_data_fixed.pb
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4
webrtc/modules/audio_processing/audio_buffer.cc
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@ -430,10 +430,10 @@ void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) {
}
if (frame->num_channels_ == num_channels_) {
Interleave(data_ptr->ibuf()->channels(), proc_num_frames_, num_channels_,
Interleave(data_ptr->ibuf()->channels(), output_num_frames_, num_channels_,
frame->data_);
} else {
UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], proc_num_frames_,
UpmixMonoToInterleaved(data_ptr->ibuf()->channels()[0], output_num_frames_,
frame->num_channels_, frame->data_);
}
}

5
webrtc/modules/audio_processing/test/audio_processing_unittest.cc
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@ -54,12 +54,7 @@ bool write_ref_data = false;
const google::protobuf::int32 kChannels[] = {1, 2};
const int kSampleRates[] = {8000, 16000, 32000, 48000};
#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
// Android doesn't support 48kHz.
const int kProcessSampleRates[] = {8000, 16000, 32000};
#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
const int kProcessSampleRates[] = {8000, 16000, 32000, 48000};
#endif
enum StreamDirection { kForward = 0, kReverse };