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Simplify AudioBuffer::mixed_low_pass_data API

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R=andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6715 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
aluebs@webrtc.org
2014-07-17 08:27:39 +00:00
parent af93fc08a1
commit 2561d52460
4 changed files with 31 additions and 69 deletions
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8
webrtc/modules/audio_processing/voice_detection_impl.cc
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@ -61,17 +61,11 @@ int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
}
assert(audio->samples_per_split_channel() <= 160);
const int16_t* mixed_data = audio->low_pass_split_data(0);
if (audio->num_channels() > 1) {
audio->CopyAndMixLowPass(1);
mixed_data = audio->mixed_low_pass_data(0);
}
// TODO(ajm): concatenate data in frame buffer here.
int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
apm_->proc_split_sample_rate_hz(),
mixed_data,
audio->mixed_low_pass_data(),
frame_size_samples_);
if (vad_ret == 0) {
stream_has_voice_ = false;