Simplify AudioBuffer::mixed_low_pass_data API
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
diff --git a/webrtc/modules/audio_processing/voice_detection_impl.cc b/webrtc/modules/audio_processing/voice_detection_impl.cc
index c6e497f..31336b4 100644
--- a/webrtc/modules/audio_processing/voice_detection_impl.cc
+++ b/webrtc/modules/audio_processing/voice_detection_impl.cc
@@ -61,17 +61,11 @@
}
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;