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Set correct sample rate in far_frame in audioproc tool.

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One debug recording with non matching sample rates between render and capture revealed a bug in modules/audio_processing/test/process_test.cc
The far_frame (render audio frame) used was loaded with the capture rate instead of the render rate with a data length mismatch error as result.

BUG=N/A
TESTED=manually on linux
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27169004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7695 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
bjornv@webrtc.org 2014-11-13 11:00:10 +00:00
parent 52bb521b47
commit ee30082af8
1 changed files with 1 additions and 1 deletions
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2
webrtc/modules/audio_processing/test/process_test.cc
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@ -610,7 +610,7 @@ void void_main(int argc, char* argv[]) {
LayoutFromChannels(msg.num_reverse_channels())));
samples_per_channel = msg.sample_rate() / 100;
far_frame.sample_rate_hz_ = msg.sample_rate();
far_frame.sample_rate_hz_ = reverse_sample_rate;
far_frame.samples_per_channel_ = reverse_sample_rate / 100;
far_frame.num_channels_ = msg.num_reverse_channels();
near_frame.sample_rate_hz_ = msg.sample_rate();