commit 47f38b129afae382d03a676777d7f9d5fc8b84f5
author bjornv@webrtc.org <bjornv@webrtc.org> Thu Nov 13 11:00:10 2014
committer bjornv@webrtc.org <bjornv@webrtc.org> Thu Nov 13 11:00:10 2014
tree d03b1628e07664f2c620ee5a55de95ee80be5274
parent 1135e2c4fd456d1bb6f15e5cf7ad621ba5eb0de5

Set correct sample rate in far_frame in audioproc tool.

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/webrtc@7695 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/audio_processing/test/process_test.cc

diff --git a/modules/audio_processing/test/process_test.cc b/modules/audio_processing/test/process_test.cc
index 76a916d..469ecf9 100644
--- a/modules/audio_processing/test/process_test.cc
+++ b/modules/audio_processing/test/process_test.cc
@@ -610,7 +610,7 @@
                               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();