commit 2c9c83d7ecd7251362d438abc70c9f50b208f316
author Andrew MacDonald <andrew@webrtc.org> Mon Mar 30 10:08:22 2015 -0700
committer Andrew MacDonald <andrew@webrtc.org> Mon Mar 30 17:08:28 2015 +0000
tree 0b0d0175075736268ff539c73e209ea0790bca9e
parent 45c6449114ab69cab755410f69567fa7e20f8106

Remove non-functional asynchronous resampling mode.

A few other cleanups, most notably using a sane parameter to specify the
number of channels.

BUG=chromium:469814
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/46729004

Cr-Commit-Position: refs/heads/master@{#8894}

webrtc/modules/utility/source/file_player_impl.cc

diff --git a/webrtc/modules/utility/source/file_player_impl.cc b/webrtc/modules/utility/source/file_player_impl.cc
index 1803e8a..df6a5bf 100644
--- a/webrtc/modules/utility/source/file_player_impl.cc
+++ b/webrtc/modules/utility/source/file_player_impl.cc
@@ -130,7 +130,7 @@
         unresampledAudioFrame.samples_per_channel_ =
             (uint16_t)lengthInBytes >> 1;
 
-    }else {
+    } else {
         // Decode will generate 10 ms of audio data. PlayoutAudioData(..)
         // expects a full frame. If the frame size is larger than 10 ms,
         // PlayoutAudioData(..) data should be called proportionally less often.
@@ -158,7 +158,7 @@
 
     int outLen = 0;
     if(_resampler.ResetIfNeeded(unresampledAudioFrame.sample_rate_hz_,
-                                frequencyInHz, kResamplerSynchronous))
+                                frequencyInHz, 1))
     {
         LOG(LS_WARNING) << "Get10msAudioFromFile() unexpected codec.";