| Index: content/renderer/media/webrtc_audio_capturer.cc
 | 
| ===================================================================
 | 
| --- content/renderer/media/webrtc_audio_capturer.cc	(revision 181072)
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| +++ content/renderer/media/webrtc_audio_capturer.cc	(working copy)
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| @@ -101,7 +101,7 @@
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|    int buffer_size = GetBufferSizeForSampleRate(sample_rate);
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|  
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|    // Configure audio parameters for the default source.
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| -  params_.Reset(format, channel_layout, sample_rate, 16, buffer_size);
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| +  params_.Reset(format, channel_layout, 0, sample_rate, 16, buffer_size);
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|  
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|    // Tell all sinks which format we use.
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|    for (SinkList::const_iterator it = sinks_.begin();
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| @@ -194,6 +194,7 @@
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|  
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|      params_.Reset(params_.format(),
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|                    channel_layout,
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| +                  0,
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|                    sample_rate,
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|                    16,  // ignored since the audio stack uses float32.
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|                    buffer_size);
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| 
 |