Pass more detailed audio hardware configuration information to the renderer

content/renderer/media/webrtc_audio_capturer.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/webrtc_audio_capturer.h"
 
 #include "base/bind.h"
 #include "base/logging.h"
 #include "base/metrics/histogram.h"
 #include "base/string_util.h"
@@ skipping 68 matching lines @@
     if (!buffer_size) {
       DLOG(ERROR) << "Unsupported sample-rate: " << sample_rate;
       return false;
     }
 
     media::AudioParameters::Format format =
         media::AudioParameters::AUDIO_PCM_LOW_LATENCY;
 
     // bits_per_sample is always 16 for now.
     int bits_per_sample = 16;
-
-    params_.Reset(format, channel_layout, 0, sample_rate, bits_per_sample,
-                  buffer_size);
+    int channels = ChannelLayoutToChannelCount(channel_layout);
+    params_.Reset(format, channel_layout, channels, 0,
+        sample_rate, bits_per_sample, buffer_size);
     buffer_.reset(new int16[params_.frames_per_buffer() * params_.channels()]);
 
     return true;
   }
 
   int16* buffer() const { return buffer_.get(); }
   const media::AudioParameters& params() const { return params_; }
 
  private:
   ~ConfiguredBuffer() {}
@@ skipping 386 matching lines @@
   if (!on_device_stopped_cb_.is_null())
     on_device_stopped_cb_.Run();
 }
 
 media::AudioParameters WebRtcAudioCapturer::audio_parameters() const {
   base::AutoLock auto_lock(lock_);
   return buffer_->params();
 }
 
 }  // namespace content