Introduce AudioHardwareConfig for renderer side audio device info.

content/renderer/media/audio_renderer_mixer_manager.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/audio_renderer_mixer_manager.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "content/renderer/media/audio_device_factory.h"
 #include "content/renderer/media/renderer_audio_output_device.h"
+#include "media/base/audio_hardware_config.h"
 #include "media/base/audio_renderer_mixer.h"
 #include "media/base/audio_renderer_mixer_input.h"
 
 namespace content {
 
-AudioRendererMixerManager::AudioRendererMixerManager(int hardware_sample_rate,
-                                                     int hardware_buffer_size)
-    : hardware_sample_rate_(hardware_sample_rate),
-      hardware_buffer_size_(hardware_buffer_size),
+AudioRendererMixerManager::AudioRendererMixerManager(
+    media::AudioHardwareConfig* hardware_config)
+    : hardware_config_(hardware_config),
       sink_for_testing_(NULL) {
 }
 
 AudioRendererMixerManager::~AudioRendererMixerManager() {
   DCHECK(mixers_.empty());
 }
 
 media::AudioRendererMixerInput* AudioRendererMixerManager::CreateInput(
     int source_render_view_id) {
   return new media::AudioRendererMixerInput(
@@ skipping 20 matching lines @@
   if (it != mixers_.end()) {
     it->second.ref_count++;
     return it->second.mixer;
   }
 
   // On Linux and ChromeOS we can rely on the playback device to handle
   // resampling, so don't waste cycles on it here.
 #if defined(OS_LINUX)
   int sample_rate = params.sample_rate();
 #else
-  int sample_rate = hardware_sample_rate_;
+  int sample_rate = hardware_config_->GetOutputSampleRate();
 #endif
 
   // Create output parameters based on the audio hardware configuration for
   // passing on to the output sink.  Force to 16-bit output for now since we
   // know that works well for WebAudio and WebRTC.
   media::AudioParameters output_params(
       media::AudioParameters::AUDIO_PCM_LOW_LATENCY, params.channel_layout(),
-      sample_rate, 16, hardware_buffer_size_);
+      sample_rate, 16, hardware_config_->GetOutputBufferSize());
 
   // If we've created invalid output parameters, simply pass on the input params
   // and let the browser side handle automatic fallback.
   if (!output_params.IsValid())
     output_params = params;
 
   media::AudioRendererMixer* mixer;
   if (sink_for_testing_) {
     mixer = new media::AudioRendererMixer(
         params, output_params, sink_for_testing_);
@@ skipping 20 matching lines @@
 
   // Only remove the mixer if AudioRendererMixerManager is the last owner.
   it->second.ref_count--;
   if (it->second.ref_count == 0) {
     delete it->second.mixer;
     mixers_.erase(it);
   }
 }
 
 }  // namespace content