Pass more detailed audio hardware configuration information to the renderer

content/test/webrtc_audio_device_test.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/test/webrtc_audio_device_test.h"
 
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/compiler_specific.h"
 #include "base/file_util.h"
 #include "base/message_loop.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/test/test_timeouts.h"
 #include "content/browser/renderer_host/media/audio_input_renderer_host.h"
 #include "content/browser/renderer_host/media/audio_mirroring_manager.h"
 #include "content/browser/renderer_host/media/audio_renderer_host.h"
 #include "content/browser/renderer_host/media/media_stream_manager.h"
 #include "content/browser/renderer_host/media/mock_media_observer.h"
+#include "content/common/media/media_param_traits.h"

[DaleCurtis, 2013/03/07 02:41:39]

??

[Chris Rogers, 2013/03/09 01:37:50]

I found that it was necessary to include this for the serializing of
AudioParameters, since we're now using them for the AudioHardwareConfig. 
There's a compile error otherwise.

 #include "content/common/view_messages.h"
 #include "content/public/browser/browser_thread.h"
 #include "content/public/common/content_paths.h"
 #include "content/public/test/mock_resource_context.h"
 #include "content/public/test/test_browser_thread.h"
 #include "content/renderer/media/audio_input_message_filter.h"
 #include "content/renderer/media/audio_message_filter.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/renderer/render_process.h"
 #include "content/renderer/render_thread_impl.h"
 #include "content/renderer/renderer_webkitplatformsupport_impl.h"
+#include "media/audio/audio_parameters.h"
 #include "media/base/audio_hardware_config.h"
 #include "net/url_request/url_request_test_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "third_party/webrtc/voice_engine/include/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/include/voe_base.h"
 #include "third_party/webrtc/voice_engine/include/voe_file.h"
 #include "third_party/webrtc/voice_engine/include/voe_network.h"
 
 #if defined(OS_WIN)
 #include "base/win/scoped_com_initializer.h"
 #endif
 
+using media::AudioParameters;
+using media::ChannelLayout;
 using testing::_;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace content {
 
 // This class is a mock of the child process singleton which is needed
 // to be able to create a RenderThread object.
 class WebRTCMockRenderProcess : public RenderProcess {
@@ skipping 202 matching lines @@
   audio_render_host_->OnChannelClosing();
   audio_render_host_->OnFilterRemoved();
   audio_input_renderer_host_->OnChannelClosing();
   audio_input_renderer_host_->OnFilterRemoved();
   channel_.reset();
   audio_render_host_ = NULL;
   audio_input_renderer_host_ = NULL;
 }
 
 void WebRTCAudioDeviceTest::OnGetAudioHardwareConfig(
-    int* output_buffer_size, int* output_sample_rate, int* input_sample_rate,
-    media::ChannelLayout* input_channel_layout) {
+    AudioParameters* input_params, AudioParameters* output_params) {
   ASSERT_TRUE(audio_hardware_config_);
 
-  *output_buffer_size = audio_hardware_config_->GetOutputBufferSize();
-  *output_sample_rate = audio_hardware_config_->GetOutputSampleRate();
+  int output_buffer_size = audio_hardware_config_->GetOutputBufferSize();
+  int output_sample_rate = audio_hardware_config_->GetOutputSampleRate();
+  ChannelLayout output_channels_layout =

[DaleCurtis, 2013/03/07 02:41:39]

Seems AudioHardwareConfig should offer to return AudioParameter objects ?

+      audio_hardware_config_->GetOutputChannelLayout();
 
   // TODO(henrika): add support for all available input devices.
-  *input_sample_rate = audio_hardware_config_->GetInputSampleRate();
-  *input_channel_layout = audio_hardware_config_->GetInputChannelLayout();
+  int input_sample_rate = audio_hardware_config_->GetInputSampleRate();
+  ChannelLayout input_channel_layout =
+      audio_hardware_config_->GetInputChannelLayout();
+
+  *input_params = AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      input_channel_layout,
+      input_sample_rate,
+      16,
+      output_buffer_size);
+
+  *output_params = AudioParameters(
+      AudioParameters::AUDIO_PCM_LOW_LATENCY,
+      output_channels_layout,
+      output_sample_rate,
+      16,
+      output_buffer_size);
 }
 
 // IPC::Listener implementation.
 bool WebRTCAudioDeviceTest::OnMessageReceived(const IPC::Message& message) {
   if (render_thread_) {
     IPC::ChannelProxy::MessageFilter* filter =
         render_thread_->audio_input_message_filter();
     if (filter->OnMessageReceived(message))
       return true;
 
@@ skipping 74 matching lines @@
 int WebRTCTransportImpl::SendPacket(int channel, const void* data, int len) {
   return network_->ReceivedRTPPacket(channel, data, len);
 }
 
 int WebRTCTransportImpl::SendRTCPPacket(int channel, const void* data,
                                         int len) {
   return network_->ReceivedRTCPPacket(channel, data, len);
 }
 
 }  // namespace content