Implicit audio output device selection for getUserMedia.

content/renderer/media/webrtc_audio_device_unittest.cc

Index: content/renderer/media/webrtc_audio_device_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index 7074c95bb09c3e28966169cb62e91300918c89ec..17661f6cef4d432404b9f5f3e6dd342677bb05f0 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -520,8 +520,14 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_StartPlayout) {
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,
+                              frames_per_buffer);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
@@ -701,8 +707,14 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,
+                              frames_per_buffer);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
@@ -778,8 +790,14 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, MAYBE_FullDuplexAudioWithAGC) {
   EXPECT_CALL(media_observer(),
       OnDeleteAudioStream(_, 1)).Times(AnyNumber());
 
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,
+                              frames_per_buffer);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
@@ -923,8 +941,15 @@ TEST_F(MAYBE_WebRTCAudioDeviceTest, WebRtcPlayoutSetupTime) {
   base::WaitableEvent event(false, false);
   scoped_ptr<MockWebRtcAudioRendererSource> renderer_source(
       new MockWebRtcAudioRendererSource(&event));
+
+  media::AudioHardwareConfig* hardware_config =
+      RenderThreadImpl::current()->GetAudioHardwareConfig();
+  int sample_rate = hardware_config->GetOutputSampleRate();
+  int frames_per_buffer = hardware_config->GetOutputBufferSize();
+
   scoped_refptr<WebRtcAudioRenderer> renderer =
-      new WebRtcAudioRenderer(kRenderViewId);
+      new WebRtcAudioRenderer(kRenderViewId, -1, sample_rate,

[Jói, 2013/09/06 14:49:26]

These parameters and the boilerplate to get the sample rate and frames per
buffer are repeated a few times in the file; maybe make a factory for
WebRtcAudioRenderer that does that work?

[tommi (sloooow) - chröme, 2013/09/06 16:56:54]

Done.  Added a factory method to WebRTCAudioDeviceTest.

+                              frames_per_buffer);
   renderer->Initialize(renderer_source.get());
 
   // Start the timer and playout.