Used 10ms native buffer size for webrtc audio renderer on Linux and Mac

content/renderer/media/webrtc_audio_renderer_unittest.cc

Index: content/renderer/media/webrtc_audio_renderer_unittest.cc
diff --git a/content/renderer/media/webrtc_audio_renderer_unittest.cc b/content/renderer/media/webrtc_audio_renderer_unittest.cc
index 90edcf29dd662c34d93e438a1af554d647f81d2e..240aeb6019222d49b902216c609aa9ea5e54771d 100644
--- a/content/renderer/media/webrtc_audio_renderer_unittest.cc
+++ b/content/renderer/media/webrtc_audio_renderer_unittest.cc
@@ -24,6 +24,9 @@ namespace content {
 
 namespace {
 
+const int kHardwareSampleRate = 44100;
+const int kHardwareBufferSize = 512;
+
 class MockAudioOutputIPC : public media::AudioOutputIPC {
  public:
   MockAudioOutputIPC() {}
@@ -88,7 +91,8 @@ class WebRtcAudioRendererTest : public testing::Test {
         factory_(new MockAudioDeviceFactory()),
         source_(new MockAudioRendererSource()),
         stream_(new rtc::RefCountedObject<MockMediaStream>("label")),
-        renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100, 441)) {
+        renderer_(new WebRtcAudioRenderer(stream_, 1, 1, 1, 44100,
+                                          kHardwareBufferSize)) {
     EXPECT_CALL(*factory_.get(), CreateOutputDevice(1))
         .WillOnce(Return(mock_output_device_.get()));
     EXPECT_CALL(*mock_output_device_.get(), Start());
@@ -151,4 +155,25 @@ TEST_F(WebRtcAudioRendererTest, MultipleRenderers) {
   }
 }
 
+// Verify that the sink of the renderer is using the expected sample rate and
+// buffer size.
+TEST_F(WebRtcAudioRendererTest, VerifySinkParameters) {
+  renderer_proxy_->Start();
+#if defined(OS_LINUX) || defined(OS_MACOSX)
+  static const int kExpectedBufferSize = kHardwareSampleRate / 100;
+#elif defined(OS_ANDROID)
+  static const int kExpectedBufferSize = 2 * kHardwareSampleRate / 100;
+#else
+  // Windows.
+  static const int kExpectedBufferSize = kHardwareBufferSize;
+#endif
+  EXPECT_EQ(kExpectedBufferSize, renderer_->buffer_size());
+  EXPECT_EQ(kHardwareSampleRate, renderer_->sample_rate());
+  EXPECT_EQ(2, renderer_->channels());
+
+  EXPECT_CALL(*mock_output_device_.get(), Stop());
+  EXPECT_CALL(*source_.get(), RemoveAudioRenderer(renderer_.get()));
+  renderer_proxy_->Stop();
+}
+
 }  // namespace content