Avoids irregular OnMoreData callbacks on Windows using Core Audio

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 30157b0c0b7235644b10b9e3a46bb5c648e2b0fc..b8f066e8de3761a24b9ec9d16254edfedd85de39 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -45,6 +45,9 @@ class AudioUtil : public AudioUtilInterface {
       const std::string& device_id) OVERRIDE {
     return media::GetAudioInputHardwareChannelLayout(device_id);
   }
+  virtual int GetAudioOutputBufferSize() OVERRIDE {
+    return media::GetAudioHardwareBufferSize();
+  }
  private:
   DISALLOW_COPY_AND_ASSIGN(AudioUtil);
 };
@@ -69,6 +72,9 @@ class AudioUtilNoHardware : public AudioUtilInterface {
       const std::string& device_id) OVERRIDE {
     return input_channel_layout_;
   }
+  virtual int GetAudioOutputBufferSize() OVERRIDE {
+    return (output_rate_ / 100);
+  }
 
  private:
   int output_rate_;
@@ -93,7 +99,7 @@ bool HardwareSampleRatesAreValid() {
   // The actual WebRTC client can limit these ranges further depending on
   // platform but this is the maximum range we support today.
   int valid_input_rates[] = {16000, 32000, 44100, 48000, 96000};

[tommi (sloooow) - chröme, 2013/01/31 13:42:08]

nit: make these const

[henrika (OOO until Aug 14), 2013/01/31 14:29:38]

Done.

-  int valid_output_rates[] = {44100, 48000, 96000};
+  int valid_output_rates[] = {16000, 32000, 44100, 48000, 96000};
 
   // Verify the input sample rate.
   int input_sample_rate = GetAudioInputSampleRate();
@@ -477,7 +483,7 @@ TEST_F(WebRTCAudioDeviceTest, DISABLED_PlayLocalFile) {
   // Play 2 seconds worth of audio and then quit.
   message_loop_.PostDelayedTask(FROM_HERE,
                                 MessageLoop::QuitClosure(),
-                                base::TimeDelta::FromSeconds(2));
+                                base::TimeDelta::FromSeconds(10));
   message_loop_.Run();
 
   renderer->Stop();
@@ -555,7 +561,7 @@ TEST_F(WebRTCAudioDeviceTest, FullDuplexAudioWithAGC) {
   LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
   message_loop_.PostDelayedTask(FROM_HERE,
                                 MessageLoop::QuitClosure(),
-                                base::TimeDelta::FromSeconds(2));
+                                base::TimeDelta::FromSeconds(10));
   message_loop_.Run();
 
   renderer->Stop();