Ensure that full-duplex audio test uses preferred buffer size.

media/audio/audio_low_latency_input_output_unittest.cc

Index: media/audio/audio_low_latency_input_output_unittest.cc
diff --git a/media/audio/audio_low_latency_input_output_unittest.cc b/media/audio/audio_low_latency_input_output_unittest.cc
index 7ca7f03635d51f7b00e420e63474d1542223544f..43c1d9c61ad4a601725bfcefd00f632f2e7c4321 100644
--- a/media/audio/audio_low_latency_input_output_unittest.cc
+++ b/media/audio/audio_low_latency_input_output_unittest.cc
@@ -291,6 +291,12 @@ class AudioInputStreamTraits {
         AudioManagerBase::kDefaultDeviceId));
   }
 
+  // TODO(henrika): add support for GetAudioInputHardwareBufferSize in media.
+  // Using same buffer size as for the output side for now.

[tommi (sloooow) - chröme, 2012/11/09 13:20:39]

nit: move this line into the implementation since it doesn't make sense for the
caller to know this.

[henrika (OOO until Aug 14), 2012/11/09 13:30:57]

Done.

+  static int HardwareBufferSize() {
+    return static_cast<int>(media::GetAudioHardwareBufferSize());
+  }
+
   static StreamType* CreateStream(AudioManager* audio_manager,
       const AudioParameters& params) {
     return audio_manager->MakeAudioInputStream(params,
@@ -306,6 +312,10 @@ class AudioOutputStreamTraits {
     return static_cast<int>(media::GetAudioHardwareSampleRate());
   }
 
+  static int HardwareBufferSize() {
+    return static_cast<int>(media::GetAudioHardwareBufferSize());
+  }
+
   static StreamType* CreateStream(AudioManager* audio_manager,
       const AudioParameters& params) {
     return audio_manager->MakeAudioOutputStream(params);
@@ -332,35 +342,9 @@ class StreamWrapper {
         channel_layout_(CHANNEL_LAYOUT_STEREO),
 #endif
         bits_per_sample_(16) {
-    // Use native/mixing sample rate and N*10ms frame size as default,
-    // where N is platform dependent.
+    // Use the preferred sample rate and buffer size.
     sample_rate_ = StreamTraits::HardwareSampleRate();
-#if defined(OS_MACOSX)
-    // 10ms buffer size works well for 44.1, 48, 96 and 192kHz.
-    samples_per_packet_ = (sample_rate_ / 100);
-#elif defined(OS_LINUX) || defined(OS_OPENBSD)
-    // 10ms buffer size works well for 44.1, 48, 96 and 192kHz.
-    samples_per_packet_ = (sample_rate_ / 100);
-#elif defined(OS_WIN)
-    if (media::IsWASAPISupported()) {
-      // WASAPI is supported for Windows Vista and higher.
-      if (sample_rate_ == 44100) {
-        // Tests have shown that the shared mode WASAPI implementation
-        // works bests for a period size of ~10.15873 ms when the sample
-        // rate is 44.1kHz.
-        samples_per_packet_ = 448;
-      } else {
-        // 10ms buffer size works well for 48, 96 and 192kHz.
-        samples_per_packet_ = (sample_rate_ / 100);
-      }
-    } else {
-      // Low-latency Wave implementation needs 30ms buffer size to
-      // ensure glitch-free output audio.
-      samples_per_packet_ = 3 * (sample_rate_ / 100);
-    }
-#elif defined(OS_ANDROID)
-      samples_per_packet_ = (sample_rate_ / 100);
-#endif
+    samples_per_packet_ = StreamTraits::HardwareBufferSize();
   }
 
   virtual ~StreamWrapper() {}