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

The existing version does not work on Mac since it uses a hard coded buffer size and that is not allowed by the lower layers.
The solution is to use the preferred buffer size instead.

BUG=none
TEST=media_unittests.exe --gtest_filter=AudioLow* --gtest_also_run_disabled_tests

Review URL: https://codereview.chromium.org/11360168

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@166906 0039d316-1c4b-4281-b951-d872f2087c98

1 files changed, 14 insertions, 28 deletions

diff --git a/media/audio/audio_low_latency_input_output_unittest.cc b/media/audio/audio_low_latency_input_output_unittest.cc
index 7ca7f03..9376db8 100644
--- a/media/audio/audio_low_latency_input_output_unittest.cc
+++ b/media/audio/audio_low_latency_input_output_unittest.cc
@@ -291,6 +291,11 @@ class AudioInputStreamTraits {
         AudioManagerBase::kDefaultDeviceId));
   }
 
+  // TODO(henrika): add support for GetAudioInputHardwareBufferSize in media.
+  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 +311,10 @@ 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 +341,12 @@ 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.
     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
+
+    // Use the preferred buffer size. Note that the input side uses the same
+    // size as the output side in this implementation.
+    samples_per_packet_ = StreamTraits::HardwareBufferSize();
   }
 
   virtual ~StreamWrapper() {}