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

media/audio/audio_low_latency_input_output_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "base/basictypes.h"
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/message_loop.h"
 #include "base/path_service.h"
@@ skipping 273 matching lines @@
 
 class AudioInputStreamTraits {
  public:
   typedef AudioInputStream StreamType;
 
   static int HardwareSampleRate() {
     return static_cast<int>(media::GetAudioInputHardwareSampleRate(
         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,
       AudioManagerBase::kDefaultDeviceId);
   }
 };
 
 class AudioOutputStreamTraits {
  public:
   typedef AudioOutputStream StreamType;
 
   static int HardwareSampleRate() {
     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);
   }
 };
 
 // Traits template holding a trait of StreamType. It encapsulates
 // AudioInputStream and AudioOutputStream stream types.
 template <typename StreamTraits>
 class StreamWrapper {
  public:
   typedef typename StreamTraits::StreamType StreamType;
 
   explicit StreamWrapper(AudioManager* audio_manager)
       :
 #if defined(OS_WIN)
         com_init_(base::win::ScopedCOMInitializer::kMTA),
 #endif
         audio_manager_(audio_manager),
         format_(AudioParameters::AUDIO_PCM_LOW_LATENCY),
 #if defined(OS_ANDROID)
         channel_layout_(CHANNEL_LAYOUT_MONO),
 #else
         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() {}
 
   // Creates an Audio[Input|Output]Stream stream object using default
   // parameters.
   StreamType* Create() {
     return CreateStream();
   }
 
@@ skipping 94 matching lines @@
 
   // All Close() operations that run on the mocked audio thread,
   // should be synchronous and not post additional close tasks to
   // mocked the audio thread. Hence, there is no need to call
   // message_loop()->RunAllPending() after the Close() methods.
   aos->Close();
   ais->Close();
 }
 
 }  // namespace media