Moved AudioUtil static functions to AudioManager interfaces

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"
 #include "base/synchronization/lock.h"
 #include "base/test/test_timeouts.h"
 #include "base/time.h"
 #include "build/build_config.h"
 #include "media/audio/audio_io.h"
 #include "media/audio/audio_manager_base.h"
-#include "media/audio/audio_util.h"
 #include "media/base/seekable_buffer.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #if defined(OS_LINUX) || defined(OS_OPENBSD)
 #include "media/audio/linux/audio_manager_linux.h"
 #elif defined(OS_MACOSX)
 #include "media/audio/mac/audio_manager_mac.h"
 #elif defined(OS_WIN)
 #include "media/audio/win/audio_manager_win.h"
@@ skipping 251 matching lines @@
   scoped_array<AudioDelayState> delay_states_;
   size_t input_elements_to_write_;
   size_t output_elements_to_write_;
   base::Time previous_write_time_;
 };
 
 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.
-  static int HardwareBufferSize() {
-    return static_cast<int>(media::GetAudioHardwareBufferSize());
+  static AudioParameters GetDefaultAudioStreamParameters(
+      AudioManager* audio_manager) {
+    return audio_manager->GetInputStreamParameters(
+        AudioManagerBase::kDefaultDeviceId);
   }
 
   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 AudioParameters GetDefaultAudioStreamParameters(
+      AudioManager* audio_manager) {
+    return audio_manager->GetDefaultOutputStreamParameters();
   }
 
   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)
       :
         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 the preferred sample rate.
-    sample_rate_ = StreamTraits::HardwareSampleRate();
+    const AudioParameters& params =
+        StreamTraits::GetDefaultAudioStreamParameters(audio_manager_);
+    sample_rate_ = params.sample_rate();
 
     // 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();
+    samples_per_packet_ = params.frames_per_buffer();
   }
 
   virtual ~StreamWrapper() {}
 
   // Creates an Audio[Input|Output]Stream stream object using default
   // parameters.
   StreamType* Create() {
     return CreateStream();
   }
 
@@ skipping 90 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()->RunUntilIdle() after the Close() methods.
   aos->Close();
   ais->Close();
 }
 
 }  // namespace media