Adding input and output audio backend to Android.

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 293 matching lines @@
 // AudioInputStream and AudioOutputStream stream types.
 template <typename StreamTraits>
 class StreamWrapper {
  public:
   typedef typename StreamTraits::StreamType StreamType;
 
   explicit StreamWrapper(AudioManager* audio_manager)
       : com_init_(ScopedCOMInitializer::kMTA),
         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.
     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
   }
 
   virtual ~StreamWrapper() {}
 
   // Creates an Audio[Input|Output]Stream stream object using default
   // parameters.
   StreamType* Create() {
     return CreateStream();
   }
@@ skipping 98 matching lines @@
   aos->Stop();
   ais->Stop();
 
   // 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();
 }