Audio software mixer.

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 @@
 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),
         channel_layout_(CHANNEL_LAYOUT_STEREO),
-        bits_per_sample_(16) {
+        bits_per_sample_(16),
+        use_browser_mixer_(false) {
     // 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)
@@ skipping 31 matching lines @@
     return CreateStream();
   }
 
   AudioParameters::Format format() const { return format_; }
   int channels() const {
     return ChannelLayoutToChannelCount(channel_layout_);
   }
   int bits_per_sample() const { return bits_per_sample_; }
   int sample_rate() const { return sample_rate_; }
   int samples_per_packet() const { return samples_per_packet_; }
+  bool use_browser_mixer() const { return use_browser_mixer_; }
 
  private:
   StreamType* CreateStream() {
     StreamType* stream = StreamTraits::CreateStream(audio_manager_,
-        AudioParameters(format_, channel_layout_, sample_rate_,
-            bits_per_sample_, samples_per_packet_));
+        AudioParameters(format_, use_browser_mixer_, channel_layout_,
+            sample_rate_, bits_per_sample_, samples_per_packet_));
     EXPECT_TRUE(stream);
     return stream;
   }
 
   ScopedCOMInitializer com_init_;
   AudioManager* audio_manager_;
   AudioParameters::Format format_;
   ChannelLayout channel_layout_;
   int bits_per_sample_;
   int sample_rate_;
   int samples_per_packet_;
+  bool use_browser_mixer_;
 };
 
 typedef StreamWrapper<AudioInputStreamTraits> AudioInputStreamWrapper;
 typedef StreamWrapper<AudioOutputStreamTraits> AudioOutputStreamWrapper;
 
 // This test is intended for manual tests and should only be enabled
 // when it is required to make a real-time test of audio in full duplex and
 // at the same time create a text file which contains measured delay values.
 // The file can later be analyzed off line using e.g. MATLAB.
 // MATLAB example:
@@ skipping 56 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();
 }