Low level windows audio support (part 1 of 2)...

media/audio/audio_output.h

 // Copyright (c) 2006-2008 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.
 
 #ifndef MEDIA_AUDIO_AUDIO_OUTPUT_H_
 #define MEDIA_AUDIO_AUDIO_OUTPUT_H_
 
 // Low-level audio output support. To make sound there are 3 objects involved:
 // - AudioSource : produces audio samples on a pull model. Implements
 //   the AudioSourceCallback interface.
@@ skipping 15 matching lines @@
 // idiosyncrasies of each platform. Non-goals:
 // - Positional, 3d audio
 // - Dependence on non-default libraries such as DirectX 9, 10, XAudio
 // - Digital signal processing or effects
 // - Extra features if a specific hardware is installed (EAX, X-fi)
 //
 // The primary client of this facility is audio coming from several tabs.
 // Specifically for this case we avoid supporting complex formats such as MP3
 // or WMA. Complex format decoding should be done by the renderers.
 
+
 // Models an audio stream that gets rendered to the audio hardware output.
 // Because we support more audio streams than physically available channels
 // a given AudioOutputStream might or might not talk directly to hardware.
 class AudioOutputStream {
  public:
   // Audio sources must implement AudioSourceCallback. This interface will be
   // called in a random thread which very likely is a high priority thread. Do
   // not rely on using this thread TLS or make calls that alter the thread
   // itself such as creating Windows or initializing COM.
   class AudioSourceCallback {
@@ skipping 53 matching lines @@
 // objects. Provides some convenience functions that avoid the need to provide
 // iterators over the existing streams.
 class AudioManager {
  public:
   enum Format {
     AUDIO_PCM_LINEAR,  // Pulse code modulation means 'raw' amplitude samples.
     AUDIO_PCM_DELTA,   // Delta-encoded pulse code modulation. 
     AUDIO_MOCK         // Creates a dummy AudioOutputStream object.
   };
 
+  // Telephone quality sample rate, mostly for speech-only audio.
+  static const int kTelephoneSampleRate = 8000;
+  // CD sampling rate is 44.1 KHz or conveniently 2x2x3x3x5x5x7x7.
+  static const int kAudioCDSampleRate = 44100;
+  // Digital Audio Tape sample rate.
+  static const int kAudioDATSampleRate = 48000;
+
   // Factory for all the supported stream formats. At this moment |channels|
   // can be 1 (mono) or 2 (stereo). The |sample_rate| is in hertz and can be
   // any value supported by the underlying platform. For some future formats
   // the |sample_rate| and |bits_per_sample| can take special values.
   // Returns NULL if the combination of the parameters is not supported, or if
   // we have reached some other platform specific limit.
   //
   // Do not free the returned AudioOutputStream. It is owned by AudioManager.
   virtual AudioOutputStream* MakeAudioStream(Format format, int channels,
                                              int sample_rate,
                                              char bits_per_sample) = 0;
 
   // Muting continues playback but effectively the volume is set to zero.
   // Un-muting returns the volume to the previous level.
   virtual void MuteAll() = 0;
   virtual void UnMuteAll() = 0;
 
+  // For testing purposes only. Returns the internal buffer of the last
+  // AUDIO_MOCK AudioOutputStream closed. Returns NULL if none closed yet.
+  // The buffer size is the same as passed to AudioOutputStream::Open().
+  virtual const void* GetLastMockBuffer() = 0;
+
+  // Get AudioManager singleton.
+  // TODO(cpu): Define threading requirements for interacting with AudioManager.
+  static AudioManager* GetAudioManager();
+
  protected:
   virtual ~AudioManager() {}
 };
 
-// Get AudioManager singleton.
-// TODO(cpu): Define threading requirements for interacting with AudioManager.
-AudioManager* GetAudioManager();
 
 #endif  // MEDIA_AUDIO_AUDIO_OUTPUT_H_