Index: media/audio/audio_io.h |
=================================================================== |
--- media/audio/audio_io.h (revision 57255) |
+++ media/audio/audio_io.h (working copy) |
@@ -153,4 +153,80 @@ |
virtual ~AudioInputStream() {} |
}; |
+// Manages all audio resources. In particular it owns the AudioOutputStream |
+// objects. Provides some convenience functions that avoid the need to provide |
+// iterators over the existing streams. |
+class AudioManager { |
+ public: |
+ enum Format { |
+ AUDIO_PCM_LINEAR = 0, // PCM is 'raw' amplitude samples. |
+ AUDIO_PCM_LOW_LATENCY, // Linear PCM, low latency requested. |
+ AUDIO_MOCK, // Creates a dummy AudioOutputStream object. |
+ AUDIO_LAST_FORMAT // Only used for validation of format. |
+ }; |
+ |
+ // Telephone quality sample rate, mostly for speech-only audio. |
+ static const uint32 kTelephoneSampleRate = 8000; |
+ // CD sampling rate is 44.1 KHz or conveniently 2x2x3x3x5x5x7x7. |
+ static const uint32 kAudioCDSampleRate = 44100; |
+ // Digital Audio Tape sample rate. |
+ static const uint32 kAudioDATSampleRate = 48000; |
+ |
+ // Returns true if the OS reports existence of audio devices. This does not |
+ // guarantee that the existing devices support all formats and sample rates. |
+ virtual bool HasAudioOutputDevices() = 0; |
+ |
+ // Returns true if the OS reports existence of audio recording devices. This |
+ // does not guarantee that the existing devices support all formats and |
+ // sample rates. |
+ virtual bool HasAudioInputDevices() = 0; |
+ |
+ // Factory for all the supported stream formats. The |channels| can be 1 to 5. |
+ // The |sample_rate| is in hertz and can be any value supported by the |
+ // 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. |
+ // |
+ // AUDIO_PCM_LOW_LATENCY can be passed to this method and it has two effects: |
+ // 1- Instead of triple buffered the audio will be double buffered. |
+ // 2- A low latency driver or alternative audio subsystem will be used when |
+ // available. |
+ // |
+ // Do not free the returned AudioOutputStream. It is owned by AudioManager. |
+ virtual AudioOutputStream* MakeAudioOutputStream(Format format, int channels, |
+ int sample_rate, |
+ char bits_per_sample) = 0; |
+ |
+ // Factory to create audio recording streams. |
+ // |channels| can be 1 or 2. |
+ // |sample_rate| is in hertz and can be any value supported by the platform. |
+ // |bits_per_sample| can be any value supported by the platform. |
+ // |samples_per_packet| is in hertz as well and can be 0 to |sample_rate|, |
+ // with 0 suggesting that the implementation use a default value for that |
+ // platform. |
+ // 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 AudioInputStream. It is owned by AudioManager. |
+ // When you are done with it, call |Stop()| and |Close()| to release it. |
+ virtual AudioInputStream* MakeAudioInputStream(Format format, int channels, |
+ int sample_rate, |
+ char bits_per_sample, |
+ uint32 samples_per_packet) = 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; |
+ |
+ // Get AudioManager singleton. |
+ // TODO(cpu): Define threading requirements for interacting with AudioManager. |
+ static AudioManager* GetAudioManager(); |
+ |
+ protected: |
+ virtual ~AudioManager() {} |
+}; |
+ |
+ |
#endif // MEDIA_AUDIO_AUDIO_IO_H_ |