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| 1 // Copyright (c) 2012 The Chromium Authors. All rights reserved. |
| 2 // Use of this source code is governed by a BSD-style license that can be |
| 3 // found in the LICENSE file. |
| 4 |
| 5 #ifndef MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_ |
| 6 #define MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_ |
| 7 |
| 8 #include "base/basictypes.h" |
| 9 #include "base/memory/ref_counted.h" |
| 10 #include "base/synchronization/lock.h" |
| 11 #include "base/time.h" |
| 12 #include "media/audio/audio_io.h" |
| 13 #include "media/audio/audio_manager.h" |
| 14 #include "media/audio/audio_output_dispatcher.h" |
| 15 #include "media/audio/audio_parameters.h" |
| 16 |
| 17 namespace media { |
| 18 |
| 19 class AudioPullFifo; |
| 20 class MultiChannelResampler; |
| 21 |
| 22 // AudioOutputResampler is a browser-side resampling and rebuffering solution |
| 23 // which ensures audio data is always output at given parameters. The rough |
| 24 // flow is: Client -> [FIFO] -> [Resampler] -> Output Device. |
| 25 // |
| 26 // The FIFO and resampler are only used when necessary. To be clear: |
| 27 // - The resampler is only used if the input and output sample rates differ. |
| 28 // - The FIFO is only used if the input and output frame sizes differ or if |
| 29 // the resampler is used. |
| 30 // |
| 31 // AOR works by intercepting the AudioSourceCallback provided to StartStream() |
| 32 // and redirecting to the appropriate resampling or FIFO callback which passes |
| 33 // through to the original callback only when necessary. |
| 34 // |
| 35 // Currently channel downmixing and upmixing is not supported. |
| 36 // TODO(dalecurtis): Add channel remixing. http://crbug.com/138762 |
| 37 class MEDIA_EXPORT AudioOutputResampler |
| 38 : public AudioOutputDispatcher, |
| 39 public AudioOutputStream::AudioSourceCallback { |
| 40 public: |
| 41 AudioOutputResampler(AudioManager* audio_manager, |
| 42 const AudioParameters& input_params, |
| 43 const AudioParameters& output_params, |
| 44 const base::TimeDelta& close_delay); |
| 45 |
| 46 // AudioOutputDispatcher interface. |
| 47 virtual bool OpenStream() OVERRIDE; |
| 48 virtual bool StartStream(AudioOutputStream::AudioSourceCallback* callback, |
| 49 AudioOutputProxy* stream_proxy) OVERRIDE; |
| 50 virtual void StopStream(AudioOutputProxy* stream_proxy) OVERRIDE; |
| 51 virtual void StreamVolumeSet(AudioOutputProxy* stream_proxy, |
| 52 double volume) OVERRIDE; |
| 53 virtual void CloseStream(AudioOutputProxy* stream_proxy) OVERRIDE; |
| 54 virtual void Shutdown() OVERRIDE; |
| 55 |
| 56 // AudioSourceCallback interface. |
| 57 virtual int OnMoreData(AudioBus* audio_bus, |
| 58 AudioBuffersState buffers_state) OVERRIDE; |
| 59 virtual void OnError(AudioOutputStream* stream, int code) OVERRIDE; |
| 60 virtual void WaitTillDataReady() OVERRIDE; |
| 61 |
| 62 private: |
| 63 friend class base::RefCountedThreadSafe<AudioOutputResampler>; |
| 64 virtual ~AudioOutputResampler(); |
| 65 |
| 66 // Called by MultiChannelResampler when more data is necessary. |
| 67 void ProvideInput(AudioBus* audio_bus); |
| 68 |
| 69 // Called by AudioPullFifo when more data is necessary. |
| 70 void SourceCallback(AudioBus* audio_bus); |
| 71 |
| 72 // Used by StopStream()/CloseStream()/Shutdown() to clear internal state. |
| 73 // TODO(dalecurtis): Probably only one of these methods needs to call this, |
| 74 // the rest should DCHECK()/CHECK() that the values were reset. |
| 75 void Reset(); |
| 76 |
| 77 // Handles resampling. |
| 78 scoped_ptr<MultiChannelResampler> resampler_; |
| 79 |
| 80 // Dispatcher to proxy all AudioOutputDispatcher calls too. |
| 81 scoped_refptr<AudioOutputDispatcher> dispatcher_; |
| 82 |
| 83 // Source callback and associated lock. |
| 84 base::Lock source_lock_; |
| 85 AudioOutputStream::AudioSourceCallback* source_callback_; |
| 86 |
| 87 // Used to buffer data between the client and the output device in cases where |
| 88 // the client buffer size is not the same as the output device buffer size. |
| 89 scoped_ptr<AudioPullFifo> audio_fifo_; |
| 90 |
| 91 // Ratio of input bytes to output bytes used to correct playback delay with |
| 92 // regard to buffering and resampling. |
| 93 double io_ratio_; |
| 94 |
| 95 // Helper values for determining playback delay adjustment. |
| 96 int input_bytes_per_frame_; |
| 97 int output_bytes_per_frame_; |
| 98 |
| 99 // Last AudioBuffersState object received via OnMoreData(), used to correct |
| 100 // playback delay by ProvideInput() and passed on to |source_callback_|. |
| 101 AudioBuffersState current_buffers_state_; |
| 102 |
| 103 // Total number of bytes (in terms of output parameters) stored in resampler |
| 104 // or FIFO buffers which have not been sent to the audio device. |
| 105 int outstanding_audio_bytes_; |
| 106 |
| 107 DISALLOW_COPY_AND_ASSIGN(AudioOutputResampler); |
| 108 }; |
| 109 |
| 110 } // namespace media |
| 111 |
| 112 #endif // MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_ |
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