| // 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. |
| |
| #ifndef MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_ |
| #define MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_ |
| |
| #include <map> |
| |
| #include "base/macros.h" |
| #include "base/memory/ref_counted.h" |
| #include "base/time/time.h" |
| #include "base/timer/timer.h" |
| #include "media/audio/audio_io.h" |
| #include "media/audio/audio_manager.h" |
| #include "media/audio/audio_output_dispatcher_impl.h" |
| #include "media/audio/audio_parameters.h" |
| |
| namespace media { |
| |
| class OnMoreDataConverter; |
| |
| // AudioOutputResampler is a browser-side resampling and buffering solution |
| // which ensures audio data is always output at given parameters. See the |
| // AudioConverter class for details on the conversion process. |
| // |
| // AOR works by intercepting the AudioSourceCallback provided to StartStream() |
| // and redirecting it through an AudioConverter instance. |total_bytes_delay| |
| // is adjusted for buffer delay caused by the conversion process. |
| // |
| // AOR will automatically fall back from AUDIO_PCM_LOW_LATENCY to |
| // AUDIO_PCM_LINEAR if the output device fails to open at the requested output |
| // parameters. If opening still fails, it will fallback to AUDIO_FAKE. |
| class MEDIA_EXPORT AudioOutputResampler : public AudioOutputDispatcher { |
| public: |
| AudioOutputResampler(AudioManager* audio_manager, |
| const AudioParameters& input_params, |
| const AudioParameters& output_params, |
| const std::string& output_device_id, |
| const base::TimeDelta& close_delay); |
| |
| // AudioOutputDispatcher interface. |
| bool OpenStream() override; |
| bool StartStream(AudioOutputStream::AudioSourceCallback* callback, |
| AudioOutputProxy* stream_proxy) override; |
| void StopStream(AudioOutputProxy* stream_proxy) override; |
| void StreamVolumeSet(AudioOutputProxy* stream_proxy, double volume) override; |
| void CloseStream(AudioOutputProxy* stream_proxy) override; |
| void Shutdown() override; |
| |
| private: |
| friend class base::RefCountedThreadSafe<AudioOutputResampler>; |
| ~AudioOutputResampler() override; |
| |
| // Converts low latency based output parameters into high latency |
| // appropriate output parameters in error situations. |
| void SetupFallbackParams(); |
| |
| // Used to reinitialize |dispatcher_|. |
| void Reinitialize(); |
| |
| // Used to initialize |dispatcher_|. |
| void Initialize(); |
| |
| // Dispatcher to proxy all AudioOutputDispatcher calls too. |
| scoped_refptr<AudioOutputDispatcherImpl> dispatcher_; |
| |
| // Map of outstanding OnMoreDataConverter objects. A new object is created |
| // on every StartStream() call and destroyed on CloseStream(). |
| typedef std::map<AudioOutputProxy*, OnMoreDataConverter*> CallbackMap; |
| CallbackMap callbacks_; |
| |
| // Used by AudioOutputDispatcherImpl; kept so we can reinitialize on the fly. |
| base::TimeDelta close_delay_; |
| |
| // AudioParameters used to setup the output stream; changed upon fallback. |
| AudioParameters output_params_; |
| |
| // The original AudioParameters we were constructed with. |
| const AudioParameters original_output_params_; |
| |
| // Whether any streams have been opened through |dispatcher_|, if so we can't |
| // fallback on future OpenStream() failures. |
| bool streams_opened_; |
| |
| // The reinitialization timer provides a way to recover from temporary failure |
| // states by clearing the dispatcher if all proxies have been closed and none |
| // have been created within |close_delay_|. Without this, audio may be lost |
| // to a fake stream indefinitely for transient errors. |
| base::Timer reinitialize_timer_; |
| |
| DISALLOW_COPY_AND_ASSIGN(AudioOutputResampler); |
| }; |
| |
| } // namespace media |
| |
| #endif // MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_ |