Introduce AudioOutputResampler for browser side resampling.

media/audio/audio_output_resampler.h

+// 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 "base/basictypes.h"
+#include "base/memory/ref_counted.h"
+#include "base/synchronization/lock.h"
+#include "base/time.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_manager.h"
+#include "media/audio/audio_output_dispatcher.h"
+#include "media/audio/audio_parameters.h"
+
+namespace media {
+
+class AudioPullFifo;
+class MultiChannelResampler;
+
+// AudioOutputResampler is a browser-side resampling and rebuffering solution
+// which ensures audio data is always output at given parameters.  The rough
+// flow is: Client -> [FIFO] -> [Resampler] -> Output Device.
+//
+// The FIFO and resampler are only used when necessary.  To be clear:
+//   - The resampler is only used if the input and output sample rates differ.
+//   - The FIFO is only used if the input and output frame sizes differ or if
+//     the resampler is used.
+//
+// AOR works by intercepting the AudioSourceCallback provided to StartStream()
+// and redirecting to the appropriate resampling or FIFO callback which passes
+// through to the original callback only when necessary.
+//
+// Currently channel downmixing and upmixing is not supported.

[scherkus (not reviewing), 2012/09/10 14:25:32]

bug etc? do we have DCHECKs / etc in place?

[DaleCurtis, 2012/09/10 14:53:51]

Left it out since this falls under the resilience use case. I'll add one and
remove it later when that CL goes in.

+class MEDIA_EXPORT AudioOutputResampler
+    : public AudioOutputDispatcher,
+      public AudioOutputStream::AudioSourceCallback {
+ public:
+  AudioOutputResampler(AudioManager* audio_manager,
+                       const AudioParameters& input_params,
+                       const AudioParameters& output_params,
+                       const base::TimeDelta& close_delay);
+
+  // AudioOutputDispatcher interface.
+  virtual bool OpenStream() OVERRIDE;
+  virtual bool StartStream(AudioOutputStream::AudioSourceCallback* callback,
+                           AudioOutputProxy* stream_proxy) OVERRIDE;
+  virtual void StopStream(AudioOutputProxy* stream_proxy) OVERRIDE;
+  virtual void StreamVolumeSet(AudioOutputProxy* stream_proxy,
+                               double volume) OVERRIDE;
+  virtual void CloseStream(AudioOutputProxy* stream_proxy) OVERRIDE;
+  virtual void Shutdown() OVERRIDE;
+
+  // AudioSourceCallback interface.
+  virtual int OnMoreData(AudioBus* audio_bus,
+                         AudioBuffersState buffers_state) OVERRIDE;
+  virtual void OnError(AudioOutputStream* stream, int code) OVERRIDE;
+  virtual void WaitTillDataReady() OVERRIDE;
+
+ private:
+  friend class base::RefCountedThreadSafe<AudioOutputResampler>;
+  virtual ~AudioOutputResampler();
+
+  // Called by MultiChannelResampler when more data is necessary.
+  void ProvideInput(AudioBus* audio_bus);
+
+  // Called by AudioPullFifo when more data is necessary.
+  void SourceCallback(AudioBus* audio_bus);
+
+  // Used by StopStream()/CloseStream()/Shutdown() to clear internal state.
+  // TODO(dalecurtis): Probably only one of these methods needs to call this,
+  // the rest should DCHECK()/CHECK() that the values were reset.
+  void Reset();
+
+  // Handles resampling.
+  scoped_ptr<MultiChannelResampler> resampler_;
+
+  // Dispatcher to proxy all AudioOutputDispatcher calls too.
+  scoped_refptr<AudioOutputDispatcher> dispatcher_;
+
+  // Source callback and associated lock.
+  base::Lock source_lock_;
+  AudioOutputStream::AudioSourceCallback* source_callback_;

[scherkus (not reviewing), 2012/09/10 14:25:32]

who owns this?

[DaleCurtis, 2012/09/10 14:53:51]

Whoever called StartStream() (which in production is just AudioOutputController)

+
+  // Used to buffer data between the client and the output device in cases where
+  // the client buffer size is not the same as the output device buffer size.
+  scoped_ptr<AudioPullFifo> audio_fifo_;
+
+  // Ratio of input bytes to output bytes used to correct playback delay with
+  // regard to buffering and resampling.
+  double io_ratio_;
+
+  // Helper values for determining playback delay adjustment.
+  int input_bytes_per_frame_;
+  int output_bytes_per_frame_;
+
+  // Last AudioBuffersState object received via OnMoreData(), used to correct
+  // playback delay by ProvideInput() and passed on to |source_callback_|.
+  AudioBuffersState current_buffers_state_;
+
+  // Total number of bytes (in terms of output parameters) stored in resampler
+  // or FIFO buffers which have not been sent to the audio device.
+  int outstanding_audio_bytes_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioOutputResampler);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_AUDIO_AUDIO_OUTPUT_RESAMPLER_H_