Collapse AudioRendererMixer and OnMoreDataResampler into AudioTransform.

media/base/audio_converter.h

Index: media/base/audio_converter.h
diff --git a/media/base/audio_converter.h b/media/base/audio_converter.h
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index 0000000000000000000000000000000000000000..2987b78ecf027c234a749fdb0d18bcd85e58f91f
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+++ b/media/base/audio_converter.h
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+// 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_BASE_AUDIO_CONVERTER_H_
+#define MEDIA_BASE_AUDIO_CONVERTER_H_
+
+#include <list>
+
+#include "base/callback.h"
+#include "base/time.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/media_export.h"
+
+namespace media {
+
+class AudioBus;
+class AudioPullFifo;
+class ChannelMixer;
+class MultiChannelResampler;
+
+// AudioConverter is a complete mixing, resampling, buffering, and channel
+// mixing solution for converting data from one set of AudioParameters to
+// another.  For efficiency pieces are only invoked when necessary; e.g. the
+// resampler is only used if the input and output sample rates differ.  Mixing
+// and channel down mixing are done prior to resampling to maximize efficiency.
+class MEDIA_EXPORT AudioConverter {
+ public:
+  class MEDIA_EXPORT InputCallback {
+   public:
+    // Method for providing more data into the converter.  Expects |audio_bus|
+    // to be completely filled with data upon return; zero padded if not enough
+    // frames are available to satisfy the request.  The return value is the
+    // volume level of the provided audio data.  If a volume level of zero is
+    // returned no further processing will be done on the provided data, else
+    // the volume level will be used to scale the provided audio data.
+    virtual double ProvideInput(AudioBus* audio_bus,
+                                base::TimeDelta buffer_delay) = 0;
+
+   protected:
+    virtual ~InputCallback() {}
+  };
+
+  // Construct an AudioConverter for converting between the given input and
+  // output parameters.  Specifying |disable_fifo| means all InputCallbacks are
+  // capable of handling arbitrary buffer size requests; i.e. one call might ask
+  // for 10 frames of data (indicated by the size of AudioBus provided) and the
+  // next might ask for 20.  In synthetic testing, disabling the FIFO yields a
+  // ~20% speed up for common cases.
+  AudioConverter(const AudioParameters& input_params,
+                 const AudioParameters& output_params,
+                 bool disable_fifo);
+  ~AudioConverter();
+
+  // Converts audio from all inputs into the |dest|.  |dest| must be sized for
+  // data matching the output AudioParameters provided during construction.
+  void Convert(AudioBus* dest);
+
+  // Add or remove an input from the converter.
+  void AddInput(InputCallback* input);
+  void RemoveInput(InputCallback* input);
+
+  // Flush all buffered data.  Automatically called when all inputs are removed.
+  void Reset();
+
+ private:
+  // Called by MultiChannelResampler when more data is necessary.
+  void ProvideInput(int resampler_frame_delay, AudioBus* audio_bus);
+
+  // Called by AudioPullFifo when more data is necessary.
+  void SourceCallback(int fifo_frame_delay, AudioBus* audio_bus);
+
+  // Set of inputs for Convert().
+  typedef std::list<InputCallback*> InputCallbackSet;
+  InputCallbackSet transform_inputs_;
+
+  // 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_;
+
+  // Handles resampling.
+  scoped_ptr<MultiChannelResampler> resampler_;
+
+  // Handles channel transforms.  |unmixed_audio_| is a temporary destination
+  // for audio data before it goes into the channel mixer.
+  scoped_ptr<ChannelMixer> channel_mixer_;
+  scoped_ptr<AudioBus> unmixed_audio_;
+
+  // Temporary AudioBus destination for mixing inputs.
+  scoped_ptr<AudioBus> mixer_input_audio_bus_;
+
+  // Since resampling is expensive, figure out if we should downmix channels
+  // before resampling.
+  bool downmix_early_;
+
+  // Used to calculate buffer delay information for InputCallbacks.
+  base::TimeDelta input_frame_duration_;
+  base::TimeDelta output_frame_duration_;
+  int resampler_frame_delay_;
+
+  const int input_channel_count_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioConverter);
+};
+
+}  // namespace media
+
+#endif  // MEDIA_BASE_AUDIO_CONVERTER_H_