Collapse AudioRendererMixer and OnMoreDataResampler into AudioTransform.

media/base/audio_transform.h

Index: media/base/audio_transform.h
diff --git a/media/base/audio_transform.h b/media/base/audio_transform.h
new file mode 100644
index 0000000000000000000000000000000000000000..bf3c12fb6653679266cc8164bc1f44701b758559
--- /dev/null
+++ b/media/base/audio_transform.h
@@ -0,0 +1,108 @@
+// 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_TRANSFORM_H_
+#define MEDIA_BASE_AUDIO_TRANSFORM_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);

[DaleCurtis, 2012/11/16 23:51:05]

Added this after benchmark shows a 20% perf hit when the FIFO is invoked.  This
is because using the fifo results in 3x the memcpy. Once from the callback, once
into the fifo, and once into the output.

+  ~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_TRANSFORM_H_