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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_
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