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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.
// 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; i.e.,
// - The resampler is only used if sample rates differ.
// - The FIFO is only used if buffer sizes differ.
// - The channel mixer is only used if channel layouts differ.
// Additionally, since resampling is the most expensive operation, input mixing
// and channel down mixing are done prior to resampling. Likewise, channel up
// mixing is performed after resampling.
#include <list>
#include "base/callback.h"
#include "base/memory/scoped_ptr.h"
#include "base/time/time.h"
#include "media/audio/audio_parameters.h"
#include "media/base/media_export.h"
namespace media {
class AudioBus;
class AudioPullFifo;
class ChannelMixer;
class MultiChannelResampler;
// Converts audio data between two AudioParameters formats. Sample usage:
// AudioParameters input(...), output(...);
// AudioConverter ac(input, output);
// scoped_ptr<AudioBus> output_audio_bus = AudioBus::Create(output);
// ac.AddInput(<AudioConverter::InputCallback* 1>);
// ac.AddInput(<AudioConverter::InputCallback* 2>);
// ac.Convert(output_audio_bus.get());
// Convert() will ask for input audio data from each InputCallback and convert
// the data into the provided AudioBus.
class MEDIA_EXPORT AudioConverter {
// Interface for inputs into the converter. Each InputCallback is added or
// removed from Convert() processing via AddInput() and RemoveInput().
class MEDIA_EXPORT InputCallback {
// 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;
virtual ~InputCallback() {}
// Constructs 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);
// Converts audio from all inputs into the |dest|. If an |initial_delay| is
// specified, it will be propagated to each input.
void Convert(AudioBus* dest);
void ConvertWithDelay(const base::TimeDelta& initial_delay, AudioBus* dest);
// Adds or removes an input from the converter. RemoveInput() will call
// Reset() if no inputs remain after the specified input is removed.
void AddInput(InputCallback* input);
void RemoveInput(InputCallback* input);
// Flushes all buffered data.
void Reset();
// The maximum size in frames that guarantees we will only make a single call
// to each input's ProvideInput for more data.
int ChunkSize() const;
bool empty() const { return transform_inputs_.empty(); }
// Provides input to the MultiChannelResampler. Called by the resampler when
// more data is necessary.
void ProvideInput(int resampler_frame_delay, AudioBus* audio_bus);
// Provides input to the AudioPullFifo. Called by the fifo when more data is
// necessary.
void SourceCallback(int fifo_frame_delay, AudioBus* audio_bus);
// (Re)creates the temporary |unmixed_audio_| buffer if necessary.
void CreateUnmixedAudioIfNecessary(int frames);
// 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_;
int chunk_size_;
// 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_;
base::TimeDelta initial_delay_;
int resampler_frame_delay_;
// Number of channels of input audio data. Set during construction via the
// value from the input AudioParameters class. Preserved to recreate internal
// AudioBus structures on demand in response to varying frame size requests.
const int input_channel_count_;
} // namespace media