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/*
* Copyright (C) 2010 Google Inc. All rights reserved.
*
* Redistribution and use in source and binary forms, with or without
* modification, are permitted provided that the following conditions
* are met:
*
* 1. Redistributions of source code must retain the above copyright
* notice, this list of conditions and the following disclaimer.
* 2. Redistributions in binary form must reproduce the above copyright
* notice, this list of conditions and the following disclaimer in the
* documentation and/or other materials provided with the distribution.
* 3. Neither the name of Apple Computer, Inc. ("Apple") nor the names of
* its contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY APPLE AND ITS CONTRIBUTORS "AS IS" AND ANY
* EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED
* WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE
* DISCLAIMED. IN NO EVENT SHALL APPLE OR ITS CONTRIBUTORS BE LIABLE FOR ANY
* DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
* (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
* LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND
* ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
* (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
* THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
#ifndef THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_BUS_H_
#define THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_BUS_H_
#include <memory>
#include "third_party/blink/renderer/platform/audio/audio_channel.h"
#include "third_party/blink/renderer/platform/wtf/noncopyable.h"
#include "third_party/blink/renderer/platform/wtf/thread_safe_ref_counted.h"
#include "third_party/blink/renderer/platform/wtf/vector.h"
namespace blink {
// An AudioBus represents a collection of one or more AudioChannels.
// The data layout is "planar" as opposed to "interleaved". An AudioBus with
// one channel is mono, an AudioBus with two channels is stereo, etc.
class PLATFORM_EXPORT AudioBus : public ThreadSafeRefCounted<AudioBus> {
WTF_MAKE_NONCOPYABLE(AudioBus);
public:
enum {
kChannelLeft = 0,
kChannelRight = 1,
kChannelCenter = 2, // center and mono are the same
kChannelMono = 2,
kChannelLFE = 3,
kChannelSurroundLeft = 4,
kChannelSurroundRight = 5,
};
enum {
kLayoutCanonical = 0
// Can define non-standard layouts here
};
enum ChannelInterpretation {
kSpeakers,
kDiscrete,
};
// allocate indicates whether or not to initially have the AudioChannels
// created with managed storage. Normal usage is to pass true here, in which
// case the AudioChannels will memory-manage their own storage. If allocate
// is false then setChannelMemory() has to be called later on for each
// channel before the AudioBus is useable...
static scoped_refptr<AudioBus> Create(unsigned number_of_channels,
size_t length,
bool allocate = true);
// Tells the given channel to use an externally allocated buffer.
void SetChannelMemory(unsigned channel_index, float* storage, size_t length);
// Channels
unsigned NumberOfChannels() const { return channels_.size(); }
AudioChannel* Channel(unsigned channel) { return channels_[channel].get(); }
const AudioChannel* Channel(unsigned channel) const {
return channels_[channel].get();
}
AudioChannel* ChannelByType(unsigned type);
const AudioChannel* ChannelByType(unsigned type) const;
// Number of sample-frames
size_t length() const { return length_; }
// resizeSmaller() can only be called with a new length <= the current length.
// The data stored in the bus will remain undisturbed.
void ResizeSmaller(size_t new_length);
// Sample-rate : 0.0 if unknown or "don't care"
float SampleRate() const { return sample_rate_; }
void SetSampleRate(float sample_rate) { sample_rate_ = sample_rate; }
// Zeroes all channels.
void Zero();
// Clears the silent flag on all channels.
void ClearSilentFlag();
// Returns true if the silent bit is set on all channels.
bool IsSilent() const;
// Returns true if the channel count and frame-size match.
bool TopologyMatches(const AudioBus& source_bus) const;
// Creates a new buffer from a range in the source buffer.
// 0 may be returned if the range does not fit in the sourceBuffer
static scoped_refptr<AudioBus> CreateBufferFromRange(
const AudioBus* source_buffer,
unsigned start_frame,
unsigned end_frame);
// Creates a new AudioBus by sample-rate converting sourceBus to the
// newSampleRate.
// setSampleRate() must have been previously called on sourceBus.
// Note: sample-rate conversion is already handled in the file-reading code
// for the mac port, so we don't need this.
static scoped_refptr<AudioBus> CreateBySampleRateConverting(
const AudioBus* source_bus,
bool mix_to_mono,
double new_sample_rate);
// Creates a new AudioBus by mixing all the channels down to mono.
// If sourceBus is already mono, then the returned AudioBus will simply be a
// copy.
static scoped_refptr<AudioBus> CreateByMixingToMono(
const AudioBus* source_bus);
// Scales all samples by the same amount.
void Scale(float scale);
// Copies the samples from the source bus to this one.
// This is just a simple per-channel copy if the number of channels match,
// otherwise an up-mix or down-mix is done.
void CopyFrom(const AudioBus& source_bus, ChannelInterpretation = kSpeakers);
// Sums the samples from the source bus to this one.
// This is just a simple per-channel summing if the number of channels match,
// otherwise an up-mix or down-mix is done.
void SumFrom(const AudioBus& source_bus, ChannelInterpretation = kSpeakers);
// Copy each channel from |source_bus| into our corresponding channel. We
// scale |source_bus| by |gain| before copying into the bus.
void CopyWithGainFrom(const AudioBus& source_bus, float gain);
// Copies the sourceBus by scaling with sample-accurate gain values.
void CopyWithSampleAccurateGainValuesFrom(const AudioBus& source_bus,
float* gain_values,
unsigned number_of_gain_values);
// Returns maximum absolute value across all channels (useful for
// normalization).
float MaxAbsValue() const;
// Makes maximum absolute value == 1.0 (if possible).
void Normalize();
static scoped_refptr<AudioBus> GetDataResource(const char* name,
float sample_rate);
protected:
AudioBus() = default;
AudioBus(unsigned number_of_channels, size_t length, bool allocate);
void DiscreteSumFrom(const AudioBus&);
// Up/down-mix by in-place summing upon the existing channel content.
// http://webaudio.github.io/web-audio-api/#channel-up-mixing-and-down-mixing
void SumFromByUpMixing(const AudioBus&);
void SumFromByDownMixing(const AudioBus&);
size_t length_;
Vector<std::unique_ptr<AudioChannel>> channels_;
int layout_;
float sample_rate_; // 0.0 if unknown or N/A
};
} // namespace blink
#endif // THIRD_PARTY_BLINK_RENDERER_PLATFORM_AUDIO_AUDIO_BUS_H_
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