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/*
* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
#include <string.h>
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/include/audio_util.h"
namespace webrtc {
// Helper to encapsulate a contiguous data buffer with access to a pointer
// array of the deinterleaved channels.
template <typename T>
class ChannelBuffer {
public:
ChannelBuffer(int samples_per_channel, int num_channels)
: data_(new T[samples_per_channel * num_channels]),
channels_(new T*[num_channels]),
samples_per_channel_(samples_per_channel),
num_channels_(num_channels) {
Initialize();
}
ChannelBuffer(const T* data, int samples_per_channel, int num_channels)
: data_(new T[samples_per_channel * num_channels]),
channels_(new T*[num_channels]),
samples_per_channel_(samples_per_channel),
num_channels_(num_channels) {
Initialize();
memcpy(data_.get(), data, length() * sizeof(T));
}
ChannelBuffer(const T* const* channels, int samples_per_channel,
int num_channels)
: data_(new T[samples_per_channel * num_channels]),
channels_(new T*[num_channels]),
samples_per_channel_(samples_per_channel),
num_channels_(num_channels) {
Initialize();
for (int i = 0; i < num_channels_; ++i)
CopyFrom(channels[i], i);
}
~ChannelBuffer() {}
void CopyFrom(const void* channel_ptr, int i) {
DCHECK_LT(i, num_channels_);
memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
}
T* data() { return data_.get(); }
const T* data() const { return data_.get(); }
const T* channel(int i) const {
DCHECK_GE(i, 0);
DCHECK_LT(i, num_channels_);
return channels_[i];
}
T* channel(int i) {
const ChannelBuffer<T>* t = this;
return const_cast<T*>(t->channel(i));
}
T* const* channels() { return channels_.get(); }
const T* const* channels() const { return channels_.get(); }
int samples_per_channel() const { return samples_per_channel_; }
int num_channels() const { return num_channels_; }
int length() const { return samples_per_channel_ * num_channels_; }
private:
void Initialize() {
memset(data_.get(), 0, sizeof(T) * length());
for (int i = 0; i < num_channels_; ++i)
channels_[i] = &data_[i * samples_per_channel_];
}
scoped_ptr<T[]> data_;
scoped_ptr<T*[]> channels_;
const int samples_per_channel_;
const int num_channels_;
};
// One int16_t and one float ChannelBuffer that are kept in sync. The sync is
// broken when someone requests write access to either ChannelBuffer, and
// reestablished when someone requests the outdated ChannelBuffer. It is
// therefore safe to use the return value of ibuf_const() and fbuf_const()
// until the next call to ibuf() or fbuf(), and the return value of ibuf() and
// fbuf() until the next call to any of the other functions.
class IFChannelBuffer {
public:
IFChannelBuffer(int samples_per_channel, int num_channels);
ChannelBuffer<int16_t>* ibuf();
ChannelBuffer<float>* fbuf();
const ChannelBuffer<int16_t>* ibuf_const() const;
const ChannelBuffer<float>* fbuf_const() const;
int num_channels() const { return ibuf_.num_channels(); }
int samples_per_channel() const { return ibuf_.samples_per_channel(); }
private:
void RefreshF() const;
void RefreshI() const;
mutable bool ivalid_;
mutable ChannelBuffer<int16_t> ibuf_;
mutable bool fvalid_;
mutable ChannelBuffer<float> fbuf_;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_CHANNEL_BUFFER_H_
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