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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/base/scoped_ptr.h"
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/test/testsupport/gtest_prod_util.h"
namespace webrtc {
// Helper to encapsulate a contiguous data buffer, full or split into frequency
// bands, with access to a pointer arrays of the deinterleaved channels and
// bands. The buffer is zero initialized at creation.
//
// The buffer structure is showed below for a 2 channel and 2 bands case:
//
// |data_|:
// { [ --- b1ch1 --- ] [ --- b2ch1 --- ] [ --- b1ch2 --- ] [ --- b2ch2 --- ] }
//
// The pointer arrays for the same example are as follows:
//
// |channels_|:
// { [ b1ch1* ] [ b1ch2* ] [ b2ch1* ] [ b2ch2* ] }
//
// |bands_|:
// { [ b1ch1* ] [ b2ch1* ] [ b1ch2* ] [ b2ch2* ] }
template <typename T>
class ChannelBuffer {
public:
ChannelBuffer(size_t num_frames,
size_t num_channels,
size_t num_bands = 1)
: data_(new T[num_frames * num_channels]()),
channels_(new T*[num_channels * num_bands]),
bands_(new T*[num_channels * num_bands]),
num_frames_(num_frames),
num_frames_per_band_(num_frames / num_bands),
num_channels_(num_channels),
num_bands_(num_bands) {
for (size_t i = 0; i < num_channels_; ++i) {
for (size_t j = 0; j < num_bands_; ++j) {
channels_[j * num_channels_ + i] =
&data_[i * num_frames_ + j * num_frames_per_band_];
bands_[i * num_bands_ + j] = channels_[j * num_channels_ + i];
}
}
}
// Returns a pointer array to the full-band channels (or lower band channels).
// Usage:
// channels()[channel][sample].
// Where:
// 0 <= channel < |num_channels_|
// 0 <= sample < |num_frames_|
T* const* channels() { return channels(0); }
const T* const* channels() const { return channels(0); }
// Returns a pointer array to the channels for a specific band.
// Usage:
// channels(band)[channel][sample].
// Where:
// 0 <= band < |num_bands_|
// 0 <= channel < |num_channels_|
// 0 <= sample < |num_frames_per_band_|
const T* const* channels(size_t band) const {
RTC_DCHECK_LT(band, num_bands_);
return &channels_[band * num_channels_];
}
T* const* channels(size_t band) {
const ChannelBuffer<T>* t = this;
return const_cast<T* const*>(t->channels(band));
}
// Returns a pointer array to the bands for a specific channel.
// Usage:
// bands(channel)[band][sample].
// Where:
// 0 <= channel < |num_channels_|
// 0 <= band < |num_bands_|
// 0 <= sample < |num_frames_per_band_|
const T* const* bands(size_t channel) const {
RTC_DCHECK_LT(channel, num_channels_);
RTC_DCHECK_GE(channel, 0u);
return &bands_[channel * num_bands_];
}
T* const* bands(size_t channel) {
const ChannelBuffer<T>* t = this;
return const_cast<T* const*>(t->bands(channel));
}
// Sets the |slice| pointers to the |start_frame| position for each channel.
// Returns |slice| for convenience.
const T* const* Slice(T** slice, size_t start_frame) const {
RTC_DCHECK_LT(start_frame, num_frames_);
for (size_t i = 0; i < num_channels_; ++i)
slice[i] = &channels_[i][start_frame];
return slice;
}
T** Slice(T** slice, size_t start_frame) {
const ChannelBuffer<T>* t = this;
return const_cast<T**>(t->Slice(slice, start_frame));
}
size_t num_frames() const { return num_frames_; }
size_t num_frames_per_band() const { return num_frames_per_band_; }
size_t num_channels() const { return num_channels_; }
size_t num_bands() const { return num_bands_; }
size_t size() const {return num_frames_ * num_channels_; }
void SetDataForTesting(const T* data, size_t size) {
RTC_CHECK_EQ(size, this->size());
memcpy(data_.get(), data, size * sizeof(*data));
}
private:
rtc::scoped_ptr<T[]> data_;
rtc::scoped_ptr<T* []> channels_;
rtc::scoped_ptr<T* []> bands_;
const size_t num_frames_;
const size_t num_frames_per_band_;
const size_t num_channels_;
const size_t num_bands_;
};
// 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(size_t num_frames, size_t num_channels, size_t num_bands = 1);
ChannelBuffer<int16_t>* ibuf();
ChannelBuffer<float>* fbuf();
const ChannelBuffer<int16_t>* ibuf_const() const;
const ChannelBuffer<float>* fbuf_const() const;
size_t num_frames() const { return ibuf_.num_frames(); }
size_t num_frames_per_band() const { return ibuf_.num_frames_per_band(); }
size_t num_channels() const { return ibuf_.num_channels(); }
size_t num_bands() const { return ibuf_.num_bands(); }
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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