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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.
*/
#include "common_audio/channel_buffer.h"
#include "rtc_base/checks.h"
namespace webrtc {
IFChannelBuffer::IFChannelBuffer(size_t num_frames,
size_t num_channels,
size_t num_bands)
: ivalid_(true),
ibuf_(num_frames, num_channels, num_bands),
fvalid_(true),
fbuf_(num_frames, num_channels, num_bands) {}
IFChannelBuffer::~IFChannelBuffer() = default;
ChannelBuffer<int16_t>* IFChannelBuffer::ibuf() {
RefreshI();
fvalid_ = false;
return &ibuf_;
}
ChannelBuffer<float>* IFChannelBuffer::fbuf() {
RefreshF();
ivalid_ = false;
return &fbuf_;
}
const ChannelBuffer<int16_t>* IFChannelBuffer::ibuf_const() const {
RefreshI();
return &ibuf_;
}
const ChannelBuffer<float>* IFChannelBuffer::fbuf_const() const {
RefreshF();
return &fbuf_;
}
void IFChannelBuffer::RefreshF() const {
if (!fvalid_) {
RTC_DCHECK(ivalid_);
fbuf_.set_num_channels(ibuf_.num_channels());
const int16_t* const* int_channels = ibuf_.channels();
float* const* float_channels = fbuf_.channels();
for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
for (size_t j = 0; j < ibuf_.num_frames(); ++j) {
float_channels[i][j] = int_channels[i][j];
}
}
fvalid_ = true;
}
}
void IFChannelBuffer::RefreshI() const {
if (!ivalid_) {
RTC_DCHECK(fvalid_);
int16_t* const* int_channels = ibuf_.channels();
ibuf_.set_num_channels(fbuf_.num_channels());
const float* const* float_channels = fbuf_.channels();
for (size_t i = 0; i < fbuf_.num_channels(); ++i) {
FloatS16ToS16(float_channels[i], ibuf_.num_frames(), int_channels[i]);
}
ivalid_ = true;
}
}
} // namespace webrtc
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