mirror of
https://github.com/danog/libtgvoip.git
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333c4a1101
Added simple audio resampler Replaced prebuilt static libs with their sources & added that to all project files (closes #5)
80 lines
2.1 KiB
C++
80 lines
2.1 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/channel_buffer.h"
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#include "webrtc/base/checks.h"
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namespace webrtc {
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IFChannelBuffer::IFChannelBuffer(size_t num_frames,
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size_t num_channels,
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size_t num_bands)
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: ivalid_(true),
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ibuf_(num_frames, num_channels, num_bands),
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fvalid_(true),
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fbuf_(num_frames, num_channels, num_bands) {}
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IFChannelBuffer::~IFChannelBuffer() = default;
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ChannelBuffer<int16_t>* IFChannelBuffer::ibuf() {
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RefreshI();
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fvalid_ = false;
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return &ibuf_;
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}
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ChannelBuffer<float>* IFChannelBuffer::fbuf() {
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RefreshF();
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ivalid_ = false;
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return &fbuf_;
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}
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const ChannelBuffer<int16_t>* IFChannelBuffer::ibuf_const() const {
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RefreshI();
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return &ibuf_;
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}
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const ChannelBuffer<float>* IFChannelBuffer::fbuf_const() const {
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RefreshF();
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return &fbuf_;
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}
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void IFChannelBuffer::RefreshF() const {
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if (!fvalid_) {
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RTC_DCHECK(ivalid_);
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fbuf_.set_num_channels(ibuf_.num_channels());
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const int16_t* const* int_channels = ibuf_.channels();
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float* const* float_channels = fbuf_.channels();
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for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
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for (size_t j = 0; j < ibuf_.num_frames(); ++j) {
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float_channels[i][j] = int_channels[i][j];
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}
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}
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fvalid_ = true;
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}
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}
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void IFChannelBuffer::RefreshI() const {
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if (!ivalid_) {
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RTC_DCHECK(fvalid_);
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int16_t* const* int_channels = ibuf_.channels();
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ibuf_.set_num_channels(fbuf_.num_channels());
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const float* const* float_channels = fbuf_.channels();
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for (size_t i = 0; i < fbuf_.num_channels(); ++i) {
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FloatS16ToS16(float_channels[i],
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ibuf_.num_frames(),
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int_channels[i]);
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}
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ivalid_ = true;
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}
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}
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} // namespace webrtc
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