/* * 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 #include "common_audio/include/audio_util.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* IFChannelBuffer::ibuf() { RefreshI(); fvalid_ = false; return &ibuf_; } ChannelBuffer* IFChannelBuffer::fbuf() { RefreshF(); ivalid_ = false; return &fbuf_; } const ChannelBuffer* IFChannelBuffer::ibuf_const() const { RefreshI(); return &ibuf_; } const ChannelBuffer* 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