From 36d22d82aa202bb199967e9512281e9a53db42c9 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Sun, 7 Apr 2024 21:33:14 +0200 Subject: Adding upstream version 115.7.0esr. Signed-off-by: Daniel Baumann --- .../libwebrtc/common_audio/channel_buffer.cc | 80 ++++++++++++++++++++++ 1 file changed, 80 insertions(+) create mode 100644 third_party/libwebrtc/common_audio/channel_buffer.cc (limited to 'third_party/libwebrtc/common_audio/channel_buffer.cc') diff --git a/third_party/libwebrtc/common_audio/channel_buffer.cc b/third_party/libwebrtc/common_audio/channel_buffer.cc new file mode 100644 index 0000000000..b9b8c25e37 --- /dev/null +++ b/third_party/libwebrtc/common_audio/channel_buffer.cc @@ -0,0 +1,80 @@ +/* + * 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 -- cgit v1.2.3