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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 17:32:43 +0000
commit6bf0a5cb5034a7e684dcc3500e841785237ce2dd (patch)
treea68f146d7fa01f0134297619fbe7e33db084e0aa /third_party/libwebrtc/common_audio/channel_buffer.cc
parentInitial commit. (diff)
downloadthunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.tar.xz
thunderbird-6bf0a5cb5034a7e684dcc3500e841785237ce2dd.zip
Adding upstream version 1:115.7.0.upstream/1%115.7.0upstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/common_audio/channel_buffer.cc')
-rw-r--r--third_party/libwebrtc/common_audio/channel_buffer.cc80
1 files changed, 80 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/channel_buffer.cc b/third_party/libwebrtc/common_audio/channel_buffer.cc
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+++ 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 <cstdint>
+
+#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<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