Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

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
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 <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