Adding upstream version 86.0.1.upstream/86.0.1upstream

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

1 files changed, 51 insertions, 0 deletions

diff --git a/third_party/libwebrtc/webrtc/common_audio/audio_util.cc b/third_party/libwebrtc/webrtc/common_audio/audio_util.cc
new file mode 100644
index 0000000000..b442a146f7
--- /dev/null
+++ b/third_party/libwebrtc/webrtc/common_audio/audio_util.cc
@@ -0,0 +1,51 @@
+/*
+ *  Copyright (c) 2013 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/include/audio_util.h"
+
+#include "typedefs.h"  // NOLINT(build/include)
+
+namespace webrtc {
+
+void FloatToS16(const float* src, size_t size, int16_t* dest) {
+  for (size_t i = 0; i < size; ++i)
+    dest[i] = FloatToS16(src[i]);
+}
+
+void S16ToFloat(const int16_t* src, size_t size, float* dest) {
+  for (size_t i = 0; i < size; ++i)
+    dest[i] = S16ToFloat(src[i]);
+}
+
+void FloatS16ToS16(const float* src, size_t size, int16_t* dest) {
+  for (size_t i = 0; i < size; ++i)
+    dest[i] = FloatS16ToS16(src[i]);
+}
+
+void FloatToFloatS16(const float* src, size_t size, float* dest) {
+  for (size_t i = 0; i < size; ++i)
+    dest[i] = FloatToFloatS16(src[i]);
+}
+
+void FloatS16ToFloat(const float* src, size_t size, float* dest) {
+  for (size_t i = 0; i < size; ++i)
+    dest[i] = FloatS16ToFloat(src[i]);
+}
+
+template <>
+void DownmixInterleavedToMono<int16_t>(const int16_t* interleaved,
+                                       size_t num_frames,
+                                       int num_channels,
+                                       int16_t* deinterleaved) {
+  DownmixInterleavedToMonoImpl<int16_t, int32_t>(interleaved, num_frames,
+                                                 num_channels, deinterleaved);
+}
+
+}  // namespace webrtc