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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h | 42 |
1 files changed, 42 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h new file mode 100644 index 0000000000..faac4bf9ff --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.h @@ -0,0 +1,42 @@ +/* + * Copyright (c) 2015 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_ +#define MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_ + +#include <vector> + +#include "api/array_view.h" +#include "modules/audio_processing/audio_buffer.h" +#include "modules/audio_processing/include/audio_processing.h" + +namespace webrtc { +namespace test { + +// Copies a vector into an audiobuffer. +void CopyVectorToAudioBuffer(const StreamConfig& stream_config, + rtc::ArrayView<const float> source, + AudioBuffer* destination); + +// Extracts a vector from an audiobuffer. +void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config, + AudioBuffer* source, + std::vector<float>* destination); + +// Sets all values in `audio_buffer` to `value`. +void FillBuffer(float value, AudioBuffer& audio_buffer); + +// Sets all values channel `channel` for `audio_buffer` to `value`. +void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer); + +} // namespace test +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_TEST_AUDIO_BUFFER_TOOLS_H_ |