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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.cc | |
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.cc')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc | 68 |
1 files changed, 68 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc new file mode 100644 index 0000000000..64fb9c7ab1 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/test/audio_buffer_tools.cc @@ -0,0 +1,68 @@ +/* + * 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. + */ + +#include "modules/audio_processing/test/audio_buffer_tools.h" + +#include <string.h> + +namespace webrtc { +namespace test { + +void SetupFrame(const StreamConfig& stream_config, + std::vector<float*>* frame, + std::vector<float>* frame_samples) { + frame_samples->resize(stream_config.num_channels() * + stream_config.num_frames()); + frame->resize(stream_config.num_channels()); + for (size_t ch = 0; ch < stream_config.num_channels(); ++ch) { + (*frame)[ch] = &(*frame_samples)[ch * stream_config.num_frames()]; + } +} + +void CopyVectorToAudioBuffer(const StreamConfig& stream_config, + rtc::ArrayView<const float> source, + AudioBuffer* destination) { + std::vector<float*> input; + std::vector<float> input_samples; + + SetupFrame(stream_config, &input, &input_samples); + + RTC_CHECK_EQ(input_samples.size(), source.size()); + memcpy(input_samples.data(), source.data(), + source.size() * sizeof(source[0])); + + destination->CopyFrom(&input[0], stream_config); +} + +void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config, + AudioBuffer* source, + std::vector<float>* destination) { + std::vector<float*> output; + + SetupFrame(stream_config, &output, destination); + + source->CopyTo(stream_config, &output[0]); +} + +void FillBuffer(float value, AudioBuffer& audio_buffer) { + for (size_t ch = 0; ch < audio_buffer.num_channels(); ++ch) { + FillBufferChannel(value, ch, audio_buffer); + } +} + +void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer) { + RTC_CHECK_LT(channel, audio_buffer.num_channels()); + for (size_t i = 0; i < audio_buffer.num_frames(); ++i) { + audio_buffer.channels()[channel][i] = value; + } +} + +} // namespace test +} // namespace webrtc |