blob: 0f0e5cd5205cf15352a60392b05dd97062451e96 [file] [log] [blame]
* 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() *
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());
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]);
} // namespace test
} // namespace webrtc