chromium/third_party/webrtc/modules/audio_processing/test/audio_buffer_tools.cc

/*
 *  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) { … }

void CopyVectorToAudioBuffer(const StreamConfig& stream_config,
                             rtc::ArrayView<const float> source,
                             AudioBuffer* destination) { … }

void ExtractVectorFromAudioBuffer(const StreamConfig& stream_config,
                                  AudioBuffer* source,
                                  std::vector<float>* destination) { … }

void FillBuffer(float value, AudioBuffer& audio_buffer) { … }

void FillBufferChannel(float value, int channel, AudioBuffer& audio_buffer) { … }

}  // namespace test
}  // namespace webrtc