chromium/third_party/webrtc/modules/audio_processing/test/test_utils.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/test_utils.h"

#include <string>
#include <utility>

#include "absl/strings/string_view.h"
#include "rtc_base/checks.h"
#include "rtc_base/system/arch.h"

namespace webrtc {

ChannelBufferWavReader::ChannelBufferWavReader(std::unique_ptr<WavReader> file)
    : … { … }

ChannelBufferWavReader::~ChannelBufferWavReader() = default;

bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) { … }

ChannelBufferWavWriter::ChannelBufferWavWriter(std::unique_ptr<WavWriter> file)
    : … { … }

ChannelBufferWavWriter::~ChannelBufferWavWriter() = default;

void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) { … }

ChannelBufferVectorWriter::ChannelBufferVectorWriter(std::vector<float>* output)
    : … { … }

ChannelBufferVectorWriter::~ChannelBufferVectorWriter() = default;

void ChannelBufferVectorWriter::Write(const ChannelBuffer<float>& buffer) { … }

FILE* OpenFile(absl::string_view filename, absl::string_view mode) { … }

void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz) { … }

}  // namespace webrtc