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

/*
 *  Copyright (c) 2014 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_TEST_UTILS_H_
#define MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_

#include <math.h>

#include <iterator>
#include <limits>
#include <memory>
#include <string>
#include <vector>

#include "absl/strings/string_view.h"
#include "api/audio/audio_processing.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/wav_file.h"

namespace webrtc {

static const AudioProcessing::Error kNoErr = …;
#define EXPECT_NOERR(expr) …

// Encapsulates samples and metadata for an integer frame.
struct Int16FrameData { … };

// Reads ChannelBuffers from a provided WavReader.
class ChannelBufferWavReader final { … };

// Writes ChannelBuffers to a provided WavWriter.
class ChannelBufferWavWriter final { … };

// Takes a pointer to a vector. Allows appending the samples of channel buffers
// to the given vector, by interleaving the samples and converting them to float
// S16.
class ChannelBufferVectorWriter final { … };

// Exits on failure; do not use in unit tests.
FILE* OpenFile(absl::string_view filename, absl::string_view mode);

void SetFrameSampleRate(Int16FrameData* frame, int sample_rate_hz);

template <typename T>
void SetContainerFormat(int sample_rate_hz,
                        size_t num_channels,
                        Int16FrameData* frame,
                        std::unique_ptr<ChannelBuffer<T> >* cb) { … }

template <typename T>
float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) { … }

}  // namespace webrtc

#endif  // MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_