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/*
* 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_frame.h"
#include "api/audio/audio_processing.h"
#include "api/audio/audio_view.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/wav_file.h"
namespace webrtc {
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(AudioProcessing::kNoError, (expr))
// Encapsulates samples and metadata for an integer frame.
struct Int16FrameData {
// Max data size that matches the data size of the AudioFrame class, providing
// storage for 8 channels of 96 kHz data.
static const int kMaxDataSizeSamples = AudioFrame::kMaxDataSizeSamples;
Int16FrameData() = default;
void CopyFrom(const Int16FrameData& src);
bool IsEqual(const Int16FrameData& frame) const;
void Scale(float f);
// Sets `samples_per_channel`, `num_channels` and, implicitly, the sample
// rate. The sample rate is set to 100x that of samples per channel. I.e. if
// samples_per_channel is 320, the sample rate will be set to 32000.
void SetProperties(size_t samples_per_channel, size_t num_channels);
size_t size() const { return view_.size(); }
size_t samples_per_channel() const { return view_.samples_per_channel(); }
size_t num_channels() const { return view_.num_channels(); }
void set_num_channels(size_t num_channels);
InterleavedView<int16_t> view() { return view_; }
InterleavedView<const int16_t> view() const { return view_; }
void FillData(int16_t value);
void FillStereoData(int16_t left, int16_t right);
// public struct members.
std::array<int16_t, kMaxDataSizeSamples> data = {};
int32_t sample_rate_hz = 0;
private:
InterleavedView<int16_t> view_;
};
// Reads ChannelBuffers from a provided WavReader.
class ChannelBufferWavReader final {
public:
explicit ChannelBufferWavReader(std::unique_ptr<WavReader> file);
~ChannelBufferWavReader();
ChannelBufferWavReader(const ChannelBufferWavReader&) = delete;
ChannelBufferWavReader& operator=(const ChannelBufferWavReader&) = delete;
// Reads data from the file according to the `buffer` format. Returns false if
// a full buffer can't be read from the file.
bool Read(ChannelBuffer<float>* buffer);
private:
std::unique_ptr<WavReader> file_;
std::vector<float> interleaved_;
};
// Writes ChannelBuffers to a provided WavWriter.
class ChannelBufferWavWriter final {
public:
explicit ChannelBufferWavWriter(std::unique_ptr<WavWriter> file);
~ChannelBufferWavWriter();
ChannelBufferWavWriter(const ChannelBufferWavWriter&) = delete;
ChannelBufferWavWriter& operator=(const ChannelBufferWavWriter&) = delete;
void Write(const ChannelBuffer<float>& buffer);
private:
std::unique_ptr<WavWriter> file_;
std::vector<float> interleaved_;
};
// 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 {
public:
explicit ChannelBufferVectorWriter(std::vector<float>* output);
ChannelBufferVectorWriter(const ChannelBufferVectorWriter&) = delete;
ChannelBufferVectorWriter& operator=(const ChannelBufferVectorWriter&) =
delete;
~ChannelBufferVectorWriter();
// Creates an interleaved copy of `buffer`, converts the samples to float S16
// and appends the result to output_.
void Write(const ChannelBuffer<float>& buffer);
private:
std::vector<float> interleaved_buffer_;
std::vector<float>* output_;
};
// Exits on failure; do not use in unit tests.
FILE* OpenFile(absl::string_view filename, absl::string_view mode);
template <typename T>
void SetContainerFormat(int sample_rate_hz,
size_t num_channels,
Int16FrameData* frame,
std::unique_ptr<ChannelBuffer<T> >* cb) {
frame->SetProperties(sample_rate_hz / 100, num_channels);
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel(), num_channels));
}
template <typename T>
float ComputeSNR(const T* ref, const T* test, size_t length, float* variance) {
float mse = 0;
float mean = 0;
*variance = 0;
for (size_t i = 0; i < length; ++i) {
T error = ref[i] - test[i];
mse += error * error;
*variance += ref[i] * ref[i];
mean += ref[i];
}
mse /= length;
*variance /= length;
mean /= length;
*variance -= mean * mean;
float snr = 100; // We assign 100 dB to the zero-error case.
if (mse > 0)
snr = 10 * log10(*variance / mse);
return snr;
}
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
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