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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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_
#include <math.h>
#include <iterator>
#include <limits>
#include <string>
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/channel_buffer.h"
#include "webrtc/common_audio/wav_file.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
static const AudioProcessing::Error kNoErr = AudioProcessing::kNoError;
#define EXPECT_NOERR(expr) EXPECT_EQ(kNoErr, (expr))
class RawFile final {
public:
explicit RawFile(const std::string& filename);
~RawFile();
void WriteSamples(const int16_t* samples, size_t num_samples);
void WriteSamples(const float* samples, size_t num_samples);
private:
FILE* file_handle_;
RTC_DISALLOW_COPY_AND_ASSIGN(RawFile);
};
void WriteIntData(const int16_t* data,
size_t length,
WavWriter* wav_file,
RawFile* raw_file);
void WriteFloatData(const float* const* data,
int samples_per_channel,
int num_channels,
WavWriter* wav_file,
RawFile* raw_file);
// Exits on failure; do not use in unit tests.
FILE* OpenFile(const std::string& filename, const char* mode);
int SamplesFromRate(int rate);
void SetFrameSampleRate(AudioFrame* frame,
int sample_rate_hz);
template <typename T>
void SetContainerFormat(int sample_rate_hz,
int num_channels,
AudioFrame* frame,
rtc::scoped_ptr<ChannelBuffer<T> >* cb) {
SetFrameSampleRate(frame, sample_rate_hz);
frame->num_channels_ = num_channels;
cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
}
AudioProcessing::ChannelLayout LayoutFromChannels(int num_channels);
template <typename T>
float ComputeSNR(const T* ref, const T* test, int length, float* variance) {
float mse = 0;
float mean = 0;
*variance = 0;
for (int 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;
}
// Returns a vector<T> parsed from whitespace delimited values in to_parse,
// or an empty vector if the string could not be parsed.
template<typename T>
std::vector<T> ParseList(const std::string& to_parse) {
std::vector<T> values;
std::istringstream str(to_parse);
std::copy(
std::istream_iterator<T>(str),
std::istream_iterator<T>(),
std::back_inserter(values));
return values;
}
// Parses the array geometry from the command line.
//
// If a vector with size != num_mics is returned, an error has occurred and an
// appropriate error message has been printed to stdout.
std::vector<Point> ParseArrayGeometry(const std::string& mic_positions,
size_t num_mics);
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_TEST_UTILS_H_