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[DEV] create basic resampling speex algorithm

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Edouard DUPIN 2015-04-20 22:10:56 +02:00
commit b358318a9d
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.gitignore vendored Normal file
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###################################
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audio/algo/speex/Resampler.cpp Normal file
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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <audio/algo/speex/Resampler.h>
#include <audio/algo/speex/debug.h>
#include <cmath>
#undef __class__
#define __class__ "algo::speex::Resampler"
audio::algo::speex::Resampler::Resampler() :
#ifdef HAVE_SPEEX_DSP
m_speexResampler(nullptr),
#endif
m_isConfigured(false) {
}
audio::algo::speex::Resampler::~Resampler() {
#ifdef HAVE_SPEEX_DSP
if (m_speexResampler != nullptr) {
speex_resampler_destroy(m_speexResampler);
m_speexResampler = nullptr;
}
#endif
}
void audio::algo::speex::Resampler::init(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality) {
#ifdef HAVE_SPEEX_DSP
if (m_speexResampler != nullptr) {
speex_resampler_destroy(m_speexResampler);
m_speexResampler = nullptr;
}
AA_SPEEX_DEBUG("Create resampler for : " << _inputSampleRate << " to " << _outputSampleRate);
int err = 0;
m_speexResampler = speex_resampler_init(_nbChannel,
_inputSampleRate,
_outputSampleRate,
_quality, &err);
m_isConfigured = true;
#else
AA_SPEEX_WARNING("SPEEX DSP lib not accessible ==> can not resample");
m_isConfigured = false;
#endif
}
std::vector<enum audio::format> audio::algo::speex::Resampler::getSupportedFormat() {
std::vector<enum audio::format> out = getNativeSupportedFormat();
return out;
}
std::vector<enum audio::format> audio::algo::speex::Resampler::getNativeSupportedFormat() {
std::vector<enum audio::format> out;
out.push_back(audio::format_float);
out.push_back(audio::format_int16);
return out;
}
void audio::algo::speex::Resampler::process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk, enum audio::format _format) {
if (m_isConfigured == false) {
AA_SPEEX_ERROR("Algo is not initialized...");
}
#ifdef HAVE_SPEEX_DSP
switch (_format) {
case audio::format_int16:
{
uint32_t nbChunkInput = _nbChunk;
uint32_t nbChunkOutput = _nbChunkOut;
int ret = speex_resampler_process_interleaved_int(m_speexResampler,
reinterpret_cast<const int16_t*>(_input),
&nbChunkInput,
reinterpret_cast<int16_t*>(_output),
&nbChunkOutput);
// Check all input and output ...
if (nbChunkInput != _nbChunk) {
AA_SPEEX_ERROR("inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _nbChunk);
// TODO : manage this case ...
}
if (nbChunkOutput == _nbChunkOut) {
AA_SPEEX_ERROR("Might have not enought data in output... output size=" << _nbChunkOut);
// TODO : manage this case ...
}
_nbChunkOut = nbChunkOutput;
}
break;
case audio::format_float:
{
uint32_t nbChunkInput = _nbChunk;
uint32_t nbChunkOutput = _nbChunkOut;
int ret = speex_resampler_process_interleaved_float(m_speexResampler,
reinterpret_cast<const float*>(_input),
&nbChunkInput,
reinterpret_cast<float*>(_output),
&nbChunkOutput);
// Check all input and output ...
if (nbChunkInput != _nbChunk) {
AA_SPEEX_ERROR("inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _nbChunk);
// TODO : manage this case ...
}
if (nbChunkOutput == _nbChunkOut) {
AA_SPEEX_ERROR("Might have not enought data in output... output size=" << _nbChunkOut);
// TODO : manage this case ...
}
_nbChunkOut = nbChunkOutput;
}
break;
default:
AA_SPEEX_ERROR("Can not Limit with unsupported format : " << _format);
break;
}
#else
AA_SPEEX_ERROR("Not build with speex DSP ... ");
_nbChunkOut = _nbChunk/10;
#endif
}

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audio/algo/speex/Resampler.h Normal file
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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#ifndef __AUDIO_ALGO_SPEEX_RESAMPLER_H__
#define __AUDIO_ALGO_SPEEX_RESAMPLER_H__
#include <etk/types.h>
#include <audio/format.h>
#include <etk/chrono.h>
#include <vector>
#ifdef HAVE_SPEEX_DSP
#include <speex/speex_resampler.h>
#endif
namespace audio {
namespace algo {
namespace speex {
class Resampler {
protected:
#ifdef HAVE_SPEEX_DSP
SpeexResamplerState* m_speexResampler;
#endif
bool m_isConfigured;
public:
Resampler();
virtual ~Resampler();
public:
/**
* @brief Initialize the Algorithm
* @param[in] _nbChannel Number of channel in the stream.
* @param[in] _inputSampleRate Input sample rate.
* @param[in] _outputSampleRate Output sample rate.
* @param[in] _quality Resampler quality [1..10].
*/
virtual void init(int8_t _nbChannel, float _inputSampleRate, float _outputSampleRate, int8_t _quality);
/**
* @brief Get list of format suported in input.
* @return list of supported format
*/
virtual std::vector<enum audio::format> getSupportedFormat();
/**
* @brief Get list of algorithm format suported. No format convertion.
* @return list of supported format
*/
virtual std::vector<enum audio::format> getNativeSupportedFormat();
/**
* @brief Main input algo process.
* @param[in,out] _output Output data.
* @param[in,out] _nbChunkOut Number of chunk allocated in the output buffer and return the number of chunk in the buffer.
* @param[in] _input Input data.
* @param[in] _nbChunk Number of chunk in the input buffer.
* @param[in] _nbChannel Number of channel in the stream.
* @param[in] _format Input data format.
*/
virtual void process(void* _output, size_t& _nbChunkOut, const void* _input, size_t _nbChunk, enum audio::format _format = audio::format_float);
};
}
}
}
#endif

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audio/algo/speex/debug.cpp Normal file
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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include "debug.h"
int32_t audio::algo::speex::getLogId() {
static int32_t g_val = etk::log::registerInstance("audio-algo-speex");
return g_val;
}

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audio/algo/speex/debug.h Normal file
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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#ifndef __AUDIO_ALGO_SPEEX_DEBUG_H__
#define __AUDIO_ALGO_SPEEX_DEBUG_H__
#include <etk/log.h>
namespace audio {
namespace algo {
namespace speex {
int32_t getLogId();
}
}
}
#define AA_SPEEX_BASE(info,data) TK_LOG_BASE(audio::algo::speex::getLogId(),info,data)
#define AA_SPEEX_PRINT(data) AA_SPEEX_BASE(-1, data)
#define AA_SPEEX_CRITICAL(data) AA_SPEEX_BASE(1, data)
#define AA_SPEEX_ERROR(data) AA_SPEEX_BASE(2, data)
#define AA_SPEEX_WARNING(data) AA_SPEEX_BASE(3, data)
#ifdef DEBUG
#define AA_SPEEX_INFO(data) AA_SPEEX_BASE(4, data)
#define AA_SPEEX_DEBUG(data) AA_SPEEX_BASE(5, data)
#define AA_SPEEX_VERBOSE(data) AA_SPEEX_BASE(6, data)
#define AA_SPEEX_TODO(data) AA_SPEEX_BASE(4, "TODO : " << data)
#else
#define AA_SPEEX_INFO(data) do { } while(false)
#define AA_SPEEX_DEBUG(data) do { } while(false)
#define AA_SPEEX_VERBOSE(data) do { } while(false)
#define AA_SPEEX_TODO(data) do { } while(false)
#endif
#define AA_SPEEX_ASSERT(cond,data) \
do { \
if (!(cond)) { \
AA_SPEEX_CRITICAL(data); \
assert(!#cond); \
} \
} while (0)
#endif

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lutin_audio_algo_speex.py Normal file
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#!/usr/bin/python
import lutinModule as module
import lutinTools as tools
import lutinDebug as debug
def get_desc():
return "audio_algo_speex : speex algos wrapper"
def create(target):
myModule = module.Module(__file__, 'audio_algo_speex', 'LIBRARY')
myModule.add_src_file([
'audio/algo/speex/debug.cpp',
'audio/algo/speex/Resampler.cpp'
])
myModule.add_module_depend(['etk', 'audio'])
myModule.add_optionnal_module_depend('speexdsp', "HAVE_SPEEX_DSP", True)
myModule.add_export_path(tools.get_current_path(__file__))
# return module
return myModule

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lutin_audio_algo_speex_test.py Normal file
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#!/usr/bin/python
import lutinModule as module
import lutinTools as tools
import lutinDebug as debug
def get_desc():
return "audio_algo_speex_test: test for speex ALGO"
def create(target):
myModule = module.Module(__file__, 'audio_algo_speex_test', 'BINARY')
myModule.add_src_file([
'test/main.cpp',
'test/debug.cpp'
])
myModule.add_module_depend(['audio_algo_speex'])
return myModule

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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include "debug.h"
int32_t appl::getLogId() {
static int32_t g_val = etk::log::registerInstance("test-speex");
return g_val;
}

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test/debug.h Normal file
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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#ifndef __APPL_DEBUG_H__
#define __APPL_DEBUG_H__
#include <etk/log.h>
namespace appl {
int32_t getLogId();
};
#define APPL_BASE(info,data) TK_LOG_BASE(appl::getLogId(),info,data)
#define APPL_PRINT(data) APPL_BASE(-1, data)
#define APPL_CRITICAL(data) APPL_BASE(1, data)
#define APPL_ERROR(data) APPL_BASE(2, data)
#define APPL_WARNING(data) APPL_BASE(3, data)
#define APPL_INFO(data) APPL_BASE(4, data)
#ifdef DEBUG
#define APPL_DEBUG(data) APPL_BASE(5, data)
#define APPL_VERBOSE(data) APPL_BASE(6, data)
#define APPL_TODO(data) APPL_BASE(4, "TODO : " << data)
#else
#define APPL_DEBUG(data) do { } while(false)
#define APPL_VERBOSE(data) do { } while(false)
#define APPL_TODO(data) do { } while(false)
#endif
#define APPL_ASSERT(cond,data) \
do { \
if (!(cond)) { \
APPL_CRITICAL(data); \
assert(!#cond); \
} \
} while (0)
#endif

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/** @file
* @author Edouard DUPIN
* @copyright 2011, Edouard DUPIN, all right reserved
* @license APACHE v2.0 (see license file)
*/
#include <test/debug.h>
#include <etk/etk.h>
#include <audio/algo/speex/Resampler.h>
#include <etk/os/FSNode.h>
#include <etk/chrono.h>
#include <unistd.h>
#undef __class__
#define __class__ "test"
class Performance {
private:
std11::chrono::steady_clock::time_point m_timeStart;
std11::chrono::steady_clock::time_point m_timeStop;
std11::chrono::nanoseconds m_totalTimeProcessing;
std11::chrono::nanoseconds m_minProcessing;
std11::chrono::nanoseconds m_maxProcessing;
int32_t m_totalIteration;
public:
Performance() :
m_totalTimeProcessing(0),
m_minProcessing(99999999999999LL),
m_maxProcessing(0),
m_totalIteration(0) {
}
void tic() {
m_timeStart = std11::chrono::steady_clock::now();
}
void toc() {
m_timeStop = std11::chrono::steady_clock::now();
std11::chrono::nanoseconds time = m_timeStop - m_timeStart;
m_minProcessing = std::min(m_minProcessing, time);
m_maxProcessing = std::max(m_maxProcessing, time);
m_totalTimeProcessing += time;
m_totalIteration++;
}
std11::chrono::nanoseconds getTotalTimeProcessing() {
return m_totalTimeProcessing;
}
std11::chrono::nanoseconds getMinProcessing() {
return m_minProcessing;
}
std11::chrono::nanoseconds getMaxProcessing() {
return m_maxProcessing;
}
int32_t getTotalIteration() {
return m_totalIteration;
}
};
float performanceResamplerStepFloat(float _sampleRateIn, float _sampleRateOut, int8_t _quality) {
std::vector<float> input;
input.resize(1024, 0);
std::vector<float> output;
output.resize(input.size()*10, 0);
double sampleRate = _sampleRateIn;
{
double phase = 0;
double baseCycle = 2.0*M_PI/sampleRate * 480.0;
for (int32_t iii=0; iii<input.size(); iii++) {
input[iii] = cos(phase) * 5.0;
phase += baseCycle;
if (phase >= 2*M_PI) {
phase -= 2*M_PI;
}
}
}
APPL_INFO("Start Resampler performance ... " << _sampleRateIn << " -> " << _sampleRateOut << " float");
Performance perfo;
audio::algo::speex::Resampler algo;
algo.init(1, _sampleRateIn, _sampleRateOut, _quality);
for (int32_t iii=0; iii<1024; ++iii) {
perfo.tic();
size_t sizeOut = output.size();
algo.process(&output[0], sizeOut, &input[0], input.size(), audio::format_float);
perfo.toc();
usleep(1000);
}
APPL_INFO(" blockSize=" << input.size() << " sample");
APPL_INFO(" min < avg < max =" << perfo.getMinProcessing().count() << "ns < "
<< perfo.getTotalTimeProcessing().count()/perfo.getTotalIteration() << "ns < "
<< perfo.getMaxProcessing().count() << "ns ");
float avg = (float(((perfo.getTotalTimeProcessing().count()/perfo.getTotalIteration())*sampleRate)/double(input.size()))/1000000000.0)*100.0;
APPL_INFO(" min < avg < max= " << (float((perfo.getMinProcessing().count()*sampleRate)/double(input.size()))/1000000000.0)*100.0 << "% < "
<< avg << "% < "
<< (float((perfo.getMaxProcessing().count()*sampleRate)/double(input.size()))/1000000000.0)*100.0 << "%");
APPL_PRINT("float : " << _sampleRateIn << " -> " << _sampleRateOut << " quality=" << int32_t(_quality) << " : " << avg << "%");
return avg;
}
float performanceResamplerStepI16(float _sampleRateIn, float _sampleRateOut, int8_t _quality) {
std::vector<int16_t> input;
input.resize(1024, 0);
std::vector<int16_t> output;
output.resize(input.size()*10, 0);
double sampleRate = _sampleRateIn;
{
double phase = 0;
double baseCycle = 2.0*M_PI/sampleRate * 480.0;
for (int32_t iii=0; iii<input.size(); iii++) {
input[iii] = cos(phase) * 30000.0;
phase += baseCycle;
if (phase >= 2*M_PI) {
phase -= 2*M_PI;
}
}
}
APPL_INFO("Start Resampler performance ... " << _sampleRateIn << " -> " << _sampleRateOut << " int16_t");
Performance perfo;
audio::algo::speex::Resampler algo;
algo.init(1, _sampleRateIn, _sampleRateOut, _quality);
for (int32_t iii=0; iii<1024; ++iii) {
perfo.tic();
size_t sizeOut = output.size();
algo.process(&output[0], sizeOut, &input[0], input.size(), audio::format_int16);
perfo.toc();
usleep(1000);
}
APPL_INFO(" blockSize=" << input.size() << " sample");
APPL_INFO(" min < avg < max =" << perfo.getMinProcessing().count() << "ns < "
<< perfo.getTotalTimeProcessing().count()/perfo.getTotalIteration() << "ns < "
<< perfo.getMaxProcessing().count() << "ns ");
float avg = (float(((perfo.getTotalTimeProcessing().count()/perfo.getTotalIteration())*sampleRate)/double(input.size()))/1000000000.0)*100.0;
APPL_INFO(" min < avg < max= " << (float((perfo.getMinProcessing().count()*sampleRate)/double(input.size()))/1000000000.0)*100.0 << "% < "
<< avg << "% < "
<< (float((perfo.getMaxProcessing().count()*sampleRate)/double(input.size()))/1000000000.0)*100.0 << "%");
APPL_PRINT("int16_t : " << _sampleRateIn << " -> " << _sampleRateOut << " quality=" << int32_t(_quality) << " : " << avg << "%");
return avg;
}
void performanceResampler() {
for (int8_t iii=1; iii<=10; ++iii) {
float modeFloat = performanceResamplerStepFloat(8000, 48000, iii);
float modeI16 = performanceResamplerStepI16(8000, 48000, iii);
modeFloat = performanceResamplerStepFloat(16000, 48000, iii);
modeI16 = performanceResamplerStepI16(16000, 48000, iii);
modeFloat = performanceResamplerStepFloat(32000, 48000, iii);
modeI16 = performanceResamplerStepI16(32000, 48000, iii);
modeFloat = performanceResamplerStepFloat(44100, 48000, iii);
modeI16 = performanceResamplerStepI16(44100, 48000, iii);
modeFloat = performanceResamplerStepFloat(48001, 48000, iii);
modeI16 = performanceResamplerStepI16(48001, 48000, iii);
modeFloat = performanceResamplerStepFloat(96000, 48000, iii);
modeI16 = performanceResamplerStepI16(96000, 48000, iii);
modeFloat = performanceResamplerStepFloat(48000, 96000, iii);
modeI16 = performanceResamplerStepI16(48000, 96000, iii);
modeFloat = performanceResamplerStepFloat(48000, 48001, iii);
modeI16 = performanceResamplerStepI16(48000, 48001, iii);
modeFloat = performanceResamplerStepFloat(48000, 44100, iii);
modeI16 = performanceResamplerStepI16(48000, 44100, iii);
modeFloat = performanceResamplerStepFloat(48000, 32000, iii);
modeI16 = performanceResamplerStepI16(48000, 32000, iii);
modeFloat = performanceResamplerStepFloat(48000, 16000, iii);
modeI16 = performanceResamplerStepI16(48000, 16000, iii);
modeFloat = performanceResamplerStepFloat(48000, 8000, iii);
modeI16 = performanceResamplerStepI16(48000, 8000, iii);
}
}
int main(int _argc, const char** _argv) {
// the only one init for etk:
etk::init(_argc, _argv);
std::string inputName = "";
std::string outputName = "output.raw";
bool performance = false;
bool perf = false;
int64_t sampleRateIn = 48000;
int64_t sampleRateOut = 48000;
int32_t nbChan = 1;
int32_t quality = 4;
std::string test = "";
for (int32_t iii=0; iii<_argc ; ++iii) {
std::string data = _argv[iii];
if (etk::start_with(data,"--in=")) {
inputName = &data[5];
} else if (etk::start_with(data,"--out=")) {
outputName = &data[6];
} else if (data == "--performance") {
performance = true;
} else if (data == "--perf") {
perf = true;
} else if (etk::start_with(data,"--test=")) {
data = &data[7];
sampleRateIn = etk::string_to_int32_t(data);
} else if (etk::start_with(data,"--sample-rate-in=")) {
data = &data[17];
sampleRateIn = etk::string_to_int32_t(data);
} else if (etk::start_with(data,"--sample-rate-out=")) {
data = &data[18];
sampleRateOut = etk::string_to_int32_t(data);
} else if (etk::start_with(data,"--nb=")) {
data = &data[5];
nbChan = etk::string_to_int32_t(data);
} else if (etk::start_with(data,"--quality=")) {
data = &data[10];
quality = etk::string_to_int32_t(data);
} else if ( data == "-h"
|| data == "--help") {
APPL_PRINT("Help : ");
APPL_PRINT(" ./xxx --fb=file.raw --mic=file.raw");
APPL_PRINT(" --in=YYY.raw input file");
APPL_PRINT(" --out=zzz.raw output file");
APPL_PRINT(" --performance Generate signal to force algo to maximum process time");
APPL_PRINT(" --perf Enable performence test (little slower but real performence test)");
APPL_PRINT(" --test=XXXX some test availlable ...");
APPL_PRINT(" RESAMPLING Test resampling data 16 bit mode");
APPL_PRINT(" --sample-rate-in=XXXX Input signal sample rate (default 48000)");
APPL_PRINT(" --sample-rate-out=XXXX Output signal sample rate (default 48000)");
APPL_PRINT(" --quality=XX Resampling quality [0..10] (default 4)");
APPL_PRINT(" --nb=XX Number of channel in the file (default 1)");
exit(0);
}
}
// PERFORMANCE test only ....
if (performance == true) {
performanceResampler();
return 0;
}
if (test == "RESAMPLING") {
APPL_INFO("Start resampling test ... ");
if (inputName == "") {
APPL_ERROR("Can not Process missing parameters...");
exit(-1);
}
APPL_INFO("Read input:");
std::vector<int16_t> inputData = etk::FSNodeReadAllDataType<int16_t>(inputName);
APPL_INFO(" " << inputData.size() << " samples");
// resize output :
std::vector<int16_t> output;
output.resize(inputData.size()*sampleRateOut/sampleRateIn+5000, 0);
// process in chunk of 256 samples
int32_t blockSize = 256*nbChan;
Performance perfo;
audio::algo::speex::Resampler algo;
algo.init(nbChan, sampleRateIn, sampleRateOut, quality);
int32_t lastPourcent = -1;
size_t outputPosition = 0;
for (int32_t iii=0; iii<inputData.size()/blockSize; ++iii) {
if (lastPourcent != 100*iii / (inputData.size()/blockSize)) {
lastPourcent = 100*iii / (inputData.size()/blockSize);
APPL_INFO("Process : " << iii*blockSize << "/" << int32_t(inputData.size()/blockSize)*blockSize << " " << lastPourcent << "/100");
} else {
APPL_VERBOSE("Process : " << iii*blockSize << "/" << int32_t(inputData.size()/blockSize)*blockSize);
}
size_t availlableSize = (output.size() - outputPosition) / nbChan;
perfo.tic();
algo.process(&output[outputPosition], availlableSize, &inputData[iii*blockSize], blockSize, audio::format_int16);
if (perf == true) {
perfo.toc();
usleep(1000);
}
outputPosition += availlableSize*nbChan;
}
if (perf == true) {
APPL_INFO("Performance Result: ");
APPL_INFO(" blockSize=" << blockSize << " sample");
APPL_INFO(" min=" << perfo.getMinProcessing().count() << " ns");
APPL_INFO(" max=" << perfo.getMaxProcessing().count() << " ns");
APPL_INFO(" avg=" << perfo.getTotalTimeProcessing().count()/perfo.getTotalIteration() << " ns");
APPL_INFO(" min=" << (float((perfo.getMinProcessing().count()*sampleRateIn)/blockSize)/1000000000.0)*100.0 << " %");
APPL_INFO(" max=" << (float((perfo.getMaxProcessing().count()*sampleRateIn)/blockSize)/1000000000.0)*100.0 << " %");
APPL_INFO(" avg=" << (float(((perfo.getTotalTimeProcessing().count()/perfo.getTotalIteration())*sampleRateIn)/blockSize)/1000000000.0)*100.0 << " %");
}
etk::FSNodeWriteAllDataType<int16_t>(outputName, output);
}
}