164 lines
6.1 KiB
C++
164 lines
6.1 KiB
C++
/** @file
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* @author Edouard DUPIN
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* @copyright 2011, Edouard DUPIN, all right reserved
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* @license APACHE v2.0 (see license file)
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*/
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#include <audio/drain/Resampler.h>
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#include <iostream>
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#include <audio/drain/debug.h>
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audio::drain::Resampler::Resampler() :
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#ifdef HAVE_SPEEX_DSP_RESAMPLE
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m_speexResampler(nullptr),
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#endif
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m_positionRead(0),
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m_positionWrite(0) {
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}
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void audio::drain::Resampler::init() {
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audio::drain::Algo::init();
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m_type = "Resampler";
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m_supportedFormat.push_back(audio::format_int16);
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m_residualTimeInResampler = audio::Duration(0);
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}
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std::shared_ptr<audio::drain::Resampler> audio::drain::Resampler::create() {
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std::shared_ptr<audio::drain::Resampler> tmp(new audio::drain::Resampler());
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tmp->init();
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return tmp;
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}
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audio::drain::Resampler::~Resampler() {
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#ifdef HAVE_SPEEX_DSP_RESAMPLE
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if (m_speexResampler != nullptr) {
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speex_resampler_destroy(m_speexResampler);
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m_speexResampler = nullptr;
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}
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#endif
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}
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void audio::drain::Resampler::configurationChange() {
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audio::drain::Algo::configurationChange();
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if (m_input.getFormat() != m_output.getFormat()) {
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DRAIN_ERROR("can not support Format Change ...");
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m_needProcess = false;
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}
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if (m_input.getFormat() != audio::format_int16) {
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DRAIN_ERROR("can not support Format other than int16_t ...");
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m_needProcess = false;
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return;
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}
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if (m_output.getMap() != m_output.getMap()) {
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DRAIN_ERROR("can not support map Change ...");
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m_needProcess = false;
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}
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if (m_input.getFrequency() == m_output.getFrequency()) {
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// nothing to process...
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m_needProcess = false;
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return;
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}
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if ( m_input.getFrequency() == 0
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|| m_output.getFrequency() == 0) {
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DRAIN_WARNING("Configure IO with 0 frequency ... " << m_input << " to " << m_output);
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return;
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}
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#ifdef HAVE_SPEEX_DSP_RESAMPLE
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if (m_speexResampler != nullptr) {
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speex_resampler_destroy(m_speexResampler);
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m_speexResampler = nullptr;
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}
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int err = 0;
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DRAIN_WARNING("Create resampler for : " << m_input << " to " << m_output);
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m_speexResampler = speex_resampler_init(m_output.getMap().size(),
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m_input.getFrequency(),
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m_output.getFrequency(),
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10, &err);
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m_residualTimeInResampler = audio::Duration(0);
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#else
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DRAIN_WARNING("SPEEX DSP lib not accessible ==> can not resample");
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m_needProcess = false;
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#endif
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}
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bool audio::drain::Resampler::process(audio::Time& _time,
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void* _input,
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size_t _inputNbChunk,
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void*& _output,
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size_t& _outputNbChunk) {
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drain::AutoLogInOut tmpLog("Resampler");
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_outputNbChunk = 2048;
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// chack if we need to process:
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if (m_needProcess == false) {
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DRAIN_WARNING("no process");
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_output = _input;
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_outputNbChunk = _inputNbChunk;
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return true;
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}
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if (_input == nullptr) {
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_output = &(m_outputData[0]);
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_outputNbChunk = 0;
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DRAIN_ERROR("null pointer input ... ");
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return false;
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}
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// Update Output time with the previous delta of the buffer
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DRAIN_VERBOSE("Resampler correct timestamp : " << _time << " ==> " << (_time - m_residualTimeInResampler));
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_time -= m_residualTimeInResampler;
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audio::Duration inTime(0, (int64_t(_inputNbChunk)*1000000000LL) / int64_t(m_input.getFrequency()));
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m_residualTimeInResampler += inTime;
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#ifdef HAVE_SPEEX_DSP_RESAMPLE
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float nbInputTime = float(_inputNbChunk)/m_input.getFrequency();
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float nbOutputSample = nbInputTime*m_output.getFrequency();
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// we add 10% of the buffer size to have all the time enought data in the output to proceed all the input data...
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_outputNbChunk = size_t(nbOutputSample*1.5f);
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DRAIN_VERBOSE(" freq in=" << m_input.getFrequency() << " out=" << m_output.getFrequency());
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DRAIN_VERBOSE(" Frame duration=" << nbInputTime);
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DRAIN_VERBOSE(" nbInput chunk=" << _inputNbChunk << " nbOutputChunk=" << nbOutputSample);
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m_outputData.resize(_outputNbChunk*m_output.getMap().size()*m_formatSize*16);
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_output = &(m_outputData[0]);
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if (m_speexResampler == nullptr) {
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DRAIN_ERROR(" No speex resampler");
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return false;
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}
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uint32_t nbChunkInput = _inputNbChunk;
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uint32_t nbChunkOutput = _outputNbChunk;
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DRAIN_VERBOSE(" >> input=" << nbChunkInput << " output=" << nbChunkOutput);
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int ret = speex_resampler_process_interleaved_int(m_speexResampler,
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static_cast<int16_t*>(_input),
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&nbChunkInput,
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static_cast<int16_t*>(_output),
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&nbChunkOutput);
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DRAIN_VERBOSE(" << input=" << nbChunkInput << " output=" << nbChunkOutput);
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// update position of data:
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m_positionWrite += nbChunkOutput;
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// Check all input and output ...
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if (nbChunkInput != _inputNbChunk) {
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DRAIN_ERROR(" inputSize (not all read ...) proceed=" << nbChunkInput << " requested=" << _inputNbChunk);
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// TODO : manage this case ...
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}
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if (nbChunkOutput == _outputNbChunk) {
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DRAIN_ERROR(" Might have not enought data in output... output size=" << _outputNbChunk);
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// TODO : manage this case ...
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}
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_outputNbChunk = nbChunkOutput;
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DRAIN_VERBOSE(" process chunk=" << nbChunkInput << " out=" << nbChunkOutput);
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audio::Duration outTime(0, (int64_t(_outputNbChunk)*1000000000LL) / int64_t(m_output.getFrequency()));
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DRAIN_VERBOSE("convert " << _inputNbChunk << " ==> " << _outputNbChunk << " " << inTime.count() << " => " << outTime.count());
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// correct time :
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m_residualTimeInResampler -= outTime;
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/*
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if (m_residualTimeInResampler.count() < 0) {
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DRAIN_TODO("manage this case ... residual time in resampler : " << m_residualTimeInResampler.count() << "ns");
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}
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*/
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return true;
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#else
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_output = _input;
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_outputNbChunk = _inputNbChunk;
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return false;
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#endif
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}
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