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/* |
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gig::Engine synthesis benchmark |
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|
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This is a benchmark for testing the performance of the sampler's current |
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Gigasampler format synthesis algorithms. It benchmarks each possible |
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"synthesis mode" (that is all possible cases like filter on / off, |
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interpolation on / off, stereo / mono). It uses fake sample data and |
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fake audio outputs, so we don't have to load a .gig file or care about |
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drivers. |
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|
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Copyright (C) 2005,2006 Christian Schoenebeck <cuse@users.sf.net> |
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*/ |
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|
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#include <math.h> |
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#include <time.h> |
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#include <stdio.h> |
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#include <malloc.h> |
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#include <string.h> |
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|
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#include <gig.h> |
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|
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#include "../src/engines/gig/SynthesisParam.h" |
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#include "../src/engines/gig/Synthesizer.h" |
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|
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#define FRAGMENTSIZE 200 |
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#define RUNS 100000 |
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|
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using namespace LinuxSampler; |
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using namespace LinuxSampler::gig; |
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|
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int16_t* pSampleInputBuf; // just a buffer with random data |
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|
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float* pOutputL; |
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float* pOutputR; |
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|
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void printmode(int mode) { |
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printf("Synthesis Mode: %d ",mode); |
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printf("(%s,DOLOOP=%s,FILTER=%s,INTERPOLATE=%s)\n", |
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(SYNTHESIS_MODE_GET_CHANNELS(mode)) ? "STEREO" : "MONO", |
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(SYNTHESIS_MODE_GET_LOOP(mode)) ? "y" : "n", |
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(SYNTHESIS_MODE_GET_FILTER(mode)) ? "y" : "n", |
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(SYNTHESIS_MODE_GET_INTERPOLATE(mode)) ? "y" : "n" |
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); |
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fflush(stdout); |
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} |
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|
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int main() { |
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pSampleInputBuf = new int16_t[FRAGMENTSIZE*2 + 100]; |
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pOutputL = (float*) memalign(16,FRAGMENTSIZE*sizeof(float)); |
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pOutputR = (float*) memalign(16,FRAGMENTSIZE*sizeof(float)); |
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|
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// prepare some input data for simulation |
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for (int i = 0; i < FRAGMENTSIZE*2; i++) { |
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pSampleInputBuf[i] = i; |
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} |
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|
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SynthesisParam* pParam = new SynthesisParam; |
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pParam->filterLeft.SetParameters(200.0f, 1.0f, 44100); |
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pParam->filterRight.SetParameters(200.0f, 1.0f, 44100); |
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pParam->fFinalPitch = 0.5f; |
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pParam->fFinalVolumeLeft = 1.0f; |
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pParam->fFinalVolumeRight = 1.0f; |
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/* For now we skip the CONFIG_INTERPOLATE_VOLUME case |
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pParam->fFinalVolumeDeltaLeft = ... |
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pParam->fFinalVolumeDeltaRight = ... */ |
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pParam->pSrc = pSampleInputBuf; |
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|
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// define some loop points |
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Loop* pLoop = new Loop; |
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pLoop->uiStart = 4; |
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pLoop->uiEnd = 20; |
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pLoop->uiSize = 16; |
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pLoop->uiTotalCycles = 0; // infinity |
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|
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for (int mode = 0; mode < 16; mode++) { |
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// zero out output buffers |
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memset(pOutputL,0,FRAGMENTSIZE*sizeof(float)); |
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memset(pOutputR,0,FRAGMENTSIZE*sizeof(float)); |
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|
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pParam->filterLeft.Reset(); |
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pParam->filterRight.Reset(); |
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|
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printf("Benchmarking "); |
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printmode(mode); |
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|
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clock_t stop_time; |
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clock_t start_time = clock(); |
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|
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for (uint i = 0; i < RUNS; i++) { |
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pParam->dPos = 0.0; |
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pParam->pOutLeft = pOutputL; |
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pParam->pOutRight = pOutputR; |
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pParam->uiToGo = FRAGMENTSIZE; |
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// now actually render audio |
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RunSynthesisFunction(mode, pParam, pLoop); |
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} |
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|
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stop_time = clock(); |
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float elapsed_time = (stop_time - start_time) / (double(CLOCKS_PER_SEC) / 1000.0); |
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printf("\t: %1.0f ms\n", elapsed_time); |
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} |
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} |
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