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+ of Leq(M) measurement according to ISO 21727:2004(E)
+ "Cinematography -- Method of measurement of perceived
+ loudness of motion-picture audio material"
+
+ Copyright (C) 2011-2013, 2017 Luca Trisciani
+
+ This program is free software: you can redistribute it and/or modify
+ it under the terms of the GNU General Public License as published by
+ the Free Software Foundation, either version 3 of the License, or
+ (at your option) any later version.
+
+ This program is distributed in the hope that it will be useful,
+ but WITHOUT ANY WARRANTY; without even the implied warranty of
+ MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ GNU General Public License for more details.
+
+ You should have received a copy of the GNU General Public License
+ along with this program. If not, see <http://www.gnu.org/licenses/>.
+
+ */
+
+
+
+#include <stdio.h>
+#include <math.h>
+#include <sndfile.h>
+#include <unistd.h>
+#include <pthread.h>
+#include <string.h>
+#include <stdlib.h>
+#include <time.h>
+#include <ctype.h>
+
+
+// Version 0.0.17 (C) Luca Trisciani 2011-2013, 2017
+// Tool from the DCP-Werkstatt Software Bundle
+
+
+
+// COMPILATION
+// compile for DEBUG with gcc -g -DEBUG -lsndfile -lfftw3 -lm -lpthread -lrt -o leqm-nrt leqm-nrt.cpp
+//otherwise gcc -lsndfile -lm -lpthread -lrt -o leqm-nrt leqm-nrt.c
+
+//#define DEBUG
+
+
+struct Sum {
+ double csum; // convolved sum
+ double sum; // flat sum
+ int nsamples;
+ double cmean; //convolved mean
+ double mean;
+ double leqm;
+ double rms;
+};
+
+struct WorkerArgs {
+ double * argbuffer;
+ int nsamples;
+ int nch;
+ int npoints;
+ double * ir;
+ struct Sum * ptrtotsum;
+ double * chconf;
+ int shorttermindex;
+ double * shorttermarray;
+ int leqm10flag;
+};
+
+int equalinterval( double * freqsamples, double * freqresp, double * eqfreqsamples, double * eqfreqresp, int points, int samplingfreq, int origpoints);
+int equalinterval2( double freqsamples[], double * freqresp, double * eqfreqsamples, double * eqfreqresp, int points, int samplingfreq, int origpoints, int bitdepthsoundfile);
+int convloglin(double * in, double * out, int points);
+double convlinlog_single(double in);
+double convloglin_single(double in);
+int convolv_buff(double * sigin, double * sigout, double * impresp, int sigin_dim, int impresp_dim);
+double inputcalib (double dbdiffch);
+int rectify(double * squared, double * inputsamples, int nsamples);
+int accumulatech(double * chaccumulator, double * inputchannel, int nsamples);
+int sumsamples(struct Sum * ts, double * inputsamples, double * cinputsamples, int nsamples);
+int meanoverduration(struct Sum * oldsum);
+void inversefft1(double * eqfreqresp, double * ir, int npoints);
+void inversefft2(double * eqfreqresp, double * ir, int npoints);
+void * worker_function(void * argfunc);
+void logleqm(FILE * filehandle, double featuretimesec, struct Sum * oldsum);
+double sumandshorttermavrg(double * channelaccumulator, int nsamples);
+void logleqm10(FILE * filehandle, double featuretimesec, double longaverage);
+
+
+pthread_mutex_t mutex = PTHREAD_MUTEX_INITIALIZER;
+
+
+int main(int argc, const char ** argv)
+{
+ int npoints = 64; // This value is low for precision. Calibration is done with 32768 point.
+ int origpoints = 21; //number of points in the standard CCIR filter
+ int samplingfreq; // this and the next is defined later taking it from sound file
+ int bitdepth;
+ // double normalizer;
+ int timing = 0;
+ struct timespec starttime;
+ int fileopenstate = 0;
+ int leqm10 = 0;
+ int leqmlog = 0;
+ int numCPU = sysconf(_SC_NPROCESSORS_ONLN) - 1;
+ double * channelconfcalvector;
+ channelconfcalvector = NULL;
+ printf("leqm-nrt Copyright (C) 2011-2013, 2017 Luca Trisciani\nThis program comes with ABSOLUTELY NO WARRANTY; for details on command line parameters -help\nThis is free software, and you are welcome to redistribute it\nunder the GPL v3 licence.\nProgram will use 1 + %d slave threads.\n", numCPU);
+ //SndfileHandle file;
+ SNDFILE *file;
+ file=NULL;
+ SF_INFO sfinfo;
+ FILE *leqm10logfile;
+ leqm10logfile = NULL;
+ FILE *leqmlogfile;
+ leqmlogfile = NULL;
+ int buffersizems = 850; //ISO 21727:2004 do not contain any indication, TASA seems to indicate 1000, p. 8
+ int buffersizesamples;
+ double tempchcal[128];
+ int numcalread = 0;
+ double * shorttermaveragedarray;
+ shorttermaveragedarray = NULL;
+ int numbershortperiods = 0;
+ int parameterstate = 0;
+ int leqnw = 0;
+
+ char soundfilename[64];
+ // This is a requirement of sndfile library, do not forget it.
+
+ memset(&sfinfo, 0, sizeof(sfinfo));
+
+
+ if (argc == 1)
+ { const char helptext[] = "Order of parameters is free.\nPossible parameters are:\n-convpoints <integer number> \tNumber of interpolation points for the filter. Default 64\n-numcpus <integer number> \tNumber of slave threads to speed up operation.\n-timing \t\t\tFor benchmarking speed.\n-chconfcal <db correction> <db correction> <etc. so many times as channels>\n-logleqm10\n-logleqm\n-buffersize <milliseconds>\n";
+ printf(helptext);
+ printf("Please indicate a sound file to be processed.\n");
+ return 0;
+ }
+
+
+ for (int in = 1; in < argc;) {
+
+ if (!(strncmp(argv[in], "-", 1) == 0)) {
+ if (fileopenstate == 0) {
+ if(! (file = sf_open(argv[in], SFM_READ, &sfinfo))) {
+ printf("Error while opening audio file, could not open %s\n.", argv[in]);
+ puts(sf_strerror(NULL));
+ return 1;
+ }
+
+ strcpy(soundfilename, argv[in]);
+ fileopenstate = 1;
+ printf("Opened file: %s\n", argv[in]);
+ printf("Sample rate: %d\n", sfinfo.samplerate);
+ printf("Channels: %d\n", sfinfo.channels);
+ printf("Format: %d\n", sfinfo.format);
+ printf("Frames: %d\n", (int) sfinfo.frames);
+ channelconfcalvector = malloc(sizeof(double) * sfinfo.channels);
+ in++;
+ continue;
+ } else {
+ free(channelconfcalvector);
+ channelconfcalvector = NULL;
+ return 0;
+ }
+ }
+
+
+ if (strcmp(argv[in], "-chconfcal") == 0) {
+ /* as the order of parameter is free I have to postpone
+ the check for consistency with the number of channels.
+ So first create a temporary array, whose number of element will be checked after
+ the parsing of the command line parameters is finished.
+ The calibration will be expressed in dB on the command line and converted to multiplier
+ here so that it can be stored as a factor in the channelconfcalvector.
+ */
+
+ in++;
+ for (;;) {
+ if (in < argc) {
+ //if (!(strncmp(argv[in], "-", 1) == 0)) { //changed this to allow negative numbers
+ if (!(strncmp(argv[in], "-", 1) == 0) || isdigit(argv[in][1])) {
+ tempchcal[numcalread++]=atof(argv[in++]);
+ } else break;
+ } else break;
+
+ } //for
+ continue;
+ }
+
+ if (strcmp(argv[in], "-convpoints") == 0) {
+ npoints = atoi(argv[in + 1]);
+ in+=2;
+ printf("Convolution points sets to %d.\n", npoints);
+ continue;
+
+ }
+ if (strcmp(argv[in], "-numcpus") == 0) {
+ numCPU= atoi(argv[in + 1]);
+ in+=2;
+ printf("Number of threads manually set to %d. Default is number of cores in the system minus one.\n", numCPU);
+ continue;
+
+ }
+ if (strcmp(argv[in], "-timing") == 0) {
+ timing = 1;
+ in++;
+ printf("Execution time will be measured.\n");
+ continue;
+
+ }
+
+ if (strcmp(argv[in], "-logleqm10") == 0) {
+ leqm10 = 1;
+ in++;
+ printf("Leq(M)10 data will be logged to the file leqm10.txt\n");
+ continue;
+
+ }
+ if (strcmp(argv[in], "-logleqm") == 0) {
+ leqmlog = 1;
+ in++;
+ printf("Leq(M) data will be logged to the file leqmlog.txt\n");
+ continue;
+
+ }
+
+ if (strcmp(argv[in], "-leqnw") == 0) {
+ leqnw = 1;
+ in++;
+ printf("Leq(nW) - unweighted - will be outputted.\n");
+ continue;
+
+ }
+
+ if (strcmp(argv[in], "-buffersize") == 0) {
+ buffersizems = atoi(argv[in + 1]);
+ in+=2;
+ printf("Buffersize will be set to %d milliseconds.\n", buffersizems);
+ continue;
+
+ }
+
+ if (parameterstate==0) {
+ break;
+ }
+ }
+// Open audio file
+
+//postprocessing parameters
+ if (numcalread == sfinfo.channels) {
+ for (int cind = 0; cind < sfinfo.channels; cind++) {
+ channelconfcalvector[cind] = convloglin_single(tempchcal[cind]);
+
+ }
+ }
+ else if ((numcalread == 0) && (sfinfo.channels == 6)) {
+ double conf51[6] = {0, 0, 0, 0, -3, -3};
+ for (int cind = 0; cind < sfinfo.channels; cind++) {
+ channelconfcalvector[cind] = convloglin_single(conf51[cind]);
+ }
+
+ } else {
+
+ printf("Either you specified a different number of calibration than number of channels in the file or you do not indicate any calibration and the program cannot infer one from the number of channels. Please specify a channel calibration on the command line.\n");
+
+ free(channelconfcalvector);
+ channelconfcalvector = NULL;
+ return 0;
+ }
+
+
+
+
+ if (leqm10) {
+ char tempstring[128];
+ strcpy(tempstring, soundfilename);
+ strcat(tempstring, ".leqm10.txt");
+ leqm10logfile = fopen(tempstring, "w");
+ if (leqm10logfile == NULL) {
+ printf("Could not open file to write log leqm10 data!\n");
+ }
+ }
+
+
+
+
+ if (leqmlog) {
+ char tempstring[128];
+ strcpy(tempstring, soundfilename);
+ strcat(tempstring, ".leqmlog.txt");
+ leqmlogfile = fopen(tempstring, "w");
+ if (leqmlogfile == NULL) {
+ printf("Could not open file to write log leqm data!\n");
+ }
+ }
+
+
+ if (timing) {
+
+ clock_gettime(CLOCK_MONOTONIC, &starttime);
+ }
+
+
+
+
+
+ // reading to a double or float buffer with sndfile take care of normalization
+ /*
+ static double buffer[BUFFER_LEN]; // it seems this must be static. I don't know why
+ */
+ double * buffer;
+ // buffer = new double [BUFFER_LEN];
+ buffersizesamples = (sfinfo.samplerate*sfinfo.channels*buffersizems)/1000;
+ if ((sfinfo.samplerate*buffersizems)%1000) {
+ printf("Please fine tune the buffersize according to the sample rate\n");
+ //close file
+ // free memory
+ // write a function to do that
+ return 1;
+ }
+
+ buffersizesamples = (sfinfo.samplerate*sfinfo.channels*buffersizems)/1000;
+ buffer = malloc(sizeof(double)*buffersizesamples);
+
+ samplingfreq = sfinfo.samplerate;
+
+ if(leqm10) {
+
+ //if duration < 10 mm exit
+
+ double featdursec = sfinfo.frames / sfinfo.samplerate;
+ if ((featdursec/60.0) < 10.0) {
+ printf("The audio file is too short to measure Leq(m10).\n");
+ return 0;
+ }
+
+ //how many short periods in overall duration
+ int remainder = sfinfo.frames % (sfinfo.samplerate*buffersizems/1000);
+ if (remainder == 0) numbershortperiods = sfinfo.frames/(sfinfo.samplerate*buffersizems/1000);
+ else numbershortperiods = sfinfo.frames/(sfinfo.samplerate*buffersizems/1000) + 1;
+
+ //allocate array
+ shorttermaveragedarray = malloc(sizeof(*shorttermaveragedarray)*numbershortperiods);
+ }
+
+
+ //End opening audio file
+
+ //ISO 21727:2004(E)
+ // M Weighting
+ double freqsamples[] = {31, 63, 100, 200, 400, 800, 1000, 2000, 3150, 4000, 5000, 6300, 7100, 8000, 9000, 10000, 12500, 14000, 16000, 20000, 31500};
+ double freqresp_db[] = {-35.5, -29.5, -25.4, -19.4, -13.4, -7.5, -5.6, 0.0, 3.4, 4.9, 6.1, 6.6, 6.4, 5.8, 4.5, 2.5, -5.6, -10.9, -17.3, -27.8, -48.3};
+
+ double * eqfreqresp_db;
+ eqfreqresp_db = malloc(sizeof(*eqfreqresp_db)*npoints);
+
+ double * eqfreqsamples;
+ eqfreqsamples = malloc(sizeof(*eqfreqsamples)*npoints);
+ double * eqfreqresp;
+ eqfreqresp = malloc(sizeof(*eqfreqresp)*npoints);
+ double * ir;
+ ir = malloc(sizeof(*ir)*npoints*2);
+
+
+// And what to do for floating point sample coding?
+
+ switch(sfinfo.format & SF_FORMAT_SUBMASK) {
+ // all signed bitdepth
+ case 0x0001:
+ bitdepth = 8;
+ break;
+ case 0x0002:
+ bitdepth = 16;
+ break;
+ case 0x0003:
+ bitdepth = 24;
+ break;
+ case 0x0004:
+ bitdepth = 32;
+ break;
+ default:
+ printf("No known bitdepth! Exiting ...\n");
+ return -1;
+ }
+
+
+
+
+ equalinterval2(freqsamples, freqresp_db, eqfreqsamples, eqfreqresp_db, npoints, samplingfreq, origpoints, bitdepth);
+ convloglin(eqfreqresp_db, eqfreqresp, npoints);
+
+ #ifdef DEBUG
+ for (int i=0; i < npoints; i++) {
+ printf("%d\t%.2f\t%.2f\t%.2f\n", i, eqfreqsamples[i], eqfreqresp_db[i], eqfreqresp[i]);
+ }
+ #endif
+
+ inversefft2(eqfreqresp, ir, npoints);
+
+// read through the entire file
+
+ struct Sum * totsum;
+ totsum = malloc(sizeof(struct Sum));
+ totsum->csum = 0.0;
+ totsum->sum = 0.0;
+ totsum->nsamples = 0;
+ totsum->cmean = 0.0;
+ totsum->mean = 0.0; // Do I write anything here?
+ totsum->leqm = 0.0;
+ totsum->rms = 0.0;
+ sf_count_t samples_read = 0;
+
+ // Main loop through audio file
+
+ int worker_id = 0;
+ pthread_t tid[numCPU];
+ struct WorkerArgs ** WorkerArgsArray;
+ WorkerArgsArray = malloc(sizeof(struct WorkerArgs *)*numCPU);
+ int staindex = 0; //shorttermarrayindex
+
+
+ while((samples_read = sf_read_double(file, buffer, buffersizesamples)) > 0) {
+
+ WorkerArgsArray[worker_id]=malloc(sizeof(struct WorkerArgs));
+ WorkerArgsArray[worker_id]->nsamples = samples_read;
+ WorkerArgsArray[worker_id]->nch = sfinfo.channels;
+ WorkerArgsArray[worker_id]->npoints=npoints;
+ WorkerArgsArray[worker_id]->ir = ir;
+ WorkerArgsArray[worker_id]->ptrtotsum = totsum;
+
+ WorkerArgsArray[worker_id]->chconf = channelconfcalvector;
+ if (leqm10) {
+ WorkerArgsArray[worker_id]->shorttermindex = staindex++;
+ WorkerArgsArray[worker_id]->leqm10flag = 1;
+ WorkerArgsArray[worker_id]->shorttermarray = shorttermaveragedarray;
+ } else {
+ WorkerArgsArray[worker_id]->shorttermindex = 0;
+ WorkerArgsArray[worker_id]->leqm10flag = 0;
+ }
+
+ WorkerArgsArray[worker_id]->argbuffer = malloc(sizeof(double)*buffersizesamples);
+ memcpy(WorkerArgsArray[worker_id]->argbuffer, buffer, samples_read*sizeof(double));
+
+
+ pthread_attr_t attr;
+ pthread_attr_init(&attr);
+ pthread_create(&tid[worker_id], &attr, worker_function, WorkerArgsArray[worker_id]);
+
+ worker_id++;
+ if (worker_id == numCPU) {
+ worker_id = 0;
+ //maybe here wait for all cores to output before going on
+ for (int idxcpu = 0; idxcpu < numCPU; idxcpu++) {
+ pthread_join(tid[idxcpu], NULL);
+ free(WorkerArgsArray[idxcpu]);
+ WorkerArgsArray[idxcpu] = NULL;
+ }
+ //simply log here your measurement it will be a multiple of your threads and your buffer
+ if (leqmlog) {
+ meanoverduration(totsum); //update leq(m) until now and log it
+ logleqm(leqmlogfile, ((double) totsum->nsamples)/((double) sfinfo.samplerate), totsum );
+ } //endlog
+ }
+
+
+
+ //end while worker_id
+ /// End looping cores
+ } // main loop through file
+
+ //here I should wait for rest worker (< numcpu)
+ //but I need to dispose of thread id.
+ if (worker_id != 0) { // worker_id = 0 means the number of samples was divisible through the number of cpus
+ for (int idxcpu = 0; idxcpu < worker_id; idxcpu++) { //worker_id is at this point one unit more than threads launched
+ pthread_join(tid[idxcpu], NULL);
+ free(WorkerArgsArray[idxcpu]);
+ WorkerArgsArray[idxcpu] = NULL;
+ }
+ //also log here for a last value
+ if (leqmlog) {
+ meanoverduration(totsum); //update leq(m) until now and log it
+ logleqm(leqmlogfile, ((double) totsum->nsamples)/((double) sfinfo.samplerate), totsum );
+ } //endlog
+ }
+ // mean of scalar sum over duration
+
+ meanoverduration(totsum);
+ if (leqnw) {
+ printf("Leq(noW): %.4f\n", totsum->rms); // Leq(no Weighting)
+ }
+ printf("Leq(M): %.4f\n", totsum->leqm);
+
+ if(timing) {
+ struct timespec stoptime;
+ long stoptimenanoseconds;
+ long executionnanoseconds;
+ clock_gettime(CLOCK_MONOTONIC, &stoptime);
+
+ if (stoptime.tv_nsec < starttime.tv_nsec) {
+ stoptimenanoseconds = 1000000000 + stoptime.tv_nsec;
+ } else {
+ stoptimenanoseconds = stoptime.tv_nsec;
+ }
+ executionnanoseconds = stoptimenanoseconds - starttime.tv_nsec;
+ printf("Total execution time is %.6f seconds\n", ((double) stoptime.tv_sec) - ((double) starttime.tv_sec) + ((double) executionnanoseconds / 1000000000.00));
+ }
+
+
+ if (leqm10) {
+
+ //Take the array with the short term accumulators
+ double interval = 10.0;
+ //create a rolling average according to rolling interval
+ int rollint; // in short 10*60 = 600 sec 600/0.850
+
+
+ //how many element of the array to consider for the rollint?
+ //that is how many buffersizems in the interval - interval could be parameterized(?)
+ double tempint = 60.0 * interval / (((double) buffersizems) /1000.0);
+ rollint = (int) tempint;
+ //dispose of the rest
+ if (tempint - ((double) rollint) > 0) {
+ rollint += 1;
+ }
+ //two loops
+ //external loop
+ int indexlong = 0;
+ while(indexlong < (numbershortperiods - rollint)) {
+
+ double accumulator = 0;
+ //internal loop
+ double averagedaccumulator = 0;
+ for (int indexshort = 0; indexshort < rollint; indexshort++) {
+
+ accumulator += shorttermaveragedarray[indexshort+indexlong];
+ } //end internal loop
+ averagedaccumulator = accumulator/((double) rollint);
+ logleqm10(leqm10logfile, ((double) (indexlong+rollint)) * ((double) buffersizems / 1000.0), averagedaccumulator);
+ indexlong++;
+ } //end external loop
+
+ fclose(leqm10logfile);
+ free(shorttermaveragedarray);
+ shorttermaveragedarray = NULL;
+ }
+
+
+ if (leqmlog) {
+
+ fclose(leqmlogfile);
+ }
+
+ sf_close(file);
+
+ free(eqfreqsamples);
+ eqfreqsamples = NULL;
+ free(eqfreqresp_db);
+ eqfreqresp_db=NULL;
+ free(eqfreqresp);
+ eqfreqresp = NULL;
+ free(ir);
+ ir = NULL;
+ free(channelconfcalvector);
+ channelconfcalvector = NULL;
+ free(WorkerArgsArray);
+ WorkerArgsArray = NULL;
+
+ free(totsum);
+ totsum = NULL;
+ free(buffer);
+ buffer=NULL;
+ return 0;
+}
+
+
+
+void * worker_function(void * argstruct) {
+
+ struct WorkerArgs * thisWorkerArgs = (struct WorkerArgs *) argstruct;
+
+ double * sumandsquarebuffer;
+ double * csumandsquarebuffer;
+ double * chsumaccumulator_norm;
+ double * chsumaccumulator_conv;
+
+
+ sumandsquarebuffer = malloc(sizeof(double)*(thisWorkerArgs->nsamples / thisWorkerArgs->nch));
+
+
+ csumandsquarebuffer = malloc(sizeof(double)*(thisWorkerArgs->nsamples / thisWorkerArgs->nch));
+
+ chsumaccumulator_norm = malloc(sizeof(double)*(thisWorkerArgs->nsamples / thisWorkerArgs->nch));
+
+ chsumaccumulator_conv = malloc(sizeof(double)*(thisWorkerArgs->nsamples / thisWorkerArgs->nch));
+
+
+
+ for (int i = 0; i < thisWorkerArgs->nsamples / thisWorkerArgs->nch; i++) {
+ sumandsquarebuffer[i] = 0.0;
+ csumandsquarebuffer[i] = 0.0;
+ chsumaccumulator_norm[i] = 0.0;
+ chsumaccumulator_conv[i] = 0.0;
+ }
+
+
+
+ for (int ch = 0; ch < thisWorkerArgs->nch; ch++) {
+
+ double * normalizedbuffer;
+ double * convolvedbuffer;
+
+
+ normalizedbuffer = malloc(sizeof(double)*(thisWorkerArgs->nsamples / thisWorkerArgs->nch));
+
+ convolvedbuffer = malloc(sizeof(double)*(thisWorkerArgs->nsamples / thisWorkerArgs->nch));
+
+
+ for (int n=ch, m= 0; n < thisWorkerArgs->nsamples; n += thisWorkerArgs->nch, m++) {
+ // use this for calibration depending on channel config for ex. chconf[6] = {1.0, 1.0, 1.0, 1.0, 0.707945784, 0.707945784} could be the default for 5.1 soundtracks
+ //so not normalized but calibrated
+ normalizedbuffer[m] = thisWorkerArgs->argbuffer[n]*thisWorkerArgs->chconf[ch]; //this scale amplitude according to specified calibration
+
+
+ }
+
+ //convolution
+ convolv_buff(normalizedbuffer, convolvedbuffer, thisWorkerArgs->ir, thisWorkerArgs->nsamples / thisWorkerArgs->nch, thisWorkerArgs->npoints * 2);
+ //rectify, square und sum
+ rectify(csumandsquarebuffer,convolvedbuffer, thisWorkerArgs->nsamples / thisWorkerArgs->nch);
+ rectify(sumandsquarebuffer,normalizedbuffer, thisWorkerArgs->nsamples / thisWorkerArgs->nch);
+
+ accumulatech(chsumaccumulator_norm, sumandsquarebuffer, thisWorkerArgs->nsamples / thisWorkerArgs->nch);
+ accumulatech(chsumaccumulator_conv, csumandsquarebuffer, thisWorkerArgs->nsamples / thisWorkerArgs->nch);
+
+
+ free(normalizedbuffer);
+ normalizedbuffer= NULL;
+
+ free(convolvedbuffer);
+ convolvedbuffer=NULL;
+
+ } // loop through channels
+
+ //Create a function for this also a tag so that the worker know if he has to do this or not
+
+ if (thisWorkerArgs->leqm10flag) {
+ thisWorkerArgs->shorttermarray[thisWorkerArgs->shorttermindex] = sumandshorttermavrg(chsumaccumulator_conv, thisWorkerArgs->nsamples / thisWorkerArgs->nch);
+ #ifdef DEBUG
+ printf("%d: %.6f\n", thisWorkerArgs->shorttermindex, thisWorkerArgs->shorttermarray[thisWorkerArgs->shorttermindex]);
+ #endif
+ }
+ pthread_mutex_lock(&mutex);
+ // this should be done under mutex conditions -> shared resources!
+ sumsamples(thisWorkerArgs->ptrtotsum, chsumaccumulator_norm, chsumaccumulator_conv, thisWorkerArgs->nsamples / thisWorkerArgs->nch);
+ pthread_mutex_unlock(&mutex);
+
+
+ free(sumandsquarebuffer);
+ sumandsquarebuffer=NULL;
+
+ free(csumandsquarebuffer);
+ csumandsquarebuffer=NULL;
+
+ free(chsumaccumulator_norm);
+ chsumaccumulator_norm=NULL;
+
+ free(chsumaccumulator_conv);
+ chsumaccumulator_conv=NULL;
+
+ free(thisWorkerArgs->argbuffer);
+ thisWorkerArgs->argbuffer = NULL;
+ // the memory pointed to by this pointer is freed in main
+ // it is the same memory for all worker
+ // but it is necessary to set pointer to NULL otherwise free will not work later (really?)
+ thisWorkerArgs->chconf = NULL;
+ pthread_exit(0);
+
+}
+
+
+//to get impulse response frequency response at equally spaced intervals is needed
+
+int equalinterval( double * freqsamples, double * freqresp, double * eqfreqsamples, double * eqfreqresp, int points, int samplingfreq, int origpoints) {
+ double freq;
+ // int findex = 0;
+ // int rindex = 0;
+ double pass = ((double) (samplingfreq >> 1)) / ((double) points);
+ for (int ieq = 0, i = 0; ieq < points; ieq++) {
+ freq = ieq*pass;
+ eqfreqsamples[ieq] = freq;
+
+ if ((freq == 0.0) || (freq < freqsamples[1])) {
+ eqfreqresp[ieq] = freqresp[0];
+ continue;
+ } else {
+
+ if ((freq >= freqsamples[i]) && (freq < freqsamples[i+1])) {
+ eqfreqresp[ieq] = ((freqresp[i+1] - freqresp[i])/(freqsamples[i+1] - freqsamples[i]))*(freq - freqsamples[i]) + freqresp[i];
+ } else if (freq >=freqsamples[i+1]) {
+ while(freq >= freqsamples[i+1]) {
+ i++;
+ if ((i + 1) >= origpoints) {
+ break;
+ }
+ }
+ if ((i+1) < origpoints) {
+ eqfreqresp[ieq] = ((freqresp[i+1] - freqresp[i])/(freqsamples[i+1] - freqsamples[i]))*(freq- freqsamples[i]) + freqresp[i];
+ } else {
+ eqfreqresp[ieq] = ((1 - freqresp[i])/(((double) (samplingfreq >> 1)) - freqsamples[i]))*(freq- freqsamples[i]) + freqresp[i];
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+
+
+
+
+//the following is different from version 1 because interpolate between db and not linear. Conversion from db to lin must be done after.
+//it is also different for the way it interpolates between DC and 31 Hz
+// Pay attention that also arguments to the functions are changed
+int equalinterval2( double freqsamples[], double freqresp_db[], double * eqfreqsamples, double * eqfreqresp, int points, int samplingfreq, int origpoints, int bitdepthsoundfile) {
+ double freq;
+
+
+ //calculate miminum attenuation depending on the bitdeph (minus one), that is −6.020599913 dB per bit in eccess to sign
+ double dcatt = ((double) (bitdepthsoundfile - 1))*(-6.020599913) + 20.00; //in dB
+ //double dcatt = -90.3;
+ double pass = ((double) (samplingfreq >> 1)) / ((double) points);
+ for (int ieq = 0, i = 0; ieq < points; ieq++) {
+ freq = ieq*pass;
+ eqfreqsamples[ieq] = freq;
+
+ if (freq == 0.0) {
+ eqfreqresp[ieq] = dcatt;
+ } else if (freq < freqsamples[0]) { // this has a lot of influence on final Leq(M) value
+ eqfreqresp[ieq] = ((freqresp_db[0] - dcatt) / (freqsamples[0] - 0)) * freq + dcatt;
+ //eqfreqresp[ieq] = freqresp_db[0]; // Is this meaningful? Shouldn't I interpolate between 0 Hz and 31 Hz? Otherwise for DC I have -35.5 dB
+ continue;
+ } else {
+
+ if ((freq >= freqsamples[i]) && (freq < freqsamples[i+1])) {
+ eqfreqresp[ieq] = ((freqresp_db[i+1] - freqresp_db[i])/(freqsamples[i+1] - freqsamples[i]))*(freq - freqsamples[i]) + freqresp_db[i];
+ } else if (freq >=freqsamples[i+1]) {
+ while(freq >= freqsamples[i+1]) {
+ i++;
+ if ((i + 1) >= origpoints) {
+ break;
+ }
+ }
+ if ((i+1) < origpoints) {
+ eqfreqresp[ieq] = ((freqresp_db[i+1] - freqresp_db[i])/(freqsamples[i+1] - freqsamples[i]))*(freq- freqsamples[i]) + freqresp_db[i];
+ } else {
+ eqfreqresp[ieq] = ((1 - freqresp_db[i])/(((double) (samplingfreq >> 1)) - freqsamples[i]))*(freq- freqsamples[i]) + freqresp_db[i];
+ }
+ }
+ }
+ }
+ return 0;
+}
+
+
+
+
+
+
+int convloglin(double * in, double * out, int points) {
+ for (int i = 0; i < points; i++) {
+ out[i] = powf(10, (in[i]/20.0));
+ }
+
+ return 0;
+}
+
+
+double convlinlog_single(double in) {
+ double out;
+ out = log(in)*20.0f;
+ return out;
+}
+
+
+double convloglin_single(double in) {
+ double out;
+ out = powf(10, in/20.0f);
+ return out;
+}
+
+// convolution
+
+int convolv_buff(double * sigin, double * sigout, double * impresp, int sigin_dim, int impresp_dim) {
+
+
+ double sum = 0.0;
+ for (int i = 0; i < sigin_dim; i++) {
+
+ int m = i;
+ for (int l = impresp_dim - 1; l >=0; l--,m++) {
+ if (m >= sigin_dim) {
+ m -= sigin_dim;
+ }
+ sum += sigin[m]*impresp[l];
+ }
+ sigout[i] = sum;
+ sum=0.0;
+ }
+ return 0;
+
+}
+
+
+void inversefft2(double * eqfreqresp, double * ir, int npoints) {
+ for (int n = 0; n < npoints; n++) {
+ double parsum = 0.0;
+ double partial = 0.0;
+
+ for (int m = 1; m <= npoints -1; m++) {
+ partial = cos(2.0*M_PI*((double) m)*( ( ((double) n) - ( ((double) npoints) * 2.0 -1 ) / 2 ) / ( ((double) npoints) * 2.0) ));
+ parsum = parsum + eqfreqresp[m]*partial;
+ }
+ ir[n] = (eqfreqresp[0] + 2.0 * parsum)/((double) npoints * 2.0);
+ #ifdef DEBUG
+ printf("%.4f\n", ir[n]);
+ #endif
+ }
+ for (int n = 0; n < npoints; n++) {
+ ir[npoints+n] = ir[npoints-(n + 1)];
+ #ifdef DEBUG
+ printf("%.4f\n", ir[npoints+n]);
+ #endif
+ }
+
+
+}
+
+// scale input according to required calibration
+// this could be different for certain digital cinema formats
+double inputcalib (double dbdiffch) {
+
+ double coeff = pow(10, dbdiffch/20);
+ return coeff;
+
+}
+
+//rectify, square and sum
+int rectify(double * squared, double * inputsamples, int nsamples){
+ for (int i = 0; i < nsamples; i++) {
+ squared[i] = (double) powf(inputsamples[i], 2);
+ }
+ return 0;
+
+}
+
+int initbuffer(double * buffertoinit, int nsamples) {
+ for (int i = 0; i < nsamples; i++) {
+ buffertoinit[i] = 0.0;
+
+ }
+ return 0;
+}
+
+int accumulatech(double * chaccumulator, double * inputchannel, int nsamples) {
+ for (int i = 0; i < nsamples; i++) {
+ chaccumulator[i] += inputchannel[i];
+ }
+ return 0;
+}
+
+int sumsamples(struct Sum * ts, double * inputsamples, double * cinputsamples, int nsamples) {
+ ts->nsamples += nsamples;
+ for (int i=0; i < nsamples; i++) {
+ ts->sum += inputsamples[i];
+ ts->csum += cinputsamples[i];
+ }
+ return 0;
+
+}
+
+int meanoverduration(struct Sum * oldsum) {
+ oldsum->mean = pow(oldsum->sum / ((double) oldsum->nsamples), 0.500);
+ oldsum->cmean = pow(oldsum->csum / ((double) oldsum->nsamples), 0.500);
+ oldsum->rms = 20*log10(oldsum->mean);
+ oldsum->leqm = 20*log10(oldsum->cmean) + 110.600;//
+ // and this must be right because M filter is -5.6 @ 1k Hz that is -25.6 dBFS and to have 85.0 as reference level we must add 25.56 + 85.00 that is 110.6 dB.
+ //this value is obtained calibrating with a -20 dBFS Dolby Tone (RMS) I think this is correct
+ //But ISO 21727:2004(E) ask for a reference level "measured using an average responding meter". So reference level is not 0.707, but 0.637 = 2/pi
+ //This is only approximate as you should use a separate calibration according to the Dolby Format. Also for SW the tone should be 100Hz (?)
+
+return 0;
+}
+
+double sumandshorttermavrg(double * channelaccumulator, int nsamples) {
+ double stsum = 0.0;
+ for (int i=0; i < nsamples; i++) {
+ stsum += channelaccumulator[i];
+
+ }
+ return stsum / (double) nsamples;
+}
+
+void logleqm(FILE * filehandle, double featuretimesec, struct Sum * oldsum) {
+
+ fprintf(filehandle, "%.4f", featuretimesec);
+ fprintf(filehandle, "\t");
+ fprintf(filehandle, "%.4f\n", oldsum->leqm);
+
+
+}
+
+void logleqm10(FILE * filehandle, double featuretimesec, double longaverage) {
+ double leqm10 = 20*log10(pow(longaverage, 0.500)) + 110.600;
+ fprintf(filehandle, "%.4f", featuretimesec);
+ fprintf(filehandle, "\t");
+ fprintf(filehandle, "%.4f\n", leqm10);
+
+}
projects / leqm-nrt.git / commitdiff