269 lines
7.0 KiB
C
269 lines
7.0 KiB
C
/*
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PiFmAdv - Advanced FM transmitter for the Raspberry Pi
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Copyright (C) 2017 Miegl
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See https://github.com/Miegl/PiFmAdv
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*/
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#include <sndfile.h>
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#include <stdlib.h>
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#include <strings.h>
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#include <math.h>
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#include "rds.h"
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#define PI 3.14159265359
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#define FIR_HALF_SIZE 30
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#define FIR_SIZE (2*FIR_HALF_SIZE-1)
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size_t length;
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// coefficients of the low-pass FIR filter
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double low_pass_fir[FIR_HALF_SIZE];
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double carrier_38[] = {0.0, 0.8660254037844386, 0.8660254037844388, 1.2246467991473532e-16, -0.8660254037844384, -0.8660254037844386};
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double carrier_19[] = {0.0, 0.5, 0.8660254037844386, 1.0, 0.8660254037844388, 0.5, 1.2246467991473532e-16, -0.5, -0.8660254037844384, -1.0, -0.8660254037844386, -0.5};
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int phase_38 = 0;
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int phase_19 = 0;
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double downsample_factor;
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double *audio_buffer;
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int audio_index = 0;
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int audio_len = 0;
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double audio_pos;
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double fir_buffer_mono[FIR_SIZE] = {0};
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double fir_buffer_stereo[FIR_SIZE] = {0};
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int fir_index = 0;
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int channels;
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double *last_buffer_val;
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double preemphasis_prewarp;
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double preemphasis_coefficient;
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SNDFILE *inf;
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double *alloc_empty_buffer(size_t length) {
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double *p = malloc(length * sizeof(double));
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if(p == NULL) return NULL;
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bzero(p, length * sizeof(double));
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return p;
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}
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int fm_mpx_open(char *filename, size_t len, int cutoff_freq, int preemphasis_corner_freq, int srate, int nochan) {
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length = len;
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if(filename != NULL) {
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// Open the input file
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SF_INFO sfinfo;
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if(filename[0] == '-' && srate > 0 && nochan > 0) {
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printf("Using stdin for raw audio input at %d Hz.\n",srate);
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sfinfo.samplerate = srate;
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sfinfo.format = SF_FORMAT_RAW | SF_FORMAT_PCM_16 | SF_ENDIAN_LITTLE;
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sfinfo.channels = nochan;
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sfinfo.frames = 0;
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}
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// stdin or file on the filesystem?
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if(filename[0] == '-') {
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if(!(inf = sf_open_fd(fileno(stdin), SFM_READ, &sfinfo, 0))) {
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fprintf(stderr, "Error: could not open stdin for audio input.\n");
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return -1;
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} else {
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printf("Using stdin for audio input.\n");
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}
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} else {
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if(!(inf = sf_open(filename, SFM_READ, &sfinfo))) {
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fprintf(stderr, "Error: could not open input file %s.\n", filename);
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return -1;
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} else {
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printf("Using audio file: %s\n", filename);
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}
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}
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int in_samplerate = sfinfo.samplerate;
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downsample_factor = 228000. / in_samplerate;
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printf("Input: %d Hz, upsampling factor: %.2f\n", in_samplerate, downsample_factor);
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channels = sfinfo.channels;
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if(channels > 1) {
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printf("%d channels, generating stereo multiplex.\n", channels);
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} else {
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printf("1 channel, monophonic operation.\n");
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}
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// Create the preemphasis
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last_buffer_val = (double*) malloc(sizeof(double)*channels);
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for(int i=0; i<channels; i++) last_buffer_val[i] = 0;
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preemphasis_prewarp = tan(PI*preemphasis_corner_freq/in_samplerate);
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preemphasis_coefficient = (1.0 + (1.0 - preemphasis_prewarp)/(1.0 + preemphasis_prewarp))/2.0;
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printf("Created preemphasis with cutoff at %.1i Hz\n", preemphasis_corner_freq);
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// Create the low-pass FIR filter
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if(in_samplerate < cutoff_freq) cutoff_freq = in_samplerate;
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// Here we divide this coefficient by two because it will be counted twice
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// when applying the filter
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low_pass_fir[FIR_HALF_SIZE-1] = 2 * cutoff_freq / 228000 /2;
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// Only store half of the filter since it is symmetric
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for(int i=1; i<FIR_HALF_SIZE; i++) {
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low_pass_fir[FIR_HALF_SIZE-1-i] =
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sin(2 * PI * cutoff_freq * i / 228000) / (PI * i) // sinc
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* (.54 - .46 * cos(2*PI * (i+FIR_HALF_SIZE) / (2*FIR_HALF_SIZE))); // Hamming window
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}
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printf("Created low-pass FIR filter for audio channels, with cutoff at %.1i Hz\n", cutoff_freq);
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audio_pos = downsample_factor;
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audio_buffer = alloc_empty_buffer(length * channels);
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if(audio_buffer == NULL) return -1;
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}
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else {
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inf = NULL;
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}
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return 0;
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}
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// samples provided by this function are in 0..10: they need to be divided by
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// 10 after.
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int fm_mpx_get_samples(double *mpx_buffer, double *rds_buffer, float mpx, int rds, int wait) {
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if(inf == NULL) {
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if(rds) get_rds_samples(mpx_buffer, length);
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return 0;
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} else {
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if(rds) get_rds_samples(rds_buffer, length);
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}
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for(int i=0; i<length; i++) {
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if(audio_pos >= downsample_factor) {
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audio_pos -= downsample_factor;
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if(audio_len <= channels) {
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for(int j=0; j<2; j++) { // one retry
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audio_len = sf_read_double(inf, audio_buffer, length);
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if (audio_len < 0) {
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fprintf(stderr, "Error reading audio\n");
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return -1;
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}
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if(audio_len == 0) {
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if( sf_seek(inf, 0, SEEK_SET) < 0 ) {
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if(wait) {
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return 0;
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} else {
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fprintf(stderr, "Could not rewind in audio file, terminating\n");
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return -1;
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}
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}
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} else {
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//apply preemphasis
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int k;
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int l;
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double tmp;
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for(k=0; k<audio_len; k+=channels) {
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for(l=0; l<channels; l++) {
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tmp = audio_buffer[k+l];
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audio_buffer[k+l] = audio_buffer[k+l] - preemphasis_coefficient*last_buffer_val[l];
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last_buffer_val[l] = tmp;
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}
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}
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break;
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}
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}
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audio_index = 0;
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} else {
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audio_index += channels;
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audio_len -= channels;
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}
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}
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// First store the current sample(s) into the FIR filter's ring buffer
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if(channels == 0) {
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fir_buffer_mono[fir_index] = audio_buffer[audio_index];
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} else {
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// In stereo operation, generate sum and difference signals
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fir_buffer_mono[fir_index] =
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audio_buffer[audio_index] + audio_buffer[audio_index+1];
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fir_buffer_stereo[fir_index] =
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audio_buffer[audio_index] - audio_buffer[audio_index+1];
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}
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fir_index++;
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if(fir_index >= FIR_SIZE) fir_index = 0;
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// Now apply the FIR low-pass filter
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/* As the FIR filter is symmetric, we do not multiply all
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the coefficients independently, but two-by-two, thus reducing
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the total number of multiplications by a factor of two
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*/
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double out_mono = 0;
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double out_stereo = 0;
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int ifbi = fir_index; // ifbi = increasing FIR Buffer Index
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int dfbi = fir_index; // dfbi = decreasing FIR Buffer Index
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for(int fi=0; fi<FIR_HALF_SIZE; fi++) { // fi = Filter Index
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dfbi--;
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if(dfbi < 0) dfbi = FIR_SIZE-1;
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out_mono +=
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low_pass_fir[fi] *
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(fir_buffer_mono[ifbi] + fir_buffer_mono[dfbi]);
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if(channels > 1) {
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out_stereo +=
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low_pass_fir[fi] *
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(fir_buffer_stereo[ifbi] + fir_buffer_stereo[dfbi]);
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}
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ifbi++;
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if(ifbi >= FIR_SIZE) ifbi = 0;
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}
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// End of FIR filter
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if (channels>1) {
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mpx_buffer[i] =
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((mpx-2)/2) * out_mono +
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((mpx-2)/2) * carrier_38[phase_38] * out_stereo +
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1 * carrier_19[phase_19];
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if (rds) {
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mpx_buffer[i] += rds_buffer[i];
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}
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phase_19++;
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phase_38++;
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if(phase_19 >= 12) phase_19 = 0;
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if(phase_38 >= 6) phase_38 = 0;
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}
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else {
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mpx_buffer[i] =
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(mpx-1) * out_mono;
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if (rds) {
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mpx_buffer[i] += rds_buffer[i];
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}
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}
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audio_pos++;
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}
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return 0;
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}
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int fm_mpx_close() {
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if(sf_close(inf) ) {
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fprintf(stderr, "Error closing audio file");
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}
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if(audio_buffer != NULL) free(audio_buffer);
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return 0;
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}
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