minimodem: use simpleaudio
This commit is contained in:
Kamal Mostafa
parent
fe4b47af20
commit
0a47a37a06
118
src/minimodem.c
118
src/minimodem.c
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@ -11,14 +11,20 @@
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#include <errno.h>
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#include <math.h>
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#include <pulse/simple.h>
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#include <pulse/error.h>
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#include <pulse/gccmacro.h>
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#include <fftw3.h>
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#include "simpleaudio.h"
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#include "tscope_print.h"
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#ifndef MAX
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#define MAX(a,b) ((a)>(b)?(a):(b))
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#endif
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#ifndef MIN
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#define MIN(a,b) ((a)<(b)?(a):(b))
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#endif
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static inline
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float
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band_mag( fftwf_complex * const cplx, unsigned int band, float scalar )
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@ -30,22 +36,11 @@ band_mag( fftwf_complex * const cplx, unsigned int band, float scalar )
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}
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int main(int argc, char*argv[]) {
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/* The sample type to use */
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static const pa_sample_spec ss = {
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.format = PA_SAMPLE_FLOAT32,
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.rate = 48000, // pulseaudio will resample its configured audio rate
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// .channels = 2 // 2 channel stereo
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.channels = 1 // pulseaudio will downmix (additively) to 1 channel
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// .channels = 3 // 2 channel stereo + 1 mixed channel
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};
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int ret = 1;
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int error;
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if ( argc < 2 ) {
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fprintf(stderr, "usage: minimodem baud_rate [ mark_hz space_hz ]\n");
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fprintf(stderr, "usage: minimodem [filename] baud_rate [ mark_hz space_hz ]\n");
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return 1;
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}
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@ -56,7 +51,27 @@ int main(int argc, char*argv[]) {
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argi++;
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}
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unsigned int decode_rate = atoi(argv[argi++]);
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simpleaudio *sa;
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char *p;
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for ( p=argv[argi]; *p; p++ )
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if ( !isdigit(*p) )
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break;
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if ( *p ) {
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sa = simpleaudio_open_source_sndfile(argv[argi]);
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argi++;
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} else {
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sa = simpleaudio_open_source_pulseaudio(argv[0], "bfsk demodulator");
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}
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if ( !sa )
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return 1;
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unsigned int sample_rate = simpleaudio_get_rate(sa);
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unsigned int nchannels = simpleaudio_get_channels(sa);
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unsigned int decode_rate;
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decode_rate = atoi(argv[argi++]);
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unsigned int band_width;
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band_width = decode_rate;
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@ -85,8 +100,6 @@ int main(int argc, char*argv[]) {
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bfsk_space_f = atoi(argv[argi++]);
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}
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unsigned int sample_rate = ss.rate;
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unsigned int bfsk_mark_band = (bfsk_mark_f +(float)band_width/2) / band_width;
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unsigned int bfsk_space_band = (bfsk_space_f +(float)band_width/2) / band_width;
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@ -101,20 +114,6 @@ int main(int argc, char*argv[]) {
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}
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/* Create the recording stream */
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pa_simple *s;
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s = pa_simple_new(NULL, argv[0], PA_STREAM_RECORD, NULL, "record", &ss, NULL, NULL, &error);
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if ( !s ) {
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fprintf(stderr, __FILE__": pa_simple_new() failed: %s\n", pa_strerror(error));
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return 1;
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}
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int pa_samplesize = pa_sample_size(&ss);
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int pa_framesize = pa_frame_size(&ss);
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int pa_nchannels = ss.channels;
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assert( pa_framesize == pa_samplesize * pa_nchannels );
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/* Create the FFT plan */
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fftwf_plan fftplan;
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@ -126,9 +125,9 @@ int main(int argc, char*argv[]) {
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unsigned int nbands = fftsize / 2 + 1;
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float *fftin = fftwf_malloc(fftsize * sizeof(float) * pa_nchannels);
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float *fftin = fftwf_malloc(fftsize * sizeof(float) * nchannels);
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fftwf_complex *fftout = fftwf_malloc(nbands * sizeof(fftwf_complex) * pa_nchannels);
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fftwf_complex *fftout = fftwf_malloc(nbands * sizeof(fftwf_complex) * nchannels);
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/*
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* works only for 1 channel:
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@ -138,8 +137,8 @@ int main(int argc, char*argv[]) {
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* works for N channels:
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*/
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fftplan = fftwf_plan_many_dft_r2c(
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/*rank*/1, &fftsize, /*howmany*/pa_nchannels,
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fftin, NULL, /*istride*/pa_nchannels, /*idist*/1,
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/*rank*/1, &fftsize, /*howmany*/nchannels,
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fftin, NULL, /*istride*/nchannels, /*idist*/1,
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fftout, NULL, /*ostride*/1, /*odist*/nbands,
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FFTW_ESTIMATE | FFTW_PRESERVE_INPUT );
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/* Nb. FFTW_PRESERVE_INPUT is needed for the "shift the input window" trick */
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@ -150,10 +149,6 @@ int main(int argc, char*argv[]) {
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}
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void *pa_samples_in = fftin; // read samples directly into fftin
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assert( pa_samplesize == sizeof(float) );
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/*
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* Prepare the input sample chunk rate
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*/
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@ -179,8 +174,9 @@ int main(int argc, char*argv[]) {
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/* pulseaudio *adds* when downmixing 2 channels to 1; if we're using
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* only one channel here, we blindly assume that pulseaudio downmixed
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* from 2, and rescale magnitudes accordingly. */
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if ( pa_nchannels == 1 )
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magscalar /= 2.0;
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// but sndfile does not do that.
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// if ( nchannels == 1 )
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// magscalar /= 2.0;
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float actual_decode_rate = (float)sample_rate / nsamples;
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fprintf(stderr, "### nsamples=%u ", nsamples);
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@ -196,7 +192,7 @@ int main(int argc, char*argv[]) {
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// sadly, COLUMNS is not exported by default (?)
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char *columns_env = getenv("COLUMNS");
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int columns = columns_env ? atoi(columns_env) : 80;
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int show_nbands = ( (columns - 2 - 10) / pa_nchannels ) - 1 - 10;
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int show_nbands = ( (columns - 2 - 10) / nchannels ) - 1 - 10;
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if ( show_nbands > nbands )
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show_nbands = nbands;
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@ -215,16 +211,12 @@ int main(int argc, char*argv[]) {
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while ( 1 ) {
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size_t nframes = nsamples;
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size_t nbytes = nframes * pa_framesize;
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bzero(fftin, (fftsize * sizeof(float) * nchannels));
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bzero(fftin, (fftsize * sizeof(float) * pa_nchannels));
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if (pa_simple_read(s, pa_samples_in, nbytes, &error) < 0) {
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fprintf(stderr, __FILE__": pa_simple_read() failed: %s\n",
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pa_strerror(error));
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ret = 1;
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size_t nframes = nsamples;
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/* read samples directly into fftin */
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if ((ret=simpleaudio_read(sa, fftin, nframes)) <= 0)
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break;
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}
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carrier_nsamples += nframes;
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#define TRICK
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@ -299,16 +291,15 @@ reprocess_audio:
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skipped_frames += nframes;
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if ( nframes ) {
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size_t nbytes = nframes * pa_framesize;
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size_t reuse_bytes = nsamples*pa_framesize - nbytes;
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memmove(pa_samples_in, pa_samples_in+nbytes, reuse_bytes);
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void *in = pa_samples_in + reuse_bytes;
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if (pa_simple_read(s, in, nbytes, &error) < 0) {
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fprintf(stderr, __FILE__": pa_simple_read() failed: %s\n",
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pa_strerror(error));
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ret = 1;
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size_t framesize = nchannels * sizeof(float);
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size_t nbytes = nframes * framesize;
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size_t reuse_bytes = nsamples*framesize - nbytes;
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memmove(fftin, fftin+nbytes, reuse_bytes);
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void *in = fftin + reuse_bytes;
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if ((ret=simpleaudio_read(sa, in, nframes)) <= 0)
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break;
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}
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carrier_nsamples += nframes;
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goto reprocess_audio;
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}
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@ -433,7 +424,10 @@ reprocess_audio:
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}
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pa_simple_free(s);
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if ( ret != 0 )
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fprintf(stderr, "simpleaudio_read: error\n");
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simpleaudio_close(sa);
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fftwf_free(fftin);
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fftwf_free(fftout);
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