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cava
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moved alsa and fifo part to separate files, it works!

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master
karlstav 11 years ago
parent 5ec1377036
commit 4c3f9ae0ee
1 changed files with 24 additions and 195 deletions
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  1. 219
      cava.c

219
cava.c
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@ -22,7 +22,10 @@
#include <pthread.h>
#include "output/terminal_ncurses.h"
#include "output/terminal_ncurses.c"
#include "input/alsa.h"
#include "input/alsa.c"
#include "input/fifo.h"
#include "input/fifo.c"
#ifdef __GNUC__
@ -33,10 +36,8 @@
#define GCC_UNUSED /* nothing */
#endif
int M = 2048;
int shared[2048];
int format = -1;
unsigned int rate = 0;
bool scientificMode = false;
@ -59,196 +60,19 @@ void sig_handler(int sig_no)
signal(sig_no, SIG_DFL);
raise(sig_no);
}
//input: FIFO
void* input_fifo(void* data)
{
int fd;
int n = 0;
signed char buf[1024];
int tempr, templ, lo;
int q, i;
int t = 0;
int size = 1024;
char *path = ((char*)data);
int bytes = 0;
int flags;
struct timespec req = { .tv_sec = 0, .tv_nsec = 10000000 };
fd = open(path, O_RDONLY);
flags = fcntl(fd, F_GETFL, 0);
fcntl(fd, F_SETFL, flags | O_NONBLOCK);
while (1) {
bytes = read(fd, buf, sizeof(buf));
if (bytes == -1) { //if no bytes read sleep 10ms and zero shared buffer
nanosleep (&req, NULL);
t++;
if (t > 10) {
for (i = 0; i < M; i++)shared[i] = 0;
t = 0;
}
} else { //if bytes read go ahead
t = 0;
for (q = 0; q < (size / 4); q++) {
tempr = ( buf[ 4 * q - 1] << 2);
lo = ( buf[4 * q ] >> 6);
if (lo < 0)lo = abs(lo) + 1;
if (tempr >= 0)tempr = tempr + lo;
else tempr = tempr - lo;
templ = ( buf[ 4 * q - 3] << 2);
lo = ( buf[ 4 * q - 2] >> 6);
if (lo < 0)lo = abs(lo) + 1;
if (templ >= 0)templ = templ + lo;
else templ = templ - lo;
shared[n] = (tempr + templ) / 2;
n++;
if (n == M - 1)n = 0;
}
}
}
close(fd);
}
// input: ALSA
void* input_alsa(void* data)
{
signed char *buffer;
snd_pcm_t *handle;
snd_pcm_hw_params_t *params;
unsigned int val;
snd_pcm_uframes_t frames;
char *device = ((char*)data);
val = 44100;
int i, n, o, size, dir, err, lo;
int tempr, templ;
int radj, ladj;
// alsa: open device to capture audio
if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_CAPTURE, 0) < 0)) {
cleanup();
fprintf(stderr,
"error opening stream: %s\n",
snd_strerror(err));
exit(EXIT_FAILURE);
}
#ifdef DEBUG
else printf("open stream successful\n");
#endif
snd_pcm_hw_params_alloca(&params); //assembling params
snd_pcm_hw_params_any (handle, params); //setting defaults or something
snd_pcm_hw_params_set_access(handle, params,
SND_PCM_ACCESS_RW_INTERLEAVED); //interleaved mode right left right left
snd_pcm_hw_params_set_format(handle, params,
SND_PCM_FORMAT_S16_LE); //trying to set 16bit
snd_pcm_hw_params_set_channels(handle, params, 2); //assuming stereo
val = 44100;
snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); //trying 44100 rate
frames = 256;
snd_pcm_hw_params_set_period_size_near(handle, params, &frames,
&dir); //number of frames pr read
err = snd_pcm_hw_params(handle, params); //attempting to set params
if (err < 0) {
cleanup();
fprintf(stderr,
"unable to set hw parameters: %s\n",
snd_strerror(err));
exit(EXIT_FAILURE);
}
snd_pcm_hw_params_get_format(params,
(snd_pcm_format_t * )&val); //getting actual format
//converting result to number of bits
if (val < 6)format = 16;
else if (val > 5 && val < 10)format = 24;
else if (val > 9)format = 32;
snd_pcm_hw_params_get_rate( params, &rate, &dir); //getting rate
snd_pcm_hw_params_get_period_size(params, &frames, &dir);
snd_pcm_hw_params_get_period_time(params, &val, &dir);
size = frames * (format / 8) * 2; // frames * bits/8 * 2 channels
buffer = malloc(size);
radj = format / 4; //adjustments for interleaved
ladj = format / 8;
o = 0;
while (1) {
err = snd_pcm_readi(handle, buffer, frames);
if (err == -EPIPE) {
/* EPIPE means overrun */
#ifdef DEBUG
fprintf(stderr, "overrun occurred\n");
#endif
snd_pcm_prepare(handle);
} else if (err < 0) {
#ifdef DEBUG
fprintf(stderr, "error from read: %s\n", snd_strerror(err));
#endif
} else if (err != (int)frames) {
#ifdef DEBUG
fprintf(stderr, "short read, read %d %d frames\n", err, (int)frames);
#endif
}
//sorting out one channel and only biggest octet
n = 0; //frame counter
for (i = 0; i < size ; i = i + (ladj) * 2) {
//first channel
tempr = ((buffer[i + (radj) - 1 ] <<
2)); //using the 10 upper bits this would give me a vert res of 1024, enough...
lo = ((buffer[i + (radj) - 2] >> 6));
if (lo < 0)lo = abs(lo) + 1;
if (tempr >= 0)tempr = tempr + lo;
if (tempr < 0)tempr = tempr - lo;
//other channel
templ = (buffer[i + (ladj) - 1] << 2);
lo = (buffer[i + (ladj) - 2] >> 6);
if (lo < 0)lo = abs(lo) + 1;
if (templ >= 0)templ = templ + lo;
else templ = templ - lo;
//adding channels and storing it in the buffer
shared[o] = (tempr + templ) / 2;
o++;
if (o == M - 1)o = 0;
n++;
}
}
}
// general: entry point
int main(int argc, char **argv)
{
int M = 2048;
pthread_t p_thread;
int thr_id GCC_UNUSED;
char *inputMethod = "alsa";
char *outputMethod = "terminal";
int im = 1;
int om = 1;
char *device = "hw:1,1";
char *path = "/tmp/mpd.fifo";
float fc[200];
float fr[200];
int lcf[200], hcf[200];
@ -302,11 +126,15 @@ Options:\n\
-v print version\n\
\n";
char ch;
struct audio_data audio;
audio.format = -1;
audio.rate = 0;
setlocale(LC_ALL, "");
for (i = 0; i < M; i++)shared[i] = 0;
for (i = 0; i < M; i++)audio.audio_out[i] = 0;
// general: handle Ctrl+C
struct sigaction action;
@ -320,7 +148,7 @@ Options:\n\
while ((c = getopt (argc, argv, "p:i:b:d:s:f:c:C:hSv")) != -1)
switch (c) {
case 'p': // argument: fifo path
path = optarg;
audio.source = optarg;
break;
case 'i': // argument: input method
im = 0;
@ -351,7 +179,7 @@ Options:\n\
if (fixedbands > 200)fixedbands = 200;
break;
case 'd': // argument: alsa device
device = optarg;
audio.source = optarg;
break;
case 's': // argument: sensitivity
sens = atoi(optarg);
@ -418,9 +246,10 @@ Options:\n\
// input: wait for the input to be ready
if (im == 1) {
audio.source = "hw:1,1";
thr_id = pthread_create(&p_thread, NULL, input_alsa,
(void*)device); //starting alsamusic listener
while (format == -1 || rate == 0) {
(void *)&audio); //starting alsamusic listener
while (audio.format == -1 || audio.rate == 0) {
req.tv_sec = 0;
req.tv_nsec = 1000000;
nanosleep (&req, NULL);
@ -441,10 +270,10 @@ Options:\n\
}
if (im == 2) {
audio.source = "/tmp/mpd.fifo";
thr_id = pthread_create(&p_thread, NULL, input_fifo,
(void*)path); //starting fifomusic listener
rate = 44100;
format = 16;
(void*)&audio); //starting fifomusic listener
audio.rate = 44100;
}
p = fftw_plan_dft_r2c_1d(M, in, *out, FFTW_MEASURE); //planning to rock
@ -521,7 +350,7 @@ Options:\n\
for (n = 0; n < bands + 1; n++) {
fc[n] = 10000 * pow(10, -2.37 + ((((float)n + 1) / ((float)bands + 1)) *
2.37)); //decided to cut it at 10k, little interesting to hear above
fr[n] = fc[n] / (rate /
fr[n] = fc[n] / (audio.rate /
2); //remember nyquist!, pr my calculations this should be rate/2 and nyquist freq in M/2 but testing shows it is not... or maybe the nq freq is in M/4
lcf[n] = fr[n] * (M /
4); //lfc stores the lower cut frequency foo each band in the fft out buffer
@ -583,9 +412,9 @@ Options:\n\
hpeak = 0;
for (i = 0; i < (2 * (M / 2 + 1)); i++) {
if (i < M) {
in[i] = shared[i];
if (shared[i] > hpeak) hpeak = shared[i];
if (shared[i] < lpeak) lpeak = shared[i];
in[i] = audio.audio_out[i];
if (audio.audio_out[i] > hpeak) hpeak = audio.audio_out[i];
if (audio.audio_out[i] < lpeak) lpeak = audio.audio_out[i];
} else in[i] = 0;
}
peak[bands] = (hpeak + abs(lpeak));

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