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Devided bass, midtones and treble in to separate fftw operations

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this commit involves some major and perhaps braking changes
audio sorting is now done by a common function in fifo.c
all other input plugins send a s16le interleaved audio buffer here,
reformating might be done here in the future.
Audio is put in to three buffers of different lenght, shorter
buffers for higher frequencies. FFT is aplied separetly and
the ouput is combined toghether in the separate_freq_bans function.
This creates better responsivnes for higher frequencies while still
 beeing able to pick up the lower ones.
master
karlstav 6 years ago
parent 8143270d74
commit 7a224dfa15
8 changed files with 367 additions and 224 deletions
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  1. 295
      cava.c
  2. 112
      input/alsa.c
  3. 133
      input/fifo.c
  4. 1
      input/fifo.h
  5. 14
      input/portaudio.c
  6. 24
      input/pulse.c
  7. 3
      input/shmem.c
  8. 9
      input/sndio.c

295
cava.c
Unescape View File

@ -153,24 +153,35 @@ static bool directory_exists(const char * path) {
#endif
int * separate_freq_bands(int FFTbufferSize, fftw_complex out[FFTbufferSize / 2 + 1],
int * separate_freq_bands(int FFTbassbufferSize, fftw_complex out_bass[FFTbassbufferSize / 2 + 1],
int FFTmidbufferSize, fftw_complex out_mid[FFTmidbufferSize / 2 + 1],
int FFTtreblebufferSize, fftw_complex out_treble[FFTtreblebufferSize / 2 + 1],
int bass_cut_off_bar, int treble_cut_off_bar,
int bars, int lcf[200], int hcf[200], double k[200], int channel,
double sens, double ignore) {
int o,i;
int o, i;
double peak[201];
static int fl[200];
static int fr[200];
double y[FFTbufferSize / 2 + 1];
double y[FFTbassbufferSize / 2 + 1];
double temp;
// process: separate frequency bands
for (o = 0; o < bars; o++) {
peak[o] = 0;
i = 0;
// process: get peaks
for (i = lcf[o]; i <= hcf[o]; i++) {
y[i] = hypot(out[i][0], out[i][1]);
if (o <= bass_cut_off_bar) {
y[i] = hypot(out_bass[i][0], out_bass[i][1]);
} else if (o > bass_cut_off_bar && o <= treble_cut_off_bar) {
y[i] = hypot(out_mid[i][0], out_mid[i][1]);
} else if (o > treble_cut_off_bar) {
y[i] = hypot(out_treble[i][0], out_treble[i][1]);
}
peak[o] += y[i]; //adding upp band
}
@ -281,8 +292,12 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
int sourceIsAuto = 1;
double smh;
double *inr, *inl;
fftw_complex *outl, *outr;
double *in_bass_r, *in_bass_l;
fftw_complex *out_bass_l, *out_bass_r;
double *in_mid_r, *in_mid_l;
fftw_complex *out_mid_l, *out_mid_r;
double *in_treble_r, *in_treble_l;
fftw_complex *out_treble_l, *out_treble_r;
struct audio_data audio;
struct config_params p;
@ -346,6 +361,9 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
// general: main loop
while (1) {
#ifdef DEBUG
printf("loading config\n");
#endif
//config: load
struct error_s error;
error.length = 0;
@ -354,17 +372,6 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
exit(EXIT_FAILURE);
}
audio.FFTbufferSize = p.FFTbufferSize;
inr = malloc(2 * (p.FFTbufferSize / 2 + 1) * sizeof(double));
inl = malloc(2 * (p.FFTbufferSize / 2 + 1) * sizeof(double));
outl = malloc(2 * (p.FFTbufferSize / 2 + 1) * sizeof(fftw_complex));
outr = malloc(2 * (p.FFTbufferSize / 2 + 1) * sizeof(fftw_complex));
fftw_plan pl = fftw_plan_dft_r2c_1d(p.FFTbufferSize, inl, outl, FFTW_MEASURE);
fftw_plan pr = fftw_plan_dft_r2c_1d(p.FFTbufferSize, inr, outr, FFTW_MEASURE);
output_mode = p.om;
@ -382,11 +389,17 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
//input: init
int bass_cut_off = 150;
int treble_cut_off = 1500;
audio.source = malloc(1 + strlen(p.audio_source));
strcpy(audio.source, p.audio_source);
audio.format = -1;
audio.rate = 0;
audio.FFTbassbufferSize = 4096;
audio.FFTmidbufferSize = 1024;
audio.FFTtreblebufferSize = 512;
audio.terminate = 0;
if (p.stereo) audio.channels = 2;
if (!p.stereo) audio.channels = 1;
@ -396,20 +409,14 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
if (strcmp(p.mono_option, "average") == 0) audio.average = true;
if (strcmp(p.mono_option, "left") == 0) audio.left = true;
if (strcmp(p.mono_option, "right") == 0) audio.right = true;
audio.bass_index = 0;
audio.mid_index = 0;
audio.treble_index = 0;
for (i = 0; i < (p.FFTbufferSize / 2 + 1); i++) {
if (i < p.FFTbufferSize) {
audio.audio_out_l[i] = 0;
audio.audio_out_r[i] = 0;
}
inl[i] = 0;
inr[i] = 0;
for (n = 0; n < 2; n++) {
outl[i][n] = 0;
outr[i][n] = 0;
}
}
#ifdef DEBUG
printf("starting audio thread\n");
#endif
#ifdef ALSA
// input_alsa: wait for the input to be ready
if (p.im == 1) {
@ -497,6 +504,83 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
exit(EXIT_FAILURE);
}
//BASS
//audio.FFTbassbufferSize = audio.rate / 20; // audio.FFTbassbufferSize;
in_bass_r = malloc(2 * (audio.FFTbassbufferSize / 2 + 1) * sizeof(double));
in_bass_l = malloc(2 * (audio.FFTbassbufferSize / 2 + 1) * sizeof(double));
out_bass_l = malloc(2 * (audio.FFTbassbufferSize / 2 + 1) * sizeof(fftw_complex));
out_bass_r = malloc(2 * (audio.FFTbassbufferSize / 2 + 1) * sizeof(fftw_complex));
fftw_plan p_bass_l = fftw_plan_dft_r2c_1d(audio.FFTbassbufferSize, in_bass_l, out_bass_l, FFTW_MEASURE);
fftw_plan p_bass_r = fftw_plan_dft_r2c_1d(audio.FFTbassbufferSize, in_bass_r, out_bass_r, FFTW_MEASURE);
//MID
//audio.FFTmidbufferSize = audio.rate / bass_cut_off; // audio.FFTbassbufferSize;
in_mid_r = malloc(2 * (audio.FFTmidbufferSize / 2 + 1) * sizeof(double));
in_mid_l = malloc(2 * (audio.FFTmidbufferSize / 2 + 1) * sizeof(double));
out_mid_l = malloc(2 * (audio.FFTmidbufferSize / 2 + 1) * sizeof(fftw_complex));
out_mid_r = malloc(2 * (audio.FFTmidbufferSize / 2 + 1) * sizeof(fftw_complex));
fftw_plan p_mid_l = fftw_plan_dft_r2c_1d(audio.FFTmidbufferSize, in_mid_l, out_mid_l, FFTW_MEASURE);
fftw_plan p_mid_r = fftw_plan_dft_r2c_1d(audio.FFTmidbufferSize, in_mid_r, out_mid_r, FFTW_MEASURE);
//TRIEBLE
//audio.FFTtreblebufferSize = audio.rate / treble_cut_off; // audio.FFTbassbufferSize;
in_treble_r = malloc(2 * (audio.FFTtreblebufferSize / 2 + 1) * sizeof(double));
in_treble_l = malloc(2 * (audio.FFTtreblebufferSize / 2 + 1) * sizeof(double));
out_treble_l = malloc(2 * (audio.FFTtreblebufferSize / 2 + 1) * sizeof(fftw_complex));
out_treble_r = malloc(2 * (audio.FFTtreblebufferSize / 2 + 1) * sizeof(fftw_complex));
fftw_plan p_treble_l = fftw_plan_dft_r2c_1d(audio.FFTtreblebufferSize, in_treble_l, out_treble_l, FFTW_MEASURE);
fftw_plan p_treble_r = fftw_plan_dft_r2c_1d(audio.FFTtreblebufferSize, in_treble_r, out_treble_r, FFTW_MEASURE);
#ifdef DEBUG
printf("got buffer size: %d, %d, %d", audio.FFTbassbufferSize, audio.FFTmidbufferSize, audio.FFTtreblebufferSize);
printf("zeroing buffers\n");
#endif
for (i = 0; i < (audio.FFTbassbufferSize / 2 + 1); i++) {
if (i < audio.FFTbassbufferSize) {
audio.audio_out_bass_l[i] = 0;
audio.audio_out_bass_r[i] = 0;
}
in_bass_l[i] = 0;
in_bass_r[i] = 0;
for (n = 0; n < 2; n++) {
out_bass_l[i][n] = 0;
out_bass_r[i][n] = 0;
}
}
for (i = 0; i < (audio.FFTmidbufferSize / 2 + 1); i++) {
if (i < audio.FFTmidbufferSize) {
audio.audio_out_mid_l[i] = 0;
audio.audio_out_mid_r[i] = 0;
}
in_mid_l[i] = 0;
in_mid_r[i] = 0;
for (n = 0; n < 2; n++) {
out_mid_l[i][n] = 0;
out_mid_r[i][n] = 0;
}
}
for (i = 0; i < (audio.FFTtreblebufferSize / 2 + 1); i++) {
if (i < audio.FFTtreblebufferSize) {
audio.audio_out_treble_l[i] = 0;
audio.audio_out_treble_r[i] = 0;
}
in_treble_l[i] = 0;
in_treble_r[i] = 0;
for (n = 0; n < 2; n++) {
out_treble_l[i][n] = 0;
out_treble_r[i][n] = 0;
}
}
bool reloadConf = false;
@ -625,6 +709,10 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
//freqconst = -2;
// process: calculate cutoff frequencies
int bass_cut_off_bar = -1;
int treble_cut_off_bar = -1;
bool first_bar = false;
int first_treble_bar = 0;
for (n = 0; n < bars + 1; n++) {
double pot = freqconst * (-1);
pot += ((float)n + 1) / ((float)bars + 1) * freqconst;
@ -634,30 +722,61 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
//and nyquist freq in M/2 but testing shows it is not...
//or maybe the nq freq is in M/4
k[n] = pow(fc[n],1);
k[n] *= (float)height / pow(2,28);
k[n] *= p.smooth[(int)floor(((double)n) * smh)];
k[n] /= log2(audio.FFTbassbufferSize);
//lfc stores the lower cut frequency foo each bar in the fft out buffer
lcf[n] = fre[n] * (p.FFTbufferSize /2) + 1;
if (n != 0) {
if (fc[n] < bass_cut_off ) {
lcf[n] = fre[n] * (audio.FFTbassbufferSize /2) + 1;
bass_cut_off_bar++;
treble_cut_off_bar++;
k[n] *= log2(audio.FFTbassbufferSize);
} else if (fc[n] > bass_cut_off && fc[n] < treble_cut_off) {
lcf[n] = fre[n] * (audio.FFTmidbufferSize /2) + 1;
treble_cut_off_bar++;
if ((treble_cut_off_bar - bass_cut_off_bar) == 1) {
first_bar = true;
hcf[n - 1] = fre[n] * (audio.FFTbassbufferSize /2);
if (hcf[n - 1] < lcf[n - 1]) hcf[n - 1] = lcf[n - 1];
} else {
first_bar = false;
}
k[n] *= log2(audio.FFTmidbufferSize);
} else {
lcf[n] = fre[n] * (audio.FFTtreblebufferSize /2) + 1;
first_treble_bar++;
if (first_treble_bar == 1) {
first_bar = true;
hcf[n - 1] = fre[n] * (audio.FFTmidbufferSize /2);
if (hcf[n - 1] < lcf[n - 1]) hcf[n - 1] = lcf[n - 1];
} else {
first_bar = false;
}
k[n] *= log2(audio.FFTtreblebufferSize);
}
if (n != 0 && !first_bar) {
hcf[n - 1] = lcf[n] - 1;
//pushing the spectrum up if the expe function gets "clumped"
if (lcf[n] <= lcf[n - 1])lcf[n] = lcf[n - 1] + 1;
hcf[n - 1] = lcf[n] - 1;
}
#ifdef DEBUG
if (n != 0) {
mvprintw(n,0,"%d: %f -> %f (%d -> %d) \n", n,
#ifdef DEBUG
if (n != 0) {
mvprintw(n,0,"%d: %f -> %f (%d -> %d) bass: %d, treble:%d \n", n,
fc[n - 1], fc[n], lcf[n - 1],
hcf[n - 1]);
}
#endif
hcf[n - 1], bass_cut_off_bar, treble_cut_off_bar);
}
#endif
}
// process: weigh signal to frequencies height and EQ
for (n = 0; n < bars; n++) {
k[n] = pow(fc[n],0.85);
k[n] *= (float)height / pow(2,28);
k[n] *= p.smooth[(int)floor(((double)n) * smh)];
}
if (p.stereo) bars = bars * 2;
@ -737,17 +856,36 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
// process: populate input buffer and check if input is present
silence = true;
for (i = 0; i < (2 * (p.FFTbufferSize / 2 + 1)); i++) {
if (i < p.FFTbufferSize) {
inl[i] = audio.audio_out_l[i];
if (p.stereo) inr[i] = audio.audio_out_r[i];
if (inl[i] || inr[i]) silence = false;
for (i = 0; i < (2 * (audio.FFTbassbufferSize / 2 + 1)); i++) {
if (i < audio.FFTbassbufferSize) {
in_bass_l[i] = audio.audio_out_bass_l[i];
if (p.stereo) in_bass_r[i] = audio.audio_out_bass_r[i];
if (in_bass_l[i] || in_bass_r[i]) silence = false;
} else {
in_bass_l[i] = 0;
if (p.stereo) in_bass_r[i] = 0;
}
}
for (i = 0; i < (2 * (audio.FFTmidbufferSize / 2 + 1)); i++) {
if (i < audio.FFTmidbufferSize) {
in_mid_l[i] = audio.audio_out_mid_l[i];
if (p.stereo) in_mid_r[i] = audio.audio_out_mid_r[i];
} else {
inl[i] = 0;
if (p.stereo) inr[i] = 0;
in_mid_l[i] = 0;
if (p.stereo) in_mid_r[i] = 0;
}
}
for (i = 0; i < (2 * (audio.FFTtreblebufferSize / 2 + 1)); i++) {
if (i < audio.FFTtreblebufferSize) {
in_treble_l[i] = audio.audio_out_treble_l[i];
if (p.stereo) in_treble_r[i] = audio.audio_out_treble_r[i];
} else {
in_treble_l[i] = 0;
if (p.stereo) in_treble_r[i] = 0;
}
}
if (silence) sleep++;
else sleep = 0;
@ -756,17 +894,34 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
// process: execute FFT and sort frequency bands
if (p.stereo) {
fftw_execute(pl);
fftw_execute(pr);
fftw_execute(p_bass_l);
fftw_execute(p_bass_r);
fftw_execute(p_mid_l);
fftw_execute(p_mid_r);
fftw_execute(p_treble_l);
fftw_execute(p_treble_r);
fl = separate_freq_bands(audio.FFTbassbufferSize, out_bass_l,
audio.FFTmidbufferSize, out_mid_l,
audio.FFTtreblebufferSize, out_treble_l,
bass_cut_off_bar, treble_cut_off_bar,
bars/2, lcf, hcf, k, 1, p.sens, p.ignore);
fr = separate_freq_bands(audio.FFTbassbufferSize, out_bass_r,
audio.FFTmidbufferSize, out_mid_r,
audio.FFTtreblebufferSize, out_treble_r,
bass_cut_off_bar, treble_cut_off_bar,
bars/2, lcf, hcf, k, 1, p.sens, p.ignore);
fl = separate_freq_bands(p.FFTbufferSize, outl,bars/2,
lcf,hcf, k, 1, p.sens, p.ignore);
fr = separate_freq_bands(p.FFTbufferSize, outr,bars/2,
lcf,hcf, k, 2, p.sens, p.ignore);
} else {
fftw_execute(pl);
fl = separate_freq_bands(p.FFTbufferSize, outl,bars,
lcf,hcf, k, 1, p.sens, p.ignore);
fftw_execute(p_bass_l);
fftw_execute(p_mid_l);
fftw_execute(p_treble_l);
fl = separate_freq_bands(audio.FFTbassbufferSize, out_bass_l,
audio.FFTmidbufferSize, out_mid_l,
audio.FFTtreblebufferSize, out_treble_l,
bass_cut_off_bar, treble_cut_off_bar,
bars, lcf, hcf, k, 1, p.sens, p.ignore);
}
@ -932,12 +1087,26 @@ as of 0.4.0 all options are specified in config file, see in '/home/username/.co
if (p.customEQ) free(p.smooth);
if (sourceIsAuto) free(audio.source);
free(inr);
free(inl);
fftw_free(outr);
fftw_free(outl);
fftw_destroy_plan(pl);
fftw_destroy_plan(pr);
free(in_bass_r);
free(in_bass_l);
fftw_free(out_bass_r);
fftw_free(out_bass_l);
fftw_destroy_plan(p_bass_l);
fftw_destroy_plan(p_bass_r);
free(in_mid_r);
free(in_mid_l);
fftw_free(out_mid_r);
fftw_free(out_mid_l);
fftw_destroy_plan(p_mid_l);
fftw_destroy_plan(p_mid_r);
free(in_treble_r);
free(in_treble_l);
fftw_free(out_treble_r);
fftw_free(out_treble_l);
fftw_destroy_plan(p_treble_l);
fftw_destroy_plan(p_treble_r);
cleanup();

112
input/alsa.c
Unescape View File

@ -67,7 +67,7 @@ static int get_certain_frame(signed char* buffer, int buffer_index, int adjustme
temp -= lo;
return temp;
}
/*
static void fill_audio_outs(struct audio_data* audio, signed char* buffer,
const int size) {
int radj = audio->format / 4; // adjustments for interleaved
@ -92,7 +92,7 @@ const int size) {
audio_out_buffer_index %= audio->FFTbufferSize;
}
}
*/
#define FRAMES_NUMBER 256
void* input_alsa(void* data) {
@ -105,7 +105,9 @@ void* input_alsa(void* data) {
initialize_audio_parameters(&handle, audio, &frames);
snd_pcm_get_params(handle, &buffer_size, &period_size);
int radj = audio->format / 4; // adjustments for interleaved
int ladj = audio->format / 8;
int16_t buf[period_size];
int32_t buffer32[period_size];
frames = period_size / ((audio->format / 8) * CHANNELS_COUNT);
@ -115,80 +117,50 @@ void* input_alsa(void* data) {
// frames * bits/8 * channels
//const int size = frames * (audio->format / 8) * CHANNELS_COUNT;
signed char* buffer = malloc(period_size);
int n = 0;
while (1) {
switch (audio->format) {
case 16:
err = snd_pcm_readi(handle, buf, frames);
for (uint16_t i = 0; i < frames * 2; i += 2) {
if (audio->channels == 1){
if (audio->average) {
audio->audio_out_l[n] = (buf[i] + buf[i + 1]) / 2;
}
if (audio->left) {
audio->audio_out_l[n] = buf[i];
}
if (audio->right) {
audio->audio_out_l[n] = buf[i + 1];
}
}
//stereo storing channels in buffer
if (audio->channels == 2) {
audio->audio_out_l[n] = buf[i];
audio->audio_out_r[n] = buf[i + 1];
}
n++;
if (n == audio->FFTbufferSize - 1) n = 0;
}
break;
case 32:
err = snd_pcm_readi(handle, buffer32, frames);
for (uint16_t i = 0; i < frames * 2; i += 2) {
if (audio->channels == 1) {
if (audio->average) {
audio->audio_out_l[n] = (buffer32[i] / pow(2,16)) / 2;
audio->audio_out_l[n] += (buffer32[i + 1] / pow(2,16)) / 2;
switch (audio->format) {
case 16:
err = snd_pcm_readi(handle, buf, frames);
break;
case 32:
err = snd_pcm_readi(handle, buffer32, frames);
for (uint16_t i = 0; i < frames * 2; i++) {
buf[i] = buffer32[i] / pow(2,16);
}
if (audio->left) {
audio->audio_out_l[n] = (buffer32[i] / pow(2,16));
}
if (audio->right) {
audio->audio_out_l[n] = (buffer32[i + 1] / pow(2,16));
break;
default:
err = snd_pcm_readi(handle, buffer, frames);
// sorting out one channel and only biggest octet
for (uint16_t i = 0; i < period_size * 2; i += ladj * 2) {
// first channel
buf[i] = get_certain_frame(buffer, i, ladj);
// second channel
buf[i + 1] = get_certain_frame(buffer, i, radj);
}
}
//stereo storing channels in buffer
if (audio->channels == 2) {
audio->audio_out_l[n] = buffer32[i] / pow(2,16);
audio->audio_out_r[n] = buffer32[i + 1] / pow(2,16);
}
n++;
if (n == audio->FFTbufferSize - 1) n = 0;
//fill_audio_outs(audio, buffer, period_size);
break;
}
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
}
break;
default:
err = snd_pcm_readi(handle, buffer, frames);
fill_audio_outs(audio, buffer, period_size);
break;
}
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
}
write_to_fftw_input_buffers(buf, frames, data);
if (audio->terminate == 1) {

133
input/fifo.c
Unescape View File

@ -5,19 +5,69 @@ int rc;
struct audio_data {
int FFTbufferSize;
int16_t audio_out_r[65536];
int16_t audio_out_l[65536];
int format;
unsigned int rate ;
char *source; //alsa device, fifo path or pulse source
int im; //input mode alsa, fifo or pulse
int channels;
int FFTbassbufferSize;
int FFTmidbufferSize;
int FFTtreblebufferSize;
int bass_index;
int mid_index;
int treble_index;
int16_t audio_out_bass_r[65536];
int16_t audio_out_bass_l[65536];
int16_t audio_out_mid_r[65536];
int16_t audio_out_mid_l[65536];
int16_t audio_out_treble_r[65536];
int16_t audio_out_treble_l[65536];
int format;
unsigned int rate ;
char *source; //alsa device, fifo path or pulse source
int im; //input mode alsa, fifo or pulse
int channels;
bool left, right, average;
int terminate; // shared variable used to terminate audio thread
char error_message[1024];
char error_message[1024];
};
int write_to_fftw_input_buffers(int16_t buf[], int16_t frames, void* data) {
struct audio_data* audio = (struct audio_data*)data;
for (uint16_t i = 0; i < frames * 2; i += 2) {
if (audio->channels == 1){
if (audio->average) {
audio->audio_out_bass_l[audio->bass_index] = (buf[i] + buf[i + 1]) / 2;
}
if (audio->left) {
audio->audio_out_bass_l[audio->bass_index] = buf[i];
}
if (audio->right) {
audio->audio_out_bass_l[audio->bass_index] = buf[i + 1];
}
}
//stereo storing channels in buffer
if (audio->channels == 2) {
audio->audio_out_bass_l[audio->bass_index] = buf[i];
audio->audio_out_bass_r[audio->bass_index] = buf[i + 1];
audio->audio_out_mid_r[audio->mid_index] = audio->audio_out_bass_r[audio->bass_index];
audio->audio_out_treble_r[audio->treble_index] = audio->audio_out_bass_r[audio->bass_index];
}
audio->audio_out_mid_l[audio->mid_index] = audio->audio_out_bass_l[audio->bass_index];
audio->audio_out_treble_l[audio->treble_index] = audio->audio_out_bass_l[audio->bass_index];
audio->bass_index++;
audio->mid_index++;
audio->treble_index++;
if (audio->bass_index == audio->FFTbassbufferSize - 1) audio->bass_index = 0;
if (audio->mid_index == audio->FFTmidbufferSize - 1) audio->mid_index = 0;
if (audio->treble_index == audio->FFTtreblebufferSize - 1) audio->treble_index = 0;
}
return 0;
}
int open_fifo(const char *path)
{
int fd = open(path, O_RDONLY);
@ -32,18 +82,12 @@ void* input_fifo(void* data)
{
struct audio_data *audio = (struct audio_data *)data;
int fd;
int n = 0;
//signed char buf[1024];
//int tempr, templ, lo, q;
int i;
int t = 0;
//int size = 1024;
int bytes = 0;
int16_t buf[BUFSIZE / 2];
struct timespec req = { .tv_sec = 0, .tv_nsec = 10000000 };
uint16_t frames = BUFSIZE / 4;
fd = open_fifo(audio->source);
@ -55,8 +99,12 @@ void* input_fifo(void* data)
nanosleep (&req, NULL);
t++;
if (t > 10) {
for (i = 0; i < audio->FFTbufferSize; i++)audio->audio_out_l[i] = 0;
for (i = 0; i < audio->FFTbufferSize; i++)audio->audio_out_r[i] = 0;
for (i = 0; i < audio->FFTbassbufferSize; i++)audio->audio_out_bass_l[i] = 0;
for (i = 0; i < audio->FFTbassbufferSize; i++)audio->audio_out_bass_r[i] = 0;
for (i = 0; i < audio->FFTmidbufferSize; i++)audio->audio_out_mid_l[i] = 0;
for (i = 0; i < audio->FFTmidbufferSize; i++)audio->audio_out_mid_l[i] = 0;
for (i = 0; i < audio->FFTtreblebufferSize; i++)audio->audio_out_treble_r[i] = 0;
for (i = 0; i < audio->FFTtreblebufferSize; i++)audio->audio_out_treble_r[i] = 0;
close(fd);
fd = open_fifo(audio->source);
t = 0;
@ -64,54 +112,7 @@ void* input_fifo(void* data)
} else { //if bytes read go ahead
t = 0;
for (i = 0; i < BUFSIZE / 2; i += 2) {
if (audio->channels == 1) audio->audio_out_l[n] = (buf[i] + buf[i + 1]) / 2;
//stereo storing channels in buffer
if (audio->channels == 2) {
audio->audio_out_l[n] = buf[i];
audio->audio_out_r[n] = buf[i + 1];
}
n++;
if (n == audio->FFTbufferSize - 1) n = 0;
}
/*
for (q = 0; q < (size / 4); q++) {
tempr = ( buf[ 4 * q + 3] << 2);
lo = ( buf[4 * q + 2] >> 6);
if (lo < 0)lo = abs(lo) + 1;
if (tempr >= 0)tempr = tempr + lo;
else tempr = tempr - lo;
templ = ( buf[ 4 * q + 1] << 2);
lo = ( buf[ 4 * q] >> 6);
if (lo < 0)lo = abs(lo) + 1;
if (templ >= 0)templ = templ + lo;
else templ = templ - lo;
if (audio->channels == 1) audio->audio_out_l[n] = (tempr +
templ) /
2;
//stereo storing channels in buffer
if (audio->channels == 2) {
audio->audio_out_l[n] = templ;
audio->audio_out_r[n] = tempr;
}
n++;
if (n == 2048 - 1)n = 0;
}
*/
write_to_fftw_input_buffers(buf, frames, audio);
}
if (audio->terminate == 1) {

1
input/fifo.h
Unescape View File

@ -1,2 +1,3 @@
//header files for fifo, part of cava
void* input_fifo(void* data);
int write_to_fftw_input_buffers(int16_t buf[], int16_t frames, void* data);

14
input/portaudio.c
Unescape View File

@ -17,18 +17,18 @@ typedef struct {
} paTestData;
static struct audio_data *audio;
static int n = 0;
//static int n = 0;
static int recordCallback(const void *inputBuffer, void *outputBuffer,
unsigned long framesPerBuffer, const PaStreamCallbackTimeInfo* timeInfo,
PaStreamCallbackFlags statusFlags, void *userData) {
paTestData *data = (paTestData*)userData;
const SAMPLE *rptr = (const SAMPLE*)inputBuffer;
SAMPLE *rptr = (SAMPLE*)inputBuffer;
long framesToCalc;
long i;
//long i;
int finished;
unsigned long framesLeft = data->maxFrameIndex - data->frameIndex;
int16_t silence_buffer[BUFSIZE] = { SAMPLE_SILENCE };
(void) outputBuffer; // Prevent unused variable warnings.
(void) timeInfo;
(void) statusFlags;
@ -43,6 +43,8 @@ static int recordCallback(const void *inputBuffer, void *outputBuffer,
}
if(inputBuffer == NULL) {
write_to_fftw_input_buffers(silence_buffer, framesToCalc, audio);
/*
for(i=0; i<framesToCalc; i++) {
if(audio->channels == 1) audio->audio_out_l[n] = SAMPLE_SILENCE;
if(audio->channels == 2) {
@ -51,7 +53,10 @@ static int recordCallback(const void *inputBuffer, void *outputBuffer,
}
if(n == BUFSIZE-1) n = 0;
}
*/
} else {
write_to_fftw_input_buffers(rptr, framesToCalc, audio);
/*
for(i=0; i<framesToCalc; i++) {
if(audio->channels == 1) {
audio->audio_out_l[n] = (rptr[0] + rptr[1]) / 2;
@ -64,6 +69,7 @@ static int recordCallback(const void *inputBuffer, void *outputBuffer,
n++;
if(n == BUFSIZE-1) n = 0;
}
*/
}
data->frameIndex += framesToCalc;

24
input/pulse.c
Unescape View File

@ -112,7 +112,6 @@ void getPulseDefaultSink(void* data) {
void* input_pulse(void* data) {
struct audio_data *audio = (struct audio_data *)data;
int i, n;
int16_t buf[BUFFERSIZE / 2];
/* The sample type to use */
@ -121,11 +120,17 @@ void* input_pulse(void* data) {
.rate = 44100,
.channels = 2
};
audio->format = 16;
static const pa_buffer_attr pb = {
.maxlength = (uint32_t) -1, //BUFSIZE * 2,
.fragsize = BUFFERSIZE
};
uint16_t frames = BUFFERSIZE / 4;
pa_simple *s = NULL;
int error;
@ -140,7 +145,6 @@ void* input_pulse(void* data) {
pthread_exit(NULL);
}
n = 0;
while (1) {
if (pa_simple_read(s, buf, sizeof(buf), &error) < 0) {
@ -152,21 +156,7 @@ void* input_pulse(void* data) {
//sorting out channels
for (i = 0; i < BUFFERSIZE / 2; i += 2) {
if (audio->channels == 1) {
audio->audio_out_l[n] = (buf[i] + buf[i + 1]) / 2;
}
//stereo storing channels in buffer
if (audio->channels == 2) {
audio->audio_out_l[n] = buf[i];
audio->audio_out_r[n] = buf[i + 1];
}
n++;
if (n == audio->FFTbufferSize - 1) n = 0;
}
write_to_fftw_input_buffers(buf, frames, data);
if (audio->terminate == 1) {
pa_simple_free(s);

3
input/shmem.c
Unescape View File

@ -60,6 +60,8 @@ void* input_shmem(void* data)
audio->rate = mmap_area->rate;
while (1) {
write_to_fftw_input_buffers(mmap_area->buffer, BUFSIZE, audio);
/*
for (i = VB_OFFSET; i < BUFSIZE+VB_OFFSET; i += 2) {
if (audio->channels == 1) {
audio->audio_out_l[n] = (mmap_area->buffer[i] + mmap_area->buffer[i + 1]) / 2;
@ -70,6 +72,7 @@ void* input_shmem(void* data)
n++;
if (n == audio->FFTbufferSize - 1) n = 0;
}
*/|
if (audio->terminate == 1) {
break;
}

9
input/sndio.c
Unescape View File

@ -11,15 +11,12 @@ void* input_sndio(void* data)
int16_t buf[256];
unsigned int i, n, channels;
assert(audio->channels > 0);
channels = audio->channels;
sio_initpar(&par);
par.sig = 1;
par.bits = 16;
par.le = 1;
par.rate = 44100;
par.rchan = channels;
par.rchan = 2;
par.appbufsz = sizeof(buf) / channels;
if ((hdl = sio_open(audio->source, SIO_REC, 0)) == NULL) {
@ -38,12 +35,15 @@ void* input_sndio(void* data)
}
n = 0;
uint16_t frames = (sizeof(buf)/sizeof(buf[0]))/channels;
while (audio->terminate != 1) {
if (sio_read(hdl, buf, sizeof(buf)) == 0) {
fprintf(stderr, __FILE__": sio_read() failed: %s\n", strerror(errno));
exit(EXIT_FAILURE);
}
write_to_fftw_input_buffers(buf, frames, audio);
/*
for (i = 0; i < sizeof(buf)/sizeof(buf[0]); i += 2) {
if (par.rchan == 1) {
// sndiod has already taken care of averaging the samples
@ -54,6 +54,7 @@ void* input_sndio(void* data)
}
n = (n + 1) % audio->FFTbufferSize;
}
*/
}
sio_stop(hdl);

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