/* Initialization and runtime control */
static int control(struct af_instance *af, int cmd, void* arg)
{
af_hrtf_t *s = af->priv;
int test_output_res;
switch(cmd) {
case AF_CONTROL_REINIT:
reset(s);
af->data->rate = 48000;
mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
if(af->data->nch == 2) {
/* 2 channel input */
if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
/* Default behavior is stereo mixing. */
s->decode_mode = HRTF_MIX_STEREO;
}
} else if (af->data->nch < 5) {
mp_audio_set_channels_old(af->data, 5);
}
mp_audio_set_format(af->data, AF_FORMAT_S16);
test_output_res = af_test_output(af, (struct mp_audio*)arg);
// after testing input set the real output format
mp_audio_set_num_channels(af->data, 2);
s->print_flag = 1;
return test_output_res;
case AF_CONTROL_RESET:
reset(s);
return AF_OK;
}
return AF_UNKNOWN;
}
/* Initialization and runtime control */
static int control(struct af_instance *af, int cmd, void* arg)
{
af_hrtf_t *s = af->setup;
int test_output_res;
char mode;
switch(cmd) {
case AF_CONTROL_REINIT:
af->data->rate = ((struct mp_audio*)arg)->rate;
if(af->data->rate != 48000) {
// automatic samplerate adjustment in the filter chain
// is not yet supported.
mp_msg(MSGT_AFILTER, MSGL_ERR,
"[hrtf] ERROR: Sampling rate is not 48000 Hz (%d)!\n",
af->data->rate);
return AF_ERROR;
}
mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch);
if(af->data->nch == 2) {
/* 2 channel input */
if(s->decode_mode != HRTF_MIX_MATRIX2CH) {
/* Default behavior is stereo mixing. */
s->decode_mode = HRTF_MIX_STEREO;
}
}
else if (af->data->nch < 5)
mp_audio_set_channels_old(af->data, 5);
mp_audio_set_format(af->data, AF_FORMAT_S16);
test_output_res = af_test_output(af, (struct mp_audio*)arg);
// after testing input set the real output format
mp_audio_set_num_channels(af->data, 2);
s->print_flag = 1;
return test_output_res;
case AF_CONTROL_COMMAND_LINE:
sscanf((char*)arg, "%c", &mode);
switch(mode) {
case 'm':
/* Use matrix rear decoding. */
s->matrix_mode = 1;
break;
case 's':
/* Input needs matrix decoding. */
s->decode_mode = HRTF_MIX_MATRIX2CH;
break;
case '0':
s->matrix_mode = 0;
break;
default:
mp_msg(MSGT_AFILTER, MSGL_ERR,
"[hrtf] Mode is neither 'm', 's', nor '0' (%c).\n",
mode);
return AF_ERROR;
}
s->print_flag = 1;
return AF_OK;
}
return AF_UNKNOWN;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_sub_t* s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
// Sanity check
if(!arg) return AF_ERROR;
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
// Design low-pass filter
s->k = 1.0;
if((-1 == af_filter_szxform(sp[0].a, sp[0].b, Q, s->fc,
(float)af->data->rate, &s->k, s->w[0])) ||
(-1 == af_filter_szxform(sp[1].a, sp[1].b, Q, s->fc,
(float)af->data->rate, &s->k, s->w[1])))
return AF_ERROR;
return af_test_output(af,(struct mp_audio*)arg);
}
case AF_CONTROL_COMMAND_LINE:{
int ch=5;
float fc=60.0;
sscanf(arg,"%f:%i", &fc , &ch);
if(AF_OK != control(af,AF_CONTROL_SUB_CH | AF_CONTROL_SET, &ch))
return AF_ERROR;
return control(af,AF_CONTROL_SUB_FC | AF_CONTROL_SET, &fc);
}
case AF_CONTROL_SUB_CH | AF_CONTROL_SET: // Requires reinit
// Sanity check
if((*(int*)arg >= AF_NCH) || (*(int*)arg < 0)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[sub] Subwoofer channel number must be between "
" 0 and %i current value is %i\n", AF_NCH-1, *(int*)arg);
return AF_ERROR;
}
s->ch = *(int*)arg;
return AF_OK;
case AF_CONTROL_SUB_CH | AF_CONTROL_GET:
*(int*)arg = s->ch;
return AF_OK;
case AF_CONTROL_SUB_FC | AF_CONTROL_SET: // Requires reinit
// Sanity check
if((*(float*)arg > 300) || (*(float*)arg < 20)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[sub] Cutoff frequency must be between 20Hz and"
" 300Hz current value is %0.2f",*(float*)arg);
return AF_ERROR;
}
// Set cutoff frequency
s->fc = *(float*)arg;
return AF_OK;
case AF_CONTROL_SUB_FC | AF_CONTROL_GET:
*(float*)arg = s->fc;
return AF_OK;
}
return AF_UNKNOWN;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_surround_t *s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
float fc;
mp_audio_copy_config(af->data, (struct mp_audio*)arg);
mp_audio_set_channels_old(af->data, ((struct mp_audio*)arg)->nch*2);
mp_audio_set_format(af->data, AF_FORMAT_FLOAT_NE);
if (af->data->nch != 4){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Only stereo input is supported.\n");
return AF_DETACH;
}
// Surround filer coefficients
fc = 2.0 * 7000.0/(float)af->data->rate;
if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Unable to design low-pass filter.\n");
return AF_ERROR;
}
// Free previous delay queues
free(s->dl);
free(s->dr);
// Allocate new delay queues
s->dl = calloc(LD,af->data->bps);
s->dr = calloc(LD,af->data->bps);
if((NULL == s->dl) || (NULL == s->dr))
mp_msg(MSGT_AFILTER, MSGL_FATAL, "[delay] Out of memory\n");
// Initialize delay queue index
if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
return AF_ERROR;
// printf("%i\n",s->wi);
s->ri = 0;
if((af->data->format != ((struct mp_audio*)arg)->format) ||
(af->data->bps != ((struct mp_audio*)arg)->bps)){
mp_audio_set_format((struct mp_audio*)arg, af->data->format);
return AF_FALSE;
}
return AF_OK;
}
case AF_CONTROL_COMMAND_LINE:{
float d = 0;
sscanf((char*)arg,"%f",&d);
if ((d < 0) || (d > 1000)){
mp_msg(MSGT_AFILTER, MSGL_ERR, "[surround] Invalid delay time, valid time values"
" are 0ms to 1000ms current value is %0.3f ms\n",d);
return AF_ERROR;
}
s->d = d;
return AF_OK;
}
}
return AF_UNKNOWN;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_center_t* s = af->priv;
switch(cmd){
case AF_CONTROL_REINIT:{
// Sanity check
if(!arg) return AF_ERROR;
af->data->rate = ((struct mp_audio*)arg)->rate;
mp_audio_set_channels_old(af->data, MPMAX(s->ch+1,((struct mp_audio*)arg)->nch));
mp_audio_set_format(af->data, AF_FORMAT_FLOAT);
return af_test_output(af,(struct mp_audio*)arg);
}
}
return AF_UNKNOWN;
}
// Initialization and runtime control
static int control(struct af_instance* af, int cmd, void* arg)
{
af_surround_t *s = af->priv;
switch(cmd){
case AF_CONTROL_REINIT:{
struct mp_audio *in = arg;
float fc;
if (!mp_chmap_is_stereo(&in->channels)) {
MP_ERR(af, "Only stereo input is supported.\n");
return AF_DETACH;
}
mp_audio_set_format(in, AF_FORMAT_FLOAT);
mp_audio_copy_config(af->data, in);
mp_audio_set_channels_old(af->data, in->nch * 2);
// Surround filer coefficients
fc = 2.0 * 7000.0/(float)af->data->rate;
if (-1 == af_filter_design_fir(L, s->w, &fc, LP|HAMMING, 0)){
MP_ERR(af, "Unable to design low-pass filter.\n");
return AF_ERROR;
}
// Free previous delay queues
free(s->dl);
free(s->dr);
// Allocate new delay queues
s->dl = calloc(LD,af->data->bps);
s->dr = calloc(LD,af->data->bps);
if((NULL == s->dl) || (NULL == s->dr))
MP_FATAL(af, "Out of memory\n");
// Initialize delay queue index
if(AF_OK != af_from_ms(1, &s->d, &s->wi, af->data->rate, 0.0, 1000.0))
return AF_ERROR;
// printf("%i\n",s->wi);
s->ri = 0;
return AF_OK;
}
}
return AF_UNKNOWN;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags){
af_surround_t* s = (af_surround_t*)af->priv;
const float* m = steering_matrix[0];
float* in = data->planes[0]; // Input audio data
float* out = NULL; // Output audio data
float* end = in + data->samples * data->nch;
int i = s->i; // Filter queue index
int ri = s->ri; // Read index for delay queue
int wi = s->wi; // Write index for delay queue
mp_audio_realloc_min(af->data, data->samples);
out = af->data->planes[0];
while(in < end){
/* Dominance:
abs(in[0]) abs(in[1]);
abs(in[0]+in[1]) abs(in[0]-in[1]);
10 * log( abs(in[0]) / (abs(in[1])|1) );
10 * log( abs(in[0]+in[1]) / (abs(in[0]-in[1])|1) ); */
/* About volume balancing...
Surround encoding does the following:
Lt=L+.707*C+.707*S, Rt=R+.707*C-.707*S
So S should be extracted as:
(Lt-Rt)
But we are splitting the S to two output channels, so we
must take 3dB off as we split it:
Ls=Rs=.707*(Lt-Rt)
Trouble is, Lt could be +1, Rt -1, so possibility that S will
overflow. So to avoid that, we cut L/R by 3dB (*.707), and S by
6dB (/2). This keeps the overall balance, but guarantees no
overflow. */
// Output front left and right
out[0] = m[0]*in[0] + m[1]*in[1];
out[1] = m[2]*in[0] + m[3]*in[1];
// Low-pass output @ 7kHz
FIR((&s->lq[i]), s->w, s->dl[wi]);
// Delay output by d ms
out[2] = s->dl[ri];
#ifdef SPLITREAR
// Low-pass output @ 7kHz
FIR((&s->rq[i]), s->w, s->dr[wi]);
// Delay output by d ms
out[3] = s->dr[ri];
#else
out[3] = -out[2];
#endif
// Update delay queues indexes
UPDATEQI(ri);
UPDATEQI(wi);
// Calculate and save surround in circular queue
#ifdef SPLITREAR
ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
#else
ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
#endif
// Next sample...
in = &in[data->nch];
out = &out[af->data->nch];
}
// Save indexes
s->i = i; s->ri = ri; s->wi = wi;
// Set output data
data->planes[0] = af->data->planes[0];
mp_audio_set_channels_old(data, af->data->nch);
return 0;
}