// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data)
{
af_data_t* c = data; // Current working data
af_delay_t* s = af->setup; // Setup for this instance
int nch = c->nch; // Number of channels
int len = c->len/c->bps; // Number of sample in data chunk
int ri = 0;
int ch,i;
for(ch=0;ch<nch;ch++){
switch(c->bps){
case 1:{
int8_t* a = c->audio;
int8_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
case 2:{
int16_t* a = c->audio;
int16_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
case 4:{
int32_t* a = c->audio;
int32_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
}
}
s->ri = ri;
return c;
}
// Filter data through filter
static int filter(struct af_instance* af, struct mp_audio* data, int flags)
{
struct mp_audio* c = data; // Current working data
af_delay_t* s = af->priv; // Setup for this instance
int nch = c->nch; // Number of channels
int len = mp_audio_psize(c)/c->bps; // Number of sample in data chunk
int ri = 0;
int ch,i;
for(ch=0;ch<nch;ch++){
switch(c->bps){
case 1:{
int8_t* a = c->planes[0];
int8_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
case 2:{
int16_t* a = c->planes[0];
int16_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
case 4:{
int32_t* a = c->planes[0];
int32_t* q = s->q[ch];
int wi = s->wi[ch];
ri = s->ri;
for(i=ch;i<len;i+=nch){
q[wi] = a[i];
a[i] = q[ri];
UPDATEQI(wi);
UPDATEQI(ri);
}
s->wi[ch] = wi;
break;
}
}
}
s->ri = ri;
return 0;
}
// Filter data through filter
static af_data_t* play(struct af_instance_s* af, af_data_t* data){
af_surround_t* s = (af_surround_t*)af->setup;
float* m = steering_matrix[0];
float* in = data->audio; // Input audio data
float* out = NULL; // Output audio data
float* end = in + data->len / sizeof(float); // Loop end
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
if (AF_OK != RESIZE_LOCAL_BUFFER(af, data))
return NULL;
out = af->data->audio;
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->audio = af->data->audio;
data->len = (data->len*af->mul.n)/af->mul.d;
data->nch = af->data->nch;
return data;
}
static int filter_frame(struct af_instance *af, struct mp_audio *data)
{
if (!data)
return 0;
struct mp_audio *outframe =
mp_audio_pool_get(af->out_pool, &af->fmt_out, data->samples);
if (!outframe) {
talloc_free(data);
return -1;
}
mp_audio_copy_attributes(outframe, data);
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 = outframe->planes[0]; // 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
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;
talloc_free(data);
af_add_output_frame(af, outframe);
return 0;
}
