mirror of https://github.com/mpv-player/mpv
278 lines
8.1 KiB
C
278 lines
8.1 KiB
C
/*
|
|
This is an libaf filter to do simple decoding of matrixed surround
|
|
sound. This will provide a (basic) surround-sound effect from
|
|
audio encoded for Dolby Surround, Pro Logic etc.
|
|
|
|
* This program is free software; you can redistribute it and/or modify
|
|
* it under the terms of the GNU General Public License as published by
|
|
* the Free Software Foundation; either version 2 of the License, or
|
|
* (at your option) any later version.
|
|
*
|
|
* This program is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
|
|
* GNU General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU General Public License
|
|
* along with this program; if not, write to the Free Software
|
|
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
|
|
|
|
Original author: Steve Davies <steve@daviesfam.org>
|
|
*/
|
|
|
|
/* The principle: Make rear channels by extracting anti-phase data
|
|
from the front channels, delay by 20ms and feed to rear in anti-phase
|
|
*/
|
|
|
|
|
|
/* SPLITREAR: Define to decode two distinct rear channels - this
|
|
doesn't work so well in practice because separation in a passive
|
|
matrix is not high. C (dialogue) to Ls and Rs 14dB or so - so
|
|
dialogue leaks to the rear. Still - give it a try and send
|
|
feedback. Comment this define for old behavior of a single
|
|
surround sent to rear in anti-phase */
|
|
#define SPLITREAR 1
|
|
|
|
#include <stdio.h>
|
|
#include <stdlib.h>
|
|
#include <string.h>
|
|
#include <unistd.h>
|
|
|
|
#include "af.h"
|
|
#include "dsp.h"
|
|
|
|
#define L 32 // Length of fir filter
|
|
#define LD 65536 // Length of delay buffer
|
|
|
|
// 32 Tap fir filter loop unrolled
|
|
#define FIR(x,w,y) \
|
|
y = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
|
|
+ w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
|
|
+ w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
|
|
+ w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] \
|
|
+ w[16]*x[16]+w[17]*x[17]+w[18]*x[18]+w[19]*x[19] \
|
|
+ w[20]*x[20]+w[21]*x[21]+w[22]*x[22]+w[23]*x[23] \
|
|
+ w[24]*x[24]+w[25]*x[25]+w[26]*x[26]+w[27]*x[27] \
|
|
+ w[28]*x[28]+w[29]*x[29]+w[30]*x[30]+w[31]*x[31])
|
|
|
|
// Add to circular queue macro + update index
|
|
#ifdef SPLITREAR
|
|
#define ADDQUE(qi,rq,lq,r,l)\
|
|
lq[qi]=lq[qi+L]=(l);\
|
|
rq[qi]=rq[qi+L]=(r);\
|
|
qi=(qi-1)&(L-1);
|
|
#else
|
|
#define ADDQUE(qi,lq,l)\
|
|
lq[qi]=lq[qi+L]=(l);\
|
|
qi=(qi-1)&(L-1);
|
|
#endif
|
|
|
|
// Macro for updating queue index in delay queues
|
|
#define UPDATEQI(qi) qi=(qi+1)&(LD-1)
|
|
|
|
// instance data
|
|
typedef struct af_surround_s
|
|
{
|
|
float lq[2*L]; // Circular queue for filtering left rear channel
|
|
float rq[2*L]; // Circular queue for filtering right rear channel
|
|
float w[L]; // FIR filter coefficients for surround sound 7kHz low-pass
|
|
float* dr; // Delay queue right rear channel
|
|
float* dl; // Delay queue left rear channel
|
|
float d; // Delay time
|
|
int i; // Position in circular buffer
|
|
int wi; // Write index for delay queue
|
|
int ri; // Read index for delay queue
|
|
}af_surround_t;
|
|
|
|
// Initialization and runtime control
|
|
static int control(struct af_instance_s* af, int cmd, void* arg)
|
|
{
|
|
af_surround_t *s = af->setup;
|
|
switch(cmd){
|
|
case AF_CONTROL_REINIT:{
|
|
float fc;
|
|
af->data->rate = ((af_data_t*)arg)->rate;
|
|
af->data->nch = ((af_data_t*)arg)->nch*2;
|
|
af->data->format = AF_FORMAT_F | AF_FORMAT_NE;
|
|
af->data->bps = 4;
|
|
|
|
if (af->data->nch != 4){
|
|
af_msg(AF_MSG_ERROR,"[surround] Only stereo input is supported.\n");
|
|
return AF_DETACH;
|
|
}
|
|
// Surround filer coefficients
|
|
fc = 2.0 * 7000.0/(float)af->data->rate;
|
|
if (-1 == design_fir(L, s->w, &fc, LP|HAMMING, 0)){
|
|
af_msg(AF_MSG_ERROR,"[surround] Unable to design low-pass filter.\n");
|
|
return AF_ERROR;
|
|
}
|
|
|
|
// Free previous delay queues
|
|
if(s->dl)
|
|
free(s->dl);
|
|
if(s->dr)
|
|
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))
|
|
af_msg(AF_MSG_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 != ((af_data_t*)arg)->format) ||
|
|
(af->data->bps != ((af_data_t*)arg)->bps)){
|
|
((af_data_t*)arg)->format = af->data->format;
|
|
((af_data_t*)arg)->bps = af->data->bps;
|
|
return AF_FALSE;
|
|
}
|
|
return AF_OK;
|
|
}
|
|
case AF_CONTROL_COMMAND_LINE:{
|
|
float d = 0;
|
|
sscanf((char*)arg,"%f",&d);
|
|
if ((d < 0) || (d > 1000)){
|
|
af_msg(AF_MSG_ERROR,"[surround] Invalid delay time, valid time values"
|
|
" are 0ms to 1000ms current value is %0.3ms\n",d);
|
|
return AF_ERROR;
|
|
}
|
|
s->d = d;
|
|
return AF_OK;
|
|
}
|
|
}
|
|
return AF_UNKNOWN;
|
|
}
|
|
|
|
// Deallocate memory
|
|
static void uninit(struct af_instance_s* af)
|
|
{
|
|
if(af->data->audio)
|
|
free(af->data->audio);
|
|
if(af->data)
|
|
free(af->data);
|
|
if(af->setup)
|
|
free(af->setup);
|
|
}
|
|
|
|
// The beginnings of an active matrix...
|
|
static float steering_matrix[][12] = {
|
|
// LL RL LR RR LS RS
|
|
// LLs RLs LRs RRs LC RC
|
|
{.707, .0, .0, .707, .5, -.5,
|
|
.5878, -.3928, .3928, -.5878, .5, .5},
|
|
};
|
|
|
|
// Experimental moving average dominance
|
|
//static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 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 open(af_instance_t* af){
|
|
af->control=control;
|
|
af->uninit=uninit;
|
|
af->play=play;
|
|
af->mul.n=2;
|
|
af->mul.d=1;
|
|
af->data=calloc(1,sizeof(af_data_t));
|
|
af->setup=calloc(1,sizeof(af_surround_t));
|
|
if(af->data == NULL || af->setup == NULL)
|
|
return AF_ERROR;
|
|
((af_surround_t*)af->setup)->d = 20;
|
|
return AF_OK;
|
|
}
|
|
|
|
af_info_t af_info_surround =
|
|
{
|
|
"Surround decoder filter",
|
|
"surround",
|
|
"Steve Davies <steve@daviesfam.org>",
|
|
"",
|
|
AF_FLAGS_NOT_REENTRANT,
|
|
open
|
|
};
|