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Speed optimizations (runs twise as fast) and bugfix (wrong cutoff frequency buffer over run noise and garbeled output when wrong input format) 

git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@8764 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
anders 2003-01-04 06:19:25 +00:00
parent c4f3964dad
commit 8845dd707c
1 changed files with 143 additions and 114 deletions
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257
libaf/af_surround.c
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@ -1,5 +1,5 @@
/*
This is an ao2 plugin to do simple decoding of matrixed surround
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.
@ -21,19 +21,17 @@
*/
/* The principle: Make rear channels by extracting anti-phase data
from the front channels, delay by 20msec and feed to rear in anti-phase
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 behaviour of a single
// surround sent to rear in anti-phase
#define SPLITREAR
/* 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>
@ -43,66 +41,106 @@
#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 msecs; // Rear channel delay in milliseconds
float* Ls_delaybuf; // circular buffer to be used for delaying Ls audio
float* Rs_delaybuf; // circular buffer to be used for delaying Rs audio
int delaybuf_len; // delaybuf buffer length in samples
int delaybuf_rpos; // offset in buffer where we are reading
int delaybuf_wpos; // offset in buffer where we are writing
float filter_coefs_surround[32]; // FIR filter coefficients for surround sound 7kHz lowpass
} af_surround_t;
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 *instance=af->setup;
af_surround_t *s = af->setup;
switch(cmd){
case AF_CONTROL_REINIT:{
float cutoff;
float fc;
af->data->rate = ((af_data_t*)arg)->rate;
af->data->nch = ((af_data_t*)arg)->nch*2;
af->data->format = ((af_data_t*)arg)->format;
af->data->bps = ((af_data_t*)arg)->bps;
af_msg(AF_MSG_DEBUG0, "[surround]: rear delay=%0.2fms.\n", instance->msecs);
af->data->format = AF_FORMAT_F | AF_FORMAT_NE;
af->data->bps = 4;
if (af->data->nch != 4){
af_msg(AF_MSG_ERROR,"Only Stereo input is supported, filter disabled.\n");
af_msg(AF_MSG_ERROR,"[surround] Only stereo input is supported.\n");
return AF_DETACH;
}
// Figure out buffer space (in int16_ts) needed for the 15msec delay
// Extra 31 samples allow for lowpass filter delay (taps-1)
// Double size to make virtual ringbuffer easier
instance->delaybuf_len = ((af->data->rate * instance->msecs / 1000)+31)*2;
// Free old buffers
if (instance->Ls_delaybuf != NULL)
free(instance->Ls_delaybuf);
if (instance->Rs_delaybuf != NULL)
free(instance->Rs_delaybuf);
// Allocate new buffers
instance->Ls_delaybuf=(void*)calloc(instance->delaybuf_len,sizeof(*instance->Ls_delaybuf));
instance->Rs_delaybuf=(void*)calloc(instance->delaybuf_len,sizeof(*instance->Rs_delaybuf));
af_msg(AF_MSG_DEBUG1, "Delay buffers are %d samples each.\n", instance->delaybuf_len);
instance->delaybuf_wpos = 0;
instance->delaybuf_rpos = 32; // compensate for fir delay
// Surround filer coefficients
cutoff = af->data->rate/7000;
if (-1 == design_fir(32, instance->filter_coefs_surround, &cutoff, LP|KAISER, 10.0)) {
af_msg(AF_MSG_ERROR,"[surround] Unable to design prototype filter.\n");
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){
af_msg(AF_MSG_ERROR,"Error setting rear delay length in af_surround. Delay has to be positive.\n");
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;
}
instance->msecs=d;
s->d = d;
return AF_OK;
}
}
@ -112,108 +150,100 @@ static int control(struct af_instance_s* af, int cmd, void* arg)
// Deallocate memory
static void uninit(struct af_instance_s* af)
{
af_surround_t *instance=af->setup;
if(af->data->audio)
free(af->data->audio);
if(af->data)
free(af->data);
if(instance->Ls_delaybuf)
free(instance->Ls_delaybuf);
if(instance->Rs_delaybuf)
free(instance->Rs_delaybuf);
free(af->setup);
if(af->setup)
free(af->setup);
}
// The beginnings of an active matrix...
static double steering_matrix[][12] = {
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 dominances
// 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* instance = (af_surround_t*)af->setup;
int16_t* in = data->audio;
int16_t* out;
int i, samples;
double *matrix = steering_matrix[0]; // later we'll index based on detected dominance
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;
// fprintf(stderr, "pl_surround: play %d bytes, %d samples\n", ao_plugin_data.len, samples);
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) ); */
samples = data->len / (data->nch * sizeof(int16_t));
/* 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. */
// Testing - place a 1kHz tone on Lt and Rt in anti-phase: should decode in S
//sinewave(in, samples, pl_surround.input_channels, 1000, 0.0, pl_surround.rate);
//sinewave(&in[1], samples, pl_surround.input_channels, 1000, PI, pl_surround.rate);
// 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];
for (i=0; i<samples; i++) {
// Low-pass output @ 7kHz
FIR((&s->lq[i]), s->w, s->dl[wi]);
// 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) );
// Delay output by d ms
out[2] = s->dl[ri];
// 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 +32767, Rt -32768, 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] = matrix[0]*in[0] + matrix[1]*in[1];
out[1] = matrix[2]*in[0] + matrix[3]*in[1];
// output Ls and Rs - from 20msec ago, lowpass filtered @ 7kHz
out[2] = fir(32, instance->filter_coefs_surround,
&instance->Ls_delaybuf[instance->delaybuf_rpos +
instance->delaybuf_len/2]);
#ifdef SPLITREAR
out[3] = fir(32, instance->filter_coefs_surround,
&instance->Rs_delaybuf[instance->delaybuf_rpos +
instance->delaybuf_len/2]);
// 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
// calculate and save surround for 20msecs time
// Update delay queues indexes
UPDATEQI(ri);
UPDATEQI(wi);
// Calculate and save surround in circular queue
#ifdef SPLITREAR
instance->Ls_delaybuf[instance->delaybuf_wpos] =
instance->Ls_delaybuf[instance->delaybuf_wpos + instance->delaybuf_len/2] =
matrix[6]*in[0] + matrix[7]*in[1];
instance->Rs_delaybuf[instance->delaybuf_wpos] =
instance->Rs_delaybuf[instance->delaybuf_wpos++ + instance->delaybuf_len/2] =
matrix[8]*in[0] + matrix[9]*in[1];
ADDQUE(i, s->rq, s->lq, m[6]*in[0]+m[7]*in[1], m[8]*in[0]+m[9]*in[1]);
#else
instance->Ls_delaybuf[instance->delaybuf_wpos] =
instance->Ls_delaybuf[instance->delaybuf_wpos++ + instance->delaybuf_len/2] =
matrix[4]*in[0] + matrix[5]*in[1];
ADDQUE(i, s->lq, m[4]*in[0]+m[5]*in[1]);
#endif
instance->delaybuf_rpos++;
instance->delaybuf_wpos %= instance->delaybuf_len/2;
instance->delaybuf_rpos %= instance->delaybuf_len/2;
// next samples...
in = &in[data->nch]; out = &out[af->data->nch];
// Next sample...
in = &in[data->nch];
out = &out[af->data->nch];
}
// Show some state
//printf("\npl_surround: delaybuf_pos=%d, samples=%d\r\033[A", pl_surround.delaybuf_pos, samples);
// 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;
@ -223,17 +253,16 @@ static af_data_t* play(struct af_instance_s* af, af_data_t* data){
}
static int open(af_instance_t* af){
af_surround_t *pl_surround;
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=pl_surround=calloc(1,sizeof(af_surround_t));
pl_surround->msecs=20;
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;
}
@ -243,6 +272,6 @@ af_info_t af_info_surround =
"surround",
"Steve Davies <steve@daviesfam.org>",
"",
AF_FLAGS_REENTRANT,
AF_FLAGS_NOT_REENTRANT,
open
};