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-rw-r--r--audio/filter/af_resample.c394
1 files changed, 0 insertions, 394 deletions
diff --git a/audio/filter/af_resample.c b/audio/filter/af_resample.c
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index 1f0b7cc942..0000000000
--- a/audio/filter/af_resample.c
+++ /dev/null
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-/*
- * This audio filter changes the sample rate.
- *
- * Copyright (C) 2002 Anders Johansson ajh@atri.curtin.edu.au
- *
- * This file is part of MPlayer.
- *
- * MPlayer 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.
- *
- * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <inttypes.h>
-
-#include "libavutil/common.h"
-#include "libavutil/mathematics.h"
-#include "af.h"
-#include "dsp.h"
-
-/* Below definition selects the length of each poly phase component.
- Valid definitions are L8 and L16, where the number denotes the
- length of the filter. This definition affects the computational
- complexity (see play()), the performance (see filter.h) and the
- memory usage. The filter length is chosen to 8 if the machine is
- slow and to 16 if the machine is fast and has MMX.
-*/
-
-#if !HAVE_MMX // This machine is slow
-#define L8
-#else
-#define L16
-#endif
-
-#include "af_resample_template.c"
-
-// Filtering types
-#define RSMP_LIN (0<<0) // Linear interpolation
-#define RSMP_INT (1<<0) // 16 bit integer
-#define RSMP_FLOAT (2<<0) // 32 bit floating point
-#define RSMP_MASK (3<<0)
-
-// Defines for sloppy or exact resampling
-#define FREQ_SLOPPY (0<<2)
-#define FREQ_EXACT (1<<2)
-#define FREQ_MASK (1<<2)
-
-// Accuracy for linear interpolation
-#define STEPACCURACY 32
-
-// local data
-typedef struct af_resample_s
-{
- void* w; // Current filter weights
- void** xq; // Circular buffers
- uint32_t xi; // Index for circular buffers
- uint32_t wi; // Index for w
- uint32_t i; // Number of new samples to put in x queue
- uint32_t dn; // Down sampling factor
- uint32_t up; // Up sampling factor
- uint64_t step; // Step size for linear interpolation
- uint64_t pt; // Pointer remainder for linear interpolation
- int setup; // Setup parameters cmdline or through postcreate
-} af_resample_t;
-
-// Fast linear interpolation resample with modest audio quality
-static int linint(struct mp_audio* c,struct mp_audio* l, af_resample_t* s)
-{
- uint32_t len = 0; // Number of input samples
- uint32_t nch = l->nch; // Words pre transfer
- uint64_t step = s->step;
- int16_t* in16 = ((int16_t*)c->audio);
- int16_t* out16 = ((int16_t*)l->audio);
- int32_t* in32 = ((int32_t*)c->audio);
- int32_t* out32 = ((int32_t*)l->audio);
- uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
- uint64_t pt = s->pt;
- uint16_t tmp;
-
- switch (nch){
- case 1:
- while(pt < end){
- out16[len++]=in16[pt>>STEPACCURACY];
- pt+=step;
- }
- s->pt=pt & ((1LL<<STEPACCURACY)-1);
- break;
- case 2:
- end/=2;
- while(pt < end){
- out32[len++]=in32[pt>>STEPACCURACY];
- pt+=step;
- }
- len=(len<<1);
- s->pt=pt & ((1LL<<STEPACCURACY)-1);
- break;
- default:
- end /=nch;
- while(pt < end){
- tmp=nch;
- do {
- tmp--;
- out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
- } while (tmp);
- len+=nch;
- pt+=step;
- }
- s->pt=pt & ((1LL<<STEPACCURACY)-1);
- }
- return len;
-}
-
-/* Determine resampling type and format */
-static int set_types(struct af_instance* af, struct mp_audio* data)
-{
- af_resample_t* s = af->setup;
- int rv = AF_OK;
- float rd = 0;
-
- // Make sure this filter isn't redundant
- if((af->data->rate == data->rate) || (af->data->rate == 0))
- return AF_DETACH;
- /* If sloppy and small resampling difference (2%) */
- rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate;
- if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) &&
- (data->format != (AF_FORMAT_FLOAT_NE))) ||
- ((s->setup & RSMP_MASK) == RSMP_LIN)){
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
- af->data->format = AF_FORMAT_S16_NE;
- af->data->bps = 2;
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Using linear interpolation. \n");
- }
- else{
- /* If the input format is float or if float is explicitly selected
- use float, otherwise use int */
- if((data->format == (AF_FORMAT_FLOAT_NE)) ||
- ((s->setup & RSMP_MASK) == RSMP_FLOAT)){
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
- af->data->format = AF_FORMAT_FLOAT_NE;
- af->data->bps = 4;
- }
- else{
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
- af->data->format = AF_FORMAT_S16_NE;
- af->data->bps = 2;
- }
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Using %s processing and %s frequecy"
- " conversion.\n",
- ((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer",
- ((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact");
- }
-
- if(af->data->format != data->format || af->data->bps != data->bps)
- rv = AF_FALSE;
- data->format = af->data->format;
- data->bps = af->data->bps;
- af->data->nch = data->nch;
- return rv;
-}
-
-// Initialization and runtime control
-static int control(struct af_instance* af, int cmd, void* arg)
-{
- switch(cmd){
- case AF_CONTROL_REINIT:{
- af_resample_t* s = af->setup;
- struct mp_audio* n = arg; // New configuration
- int i,d = 0;
- int rv = AF_OK;
-
- // Free space for circular buffers
- if(s->xq){
- free(s->xq[0]);
- free(s->xq);
- s->xq = NULL;
- }
-
- if(AF_DETACH == (rv = set_types(af,n)))
- return AF_DETACH;
-
- // If linear interpolation
- if((s->setup & RSMP_MASK) == RSMP_LIN){
- s->pt=0LL;
- s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
- mp_msg(MSGT_AFILTER, MSGL_DBG2, "[resample] Linear interpolation step: 0x%016"PRIX64".\n",
- s->step);
- af->mul = (double)af->data->rate / n->rate;
- return rv;
- }
-
- // Calculate up and down sampling factors
- d=av_gcd(af->data->rate,n->rate);
-
- // If sloppy resampling is enabled limit the upsampling factor
- if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){
- int up=af->data->rate/2;
- int dn=n->rate/2;
- int m=2;
- while(af->data->rate/(d*m) > 5000){
- d=av_gcd(up,dn);
- up/=2; dn/=2; m*=2;
- }
- d*=m;
- }
-
- // Create space for circular buffers
- s->xq = malloc(n->nch*sizeof(void*));
- s->xq[0] = calloc(n->nch, 2*L*af->data->bps);
- for(i=1;i<n->nch;i++)
- s->xq[i] = (uint8_t *)s->xq[i-1] + 2*L*af->data->bps;
- s->xi = 0;
-
- // Check if the design needs to be redone
- if(s->up != af->data->rate/d || s->dn != n->rate/d){
- float* w;
- float* wt;
- float fc;
- int j;
- s->up = af->data->rate/d;
- s->dn = n->rate/d;
- s->wi = 0;
- s->i = 0;
-
- // Calculate cutoff frequency for filter
- fc = 1/(float)(max(s->up,s->dn));
- // Allocate space for polyphase filter bank and prototype filter
- w = malloc(sizeof(float) * s->up *L);
- free(s->w);
- s->w = malloc(L*s->up*af->data->bps);
-
- // Design prototype filter type using Kaiser window with beta = 10
- if(NULL == w || NULL == s->w ||
- -1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] Unable to design prototype filter.\n");
- return AF_ERROR;
- }
- // Copy data from prototype to polyphase filter
- wt=w;
- for(j=0;j<L;j++){//Columns
- for(i=0;i<s->up;i++){//Rows
- if((s->setup & RSMP_MASK) == RSMP_INT){
- float t=(float)s->up*32767.0*(*wt);
- ((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
- }
- else
- ((float*)s->w)[i*L+j] = (float)s->up*(*wt);
- wt++;
- }
- }
- free(w);
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] New filter designed up: %i "
- "down: %i\n", s->up, s->dn);
- }
-
- // Set multiplier and delay
- af->delay = 0; // not set correctly, but shouldn't be too large anyway
- af->mul = (double)s->up / s->dn;
- return rv;
- }
- case AF_CONTROL_COMMAND_LINE:{
- af_resample_t* s = af->setup;
- int rate=0;
- int type=RSMP_INT;
- int sloppy=1;
- sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type);
- s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) |
- (clamp(type,RSMP_LIN,RSMP_FLOAT));
- return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
- }
- case AF_CONTROL_POST_CREATE:
- if((((struct af_cfg*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
- ((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
- return AF_OK;
- case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
- // Reinit must be called after this function has been called
-
- // Sanity check
- if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] The output sample frequency "
- "must be between 8kHz and 192kHz. Current value is %i \n",
- ((int*)arg)[0]);
- return AF_ERROR;
- }
-
- af->data->rate=((int*)arg)[0];
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Changing sample rate "
- "to %iHz\n",af->data->rate);
- return AF_OK;
- }
- return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance* af)
-{
- af_resample_t *s = af->setup;
- if (s) {
- if (s->xq) free(s->xq[0]);
- free(s->xq);
- free(s->w);
- free(s);
- }
- if(af->data)
- free(af->data->audio);
- free(af->data);
-}
-
-// Filter data through filter
-static struct mp_audio* play(struct af_instance* af, struct mp_audio* data)
-{
- int len = 0; // Length of output data
- struct mp_audio* c = data; // Current working data
- struct mp_audio* l = af->data; // Local data
- af_resample_t* s = af->setup;
-
- if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
- return NULL;
-
- // Run resampling
- switch(s->setup & RSMP_MASK){
- case(RSMP_INT):
-# define FORMAT_I 1
- if(s->up>s->dn){
-# define UP
-# include "af_resample_template.c"
-# undef UP
- }
- else{
-# define DN
-# include "af_resample_template.c"
-# undef DN
- }
- break;
- case(RSMP_FLOAT):
-# undef FORMAT_I
-# define FORMAT_F 1
- if(s->up>s->dn){
-# define UP
-# include "af_resample_template.c"
-# undef UP
- }
- else{
-# define DN
-# include "af_resample_template.c"
-# undef DN
- }
- break;
- case(RSMP_LIN):
- len = linint(c, l, s);
- break;
- }
-
- // Set output data
- c->audio = l->audio;
- c->len = len*l->bps;
- c->rate = l->rate;
-
- return c;
-}
-
-// Allocate memory and set function pointers
-static int af_open(struct af_instance* af){
- af->control=control;
- af->uninit=uninit;
- af->play=play;
- af->mul=1;
- af->data=calloc(1,sizeof(struct mp_audio));
- af->setup=calloc(1,sizeof(af_resample_t));
- if(af->data == NULL || af->setup == NULL)
- return AF_ERROR;
- ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
- return AF_OK;
-}
-
-// Description of this plugin
-struct af_info af_info_resample = {
- "Sample frequency conversion",
- "resample",
- "Anders",
- "",
- AF_FLAGS_REENTRANT,
- af_open
-};