diff options
Diffstat (limited to 'audio/filter/af_resample.c')
-rw-r--r-- | audio/filter/af_resample.c | 394 |
1 files changed, 0 insertions, 394 deletions
diff --git a/audio/filter/af_resample.c b/audio/filter/af_resample.c deleted file mode 100644 index 1f0b7cc942..0000000000 --- a/audio/filter/af_resample.c +++ /dev/null @@ -1,394 +0,0 @@ -/* - * 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 -}; |