mirror of
https://github.com/mpv-player/mpv
synced 2024-12-22 14:52:43 +00:00
2d3d563b15
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@21074 b3059339-0415-0410-9bf9-f77b7e298cf2
382 lines
10 KiB
C
382 lines
10 KiB
C
/*=============================================================================
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//
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// This software has been released under the terms of the GNU General Public
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// license. See http://www.gnu.org/copyleft/gpl.html for details.
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//
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// Copyright 2002 Anders Johansson ajh@atri.curtin.edu.au
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//
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//=============================================================================
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*/
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/* This audio filter changes the sample rate. */
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#include <stdio.h>
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#include <stdlib.h>
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#include <unistd.h>
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#include <inttypes.h>
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#include "af.h"
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#include "dsp.h"
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/* Below definition selects the length of each poly phase component.
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Valid definitions are L8 and L16, where the number denotes the
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length of the filter. This definition affects the computational
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complexity (see play()), the performance (see filter.h) and the
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memory usage. The filterlenght is choosen to 8 if the machine is
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slow and to 16 if the machine is fast and has MMX.
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*/
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#if !defined(HAVE_MMX) // This machine is slow
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#define L8
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#else
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#define L16
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#endif
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#include "af_resample.h"
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// Filtering types
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#define RSMP_LIN (0<<0) // Linear interpolation
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#define RSMP_INT (1<<0) // 16 bit integer
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#define RSMP_FLOAT (2<<0) // 32 bit floating point
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#define RSMP_MASK (3<<0)
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// Defines for sloppy or exact resampling
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#define FREQ_SLOPPY (0<<2)
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#define FREQ_EXACT (1<<2)
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#define FREQ_MASK (1<<2)
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// Accuracy for linear interpolation
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#define STEPACCURACY 32
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// local data
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typedef struct af_resample_s
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{
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void* w; // Current filter weights
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void** xq; // Circular buffers
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uint32_t xi; // Index for circular buffers
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uint32_t wi; // Index for w
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uint32_t i; // Number of new samples to put in x queue
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uint32_t dn; // Down sampling factor
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uint32_t up; // Up sampling factor
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uint64_t step; // Step size for linear interpolation
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uint64_t pt; // Pointer remainder for linear interpolation
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int setup; // Setup parameters cmdline or through postcreate
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} af_resample_t;
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// Fast linear interpolation resample with modest audio quality
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static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
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{
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uint32_t len = 0; // Number of input samples
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uint32_t nch = l->nch; // Words pre transfer
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uint64_t step = s->step;
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int16_t* in16 = ((int16_t*)c->audio);
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int16_t* out16 = ((int16_t*)l->audio);
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int32_t* in32 = ((int32_t*)c->audio);
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int32_t* out32 = ((int32_t*)l->audio);
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uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
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uint64_t pt = s->pt;
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uint16_t tmp;
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switch (nch){
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case 1:
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while(pt < end){
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out16[len++]=in16[pt>>STEPACCURACY];
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pt+=step;
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}
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s->pt=pt & ((1LL<<STEPACCURACY)-1);
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break;
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case 2:
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end/=2;
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while(pt < end){
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out32[len++]=in32[pt>>STEPACCURACY];
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pt+=step;
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}
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len=(len<<1);
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s->pt=pt & ((1LL<<STEPACCURACY)-1);
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break;
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default:
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end /=nch;
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while(pt < end){
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tmp=nch;
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do {
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tmp--;
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out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
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} while (tmp);
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len+=nch;
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pt+=step;
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}
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s->pt=pt & ((1LL<<STEPACCURACY)-1);
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}
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return len;
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}
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/* Determine resampling type and format */
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static int set_types(struct af_instance_s* af, af_data_t* data)
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{
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af_resample_t* s = af->setup;
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int rv = AF_OK;
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float rd = 0;
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// Make sure this filter isn't redundant
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if((af->data->rate == data->rate) || (af->data->rate == 0))
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return AF_DETACH;
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/* If sloppy and small resampling difference (2%) */
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rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate;
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if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) &&
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(data->format != (AF_FORMAT_FLOAT_NE))) ||
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((s->setup & RSMP_MASK) == RSMP_LIN)){
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s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
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af->data->format = AF_FORMAT_S16_NE;
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af->data->bps = 2;
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af_msg(AF_MSG_VERBOSE,"[resample] Using linear interpolation. \n");
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}
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else{
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/* If the input format is float or if float is explicitly selected
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use float, otherwise use int */
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if((data->format == (AF_FORMAT_FLOAT_NE)) ||
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((s->setup & RSMP_MASK) == RSMP_FLOAT)){
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s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
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af->data->format = AF_FORMAT_FLOAT_NE;
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af->data->bps = 4;
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}
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else{
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s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
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af->data->format = AF_FORMAT_S16_NE;
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af->data->bps = 2;
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}
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af_msg(AF_MSG_VERBOSE,"[resample] Using %s processing and %s frequecy"
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" conversion.\n",
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((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer",
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((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact");
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}
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if(af->data->format != data->format || af->data->bps != data->bps)
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rv = AF_FALSE;
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data->format = af->data->format;
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data->bps = af->data->bps;
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af->data->nch = data->nch;
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return rv;
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}
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// Initialization and runtime control
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static int control(struct af_instance_s* af, int cmd, void* arg)
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{
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switch(cmd){
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case AF_CONTROL_REINIT:{
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af_resample_t* s = (af_resample_t*)af->setup;
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af_data_t* n = (af_data_t*)arg; // New configureation
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int i,d = 0;
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int rv = AF_OK;
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// Free space for circular bufers
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if(s->xq){
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for(i=1;i<af->data->nch;i++)
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if(s->xq[i])
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free(s->xq[i]);
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free(s->xq);
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s->xq = NULL;
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}
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if(AF_DETACH == (rv = set_types(af,n)))
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return AF_DETACH;
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// If linear interpolation
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if((s->setup & RSMP_MASK) == RSMP_LIN){
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s->pt=0LL;
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s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
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af_msg(AF_MSG_DEBUG0,"[resample] Linear interpolation step: 0x%016"PRIX64".\n",
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s->step);
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af->mul.n = af->data->rate;
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af->mul.d = n->rate;
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af_frac_cancel(&af->mul);
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return rv;
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}
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// Calculate up and down sampling factors
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d=af_gcd(af->data->rate,n->rate);
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// If sloppy resampling is enabled limit the upsampling factor
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if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){
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int up=af->data->rate/2;
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int dn=n->rate/2;
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int m=2;
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while(af->data->rate/(d*m) > 5000){
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d=af_gcd(up,dn);
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up/=2; dn/=2; m*=2;
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}
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d*=m;
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}
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// Create space for circular bufers
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s->xq = malloc(n->nch*sizeof(void*));
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for(i=0;i<n->nch;i++)
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s->xq[i] = malloc(2*L*af->data->bps);
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s->xi = 0;
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// Check if the the design needs to be redone
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if(s->up != af->data->rate/d || s->dn != n->rate/d){
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float* w;
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float* wt;
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float fc;
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int j;
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s->up = af->data->rate/d;
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s->dn = n->rate/d;
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s->wi = 0;
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s->i = 0;
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// Calculate cuttof frequency for filter
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fc = 1/(float)(max(s->up,s->dn));
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// Allocate space for polyphase filter bank and protptype filter
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w = malloc(sizeof(float) * s->up *L);
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if(NULL != s->w)
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free(s->w);
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s->w = malloc(L*s->up*af->data->bps);
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// Design prototype filter type using Kaiser window with beta = 10
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if(NULL == w || NULL == s->w ||
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-1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
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af_msg(AF_MSG_ERROR,"[resample] Unable to design prototype filter.\n");
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return AF_ERROR;
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}
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// Copy data from prototype to polyphase filter
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wt=w;
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for(j=0;j<L;j++){//Columns
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for(i=0;i<s->up;i++){//Rows
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if((s->setup & RSMP_MASK) == RSMP_INT){
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float t=(float)s->up*32767.0*(*wt);
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((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
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}
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else
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((float*)s->w)[i*L+j] = (float)s->up*(*wt);
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wt++;
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}
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}
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free(w);
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af_msg(AF_MSG_VERBOSE,"[resample] New filter designed up: %i "
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"down: %i\n", s->up, s->dn);
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}
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// Set multiplier and delay
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af->delay = (double)(1000*L/2)/((double)n->rate);
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af->mul.n = s->up;
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af->mul.d = s->dn;
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return rv;
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}
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case AF_CONTROL_COMMAND_LINE:{
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af_resample_t* s = (af_resample_t*)af->setup;
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int rate=0;
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int type=RSMP_INT;
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int sloppy=1;
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sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type);
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s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) |
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(clamp(type,RSMP_LIN,RSMP_FLOAT));
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return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
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}
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case AF_CONTROL_POST_CREATE:
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if((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
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((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
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return AF_OK;
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case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
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// Reinit must be called after this function has been called
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// Sanity check
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if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
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af_msg(AF_MSG_ERROR,"[resample] The output sample frequency "
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"must be between 8kHz and 192kHz. Current value is %i \n",
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((int*)arg)[0]);
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return AF_ERROR;
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}
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af->data->rate=((int*)arg)[0];
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af_msg(AF_MSG_VERBOSE,"[resample] Changing sample rate "
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"to %iHz\n",af->data->rate);
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return AF_OK;
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}
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return AF_UNKNOWN;
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}
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// Deallocate memory
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static void uninit(struct af_instance_s* af)
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{
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if(af->data)
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free(af->data);
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}
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// Filter data through filter
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static af_data_t* play(struct af_instance_s* af, af_data_t* data)
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{
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int len = 0; // Length of output data
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af_data_t* c = data; // Current working data
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af_data_t* l = af->data; // Local data
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af_resample_t* s = (af_resample_t*)af->setup;
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if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
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return NULL;
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// Run resampling
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switch(s->setup & RSMP_MASK){
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case(RSMP_INT):
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# define FORMAT_I 1
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if(s->up>s->dn){
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# define UP
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# include "af_resample.h"
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# undef UP
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}
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else{
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# define DN
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# include "af_resample.h"
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# undef DN
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}
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break;
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case(RSMP_FLOAT):
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# undef FORMAT_I
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# define FORMAT_F 1
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if(s->up>s->dn){
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# define UP
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# include "af_resample.h"
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# undef UP
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}
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else{
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# define DN
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# include "af_resample.h"
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# undef DN
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}
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break;
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case(RSMP_LIN):
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len = linint(c, l, s);
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break;
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}
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// Set output data
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c->audio = l->audio;
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c->len = len*l->bps;
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c->rate = l->rate;
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return c;
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}
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// Allocate memory and set function pointers
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static int open(af_instance_t* af){
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af->control=control;
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af->uninit=uninit;
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af->play=play;
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af->mul.n=1;
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af->mul.d=1;
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af->data=calloc(1,sizeof(af_data_t));
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af->setup=calloc(1,sizeof(af_resample_t));
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if(af->data == NULL || af->setup == NULL)
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return AF_ERROR;
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((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
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return AF_OK;
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}
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// Description of this plugin
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af_info_t af_info_resample = {
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"Sample frequency conversion",
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"resample",
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"Anders",
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"",
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AF_FLAGS_REENTRANT,
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open
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};
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