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mpv/libao2/pl_resample.c
alex d65c8518de 64bit libao2 fix by Jens Axboe <mplayer-dev@kernel.dk>
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@9634 b3059339-0415-0410-9bf9-f77b7e298cf2
2003-03-21 16:42:50 +00:00

286 lines
7.9 KiB
C

/*=============================================================================
//
// This software has been released under the terms of the GNU Public
// license. See http://www.gnu.org/copyleft/gpl.html for details.
//
// Copyright 2001 Anders Johansson ajh@atri.curtin.edu.au
//
//=============================================================================
*/
/* This audio output plugin changes the sample rate. The output
samplerate from this plugin is specified by using the switch
`fout=F' where F is the desired output sample frequency
*/
#define PLUGIN
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include <inttypes.h>
#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
static ao_info_t info =
{
"Sample frequency conversion audio plugin",
"resample",
"Anders",
""
};
LIBAO_PLUGIN_EXTERN(resample)
#define min(a,b) (((a) < (b)) ? (a) : (b))
#define max(a,b) (((a) > (b)) ? (a) : (b))
/* 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 filterlenght is choosen to 8 if the machine is
slow and to 16 if the machine is fast and has MMX.
*/
#if !defined(HAVE_SSE) && !defined(HAVE_3DNOW) //This machine is slow
#define W W8 // Filter bank parameters
#define L 8 // Filter length
#ifdef HAVE_MMX
#define FIR(x,w,y) *y=(int16_t)firn(x,w,8);
#else /* HAVE_MMX */
// Unrolled loop to speed up execution
#define FIR(x,w,y){ \
int16_t a = (w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3]) >> 16; \
int16_t b = (w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7]) >> 16; \
y[0] = a+b; \
}
#endif /* HAVE_MMX */
#else /* Fast machine */
#define W W16
#define L 16
#define FIR(x,w,y) *y=(int16_t)firn(x,w,16);
#endif
#define CH 6 // Max number of channels
#define UP 128 /* Up sampling factor. Increasing this value will
improve frequency accuracy. Think about the L1
cashing of filter parameters - how big can it be? */
#include "fir.h"
#include "filter.h"
// local data
typedef struct pl_resample_s
{
int16_t* data; // Data buffer
int16_t* w; // Current filter weights
uint16_t dn; // Down sampling factor
uint16_t up; // Up sampling factor
int channels; // Number of channels
int len; // Lenght of buffer
int16_t ws[UP*L]; // List of all available filters
int16_t xs[CH][L*2]; // Circular buffers
} pl_resample_t;
static pl_resample_t pl_resample = {NULL,NULL,1,1,1,0,W};
// to set/get/query special features/parameters
static int control(int cmd,void *arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
if(pl_resample.data)
free(pl_resample.data);
pl_resample.len = ao_plugin_data.len;
pl_resample.data=(int16_t*)malloc(pl_resample.len);
if(!pl_resample.data)
return CONTROL_ERROR;
ao_plugin_data.len = (int)((double)ao_plugin_data.len *
((double)pl_resample.dn)/
((double)pl_resample.up));
return CONTROL_OK;
}
return -1;
}
// open & setup audio device
// return: 1=success 0=fail
static int init(){
int fin=ao_plugin_data.rate;
int fout=ao_plugin_cfg.pl_resample_fout;
pl_resample.w=pl_resample.ws;
pl_resample.up=UP;
// Sheck input format
if(ao_plugin_data.format != AFMT_S16_NE){
fprintf(stderr,"[pl_resample] Input audio format not yet suported. \n");
return 0;
}
// Sanity check and calculate down sampling factor
if((float)max(fin,fout)/(float)min(fin,fout) > 10){
fprintf(stderr,"[pl_resample] The difference between fin and fout is too large.\n");
return 0;
}
pl_resample.dn=(int)(0.5+((float)(fin*pl_resample.up))/((float)fout));
pl_resample.channels=ao_plugin_data.channels;
if(ao_plugin_data.channels>CH){
fprintf(stderr,"[pl_resample] Too many channels, max is 6.\n");
return 0;
}
// Tell the world what we are up to
printf("[pl_resample] Up=%i, Down=%i, True fout=%f\n",
pl_resample.up,pl_resample.dn,
((float)fin*pl_resample.up)/((float)pl_resample.dn));
// This plugin changes buffersize and adds some delay
ao_plugin_data.sz_mult/=((float)pl_resample.up)/((float)pl_resample.dn);
ao_plugin_data.delay_fix-= ((float)L/2) * (1/fout);
ao_plugin_data.rate=fout;
return 1;
}
// close plugin
static void uninit(){
if(pl_resample.data)
free(pl_resample.data);
pl_resample.data=NULL;
}
// empty buffers
static void reset(){
}
/* forward declarations */
int upsample();
int downsample();
// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
// FIXME: this routine needs to be optimized (it is probably possible to do a lot here)
static int play(){
if(pl_resample.up==pl_resample.dn){
register int16_t* in = ((int16_t*)ao_plugin_data.data);
register int16_t* end = in+ao_plugin_data.len/2;
while(in < end) *(in++)>>=1;
return 1;
}
if(pl_resample.up>pl_resample.dn)
return upsample();
// if(pl_resample.up<pl_resample.dn)
return downsample();
}
int upsample(){
static uint16_t pwi = 0; // Index for w
static uint16_t pxi = 0; // Index for circular queue
uint16_t ci = pl_resample.channels; // Index for channels
uint16_t nch = pl_resample.channels; // Number of channels
uint16_t len = 0; // Number of input samples
uint16_t inc = pl_resample.up/pl_resample.dn;
uint16_t level = pl_resample.up%pl_resample.dn;
uint16_t up = pl_resample.up;
uint16_t dn = pl_resample.dn;
register int16_t* w = pl_resample.w;
register uint16_t wi,xi; // Temporary indexes
// Index current channel
while(ci--){
// Temporary pointers
register int16_t* x = pl_resample.xs[ci];
register int16_t* in = ((int16_t*)ao_plugin_data.data)+ci;
register int16_t* out = pl_resample.data+ci;
int16_t* end = in+ao_plugin_data.len/2; // Block loop end
wi = pwi; xi = pxi;
while(in < end){
register uint16_t i = inc;
if(wi<level) i++;
xi=updateq(x,in,xi,L);
in+=nch;
while(i--){
// Run the FIR filter
FIR((&x[xi]),(&w[wi*L]),out);
len++; out+=nch;
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
}
}
}
// Save values that needs to be kept for next time
pwi = wi;
pxi = xi;
// Set new data
ao_plugin_data.len=len*2;
ao_plugin_data.data=pl_resample.data;
return 1;
}
int downsample(){
static uint16_t pwi = 0; // Index for w
static uint16_t pxi = 0; // Index for circular queue
static uint16_t pi = 1; // Number of new samples to put in x queue
uint16_t ci = pl_resample.channels; // Index for channels
uint16_t len = 0; // Number of input samples
uint16_t nch = pl_resample.channels; // Number of channels
uint16_t inc = pl_resample.dn/pl_resample.up;
uint16_t level = pl_resample.dn%pl_resample.up;
uint16_t up = pl_resample.up;
uint16_t dn = pl_resample.dn;
register uint16_t i,wi,xi; // Temporary indexes
// Index current channel
while(ci--){
// Temporary pointers
register int16_t* x = pl_resample.xs[ci];
register int16_t* in = ((int16_t*)ao_plugin_data.data)+ci;
register int16_t* out = pl_resample.data+ci;
// Block loop end
register int16_t* end = in+ao_plugin_data.len/2;
i = pi; wi = pwi; xi = pxi;
while(in < end){
xi=updateq(x,in,xi,L);
in+=nch;
if(!--i){
// Run the FIR filter
FIR((&x[xi]),(&pl_resample.w[wi*L]),out);
len++; out+=nch;
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
// Insert i number of new samples in queue
i = inc;
if(wi<level) i++;
}
}
}
// Save values that needs to be kept for next time
pwi = wi;
pxi = xi;
pi = i;
// Set new data
ao_plugin_data.len=len*2;
ao_plugin_data.data=pl_resample.data;
return 1;
}