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Code Issues Releases Wiki Activity

Sync problem when using fractional resampling fixed + speed increased.

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git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@4172 b3059339-0415-0410-9bf9-f77b7e298cf2
This commit is contained in:
anders 2002-01-15 13:21:45 +00:00
parent 47aa9aeb74
commit f36628b8fe
2 changed files with 93 additions and 44 deletions
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2
libao2/fir.h
View File

@ -111,7 +111,7 @@
#define SAVE_QUE(x)
#define UPDATE_QUE(inm) \
xi=(--xi)&(L-1); \
x[xi]=x[xi+L]=*inm;
x[xi]=x[xi+L]=*(inm);
#ifdef L4
#define FIR(x,w,y) \

135
libao2/pl_resample.c
View File

@ -24,7 +24,7 @@
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "afmt.h"
//#include "../config.h"
// #include "../config.h"
static ao_info_t info =
{
@ -46,15 +46,16 @@ LIBAO_PLUGIN_EXTERN(resample)
memory usage. For now the filterlenght is choosen to 4 and without
assembly optimization if no SSE is present.
*/
#ifdef HAVE_SSE
// #ifdef HAVE_SSE
#define L8 1 // Filter bank type
#define W W8 // Filter bank parameters
#define L 8 // Filter length
#else
#define L4 1
#define W W4
#define L 4
#endif
// #else
// #define L4 1
// #define W W4
// #define L 4
// #endif
#define CH 6 // Max number of channels
#define UP 128 /* Up sampling factor. Increasing this value will
@ -73,12 +74,11 @@ typedef struct pl_resample_s
uint16_t up; // Up sampling factor
int channels; // Number of channels
int len; // Lenght of buffer
int bypass; // Bypass this plugin
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,1,1,1,0,0,W};
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,int arg){
@ -117,11 +117,7 @@ static int init(){
return 0;
}
pl_resample.dn=(int)(0.5+((float)(fin*pl_resample.up))/((float)fout));
if(pl_resample.dn == pl_resample.up){
fprintf(stderr,"[pl_resample] Fin is too close to fout no conversion is needed.\n");
pl_resample.bypass=1;
return 1;
}
pl_resample.channels=ao_plugin_data.channels;
if(ao_plugin_data.channels>CH){
fprintf(stderr,"[pl_resample] Too many channels, max is 6.\n");
@ -155,12 +151,79 @@ static void reset(){
// 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=(*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;
LOAD_QUE(x);
while(in < end){
register uint16_t i = inc;
if(wi<level) i++;
UPDATE_QUE(in);
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_QUE(x);
}
// 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 output samples
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;
@ -169,8 +232,6 @@ static int play(){
register uint16_t i,wi,xi; // Temporary indexes
if(pl_resample.bypass)
return 1;
// Index current channel
while(ci--){
@ -183,33 +244,26 @@ static int play(){
i = pi; wi = pwi; xi = pxi;
LOAD_QUE(x);
if(0!=i) goto L1;
while(in < end){
// Update wi to point at the correct polyphase component
wi=(wi+dn)%up;
/* Update circular buffer x. This loop will be updated 0 or 1 time
for upsamling and inc or inc + 1 times for downsampling */
if(wi<level) goto L3;
if(0==i) goto L2;
L1: i--;
L3: UPDATE_QUE(in);
in+=nch;
if(in >= end) goto L2;
if(i) goto L1;
L2: if(i) goto L5;
i=inc;
UPDATE_QUE(in);
in+=nch;
if(!--i){
// Run the FIR filter
FIR((&x[xi]),(&pl_resample.w[wi*L]),out);
len++; out+=nch;
/* Get the correct polyphase component and the correct startpoint
in the circular bufer and 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++;
}
}
L5:
SAVE_QUE(x);
}
// Save values that needs to be kept for next time
pwi = wi;
pxi = xi;
@ -219,8 +273,3 @@ L5:
ao_plugin_data.data=pl_resample.data;
return 1;
}