[Mplayerxp-cvslog] SF.net SVN: mplayerxp:[301] mplayerxp
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[Mplayerxp-cvslog] SF.net SVN: mplayerxp:[301] mplayerxp
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From: <nic...@us...> - 2012-11-07 15:12:00
|
Revision: 301
http://mplayerxp.svn.sourceforge.net/mplayerxp/?rev=301&view=rev
Author: nickols_k
Date: 2012-11-07 15:11:47 +0000 (Wed, 07 Nov 2012)
Log Message:
-----------
bound arrays test: use libswresample for resampler
Modified Paths:
--------------
mplayerxp/Makefile
mplayerxp/configure
mplayerxp/mplayer.c
mplayerxp/postproc/af.c
mplayerxp/postproc/af_channels.c
mplayerxp/postproc/af_crystality.c
mplayerxp/postproc/af_echo3d.c
mplayerxp/postproc/af_format.c
mplayerxp/postproc/af_resample.c
Removed Paths:
-------------
mplayerxp/postproc/af_resample.h
Modified: mplayerxp/Makefile
===================================================================
--- mplayerxp/Makefile 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/Makefile 2012-11-07 15:11:47 UTC (rev 301)
@@ -30,6 +30,7 @@
FF_LIBS = ../ffmpeg/libavcodec/libavcodec$(FF_SUFFIX).a \
../ffmpeg/libswscale/libswscale$(FF_SUFFIX).a \
+ ../ffmpeg/libswresample/libswresample$(FF_SUFFIX).a \
../ffmpeg/libpostproc/libpostproc$(FF_SUFFIX).a \
../ffmpeg/libavformat/libavformat$(FF_SUFFIX).a \
../ffmpeg/libavutil/libavutil$(FF_SUFFIX).a
Modified: mplayerxp/configure
===================================================================
--- mplayerxp/configure 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/configure 2012-11-07 15:11:47 UTC (rev 301)
@@ -320,7 +320,7 @@
add_cflags "-I$srcdir/../ffmpeg"
print_config HAVE_ mp_config.h mp_config.mak ffmpeg
# Configuring external ffmpeg stuff
-ffmpeg_args="--enable-static --disable-shared --enable-gpl --enable-pthreads --disable-doc --disable-ffmpeg --disable-ffplay --disable-ffserver --disable-ffprobe --disable-avdevice --disable-avfilter --disable-avresample --disable-swresample"
+ffmpeg_args="--enable-static --disable-shared --enable-gpl --enable-pthreads --disable-doc --disable-ffmpeg --disable-ffplay --disable-ffserver --disable-ffprobe --disable-avdevice --disable-avfilter --disable-avresample"
if test -n $host ; then
_arch=$host_arch
x86_32 && _arch="i686"
@@ -332,7 +332,7 @@
test "$debug" != "0" && ffmpeg_args="$ffmpeg_args --enable-debug=$debug"
enabled $profile && ffmpeg_args="$ffmpeg_args --enable-profile"
if test -f "../ffmpeg/libavcodec/libavcodec$ff_libsuffix.a"; then
-echocheck "ffmpeg was already configured for $ff_libsuffix"
+echocheck "ffmpeg was already configured for $_arch"
else
echocheck "configuring ffmpeg stuff: --cc=\"$cc\" $ffmpeg_args"
cd ../ffmpeg
Modified: mplayerxp/mplayer.c
===================================================================
--- mplayerxp/mplayer.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/mplayer.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -1656,6 +1656,8 @@
mp_conf.malloc_debug=0;
for(i=0;i<argc;i++) if(strcmp(argv[i],"-core.malloc-debug")==0) { mp_conf.malloc_debug=1; break; }
mp_init_malloc(argv[0],1000,10,mp_conf.malloc_debug?MPA_FLG_BACKTRACE:MPA_FLG_RANDOMIZER);
+// mp_init_malloc(argv[0],1000,10,MPA_FLG_BOUNDS_CHECK);
+// mp_init_malloc(argv[0],1000,10,MPA_FLG_BEFORE_CHECK);
mpxp_init_structs();
priv_t*priv=mp_data->priv;
Modified: mplayerxp/postproc/af.c
===================================================================
--- mplayerxp/postproc/af.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -227,7 +227,7 @@
else
s->last=af->prev;
- // Uninitialize af and mp_free memory
+ // Uninitialize af and mp_free memory
af->uninit(af);
mp_free(af);
}
@@ -245,15 +245,14 @@
af_data_t in; // Format of the input to current filter
int rv=0; // Return value
- // Check if this is the first filter
- if(!af->prev)
- memcpy(&in,&(s->input),sizeof(af_data_t));
- else
- memcpy(&in,af->prev->data,sizeof(af_data_t));
+ // Check if this is the first filter
+ if(!af->prev) memcpy(&in,&(s->input),sizeof(af_data_t));
+ else memcpy(&in,af->prev->data,sizeof(af_data_t));
+
// Reset just in case...
in.audio=NULL;
in.len=0;
-
+
rv = af->control(af,AF_CONTROL_REINIT,&in);
switch(rv){
case CONTROL_OK:
@@ -264,58 +263,44 @@
af_instance_t* _new = NULL;
// Insert channels filter
if((af->prev?af->prev->data->nch:s->input.nch) != in.nch){
- // Create channels filter
- if(NULL == (_new = af_prepend(s,af,"channels")))
- return CONTROL_ERROR;
- // Set number of output channels
- if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_CHANNELS,&in.nch)))
- return rv;
+ if(NULL == (_new = af_prepend(s,af,"channels"))) return CONTROL_ERROR;
+ if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_CHANNELS,&in.nch))) return rv;
// Initialize channels filter
if(!_new->prev) memcpy(&in,&(s->input),sizeof(af_data_t));
else memcpy(&in,_new->prev->data,sizeof(af_data_t));
- if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_REINIT,&in)))
- return rv;
+ if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_REINIT,&in))) return rv;
}
// Insert rate filter
if((af->prev?af->prev->data->rate:s->input.rate) != in.rate){
- // Create channels filter
- if(NULL == (_new = af_prepend(s,af,"resample")))
- return CONTROL_ERROR;
- // Set number of output channels
- if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_RESAMPLE_RATE,&in.rate)))
- return rv;
+ if(NULL == (_new = af_prepend(s,af,"resample"))) return CONTROL_ERROR;
+ if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_RESAMPLE_RATE,&in.rate))) return rv;
// Initialize channels filter
if(!_new->prev) memcpy(&in,&(s->input),sizeof(af_data_t));
else memcpy(&in,_new->prev->data,sizeof(af_data_t));
if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_REINIT,&in))) {
+#if 0
af_instance_t* af2 = af_prepend(s,af,"format");
// Init the new filter
if(af2) {
if((CONTROL_OK != af2->control(af2,AF_CONTROL_FORMAT_BPS,&(in.bps)))
- || (CONTROL_OK != af2->control(af2,AF_CONTROL_FORMAT_FMT,&(in.format))))
- return -1;
- if(CONTROL_OK != af_reinit(s,af2))
- return -1;
+ || (CONTROL_OK != af2->control(af2,AF_CONTROL_FORMAT_FMT,&(in.format)))) return -1;
+ if(CONTROL_OK != af_reinit(s,af2)) return -1;
rv = _new->control(_new,AF_CONTROL_REINIT,&in);
}
+#endif
return rv;
}
}
// Insert format filter
- if(((af->prev?af->prev->data->format:s->input.format) != in.format) ||
+ if(((af->prev?af->prev->data->format:s->input.format) != in.format) ||
((af->prev?af->prev->data->bps:s->input.bps) != in.bps)){
- // Create format filter
- if(NULL == (_new = af_prepend(s,af,"format")))
- return CONTROL_ERROR;
- // Set output bits per sample
- if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_FORMAT_BPS,&in.bps)) ||
- CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_FORMAT_FMT,&in.format)))
- return rv;
+ if(NULL == (_new = af_prepend(s,af,"format"))) return CONTROL_ERROR;
+ if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_FORMAT_BPS,&in.bps)) ||
+ CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_FORMAT_FMT,&in.format))) return rv;
// Initialize format filter
if(!_new->prev) memcpy(&in,&(s->input),sizeof(af_data_t));
else memcpy(&in,_new->prev->data,sizeof(af_data_t));
- if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_REINIT,&in)))
- return rv;
+ if(CONTROL_OK != (rv = _new->control(_new,AF_CONTROL_REINIT,&in))) return rv;
}
if(!_new){ // Should _never_ happen
MSG_ERR("[libaf] Unable to correct audio format. "
Modified: mplayerxp/postproc/af_channels.c
===================================================================
--- mplayerxp/postproc/af_channels.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af_channels.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -249,12 +249,12 @@
static af_data_t* __FASTCALL__ play(struct af_instance_s* af, af_data_t* data,int final)
{
af_data_t* c = data; // Current working data
- af_data_t* l = af->data; // Local data
af_channels_t* s = af->setup;
int i;
if(CONTROL_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
+ af_data_t* l = af->data; // Local data
// Reset unused channels
memset(l->audio,0,(c->len*af->mul.n)/af->mul.d);
Modified: mplayerxp/postproc/af_crystality.c
===================================================================
--- mplayerxp/postproc/af_crystality.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af_crystality.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -331,7 +331,7 @@
int acount; // counter
float lval, rval; // value
float sal, sar; // sum
- float al, ar;
+ float al, ar;
float a1l, a1r;
};
Modified: mplayerxp/postproc/af_echo3d.c
===================================================================
--- mplayerxp/postproc/af_echo3d.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af_echo3d.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -166,7 +166,7 @@
af->data->format = AF_FORMAT_NE | AF_FORMAT_F;
af->data->bps = 4;
init_echo3d(s,af->data->rate);
-
+
return af_test_output(af,arg);
}
case AF_CONTROL_COMMAND_LINE:{
@@ -182,7 +182,7 @@
return CONTROL_UNKNOWN;
}
-// Deallocate memory
+// Deallocate memory
static void __FASTCALL__ uninit(struct af_instance_s* af)
{
if(af->data)
Modified: mplayerxp/postproc/af_format.c
===================================================================
--- mplayerxp/postproc/af_format.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af_format.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -280,11 +280,11 @@
// Helper functions to check sanity for input arguments
// Sanity check for bytes per sample
-static int __FASTCALL__ check_bps(int bps)
+static int __FASTCALL__ check_bps(const char *pfx,int bps)
{
if(bps != 4 && bps != 3 && bps != 2 && bps != 1){
- MSG_ERR("[format] The number of bytes per sample"
- " must be 1, 2, 3 or 4. Current value is %i \n",bps);
+ MSG_ERR("[%s-format] The number of bytes per sample"
+ " must be 1, 2, 3 or 4. Current value is %i \n",pfx,bps);
return CONTROL_ERROR;
}
return CONTROL_OK;
@@ -311,15 +311,15 @@
char buf1[256],buf2[256];
switch(cmd){
case AF_CONTROL_REINIT:{
- // Make sure this filter isn't redundant
+ // Make sure this filter isn't redundant
if(af->data->format == ((af_data_t*)arg)->format &&
af->data->bps == ((af_data_t*)arg)->bps)
return CONTROL_DETACH;
// Check for errors in configuraton
- if((CONTROL_OK != check_bps(((af_data_t*)arg)->bps)) ||
+ if((CONTROL_OK != check_bps("new",((af_data_t*)arg)->bps)) ||
(CONTROL_OK != check_format(((af_data_t*)arg)->format)) ||
- (CONTROL_OK != check_bps(af->data->bps)) ||
+ (CONTROL_OK != check_bps("prev",af->data->bps)) ||
(CONTROL_OK != check_format(af->data->format)))
return CONTROL_ERROR;
@@ -341,7 +341,7 @@
int format = AF_FORMAT_NE;
// Convert string to format
format = str2fmt((char *)arg,&bps);
-
+
if((CONTROL_OK != af->control(af,AF_CONTROL_FORMAT_BPS | AF_CONTROL_SET,&bps)) ||
(CONTROL_OK != af->control(af,AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET,&format)))
return CONTROL_ERROR;
@@ -349,9 +349,9 @@
}
case AF_CONTROL_FORMAT_BPS | AF_CONTROL_SET:
// Reinit must be called after this function has been called
-
+
// Check for errors in configuraton
- if(CONTROL_OK != check_bps(*(int*)arg))
+ if(CONTROL_OK != check_bps("arg",*(int*)arg))
return CONTROL_ERROR;
af->data->bps = *(int*)arg;
@@ -370,7 +370,7 @@
return CONTROL_UNKNOWN;
}
-// Deallocate memory
+// Deallocate memory
static void __FASTCALL__ uninit(struct af_instance_s* af)
{
if(af->data)
@@ -382,12 +382,12 @@
// Filter data through filter
static af_data_t* __FASTCALL__ play(struct af_instance_s* af, af_data_t* data,int final)
{
- af_data_t* l = af->data; // Local data
af_data_t* c = data; // Current working data
int len = c->len/c->bps; // Lenght in samples of current audio block
if(CONTROL_OK != RESIZE_LOCAL_BUFFER(af,data))
return NULL;
+ af_data_t* l = af->data; // Local data
// Change to cpu native endian format
if((c->format&AF_FORMAT_END_MASK)!=AF_FORMAT_NE)
@@ -429,7 +429,7 @@
break;
default:
// Input must be int
-
+
// Change signed/unsigned
if((c->format&AF_FORMAT_SIGN_MASK) != (l->format&AF_FORMAT_SIGN_MASK)){
if((c->format&AF_FORMAT_SIGN_MASK) == AF_FORMAT_US)
@@ -593,5 +593,5 @@
for(i=0;i<len;i++)
((int32_t*)out)[i]=(int32_t)(INT_MIN+((uint32_t*)in)[i]);
break;
- }
+ }
}
Modified: mplayerxp/postproc/af_resample.c
===================================================================
--- mplayerxp/postproc/af_resample.c 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af_resample.c 2012-11-07 15:11:47 UTC (rev 301)
@@ -11,433 +11,189 @@
/* This audio filter changes the sample rate. */
#include <stdio.h>
#include <stdlib.h>
+#include <signal.h>
#include <unistd.h>
#include <inttypes.h>
+#include "libavutil/audioconvert.h"
+#include "libswresample/swresample.h"
+
#include "af.h"
#include "dsp.h"
#include "osdep/mplib.h"
#include "pp_msg.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 filterlenght is choosen to 8 if the machine is
- slow and to 16 if the machine is fast and has MMX.
-*/
+uint64_t layouts[]={
+ 0,
+ AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_4POINT0,
+ AV_CH_LAYOUT_5POINT0,
+ AV_CH_LAYOUT_5POINT1,
+ AV_CH_LAYOUT_7POINT0,
+ AV_CH_LAYOUT_7POINT1
+};
-#if __CPU__ < 686 // This machine is slow
-#define L8
-#else
-#define L16
-#endif
-
-#include "af_resample.h"
-
-// 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
-{
- any_t* w; // Current filter weights
- any_t** 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
- int ifreq;
-} af_resample_t;
-
-// Euclids algorithm for calculating Greatest Common Divisor GCD(a,b)
-// Extended for negative and 0 values. If both are 0 the result is 1.
-// The sign of the result will be so that it has the same sign as b.
-static inline int gcd(register int a, register int b)
-{
- int b_org = b;
- while (b != 0) {
- a %= b;
- if (a == 0)
- break;
- b %= a;
- }
- // the result is either in a or b. As the other one is 0 just add them.
- a += b;
- if (!a)
- return 1;
- if (a * b_org < 0)
- return 1;//-a;
- return a;
+static uint64_t get_ch_layout(unsigned nch) {
+ if(nch < sizeof(layouts)/sizeof(uint64_t))
+ return layouts[nch];
+ return 0;
}
-/**
- * \brief cancel down a fraction f
- * \param f fraction to cancel down
- * \ingroup af_filter
- */
-static void frac_cancel(frac_t *f) {
- int _gcd = gcd(f->n, f->d);
- f->n /= _gcd;
- f->d /= _gcd;
-}
-
-// Fast linear interpolation resample with modest audio quality
-static int __FASTCALL__ linint(af_data_t* c,af_data_t* 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;
+static enum AVSampleFormat get_sample_format(unsigned fmt,unsigned nbps) {
+ switch(nbps) {
+ case 1: if((fmt&AF_FORMAT_SIGN_MASK) == AF_FORMAT_US) return AV_SAMPLE_FMT_U8;
+ break;
+ case 2: if((fmt&AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI &&
+ (fmt&AF_FORMAT_POINT_MASK) == AF_FORMAT_I &&
+ (fmt&AF_FORMAT_END_MASK) == AF_FORMAT_NE) return AV_SAMPLE_FMT_S16;
+ break;
+ case 4: if((fmt&AF_FORMAT_POINT_MASK) == AF_FORMAT_I) {
+ if((fmt&AF_FORMAT_SIGN_MASK) == AF_FORMAT_SI &&
+ (fmt&AF_FORMAT_POINT_MASK) == AF_FORMAT_I &&
+ (fmt&AF_FORMAT_END_MASK) == AF_FORMAT_NE) return AV_SAMPLE_FMT_S32;
+ }
+ else return AV_SAMPLE_FMT_FLT;
+ break;
+ case 8: if((fmt&AF_FORMAT_POINT_MASK) == AF_FORMAT_F) return AV_SAMPLE_FMT_DBL;
+ break;
}
- 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;
+ return AV_SAMPLE_FMT_NONE;
}
-/* Determine resampling type and format */
-static int __FASTCALL__ set_types(struct af_instance_s* af, af_data_t* data)
-{
- af_resample_t* s = af->setup;
- int rv = CONTROL_OK;
- float rd = 0;
+// local data
+typedef struct af_resample_s {
+ struct SwrContext* ctx;
+ unsigned irate;
+ unsigned inch;
+ unsigned ifmt;
+ unsigned ibps;
+} af_resample_t;
- // Make sure this filter isn't redundant
- if((af->data->rate == data->rate) || (af->data->rate == 0))
- return CONTROL_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_NE | AF_FORMAT_F))) ||
- ((s->setup & RSMP_MASK) == RSMP_LIN)){
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
- af->data->format = AF_FORMAT_NE | AF_FORMAT_SI;
- af->data->bps = 2;
- MSG_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_NE | AF_FORMAT_F)) ||
- ((s->setup & RSMP_MASK) == RSMP_FLOAT)){
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
- af->data->format = AF_FORMAT_NE | AF_FORMAT_F;
- af->data->bps = 4;
- }
- else{
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
- af->data->format = AF_FORMAT_NE | AF_FORMAT_SI;
- af->data->bps = 2;
- }
- MSG_V("[resample] Using %s processing and %s frequency 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) {
- char buff[256];
- MSG_V("[resample] doesn't fork with '%s' input format\n",fmt2str(data->format,data->bps,buff,sizeof(buff)));
- rv = CONTROL_FALSE;
- }
- data->format = af->data->format;
- data->bps = af->data->bps;
- af->data->nch = data->nch;
- return rv;
-}
-
// Initialization and runtime control
static ControlCodes __FASTCALL__ control(struct af_instance_s* af, int cmd, any_t* arg)
{
- af_resample_t* s = (af_resample_t*)af->setup;
- switch(cmd){
- case AF_CONTROL_REINIT:{
- af_data_t* n = (af_data_t*)arg; // New configuration
- int i,d = 0;
- int rv = CONTROL_OK;
+ af_resample_t* s = (af_resample_t*)af->setup;
+ switch(cmd){
+ case AF_CONTROL_REINIT: {
+ enum AVSampleFormat avfmt;
+ uint64_t nch;
+ af_data_t* n = (af_data_t*)arg; // New configuration
+ int rv = CONTROL_OK;
- // Free space for circular bufers
- s->ifreq=n->rate;
- if(s->xq){
- for(i=1;i<af->data->nch;i++)
- if(s->xq[i])
- mp_free(s->xq[i]);
- mp_free(s->xq);
- }
+ if(s->ctx) { swr_free(&s->ctx); s->ctx=NULL; }
+ // Make sure this filter isn't redundant
+ if((af->data->rate == n->rate) || (af->data->rate == 0)) {
+ MSG_V("[af_resample] detach due: %i -> %i Hz\n",
+ af->data->rate,n->rate);
+ return CONTROL_DETACH;
+ }
+ avfmt=get_sample_format(n->format,n->bps);
+ nch=get_ch_layout(n->nch);
+ if(avfmt==AV_SAMPLE_FMT_NONE) rv=CONTROL_ERROR;
+ if(nch==0) rv=CONTROL_ERROR;
+ if(rv!=CONTROL_OK) {
+ char buff[256];
+ MSG_V("[af_resample] doesn't work with '%s' x %i\n"
+ ,fmt2str(n->format,n->bps,buff,sizeof(buff))
+ ,n->nch);
+ }
+ s->ctx = swr_alloc_set_opts(NULL,
+ nch, avfmt,af->data->rate,
+ nch, avfmt,n->rate,
+ 0, NULL);
+ if(swr_init(s->ctx)<0) {
+ MSG_ERR("[af_resample] Cannot init swr_init\n");
+ rv=CONTROL_ERROR;
+ }
- if(CONTROL_DETACH == (rv = set_types(af,n)))
- return CONTROL_DETACH;
+ af->data->format = n->format;
+ af->data->bps = n->bps;
+ af->data->nch = n->nch;
- // 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;
- MSG_DBG2("[resample] Linear interpolation step: 0x%016X.\n",
- s->step);
- af->mul.n = af->data->rate;
- af->mul.d = n->rate;
- frac_cancel(&af->mul);
- return rv;
- }
-
- // Calculate up and down sampling factors
- d=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) > 50){
- d=gcd(up,dn);
- up/=2; dn/=2; m*=2;
- }
- d*=m;
- }
-
- // Create space for circular bufers
- s->xq = mp_malloc(n->nch*sizeof(any_t*));
- for(i=0;i<n->nch;i++)
- s->xq[i] = mp_malloc(2*L*af->data->bps);
- s->xi = 0;
-
- // Check if the 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;
- if(af->data->rate/d && n->rate/d)
- {
- s->up = af->data->rate/d;
- s->dn = n->rate/d;
- }
- else if(af->data->rate/d)
- {
- s->up=(af->data->rate/d)*(d/n->rate);
- s->dn=1;
- }
- else
- {
- s->up=1;
- s->dn=n->rate/d*(d/af->data->rate);
- }
-
- // Calculate cuttof frequency for filter
- fc = 1/(float)(max(s->up,s->dn));
- // Allocate space for polyphase filter bank and protptype filter
- w = mp_malloc(sizeof(float) * s->up *L);
- if(NULL != s->w)
- mp_free(s->w);
- s->w = mp_malloc(L*s->up*af->data->bps);
-
- // Design prototype filter type using Kaiser window with beta = 10
- if(NULL == w || NULL == s->w ||
- -1 == design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
- MSG_ERR("[resample] Unable to design prototype filter.\n");
- return CONTROL_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++;
+ s->irate=n->rate;
+ s->inch=n->nch;
+ s->ifmt=n->format;
+ s->ibps=n->bps;
+ // Set multiplier and delay
+ af->delay = (double)swr_get_delay(s->ctx,1000);
+ af->mul.n = af->data->rate;
+ af->mul.d = n->rate;
+ return rv;
}
- }
- mp_free(w);
+ case AF_CONTROL_SHOWCONF:
+ MSG_INFO("[af_resample] New filter designed (%i -> %i Hz)\n", s->irate,af->data->rate);
+ return CONTROL_OK;
+ case AF_CONTROL_COMMAND_LINE:{
+ int rate=0;
+ sscanf((char*)arg,"%i", &rate);
+ return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
+ }
+ case AF_CONTROL_POST_CREATE: return CONTROL_OK;
+ case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET: {
+ af->data->rate = ((int*)arg)[0];
+ return CONTROL_OK;
+ }
+ default: break;
}
-
- // Set multiplier and delay
- af->delay = (double)(1000*L/2)/((double)n->rate);
- af->mul.n = s->up;
- af->mul.d = s->dn;
- return rv;
- }
- case AF_CONTROL_SHOWCONF:
- MSG_INFO("[af_resample] New filter designed up: %i down: %i (%i -> %i Hz)\n", s->up, s->dn,s->ifreq,af->data->rate);
- return CONTROL_OK;
- case AF_CONTROL_COMMAND_LINE:{
- af_resample_t* s = (af_resample_t*)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((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
- ((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
- return CONTROL_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] < 4000 || ((int*)arg)[0] > 192000){
- MSG_ERR("[resample] The output sample frequency "
- "must be between 4kHz and 192kHz. Current value is %i \n",
- ((int*)arg)[0]);
- return CONTROL_ERROR;
- }
-
- af->data->rate=((int*)arg)[0];
- MSG_V("[resample] Changing sample rate "
- "to %iHz\n",af->data->rate);
- return CONTROL_OK;
- default: break;
- }
- return CONTROL_UNKNOWN;
+ return CONTROL_UNKNOWN;
}
// Deallocate memory
static void __FASTCALL__ uninit(struct af_instance_s* af)
{
- if(af->data)
- mp_free(af->data);
+ af_resample_t* s = (af_resample_t*)af->setup;
+ if(af->data) {
+ if(af->data->audio) mp_free(af->data->audio);
+ mp_free(af->data);
+ }
+ if(s->ctx) swr_free(&s->ctx);
+ s->ctx=NULL;
}
// Filter data through filter
static af_data_t* __FASTCALL__ play(struct af_instance_s* af, af_data_t* data,int final)
{
- int len = 0; // Length of output data
- af_data_t* c = data; // Current working data
- af_data_t* l = af->data; // Local data
- af_resample_t* s = (af_resample_t*)af->setup;
- uint32_t ci = l->nch; // Index for channels
- uint32_t nch = l->nch; // Number of channels
- uint32_t up = s->up;
- uint32_t dn = s->dn;
- uint32_t ns = c->len/l->bps;
- register int32_t i = 0;
- register uint32_t wi = 0;
- register uint32_t xi = 0;
+ int rc;
+ af_data_t* c = data; // Current working data
+ af_resample_t* s = (af_resample_t*)af->setup;
- if(CONTROL_OK != RESIZE_LOCAL_BUFFER(af,data))
- return NULL;
+ if (CONTROL_OK != RESIZE_LOCAL_BUFFER(af, data)) return NULL;
+ af_data_t* l = af->data; // Local data
+ uint8_t* ain[SWR_CH_MAX];
+ const uint8_t* aout[SWR_CH_MAX];
- // Run resampling
- switch(s->setup & RSMP_MASK){
- case(RSMP_INT):
-# define FORMAT_I 1
- if(s->up>s->dn){
-# define UP
-# include "af_resample.h"
-# undef UP
- }
- else{
-# define DN
-# include "af_resample.h"
-# undef DN
- }
- break;
- case(RSMP_FLOAT):
-# undef FORMAT_I
-# define FORMAT_F 1
- if(s->up>s->dn){
-# define UP
-# include "af_resample.h"
-# undef UP
- }
- else{
-# define DN
-# include "af_resample.h"
-# undef DN
- }
- break;
- case(RSMP_LIN):
- len = linint(c, l, s);
- break;
- }
+ aout[0]=l->audio;
+ ain[0]=c->audio;
- // Save values that needs to be kept for next time
- s->wi = wi;
- s->xi = xi;
- // Set output data
- c->audio = l->audio;
- c->len = len*l->bps;
- c->rate = l->rate;
+ rc=swr_convert(s->ctx,aout,l->len/(l->nch*l->bps),ain,c->len/(c->nch*c->bps));
+ if(rc<0) MSG_ERR("%i=swr_convert\n",rc);
+ else l->len=rc*l->nch*l->bps;
- return c;
+ return l;
}
// Allocate memory and set function pointers
static ControlCodes __FASTCALL__ open(af_instance_t* af){
- af->control=control;
- af->uninit=uninit;
- af->play=play;
- af->mul.n=1;
- af->mul.d=1;
- af->data=mp_calloc(1,sizeof(af_data_t));
- af->setup=mp_calloc(1,sizeof(af_resample_t));
- if(af->data == NULL || af->setup == NULL)
- return CONTROL_ERROR;
- ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
- return CONTROL_OK;
+ af->control=control;
+ af->uninit=uninit;
+ af->play=play;
+ af->mul.n=1;
+ af->mul.d=1;
+ af->data=mp_calloc(1,sizeof(af_data_t));
+ af->setup=mp_calloc(1,sizeof(af_resample_t));
+ if(af->data == NULL || af->setup == NULL) return CONTROL_ERROR;
+ return CONTROL_OK;
}
// Description of this plugin
const af_info_t af_info_resample = {
- "Sample frequency conversion",
- "resample",
- "Anders",
- "",
- AF_FLAGS_REENTRANT,
- open
+ "Sample frequency conversion",
+ "resample",
+ "Anders",
+ "",
+ AF_FLAGS_REENTRANT,
+ open
};
Deleted: mplayerxp/postproc/af_resample.h
===================================================================
--- mplayerxp/postproc/af_resample.h 2012-11-07 15:09:02 UTC (rev 300)
+++ mplayerxp/postproc/af_resample.h 2012-11-07 15:11:47 UTC (rev 301)
@@ -1,144 +0,0 @@
-/*=============================================================================
-//
-// This software has been released under the terms of the GNU General Public
-// license. See http://www.gnu.org/copyleft/gpl.html for details.
-//
-// Copyright 2002 Anders Johansson aj...@at...
-//
-//=============================================================================
-*/
-
-/* This file contains the resampling engine, the sample format is
- controlled by the FORMAT parameter, the filter length by the L
- parameter and the resampling type by UP and DN. This file should
- only be included by af_resample.c
-*/
-
-#undef L
-#undef SHIFT
-#undef FORMAT
-#undef FIR
-#undef ADDQUE
-
-/* The lenght Lxx definition selects the length of each poly phase
- component. Valid definitions are L8 and L16 where the number
- defines the nuber of taps. This definition affects the
- computational complexity, the performance and the memory usage.
-*/
-
-/* The FORMAT_x parameter selects the sample format type currently
- float and int16 are supported. Thes two formats are selected by
- defining eiter FORMAT_F or FORMAT_I. The advantage of using float
- is that the amplitude and therefore the SNR isn't affected by the
- filtering, the disadvantage is that it is a lot slower.
-*/
-
-#if defined(FORMAT_I)
-#define SHIFT >>16
-#define FORMAT int16_t
-#if defined( ARCH_X86 ) || defined(ARCH_X86_64)
-#define FIR(x,w,y) {y[0]=FIR_i16(x,w);}
-#endif
-#else
-#define SHIFT
-#define FORMAT float
-#if defined( ARCH_X86 ) || defined(ARCH_X86_64)
-#define FIR(x,w,y) {y[0]=FIR_f32(x,w);}
-#endif
-#endif
-
-// Short filter
-#if defined(L8)
-
-#define L 8 // Filter length
-// Unrolled loop to speed up execution
-#if !(defined( ARCH_X86 ) || defined(ARCH_X86_64))
-#define FIR(x,w,y) \
- (y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
- + w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT
-#endif
-
-
-#else /* L8/L16 */
-
-#define L 16
-// Unrolled loop to speed up execution
-#if !(defined( ARCH_X86 ) || defined(ARCH_X86_64))
-#define FIR(x,w,y) \
- y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
- + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
- + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
- + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) SHIFT
-#endif
-#endif /* L8/L16 */
-
-// Macro to add data to circular que
-#define ADDQUE(xi,xq,in)\
- xq[xi]=xq[(xi)+L]=*(in);\
- xi=((xi)-1)&(L-1);
-
-#if defined(UP)
-
- uint32_t inc = s->up/s->dn;
- uint32_t level = s->up%s->dn;
- register FORMAT* w = s->w;
-
- // Index current channel
- while(ci--){
- // Temporary pointers
- register FORMAT* x = s->xq[ci];
- register FORMAT* in = ((FORMAT*)c->audio)+ci;
- register FORMAT* out = ((FORMAT*)l->audio)+ci;
- FORMAT* end = in+ns; // Block loop end
- wi = s->wi; xi = s->xi;
-
- while(in < end){
- register uint32_t i = inc;
- if(wi<level) i++;
-
- ADDQUE(xi,x,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;
- }
- }
-
- }
-#endif /* UP */
-
-#if defined(DN) /* DN */
- uint32_t inc = s->dn/s->up;
- uint32_t level = s->dn%s->up;
- FORMAT* w = s->w;
-
- // Index current channel
- while(ci--){
- // Temporary pointers
- register FORMAT* x = s->xq[ci];
- register FORMAT* in = ((FORMAT*)c->audio)+ci;
- register FORMAT* out = ((FORMAT*)l->audio)+ci;
- register FORMAT* end = in+ns; // Block loop end
- i = s->i; wi = s->wi; xi = s->xi;
-
- while(in < end){
-
- ADDQUE(xi,x,in);
- in+=nch;
- if((--i)<=0){
- // Run the FIR filter
- if(out<l->audio+l->len-L) FIR((&x[xi]),(&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++;
- }
- }
- }
- s->i = i;
-#endif /* DN */
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