[Mplayerxp-cvslog] CVS: mplayerxp/mplayerxp/libao2 ao_accel.c,NONE,1.1 ao_accel_template.c,NONE,1.1
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[Mplayerxp-cvslog] CVS: mplayerxp/mplayerxp/libao2 ao_accel.c,NONE,1.1 ao_accel_template.c,NONE,1.1 pl_lp.c,NONE,1.1 ao_alsa9.c,1.8,1.9 ao_null.c,1.3,1.4 ao_oss.c,1.7,1.8 ao_plugin.c,1.5,1.6 ao_sdl.c,1.7,1.8 audio_out.c,1.4,1.5 audio_plugin.h,1.2,1.3 audio_plugin_internal.h,1.2,1.3 fir.h,1.1.1.1,1.2 Makefile,1.4,1.5 pl_delay.c,1.4,1.5 pl_eq.c,1.3,1.4 pl_extrastereo.c,1.4,1.5 pl_format.c,1.5,1.6 pl_resample.c,1.3,1.4 pl_surround.c,1.6,1.7 ao_mpegpes.c,1.3,NONE ao_pcm.c,1.5,NONE
Update of /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2
In directory sc8-pr-cvs1:/tmp/cvs-serv22640
Modified Files:
ao_alsa9.c ao_null.c ao_oss.c ao_plugin.c ao_sdl.c audio_out.c
audio_plugin.h audio_plugin_internal.h fir.h Makefile
pl_delay.c pl_eq.c pl_extrastereo.c pl_format.c pl_resample.c
pl_surround.c
Added Files:
ao_accel.c ao_accel_template.c pl_lp.c
Removed Files:
ao_mpegpes.c ao_pcm.c
Log Message:
sync with mphq and lot of my improvements include new audio plugin
--- NEW FILE: ao_accel.c ---
#include <inttypes.h>
#include "../config.h"
#include "../cpudetect.h"
#include "fir.h"
#ifdef ARCH_X86
#define CAN_COMPILE_X86_ASM
#endif
#undef HAVE_MMX
#undef HAVE_MMX2
#undef HAVE_3DNOW
#undef HAVE_SSE
#define RENAME(a) a ## _c
#include "ao_accel_template.c"
#ifdef CAN_COMPILE_X86_ASM
//MMX versions
#undef RENAME
#define HAVE_MMX
#undef HAVE_MMX2
#undef HAVE_3DNOW
#define RENAME(a) a ## _MMX
#include "ao_accel_template.c"
#endif
static int32_t init_firn(int16_t* x, int16_t* w, int n)
{
#ifdef CAN_COMPILE_X86_ASM
if(gCpuCaps.hasMMX) firn = firn_MMX;
else
#endif //CAN_COMPILE_X86_ASM
firn = firn_c;
return (*firn)(x,w,n);
}
int32_t (*firn)(int16_t* x, int16_t* w, int n)=init_firn;
--- NEW FILE: ao_accel_template.c ---
static int32_t RENAME(firn)(int16_t* x, int16_t* w, int n)
{
register int32_t y=0;
register int i,nn=(n/2)*2;
#ifdef HAVE_MMX
/* Prologue */
asm volatile(
"pxor %%mm4, %%mm4\n"
"pxor %%mm5, %%mm5"
:::"memory"); /* Clear buffer yt */
/* Main loop (Note: the case n==8 is most often (for resampling stereo 48KHz to 44.1KHz for example)*/
for(i=0;i<nn;i+=8){
asm volatile(
" movq (%1), %%mm0\n" /* Load x(n:n+4) */
" movq 8(%1), %%mm1\n"
" pmaddwd (%0), %%mm0\n" /* yt(n:n+1)=sum(x(n:n+4).*w(n:n+4)) */
" pmaddwd 8(%0), %%mm1\n"
" psrld $16, %%mm0\n" /* yt(n:n+1)=yt(n:n+1)>>16 */
" psrld $16, %%mm1\n"
" paddd %%mm0, %%mm4\n" /* yt(n:n+1)=yt(n-2:n-1)+yt(n:n+1) */
" paddd %%mm1, %%mm5\n"
:: "g" (&w[i]), "g" (&x[i]));
}
/* Tail loop */
for(;i<n;i+=4){
asm volatile(
" movq (%1), %%mm0\n" /* Load x(n:n+4) */
" pmaddwd (%0), %%mm0\n" /* yt(n:n+1)=sum(x(n:n+4).*w(n:n+4)) */
" psrld $16, %%mm0\n" /* yt(n:n+1)=yt(n:n+1)>>16 */
" paddd %%mm0, %%mm4\n" /* yt(n:n+1)=yt(n-2:n-1)+yt(n:n+1) */
:: "g" (&w[i]), "g" (&x[i]));
}
/* Epilogue */
asm volatile(
" movq %%mm4, %%mm0\n"
" movq %%mm5, %%mm1\n"
" punpckhdq %%mm4, %%mm0\n"
" punpckhdq %%mm5, %%mm1\n"
" paddd %%mm0, %%mm4\n" /* yt(n)=yt(n)+yt(n+1) */
" paddd %%mm1, %%mm5\n"
" paddd %%mm5, %%mm4\n"
" movd %%mm4, %0\n" /* y=yt */
" emms\n"
: "=&r" (y));
#else /* HAVE_MMX */
for(i=0;i<nn;i+=8)
y+=(w[i+0]*x[i+0]+w[i+1]*x[i+1]+w[i+2]*x[i+2]+w[i+3]*x[i+3])>>16+
(w[i+4]*x[i+4]+w[i+5]*x[i+5]+w[i+6]*x[i+6]+w[i+7]*x[i+7])>>16;
for(;i<n;i+=4)
y+=(w[i]*x[i]+w[i+1]*x[i+1]+w[i+2]*x[i+2]+w[i+3]*x[i+3])>>16;
#endif /* HAVE_MMX */
return y;
}
--- NEW FILE: pl_lp.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 2003 Nick Kurshev
//
//=============================================================================
*/
/* LongPlay - This audio output plugin fakes changing of sample rate.
The output should be played longer than original sound track so
it may help to avoid framedropping by longer movie playback
*/
#define PLUGIN
#include "../config.h"
#include "audio_out.h"
#include "audio_plugin.h"
#include "audio_plugin_internal.h"
#include "libmpdemux/stream.h"
#include "libmpdemux/demuxer.h"
#include "../libmpdemux/stheader.h"
#include "ao_msg.h"
static ao_info_t info =
{
"LongPlay audio plugin",
"lp",
"Nick Kurshev",
""
};
LIBAO_PLUGIN_EXTERN(lp)
static int control(int cmd,long arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
static int init(){
int fin=ao_plugin_data.rate;
int fout=ao_plugin_cfg.pl_resample_fout;
/* Tell the world what we are up to */
MSG_INFO("[pl_lp] In=%i, Faked Out=%i\n",fin,fout);
ao_plugin_data.rate=fout;
ao_plugin_data.delay_mult*=(float)fout/(float)fin;
return 1;
}
static void uninit(){}
static void reset(){}
static int play(){
/* Do nothing here */
return 1;
}
Index: ao_alsa9.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/ao_alsa9.c,v
retrieving revision 1.8
retrieving revision 1.9
diff -p -u -d -r1.8 -r1.9
--- ao_alsa9.c 11 Feb 2003 15:14:24 -0000 1.8
+++ ao_alsa9.c 29 Apr 2003 17:25:14 -0000 1.9
@@ -54,6 +54,8 @@ static int alsa_fragsize = 4096;
static int alsa_fragcount = 16;
static int chunk_size = 1024; //is alsa_fragsize / 4
+#define MIN_CHUNK_SIZE 1024
+
static size_t bits_per_sample, bytes_per_sample, bits_per_frame;
static size_t chunk_bytes;
@@ -416,7 +418,7 @@ static int init(int rate_hz, int channel
" mmap-mode and the device-name as first card third device\n");
return(0);
} else {
- printf("alsa-init: soundcard set to %s\n", alsa_device);
+ MSG_INFO("alsa-init: soundcard set to %s\n", alsa_device);
}
//setting modes for block or nonblock-mode
@@ -725,23 +727,17 @@ static void reset()
{
int err;
- if (!ao_noblock) {
- //drain causes error in nonblock-mode!
- if ((err = snd_pcm_drain(alsa_handler)) < 0)
- {
- MSG_ERR("alsa-pause: pcm drain error: %s\n", snd_strerror(err));
- return;
- }
-
- if ((err = snd_pcm_prepare(alsa_handler)) < 0)
- {
- MSG_ERR("alsa-reset: pcm prepare error: %s\n", snd_strerror(err));
- return;
- }
- } else {
- MSG_V("alsa-reset: reset nonblocked");
- return;
+ if ((err = snd_pcm_drop(alsa_handler)) < 0)
+ {
+ MSG_ERR("alsa-reset: pcm drop error: %s\n", snd_strerror(err));
+ return;
}
+ if ((err = snd_pcm_prepare(alsa_handler)) < 0)
+ {
+ MSG_ERR("alsa-reset: pcm prepare error: %s\n", snd_strerror(err));
+ return;
+ }
+ return;
}
#ifdef USE_POLL
@@ -1060,6 +1056,10 @@ static int get_space()
ret = 0;
}
+ // workaround for too small value returned
+ if (ret < MIN_CHUNK_SIZE)
+ ret = 0;
+
return(ret);
}
Index: ao_null.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/ao_null.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -p -u -d -r1.3 -r1.4
--- ao_null.c 30 Jun 2002 21:21:07 -0000 1.3
+++ ao_null.c 29 Apr 2003 17:25:14 -0000 1.4
@@ -19,7 +19,7 @@ LIBAO_EXTERN(null)
struct timeval last_tv;
int buffer;
-static int drain(){
+static void drain(){
struct timeval now_tv;
int temp, temp2;
Index: ao_oss.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/ao_oss.c,v
retrieving revision 1.7
retrieving revision 1.8
diff -p -u -d -r1.7 -r1.8
--- ao_oss.c 30 Jun 2002 21:21:07 -0000 1.7
+++ ao_oss.c 29 Apr 2003 17:25:14 -0000 1.8
@@ -18,8 +18,6 @@
#include "audio_out.h"
#include "audio_out_internal.h"
-extern int verbose;
-
static ao_info_t info =
{
"OSS/ioctl audio output",
@@ -50,7 +48,7 @@ static int control(int cmd,long arg){
case AOCONTROL_SET_VOLUME:
{
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
- int fd, v, mcmd, devs;
+ int fd, v, devs;
if(ao_data.format == AFMT_AC3)
return CONTROL_TRUE;
@@ -90,8 +88,7 @@ static int control(int cmd,long arg){
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
- if (verbose)
- MSG_INFO("ao2: %d Hz %d chans %s\n",rate,channels,
+ MSG_V("ao2: %d Hz %d chans %s\n",rate,channels,
audio_out_format_name(format));
if (ao_subdevice)
@@ -120,9 +117,9 @@ static int init(int rate,int channels,in
}
#endif
- ao_data.bps=channels;
- if(format != AFMT_U8 && format != AFMT_S8)
- ao_data.bps*=2;
+#if defined(FD_CLOEXEC) && defined(F_SETFD)
+ fcntl(audio_fd, F_SETFD, FD_CLOEXEC);
+#endif
if(format == AFMT_AC3) {
ao_data.samplerate=rate;
@@ -133,7 +130,7 @@ ac3_retry:
ao_data.format=format;
if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format)<0 ||
ao_data.format != format) if(format == AFMT_AC3){
- MSG_ERR("Can't set audio device %s to AC3 output, trying S16...\n", dsp);
+ MSG_WARN("Can't set audio device %s to AC3 output, trying S16...\n", dsp);
#ifdef WORDS_BIGENDIAN
format=AFMT_S16_BE;
#else
@@ -141,36 +138,37 @@ ac3_retry:
#endif
goto ac3_retry;
}
- MSG_INFO("audio_setup: sample format: %s (requested: %s)\n",
+ MSG_V("audio_setup: sample format: %s (requested: %s)\n",
audio_out_format_name(ao_data.format), audio_out_format_name(format));
+ ao_data.channels = channels;
if(format != AFMT_AC3) {
// We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it
- ao_data.channels = channels;
if (ao_data.channels > 2) {
if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 ||
ao_data.channels != channels ) {
- MSG_FATAL("audio_setup: Failed to set audio device to %d channels\n", channels);
+ MSG_ERR("audio_setup: Failed to set audio device to %d channels\n", channels);
return 0;
}
}
else {
int c = ao_data.channels-1;
if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) {
- MSG_FATAL("audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
+ MSG_ERR("audio_setup: Failed to set audio device to %d channels\n", ao_data.channels);
return 0;
}
+ ao_data.channels=c+1;
}
- MSG_INFO("audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
+ MSG_V("audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels);
// set rate
ao_data.samplerate=rate;
ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
- MSG_INFO("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
+ MSG_V("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate);
}
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){
int r=0;
- MSG_ERR("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
+ MSG_WARN("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
MSG_V("audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
} else {
@@ -201,25 +199,31 @@ ac3_retry:
}
free(data);
if(ao_data.buffersize==0){
- MSG_FATAL("\n *** Your audio driver DOES NOT support select() ***\n");
- MSG_FATAL("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
+ MSG_ERR("\n *** Your audio driver DOES NOT support select() ***\n"
+ "Recompile mplayerxp with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
return 0;
}
#endif
}
+ ao_data.bps=ao_data.channels;
+ if(ao_data.format != AFMT_U8 && ao_data.format != AFMT_S8)
+ ao_data.bps*=2;
+
ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down
- ao_data.bps*=rate;
+ ao_data.bps*=ao_data.samplerate;
return 1;
}
// close audio device
static void uninit(){
+ if(audio_fd == -1) return;
#ifdef SNDCTL_DSP_RESET
ioctl(audio_fd, SNDCTL_DSP_RESET, NULL);
#endif
close(audio_fd);
+ audio_fd = -1;
}
// stop playing and empty buffers (for seeking/pause)
@@ -253,6 +257,7 @@ static void audio_pause()
// resume playing, after audio_pause()
static void audio_resume()
{
+ reset();
}
Index: ao_plugin.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/ao_plugin.c,v
retrieving revision 1.5
retrieving revision 1.6
diff -p -u -d -r1.5 -r1.6
--- ao_plugin.c 30 Jun 2002 21:21:07 -0000 1.5
+++ ao_plugin.c 29 Apr 2003 17:25:14 -0000 1.6
@@ -124,7 +124,7 @@ static int init(int rate,int channels,in
// Create list of plugins from cfg option
int i=0;
if(ao_plugin_cfg.plugin_list){
- config = malloc(strlen(ao_plugin_cfg.plugin_list));
+ config = malloc(strlen(ao_plugin_cfg.plugin_list)+1);
if(!config) return 0;
if(!strcpy(config, ao_plugin_cfg.plugin_list) || !add_plugin(i,config)){
free(config);
@@ -148,15 +148,15 @@ static int init(int rate,int channels,in
ao_plugin_data.delay_mult=1;
ao_plugin_data.delay_fix=0;
- for(i=0;i<NPL,plugin(i);i++){
+ for(i=0;i<NPL && plugin(i);i++){
use_plugin[i]=plugin(i)->init();
if(!use_plugin[i]) plugin(i)->uninit();
}
npl=i;
- for(i=0;i<npl,plugin(i);i++)
+ for(i=0;i<npl && plugin(i);i++)
if(!use_plugin[i+unused]){
unused++;
- for(x=i;x<npl,plugin(x+1);x++) plugin(x)=plugin(x+1);
+ for(x=i;x<npl && plugin(x+1);x++) plugin(x)=plugin(x+1);
plugin(x)=NULL;
npl--;
i--;
Index: ao_sdl.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/ao_sdl.c,v
retrieving revision 1.7
retrieving revision 1.8
diff -p -u -d -r1.7 -r1.8
--- ao_sdl.c 30 Jun 2002 21:21:07 -0000 1.7
+++ ao_sdl.c 29 Apr 2003 17:25:14 -0000 1.8
@@ -30,9 +30,6 @@ static ao_info_t info =
LIBAO_EXTERN(sdl)
-
-extern int verbose;
-
// Samplesize used by the SDLlib AudioSpec struct
#define SAMPLESIZE 1024
@@ -95,8 +92,8 @@ static int read_buffer(unsigned char* da
// end ring buffer stuff
-#if defined(sun) && defined(__svr4__)
-/* setenv is missing on solaris */
+#if defined(HPUX) || defined(sun) && defined(__svr4__)
+/* setenv is missing on solaris and HPUX */
static void setenv(const char *name, const char *val, int _xx)
{
int len = strlen(name) + strlen(val) + 2;
@@ -143,7 +140,7 @@ void outputaudio(void *unused, Uint8 *st
static int init(int rate,int channels,int format,int flags){
/* SDL Audio Specifications */
- SDL_AudioSpec aspec;
+ SDL_AudioSpec aspec, obtained;
char drv_name[80];
int i;
/* Allocate ring-buffer memory */
@@ -153,6 +150,10 @@ static int init(int rate,int channels,in
setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
}
+ ao_data.channels=channels;
+ ao_data.samplerate=rate;
+ ao_data.format=format;
+
ao_data.bps=channels*rate;
if(format != AFMT_U8 && format != AFMT_S8)
ao_data.bps*=2;
@@ -205,14 +206,41 @@ void callback(void *userdata, Uint8 *str
}
/* Open the audio device and start playing sound! */
- if(SDL_OpenAudio(&aspec, NULL) < 0) {
+ if(SDL_OpenAudio(&aspec, &obtained) < 0) {
MSG_ERR("SDL: Unable to open audio: %s\n", SDL_GetError());
return(0);
}
- if(verbose) MSG_V("SDL: buf size = %d\n",aspec.size);
- if(ao_data.buffersize==-1) ao_data.buffersize=aspec.size;
-
+ /* did we got what we wanted ? */
+ ao_data.channels=obtained.channels;
+ ao_data.samplerate=obtained.freq;
+
+ switch(obtained.format) {
+ case AUDIO_U8 :
+ ao_data.format = AFMT_U8;
+ break;
+ case AUDIO_S16LSB :
+ ao_data.format = AFMT_S16_LE;
+ break;
+ case AUDIO_S16MSB :
+ ao_data.format = AFMT_S16_BE;
+ break;
+ case AUDIO_S8 :
+ ao_data.format = AFMT_S8;
+ break;
+ case AUDIO_U16LSB :
+ ao_data.format = AFMT_U16_LE;
+ break;
+ case AUDIO_U16MSB :
+ ao_data.format = AFMT_U16_BE;
+ break;
+ default:
+ MSG_WARN("SDL: Unsupported SDL audio format: 0x%x.\n", obtained.format);
+ return 0;
+ }
+
+ MSG_V("SDL: buf size = %d\n",aspec.size);
+ ao_data.buffersize=obtained.size;
SDL_AudioDriverName(drv_name, sizeof(drv_name));
MSG_OK("SDL: using %s audio driver (%iHz %s \"%s\")\n"
@@ -229,7 +257,7 @@ void callback(void *userdata, Uint8 *str
// close audio device
static void uninit(){
- if(verbose) MSG_INFO("SDL: Audio Subsystem shutting down!\n");
+ MSG_V("SDL: Audio Subsystem shutting down!\n");
SDL_CloseAudio();
SDL_QuitSubSystem(SDL_INIT_AUDIO);
}
@@ -251,13 +279,13 @@ static void reset(){
// stop playing, keep buffers (for pause)
static void audio_pause()
{
- SDL_LockAudio();
+ SDL_PauseAudio(1);
}
// resume playing, after audio_pause()
static void audio_resume()
{
- SDL_UnlockAudio();
+ SDL_PauseAudio(0);
}
@@ -290,9 +318,3 @@ static int play(void* data,int len,int f
static float get_delay(){
return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
}
-
-
-
-
-
-
Index: audio_out.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/audio_out.c,v
retrieving revision 1.4
retrieving revision 1.5
diff -p -u -d -r1.4 -r1.5
--- audio_out.c 11 Feb 2003 15:14:24 -0000 1.4
+++ audio_out.c 29 Apr 2003 17:25:14 -0000 1.5
@@ -14,18 +14,11 @@ extern ao_functions_t audio_out_oss;
#endif
extern ao_functions_t audio_out_null;
#ifdef HAVE_ALSA
- extern ao_functions_t audio_out_alsa;
-#endif
-#ifdef HAVE_ESD
- extern ao_functions_t audio_out_esd;
+extern ao_functions_t audio_out_alsa;
#endif
#ifdef HAVE_SDL
extern ao_functions_t audio_out_sdl;
#endif
-#ifdef HAVE_DVB
-extern ao_functions_t audio_out_mpegpes;
-#endif
-extern ao_functions_t audio_out_pcm;
extern ao_functions_t audio_out_pss;
extern ao_functions_t audio_out_plugin;
@@ -41,14 +34,6 @@ ao_functions_t* audio_out_drivers[] =
#ifdef HAVE_ALSA
&audio_out_alsa,
#endif
-#ifdef HAVE_ESD
- &audio_out_esd,
-#endif
-#ifdef HAVE_DVB
- &audio_out_mpegpes,
-#endif
- &audio_out_pcm,
&audio_out_plugin,
-// &audio_out_pss,
NULL
};
Index: audio_plugin.h
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/audio_plugin.h,v
retrieving revision 1.2
retrieving revision 1.3
diff -p -u -d -r1.2 -r1.3
--- audio_plugin.h 17 Jun 2002 07:43:14 -0000 1.2
+++ audio_plugin.h 29 Apr 2003 17:25:15 -0000 1.3
@@ -5,11 +5,11 @@
typedef struct ao_plugin_functions_s
{
ao_info_t *info;
- int (*control)(int cmd,int arg);
- int (*init)();
- void (*uninit)();
- void (*reset)();
- int (*play)();
+ int (*control)(int cmd,long arg);
+ int (*init)(void);
+ void (*uninit)(void);
+ void (*reset)(void);
+ int (*play)(void);
} ao_plugin_functions_t;
// Global data for all audio plugins
@@ -56,7 +56,7 @@ extern ao_plugin_cfg_t ao_plugin_cfg;
// This block should not be available in the pl_xxxx files
// due to compilation issues
#ifndef PLUGIN
-#define NPL 8+1 // Number of PLugins ( +1 list ends with NULL )
+#define NPL 9+1 // Number of PLugins ( +1 list ends with NULL )
// List of plugins
extern ao_plugin_functions_t audio_plugin_delay;
extern ao_plugin_functions_t audio_plugin_format;
@@ -66,6 +66,7 @@ extern ao_plugin_functions_t audio_plugi
extern ao_plugin_functions_t audio_plugin_extrastereo;
extern ao_plugin_functions_t audio_plugin_volnorm;
extern ao_plugin_functions_t audio_plugin_eq;
+extern ao_plugin_functions_t audio_plugin_lp;
#define AO_PLUGINS { \
&audio_plugin_delay, \
@@ -76,6 +77,7 @@ extern ao_plugin_functions_t audio_plugi
&audio_plugin_extrastereo, \
&audio_plugin_volnorm, \
&audio_plugin_eq, \
+ &audio_plugin_lp, \
NULL \
}
#endif /* PLUGIN */
@@ -83,6 +85,8 @@ extern ao_plugin_functions_t audio_plugi
// Control parameters used by the plugins
#define AOCONTROL_PLUGIN_SET_LEN 1 // All plugins must respond to this parameter
+#define AOCONTROL_PLUGIN_ES_SET 4 // set extra stereo coefficient
+#define AOCONTROL_PLUGIN_ES_GET 5 // get extra stereo coefficient
#endif
Index: audio_plugin_internal.h
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/audio_plugin_internal.h,v
retrieving revision 1.2
retrieving revision 1.3
diff -p -u -d -r1.2 -r1.3
--- audio_plugin_internal.h 30 Jun 2002 21:21:07 -0000 1.2
+++ audio_plugin_internal.h 29 Apr 2003 17:25:15 -0000 1.3
@@ -1,9 +1,9 @@
// prototypes:
static int control(int cmd,long arg);
-static int init();
-static void uninit();
-static void reset();
-static int play();
+static int init(void);
+static void uninit(void);
+static void reset(void);
+static int play(void);
#define LIBAO_PLUGIN_EXTERN(x) ao_plugin_functions_t audio_plugin_##x =\
{\
Index: fir.h
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/fir.h,v
retrieving revision 1.1.1.1
retrieving revision 1.2
diff -p -u -d -r1.1.1.1 -r1.2
--- fir.h 23 Mar 2002 11:09:20 -0000 1.1.1.1
+++ fir.h 29 Apr 2003 17:25:15 -0000 1.2
@@ -19,53 +19,15 @@ must be multiple of 4 (pad fiter with ze
Sometimes it works with filters longer than 4*16 (the problem is
overshoot and the acumulated energy in the filter taps). */
-#ifdef HAVE_MMX
-inline int32_t firn(int16_t* x, int16_t* w, int16_t n)
-{
- register int32_t y; // Output
- // Prologue
- asm volatile(" pxor %mm1, %mm1;\n" ); // Clear buffer yt
- // Main loop
- while((n-=4)>=0){
- asm volatile(
- " movq (%1), %%mm0;\n" // Load x(n:n+4)
- " pmaddwd (%0), %%mm0;\n" // yt(n:n+1)=sum(x(n:n+4).*w(n:n+4))
- " psrld $16, %%mm0;\n" // yt(n:n+1)=yt(n:n+1)>>16
- " paddd %%mm0, %%mm1;\n" // yt(n:n+1)=yt(n-2:n-1)+yt(n:n+1)
- :: "r" (w), "r" (x));
- w+=4; x+=4;
- }
- // Epilogue
- asm volatile(
- " movq %%mm1, %%mm0;\n"
- " punpckhdq %%mm1, %%mm0;\n"
- " paddd %%mm0, %%mm1;\n" //yt(n)=yt(n)+yt(n+1)
- " movd %%mm1, %0 ;\n" //y=yt
- " emms ;\n"
- : "=&r" (y));
- return y;
-}
-
-#else /* HAVE_MMX */
-
-// Same thing as above but in C
-inline int32_t firn(int16_t* x, int16_t* w, int16_t n)
-{
- register int32_t y=0;
- while((n-=4) >=0)
- y+=w[n]*x[n]+w[n+1]*x[n+1]+w[n+2]*x[n+2]+w[n+3]*x[n+3] >> 16;
- return y;
-}
-
-#endif /* HAVE_MMX */
+extern int32_t (*firn)(int16_t* x, int16_t* w, int n);
/* Add new data to circular queue designed to be used with a FIR
filter. xq is the circular queue, in pointing at the new sample, xi
current index for in xq and l the lenght of the filter */
-inline uint16_t updateq(int16_t* xq, int16_t* in, uint16_t xi, uint16_t l)
+static inline uint16_t updateq(int16_t* xq, int16_t* in, uint16_t xi, uint16_t l)
{
xq[xi]=xq[xi+l]=*in;
- return (--xi)&(l-1); \
+ return (--xi)&(l-1);
}
#endif /* __FIR_H__ */
Index: Makefile
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/Makefile,v
retrieving revision 1.4
retrieving revision 1.5
diff -p -u -d -r1.4 -r1.5
--- Makefile 6 Jan 2003 08:28:38 -0000 1.4
+++ Makefile 29 Apr 2003 17:25:15 -0000 1.5
@@ -4,7 +4,7 @@ include config.mak
LIBNAME = libao2.a
# TODO: moveout ao_sdl.c so it's only used when SDL is detected
-SRCS=afmt.c audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c pl_delay.c pl_format.c pl_surround.c remez.c pl_resample.c pl_volume.c pl_extrastereo.c pl_volnorm.c pl_eq.c $(OPTIONAL_SRCS)
+SRCS=afmt.c audio_out.c ao_null.c ao_plugin.c pl_delay.c pl_format.c pl_surround.c remez.c ao_accel.c pl_lp.c pl_resample.c pl_volume.c pl_extrastereo.c pl_volnorm.c pl_eq.c $(OPTIONAL_SRCS)
OBJS=$(SRCS:.c=.o)
Index: pl_delay.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/pl_delay.c,v
retrieving revision 1.4
retrieving revision 1.5
diff -p -u -d -r1.4 -r1.5
--- pl_delay.c 30 Jun 2002 21:21:07 -0000 1.4
+++ pl_delay.c 29 Apr 2003 17:25:15 -0000 1.5
@@ -56,7 +56,6 @@ static int control(int cmd,long arg){
// open & setup audio device
// return: 1=success 0=fail
static int init(){
- int i=0;
float time_delay; // The time in [s] this plugin delays the output data
/* If the output format of any of the below parameters differs from
@@ -93,11 +92,8 @@ static void uninit(){
// empty buffers
static void reset(){
- int i = 0;
- for(i=0;i<pl_delay.len;i++)
- ((char*)pl_delay.data)[i]=0;
- for(i=0;i<ao_plugin_cfg.pl_delay_len;i++)
- ((char*)pl_delay.delay)[i]=0;
+ memset(pl_delay.data,0,pl_delay.len);
+ memset(pl_delay.delay,0,ao_plugin_cfg.pl_delay_len);
}
// processes 'ao_plugin_data.len' bytes of 'data'
Index: pl_eq.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/pl_eq.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -p -u -d -r1.3 -r1.4
--- pl_eq.c 17 Sep 2002 16:33:10 -0000 1.3
+++ pl_eq.c 29 Apr 2003 17:25:15 -0000 1.4
@@ -201,17 +201,16 @@ static int play(){
// Pointer to circular buffer wq
register int16_t* wq = pl_eq.wq[ci][k];
// Calculate output from AR part of current filter
- register int32_t xt = (x*pl_eq.b[k][0]) >> 4;
+ register int32_t xt = (x*pl_eq.b[k][0]) / 48;
register int32_t w = xt + wq[0]*pl_eq.a[k][0] + wq[1]*pl_eq.a[k][1];
// Calculate output form MA part of current filter
yt+=(((w + wq[1]*pl_eq.b[k][1]) >> 10)*g[k]) >> 12;
// Update circular buffer
- wq[1] = wq[0]; wq[0] = w >> 14;
+ wq[1] = wq[0]; wq[0] = w / 24576;
}
// Calculate output
- *out=(int16_t)(yt+x);
- out+=nch;
+ *out=(int16_t)(yt * 0.25 + x * 0.5);
}
}
return 1;
Index: pl_extrastereo.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/pl_extrastereo.c,v
retrieving revision 1.4
retrieving revision 1.5
diff -p -u -d -r1.4 -r1.5
--- pl_extrastereo.c 17 Sep 2002 16:33:10 -0000 1.4
+++ pl_extrastereo.c 29 Apr 2003 17:25:15 -0000 1.5
@@ -45,6 +45,12 @@ static int control(int cmd,long arg){
switch(cmd){
case AOCONTROL_PLUGIN_SET_LEN:
return CONTROL_OK;
+ case AOCONTROL_PLUGIN_ES_SET:
+ pl_extrastereo.mul=*((float*)arg);
+ return CONTROL_OK;
+ case AOCONTROL_PLUGIN_ES_GET:
+ *((float*)arg)=pl_extrastereo.mul;
+ return CONTROL_OK;
}
return CONTROL_UNKNOWN;
}
Index: pl_format.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/pl_format.c,v
retrieving revision 1.5
retrieving revision 1.6
diff -p -u -d -r1.5 -r1.6
--- pl_format.c 17 Sep 2002 16:33:10 -0000 1.5
+++ pl_format.c 29 Apr 2003 17:25:15 -0000 1.6
@@ -253,7 +253,7 @@ static int play(){
switch(pl_format.in&SIGN_MASK){
case(US):
for(i=0;i<len;i++)
- ((int32_t*)in_data)[i]=(int32_t)(-(1<<31-1)+((uint32_t*)in_data)[i]);
+ ((int32_t*)in_data)[i]=(int32_t)(((uint32_t*)in_data)[i]-0x80000000);
break;
case(SI):
for(i=0;i<len;i++)
Index: pl_resample.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/pl_resample.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -p -u -d -r1.3 -r1.4
--- pl_resample.c 30 Jun 2002 21:21:07 -0000 1.3
+++ pl_resample.c 29 Apr 2003 17:25:15 -0000 1.4
@@ -161,22 +161,6 @@ static void uninit(){
static void reset(){
}
-// 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=(*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
@@ -281,3 +265,20 @@ int downsample(){
ao_plugin_data.data=pl_resample.data;
return 1;
}
+
+// 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=(*in++)>>1;
+ return 1;
+ }
+ if(pl_resample.up>pl_resample.dn)
+ return upsample();
+ else
+ return downsample();
+}
+
Index: pl_surround.c
===================================================================
RCS file: /cvsroot/mplayerxp/mplayerxp/mplayerxp/libao2/pl_surround.c,v
retrieving revision 1.6
retrieving revision 1.7
diff -p -u -d -r1.6 -r1.7
--- pl_surround.c 17 Sep 2002 16:33:10 -0000 1.6
+++ pl_surround.c 29 Apr 2003 17:25:15 -0000 1.7
@@ -169,9 +169,6 @@ static double steering_matrix[][12] = {
{.707, .0, .0, .707, .5, -.5, .5878, -.3928, .3928, -.5878, .5, .5},
};
-// Experimental moving average dominances
-static int amp_L = 0, amp_R = 0, amp_C = 0, amp_S = 0;
-
// processes 'ao_plugin_data.len' bytes of 'data'
// called for every block of data
static int play(){
--- ao_mpegpes.c DELETED ---
--- ao_pcm.c DELETED ---
|