[Mplayer-cvslog] CVS: main/libao2 Makefile,1.11,1.12 ao_alsa5.c,1.7,1.8 ao_alsa9.c,1.10,1.11 ao_mpegpes.c,1.1,1.2 ao_null.c,1.4,1.5 ao_oss.c,1.11,1.12 ao_pcm.c,1.3,1.4 ao_sdl.c,1.13,1.14 ao_sgi.c,1.1,1.2 ao_sun.c,1.9,1.10 audio_out.c,1.18,1.19 audio_out.h,1.6,1.7 audio_out_internal.h,1.2,1.3
Anders Johansson
anders at mplayer.dev.hu
Sat Nov 24 06:21:25 CET 2001
Update of /cvsroot/mplayer/main/libao2
In directory mplayer:/var/tmp.root/cvs-serv31552/libao2
Modified Files:
Makefile ao_alsa5.c ao_alsa9.c ao_mpegpes.c ao_null.c ao_oss.c
ao_pcm.c ao_sdl.c ao_sgi.c ao_sun.c audio_out.c audio_out.h
audio_out_internal.h
Log Message:
interface to libao2 changed ao_plugin added
Index: Makefile
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/Makefile,v
retrieving revision 1.11
retrieving revision 1.12
diff -u -r1.11 -r1.12
--- Makefile 18 Nov 2001 18:58:03 -0000 1.11
+++ Makefile 24 Nov 2001 05:21:21 -0000 1.12
@@ -4,7 +4,7 @@
LIBNAME = libao2.a
# TODO: moveout ao_sdl.c so it's only used when SDL is detected
-SRCS=afmt.c audio_out.c ao_null.c ao_pcm.c ao_mpegpes.c $(OPTIONAL_SRCS)
+SRCS=afmt.c audio_out.c ao_mpegpes.c ao_null.c ao_pcm.c ao_plugin.c $(OPTIONAL_SRCS)
OBJS=$(SRCS:.c=.o)
CFLAGS = $(OPTFLAGS) -I. -I.. $(SDL_INC) $(EXTRA_INC)
@@ -38,3 +38,4 @@
ifneq ($(wildcard .depend),)
include .depend
endif
+
Index: ao_alsa5.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_alsa5.c,v
retrieving revision 1.7
retrieving revision 1.8
diff -u -r1.7 -r1.8
--- ao_alsa5.c 23 Nov 2001 16:35:51 -0000 1.7
+++ ao_alsa5.c 24 Nov 2001 05:21:21 -0000 1.8
@@ -27,15 +27,6 @@
LIBAO_EXTERN(alsa5)
-/* global variables:
- ao_samplerate
- ao_channels
- ao_format
- ao_bps
- ao_outburst
- ao_buffersize
-*/
-
static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
static int alsa_rate = SND_PCM_RATE_CONTINUOUS;
@@ -74,12 +65,12 @@
return(0);
}
- ao_format = format;
- ao_channels = channels - 1;
- ao_samplerate = rate_hz;
- ao_bps = ao_samplerate*(ao_channels+1);
- ao_outburst = OUTBURST;
- ao_buffersize = 16384;
+ ao_data.format = format;
+ ao_data.channels = channels - 1;
+ ao_data.samplerate = rate_hz;
+ ao_data.bps = ao_data.samplerate*(ao_data.channels+1);
+ ao_data.outburst = OUTBURST;
+ ao_data.buffersize = 16384;
memset(&alsa_format, 0, sizeof(alsa_format));
switch (format)
@@ -111,7 +102,7 @@
{
case SND_PCM_SFMT_S16_LE:
case SND_PCM_SFMT_U16_LE:
- ao_bps *= 2;
+ ao_data.bps *= 2;
break;
case -1:
printf("alsa-init: invalid format (%s) requested - output disabled\n",
@@ -161,8 +152,8 @@
break;
}
- alsa_format.rate = ao_samplerate;
- alsa_format.voices = ao_channels*2;
+ alsa_format.rate = ao_data.samplerate;
+ alsa_format.voices = ao_data.channels*2;
alsa_format.interleave = 1;
if ((err = snd_pcm_open(&alsa_handler, 0, 0, SND_PCM_OPEN_PLAYBACK)) < 0)
@@ -189,13 +180,15 @@
printf("alsa-init: pcm channel info error: %s\n", snd_strerror(err));
return(0);
}
+
#ifndef __QNX__
if (chninfo.buffer_size)
- ao_buffersize = chninfo.buffer_size;
+ ao_data.buffersize = chninfo.buffer_size;
#endif
+
if (verbose)
printf("alsa-init: setting preferred buffer size from driver: %d bytes\n",
- ao_buffersize);
+ ao_data.buffersize);
}
memset(¶ms, 0, sizeof(params));
@@ -204,7 +197,7 @@
params.format = alsa_format;
params.start_mode = SND_PCM_START_DATA;
params.stop_mode = SND_PCM_STOP_ROLLOVER;
- params.buf.stream.queue_size = ao_buffersize;
+ params.buf.stream.queue_size = ao_data.buffersize;
params.buf.stream.fill = SND_PCM_FILL_NONE;
if ((err = snd_pcm_channel_params(alsa_handler, ¶ms)) < 0)
@@ -217,8 +210,8 @@
setup.channel = SND_PCM_CHANNEL_PLAYBACK;
setup.mode = SND_PCM_MODE_STREAM;
setup.format = alsa_format;
- setup.buf.stream.queue_size = ao_buffersize;
- setup.msbits_per_sample = ao_bps;
+ setup.buf.stream.queue_size = ao_data.buffersize;
+ setup.msbits_per_sample = ao_data.bps;
if ((err = snd_pcm_channel_setup(alsa_handler, &setup)) < 0)
{
@@ -233,7 +226,7 @@
}
printf("AUDIO: %d Hz/%d channels/%d bps/%d bytes buffer/%s\n",
- ao_samplerate, ao_channels+1, ao_bps, ao_buffersize,
+ ao_data.samplerate, ao_data.channels+1, ao_data.bps, ao_data.buffersize,
snd_pcm_get_format_name(alsa_format.format));
return(1);
}
@@ -357,15 +350,15 @@
return(ch_stat.free);
}
-/* how many unplayed bytes are in the buffer */
-static int get_delay()
+/* delay in seconds between first and last sample in buffer */
+static float get_delay()
{
snd_pcm_channel_status_t ch_stat;
ch_stat.channel = SND_PCM_CHANNEL_PLAYBACK;
if (snd_pcm_channel_status(alsa_handler, &ch_stat) < 0)
- return(ao_buffersize); /* error occured */
+ return((float)ao_data.buffersize/(float)ao_data.bps); /* error occured */
else
- return(ch_stat.count);
+ return((float)ch_stat.count/(float)ao_data.bps);
}
Index: ao_alsa9.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_alsa9.c,v
retrieving revision 1.10
retrieving revision 1.11
diff -u -r1.10 -r1.11
--- ao_alsa9.c 15 Oct 2001 11:17:19 -0000 1.10
+++ ao_alsa9.c 24 Nov 2001 05:21:22 -0000 1.11
@@ -32,14 +32,6 @@
LIBAO_EXTERN(alsa9)
-/* global variables:
- ao_samplerate
- ao_channels
- ao_format
- ao_bps
- ao_outburst
- ao_buffersize
-*/
static snd_pcm_t *alsa_handler;
static snd_pcm_format_t alsa_format;
@@ -67,7 +59,7 @@
strncpy(alsa_device, (char *)arg, ALSA_DEVICE_SIZE);
uninit();
- ret = init(ao_samplerate, ao_channels, ao_format, 0);
+ ret = init(ao_data.samplerate, ao_data.channels, ao_data.format, 0);
if (ret == 0)
return(CONTROL_ERROR);
else
@@ -112,12 +104,12 @@
return(0);
}
- ao_samplerate = rate_hz;
- ao_bps = channels; /* really this is bytes per frame so bad varname */
- ao_format = format;
- ao_channels = channels;
- ao_outburst = OUTBURST;
- ao_buffersize = 16384;
+ ao_data.samplerate = rate_hz;
+ ao_data.bps = channels; /* really this is bytes per frame so bad varname */
+ ao_data.format = format;
+ ao_data.channels = channels;
+ ao_data.outburst = OUTBURST;
+ ao_data.buffersize = 16384;
switch (format)
{
@@ -148,7 +140,7 @@
{
case SND_PCM_FORMAT_S16_LE:
case SND_PCM_FORMAT_U16_LE:
- ao_bps *= 2;
+ ao_data.bps *= 2;
break;
case -1:
printf("alsa-init: invalid format (%s) requested - output disabled\n",
@@ -164,8 +156,8 @@
return(0);
}
- if (ao_subdevice != NULL)
- alsa_device = ao_subdevice;
+ if (ao_data.subdevice != NULL)
+ alsa_device = ao_data.subdevice;
if (alsa_device == NULL)
{
@@ -219,14 +211,14 @@
}
if ((err = snd_pcm_hw_params_set_channels(alsa_handler, alsa_hwparams,
- ao_channels)) < 0)
+ ao_data.channels)) < 0)
{
printf("alsa-init: unable to set channels: %s\n",
snd_strerror(err));
return(0);
}
- if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_samplerate, 0)) < 0)
+ if ((err = snd_pcm_hw_params_set_rate_near(alsa_handler, alsa_hwparams, ao_data.samplerate, 0)) < 0)
/* was originally only snd_pcm_hw_params_set_rate jp*/
{
printf("alsa-init: unable to set samplerate-2: %s\n",
@@ -259,9 +251,9 @@
return(0);
} else
{
- ao_buffersize = err;
+ ao_data.buffersize = err;
if (verbose)
- printf("alsa-init: got buffersize %i\n", ao_buffersize);
+ printf("alsa-init: got buffersize %i\n", ao_data.buffersize);
}
#endif
@@ -278,7 +270,7 @@
alsa_buffer_time = err;
if ((err = snd_pcm_hw_params_set_period_time_near(alsa_handler, alsa_hwparams, alsa_buffer_time/4, 0)) < 0)
- /* original: alsa_buffer_time/ao_bps */
+ /* original: alsa_buffer_time/ao_data.bps */
{
printf("alsa-init: unable to set period time: %s\n",
snd_strerror(err));
@@ -299,7 +291,7 @@
#ifdef sw_params
{
chunk_size = snd_pcm_hw_params_get_period_size(alsa_hwparams, 0);
- start_threshold = (double) ao_samplerate * start_delay / 1000000;
+ start_threshold = (double) ao_data.samplerate * start_delay / 1000000;
xfer_align = snd_pcm_sw_params_get_xfer_align(alsa_swparams);
if ((err = snd_pcm_sw_params_current(alsa_handler, alsa_swparams)) < 0)
@@ -369,7 +361,7 @@
}
#endif
printf("AUDIO: %d Hz/%d channels/%d bpf/%d bytes buffer/%s\n",
- ao_samplerate, ao_channels, ao_bps, ao_buffersize,
+ ao_data.samplerate, ao_data.channels, ao_data.bps, ao_data.buffersize,
snd_pcm_format_description(alsa_format));
return(1);
}
@@ -488,7 +480,7 @@
got_len = snd_pcm_writei(alsa_handler, data, len / 4);
- //if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps)) {
+ //if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps)) {
//SHOULD BE FIXED
if (got_len == -EPIPE) /* underrun? */
{
@@ -498,7 +490,7 @@
printf("alsa-play: playback prepare error: %s\n", snd_strerror(got_len));
return(0);
}
- if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_bps))) != (len/ao_bps))
+ if ((got_len = snd_pcm_writei(alsa_handler, data, (len/ao_data.bps))) != (len/ao_data.bps))
{
printf("alsa-play: write error after reset: %s - giving up\n",
snd_strerror(got_len));
@@ -533,7 +525,7 @@
case SND_PCM_STATE_OPEN:
case SND_PCM_STATE_PREPARED:
case SND_PCM_STATE_RUNNING:
- ret = snd_pcm_status_get_avail(status) * ao_bps;
+ ret = snd_pcm_status_get_avail(status) * ao_data.bps;
break;
default:
ret = 0;
@@ -546,11 +538,11 @@
return(ret);
}
-/* how many unplayed bytes are in the buffer */
-static int get_delay()
+/* delay in seconds between first and last sample in buffer */
+static float get_delay()
{
snd_pcm_status_t *status;
- int ret;
+ float ret;
if ((ret = snd_pcm_status_malloc(&status)) < 0)
{
@@ -569,7 +561,7 @@
case SND_PCM_STATE_OPEN:
case SND_PCM_STATE_PREPARED:
case SND_PCM_STATE_RUNNING:
- ret = snd_pcm_status_get_delay(status) * ao_bps;
+ ret = (float)snd_pcm_status_get_delay(status)/(float)ao_data.samplerate;
break;
default:
ret = 0;
Index: ao_mpegpes.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_mpegpes.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -r1.1 -r1.2
--- ao_mpegpes.c 5 Nov 2001 03:01:11 -0000 1.1
+++ ao_mpegpes.c 24 Nov 2001 05:21:22 -0000 1.2
@@ -16,13 +16,6 @@
LIBAO_EXTERN(mpegpes)
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
// to set/get/query special features/parameters
static int control(int cmd,int arg){
@@ -33,8 +26,8 @@
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
- ao_outburst=2000;
- ao_format=format;
+ ao_data.outburst=2000;
+ ao_data.format=format;
return 1;
}
@@ -67,10 +60,10 @@
// return: how many bytes can be played without blocking
static int get_space(){
- float x=(float)(vo_pts-ao_pts)/90000.0-0.5;
+ float x=(float)(vo_pts-ao_data.pts)/90000.0-0.5;
int y;
if(x<=0) return 0;
- y=48000*4*x;y/=ao_outburst;y*=ao_outburst;
+ y=48000*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
// printf("diff: %5.3f -> %d \n",x,y);
return y;
}
@@ -79,20 +72,20 @@
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){
- if(ao_format==AFMT_MPEG)
- send_pes_packet(data,len,0x1C0,ao_pts);
+ if(ao_data.format==AFMT_MPEG)
+ send_pes_packet(data,len,0x1C0,ao_data.pts);
else {
int i;
unsigned short *s=data;
for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be
- send_lpcm_packet(data,len,0xA0,ao_pts);
+ send_lpcm_packet(data,len,0xA0,ao_data.pts);
}
return len;
}
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
- return 0;
+ return 0.0;
}
Index: ao_null.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_null.c,v
retrieving revision 1.4
retrieving revision 1.5
diff -u -r1.4 -r1.5
--- ao_null.c 29 Aug 2001 21:08:12 -0000 1.4
+++ ao_null.c 24 Nov 2001 05:21:22 -0000 1.5
@@ -14,13 +14,6 @@
LIBAO_EXTERN(null)
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
// to set/get/query special features/parameters
static int control(int cmd,int arg){
@@ -31,7 +24,7 @@
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
- ao_outburst=4096;
+ ao_data.outburst=4096;
return 0;
}
@@ -61,7 +54,7 @@
// return: how many bytes can be played without blocking
static int get_space(){
- return ao_outburst;
+ return ao_data.outburst;
}
// plays 'len' bytes of 'data'
@@ -72,10 +65,10 @@
return len;
}
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
- return 0;
+ return 0.0;
}
Index: ao_oss.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_oss.c,v
retrieving revision 1.11
retrieving revision 1.12
diff -u -r1.11 -r1.12
--- ao_oss.c 4 Oct 2001 18:25:50 -0000 1.11
+++ ao_oss.c 24 Nov 2001 05:21:22 -0000 1.12
@@ -28,14 +28,6 @@
LIBAO_EXTERN(oss)
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
-
static char *dsp="/dev/dsp";
static audio_buf_info zz;
static int audio_fd=-1;
@@ -57,7 +49,7 @@
ao_control_vol_t *vol = (ao_control_vol_t *)arg;
int fd, v, mcmd, devs;
- if(ao_format == AFMT_AC3)
+ if(ao_data.format == AFMT_AC3)
return CONTROL_TRUE;
if ((fd = open("/dev/mixer", O_RDONLY)) > 0)
@@ -118,58 +110,62 @@
return 0;
}
- ao_format=format;
- ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_format);
- if(format == AFMT_AC3 && ao_format != AFMT_AC3) {
+ ao_data.bps=(channels+1)*rate;
+ if(format != AFMT_U8 && format != AFMT_S8)
+ ao_data.bps*=2;
+
+ ao_data.format=format;
+ ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
+ if(format == AFMT_AC3 && ao_data.format != AFMT_AC3) {
printf("Can't set audio device %s to AC3 output\n", dsp);
return 0;
}
printf("audio_setup: sample format: %s (requested: %s)\n",
- audio_out_format_name(ao_format), audio_out_format_name(format));
+ audio_out_format_name(ao_data.format), audio_out_format_name(format));
if(format != AFMT_AC3) {
- ao_channels=channels-1;
- ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_channels);
+ ao_data.channels=channels-1;
+ ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
// set rate
- ao_samplerate=rate;
- ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_samplerate);
- printf("audio_setup: using %d Hz samplerate (requested: %d)\n",ao_samplerate,rate);
+ ao_data.samplerate=rate;
+ ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
+ printf("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;
printf("audio_setup: driver doesn't support SNDCTL_DSP_GETOSPACE :-(\n");
if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){
- printf("audio_setup: %d bytes/frag (config.h)\n",ao_outburst);
+ printf("audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst);
} else {
- ao_outburst=r;
- printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_outburst);
+ ao_data.outburst=r;
+ printf("audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst);
}
} else {
printf("audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n",
zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes);
- if(ao_buffersize==-1) ao_buffersize=zz.bytes;
- ao_outburst=zz.fragsize;
+ if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes;
+ ao_data.outburst=zz.fragsize;
}
- if(ao_buffersize==-1){
+ if(ao_data.buffersize==-1){
// Measuring buffer size:
void* data;
- ao_buffersize=0;
+ ao_data.buffersize=0;
#ifdef HAVE_AUDIO_SELECT
- data=malloc(ao_outburst); memset(data,0,ao_outburst);
- while(ao_buffersize<0x40000){
+ data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst);
+ while(ao_data.buffersize<0x40000){
fd_set rfds;
struct timeval tv;
FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
tv.tv_sec=0; tv.tv_usec = 0;
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
- write(audio_fd,data,ao_outburst);
- ao_buffersize+=ao_outburst;
+ write(audio_fd,data,ao_data.outburst);
+ ao_data.buffersize+=ao_data.outburst;
}
free(data);
- if(ao_buffersize==0){
+ if(ao_data.buffersize==0){
printf("\n *** Your audio driver DOES NOT support select() ***\n");
printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
return 0;
@@ -197,10 +193,10 @@
return;
}
- ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_format);
- if(ao_format != AFMT_AC3) {
- ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_channels);
- ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_samplerate);
+ ioctl (audio_fd, SNDCTL_DSP_SETFMT, &ao_data.format);
+ if(ao_data.format != AFMT_AC3) {
+ ioctl (audio_fd, SNDCTL_DSP_STEREO, &ao_data.channels);
+ ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate);
}
}
@@ -219,7 +215,7 @@
// return: how many bytes can be played without blocking
static int get_space(){
- int playsize=ao_outburst;
+ int playsize=ao_data.outburst;
#ifdef SNDCTL_DSP_GETOSPACE
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){
@@ -240,35 +236,38 @@
}
#endif
- return ao_outburst;
+ return ao_data.outburst;
}
// plays 'len' bytes of 'data'
// it should round it down to outburst*n
// return: number of bytes played
static int play(void* data,int len,int flags){
- len/=ao_outburst;
- len=write(audio_fd,data,len*ao_outburst);
+ len/=ao_data.outburst;
+ len=write(audio_fd,data,len*ao_data.outburst);
return len;
}
static int audio_delay_method=2;
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
+ /* Calculate how many bytes/second is sent out */
if(audio_delay_method==2){
- //
int r=0;
if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1)
- return r;
+ return ((float)r)/(float)ao_data.bps;
audio_delay_method=1; // fallback if not supported
}
if(audio_delay_method==1){
// SNDCTL_DSP_GETOSPACE
if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1)
- return ao_buffersize-zz.bytes;
+ return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps;
audio_delay_method=0; // fallback if not supported
}
- return ao_buffersize;
+ return ((float)ao_data.buffersize)/(float)ao_data.bps;
}
+
+
+
Index: ao_pcm.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_pcm.c,v
retrieving revision 1.3
retrieving revision 1.4
diff -u -r1.3 -r1.4
--- ao_pcm.c 12 Jun 2001 14:37:32 -0000 1.3
+++ ao_pcm.c 24 Nov 2001 05:21:22 -0000 1.4
@@ -14,14 +14,6 @@
LIBAO_EXTERN(pcm)
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
-
char *ao_outputfilename = NULL;
int ao_pcm_waveheader = 1;
@@ -90,7 +82,7 @@
printf("PCM: Info - to write WAVE files use -waveheader (default), for RAW PCM -nowaveheader.\n");
fp = fopen(ao_outputfilename, "wb");
- ao_outburst = 4096;
+ ao_data.outburst = 4096;
if(fp) {
@@ -134,7 +126,7 @@
// return: how many bytes can be played without blocking
static int get_space(){
- return ao_outburst;
+ return ao_data.outburst;
}
// plays 'len' bytes of 'data'
@@ -151,10 +143,10 @@
return len;
}
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
- return 0;
+ return 0.0;
}
Index: ao_sdl.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sdl.c,v
retrieving revision 1.13
retrieving revision 1.14
diff -u -r1.13 -r1.14
--- ao_sdl.c 27 Jun 2001 11:40:15 -0000 1.13
+++ ao_sdl.c 24 Nov 2001 05:21:22 -0000 1.14
@@ -29,13 +29,6 @@
LIBAO_EXTERN(sdl)
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
extern int verbose;
@@ -150,6 +143,10 @@
setenv("SDL_AUDIODRIVER", ao_subdevice, 1);
printf("SDL: using %s audio driver\n", ao_subdevice);
}
+
+ ao_data.bps=(channels+1)*rate;
+ if(format != AFMT_U8 && format != AFMT_S8)
+ ao_data.bps*=2;
/* The desired audio format (see SDL_AudioSpec) */
switch(format) {
@@ -205,7 +202,7 @@
}
if(verbose) printf("SDL: buf size = %d\n",aspec.size);
- if(ao_buffersize==-1) ao_buffersize=aspec.size;
+ if(ao_data.buffersize==-1) ao_data.buffersize=aspec.size;
/* unsilence audio, if callback is ready */
SDL_PauseAudio(0);
@@ -278,9 +275,9 @@
#endif
}
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
- return buffered_bytes + ao_buffersize;
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
+ return (float)(buffered_bytes + ao_data.buffersize)/(float)ao_data.bps;
}
Index: ao_sgi.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sgi.c,v
retrieving revision 1.1
retrieving revision 1.2
diff -u -r1.1 -r1.2
--- ao_sgi.c 24 Oct 2001 14:06:36 -0000 1.1
+++ ao_sgi.c 24 Nov 2001 05:21:22 -0000 1.2
@@ -22,13 +22,6 @@
LIBAO_EXTERN(sgi)
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
static ALconfig ao_config;
static ALport ao_port;
@@ -86,9 +79,9 @@
}
- ao_buffersize=131072;
- ao_outburst = ao_buffersize/16;
- ao_channels = channels;
+ ao_data.buffersize=131072;
+ ao_data.outburst = ao_data.buffersize/16;
+ ao_data.channels = channels;
ao_config = alNewConfig();
@@ -164,7 +157,7 @@
// printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_outburst);
// printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port));
- return alGetFillable(ao_port)*(2*ao_channels);
+ return alGetFillable(ao_port)*(2*ao_data.channels);
}
@@ -177,14 +170,14 @@
// printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config);
// printf("channels %d\n", ao_channels);
- alWriteFrames(ao_port, data, len/(2*ao_channels));
+ alWriteFrames(ao_port, data, len/(2*ao_data.channels));
return len;
}
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
// printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize);
Index: ao_sun.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/ao_sun.c,v
retrieving revision 1.9
retrieving revision 1.10
diff -u -r1.9 -r1.10
--- ao_sun.c 31 Aug 2001 20:34:17 -0000 1.9
+++ ao_sun.c 24 Nov 2001 05:21:22 -0000 1.10
@@ -42,18 +42,11 @@
#endif
-// there are some globals:
-// ao_samplerate
-// ao_channels
-// ao_format
-// ao_bps
-// ao_outburst
-// ao_buffersize
-
static char *audio_dev = "/dev/audio";
static int queued_bursts = 0;
static int queued_samples = 0;
static int bytes_per_sample = 0;
+static int byte_per_sec = 0;
static int convert_u8_s8;
static int audio_fd = -1;
static enum {
@@ -230,7 +223,6 @@
static int init(int rate,int channels,int format,int flags){
audio_info_t info;
- int byte_per_sec;
int ok;
if (ao_subdevice) audio_dev = ao_subdevice;
@@ -252,13 +244,13 @@
ioctl(audio_fd, AUDIO_DRAIN, 0);
AUDIO_INITINFO(&info);
- info.play.encoding = oss2sunfmt(ao_format = format);
+ info.play.encoding = oss2sunfmt(ao_data.format = format);
info.play.precision =
(format==AFMT_S16_LE || format==AFMT_S16_BE
? AUDIO_PRECISION_16
: AUDIO_PRECISION_8);
- info.play.channels = ao_channels = channels;
- info.play.sample_rate = ao_samplerate = rate;
+ info.play.channels = ao_data.channels = channels;
+ info.play.sample_rate = ao_data.samplerate = rate;
convert_u8_s8 = 0;
ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0;
if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) {
@@ -278,37 +270,37 @@
bytes_per_sample = channels * info.play.precision / 8;
byte_per_sec = bytes_per_sample * rate;
- ao_outburst = byte_per_sec > 100000 ? 16384 : 8192;
+ ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192;
#ifdef __not_used__
/*
- * hmm, ao_buffersize is currently not used in this driver, do there's
+ * hmm, ao_data.buffersize is currently not used in this driver, do there's
* no need to measure it
*/
- if(ao_buffersize==-1){
+ if(ao_data.buffersize==-1){
// Measuring buffer size:
void* data;
- ao_buffersize=0;
+ ao_data.buffersize=0;
#ifdef HAVE_AUDIO_SELECT
- data = malloc(ao_outburst);
- memset(data, format==AFMT_U8 ? 0x80 : 0, ao_outburst);
- while(ao_buffersize<0x40000){
+ data = malloc(ao_data.outburst);
+ memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst);
+ while(ao_data.buffersize<0x40000){
fd_set rfds;
struct timeval tv;
FD_ZERO(&rfds); FD_SET(audio_fd,&rfds);
tv.tv_sec=0; tv.tv_usec = 0;
if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break;
- write(audio_fd,data,ao_outburst);
- ao_buffersize+=ao_outburst;
+ write(audio_fd,data,ao_data.outburst);
+ ao_data.buffersize+=ao_data.outburst;
}
free(data);
- if(ao_buffersize==0){
+ if(ao_data.buffersize==0){
printf("\n *** Your audio driver DOES NOT support select() ***\n");
printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n");
return 0;
}
#ifdef __svr4__
- // remove the 0 bytes from the above ao_buffersize measurement from the
+ // remove the 0 bytes from the above ao_data.buffersize measurement from the
// audio driver's STREAMS queue
ioctl(audio_fd, I_FLUSH, FLUSHW);
#endif
@@ -352,13 +344,13 @@
ioctl(audio_fd, AUDIO_DRAIN, 0);
AUDIO_INITINFO(&info);
- info.play.encoding = oss2sunfmt(ao_format);
+ info.play.encoding = oss2sunfmt(ao_data.format);
info.play.precision =
- (ao_format==AFMT_S16_LE || ao_format==AFMT_S16_BE
+ (ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE
? AUDIO_PRECISION_16
: AUDIO_PRECISION_8);
- info.play.channels = ao_channels;
- info.play.sample_rate = ao_samplerate;
+ info.play.channels = ao_data.channels;
+ info.play.sample_rate = ao_data.samplerate;
info.play.samples = 0;
info.play.eof = 0;
info.play.error = 0;
@@ -388,7 +380,7 @@
// return: how many bytes can be played without blocking
static int get_space(){
- int playsize = ao_outburst;
+ int playsize = ao_data.outburst;
audio_info_t info;
// check buffer
@@ -408,7 +400,7 @@
if (queued_bursts - info.play.eof > 2)
return 0;
- return ao_outburst;
+ return ao_data.outburst;
}
// plays 'len' bytes of 'data'
@@ -421,13 +413,13 @@
int native_endian = AFMT_S16_LE;
#endif
- if (len < ao_outburst) return 0;
- len /= ao_outburst;
- len *= ao_outburst;
+ if (len < ao_data.outburst) return 0;
+ len /= ao_data.outburst;
+ len *= ao_data.outburst;
/* 16-bit format using the 'wrong' byteorder? swap words */
- if ((ao_format == AFMT_S16_LE || ao_format == AFMT_S16_BE)
- && ao_format != native_endian) {
+ if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE)
+ && ao_data.format != native_endian) {
static void *swab_buf;
static int swab_len;
if (len > swab_len) {
@@ -440,7 +432,7 @@
}
swab(data, swab_buf, len);
data = swab_buf;
- } else if (ao_format == AFMT_U8 && convert_u8_s8) {
+ } else if (ao_data.format == AFMT_U8 && convert_u8_s8) {
int i;
unsigned char *p = data;
@@ -460,13 +452,13 @@
}
-// return: how many unplayed bytes are in the buffer
-static int get_delay(){
+// return: delay in seconds between first and last sample in buffer
+static float get_delay(){
audio_info_t info;
ioctl(audio_fd, AUDIO_GETINFO, &info);
if (info.play.samples && enable_sample_timing == RTSC_ENABLED)
- return (queued_samples - info.play.samples) * bytes_per_sample;
+ return (float)(queued_samples - info.play.samples) / (float)byte_per_sec;
else
- return (queued_bursts - info.play.eof) * ao_outburst;
+ return (flaot)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec;
}
Index: audio_out.c
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/audio_out.c,v
retrieving revision 1.18
retrieving revision 1.19
diff -u -r1.18 -r1.19
--- audio_out.c 14 Nov 2001 13:07:08 -0000 1.18
+++ audio_out.c 24 Nov 2001 05:21:22 -0000 1.19
@@ -6,19 +6,12 @@
#include "afmt.h"
// there are some globals:
-int ao_samplerate=0;
-int ao_channels=0;
-int ao_format=0;
-int ao_bps=0;
-int ao_outburst=OUTBURST; // config.h default
-int ao_buffersize=-1;
-int ao_pts=0;
+ao_data_t ao_data={0,0,0,0,OUTBURST,-1,0};
char *ao_subdevice = NULL;
#ifdef USE_OSS_AUDIO
extern ao_functions_t audio_out_oss;
#endif
-//extern ao_functions_t audio_out_ossold;
extern ao_functions_t audio_out_null;
#ifdef HAVE_ALSA5
extern ao_functions_t audio_out_alsa5;
@@ -42,10 +35,9 @@
extern ao_functions_t audio_out_dxr3;
#endif
extern ao_functions_t audio_out_pcm;
-#ifndef USE_LIBVO2
extern ao_functions_t audio_out_mpegpes;
-#endif
extern ao_functions_t audio_out_pss;
+extern ao_functions_t audio_out_plugin;
ao_functions_t* audio_out_drivers[] =
{
@@ -75,9 +67,11 @@
&audio_out_dxr3,
#endif
&audio_out_pcm,
-#ifndef USE_LIBVO2
&audio_out_mpegpes,
-#endif
+ &audio_out_plugin,
// &audio_out_pss,
NULL
};
+
+
+
Index: audio_out.h
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/audio_out.h,v
retrieving revision 1.6
retrieving revision 1.7
diff -u -r1.6 -r1.7
--- audio_out.h 5 Nov 2001 03:01:11 -0000 1.6
+++ audio_out.h 24 Nov 2001 05:21:22 -0000 1.7
@@ -10,6 +10,7 @@
const char *comment;
} ao_info_t;
+/* interface towards mplayer and */
typedef struct ao_functions_s
{
ao_info_t *info;
@@ -19,25 +20,32 @@
void (*reset)();
int (*get_space)();
int (*play)(void* data,int len,int flags);
- int (*get_delay)();
+ float (*get_delay)();
void (*pause)();
void (*resume)();
} ao_functions_t;
+/* global data used by mplayer and plugins */
+typedef struct ao_data_s
+{
+ int samplerate;
+ int channels;
+ int format;
+ int bps;
+ int outburst;
+ int buffersize;
+ int pts;
+} ao_data_t;
+
+extern char *ao_subdevice;
+extern ao_data_t ao_data;
+
// prototypes
extern char *audio_out_format_name(int format);
// NULL terminated array of all drivers
extern ao_functions_t* audio_out_drivers[];
-extern int ao_samplerate;
-extern int ao_channels;
-extern int ao_format;
-extern int ao_bps;
-extern int ao_outburst;
-extern int ao_buffersize;
-extern int ao_pts;
-extern char *ao_subdevice;
#define CONTROL_OK 1
#define CONTROL_TRUE 1
Index: audio_out_internal.h
===================================================================
RCS file: /cvsroot/mplayer/main/libao2/audio_out_internal.h,v
retrieving revision 1.2
retrieving revision 1.3
diff -u -r1.2 -r1.3
--- audio_out_internal.h 5 Jun 2001 18:40:44 -0000 1.2
+++ audio_out_internal.h 24 Nov 2001 05:21:22 -0000 1.3
@@ -7,7 +7,7 @@
static void reset();
static int get_space();
static int play(void* data,int len,int flags);
-static int get_delay();
+static float get_delay();
static void audio_pause();
static void audio_resume();
More information about the MPlayer-cvslog
mailing list