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mpv/libao2/ao_mpegpes.c
alex 507121f7fe removing AFMT_ dependancy
git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@14246 b3059339-0415-0410-9bf9-f77b7e298cf2
2004-12-27 17:30:15 +00:00

196 lines
4.4 KiB
C

#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include "config.h"
#ifdef HAVE_DVB_HEAD
#define HAVE_DVB 1
#endif
#ifdef HAVE_DVB
#include <sys/ioctl.h>
#endif
#include "audio_out.h"
#include "audio_out_internal.h"
#include "libaf/af_format.h"
#include "mp_msg.h"
#include "help_mp.h"
#ifdef HAVE_DVB
#ifndef HAVE_DVB_HEAD
#include <ost/audio.h>
audioMixer_t dvb_mixer={255,255};
#else
#include <linux/dvb/audio.h>
audio_mixer_t dvb_mixer={255,255};
#endif
#endif
extern int vo_mpegpes_fd;
extern int vo_mpegpes_fd2;
#include <errno.h>
static ao_info_t info =
{
#ifdef HAVE_DVB
"DVB audio output",
#else
"Mpeg-PES audio output",
#endif
"mpegpes",
"A'rpi",
""
};
LIBAO_EXTERN(mpegpes)
// to set/get/query special features/parameters
static int control(int cmd,void *arg){
#ifdef HAVE_DVB
switch(cmd){
case AOCONTROL_GET_VOLUME:
if(vo_mpegpes_fd2>=0){
((ao_control_vol_t*)(arg))->left=dvb_mixer.volume_left/2.56;
((ao_control_vol_t*)(arg))->right=dvb_mixer.volume_right/2.56;
return CONTROL_OK;
}
return CONTROL_ERROR;
case AOCONTROL_SET_VOLUME:
if(vo_mpegpes_fd2>=0){
dvb_mixer.volume_left=((ao_control_vol_t*)(arg))->left*2.56;
dvb_mixer.volume_right=((ao_control_vol_t*)(arg))->right*2.56;
if(dvb_mixer.volume_left>255) dvb_mixer.volume_left=255;
if(dvb_mixer.volume_right>255) dvb_mixer.volume_right=255;
// printf("Setting DVB volume: %d ; %d \n",dvb_mixer.volume_left,dvb_mixer.volume_right);
if ( (ioctl(vo_mpegpes_fd2,AUDIO_SET_MIXER, &dvb_mixer) < 0)){
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_CantSetMixer,
strerror(errno));
return CONTROL_ERROR;
}
return CONTROL_OK;
}
return CONTROL_ERROR;
}
#endif
return CONTROL_UNKNOWN;
}
static int freq=0;
static int freq_id=0;
// open & setup audio device
// return: 1=success 0=fail
static int init(int rate,int channels,int format,int flags){
#ifdef HAVE_DVB
if(vo_mpegpes_fd2<0) return 0; // couldn't open audio dev
#else
if(vo_mpegpes_fd<0) return 0; // no file
#endif
ao_data.channels=2;
ao_data.outburst=2000;
switch(format){
case AF_FORMAT_S16_LE:
case AF_FORMAT_S16_BE:
case AF_FORMAT_MPEG2:
case AF_FORMAT_AC3:
ao_data.format=format;
break;
default:
ao_data.format=AF_FORMAT_S16_BE;
}
retry:
switch(rate){
case 48000: freq_id=0;break;
case 96000: freq_id=1;break;
case 44100: freq_id=2;break;
case 32000: freq_id=3;break;
default:
mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_MPEGPES_UnsupSamplerate, rate);
#if 0
if(rate>48000) rate=96000; else
if(rate>44100) rate=48000; else
if(rate>32000) rate=44100; else
rate=32000;
goto retry;
#else
rate=48000; freq_id=0;
#endif
}
ao_data.bps=rate*2*2;
freq=ao_data.samplerate=rate;
return 1;
}
// close audio device
static void uninit(int immed){
}
// stop playing and empty buffers (for seeking/pause)
static void reset(){
}
// stop playing, keep buffers (for pause)
static void audio_pause()
{
// for now, just call reset();
reset();
}
// resume playing, after audio_pause()
static void audio_resume()
{
}
void send_pes_packet(unsigned char* data,int len,int id,int timestamp);
void send_lpcm_packet(unsigned char* data,int len,int id,int timestamp,int freq_id);
extern int vo_pts;
// return: how many bytes can be played without blocking
static int get_space(){
float x=(float)(vo_pts-ao_data.pts)/90000.0;
int y;
// printf("vo_pts: %5.3f ao_pts: %5.3f\n",vo_pts/90000.0,ao_data.pts/90000.0);
if(x<=0) return 0;
y=freq*4*x;y/=ao_data.outburst;y*=ao_data.outburst;
if(y>32000) y=32000;
// printf("diff: %5.3f -> %d \n",x,y);
return y;
}
// 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){
// printf("\nao_mpegpes: play(%d) freq=%d\n",len,freq_id);
if(ao_data.format==AF_FORMAT_MPEG2)
send_pes_packet(data,len,0x1C0,ao_data.pts);
else {
int i;
unsigned short *s=data;
// if(len>2000) len=2000;
// printf("ao_mpegpes: len=%d \n",len);
if(ao_data.format==AF_FORMAT_S16_LE || ao_data.format==AF_FORMAT_AC3)
for(i=0;i<len/2;i++) s[i]=(s[i]>>8)|(s[i]<<8); // le<->be
send_lpcm_packet(data,len,0xA0,ao_data.pts,freq_id);
}
return len;
}
// return: delay in seconds between first and last sample in buffer
static float get_delay(){
return 0.0;
}