ao_mpegpes.c

自己移植的linux下的流媒体播放器原代码,支持mms协议,支持ftp和http协议.

#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)( 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;
}