Mercurial > mplayer.hg
view libao2/ao_mpegpes.c @ 13784:7d3b84ddd2fd
Remove hardcoded filenames in favor of command line parameters, some error
checking added, patch by Reza Jelveh.
author | diego |
---|---|
date | Thu, 28 Oct 2004 01:02:24 +0000 |
parents | c1955840883d |
children | a92101a7eb49 |
line wrap: on
line source
#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 "afmt.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 AFMT_S16_LE: case AFMT_S16_BE: case AFMT_MPEG: case AFMT_AC3: ao_data.format=format; break; default: ao_data.format=AFMT_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==AFMT_MPEG) 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==AFMT_S16_LE || ao_data.format==AFMT_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; }