Mercurial > mplayer.hg
view libmpcodecs/ad_libvorbis.c @ 9695:68c44d3530c8
Improved xscreensaver support.
With this we no longer have to kill xscreensaver, but just send it a message
periodically.
author | ranma |
---|---|
date | Wed, 26 Mar 2003 17:02:28 +0000 |
parents | f49a2bf04229 |
children | e94036364011 |
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#include <stdio.h> #include <stdlib.h> #include <unistd.h> #include <stdarg.h> #include <math.h> #include "config.h" #include "ad_internal.h" #ifdef USE_SETLOCALE #include <locale.h> #endif #ifdef HAVE_OGGVORBIS static ad_info_t info = { "Ogg/Vorbis audio decoder", "libvorbis", "Felix Buenemann, A'rpi", "libvorbis", "" }; LIBAD_EXTERN(libvorbis) #ifdef TREMOR #include <tremor/ivorbiscodec.h> #else #include <vorbis/codec.h> #endif // This struct is also defined in demux_ogg.c => common header ? typedef struct ov_struct_st { vorbis_info vi; /* struct that stores all the static vorbis bitstream settings */ vorbis_comment vc; /* struct that stores all the bitstream user comments */ vorbis_dsp_state vd; /* central working state for the packet->PCM decoder */ vorbis_block vb; /* local working space for packet->PCM decode */ float rg_scale; /* replaygain scale */ #ifdef TREMOR int rg_scale_int; #endif } ov_struct_t; static int read_vorbis_comment( char* ptr, char* comment, char* format, ... ) { va_list va; int clen, ret; va_start( va, format ); clen = strlen( comment ); #ifdef USE_SETLOCALE setlocale( LC_NUMERIC, "C" ); #endif ret = strncasecmp( ptr, comment, clen) == 0 ? vsscanf( ptr+clen, format, va ) : 0; #ifdef USE_SETLOCALE setlocale( LC_NUMERIC, "" ); #endif va_end( va ); return ret; } static int preinit(sh_audio_t *sh) { sh->audio_out_minsize=1024*4; // 1024 samples/frame return 1; } static int init(sh_audio_t *sh) { ogg_packet op; vorbis_comment vc; struct ov_struct_st *ov; #define ERROR() { \ vorbis_comment_clear(&vc); \ vorbis_info_clear(&ov->vi); \ free(ov); \ return 0; \ } /// Init the decoder with the 3 header packets ov = (struct ov_struct_st*)malloc(sizeof(struct ov_struct_st)); vorbis_info_init(&ov->vi); vorbis_comment_init(&vc); op.bytes = ds_get_packet(sh->ds,&op.packet); op.b_o_s = 1; /// Header if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: initial (identification) header broken!\n"); ERROR(); } op.bytes = ds_get_packet(sh->ds,&op.packet); op.b_o_s = 0; /// Comments if(vorbis_synthesis_headerin(&ov->vi,&vc,&op) <0) { mp_msg(MSGT_DECAUDIO,MSGL_ERR,"OggVorbis: comment header broken!\n"); ERROR(); } op.bytes = ds_get_packet(sh->ds,&op.packet); //// Codebook if(vorbis_synthesis_headerin(&ov->vi,&vc,&op)<0) { mp_msg(MSGT_DECAUDIO,MSGL_WARN,"OggVorbis: codebook header broken!\n"); ERROR(); } else { /// Print the infos float rg_gain=0.f, rg_peak=0.f; char **ptr=vc.user_comments; while(*ptr){ mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbisComment: %s\n",*ptr); /* replaygain */ read_vorbis_comment( *ptr, "replaygain_album_gain=", "%f", &rg_gain ); read_vorbis_comment( *ptr, "rg_audiophile=", "%f", &rg_gain ); if( !rg_gain ) { read_vorbis_comment( *ptr, "replaygain_track_gain=", "%f", &rg_gain ); read_vorbis_comment( *ptr, "rg_radio=", "%f", &rg_gain ); } read_vorbis_comment( *ptr, "replaygain_album_peak=", "%f", &rg_peak ); if( !rg_peak ) { read_vorbis_comment( *ptr, "replaygain_track_peak=", "%f", &rg_peak ); read_vorbis_comment( *ptr, "rg_peak=", "%f", &rg_peak ); } ++ptr; } /* replaygain: scale */ if(!rg_gain) ov->rg_scale = 1.f; /* just in case pow() isn't standard-conformant */ else ov->rg_scale = pow(10.f, rg_gain/20); /* replaygain: anticlip */ if(ov->rg_scale * rg_peak > 1.f) ov->rg_scale = 1.f / rg_peak; /* replaygain: security */ if(ov->rg_scale > 15.) ov->rg_scale = 15.; #ifdef TREMOR ov->rg_scale_int = (int)(ov->rg_scale*64.f); #endif mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Bitstream is %d channel%s, %dHz, %dbit/s %cBR\n",(int)ov->vi.channels,ov->vi.channels>1?"s":"",(int)ov->vi.rate,(int)ov->vi.bitrate_nominal, (ov->vi.bitrate_lower!=ov->vi.bitrate_nominal)||(ov->vi.bitrate_upper!=ov->vi.bitrate_nominal)?'V':'C'); if(rg_gain || rg_peak) mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Gain = %+.2f dB, Peak = %.4f, Scale = %.2f\n", rg_gain, rg_peak, ov->rg_scale); mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Encoded by: %s\n",vc.vendor); } vorbis_comment_clear(&vc); // printf("lower=%d upper=%d \n",(int)ov->vi.bitrate_lower,(int)ov->vi.bitrate_upper); // Setup the decoder sh->channels=ov->vi.channels; sh->samplerate=ov->vi.rate; // assume 128kbit if bitrate not specified in the header sh->i_bps=((ov->vi.bitrate_nominal>0) ? ov->vi.bitrate_nominal : 128000)/8; sh->context = ov; /// Finish the decoder init vorbis_synthesis_init(&ov->vd,&ov->vi); vorbis_block_init(&ov->vd,&ov->vb); mp_msg(MSGT_DECAUDIO,MSGL_V,"OggVorbis: Init OK!\n"); return 1; } static void uninit(sh_audio_t *sh) { struct ov_struct_st *ov = sh->context; vorbis_block_clear(&ov->vb); vorbis_info_clear(&ov->vi); free(ov); } static int control(sh_audio_t *sh,int cmd,void* arg, ...) { switch(cmd) { #if 0 case ADCTRL_RESYNC_STREAM: return CONTROL_TRUE; case ADCTRL_SKIP_FRAME: return CONTROL_TRUE; #endif } return CONTROL_UNKNOWN; } static int decode_audio(sh_audio_t *sh,unsigned char *buf,int minlen,int maxlen) { int len = 0; int samples; float **pcm; float scale; struct ov_struct_st *ov = sh->context; while(len < minlen) { while((samples=vorbis_synthesis_pcmout(&ov->vd,&pcm))<=0){ ogg_packet op; memset(&op,0,sizeof(op)); //op.b_o_s = op.e_o_s = 0; op.bytes = ds_get_packet(sh->ds,&op.packet); if(op.bytes<=0) break; if(vorbis_synthesis(&ov->vb,&op)==0) /* test for success! */ vorbis_synthesis_blockin(&ov->vd,&ov->vb); } if(samples<=0) break; // error/EOF while(samples>0){ int i,j; int clipflag=0; int convsize=(maxlen-len)/(2*ov->vi.channels); // max size! int bout=((samples<convsize)?samples:convsize); if(bout<=0) break; // no buffer space /* convert floats to 16 bit signed ints (host order) and interleave */ #ifdef TREMOR if (ov->rg_scale_int == 64) { for(i=0;i<ov->vi.channels;i++){ ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); ogg_int16_t *ptr=convbuffer+i; ogg_int32_t *mono=pcm[i]; for(j=0;j<bout;j++){ int val=mono[j]>>9; /* might as well guard against clipping */ if(val>32767){ val=32767; clipflag=1; } if(val<-32768){ val=-32768; clipflag=1; } *ptr=val; ptr+=ov->vi.channels; } } } else #endif /* TREMOR */ { #ifndef TREMOR scale = 32767.f * ov->rg_scale; #endif for(i=0;i<ov->vi.channels;i++){ ogg_int16_t *convbuffer=(ogg_int16_t *)(&buf[len]); ogg_int16_t *ptr=convbuffer+i; #ifdef TREMOR ogg_int32_t *mono=pcm[i]; for(j=0;j<bout;j++){ int val=(mono[j]*ov->rg_scale_int)>>(9+6); #else float *mono=pcm[i]; for(j=0;j<bout;j++){ int val=mono[j]*scale; /* might as well guard against clipping */ if(val>32767){ val=32767; clipflag=1; } if(val<-32768){ val=-32768; clipflag=1; } #endif /* TREMOR */ *ptr=val; ptr+=ov->vi.channels; } } } if(clipflag) mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"Clipping in frame %ld\n",(long)(ov->vd.sequence)); len+=2*ov->vi.channels*bout; mp_msg(MSGT_DECAUDIO,MSGL_DBG2,"\n[decoded: %d / %d ]\n",bout,samples); samples-=bout; vorbis_synthesis_read(&ov->vd,bout); /* tell libvorbis how many samples we actually consumed */ } //while(samples>0) // if (!samples) break; // why? how? } return len; } #endif /* !HAVE_OGGVORBIS */