/* the Music Player Daemon (MPD) * (c)2003-2004 by Warren Dukes (shank@mercury.chem.pitt.edu) * This project's homepage is: http://www.musicpd.org * * libaudiofile (wave) support added by Eric Wong <normalperson@yhbt.net> * * This program is free software; you can redistribute it and/or modify * it under the terms of the GNU General Public License as published by * the Free Software Foundation; either version 2 of the License, or * (at your option) any later version. * * This program is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the * GNU General Public License for more details. * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include "audiofile_decode.h" #ifdef HAVE_AUDIOFILE #include "command.h" #include "utils.h" #include "audio.h" #include "log.h" #include "pcm_utils.h" #include <stdio.h> #include <unistd.h> #include <stdlib.h> #include <string.h> #include <sys/types.h> #include <sys/stat.h> #include <unistd.h> #include <audiofile.h> int getAudiofileTotalTime(char * file) { int time; AFfilehandle af_fp = afOpenFile(file, "r", NULL); if(af_fp == AF_NULL_FILEHANDLE) { return -1; } time = (int) ((double)afGetFrameCount(af_fp,AF_DEFAULT_TRACK) /afGetRate(af_fp,AF_DEFAULT_TRACK)); afCloseFile(af_fp); return time; } int audiofile_decode(OutputBuffer * cb, DecoderControl * dc) { int fs, frame_count; AFfilehandle af_fp; int bits; mpd_uint16 bitRate; struct stat st; if(stat(dc->file,&st) < 0) { ERROR("failed to stat: %s\n",dc->file); return -1; } af_fp = afOpenFile(dc->file,"r", NULL); if(af_fp == AF_NULL_FILEHANDLE) { ERROR("failed to open: %s\n",dc->file); return -1; } afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits); dc->audioFormat.bits = bits; dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK); dc->audioFormat.channels = afGetChannels(af_fp,AF_DEFAULT_TRACK); getOutputAudioFormat(&(dc->audioFormat),&(cb->audioFormat)); frame_count = afGetFrameCount(af_fp,AF_DEFAULT_TRACK); dc->totalTime = ((float)frame_count/(float)dc->audioFormat.sampleRate); bitRate = st.st_size*8.0/dc->totalTime/1000.0+0.5; if (dc->audioFormat.bits != 8 && dc->audioFormat.bits != 16) { ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n", dc->file,dc->audioFormat.bits); afCloseFile(af_fp); return -1; } fs = (int)afGetFrameSize(af_fp, AF_DEFAULT_TRACK,1); dc->state = DECODE_STATE_DECODE; { int ret, eof = 0, current = 0; unsigned char chunk[CHUNK_SIZE]; while(!eof) { if(dc->seek) { clearOutputBuffer(cb); current = dc->seekWhere * dc->audioFormat.sampleRate; afSeekFrame(af_fp, AF_DEFAULT_TRACK,current); dc->seek = 0; } ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk, CHUNK_SIZE/fs); if(ret<=0) eof = 1; else { current += ret; sendDataToOutputBuffer(cb,dc,chunk,ret*fs, (float)current / (float)dc->audioFormat.sampleRate, bitRate); if(dc->stop) break; else if(dc->seek) continue; } } flushOutputBuffer(cb); if(dc->seek) dc->seek = 0; if(dc->stop) { dc->state = DECODE_STATE_STOP; dc->stop = 0; } else dc->state = DECODE_STATE_STOP; } afCloseFile(af_fp); return 0; } #endif /* HAVE_AUDIOFILE */ /* vim:set shiftwidth=4 tabstop=8 expandtab: */