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author | Warren Dukes <warren.dukes@gmail.com> | 2004-02-23 23:41:20 +0000 |
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committer | Warren Dukes <warren.dukes@gmail.com> | 2004-02-23 23:41:20 +0000 |
commit | d35747a40c7dea12de95f78a4f283a45ef687597 (patch) | |
tree | 3a85f138e68c7ef3e5a9ece863ed7266ad1cd7ce /src/audiofile_decode.c | |
download | mpd-d35747a40c7dea12de95f78a4f283a45ef687597.tar.gz mpd-d35747a40c7dea12de95f78a4f283a45ef687597.tar.xz mpd-d35747a40c7dea12de95f78a4f283a45ef687597.zip |
import from SF CVS
git-svn-id: https://svn.musicpd.org/mpd/trunk@1 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/audiofile_decode.c')
-rw-r--r-- | src/audiofile_decode.c | 138 |
1 files changed, 138 insertions, 0 deletions
diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c new file mode 100644 index 000000000..717237227 --- /dev/null +++ b/src/audiofile_decode.c @@ -0,0 +1,138 @@ +/* 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 + */ + +#ifdef HAVE_AUDIOFILE + +#include "audiofile_decode.h" + +#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 <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(Buffer * cb, AudioFormat * af, DecoderControl * dc) +{ + int fs, frame_count; + AFfilehandle af_fp; + + 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, &af->bits); + af->sampleRate = (int)afGetRate(af_fp, AF_DEFAULT_TRACK); + af->channels = afGetChannels(af_fp,AF_DEFAULT_TRACK); + + frame_count = afGetFrameCount(af_fp,AF_DEFAULT_TRACK); + + cb->totalTime = ((float)frame_count/(float)af->sampleRate); + + if (af->bits != 8 && af->bits != 16) { + ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n", + dc->file,af->bits); + afCloseFile(af_fp); + return -1; + } + + fs = (int)afGetFrameSize(af_fp, AF_DEFAULT_TRACK,1); + + dc->state = DECODE_STATE_DECODE; + dc->start = 0; + { + int ret, eof = 0, current = 0; + unsigned char chunk[CHUNK_SIZE]; + + while(!eof) { + if(dc->seek) { + cb->end = 0; + cb->wrap = 0; + current = dc->seekWhere * af->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 { + while(cb->begin==cb->end && cb->wrap && + !dc->stop && !dc->seek){ + usleep(1000); + } + if(dc->stop) break; + else if(dc->seek) continue; + +#ifdef WORDS_BIGENDIAN + pcm_changeBufferEndianness(chunk,CHUNK_SIZE, + af->bits); +#endif + memcpy(cb->chunks+cb->end*CHUNK_SIZE,chunk, + CHUNK_SIZE); + cb->chunkSize[cb->end] = CHUNK_SIZE; + + current += ret; + cb->times[cb->end] = (float)current/(float)af->sampleRate; + + ++cb->end; + + if(cb->end>=BUFFERED_CHUNKS) { + cb->end = 0; + cb->wrap = 1; + } + } + } + + 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 */ |