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author | Warren Dukes <warren.dukes@gmail.com> | 2004-05-31 02:31:55 +0000 |
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committer | Warren Dukes <warren.dukes@gmail.com> | 2004-05-31 02:31:55 +0000 |
commit | 5d392c70cbea09d81e6e5bb7f0a0bd075fcd6f8d (patch) | |
tree | 214e7ad2ff7d9cc351357a341239d2d190fc9654 /src/audiofile_decode.c | |
parent | 97fe75a0bf4ce5a0769a7509f758eda3f52fd6b3 (diff) | |
download | mpd-5d392c70cbea09d81e6e5bb7f0a0bd075fcd6f8d.tar.gz mpd-5d392c70cbea09d81e6e5bb7f0a0bd075fcd6f8d.tar.xz mpd-5d392c70cbea09d81e6e5bb7f0a0bd075fcd6f8d.zip |
audiofile_plugin
git-svn-id: https://svn.musicpd.org/mpd/trunk@1248 09075e82-0dd4-0310-85a5-a0d7c8717e4f
Diffstat (limited to 'src/audiofile_decode.c')
-rw-r--r-- | src/audiofile_decode.c | 143 |
1 files changed, 0 insertions, 143 deletions
diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c deleted file mode 100644 index 8e6c779b9..000000000 --- a/src/audiofile_decode.c +++ /dev/null @@ -1,143 +0,0 @@ -/* 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, - NULL, - dc, - 1, - chunk, - ret*fs, - (float)current / - (float)dc->audioFormat.sampleRate, - bitRate); - if(dc->stop) break; - } - } - - flushOutputBuffer(cb); - - /*if(dc->seek) { - dc->seekError = 1; - 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: */ |