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-rw-r--r--src/audiofile_decode.c138
1 files changed, 138 insertions, 0 deletions
diff --git a/src/audiofile_decode.c b/src/audiofile_decode.c
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+/* 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 */