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-rw-r--r--src/inputPlugins/audiofile_plugin.c180
1 files changed, 180 insertions, 0 deletions
diff --git a/src/inputPlugins/audiofile_plugin.c b/src/inputPlugins/audiofile_plugin.c
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index 000000000..8c1089e1b
--- /dev/null
+++ b/src/inputPlugins/audiofile_plugin.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
+ */
+
+#include "../inputPlugin.h"
+
+#ifdef HAVE_AUDIOFILE
+
+#include "../utils.h"
+#include "../audio.h"
+#include "../log.h"
+#include "../pcm_utils.h"
+#include "../playerData.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;
+}
+
+MpdTag * audiofileTagDup(char * file) {
+ MpdTag * ret = NULL;
+ int time = getAudiofileTotalTime(file);
+
+ if (time>=0) {
+ if(!ret) ret = newMpdTag();
+ ret->time = time;
+ }
+
+ return ret;
+}
+
+char * audiofileSuffixes[] = {"wav", NULL};
+
+InputPlugin audiofilePlugin =
+{
+ "audiofile",
+ NULL,
+ audiofile_decode,
+ audiofileTagDup,
+ INPUT_PLUGIN_STREAM_FILE,
+ audiofileSuffixes,
+ NULL
+};
+
+#else
+
+InputPlugin audiofilePlugin =
+{
+ NULL,
+ NULL,
+ NULL,
+ NULL,
+ 0,
+ NULL,
+ NULL
+};
+
+#endif /* HAVE_AUDIOFILE */