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/* the Music Player Daemon (MPD)
* Copyright (C) 2003-2007 by Warren Dukes (warren.dukes@gmail.com)
* 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>
static 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;
}
static int audiofile_decode(OutputBuffer * cb, DecoderControl * dc, char *path)
{
int fs, frame_count;
AFfilehandle af_fp;
int bits;
mpd_uint16 bitRate;
struct stat st;
if (stat(path, &st) < 0) {
ERROR("failed to stat: %s\n", path);
return -1;
}
af_fp = afOpenFile(path, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
ERROR("failed to open: %s\n", path);
return -1;
}
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
dc->audioFormat.bits = bits;
dc->audioFormat.sampleRate = afGetRate(af_fp, AF_DEFAULT_TRACK);
dc->audioFormat.channels = afGetVirtualChannels(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",
path, dc->audioFormat.bits);
afCloseFile(af_fp);
return -1;
}
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
dc->state = DECODE_STATE_DECODE;
{
int ret, eof = 0, current = 0;
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,
NULL);
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;
}
static MpdTag *audiofileTagDup(char *file)
{
MpdTag *ret = NULL;
int time = getAudiofileTotalTime(file);
if (time >= 0) {
if (!ret)
ret = newMpdTag();
ret->time = time;
} else {
DEBUG
("audiofileTagDup: Failed to get total song time from: %s\n",
file);
}
return ret;
}
static char *audiofileSuffixes[] = { "wav", "au", "aiff", "aif", NULL };
InputPlugin audiofilePlugin = {
"audiofile",
NULL,
NULL,
NULL,
NULL,
audiofile_decode,
audiofileTagDup,
INPUT_PLUGIN_STREAM_FILE,
audiofileSuffixes,
NULL
};
#else
InputPlugin audiofilePlugin;
#endif /* HAVE_AUDIOFILE */
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