/* 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 "../decoder_api.h"
#ifdef HAVE_AUDIOFILE
#include "../log.h"
#include <audiofile.h>
static int getAudiofileTotalTime(char *file)
{
int total_time;
AFfilehandle af_fp = afOpenFile(file, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
return -1;
}
total_time = (int)
((double)afGetFrameCount(af_fp, AF_DEFAULT_TRACK)
/ afGetRate(af_fp, AF_DEFAULT_TRACK));
afCloseFile(af_fp);
return total_time;
}
static int audiofile_decode(struct decoder * decoder, 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 = (mpd_uint8)bits;
dc.audioFormat.sampleRate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
dc.audioFormat.channels =
(mpd_uint8)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
getOutputAudioFormat(&(dc.audioFormat), &(ob.audioFormat));
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
dc.totalTime =
((float)frame_count / (float)dc.audioFormat.sampleRate);
bitRate = (mpd_uint16)(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);
decoder_initialized(decoder);
{
int ret, eof = 0, current = 0;
char chunk[CHUNK_SIZE];
while (!eof) {
if (dc.command == DECODE_COMMAND_SEEK) {
decoder_clear(decoder);
current = dc.seekWhere *
dc.audioFormat.sampleRate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
dc_command_finished();
}
ret =
afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
eof = 1;
else {
current += ret;
decoder_data(decoder, NULL,
1,
chunk, ret * fs,
(float)current /
(float)dc.audioFormat.
sampleRate, bitRate,
NULL);
if (dc.command == DECODE_COMMAND_STOP)
break;
}
}
decoder_flush(decoder);
}
afCloseFile(af_fp);
return 0;
}
static MpdTag *audiofileTagDup(char *file)
{
MpdTag *ret = NULL;
int total_time = getAudiofileTotalTime(file);
if (total_time >= 0) {
if (!ret)
ret = newMpdTag();
ret->time = total_time;
} else {
DEBUG
("audiofileTagDup: Failed to get total song time from: %s\n",
file);
}
return ret;
}
static const 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 */