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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 "../decoder_api.h"
#include "../log.h"
#include <sys/stat.h>
#include <audiofile.h>
/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
#define CHUNK_SIZE 1020
static int getAudiofileTotalTime(const 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 void
audiofile_decode(struct decoder *decoder, const char *path)
{
int fs, frame_count;
AFfilehandle af_fp;
int bits;
struct audio_format audio_format;
float total_time;
uint16_t bitRate;
struct stat st;
int ret, current = 0;
char chunk[CHUNK_SIZE];
if (stat(path, &st) < 0) {
ERROR("failed to stat: %s\n", path);
return;
}
af_fp = afOpenFile(path, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
ERROR("failed to open: %s\n", path);
return;
}
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, 16);
afGetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
audio_format.bits = (uint8_t)bits;
audio_format.sample_rate =
(unsigned int)afGetRate(af_fp, AF_DEFAULT_TRACK);
audio_format.channels =
(uint8_t)afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK);
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
total_time = ((float)frame_count / (float)audio_format.sample_rate);
bitRate = (uint16_t)(st.st_size * 8.0 / total_time / 1000.0 + 0.5);
if (audio_format.bits != 8 && audio_format.bits != 16) {
ERROR("Only 8 and 16-bit files are supported. %s is %i-bit\n",
path, audio_format.bits);
afCloseFile(af_fp);
return;
}
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, true, total_time);
do {
if (decoder_get_command(decoder) == DECODE_COMMAND_SEEK) {
current = decoder_seek_where(decoder) *
audio_format.sample_rate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
decoder_command_finished(decoder);
}
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
break;
current += ret;
decoder_data(decoder, NULL,
chunk, ret * fs,
(float)current / (float)audio_format.sample_rate,
bitRate, NULL);
} while (decoder_get_command(decoder) != DECODE_COMMAND_STOP);
afCloseFile(af_fp);
}
static struct tag *audiofileTagDup(const char *file)
{
struct tag *ret = NULL;
int total_time = getAudiofileTotalTime(file);
if (total_time >= 0) {
ret = tag_new();
ret->time = total_time;
} else {
DEBUG
("audiofileTagDup: Failed to get total song time from: %s\n",
file);
}
return ret;
}
static const char *const audiofileSuffixes[] = {
"wav", "au", "aiff", "aif", NULL
};
const struct decoder_plugin audiofilePlugin = {
.name = "audiofile",
.file_decode = audiofile_decode,
.tag_dup = audiofileTagDup,
.suffixes = audiofileSuffixes,
};
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