mpd/src/decoder/audiofile_plugin.c
Avuton Olrich 0aee49bdf8 all: Update copyright header.
This updates the copyright header to all be the same, which is
pretty much an update of where to mail request for a copy of the GPL
and the years of the MPD project. This also puts all committers under
'The Music Player Project' umbrella. These entries should go
individually in the AUTHORS file, for consistancy.
2009-03-13 11:51:55 -07:00

222 lines
5.5 KiB
C

/*
* Copyright (C) 2003-2009 The Music Player Daemon Project
* http://www.musicpd.org
*
* 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.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "../decoder_api.h"
#include <audiofile.h>
#include <af_vfs.h>
#include <assert.h>
#include <glib.h>
#undef G_LOG_DOMAIN
#define G_LOG_DOMAIN "audiofile"
/* pick 1020 since its devisible for 8,16,24, and 32-bit audio */
#define CHUNK_SIZE 1020
static int audiofile_get_duration(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 ssize_t
audiofile_file_read(AFvirtualfile *vfile, void *data, size_t nbytes)
{
struct input_stream *is = (struct input_stream *) vfile->closure;
return input_stream_read(is, data, nbytes);
}
static long
audiofile_file_length(AFvirtualfile *vfile)
{
struct input_stream *is = (struct input_stream *) vfile->closure;
return is->size;
}
static long
audiofile_file_tell(AFvirtualfile *vfile)
{
struct input_stream *is = (struct input_stream *) vfile->closure;
return is->offset;
}
static void
audiofile_file_destroy(AFvirtualfile *vfile)
{
assert(vfile->closure != NULL);
vfile->closure = NULL;
}
static long
audiofile_file_seek(AFvirtualfile *vfile, long offset, int is_relative)
{
struct input_stream *is = (struct input_stream *) vfile->closure;
int whence = (is_relative ? SEEK_CUR : SEEK_SET);
if (input_stream_seek(is, offset, whence)) {
return is->offset;
} else {
return -1;
}
}
static AFvirtualfile *
setup_virtual_fops(struct input_stream *stream)
{
AFvirtualfile *vf = g_malloc(sizeof(AFvirtualfile));
vf->closure = stream;
vf->write = NULL;
vf->read = audiofile_file_read;
vf->length = audiofile_file_length;
vf->destroy = audiofile_file_destroy;
vf->seek = audiofile_file_seek;
vf->tell = audiofile_file_tell;
return vf;
}
static void
audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
{
AFvirtualfile *vf;
int fs, frame_count;
AFfilehandle af_fp;
int bits;
struct audio_format audio_format;
float total_time;
uint16_t bit_rate;
int ret, current = 0;
char chunk[CHUNK_SIZE];
enum decoder_command cmd;
if (!is->seekable) {
g_warning("not seekable");
return;
}
vf = setup_virtual_fops(is);
af_fp = afOpenVirtualFile(vf, "r", NULL);
if (af_fp == AF_NULL_FILEHANDLE) {
g_warning("failed to input stream\n");
return;
}
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
if (!audio_valid_sample_format(bits)) {
g_debug("input file has %d bit samples, converting to 16",
bits);
bits = 16;
}
afSetVirtualSampleFormat(af_fp, AF_DEFAULT_TRACK,
AF_SAMPFMT_TWOSCOMP, bits);
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);
if (!audio_format_valid(&audio_format)) {
g_warning("Invalid audio format: %u:%u:%u\n",
audio_format.sample_rate, audio_format.bits,
audio_format.channels);
afCloseFile(af_fp);
return;
}
frame_count = afGetFrameCount(af_fp, AF_DEFAULT_TRACK);
total_time = ((float)frame_count / (float)audio_format.sample_rate);
bit_rate = (uint16_t)(is->size * 8.0 / total_time / 1000.0 + 0.5);
fs = (int)afGetVirtualFrameSize(af_fp, AF_DEFAULT_TRACK, 1);
decoder_initialized(decoder, &audio_format, true, total_time);
do {
ret = afReadFrames(af_fp, AF_DEFAULT_TRACK, chunk,
CHUNK_SIZE / fs);
if (ret <= 0)
break;
current += ret;
cmd = decoder_data(decoder, NULL,
chunk, ret * fs,
(float)current /
(float)audio_format.sample_rate,
bit_rate, NULL);
if (cmd == DECODE_COMMAND_SEEK) {
current = decoder_seek_where(decoder) *
audio_format.sample_rate;
afSeekFrame(af_fp, AF_DEFAULT_TRACK, current);
decoder_command_finished(decoder);
cmd = DECODE_COMMAND_NONE;
}
} while (cmd == DECODE_COMMAND_NONE);
afCloseFile(af_fp);
}
static struct tag *audiofile_tag_dup(const char *file)
{
struct tag *ret = NULL;
int total_time = audiofile_get_duration(file);
if (total_time >= 0) {
ret = tag_new();
ret->time = total_time;
} else {
g_debug("Failed to get total song time from: %s\n",
file);
}
return ret;
}
static const char *const audiofile_suffixes[] = {
"wav", "au", "aiff", "aif", NULL
};
static const char *const audiofile_mime_types[] = {
"audio/x-wav",
"audio/x-aiff",
NULL
};
const struct decoder_plugin audiofilePlugin = {
.name = "audiofile",
.stream_decode = audiofile_stream_decode,
.tag_dup = audiofile_tag_dup,
.suffixes = audiofile_suffixes,
.mime_types = audiofile_mime_types,
};