/*
* 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 "config.h"
#include "decoder_api.h"
#include "audio_check.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 length)
{
struct input_stream *is = (struct input_stream *) vfile->closure;
GError *error = NULL;
size_t nbytes;
nbytes = input_stream_read(is, data, length, &error);
if (nbytes == 0 && error != NULL) {
g_warning("%s", error->message);
g_error_free(error);
return -1;
}
return 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, NULL)) {
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 enum sample_format
audiofile_bits_to_sample_format(int bits)
{
switch (bits) {
case 8:
return SAMPLE_FORMAT_S8;
case 16:
return SAMPLE_FORMAT_S16;
case 24:
return SAMPLE_FORMAT_S24_P32;
case 32:
return SAMPLE_FORMAT_S32;
}
return SAMPLE_FORMAT_UNDEFINED;
}
static enum sample_format
audiofile_setup_sample_format(AFfilehandle af_fp)
{
int fs, bits;
afGetSampleFormat(af_fp, AF_DEFAULT_TRACK, &fs, &bits);
if (!audio_valid_sample_format(audiofile_bits_to_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);
return audiofile_bits_to_sample_format(bits);
}
static void
audiofile_stream_decode(struct decoder *decoder, struct input_stream *is)
{
GError *error = NULL;
AFvirtualfile *vf;
int fs, frame_count;
AFfilehandle af_fp;
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;
}
if (!audio_format_init_checked(&audio_format,
afGetRate(af_fp, AF_DEFAULT_TRACK),
audiofile_setup_sample_format(af_fp),
afGetVirtualChannels(af_fp, AF_DEFAULT_TRACK),
&error)) {
g_warning("%s", error->message);
g_error_free(error);
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 audiofile_decoder_plugin = {
.name = "audiofile",
.stream_decode = audiofile_stream_decode,
.tag_dup = audiofile_tag_dup,
.suffixes = audiofile_suffixes,
.mime_types = audiofile_mime_types,
};