/*
* This file is part of mpv.
*
* mpv 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.
*
* mpv 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 mpv. If not, see .
*/
#include
#include
#include "audio/format.h"
#include "af.h"
#include "osdep/endian.h"
static bool test_conversion(int src_format, int dst_format)
{
return (src_format == AF_FORMAT_U24 && dst_format == AF_FORMAT_U32) ||
(src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
(src_format == AF_FORMAT_U32 && dst_format == AF_FORMAT_U24) ||
(src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24);
}
static int control(struct af_instance *af, int cmd, void *arg)
{
switch (cmd) {
case AF_CONTROL_REINIT: {
struct mp_audio *in = arg;
struct mp_audio orig_in = *in;
struct mp_audio *out = af->data;
if (!test_conversion(in->format, out->format))
return AF_DETACH;
if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_24BIT) {
mp_audio_set_format(out, af_fmt_change_bits(in->format, 32));
} else if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_32BIT) {
mp_audio_set_format(out, af_fmt_change_bits(in->format, 24));
} else {
abort();
}
out->rate = in->rate;
mp_audio_set_channels(out, &in->channels);
assert(test_conversion(in->format, out->format));
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_SET_FORMAT: {
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
}
return AF_UNKNOWN;
}
// The LSB is always ignored.
#if BYTE_ORDER == BIG_ENDIAN
#define SHIFT(x) ((3-(x))*8)
#else
#define SHIFT(x) (((x)+1)*8)
#endif
static int filter(struct af_instance *af, struct mp_audio *data, int flags)
{
mp_audio_realloc_min(af->data, data->samples);
struct mp_audio *out = af->data;
size_t len = mp_audio_psize(data) / data->bps;
if (data->bps == 4) {
for (int s = 0; s < len; s++) {
uint32_t val = *((uint32_t *)data->planes[0] + s);
uint8_t *ptr = (uint8_t *)out->planes[0] + s * 3;
ptr[0] = val >> SHIFT(0);
ptr[1] = val >> SHIFT(1);
ptr[2] = val >> SHIFT(2);
}
mp_audio_set_format(data, af_fmt_change_bits(data->format, 24));
} else {
for (int s = 0; s < len; s++) {
uint8_t *ptr = (uint8_t *)data->planes[0] + s * 3;
uint32_t val = ptr[0] << SHIFT(0)
| ptr[1] << SHIFT(1)
| ptr[2] << SHIFT(2);
*((uint32_t *)out->planes[0] + s) = val;
}
mp_audio_set_format(data, af_fmt_change_bits(data->format, 32));
}
data->planes[0] = out->planes[0];
return 0;
}
static int af_open(struct af_instance *af)
{
af->control = control;
af->filter = filter;
return AF_OK;
}
const struct af_info af_info_convert24 = {
.info = "Convert between 24 and 32 bit sample format",
.name = "convert24",
.open = af_open,
.test_conversion = test_conversion,
};