1
mirror of https://github.com/mpv-player/mpv synced 2024-07-15 22:21:38 +02:00
mpv/audio/filter/af_convertsignendian.c

132 lines
3.7 KiB
C
Raw Normal View History

audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
/*
* 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 <http://www.gnu.org/licenses/>.
*/
#include <stdlib.h>
#include <assert.h>
#include "af.h"
#include "audio/format.h"
#include "compat/mpbswap.h"
static bool test_conversion(int src_format, int dst_format)
{
if ((src_format & AF_FORMAT_PLANAR) ||
(dst_format & AF_FORMAT_PLANAR))
return false;
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
int src_noend = src_format & ~AF_FORMAT_END_MASK;
int dst_noend = dst_format & ~AF_FORMAT_END_MASK;
// We can swap endian for all formats, but sign only for integer formats.
if (src_noend == dst_noend)
return true;
if (((src_noend & ~AF_FORMAT_SIGN_MASK) ==
(dst_noend & ~AF_FORMAT_SIGN_MASK)) &&
((src_noend & AF_FORMAT_POINT_MASK) == AF_FORMAT_I))
return true;
return false;
}
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;
out->rate = in->rate;
mp_audio_set_channels(out, &in->channels);
return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
}
case AF_CONTROL_SET_FORMAT: {
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
mp_audio_set_format(af->data, *(int*)arg);
return AF_OK;
}
}
return AF_UNKNOWN;
}
static void endian(void *data, int len, int bps)
{
switch (bps) {
case 2:
for (int i = 0; i < len; i++) {
((uint16_t*)data)[i] = bswap_16(((uint16_t *)data)[i]);
}
break;
case 3:
for(int i = 0; i < len; i++) {
uint8_t s = ((uint8_t *)data)[3 * i];
((uint8_t *)data)[3 * i] = ((uint8_t *)data)[3 * i + 2];
((uint8_t *)data)[3 * i + 2] = s;
}
break;
case 4:
for(int i = 0; i < len; i++) {
((uint32_t*)data)[i] = bswap_32(((uint32_t *)data)[i]);
}
break;
}
}
static void si2us(void *data, int len, int bps, bool le)
{
ptrdiff_t i = -(len * bps);
uint8_t *p = &((uint8_t *)data)[len * bps];
if (le && bps > 1)
p += bps - 1;
if (len <= 0)
return;
do {
p[i] ^= 0x80;
} while (i += bps);
}
static int filter(struct af_instance *af, struct mp_audio *data, int flags)
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
{
int infmt = data->format;
int outfmt = af->data->format;
size_t len = data->samples * data->nch;
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
if ((infmt & AF_FORMAT_END_MASK) != (outfmt & AF_FORMAT_END_MASK))
endian(data->planes[0], len, data->bps);
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK))
si2us(data->planes[0], len, data->bps,
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
(outfmt & AF_FORMAT_END_MASK) == AF_FORMAT_LE);
mp_audio_set_format(data, outfmt);
return 0;
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
}
static int af_open(struct af_instance *af)
{
af->control = control;
af->filter = filter;
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
return AF_OK;
}
struct af_info af_info_convertsignendian = {
.info = "Convert between sample format sign/endian",
.name = "convertsignendian",
.open = af_open,
audio/filter: split af_format into separate filters, rename af_force af_format is the old audio conversion filter. It could do all possible conversions supported by the audio chain. However, ever since the addition of af_lavrresample, most conversions are done by libav/swresample, and af_format is used as fallback. Separate out the fallback cases and remove af_format. af_convert24 does 24 bit <-> 32 bit conversions, while af_convertsignendian does sign and endian conversions. Maybe the way the conversions are split sounds a bit odd. But the former changes the size of the audio data, while the latter is fully in-place, so there's at least different buffer management. This requires a quite complicated algorithm to make sure all these "partial" conversion filters can actually get from one format to another. E.g. s24le->s32be always requires convertsignendian and convert24, but af.c has no idea what the intermediate format should be. So I added a graph search (trying every possible format and filter) to determine required format and filter. When I wrote this, it seemed this was still better than messing everything into af_lavrresample, but maybe this is overkill and I'll change my opinion. For now, it seems nice to get rid of af_format though. The AC3->IEC61937 conversion isn't supported anymore, but I don't think this is needed anywhere. Most AOs test all formats explicitly, or use the AF_FORMAT_IS_IEC61937() macro (which includes AC3). One positive consequence of this change is that conversions always include dithering (done by libav/swresample), instead of possibly going through af_format, which doesn't do anything fancy. Rename af_force to af_format. It's essentially compatible with command line uses of af_format. We retain a compatibility alias for af_force.
2013-10-22 01:20:43 +02:00
.test_conversion = test_conversion,
};