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
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/*
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* This file is part of mpv.
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*
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* mpv is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* mpv is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with mpv. If not, see <http://www.gnu.org/licenses/>.
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*/
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#include <stdlib.h>
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#include <assert.h>
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#include "af.h"
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#include "audio/format.h"
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#include "compat/mpbswap.h"
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static bool test_conversion(int src_format, int dst_format)
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{
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int src_noend = src_format & ~AF_FORMAT_END_MASK;
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int dst_noend = dst_format & ~AF_FORMAT_END_MASK;
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// We can swap endian for all formats, but sign only for integer formats.
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if (src_noend == dst_noend)
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return true;
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if (((src_noend & ~AF_FORMAT_SIGN_MASK) ==
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(dst_noend & ~AF_FORMAT_SIGN_MASK)) &&
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((src_noend & AF_FORMAT_POINT_MASK) == AF_FORMAT_I))
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return true;
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return false;
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}
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static int control(struct af_instance *af, int cmd, void *arg)
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{
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switch (cmd) {
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case AF_CONTROL_REINIT: {
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struct mp_audio *in = arg;
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struct mp_audio orig_in = *in;
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struct mp_audio *out = af->data;
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if (!test_conversion(in->format, out->format))
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return AF_DETACH;
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out->rate = in->rate;
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mp_audio_set_channels(out, &in->channels);
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return mp_audio_config_equals(in, &orig_in) ? AF_OK : AF_FALSE;
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}
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case AF_CONTROL_FORMAT_FMT | AF_CONTROL_SET: {
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mp_audio_set_format(af->data, *(int*)arg);
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return AF_OK;
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}
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}
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return AF_UNKNOWN;
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}
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static void endian(void *data, int len, int bps)
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{
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switch (bps) {
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case 2:
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for (int i = 0; i < len; i++) {
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((uint16_t*)data)[i] = bswap_16(((uint16_t *)data)[i]);
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}
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break;
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case 3:
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for(int i = 0; i < len; i++) {
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uint8_t s = ((uint8_t *)data)[3 * i];
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((uint8_t *)data)[3 * i] = ((uint8_t *)data)[3 * i + 2];
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((uint8_t *)data)[3 * i + 2] = s;
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}
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break;
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case 4:
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for(int i = 0; i < len; i++) {
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((uint32_t*)data)[i] = bswap_32(((uint32_t *)data)[i]);
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}
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break;
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}
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}
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static void si2us(void *data, int len, int bps, bool le)
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{
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ptrdiff_t i = -(len * bps);
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uint8_t *p = &((uint8_t *)data)[len * bps];
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if (le && bps > 1)
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p += bps - 1;
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if (len <= 0)
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return;
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do {
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p[i] ^= 0x80;
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} while (i += bps);
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}
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static struct mp_audio *play(struct af_instance *af, struct mp_audio *data)
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{
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int infmt = data->format;
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int outfmt = af->data->format;
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size_t len = data->len / data->bps;
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if ((infmt & AF_FORMAT_END_MASK) != (outfmt & AF_FORMAT_END_MASK))
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endian(data->audio, len, data->bps);
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if ((infmt & AF_FORMAT_SIGN_MASK) != (outfmt & AF_FORMAT_SIGN_MASK))
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si2us(data->audio, len, data->bps,
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(outfmt & AF_FORMAT_END_MASK) == AF_FORMAT_LE);
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mp_audio_set_format(data, outfmt);
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return data;
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}
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static int af_open(struct af_instance *af)
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{
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af->control = control;
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af->play = play;
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af->mul = 1;
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af->data = talloc_zero(af, struct mp_audio);
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return AF_OK;
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}
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struct af_info af_info_convertsignendian = {
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2013-10-23 19:05:47 +02:00
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.info = "Convert between sample format sign/endian",
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.name = "convertsignendian",
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.open = af_open,
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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
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.test_conversion = test_conversion,
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};
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