mirror of https://github.com/mpv-player/mpv
af_convert24: remove this filter
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5a9f817bfd
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@ -263,13 +263,6 @@ Available filters are:
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used to do conversion itself, unlike this one which lets the filter system
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handle the conversion.
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``convert24``
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Filter for internal use only. Converts between 24-bit and 32-bit sample
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formats.
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``convertsign``
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Filter for internal use only. Converts between signed/unsigned formats.
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``volume[=<volumedb>[:...]]``
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Implements software volume control. Use this filter with caution since it
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can reduce the signal to noise ratio of the sound. In most cases it is
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@ -123,7 +123,6 @@ SOURCES = audio/audio.c \
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audio/filter/af.c \
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audio/filter/af_center.c \
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audio/filter/af_channels.c \
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audio/filter/af_convert24.c \
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audio/filter/af_delay.c \
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audio/filter/af_dummy.c \
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audio/filter/af_equalizer.c \
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@ -55,7 +55,6 @@ extern const struct af_info af_info_karaoke;
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extern const struct af_info af_info_scaletempo;
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extern const struct af_info af_info_bs2b;
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extern const struct af_info af_info_lavfi;
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extern const struct af_info af_info_convert24;
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extern const struct af_info af_info_rubberband;
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static const struct af_info *const filter_list[] = {
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@ -91,8 +90,6 @@ static const struct af_info *const filter_list[] = {
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#if HAVE_LIBAVFILTER
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&af_info_lavfi,
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#endif
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// Must come last, because they're fallback format conversion filter
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&af_info_convert24,
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NULL
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};
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@ -1,120 +0,0 @@
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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 "audio/format.h"
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#include "af.h"
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#include "osdep/endian.h"
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static bool test_conversion(int src_format, int dst_format)
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{
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return (src_format == AF_FORMAT_S24 && dst_format == AF_FORMAT_S32) ||
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(src_format == AF_FORMAT_S32 && dst_format == AF_FORMAT_S24);
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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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if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_24BIT) {
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mp_audio_set_format(out, af_fmt_change_bits(in->format, 32));
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} else if ((in->format & AF_FORMAT_BITS_MASK) == AF_FORMAT_32BIT) {
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mp_audio_set_format(out, af_fmt_change_bits(in->format, 24));
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} else {
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abort();
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}
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out->rate = in->rate;
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mp_audio_set_channels(out, &in->channels);
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assert(test_conversion(in->format, out->format));
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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_SET_FORMAT: {
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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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// The LSB is always ignored.
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#if BYTE_ORDER == BIG_ENDIAN
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#define SHIFT(x) ((3-(x))*8)
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#else
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#define SHIFT(x) (((x)+1)*8)
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#endif
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static int filter(struct af_instance *af, struct mp_audio *data)
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{
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if (!data)
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return 0;
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struct mp_audio *out =
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mp_audio_pool_get(af->out_pool, af->data, data->samples);
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if (!out) {
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talloc_free(data);
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return -1;
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}
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mp_audio_copy_attributes(out, data);
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size_t len = mp_audio_psize(data) / data->bps;
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if (data->bps == 4) {
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for (int s = 0; s < len; s++) {
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uint32_t val = *((uint32_t *)data->planes[0] + s);
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uint8_t *ptr = (uint8_t *)out->planes[0] + s * 3;
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ptr[0] = val >> SHIFT(0);
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ptr[1] = val >> SHIFT(1);
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ptr[2] = val >> SHIFT(2);
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}
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} else {
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for (int s = 0; s < len; s++) {
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uint8_t *ptr = (uint8_t *)data->planes[0] + s * 3;
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uint32_t val = ptr[0] << SHIFT(0)
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| ptr[1] << SHIFT(1)
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| ptr[2] << SHIFT(2);
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*((uint32_t *)out->planes[0] + s) = val;
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}
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}
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talloc_free(data);
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af_add_output_frame(af, out);
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return 0;
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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->filter_frame = filter;
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return AF_OK;
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}
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const struct af_info af_info_convert24 = {
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.info = "Convert between 24 and 32 bit sample format",
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.name = "convert24",
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.open = af_open,
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.test_conversion = test_conversion,
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};
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@ -103,7 +103,6 @@ def build(ctx):
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( "audio/filter/af_bs2b.c", "libbs2b" ),
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( "audio/filter/af_center.c" ),
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( "audio/filter/af_channels.c" ),
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( "audio/filter/af_convert24.c" ),
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( "audio/filter/af_delay.c" ),
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( "audio/filter/af_drc.c" ),
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( "audio/filter/af_dummy.c" ),
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