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ffmpeg.wasm-core
Commit c79eddaf authored by Clément Bœsch's avatar Clément Bœsch
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lavfi/aconvert: use libswresample. 

This commit also drops the planar parameter; you now need to use the 'p'
suffix in order to request a planar sample format.
parent e96be840
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Showing with 44 additions and 459 deletions
configure
View file @ c79eddaf
......@@ -1648,6 +1648,7 @@ tls_protocol_select="tcp_protocol"
udp_protocol_deps="network"
# filters
aconvert_filter_deps="swresample"
amovie_filter_deps="avcodec avformat"
aresample_filter_deps="swresample"
ass_filter_deps="libass"
......
doc/filters.texi
View file @ c79eddaf
......@@ -104,17 +104,15 @@ Below is a description of the currently available audio filters.
Convert the input audio format to the specified formats.
The filter accepts a string of the form:
"@var{sample_format}:@var{channel_layout}:@var{packing_format}".
"@var{sample_format}:@var{channel_layout}".
@var{sample_format} specifies the sample format, and can be a string or
the corresponding numeric value defined in @file{libavutil/samplefmt.h}.
@var{sample_format} specifies the sample format, and can be a string or the
corresponding numeric value defined in @file{libavutil/samplefmt.h}. Use 'p'
suffix for a planar sample format.
@var{channel_layout} specifies the channel layout, and can be a string
or the corresponding number value defined in @file{libavutil/audioconvert.h}.
@var{packing_format} specifies the type of packing in output, can be one
of "planar" or "packed", or the corresponding numeric values "0" or "1".
The special parameter "auto", signifies that the filter will
automatically select the output format depending on the output filter.
......@@ -122,16 +120,15 @@ Some examples follow.
@itemize
@item
Convert input to unsigned 8-bit, stereo, packed:
Convert input to float, planar, stereo:
@example
aconvert=u8:stereo:packed
aconvert=fltp:stereo
@end example
@item
Convert input to unsigned 8-bit, automatically select out channel layout
and packing format:
Convert input to unsigned 8-bit, automatically select out channel layout:
@example
aconvert=u8:auto:auto
aconvert=u8:auto
@end example
@end itemize
......
libavfilter/Makefile
View file @ c79eddaf
......@@ -3,7 +3,7 @@ include $(SUBDIR)../config.mak
NAME = avfilter
FFLIBS = avutil
FFLIBS-$(CONFIG_ACONVERT_FILTER) += avcodec
FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample
FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
FFLIBS-$(CONFIG_ARESAMPLE_FILTER) += swresample
FFLIBS-$(CONFIG_MOVIE_FILTER) += avformat avcodec
......
libavfilter/af_aconvert.c
View file @ c79eddaf
This diff is collapsed.
libavfilter/af_aconvert_rematrix.c deleted 100644 → 0
View file @ e96be840
/*
* Copyright (c) 2011 Mina Nagy Zaki
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* audio rematrixing functions, based on functions from libavcodec/resample.c
*/
#if defined(FLOATING)
# define DIV2 /2
#else
# define DIV2 >>1
#endif
REMATRIX_FUNC_SIG(stereo_to_mono_packed)
{
while (nb_samples >= 4) {
outp[0][0] = (inp[0][0] + inp[0][1]) DIV2;
outp[0][1] = (inp[0][2] + inp[0][3]) DIV2;
outp[0][2] = (inp[0][4] + inp[0][5]) DIV2;
outp[0][3] = (inp[0][6] + inp[0][7]) DIV2;
outp[0] += 4;
inp[0] += 8;
nb_samples -= 4;
}
while (nb_samples--) {
outp[0][0] = (inp[0][0] + inp[0][1]) DIV2;
outp[0]++;
inp[0] += 2;
}
}
REMATRIX_FUNC_SIG(stereo_downmix_packed)
{
while (nb_samples--) {
*outp[0]++ = inp[0][0];
*outp[0]++ = inp[0][1];
inp[0] += aconvert->in_nb_channels;
}
}
REMATRIX_FUNC_SIG(mono_to_stereo_packed)
{
while (nb_samples >= 4) {
outp[0][0] = outp[0][1] = inp[0][0];
outp[0][2] = outp[0][3] = inp[0][1];
outp[0][4] = outp[0][5] = inp[0][2];
outp[0][6] = outp[0][7] = inp[0][3];
outp[0] += 8;
inp[0] += 4;
nb_samples -= 4;
}
while (nb_samples--) {
outp[0][0] = outp[0][1] = inp[0][0];
outp[0] += 2;
inp[0] += 1;
}
}
/**
* This is for when we have more than 2 input channels, need to downmix to mono
* and do not have a conversion formula available. We just use first two input
* channels - left and right. This is a placeholder until more conversion
* functions are written.
*/
REMATRIX_FUNC_SIG(mono_downmix_packed)
{
while (nb_samples--) {
outp[0][0] = (inp[0][0] + inp[0][1]) DIV2;
inp[0] += aconvert->in_nb_channels;
outp[0]++;
}
}
REMATRIX_FUNC_SIG(mono_downmix_planar)
{
FMT_TYPE *out = outp[0];
while (nb_samples >= 4) {
out[0] = (inp[0][0] + inp[1][0]) DIV2;
out[1] = (inp[0][1] + inp[1][1]) DIV2;
out[2] = (inp[0][2] + inp[1][2]) DIV2;
out[3] = (inp[0][3] + inp[1][3]) DIV2;
out += 4;
inp[0] += 4;
inp[1] += 4;
nb_samples -= 4;
}
while (nb_samples--) {
out[0] = (inp[0][0] + inp[1][0]) DIV2;
out++;
inp[0]++;
inp[1]++;
}
}
/* Stereo to 5.1 output */
REMATRIX_FUNC_SIG(stereo_to_surround_5p1_packed)
{
while (nb_samples--) {
outp[0][0] = inp[0][0]; /* left */
outp[0][1] = inp[0][1]; /* right */
outp[0][2] = (inp[0][0] + inp[0][1]) DIV2; /* center */
outp[0][3] = 0; /* low freq */
outp[0][4] = 0; /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left */
outp[0][5] = 0; /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */
inp[0] += 2;
outp[0] += 6;
}
}
REMATRIX_FUNC_SIG(stereo_to_surround_5p1_planar)
{
while (nb_samples--) {
*outp[0]++ = *inp[0]; /* left */
*outp[1]++ = *inp[1]; /* right */
*outp[2]++ = (*inp[0] + *inp[1]) DIV2; /* center */
*outp[3]++ = 0; /* low freq */
*outp[4]++ = 0; /* FIXME: left surround: -3dB or -6dB or -9dB of stereo left */
*outp[5]++ = 0; /* FIXME: right surroud: -3dB or -6dB or -9dB of stereo right */
inp[0]++; inp[1]++;
}
}
/*
5.1 to stereo input: [fl, fr, c, lfe, rl, rr]
- Left = front_left + rear_gain * rear_left + center_gain * center
- Right = front_right + rear_gain * rear_right + center_gain * center
Where rear_gain is usually around 0.5-1.0 and
center_gain is almost always 0.7 (-3 dB)
*/
REMATRIX_FUNC_SIG(surround_5p1_to_stereo_packed)
{
while (nb_samples--) {
*outp[0]++ = inp[0][0] + (0.5 * inp[0][4]) + (0.7 * inp[0][2]); //FIXME CLIPPING!
*outp[0]++ = inp[0][1] + (0.5 * inp[0][5]) + (0.7 * inp[0][2]); //FIXME CLIPPING!
inp[0] += 6;
}
}
REMATRIX_FUNC_SIG(surround_5p1_to_stereo_planar)
{
while (nb_samples--) {
*outp[0]++ = *inp[0] + (0.5 * *inp[4]) + (0.7 * *inp[2]); //FIXME CLIPPING!
*outp[1]++ = *inp[1] + (0.5 * *inp[5]) + (0.7 * *inp[2]); //FIXME CLIPPING!
inp[0]++; inp[1]++; inp[2]++; inp[3]++; inp[4]++; inp[5]++;
}
}
#undef DIV2
#undef REMATRIX_FUNC_NAME
#undef FMT_TYPE
libavfilter/version.h
View file @ c79eddaf
......@@ -29,7 +29,7 @@
#include "libavutil/avutil.h"
#define LIBAVFILTER_VERSION_MAJOR 2
#define LIBAVFILTER_VERSION_MINOR 60
#define LIBAVFILTER_VERSION_MINOR 61
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
......
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