[FFmpeg-cvslog] avfilter: add SOFAlizer audio filter
Paul B Mahol
git at videolan.org
Sat Dec 12 21:09:52 CET 2015
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Dec 9 13:40:03 2015 +0100| [0a19538bcf401afd369a597fe1fa06172368e46f] | committer: Paul B Mahol
avfilter: add SOFAlizer audio filter
Signed-off-by: Paul B Mahol <onemda at gmail.com>
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=0a19538bcf401afd369a597fe1fa06172368e46f
---
Changelog | 1 +
configure | 4 +
doc/filters.texi | 29 ++
libavfilter/Makefile | 1 +
libavfilter/af_sofalizer.c | 1018 ++++++++++++++++++++++++++++++++++++++++++++
libavfilter/allfilters.c | 1 +
libavfilter/formats.c | 11 +
libavfilter/formats.h | 3 +
libavfilter/version.h | 2 +-
9 files changed, 1069 insertions(+), 1 deletion(-)
diff --git a/Changelog b/Changelog
index a262896..1770068 100644
--- a/Changelog
+++ b/Changelog
@@ -44,6 +44,7 @@ version <next>:
- mips32r5 option has been removed
- mips64r6 option has been removed
- DXVA2-accelerated VP9 decoding
+- SOFAlizer: virtual binaural acoustics filter
version 2.8:
diff --git a/configure b/configure
index 3cefa7c..43fa9a6 100755
--- a/configure
+++ b/configure
@@ -279,6 +279,7 @@ External library support:
--disable-lzma disable lzma [autodetect]
--enable-decklink enable Blackmagic DeckLink I/O support [no]
--enable-mmal enable decoding via MMAL [no]
+ --enable-netcdf enable NetCDF, needed for sofalizer filter [no]
--enable-nvenc enable NVIDIA NVENC support [no]
--enable-openal enable OpenAL 1.1 capture support [no]
--enable-opencl enable OpenCL code
@@ -1503,6 +1504,7 @@ EXTERNAL_LIBRARY_LIST="
libzvbi
lzma
mmal
+ netcdf
nvenc
openal
opencl
@@ -2890,6 +2892,7 @@ showfreqs_filter_deps="avcodec"
showfreqs_filter_select="fft"
showspectrum_filter_deps="avcodec"
showspectrum_filter_select="rdft"
+sofalizer_filter_deps="netcdf"
spp_filter_deps="gpl avcodec"
spp_filter_select="fft idctdsp fdctdsp me_cmp pixblockdsp"
stereo3d_filter_deps="gpl"
@@ -5494,6 +5497,7 @@ enabled mmal && { check_lib interface/mmal/mmal.h mmal_port_connect
check_lib interface/mmal/mmal.h mmal_port_connect ; }
check_lib interface/mmal/mmal.h mmal_port_connect ; } ||
die "ERROR: mmal not found"; }
+enabled netcdf && require_pkg_config netcdf netcdf.h nc_inq_libvers
enabled nvenc && { check_header nvEncodeAPI.h || die "ERROR: nvEncodeAPI.h not found."; } &&
{ check_cpp_condition nvEncodeAPI.h "NVENCAPI_MAJOR_VERSION >= 5" ||
die "ERROR: NVENC API version 4 or older is not supported"; } &&
diff --git a/doc/filters.texi b/doc/filters.texi
index 2aebd8e..ba2ffc4 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2889,6 +2889,35 @@ silenceremove=1:5:0.02
@end example
@end itemize
+ at section sofalizer
+
+SOFAlizer uses head-related transfer functions (HRTFs) to create virtual
+loudspeakers around the user for binaural listening via headphones (audio
+formats up to 9 channels supported).
+The HRTFs are stored in SOFA files (see www.sofacoustics.org for a database).
+SOFAlizer is developed at the Acoustics Research Institute (ARI) of the
+Austrian Academy of Sciences.
+
+The filter accepts the following options:
+
+ at table @option
+ at item sofa
+Set the SOFA file used for rendering.
+
+ at item gain
+Set gain applied to audio. Value is in dB. Default is 0.
+
+ at item rotation
+Set rotation of virtual loudspeakers in deg. Default is 0.
+
+ at item elevation
+Set elevation of virtual speakers in deg. Default is 0.
+
+ at item radius
+Set distance in meters between loudspeakers and the listener with near-field
+HRTFs. Default is 1.
+ at end table
+
@section stereotools
This filter has some handy utilities to manage stereo signals, for converting
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 8884d1d..d7a3f61 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -87,6 +87,7 @@ OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o
OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
+OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o
OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
diff --git a/libavfilter/af_sofalizer.c b/libavfilter/af_sofalizer.c
new file mode 100644
index 0000000..b81521c
--- /dev/null
+++ b/libavfilter/af_sofalizer.c
@@ -0,0 +1,1018 @@
+/*****************************************************************************
+ * sofalizer.c : SOFAlizer filter for virtual binaural acoustics
+ *****************************************************************************
+ * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda,
+ * Acoustics Research Institute (ARI), Vienna, Austria
+ *
+ * Authors: Andreas Fuchs <andi.fuchs.mail at gmail.com>
+ * Wolfgang Hrauda <wolfgang.hrauda at gmx.at>
+ *
+ * SOFAlizer project coordinator at ARI, main developer of SOFA:
+ * Piotr Majdak <piotr at majdak.at>
+ *
+ * This program 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.
+ *
+ * This program 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 this program; if not, write to the Free Software Foundation,
+ * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
+ *****************************************************************************/
+
+#include <math.h>
+#include <netcdf.h>
+
+#include "libavutil/float_dsp.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "internal.h"
+#include "audio.h"
+
+typedef struct NCSofa { /* contains data of one SOFA file */
+ int ncid; /* netCDF ID of the opened SOFA file */
+ int n_samples; /* length of one impulse response (IR) */
+ int m_dim; /* number of measurement positions */
+ int *data_delay; /* broadband delay of each IR */
+ /* all measurement positions for each receiver (i.e. ear): */
+ float *sp_a; /* azimuth angles */
+ float *sp_e; /* elevation angles */
+ float *sp_r; /* radii */
+ /* data at each measurement position for each receiver: */
+ float *data_ir; /* IRs (time-domain) */
+} NCSofa;
+
+typedef struct SOFAlizerContext {
+ const AVClass *class;
+
+ char *filename; /* name of SOFA file */
+ NCSofa sofa; /* contains data of the SOFA file */
+
+ const int8_t *reorder; /* reorder in SOFA channel order */
+ int sample_rate; /* sample rate from SOFA file */
+ float *speaker_pos; /* positions of the virtual loudspekaers */
+ float gain_lfe; /* gain applied to LFE channel */
+
+ int n_conv; /* number of channels to convolute */
+
+ /* buffer variables (for convolution) */
+ float *ringbuffer[2]; /* buffers input samples, length of one buffer: */
+ /* no. input ch. (incl. LFE) x buffer_length */
+ int write[2]; /* current write position to ringbuffer */
+ int buffer_length; /* is: longest IR plus max. delay in all SOFA files */
+ /* then choose next power of 2 */
+
+ /* netCDF variables */
+ int *delay[2]; /* broadband delay for each channel/IR to be convolved */
+
+ float *data_ir[2]; /* IRs for all channels to be convolved */
+ /* (this excludes the LFE) */
+ float *temp_src[2];
+
+ /* control variables */
+ float gain; /* filter gain (in dB) */
+ float rotation; /* rotation of virtual loudspeakers (in degrees) */
+ float elevation; /* elevation of virtual loudspeakers (in deg.) */
+ float radius; /* distance virtual loudspeakers to listener (in metres) */
+
+ int lfe; /* whether or not the LFE channel is used */
+
+ AVFloatDSPContext *fdsp;
+} SOFAlizerContext;
+
+static int close_sofa(struct NCSofa *sofa)
+{
+ av_freep(&sofa->data_delay);
+ av_freep(&sofa->sp_a);
+ av_freep(&sofa->sp_e);
+ av_freep(&sofa->sp_r);
+ av_freep(&sofa->data_ir);
+ nc_close(sofa->ncid);
+ sofa->ncid = 0;
+
+ return 0;
+}
+
+static int load_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ /* variables associated with content of SOFA file: */
+ int ncid, n_dims, n_vars, n_gatts, n_unlim_dim_id, status;
+ char data_delay_dim_name[NC_MAX_NAME];
+ float *sp_a, *sp_e, *sp_r, *data_ir;
+ char *sofa_conventions;
+ char dim_name[NC_MAX_NAME]; /* names of netCDF dimensions */
+ size_t *dim_length; /* lengths of netCDF dimensions */
+ char *psz_conventions;
+ unsigned int sample_rate;
+ int data_delay_dim_id[2];
+ int samplingrate_id;
+ int data_delay_id;
+ int n_samples;
+ int m_dim_id = -1;
+ int n_dim_id = -1;
+ int data_ir_id;
+ size_t att_len;
+ int m_dim;
+ int *data_delay;
+ int sp_id;
+ int i, ret;
+
+ s->sofa.ncid = 0;
+ status = nc_open(filename, NC_NOWRITE, &ncid); /* open SOFA file read-only */
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);
+ return AVERROR(EINVAL);
+ }
+
+ /* get number of dimensions, vars, global attributes and Id of unlimited dimensions: */
+ nc_inq(ncid, &n_dims, &n_vars, &n_gatts, &n_unlim_dim_id);
+
+ /* -- get number of measurements ("M") and length of one IR ("N") -- */
+ dim_length = av_malloc_array(n_dims, sizeof(*dim_length));
+ if (!dim_length) {
+ nc_close(ncid);
+ return AVERROR(ENOMEM);
+ }
+
+ for (i = 0; i < n_dims; i++) { /* go through all dimensions of file */
+ nc_inq_dim(ncid, i, (char *)&dim_name, &dim_length[i]); /* get dimensions */
+ if (!strncmp("M", (const char *)&dim_name, 1)) /* get ID of dimension "M" */
+ m_dim_id = i;
+ if (!strncmp("N", (const char *)&dim_name, 1)) /* get ID of dimension "N" */
+ n_dim_id = i;
+ }
+
+ if ((m_dim_id == -1) || (n_dim_id == -1)) { /* dimension "M" or "N" couldn't be found */
+ av_log(ctx, AV_LOG_ERROR, "Can't find required dimensions in SOFA file.\n");
+ av_freep(&dim_length);
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+
+ n_samples = dim_length[n_dim_id]; /* get number of measurements */
+ m_dim = dim_length[m_dim_id]; /* get length of one IR */
+
+ av_freep(&dim_length);
+
+ /* -- check file type -- */
+ /* get length of attritube "Conventions" */
+ status = nc_inq_attlen(ncid, NC_GLOBAL, "Conventions", &att_len);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"Conventions\".\n");
+ nc_close(ncid);
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* check whether file is SOFA file */
+ psz_conventions = av_malloc(att_len + 1);
+ if (!psz_conventions) {
+ nc_close(ncid);
+ return AVERROR(ENOMEM);
+ }
+
+ nc_get_att_text(ncid, NC_GLOBAL, "Conventions", psz_conventions);
+ *(psz_conventions + att_len) = 0;
+ if (strncmp("SOFA", psz_conventions, 4)) {
+ av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n");
+ av_freep(&psz_conventions);
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+ av_freep(&psz_conventions);
+
+ status = nc_inq_attlen(ncid, NC_GLOBAL, "SOFAConventions", &att_len);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"SOFAConventions\".\n");
+ nc_close(ncid);
+ return AVERROR_INVALIDDATA;
+ }
+
+ sofa_conventions = av_malloc(att_len + 1);
+ if (!sofa_conventions) {
+ nc_close(ncid);
+ return AVERROR(ENOMEM);
+ }
+
+ nc_get_att_text(ncid, NC_GLOBAL, "SOFAConventions", sofa_conventions);
+ *(sofa_conventions + att_len) = 0;
+ if (strncmp("SimpleFreeFieldHRIR", sofa_conventions, att_len)) {
+ av_log(ctx, AV_LOG_ERROR, "Not a SimpleFreeFieldHRIR file!\n");
+ av_freep(&sofa_conventions);
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+ av_freep(&sofa_conventions);
+
+ /* -- get sampling rate of HRTFs -- */
+ /* read ID, then value */
+ status = nc_inq_varid(ncid, "Data.SamplingRate", &samplingrate_id);
+ status += nc_get_var_uint(ncid, samplingrate_id, &sample_rate);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.SamplingRate.\n");
+ nc_close(ncid);
+ return AVERROR(EINVAL);
+ }
+ *samplingrate = sample_rate; /* remember sampling rate */
+
+ /* -- allocate memory for one value for each measurement position: -- */
+ sp_a = s->sofa.sp_a = av_malloc_array(m_dim, sizeof(float));
+ sp_e = s->sofa.sp_e = av_malloc_array(m_dim, sizeof(float));
+ sp_r = s->sofa.sp_r = av_malloc_array(m_dim, sizeof(float));
+ /* delay and IR values required for each ear and measurement position: */
+ data_delay = s->sofa.data_delay = av_calloc(m_dim, 2 * sizeof(int));
+ data_ir = s->sofa.data_ir = av_malloc_array(m_dim * n_samples, sizeof(float) * 2);
+ s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
+ s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
+
+ if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir ||
+ !s->temp_src[0] || !s->temp_src[1]) {
+ /* if memory could not be allocated */
+ close_sofa(&s->sofa);
+ return AVERROR(ENOMEM);
+ }
+
+ /* get impulse responses (HRTFs): */
+ /* get corresponding ID */
+ status = nc_inq_varid(ncid, "Data.IR", &data_ir_id);
+ status += nc_get_var_float(ncid, data_ir_id, data_ir); /* read and store IRs */
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.IR!\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* get source positions of the HRTFs in the SOFA file: */
+ status = nc_inq_varid(ncid, "SourcePosition", &sp_id); /* get corresponding ID */
+ status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 0 } ,
+ (size_t[2]){ m_dim, 1}, sp_a); /* read & store azimuth angles */
+ status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 1 } ,
+ (size_t[2]){ m_dim, 1}, sp_e); /* read & store elevation angles */
+ status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 2 } ,
+ (size_t[2]){ m_dim, 1}, sp_r); /* read & store radii */
+ if (status != NC_NOERR) { /* if any source position variable coudn't be read */
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read SourcePosition.\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* read Data.Delay, check for errors and fit it to data_delay */
+ status = nc_inq_varid(ncid, "Data.Delay", &data_delay_id);
+ status += nc_inq_vardimid(ncid, data_delay_id, &data_delay_dim_id[0]);
+ status += nc_inq_dimname(ncid, data_delay_dim_id[0], data_delay_dim_name);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay.\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* Data.Delay dimension check */
+ /* dimension of Data.Delay is [I R]: */
+ if (!strncmp(data_delay_dim_name, "I", 2)) {
+ /* check 2 characters to assure string is 0-terminated after "I" */
+ int delay[2]; /* delays get from SOFA file: */
+
+ av_log(ctx, AV_LOG_DEBUG, "Data.Delay has dimension [I R]\n");
+ status = nc_get_var_int(ncid, data_delay_id, &delay[0]);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+ int *data_delay_r = data_delay + m_dim;
+ for (i = 0; i < m_dim; i++) { /* extend given dimension [I R] to [M R] */
+ /* assign constant delay value for all measurements to data_delay fields */
+ data_delay[i] = delay[0];
+ data_delay_r[i] = delay[1];
+ }
+ /* dimension of Data.Delay is [M R] */
+ } else if (!strncmp(data_delay_dim_name, "M", 2)) {
+ av_log(ctx, AV_LOG_ERROR, "Data.Delay in dimension [M R]\n");
+ /* get delays from SOFA file: */
+ status = nc_get_var_int(ncid, data_delay_id, data_delay);
+ if (status != NC_NOERR) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+ } else { /* dimension of Data.Delay is neither [I R] nor [M R] */
+ av_log(ctx, AV_LOG_ERROR, "Data.Delay does not have the required dimensions [I R] or [M R].\n");
+ ret = AVERROR(EINVAL);
+ goto error;
+ }
+
+ /* save information in SOFA struct: */
+ s->sofa.m_dim = m_dim; /* no. measurement positions */
+ s->sofa.n_samples = n_samples; /* length on one IR */
+ s->sofa.ncid = ncid; /* netCDF ID of SOFA file */
+ nc_close(ncid); /* close SOFA file */
+
+ return 0;
+
+error:
+ close_sofa(&s->sofa);
+ return ret;
+}
+
+static const int8_t reorder[18][9] = {
+ { 0, -1, -1, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, -1, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, -1, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 2, 3, -1, -1, -1, -1, -1 },
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1 },
+ { 0, 1, 3, 4, 2, -1, -1, -1, -1 },
+ { 0, 1, 4, 5, 2, 3, -1, -1, -1 },
+ { 0, 1, 4, 5, 2, 3, -1, -1, -1 },
+ { 0, 1, 5, 6, 4, 2, 3, -1, -1 },
+ { 0, 1, 5, 6, 3, 4, 2, -1, -1 },
+ { 0, 1, 6, 7, 4, 5, 2, 3, -1 },
+ { 0, 1, 2, 3, 4, 5, 6, 7, 8 },
+ { 0, 1, 2, 3, 4, 5, 6, 7, -1 },
+ { 0, 1, 3, 4, 2, 5, -1, -1, -1 },
+ { 0, 1, 4, 5, 2, 6, 3, -1, -1 },
+};
+
+static int get_speaker_pos(AVFilterContext *ctx, float *speaker_pos)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ uint64_t channels_layout = ctx->inputs[0]->channel_layout;
+ float pos_temp[9];
+ int nb_input_channels = ctx->inputs[0]->channels; /* get no. input channels */
+ int n_conv = nb_input_channels;
+
+ if (channels_layout & AV_CH_LOW_FREQUENCY) { /* if LFE is used */
+ /* decrease number of channels to be convolved: */
+ n_conv = nb_input_channels - 1;
+ }
+
+ /* set speaker positions according to input channel configuration: */
+ switch (channels_layout) {
+ case AV_CH_LAYOUT_MONO:
+ pos_temp[0] = 0;
+ break;
+ case AV_CH_LAYOUT_STEREO:
+ case AV_CH_LAYOUT_2POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ break;
+ case AV_CH_LAYOUT_SURROUND:
+ case AV_CH_LAYOUT_3POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 0;
+ break;
+ case AV_CH_LAYOUT_2_1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 180;
+ break;
+ case AV_CH_LAYOUT_2_2:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 90;
+ pos_temp[3] = 270;
+ break;
+ case AV_CH_LAYOUT_QUAD:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 120;
+ pos_temp[3] = 240;
+ break;
+ case AV_CH_LAYOUT_4POINT0:
+ case AV_CH_LAYOUT_4POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 0;
+ pos_temp[3] = 180;
+ break;
+ case AV_CH_LAYOUT_5POINT0:
+ case AV_CH_LAYOUT_5POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 90;
+ pos_temp[3] = 270;
+ pos_temp[4] = 0;
+ break;
+ case AV_CH_LAYOUT_5POINT0_BACK:
+ case AV_CH_LAYOUT_5POINT1_BACK:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 120;
+ pos_temp[3] = 240;
+ pos_temp[4] = 0;
+ break;
+ case AV_CH_LAYOUT_6POINT0:
+ case AV_CH_LAYOUT_6POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 90;
+ pos_temp[3] = 270;
+ pos_temp[4] = 0;
+ pos_temp[5] = 180;
+ break;
+ case AV_CH_LAYOUT_6POINT1_BACK:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 120;
+ pos_temp[3] = 240;
+ pos_temp[4] = 0;
+ pos_temp[4] = 180;
+ break;
+ case AV_CH_LAYOUT_HEXAGONAL:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 120;
+ pos_temp[3] = 240;
+ pos_temp[4] = 0;
+ pos_temp[5] = 180;
+ break;
+ case AV_CH_LAYOUT_7POINT0:
+ case AV_CH_LAYOUT_7POINT1:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 90;
+ pos_temp[3] = 270;
+ pos_temp[4] = 150;
+ pos_temp[5] = 210;
+ pos_temp[6] = 0;
+ break;
+ case AV_CH_LAYOUT_OCTAGONAL:
+ pos_temp[0] = 30;
+ pos_temp[1] = 330;
+ pos_temp[2] = 0;
+ pos_temp[3] = 150;
+ pos_temp[4] = 210;
+ pos_temp[5] = 180;
+ pos_temp[6] = 90;
+ pos_temp[7] = 270;
+ break;
+ default:
+ return -1;
+ }
+
+ switch (channels_layout) {
+ case AV_CH_LAYOUT_MONO:
+ s->reorder = reorder[0];
+ break;
+ case AV_CH_LAYOUT_STEREO:
+ s->reorder = reorder[1];
+ break;
+ case AV_CH_LAYOUT_2_1:
+ case AV_CH_LAYOUT_2POINT1:
+ s->reorder = reorder[2];
+ break;
+ case AV_CH_LAYOUT_SURROUND:
+ s->reorder = reorder[3];
+ break;
+ case AV_CH_LAYOUT_3POINT1:
+ case AV_CH_LAYOUT_2_2:
+ s->reorder = reorder[4];
+ break;
+ case AV_CH_LAYOUT_QUAD:
+ s->reorder = reorder[5];
+ break;
+ case AV_CH_LAYOUT_4POINT0:
+ s->reorder = reorder[6];
+ break;
+ case AV_CH_LAYOUT_4POINT1:
+ s->reorder = reorder[7];
+ break;
+ case AV_CH_LAYOUT_5POINT0:
+ case AV_CH_LAYOUT_5POINT0_BACK:
+ s->reorder = reorder[8];
+ break;
+ case AV_CH_LAYOUT_5POINT1:
+ case AV_CH_LAYOUT_5POINT1_BACK:
+ s->reorder = reorder[9];
+ break;
+ case AV_CH_LAYOUT_6POINT0:
+ s->reorder = reorder[10];
+ break;
+ case AV_CH_LAYOUT_HEXAGONAL:
+ s->reorder = reorder[16];
+ break;
+ case AV_CH_LAYOUT_6POINT1:
+ s->reorder = reorder[11];
+ break;
+ case AV_CH_LAYOUT_6POINT1_BACK:
+ s->reorder = reorder[17];
+ break;
+ case AV_CH_LAYOUT_7POINT0:
+ s->reorder = reorder[12];
+ break;
+ case AV_CH_LAYOUT_7POINT1:
+ s->reorder = reorder[13];
+ break;
+ case AV_CH_LAYOUT_OCTAGONAL:
+ s->reorder = reorder[15];
+ break;
+ default:
+ return -1;
+ }
+
+ memcpy(speaker_pos, pos_temp, n_conv * sizeof(float));
+
+ return 0;
+
+}
+
+static int max_delay(struct NCSofa *sofa)
+{
+ int i, max = 0;
+
+ for (i = 0; i < sofa->m_dim * 2; i++) {
+ /* search maximum delay in given SOFA file */
+ max = FFMAX(max, sofa->data_delay[i]);
+ }
+
+ return max;
+}
+
+static int find_m(SOFAlizerContext *s, int azim, int elev, float radius)
+{
+ /* get source positions and M of currently selected SOFA file */
+ float *sp_a = s->sofa.sp_a; /* azimuth angle */
+ float *sp_e = s->sofa.sp_e; /* elevation angle */
+ float *sp_r = s->sofa.sp_r; /* radius */
+ int m_dim = s->sofa.m_dim; /* no. measurements */
+ int best_id = 0; /* index m currently closest to desired source pos. */
+ float delta = 1000; /* offset between desired and currently best pos. */
+ float current;
+ int i;
+
+ for (i = 0; i < m_dim; i++) {
+ /* search through all measurements in currently selected SOFA file */
+ /* distance of current to desired source position: */
+ current = fabs(sp_a[i] - azim) +
+ fabs(sp_e[i] - elev) +
+ fabs(sp_r[i] - radius);
+ if (current <= delta) {
+ /* if current distance is smaller than smallest distance so far */
+ delta = current;
+ best_id = i; /* remember index */
+ }
+ }
+
+ return best_id;
+}
+
+static int compensate_volume(AVFilterContext *ctx)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ float compensate;
+ float energy = 0;
+ float *ir;
+ int m, j;
+
+ if (s->sofa.ncid) {
+ /* find IR at front center position in the SOFA file (IR closest to 0°,0°,1m) */
+ struct NCSofa *sofa = &s->sofa;
+ m = find_m(s, 0, 0, 1);
+ /* get energy of that IR and compensate volume */
+ ir = sofa->data_ir + 2 * m * sofa->n_samples;
+ for (j = 0; j < sofa->n_samples; j++) {
+ energy += *(ir + j) * *(ir + j);
+ }
+ compensate = 256 / (sofa->n_samples * sqrt(energy));
+ av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", compensate);
+ ir = sofa->data_ir;
+ for (j = 0; j < sofa->n_samples * sofa->m_dim * 2; j++) {
+ ir[j] *= compensate; /* apply volume compensation to IRs */
+ }
+ }
+
+ return 0;
+}
+
+typedef struct ThreadData {
+ AVFrame *in, *out;
+ int *write;
+ int **delay;
+ float **ir;
+ int *n_clippings;
+ float **ringbuffer;
+ float **temp_src;
+} ThreadData;
+
+static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
+{
+ SOFAlizerContext *s = ctx->priv;
+ ThreadData *td = arg;
+ AVFrame *in = td->in, *out = td->out;
+ int offset = jobnr;
+ int *write = &td->write[jobnr];
+ const int *const delay = td->delay[jobnr];
+ const float *const ir = td->ir[jobnr];
+ int *n_clippings = &td->n_clippings[jobnr];
+ float *ringbuffer = td->ringbuffer[jobnr];
+ float *temp_src = td->temp_src[jobnr];
+ const int n_samples = s->sofa.n_samples; /* length of one IR */
+ const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
+ float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
+ int in_channels = in->channels; /* number of input channels */
+ /* ring buffer length is: longest IR plus max. delay -> next power of 2 */
+ int buffer_length = s->buffer_length;
+ /* -1 for AND instead of MODULO (applied to powers of 2): */
+ uint32_t modulo = (uint32_t)buffer_length - 1;
+ float *buffer[10]; /* holds ringbuffer for each input channel */
+ int wr = *write;
+ int read;
+ int i, j, l;
+
+ dst += offset;
+ for (l = 0; l < in_channels; l++) {
+ /* get starting address of ringbuffer for each input channel */
+ buffer[l] = ringbuffer + l * buffer_length;
+ }
+
+ for (i = 0; i < in->nb_samples; i++) {
+ const float *temp_ir = ir; /* using same set of IRs for each sample */
+
+ *dst = 0;
+ for (l = 0; l < in_channels; l++) {
+ /* write current input sample to ringbuffer (for each channel) */
+ *(buffer[l] + wr) = src[s->reorder[l]];
+ }
+
+ /* loop goes through all channels to be convolved (excl. LFE): */
+ for (l = 0; l < s->n_conv; l++) {
+ const float *const bptr = buffer[l];
+
+ /* current read position in ringbuffer: input sample write position
+ * - delay for l-th ch. + diff. betw. IR length and buffer length
+ * (mod buffer length) */
+ read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo;
+
+ for (j = 0; j < n_samples; j++)
+ temp_src[j] = bptr[(read + j) & modulo];
+
+ /* multiply signal and IR, and add up the results */
+ dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);
+ temp_ir += n_samples;
+ }
+
+ if (s->lfe) { /* LFE */
+ /* apply gain to LFE signal and add to output buffer */
+ *dst += *(buffer[s->n_conv] + wr) * s->gain_lfe;
+ }
+
+ /* clippings counter */
+ if (fabs(*dst) > 1)
+ *n_clippings += 1;
+
+ /* move output buffer pointer by +2 to get to next sample of processed channel: */
+ dst += 2;
+ src += in_channels;
+ wr = (wr + 1) & modulo; /* update ringbuffer write position */
+ }
+
+ *write = wr; /* remember write position in ringbuffer for next call */
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SOFAlizerContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int n_clippings[2] = { 0 };
+ ThreadData td;
+ AVFrame *out;
+
+ out = ff_get_audio_buffer(outlink, in->nb_samples);
+ if (!out) {
+ av_frame_free(&in);
+ return AVERROR(ENOMEM);
+ }
+ av_frame_copy_props(out, in);
+
+ td.in = in; td.out = out; td.write = s->write;
+ td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
+ td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
+
+ ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2);
+ emms_c();
+
+ /* display error message if clipping occured */
+ if (n_clippings[0] + n_clippings[1] > 0) {
+ av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",
+ n_clippings[0] + n_clippings[1], out->nb_samples * 2);
+ }
+
+ av_frame_free(&in);
+ return ff_filter_frame(outlink, out);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ int ret, sample_rates[] = { 48000, -1 };
+ static const uint64_t channel_layouts[] = { AV_CH_LAYOUT_MONO,
+ AV_CH_LAYOUT_STEREO,
+ AV_CH_LAYOUT_2POINT1,
+ AV_CH_LAYOUT_SURROUND,
+ AV_CH_LAYOUT_2_1,
+ AV_CH_LAYOUT_4POINT0,
+ AV_CH_LAYOUT_QUAD,
+ AV_CH_LAYOUT_2_2,
+ AV_CH_LAYOUT_3POINT1,
+ AV_CH_LAYOUT_5POINT0_BACK,
+ AV_CH_LAYOUT_5POINT0,
+ AV_CH_LAYOUT_4POINT1,
+ AV_CH_LAYOUT_5POINT1_BACK,
+ AV_CH_LAYOUT_5POINT1,
+ AV_CH_LAYOUT_6POINT0,
+ AV_CH_LAYOUT_HEXAGONAL,
+ AV_CH_LAYOUT_6POINT1,
+ AV_CH_LAYOUT_6POINT1_BACK,
+ AV_CH_LAYOUT_7POINT0,
+ AV_CH_LAYOUT_7POINT1,
+ AV_CH_LAYOUT_OCTAGONAL,
+ 0, };
+
+ ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
+ if (ret)
+ return ret;
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret)
+ return ret;
+
+ layouts = ff_make_formatu64_list(channel_layouts);
+ if (!layouts)
+ return AVERROR(ENOMEM);
+
+ ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts);
+ if (ret)
+ return ret;
+
+ layouts = NULL;
+ ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);
+ if (ret)
+ return ret;
+
+ ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);
+ if (ret)
+ return ret;
+
+ sample_rates[0] = s->sample_rate;
+ formats = ff_make_format_list(sample_rates);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static int load_data(AVFilterContext *ctx, int azim, int elev, float radius)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ const int n_samples = s->sofa.n_samples;
+ int n_conv = s->n_conv; /* no. channels to convolve (excl. LFE) */
+ int delay_l[10]; /* broadband delay for each IR */
+ int delay_r[10];
+ int nb_input_channels = ctx->inputs[0]->channels; /* no. input channels */
+ float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */
+ float *data_ir_l = NULL;
+ float *data_ir_r = NULL;
+ int offset = 0; /* used for faster pointer arithmetics in for-loop */
+ int m[s->n_conv]; /* measurement index m of IR closest to required source positions */
+ int i, j, azim_orig = azim;
+
+ if (!s->sofa.ncid) { /* if an invalid SOFA file has been selected */
+ av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n");
+ return AVERROR_INVALIDDATA;
+ }
+
+ /* get temporary IR for L and R channel */
+ data_ir_l = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_l));
+ data_ir_r = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_r));
+ if (!data_ir_r || !data_ir_l) {
+ av_free(data_ir_l);
+ av_free(data_ir_r);
+ return AVERROR(ENOMEM);
+ }
+
+ for (i = 0; i < s->n_conv; i++) {
+ /* load and store IRs and corresponding delays */
+ azim = (int)(s->speaker_pos[i] + azim_orig) % 360;
+ /* get id of IR closest to desired position */
+ m[i] = find_m(s, azim, elev, radius);
+
+ /* load the delays associated with the current IRs */
+ delay_l[i] = *(s->sofa.data_delay + 2 * m[i]);
+ delay_r[i] = *(s->sofa.data_delay + 2 * m[i] + 1);
+
+ offset = i * n_samples; /* no. samples already written */
+ for (j = 0; j < n_samples; j++) {
+ /* load reversed IRs of the specified source position
+ * sample-by-sample for left and right ear; and apply gain */
+ *(data_ir_l + offset + j) = /* left channel */
+ *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j) * gain_lin;
+ *(data_ir_r + offset + j) = /* right channel */
+ *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j + n_samples) * gain_lin;
+ }
+
+ av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n",
+ m[i], *(s->sofa.sp_a + m[i]), *(s->sofa.sp_e + m[i]), *(s->sofa.sp_r + m[i]));
+ }
+
+ /* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */
+ memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * n_samples);
+ memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * n_samples);
+
+ av_free(data_ir_l); /* free temporary IR memory */
+ av_free(data_ir_r);
+
+ memcpy(s->delay[0], &delay_l[0], sizeof(int) * s->n_conv);
+ memcpy(s->delay[1], &delay_r[0], sizeof(int) * s->n_conv);
+
+ return 0;
+}
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ SOFAlizerContext *s = ctx->priv;
+ int ret;
+
+ /* load SOFA file, */
+ /* initialize file IDs to 0 before attempting to load SOFA files,
+ * this assures that in case of error, only the memory of already
+ * loaded files is free'd */
+ s->sofa.ncid = 0;
+ ret = load_sofa(ctx, s->filename, &s->sample_rate);
+ if (ret) {
+ /* file loading error */
+ av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename);
+ } else { /* no file loading error, resampling not required */
+ av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename);
+ }
+
+ if (ret) {
+ av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n");
+ return ret;
+ }
+
+ s->fdsp = avpriv_float_dsp_alloc(0);
+ if (!s->fdsp)
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static inline unsigned clz(unsigned x)
+{
+ unsigned i = sizeof(x) * 8;
+
+ while (x) {
+ x >>= 1;
+ i--;
+ }
+
+ return i;
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ SOFAlizerContext *s = ctx->priv;
+ int nb_input_channels = inlink->channels; /* no. input channels */
+ int n_max_ir = 0;
+ int n_current;
+ int n_max = 0;
+ int ret;
+
+ /* gain -3 dB per channel, -6 dB to get LFE on a similar level */
+ s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6) / 20 * M_LN10);
+
+ s->lfe = !!(inlink->channel_layout & AV_CH_LOW_FREQUENCY);
+ /* LFE is an input channel but requires no convolution */
+ s->n_conv = nb_input_channels - s->lfe;
+
+ /* get size of ringbuffer (longest IR plus max. delay) */
+ /* then choose next power of 2 for performance optimization */
+ n_current = s->sofa.n_samples + max_delay(&s->sofa);
+ if (n_current > n_max) {
+ /* length of longest IR plus max. delay (in all SOFA files) */
+ n_max = n_current;
+ /* length of longest IR (without delay, in all SOFA files) */
+ n_max_ir = s->sofa.n_samples;
+ }
+ /* buffer length is longest IR plus max. delay -> next power of 2
+ (32 - count leading zeros gives required exponent) */
+ s->buffer_length = exp2(32 - clz((uint32_t)n_max));
+
+ /* Allocate memory for the impulse responses, delays and the ringbuffers */
+ /* size: (longest IR) * (number of channels to convolute), without LFE */
+ s->data_ir[0] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv);
+ s->data_ir[1] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv);
+ /* length: number of channels to convolute */
+ s->delay[0] = av_malloc_array(s->n_conv, sizeof(float));
+ s->delay[1] = av_malloc_array(s->n_conv, sizeof(float));
+ /* length: (buffer length) * (number of input channels),
+ * OR: buffer length (if frequency domain processing)
+ * calloc zero-initializes the buffer */
+ s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
+ s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);
+ /* length: number of channels to convolute */
+ s->speaker_pos = av_malloc_array(s->n_conv, sizeof(*s->speaker_pos));
+
+ /* memory allocation failed: */
+ if (!s->data_ir[0] || !s->data_ir[1] || !s->delay[1] ||
+ !s->delay[0] || !s->ringbuffer[0] || !s->ringbuffer[1] ||
+ !s->speaker_pos)
+ return AVERROR(ENOMEM);
+
+ compensate_volume(ctx);
+
+ /* get speaker positions */
+ if ((ret = get_speaker_pos(ctx, s->speaker_pos)) < 0) {
+ av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");
+ return ret;
+ }
+ /* load IRs to data_ir[0] and data_ir[1] for required directions */
+ /* only load IRs if time-domain convolution is used. */
+ if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius)) < 0)
+ return ret;
+
+ av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n",
+ inlink->sample_rate, s->n_conv, nb_input_channels, s->buffer_length);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ SOFAlizerContext *s = ctx->priv;
+
+ if (s->sofa.ncid) {
+ av_freep(&s->sofa.sp_a);
+ av_freep(&s->sofa.sp_e);
+ av_freep(&s->sofa.sp_r);
+ av_freep(&s->sofa.data_delay);
+ av_freep(&s->sofa.data_ir);
+ }
+ av_freep(&s->delay[0]);
+ av_freep(&s->delay[1]);
+ av_freep(&s->data_ir[0]);
+ av_freep(&s->data_ir[1]);
+ av_freep(&s->ringbuffer[0]);
+ av_freep(&s->ringbuffer[1]);
+ av_freep(&s->speaker_pos);
+ av_freep(&s->temp_src[0]);
+ av_freep(&s->temp_src[1]);
+ av_freep(&s->fdsp);
+}
+
+#define OFFSET(x) offsetof(SOFAlizerContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption sofalizer_options[] = {
+ { "sofa", "sofa filename", OFFSET(filename), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS },
+ { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS },
+ { "rotation", "set rotation" , OFFSET(rotation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -360, 360, .flags = FLAGS },
+ { "elevation", "set elevation", OFFSET(elevation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -90, 90, .flags = FLAGS },
+ { "radius", "set radius", OFFSET(radius), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 3, .flags = FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(sofalizer);
+
+static const AVFilterPad inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_input,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_sofalizer = {
+ .name = "sofalizer",
+ .description = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."),
+ .priv_size = sizeof(SOFAlizerContext),
+ .priv_class = &sofalizer_class,
+ .init = init,
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .inputs = inputs,
+ .outputs = outputs,
+ .flags = AVFILTER_FLAG_SLICE_THREADS,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 0eeef53..131e067 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -109,6 +109,7 @@ void avfilter_register_all(void)
REGISTER_FILTER(SIDECHAINGATE, sidechaingate, af);
REGISTER_FILTER(SILENCEDETECT, silencedetect, af);
REGISTER_FILTER(SILENCEREMOVE, silenceremove, af);
+ REGISTER_FILTER(SOFALIZER, sofalizer, af);
REGISTER_FILTER(STEREOTOOLS, stereotools, af);
REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
REGISTER_FILTER(TREBLE, treble, af);
diff --git a/libavfilter/formats.c b/libavfilter/formats.c
index 95a6b11..a2b19e7 100644
--- a/libavfilter/formats.c
+++ b/libavfilter/formats.c
@@ -289,6 +289,17 @@ AVFilterFormats *ff_make_format_list(const int *fmts)
return formats;
}
+AVFilterChannelLayouts *ff_make_formatu64_list(const uint64_t *fmts)
+{
+ MAKE_FORMAT_LIST(AVFilterChannelLayouts,
+ channel_layouts, nb_channel_layouts);
+ if (count)
+ memcpy(formats->channel_layouts, fmts,
+ sizeof(*formats->channel_layouts) * count);
+
+ return formats;
+}
+
AVFilterChannelLayouts *avfilter_make_format64_list(const int64_t *fmts)
{
MAKE_FORMAT_LIST(AVFilterChannelLayouts,
diff --git a/libavfilter/formats.h b/libavfilter/formats.h
index 3d730f3..ce09d3c 100644
--- a/libavfilter/formats.h
+++ b/libavfilter/formats.h
@@ -141,6 +141,9 @@ AVFilterChannelLayouts *ff_all_channel_counts(void);
av_warn_unused_result
AVFilterChannelLayouts *avfilter_make_format64_list(const int64_t *fmts);
+av_warn_unused_result
+AVFilterChannelLayouts *ff_make_formatu64_list(const uint64_t *fmts);
+
/**
* A helper for query_formats() which sets all links to the same list of channel
diff --git a/libavfilter/version.h b/libavfilter/version.h
index a2c9462..a7832e0 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 6
-#define LIBAVFILTER_VERSION_MINOR 20
+#define LIBAVFILTER_VERSION_MINOR 21
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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