a55c953ef0
Signed-off-by: Paul B Mahol <onemda@gmail.com>
1229 lines
46 KiB
C
1229 lines
46 KiB
C
/*****************************************************************************
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* sofalizer.c : SOFAlizer filter for virtual binaural acoustics
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*****************************************************************************
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* Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda,
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* Acoustics Research Institute (ARI), Vienna, Austria
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*
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* Authors: Andreas Fuchs <andi.fuchs.mail@gmail.com>
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* Wolfgang Hrauda <wolfgang.hrauda@gmx.at>
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*
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* SOFAlizer project coordinator at ARI, main developer of SOFA:
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* Piotr Majdak <piotr@majdak.at>
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*
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* This program is free software; you can redistribute it and/or modify it
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* under the terms of the GNU Lesser General Public License as published by
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* the Free Software Foundation; either version 2.1 of the License, or
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* (at your option) any later version.
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*
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* This program 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 Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public License
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* along with this program; if not, write to the Free Software Foundation,
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* Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.
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*****************************************************************************/
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#include <math.h>
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#include <netcdf.h>
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#include "libavcodec/avfft.h"
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#include "libavutil/avstring.h"
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#include "libavutil/channel_layout.h"
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#include "libavutil/float_dsp.h"
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#include "libavutil/intmath.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "audio.h"
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#define TIME_DOMAIN 0
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#define FREQUENCY_DOMAIN 1
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typedef struct NCSofa { /* contains data of one SOFA file */
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int ncid; /* netCDF ID of the opened SOFA file */
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int n_samples; /* length of one impulse response (IR) */
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int m_dim; /* number of measurement positions */
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int *data_delay; /* broadband delay of each IR */
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/* all measurement positions for each receiver (i.e. ear): */
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float *sp_a; /* azimuth angles */
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float *sp_e; /* elevation angles */
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float *sp_r; /* radii */
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/* data at each measurement position for each receiver: */
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float *data_ir; /* IRs (time-domain) */
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} NCSofa;
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typedef struct VirtualSpeaker {
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uint8_t set;
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float azim;
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float elev;
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} VirtualSpeaker;
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typedef struct SOFAlizerContext {
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const AVClass *class;
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char *filename; /* name of SOFA file */
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NCSofa sofa; /* contains data of the SOFA file */
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int sample_rate; /* sample rate from SOFA file */
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float *speaker_azim; /* azimuth of the virtual loudspeakers */
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float *speaker_elev; /* elevation of the virtual loudspeakers */
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char *speakers_pos; /* custom positions of the virtual loudspeakers */
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float gain_lfe; /* gain applied to LFE channel */
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int lfe_channel; /* LFE channel position in channel layout */
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int n_conv; /* number of channels to convolute */
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/* buffer variables (for convolution) */
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float *ringbuffer[2]; /* buffers input samples, length of one buffer: */
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/* no. input ch. (incl. LFE) x buffer_length */
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int write[2]; /* current write position to ringbuffer */
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int buffer_length; /* is: longest IR plus max. delay in all SOFA files */
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/* then choose next power of 2 */
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int n_fft; /* number of samples in one FFT block */
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/* netCDF variables */
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int *delay[2]; /* broadband delay for each channel/IR to be convolved */
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float *data_ir[2]; /* IRs for all channels to be convolved */
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/* (this excludes the LFE) */
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float *temp_src[2];
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FFTComplex *temp_fft[2];
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/* control variables */
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float gain; /* filter gain (in dB) */
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float rotation; /* rotation of virtual loudspeakers (in degrees) */
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float elevation; /* elevation of virtual loudspeakers (in deg.) */
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float radius; /* distance virtual loudspeakers to listener (in metres) */
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int type; /* processing type */
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VirtualSpeaker vspkrpos[64];
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FFTContext *fft[2], *ifft[2];
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FFTComplex *data_hrtf[2];
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AVFloatDSPContext *fdsp;
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} SOFAlizerContext;
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static int close_sofa(struct NCSofa *sofa)
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{
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av_freep(&sofa->data_delay);
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av_freep(&sofa->sp_a);
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av_freep(&sofa->sp_e);
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av_freep(&sofa->sp_r);
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av_freep(&sofa->data_ir);
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nc_close(sofa->ncid);
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sofa->ncid = 0;
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return 0;
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}
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static int load_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
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{
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struct SOFAlizerContext *s = ctx->priv;
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/* variables associated with content of SOFA file: */
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int ncid, n_dims, n_vars, n_gatts, n_unlim_dim_id, status;
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char data_delay_dim_name[NC_MAX_NAME];
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float *sp_a, *sp_e, *sp_r, *data_ir;
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char *sofa_conventions;
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char dim_name[NC_MAX_NAME]; /* names of netCDF dimensions */
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size_t *dim_length; /* lengths of netCDF dimensions */
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char *text;
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unsigned int sample_rate;
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int data_delay_dim_id[2];
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int samplingrate_id;
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int data_delay_id;
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int n_samples;
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int m_dim_id = -1;
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int n_dim_id = -1;
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int data_ir_id;
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size_t att_len;
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int m_dim;
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int *data_delay;
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int sp_id;
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int i, ret;
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s->sofa.ncid = 0;
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status = nc_open(filename, NC_NOWRITE, &ncid); /* open SOFA file read-only */
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);
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return AVERROR(EINVAL);
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}
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/* get number of dimensions, vars, global attributes and Id of unlimited dimensions: */
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nc_inq(ncid, &n_dims, &n_vars, &n_gatts, &n_unlim_dim_id);
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/* -- get number of measurements ("M") and length of one IR ("N") -- */
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dim_length = av_malloc_array(n_dims, sizeof(*dim_length));
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if (!dim_length) {
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nc_close(ncid);
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return AVERROR(ENOMEM);
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}
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for (i = 0; i < n_dims; i++) { /* go through all dimensions of file */
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nc_inq_dim(ncid, i, (char *)&dim_name, &dim_length[i]); /* get dimensions */
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if (!strncmp("M", (const char *)&dim_name, 1)) /* get ID of dimension "M" */
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m_dim_id = i;
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if (!strncmp("N", (const char *)&dim_name, 1)) /* get ID of dimension "N" */
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n_dim_id = i;
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}
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if ((m_dim_id == -1) || (n_dim_id == -1)) { /* dimension "M" or "N" couldn't be found */
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av_log(ctx, AV_LOG_ERROR, "Can't find required dimensions in SOFA file.\n");
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av_freep(&dim_length);
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nc_close(ncid);
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return AVERROR(EINVAL);
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}
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n_samples = dim_length[n_dim_id]; /* get length of one IR */
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m_dim = dim_length[m_dim_id]; /* get number of measurements */
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av_freep(&dim_length);
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/* -- check file type -- */
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/* get length of attritube "Conventions" */
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status = nc_inq_attlen(ncid, NC_GLOBAL, "Conventions", &att_len);
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"Conventions\".\n");
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nc_close(ncid);
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return AVERROR_INVALIDDATA;
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}
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/* check whether file is SOFA file */
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text = av_malloc(att_len + 1);
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if (!text) {
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nc_close(ncid);
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return AVERROR(ENOMEM);
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}
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nc_get_att_text(ncid, NC_GLOBAL, "Conventions", text);
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*(text + att_len) = 0;
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if (strncmp("SOFA", text, 4)) {
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av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n");
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av_freep(&text);
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nc_close(ncid);
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return AVERROR(EINVAL);
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}
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av_freep(&text);
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status = nc_inq_attlen(ncid, NC_GLOBAL, "License", &att_len);
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if (status == NC_NOERR) {
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text = av_malloc(att_len + 1);
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if (text) {
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nc_get_att_text(ncid, NC_GLOBAL, "License", text);
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*(text + att_len) = 0;
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av_log(ctx, AV_LOG_INFO, "SOFA file License: %s\n", text);
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av_freep(&text);
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}
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}
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status = nc_inq_attlen(ncid, NC_GLOBAL, "SourceDescription", &att_len);
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if (status == NC_NOERR) {
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text = av_malloc(att_len + 1);
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if (text) {
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nc_get_att_text(ncid, NC_GLOBAL, "SourceDescription", text);
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*(text + att_len) = 0;
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av_log(ctx, AV_LOG_INFO, "SOFA file SourceDescription: %s\n", text);
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av_freep(&text);
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}
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}
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status = nc_inq_attlen(ncid, NC_GLOBAL, "Comment", &att_len);
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if (status == NC_NOERR) {
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text = av_malloc(att_len + 1);
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if (text) {
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nc_get_att_text(ncid, NC_GLOBAL, "Comment", text);
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*(text + att_len) = 0;
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av_log(ctx, AV_LOG_INFO, "SOFA file Comment: %s\n", text);
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av_freep(&text);
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}
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}
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status = nc_inq_attlen(ncid, NC_GLOBAL, "SOFAConventions", &att_len);
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"SOFAConventions\".\n");
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nc_close(ncid);
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return AVERROR_INVALIDDATA;
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}
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sofa_conventions = av_malloc(att_len + 1);
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if (!sofa_conventions) {
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nc_close(ncid);
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return AVERROR(ENOMEM);
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}
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nc_get_att_text(ncid, NC_GLOBAL, "SOFAConventions", sofa_conventions);
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*(sofa_conventions + att_len) = 0;
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if (strncmp("SimpleFreeFieldHRIR", sofa_conventions, att_len)) {
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av_log(ctx, AV_LOG_ERROR, "Not a SimpleFreeFieldHRIR file!\n");
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av_freep(&sofa_conventions);
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nc_close(ncid);
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return AVERROR(EINVAL);
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}
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av_freep(&sofa_conventions);
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/* -- get sampling rate of HRTFs -- */
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/* read ID, then value */
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status = nc_inq_varid(ncid, "Data.SamplingRate", &samplingrate_id);
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status += nc_get_var_uint(ncid, samplingrate_id, &sample_rate);
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.SamplingRate.\n");
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nc_close(ncid);
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return AVERROR(EINVAL);
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}
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*samplingrate = sample_rate; /* remember sampling rate */
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/* -- allocate memory for one value for each measurement position: -- */
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sp_a = s->sofa.sp_a = av_malloc_array(m_dim, sizeof(float));
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sp_e = s->sofa.sp_e = av_malloc_array(m_dim, sizeof(float));
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sp_r = s->sofa.sp_r = av_malloc_array(m_dim, sizeof(float));
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/* delay and IR values required for each ear and measurement position: */
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data_delay = s->sofa.data_delay = av_calloc(m_dim, 2 * sizeof(int));
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data_ir = s->sofa.data_ir = av_calloc(m_dim * FFALIGN(n_samples, 16), sizeof(float) * 2);
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if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir) {
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/* if memory could not be allocated */
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close_sofa(&s->sofa);
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return AVERROR(ENOMEM);
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}
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/* get impulse responses (HRTFs): */
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/* get corresponding ID */
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status = nc_inq_varid(ncid, "Data.IR", &data_ir_id);
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status += nc_get_var_float(ncid, data_ir_id, data_ir); /* read and store IRs */
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.IR!\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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/* get source positions of the HRTFs in the SOFA file: */
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status = nc_inq_varid(ncid, "SourcePosition", &sp_id); /* get corresponding ID */
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status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 0 } ,
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(size_t[2]){ m_dim, 1}, sp_a); /* read & store azimuth angles */
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status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 1 } ,
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(size_t[2]){ m_dim, 1}, sp_e); /* read & store elevation angles */
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status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 2 } ,
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(size_t[2]){ m_dim, 1}, sp_r); /* read & store radii */
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if (status != NC_NOERR) { /* if any source position variable coudn't be read */
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av_log(ctx, AV_LOG_ERROR, "Couldn't read SourcePosition.\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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/* read Data.Delay, check for errors and fit it to data_delay */
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status = nc_inq_varid(ncid, "Data.Delay", &data_delay_id);
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status += nc_inq_vardimid(ncid, data_delay_id, &data_delay_dim_id[0]);
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status += nc_inq_dimname(ncid, data_delay_dim_id[0], data_delay_dim_name);
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay.\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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/* Data.Delay dimension check */
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/* dimension of Data.Delay is [I R]: */
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if (!strncmp(data_delay_dim_name, "I", 2)) {
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/* check 2 characters to assure string is 0-terminated after "I" */
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int delay[2]; /* delays get from SOFA file: */
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int *data_delay_r;
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av_log(ctx, AV_LOG_DEBUG, "Data.Delay has dimension [I R]\n");
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status = nc_get_var_int(ncid, data_delay_id, &delay[0]);
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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data_delay_r = data_delay + m_dim;
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for (i = 0; i < m_dim; i++) { /* extend given dimension [I R] to [M R] */
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/* assign constant delay value for all measurements to data_delay fields */
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data_delay[i] = delay[0];
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data_delay_r[i] = delay[1];
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}
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/* dimension of Data.Delay is [M R] */
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} else if (!strncmp(data_delay_dim_name, "M", 2)) {
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av_log(ctx, AV_LOG_ERROR, "Data.Delay in dimension [M R]\n");
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/* get delays from SOFA file: */
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status = nc_get_var_int(ncid, data_delay_id, data_delay);
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if (status != NC_NOERR) {
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av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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} else { /* dimension of Data.Delay is neither [I R] nor [M R] */
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av_log(ctx, AV_LOG_ERROR, "Data.Delay does not have the required dimensions [I R] or [M R].\n");
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ret = AVERROR(EINVAL);
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goto error;
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}
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/* save information in SOFA struct: */
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s->sofa.m_dim = m_dim; /* no. measurement positions */
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s->sofa.n_samples = n_samples; /* length on one IR */
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s->sofa.ncid = ncid; /* netCDF ID of SOFA file */
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nc_close(ncid); /* close SOFA file */
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av_log(ctx, AV_LOG_DEBUG, "m_dim: %d n_samples %d\n", m_dim, n_samples);
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return 0;
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error:
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close_sofa(&s->sofa);
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return ret;
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}
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|
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static int parse_channel_name(char **arg, int *rchannel)
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{
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char buf[8];
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int len, i, channel_id = 0;
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int64_t layout, layout0;
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/* try to parse a channel name, e.g. "FL" */
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if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) {
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layout0 = layout = av_get_channel_layout(buf);
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/* channel_id <- first set bit in layout */
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for (i = 32; i > 0; i >>= 1) {
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if (layout >= (int64_t)1 << i) {
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channel_id += i;
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layout >>= i;
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}
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}
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/* reject layouts that are not a single channel */
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if (channel_id >= 64 || layout0 != (int64_t)1 << channel_id)
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return AVERROR(EINVAL);
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*rchannel = channel_id;
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*arg += len;
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return 0;
|
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}
|
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return AVERROR(EINVAL);
|
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}
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|
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static void parse_speaker_pos(AVFilterContext *ctx, int64_t in_channel_layout)
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{
|
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SOFAlizerContext *s = ctx->priv;
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char *arg, *tokenizer, *p, *args = av_strdup(s->speakers_pos);
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|
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if (!args)
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return;
|
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p = args;
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|
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while ((arg = av_strtok(p, "|", &tokenizer))) {
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float azim, elev;
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int out_ch_id;
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p = NULL;
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if (parse_channel_name(&arg, &out_ch_id))
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continue;
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if (sscanf(arg, "%f %f", &azim, &elev) == 2) {
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s->vspkrpos[out_ch_id].set = 1;
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s->vspkrpos[out_ch_id].azim = azim;
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s->vspkrpos[out_ch_id].elev = elev;
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} else if (sscanf(arg, "%f", &azim) == 1) {
|
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s->vspkrpos[out_ch_id].set = 1;
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s->vspkrpos[out_ch_id].azim = azim;
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s->vspkrpos[out_ch_id].elev = 0;
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}
|
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}
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|
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av_free(args);
|
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}
|
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|
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static int get_speaker_pos(AVFilterContext *ctx,
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float *speaker_azim, float *speaker_elev)
|
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{
|
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struct SOFAlizerContext *s = ctx->priv;
|
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uint64_t channels_layout = ctx->inputs[0]->channel_layout;
|
|
float azim[16] = { 0 };
|
|
float elev[16] = { 0 };
|
|
int m, ch, n_conv = ctx->inputs[0]->channels; /* get no. input channels */
|
|
|
|
if (n_conv > 16)
|
|
return AVERROR(EINVAL);
|
|
|
|
s->lfe_channel = -1;
|
|
|
|
if (s->speakers_pos)
|
|
parse_speaker_pos(ctx, channels_layout);
|
|
|
|
/* set speaker positions according to input channel configuration: */
|
|
for (m = 0, ch = 0; ch < n_conv && m < 64; m++) {
|
|
uint64_t mask = channels_layout & (1 << m);
|
|
|
|
switch (mask) {
|
|
case AV_CH_FRONT_LEFT: azim[ch] = 30; break;
|
|
case AV_CH_FRONT_RIGHT: azim[ch] = 330; break;
|
|
case AV_CH_FRONT_CENTER: azim[ch] = 0; break;
|
|
case AV_CH_LOW_FREQUENCY:
|
|
case AV_CH_LOW_FREQUENCY_2: s->lfe_channel = ch; break;
|
|
case AV_CH_BACK_LEFT: azim[ch] = 150; break;
|
|
case AV_CH_BACK_RIGHT: azim[ch] = 210; break;
|
|
case AV_CH_BACK_CENTER: azim[ch] = 180; break;
|
|
case AV_CH_SIDE_LEFT: azim[ch] = 90; break;
|
|
case AV_CH_SIDE_RIGHT: azim[ch] = 270; break;
|
|
case AV_CH_FRONT_LEFT_OF_CENTER: azim[ch] = 15; break;
|
|
case AV_CH_FRONT_RIGHT_OF_CENTER: azim[ch] = 345; break;
|
|
case AV_CH_TOP_CENTER: azim[ch] = 0;
|
|
elev[ch] = 90; break;
|
|
case AV_CH_TOP_FRONT_LEFT: azim[ch] = 30;
|
|
elev[ch] = 45; break;
|
|
case AV_CH_TOP_FRONT_CENTER: azim[ch] = 0;
|
|
elev[ch] = 45; break;
|
|
case AV_CH_TOP_FRONT_RIGHT: azim[ch] = 330;
|
|
elev[ch] = 45; break;
|
|
case AV_CH_TOP_BACK_LEFT: azim[ch] = 150;
|
|
elev[ch] = 45; break;
|
|
case AV_CH_TOP_BACK_RIGHT: azim[ch] = 210;
|
|
elev[ch] = 45; break;
|
|
case AV_CH_TOP_BACK_CENTER: azim[ch] = 180;
|
|
elev[ch] = 45; break;
|
|
case AV_CH_WIDE_LEFT: azim[ch] = 90; break;
|
|
case AV_CH_WIDE_RIGHT: azim[ch] = 270; break;
|
|
case AV_CH_SURROUND_DIRECT_LEFT: azim[ch] = 90; break;
|
|
case AV_CH_SURROUND_DIRECT_RIGHT: azim[ch] = 270; break;
|
|
case AV_CH_STEREO_LEFT: azim[ch] = 90; break;
|
|
case AV_CH_STEREO_RIGHT: azim[ch] = 270; break;
|
|
case 0: break;
|
|
default:
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
if (s->vspkrpos[m].set) {
|
|
azim[ch] = s->vspkrpos[m].azim;
|
|
elev[ch] = s->vspkrpos[m].elev;
|
|
}
|
|
|
|
if (mask)
|
|
ch++;
|
|
}
|
|
|
|
memcpy(speaker_azim, azim, n_conv * sizeof(float));
|
|
memcpy(speaker_elev, elev, 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;
|
|
|
|
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;
|
|
if (sofa->n_samples & 31) {
|
|
energy = avpriv_scalarproduct_float_c(ir, ir, sofa->n_samples);
|
|
} else {
|
|
energy = s->fdsp->scalarproduct_float(ir, ir, sofa->n_samples);
|
|
}
|
|
compensate = 256 / (sofa->n_samples * sqrt(energy));
|
|
av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", compensate);
|
|
ir = sofa->data_ir;
|
|
/* apply volume compensation to IRs */
|
|
if (sofa->n_samples & 31) {
|
|
int i;
|
|
for (i = 0; i < sofa->n_samples * sofa->m_dim * 2; i++) {
|
|
ir[i] = ir[i] * compensate;
|
|
}
|
|
} else {
|
|
s->fdsp->vector_fmul_scalar(ir, ir, compensate, sofa->n_samples * sofa->m_dim * 2);
|
|
emms_c();
|
|
}
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
typedef struct ThreadData {
|
|
AVFrame *in, *out;
|
|
int *write;
|
|
int **delay;
|
|
float **ir;
|
|
int *n_clippings;
|
|
float **ringbuffer;
|
|
float **temp_src;
|
|
FFTComplex **temp_fft;
|
|
} 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 */
|
|
const int in_channels = s->n_conv; /* number of input channels */
|
|
/* ring buffer length is: longest IR plus max. delay -> next power of 2 */
|
|
const int buffer_length = s->buffer_length;
|
|
/* -1 for AND instead of MODULO (applied to powers of 2): */
|
|
const uint32_t modulo = (uint32_t)buffer_length - 1;
|
|
float *buffer[16]; /* holds ringbuffer for each input channel */
|
|
int wr = *write;
|
|
int read;
|
|
int i, 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[l];
|
|
}
|
|
|
|
/* loop goes through all channels to be convolved */
|
|
for (l = 0; l < in_channels; l++) {
|
|
const float *const bptr = buffer[l];
|
|
|
|
if (l == s->lfe_channel) {
|
|
/* LFE is an input channel but requires no convolution */
|
|
/* apply gain to LFE signal and add to output buffer */
|
|
*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
|
|
temp_ir += FFALIGN(n_samples, 16);
|
|
continue;
|
|
}
|
|
|
|
/* 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;
|
|
|
|
if (read + n_samples < buffer_length) {
|
|
memcpy(temp_src, bptr + read, n_samples * sizeof(*temp_src));
|
|
} else {
|
|
int len = FFMIN(n_samples - (read % n_samples), buffer_length - read);
|
|
|
|
memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
|
|
memcpy(temp_src + len, bptr, (n_samples - len) * sizeof(*temp_src));
|
|
}
|
|
|
|
/* multiply signal and IR, and add up the results */
|
|
dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);
|
|
temp_ir += FFALIGN(n_samples, 16);
|
|
}
|
|
|
|
/* 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 sofalizer_fast_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];
|
|
FFTComplex *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */
|
|
int *n_clippings = &td->n_clippings[jobnr];
|
|
float *ringbuffer = td->ringbuffer[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 */
|
|
const int in_channels = s->n_conv; /* number of input channels */
|
|
/* ring buffer length is: longest IR plus max. delay -> next power of 2 */
|
|
const int buffer_length = s->buffer_length;
|
|
/* -1 for AND instead of MODULO (applied to powers of 2): */
|
|
const uint32_t modulo = (uint32_t)buffer_length - 1;
|
|
FFTComplex *fft_in = s->temp_fft[jobnr]; /* temporary array for FFT input/output data */
|
|
FFTContext *ifft = s->ifft[jobnr];
|
|
FFTContext *fft = s->fft[jobnr];
|
|
const int n_conv = s->n_conv;
|
|
const int n_fft = s->n_fft;
|
|
int wr = *write;
|
|
int n_read;
|
|
int i, j;
|
|
|
|
dst += offset;
|
|
|
|
/* find minimum between number of samples and output buffer length:
|
|
* (important, if one IR is longer than the output buffer) */
|
|
n_read = FFMIN(s->sofa.n_samples, in->nb_samples);
|
|
for (j = 0; j < n_read; j++) {
|
|
/* initialize output buf with saved signal from overflow buf */
|
|
dst[2 * j] = ringbuffer[wr];
|
|
ringbuffer[wr] = 0.0; /* re-set read samples to zero */
|
|
/* update ringbuffer read/write position */
|
|
wr = (wr + 1) & modulo;
|
|
}
|
|
|
|
/* initialize rest of output buffer with 0 */
|
|
for (j = n_read; j < in->nb_samples; j++) {
|
|
dst[2 * j] = 0;
|
|
}
|
|
|
|
for (i = 0; i < n_conv; i++) {
|
|
if (i == s->lfe_channel) { /* LFE */
|
|
for (j = 0; j < in->nb_samples; j++) {
|
|
/* apply gain to LFE signal and add to output buffer */
|
|
dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
|
|
}
|
|
continue;
|
|
}
|
|
|
|
/* outer loop: go through all input channels to be convolved */
|
|
offset = i * n_fft; /* no. samples already processed */
|
|
|
|
/* fill FFT input with 0 (we want to zero-pad) */
|
|
memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
|
|
|
|
for (j = 0; j < in->nb_samples; j++) {
|
|
/* prepare input for FFT */
|
|
/* write all samples of current input channel to FFT input array */
|
|
fft_in[j].re = src[j * in_channels + i];
|
|
}
|
|
|
|
/* transform input signal of current channel to frequency domain */
|
|
av_fft_permute(fft, fft_in);
|
|
av_fft_calc(fft, fft_in);
|
|
for (j = 0; j < n_fft; j++) {
|
|
const float re = fft_in[j].re;
|
|
const float im = fft_in[j].im;
|
|
|
|
/* complex multiplication of input signal and HRTFs */
|
|
/* output channel (real): */
|
|
fft_in[j].re = re * (hrtf + offset + j)->re - im * (hrtf + offset + j)->im;
|
|
/* output channel (imag): */
|
|
fft_in[j].im = re * (hrtf + offset + j)->im + im * (hrtf + offset + j)->re;
|
|
}
|
|
|
|
/* transform output signal of current channel back to time domain */
|
|
av_fft_permute(ifft, fft_in);
|
|
av_fft_calc(ifft, fft_in);
|
|
|
|
for (j = 0; j < in->nb_samples; j++) {
|
|
/* write output signal of current channel to output buffer */
|
|
dst[2 * j] += fft_in[j].re / (float)n_fft;
|
|
}
|
|
|
|
for (j = 0; j < n_samples - 1; j++) { /* overflow length is IR length - 1 */
|
|
/* write the rest of output signal to overflow buffer */
|
|
int write_pos = (wr + j) & modulo;
|
|
|
|
*(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re / (float)n_fft;
|
|
}
|
|
}
|
|
|
|
/* go through all samples of current output buffer: count clippings */
|
|
for (i = 0; i < out->nb_samples; i++) {
|
|
/* clippings counter */
|
|
if (fabs(*dst) > 1) { /* if current output sample > 1 */
|
|
*n_clippings = *n_clippings + 1;
|
|
}
|
|
|
|
/* move output buffer pointer by +2 to get to next sample of processed channel: */
|
|
dst += 2;
|
|
}
|
|
|
|
/* remember read/write position in ringbuffer for next call */
|
|
*write = wr;
|
|
|
|
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;
|
|
td.temp_fft = s->temp_fft;
|
|
|
|
if (s->type == TIME_DOMAIN) {
|
|
ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2);
|
|
} else {
|
|
ctx->internal->execute(ctx, sofalizer_fast_convolute, &td, NULL, 2);
|
|
}
|
|
emms_c();
|
|
|
|
/* display error message if clipping occurred */
|
|
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 };
|
|
|
|
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_all_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 */
|
|
int n_fft = s->n_fft;
|
|
int delay_l[16]; /* broadband delay for each IR */
|
|
int delay_r[16];
|
|
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 */
|
|
FFTComplex *data_hrtf_l = NULL;
|
|
FFTComplex *data_hrtf_r = NULL;
|
|
FFTComplex *fft_in_l = NULL;
|
|
FFTComplex *fft_in_r = NULL;
|
|
float *data_ir_l = NULL;
|
|
float *data_ir_r = NULL;
|
|
int offset = 0; /* used for faster pointer arithmetics in for-loop */
|
|
int m[16]; /* measurement index m of IR closest to required source positions */
|
|
int i, j, azim_orig = azim, elev_orig = elev;
|
|
|
|
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;
|
|
}
|
|
|
|
if (s->type == TIME_DOMAIN) {
|
|
s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
|
|
s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));
|
|
|
|
/* get temporary IR for L and R channel */
|
|
data_ir_l = av_calloc(n_conv * FFALIGN(n_samples, 16), sizeof(*data_ir_l));
|
|
data_ir_r = av_calloc(n_conv * FFALIGN(n_samples, 16), sizeof(*data_ir_r));
|
|
if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {
|
|
av_free(data_ir_l);
|
|
av_free(data_ir_r);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
} else {
|
|
/* get temporary HRTF memory for L and R channel */
|
|
data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv);
|
|
data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv);
|
|
if (!data_hrtf_r || !data_hrtf_l) {
|
|
av_free(data_hrtf_l);
|
|
av_free(data_hrtf_r);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
}
|
|
|
|
for (i = 0; i < s->n_conv; i++) {
|
|
/* load and store IRs and corresponding delays */
|
|
azim = (int)(s->speaker_azim[i] + azim_orig) % 360;
|
|
elev = (int)(s->speaker_elev[i] + elev_orig) % 90;
|
|
/* 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);
|
|
|
|
if (s->type == TIME_DOMAIN) {
|
|
offset = i * FFALIGN(n_samples, 16); /* 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;
|
|
}
|
|
} else {
|
|
fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
|
|
fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));
|
|
if (!fft_in_l || !fft_in_r) {
|
|
av_free(data_hrtf_l);
|
|
av_free(data_hrtf_r);
|
|
av_free(fft_in_l);
|
|
av_free(fft_in_r);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
offset = i * n_fft; /* no. samples already written */
|
|
for (j = 0; j < n_samples; j++) {
|
|
/* load non-reversed IRs of the specified source position
|
|
* sample-by-sample and apply gain,
|
|
* L channel is loaded to real part, R channel to imag part,
|
|
* IRs ared shifted by L and R delay */
|
|
fft_in_l[delay_l[i] + j].re = /* left channel */
|
|
*(s->sofa.data_ir + 2 * m[i] * n_samples + j) * gain_lin;
|
|
fft_in_r[delay_r[i] + j].re = /* right channel */
|
|
*(s->sofa.data_ir + (2 * m[i] + 1) * n_samples + j) * gain_lin;
|
|
}
|
|
|
|
/* actually transform to frequency domain (IRs -> HRTFs) */
|
|
av_fft_permute(s->fft[0], fft_in_l);
|
|
av_fft_calc(s->fft[0], fft_in_l);
|
|
memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));
|
|
av_fft_permute(s->fft[0], fft_in_r);
|
|
av_fft_calc(s->fft[0], fft_in_r);
|
|
memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
|
|
}
|
|
|
|
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]));
|
|
}
|
|
|
|
if (s->type == TIME_DOMAIN) {
|
|
/* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */
|
|
memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * FFALIGN(n_samples, 16));
|
|
memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * FFALIGN(n_samples, 16));
|
|
|
|
av_freep(&data_ir_l); /* free temporary IR memory */
|
|
av_freep(&data_ir_r);
|
|
} else {
|
|
s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));
|
|
s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));
|
|
if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
|
|
av_freep(&data_hrtf_l);
|
|
av_freep(&data_hrtf_r);
|
|
av_freep(&fft_in_l);
|
|
av_freep(&fft_in_r);
|
|
return AVERROR(ENOMEM); /* memory allocation failed */
|
|
}
|
|
|
|
memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */
|
|
sizeof(FFTComplex) * n_conv * n_fft); /* filter struct */
|
|
memcpy(s->data_hrtf[1], data_hrtf_r,
|
|
sizeof(FFTComplex) * n_conv * n_fft);
|
|
|
|
av_freep(&data_hrtf_l); /* free temporary HRTF memory */
|
|
av_freep(&data_hrtf_r);
|
|
|
|
av_freep(&fft_in_l); /* free temporary FFT memory */
|
|
av_freep(&fft_in_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;
|
|
|
|
if (!s->filename) {
|
|
av_log(ctx, AV_LOG_ERROR, "Valid SOFA filename must be set.\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
/* 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 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;
|
|
|
|
if (s->type == FREQUENCY_DOMAIN) {
|
|
inlink->partial_buf_size =
|
|
inlink->min_samples =
|
|
inlink->max_samples = inlink->sample_rate;
|
|
}
|
|
|
|
/* 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->n_conv = nb_input_channels;
|
|
|
|
/* 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 = 1 << (32 - ff_clz(n_max));
|
|
s->n_fft = 1 << (32 - ff_clz(n_max + inlink->sample_rate));
|
|
|
|
if (s->type == FREQUENCY_DOMAIN) {
|
|
av_fft_end(s->fft[0]);
|
|
av_fft_end(s->fft[1]);
|
|
s->fft[0] = av_fft_init(log2(s->n_fft), 0);
|
|
s->fft[1] = av_fft_init(log2(s->n_fft), 0);
|
|
av_fft_end(s->ifft[0]);
|
|
av_fft_end(s->ifft[1]);
|
|
s->ifft[0] = av_fft_init(log2(s->n_fft), 1);
|
|
s->ifft[1] = av_fft_init(log2(s->n_fft), 1);
|
|
|
|
if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {
|
|
av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
}
|
|
|
|
/* Allocate memory for the impulse responses, delays and the ringbuffers */
|
|
/* size: (longest IR) * (number of channels to convolute) */
|
|
s->data_ir[0] = av_calloc(FFALIGN(n_max_ir, 16), sizeof(float) * s->n_conv);
|
|
s->data_ir[1] = av_calloc(FFALIGN(n_max_ir, 16), 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 */
|
|
|
|
if (s->type == TIME_DOMAIN) {
|
|
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);
|
|
} else {
|
|
s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));
|
|
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
|
|
s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
|
|
s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));
|
|
if (!s->temp_fft[0] || !s->temp_fft[1])
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
/* length: number of channels to convolute */
|
|
s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim));
|
|
s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev));
|
|
|
|
/* 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_azim || !s->speaker_elev)
|
|
return AVERROR(ENOMEM);
|
|
|
|
compensate_volume(ctx);
|
|
|
|
/* get speaker positions */
|
|
if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 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 */
|
|
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_fft_end(s->ifft[0]);
|
|
av_fft_end(s->ifft[1]);
|
|
av_fft_end(s->fft[0]);
|
|
av_fft_end(s->fft[1]);
|
|
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_azim);
|
|
av_freep(&s->speaker_elev);
|
|
av_freep(&s->temp_src[0]);
|
|
av_freep(&s->temp_src[1]);
|
|
av_freep(&s->temp_fft[0]);
|
|
av_freep(&s->temp_fft[1]);
|
|
av_freep(&s->data_hrtf[0]);
|
|
av_freep(&s->data_hrtf[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 },
|
|
{ "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" },
|
|
{ "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" },
|
|
{ "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" },
|
|
{ "speakers", "set speaker custom positions", OFFSET(speakers_pos), AV_OPT_TYPE_STRING, {.str=0}, 0, 0, .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,
|
|
};
|