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ffmpeg/libavfilter/af_sofalizer.c
Paul B Mahol 04cc4d0550 avfilter/af_sofalizer: check av_fft_init return code 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
2015-12-15 10:04:01 +01:00

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/*****************************************************************************
* 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@gmail.com>
* Wolfgang Hrauda <wolfgang.hrauda@gmx.at>
*
* SOFAlizer project coordinator at ARI, main developer of SOFA:
* Piotr Majdak <piotr@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 "libavcodec/avfft.h"
#include "libavutil/float_dsp.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "internal.h"
#include "audio.h"
#define TIME_DOMAIN 0
#define FREQUENCY_DOMAIN 1
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 */
int sample_rate; /* sample rate from SOFA file */
float *speaker_azim; /* azimuth of the virtual loudspeakers */
float *speaker_elev; /* elevation of the virtual loudspeakers */
float gain_lfe; /* gain applied to LFE channel */
int lfe_channel; /* LFE channel position in channel layout */
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 */
int n_fft; /* number of samples in one FFT block */
/* 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];
FFTComplex *temp_fft[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 type; /* processing type */
FFTContext *fft[2], *ifft[2];
FFTComplex *data_hrtf[2];
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 *text;
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 length of one IR */
m_dim = dim_length[m_dim_id]; /* get number of measurements */
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 */
text = av_malloc(att_len + 1);
if (!text) {
nc_close(ncid);
return AVERROR(ENOMEM);
}
nc_get_att_text(ncid, NC_GLOBAL, "Conventions", text);
*(text + att_len) = 0;
if (strncmp("SOFA", text, 4)) {
av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n");
av_freep(&text);
nc_close(ncid);
return AVERROR(EINVAL);
}
av_freep(&text);
status = nc_inq_attlen(ncid, NC_GLOBAL, "License", &att_len);
if (status == NC_NOERR) {
text = av_malloc(att_len + 1);
if (text) {
nc_get_att_text(ncid, NC_GLOBAL, "License", text);
*(text + att_len) = 0;
av_log(ctx, AV_LOG_INFO, "SOFA file License: %s\n", text);
av_freep(&text);
}
}
status = nc_inq_attlen(ncid, NC_GLOBAL, "SourceDescription", &att_len);
if (status == NC_NOERR) {
text = av_malloc(att_len + 1);
if (text) {
nc_get_att_text(ncid, NC_GLOBAL, "SourceDescription", text);
*(text + att_len) = 0;
av_log(ctx, AV_LOG_INFO, "SOFA file SourceDescription: %s\n", text);
av_freep(&text);
}
}
status = nc_inq_attlen(ncid, NC_GLOBAL, "Comment", &att_len);
if (status == NC_NOERR) {
text = av_malloc(att_len + 1);
if (text) {
nc_get_att_text(ncid, NC_GLOBAL, "Comment", text);
*(text + att_len) = 0;
av_log(ctx, AV_LOG_INFO, "SOFA file Comment: %s\n", text);
av_freep(&text);
}
}
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);
if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir) {
/* 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 int get_speaker_pos(AVFilterContext *ctx,
float *speaker_azim, float *speaker_elev)
{
struct SOFAlizerContext *s = ctx->priv;
uint64_t channels_layout = ctx->inputs[0]->channel_layout;
float azim[10] = { 0 };
float elev[10] = { 0 };
int n_conv = ctx->inputs[0]->channels; /* get no. input channels */
s->lfe_channel = -1;
/* set speaker positions according to input channel configuration: */
switch (channels_layout) {
case AV_CH_LAYOUT_MONO:
azim[0] = 0;
break;
case AV_CH_LAYOUT_2POINT1:
s->lfe_channel = 2;
case AV_CH_LAYOUT_STEREO:
azim[0] = 30;
azim[1] = 330;
break;
case AV_CH_LAYOUT_3POINT1:
s->lfe_channel = 3;
case AV_CH_LAYOUT_SURROUND:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
break;
case AV_CH_LAYOUT_2_1:
azim[0] = 30;
azim[1] = 330;
azim[2] = 180;
break;
case AV_CH_LAYOUT_2_2:
azim[0] = 30;
azim[1] = 330;
azim[2] = 90;
azim[3] = 270;
break;
case AV_CH_LAYOUT_QUAD:
azim[0] = 30;
azim[1] = 330;
azim[2] = 120;
azim[3] = 240;
break;
case AV_CH_LAYOUT_4POINT1:
s->lfe_channel = 3;
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[4] = 180;
break;
case AV_CH_LAYOUT_4POINT0:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 180;
break;
case AV_CH_LAYOUT_5POINT1:
s->lfe_channel = 3;
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[4] = 90;
azim[5] = 270;
break;
case AV_CH_LAYOUT_5POINT0:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 90;
azim[4] = 270;
break;
case AV_CH_LAYOUT_5POINT1_BACK:
s->lfe_channel = 3;
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[4] = 120;
azim[5] = 240;
break;
case AV_CH_LAYOUT_5POINT0_BACK:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 120;
azim[4] = 240;
break;
case AV_CH_LAYOUT_6POINT1:
s->lfe_channel = 3;
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[4] = 180;
azim[5] = 90;
azim[6] = 270;
break;
case AV_CH_LAYOUT_6POINT0:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 180;
azim[4] = 90;
azim[5] = 270;
break;
case AV_CH_LAYOUT_6POINT1_BACK:
s->lfe_channel = 3;
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[4] = 120;
azim[5] = 240;
azim[6] = 180;
break;
case AV_CH_LAYOUT_HEXAGONAL:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 120;
azim[4] = 240;
azim[5] = 180;
break;
case AV_CH_LAYOUT_7POINT1:
s->lfe_channel = 3;
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[4] = 150;
azim[5] = 210;
azim[6] = 90;
azim[7] = 270;
break;
case AV_CH_LAYOUT_7POINT0:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 150;
azim[4] = 210;
azim[5] = 90;
azim[6] = 270;
break;
case AV_CH_LAYOUT_OCTAGONAL:
azim[0] = 30;
azim[1] = 330;
azim[2] = 0;
azim[3] = 150;
azim[4] = 210;
azim[5] = 180;
azim[6] = 90;
azim[7] = 270;
break;
default:
return -1;
}
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, 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;
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[10]; /* 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 += n_samples;
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 += n_samples;
}
/* 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 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 */
int n_fft = s->n_fft;
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 */
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[10]; /* 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_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 || !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 * 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;
}
} 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 * n_samples);
memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * n_samples);
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;
/* 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;
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 = exp2(32 - clz((uint32_t)n_max));
s->n_fft = exp2(32 - clz((uint32_t)(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.\n");
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_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 */
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" },
{ 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,
};
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