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FFmpeg/libavfilter/af_sofalizer.c

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  1. /*****************************************************************************

  2.  * sofalizer.c : SOFAlizer filter for virtual binaural acoustics

  3.  *****************************************************************************

  4.  * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda,

  5.  *                         Acoustics Research Institute (ARI), Vienna, Austria

  6.  *

  7.  * Authors: Andreas Fuchs <andi.fuchs.mail@gmail.com>

  8.  *          Wolfgang Hrauda <wolfgang.hrauda@gmx.at>

  9.  *

  10.  * SOFAlizer project coordinator at ARI, main developer of SOFA:

  11.  *          Piotr Majdak <piotr@majdak.at>

  12.  *

  13.  * This program is free software; you can redistribute it and/or modify it

  14.  * under the terms of the GNU Lesser General Public License as published by

  15.  * the Free Software Foundation; either version 2.1 of the License, or

  16.  * (at your option) any later version.

  17.  *

  18.  * This program is distributed in the hope that it will be useful,

  19.  * but WITHOUT ANY WARRANTY; without even the implied warranty of

  20.  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the

  21.  * GNU Lesser General Public License for more details.

  22.  *

  23.  * You should have received a copy of the GNU Lesser General Public License

  24.  * along with this program; if not, write to the Free Software Foundation,

  25.  * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA.

  26.  *****************************************************************************/

  27. 

  28. #include <math.h>

  29. #include <netcdf.h>

  30. 

  31. #include "libavcodec/avfft.h"

  32. #include "libavutil/float_dsp.h"

  33. #include "libavutil/intmath.h"

  34. #include "libavutil/opt.h"

  35. #include "avfilter.h"

  36. #include "internal.h"

  37. #include "audio.h"

  38. 

  39. #define TIME_DOMAIN      0

  40. #define FREQUENCY_DOMAIN 1

  41. 

  42. typedef struct NCSofa {  /* contains data of one SOFA file */

  43.     int ncid;            /* netCDF ID of the opened SOFA file */

  44.     int n_samples;       /* length of one impulse response (IR) */

  45.     int m_dim;           /* number of measurement positions */

  46.     int *data_delay;     /* broadband delay of each IR */

  47.                          /* all measurement positions for each receiver (i.e. ear): */

  48.     float *sp_a;         /* azimuth angles */

  49.     float *sp_e;         /* elevation angles */

  50.     float *sp_r;         /* radii */

  51.                          /* data at each measurement position for each receiver: */

  52.     float *data_ir;      /* IRs (time-domain) */

  53. } NCSofa;

  54. 

  55. typedef struct SOFAlizerContext {

  56.     const AVClass *class;

  57. 

  58.     char *filename;             /* name of SOFA file */

  59.     NCSofa sofa;                /* contains data of the SOFA file */

  60. 

  61.     int sample_rate;            /* sample rate from SOFA file */

  62.     float *speaker_azim;        /* azimuth of the virtual loudspeakers */

  63.     float *speaker_elev;        /* elevation of the virtual loudspeakers */

  64.     float gain_lfe;             /* gain applied to LFE channel */

  65.     int lfe_channel;            /* LFE channel position in channel layout */

  66. 

  67.     int n_conv;                 /* number of channels to convolute */

  68. 

  69.                                 /* buffer variables (for convolution) */

  70.     float *ringbuffer[2];       /* buffers input samples, length of one buffer: */

  71.                                 /* no. input ch. (incl. LFE) x buffer_length */

  72.     int write[2];               /* current write position to ringbuffer */

  73.     int buffer_length;          /* is: longest IR plus max. delay in all SOFA files */

  74.                                 /* then choose next power of 2 */

  75.     int n_fft;                  /* number of samples in one FFT block */

  76. 

  77.                                 /* netCDF variables */

  78.     int *delay[2];              /* broadband delay for each channel/IR to be convolved */

  79. 

  80.     float *data_ir[2];          /* IRs for all channels to be convolved */

  81.                                 /* (this excludes the LFE) */

  82.     float *temp_src[2];

  83.     FFTComplex *temp_fft[2];

  84. 

  85.                          /* control variables */

  86.     float gain;          /* filter gain (in dB) */

  87.     float rotation;      /* rotation of virtual loudspeakers (in degrees)  */

  88.     float elevation;     /* elevation of virtual loudspeakers (in deg.) */

  89.     float radius;        /* distance virtual loudspeakers to listener (in metres) */

  90.     int type;            /* processing type */

  91. 

  92.     FFTContext *fft[2], *ifft[2];

  93.     FFTComplex *data_hrtf[2];

  94. 

  95.     AVFloatDSPContext *fdsp;

  96. } SOFAlizerContext;

  97. 

  98. static int close_sofa(struct NCSofa *sofa)

  99. {

  100.     av_freep(&sofa->data_delay);

  101.     av_freep(&sofa->sp_a);

  102.     av_freep(&sofa->sp_e);

  103.     av_freep(&sofa->sp_r);

  104.     av_freep(&sofa->data_ir);

  105.     nc_close(sofa->ncid);

  106.     sofa->ncid = 0;

  107. 

  108.     return 0;

  109. }

  110. 

  111. static int load_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)

  112. {

  113.     struct SOFAlizerContext *s = ctx->priv;

  114.     /* variables associated with content of SOFA file: */

  115.     int ncid, n_dims, n_vars, n_gatts, n_unlim_dim_id, status;

  116.     char data_delay_dim_name[NC_MAX_NAME];

  117.     float *sp_a, *sp_e, *sp_r, *data_ir;

  118.     char *sofa_conventions;

  119.     char dim_name[NC_MAX_NAME];   /* names of netCDF dimensions */

  120.     size_t *dim_length;           /* lengths of netCDF dimensions */

  121.     char *text;

  122.     unsigned int sample_rate;

  123.     int data_delay_dim_id[2];

  124.     int samplingrate_id;

  125.     int data_delay_id;

  126.     int n_samples;

  127.     int m_dim_id = -1;

  128.     int n_dim_id = -1;

  129.     int data_ir_id;

  130.     size_t att_len;

  131.     int m_dim;

  132.     int *data_delay;

  133.     int sp_id;

  134.     int i, ret;

  135. 

  136.     s->sofa.ncid = 0;

  137.     status = nc_open(filename, NC_NOWRITE, &ncid); /* open SOFA file read-only */

  138.     if (status != NC_NOERR) {

  139.         av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);

  140.         return AVERROR(EINVAL);

  141.     }

  142. 

  143.     /* get number of dimensions, vars, global attributes and Id of unlimited dimensions: */

  144.     nc_inq(ncid, &n_dims, &n_vars, &n_gatts, &n_unlim_dim_id);

  145. 

  146.     /* -- get number of measurements ("M") and length of one IR ("N") -- */

  147.     dim_length = av_malloc_array(n_dims, sizeof(*dim_length));

  148.     if (!dim_length) {

  149.         nc_close(ncid);

  150.         return AVERROR(ENOMEM);

  151.     }

  152. 

  153.     for (i = 0; i < n_dims; i++) { /* go through all dimensions of file */

  154.         nc_inq_dim(ncid, i, (char *)&dim_name, &dim_length[i]); /* get dimensions */

  155.         if (!strncmp("M", (const char *)&dim_name, 1)) /* get ID of dimension "M" */

  156.             m_dim_id = i;

  157.         if (!strncmp("N", (const char *)&dim_name, 1)) /* get ID of dimension "N" */

  158.             n_dim_id = i;

  159.     }

  160. 

  161.     if ((m_dim_id == -1) || (n_dim_id == -1)) { /* dimension "M" or "N" couldn't be found */

  162.         av_log(ctx, AV_LOG_ERROR, "Can't find required dimensions in SOFA file.\n");

  163.         av_freep(&dim_length);

  164.         nc_close(ncid);

  165.         return AVERROR(EINVAL);

  166.     }

  167. 

  168.     n_samples = dim_length[n_dim_id]; /* get length of one IR */

  169.     m_dim     = dim_length[m_dim_id]; /* get number of measurements */

  170. 

  171.     av_freep(&dim_length);

  172. 

  173.     /* -- check file type -- */

  174.     /* get length of attritube "Conventions" */

  175.     status = nc_inq_attlen(ncid, NC_GLOBAL, "Conventions", &att_len);

  176.     if (status != NC_NOERR) {

  177.         av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"Conventions\".\n");

  178.         nc_close(ncid);

  179.         return AVERROR_INVALIDDATA;

  180.     }

  181. 

  182.     /* check whether file is SOFA file */

  183.     text = av_malloc(att_len + 1);

  184.     if (!text) {

  185.         nc_close(ncid);

  186.         return AVERROR(ENOMEM);

  187.     }

  188. 

  189.     nc_get_att_text(ncid, NC_GLOBAL, "Conventions", text);

  190.     *(text + att_len) = 0;

  191.     if (strncmp("SOFA", text, 4)) {

  192.         av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n");

  193.         av_freep(&text);

  194.         nc_close(ncid);

  195.         return AVERROR(EINVAL);

  196.     }

  197.     av_freep(&text);

  198. 

  199.     status = nc_inq_attlen(ncid, NC_GLOBAL, "License", &att_len);

  200.     if (status == NC_NOERR) {

  201.         text = av_malloc(att_len + 1);

  202.         if (text) {

  203.             nc_get_att_text(ncid, NC_GLOBAL, "License", text);

  204.             *(text + att_len) = 0;

  205.             av_log(ctx, AV_LOG_INFO, "SOFA file License: %s\n", text);

  206.             av_freep(&text);

  207.         }

  208.     }

  209. 

  210.     status = nc_inq_attlen(ncid, NC_GLOBAL, "SourceDescription", &att_len);

  211.     if (status == NC_NOERR) {

  212.         text = av_malloc(att_len + 1);

  213.         if (text) {

  214.             nc_get_att_text(ncid, NC_GLOBAL, "SourceDescription", text);

  215.             *(text + att_len) = 0;

  216.             av_log(ctx, AV_LOG_INFO, "SOFA file SourceDescription: %s\n", text);

  217.             av_freep(&text);

  218.         }

  219.     }

  220. 

  221.     status = nc_inq_attlen(ncid, NC_GLOBAL, "Comment", &att_len);

  222.     if (status == NC_NOERR) {

  223.         text = av_malloc(att_len + 1);

  224.         if (text) {

  225.             nc_get_att_text(ncid, NC_GLOBAL, "Comment", text);

  226.             *(text + att_len) = 0;

  227.             av_log(ctx, AV_LOG_INFO, "SOFA file Comment: %s\n", text);

  228.             av_freep(&text);

  229.         }

  230.     }

  231. 

  232.     status = nc_inq_attlen(ncid, NC_GLOBAL, "SOFAConventions", &att_len);

  233.     if (status != NC_NOERR) {

  234.         av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"SOFAConventions\".\n");

  235.         nc_close(ncid);

  236.         return AVERROR_INVALIDDATA;

  237.     }

  238. 

  239.     sofa_conventions = av_malloc(att_len + 1);

  240.     if (!sofa_conventions) {

  241.         nc_close(ncid);

  242.         return AVERROR(ENOMEM);

  243.     }

  244. 

  245.     nc_get_att_text(ncid, NC_GLOBAL, "SOFAConventions", sofa_conventions);

  246.     *(sofa_conventions + att_len) = 0;

  247.     if (strncmp("SimpleFreeFieldHRIR", sofa_conventions, att_len)) {

  248.         av_log(ctx, AV_LOG_ERROR, "Not a SimpleFreeFieldHRIR file!\n");

  249.         av_freep(&sofa_conventions);

  250.         nc_close(ncid);

  251.         return AVERROR(EINVAL);

  252.     }

  253.     av_freep(&sofa_conventions);

  254. 

  255.     /* -- get sampling rate of HRTFs -- */

  256.     /* read ID, then value */

  257.     status  = nc_inq_varid(ncid, "Data.SamplingRate", &samplingrate_id);

  258.     status += nc_get_var_uint(ncid, samplingrate_id, &sample_rate);

  259.     if (status != NC_NOERR) {

  260.         av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.SamplingRate.\n");

  261.         nc_close(ncid);

  262.         return AVERROR(EINVAL);

  263.     }

  264.     *samplingrate = sample_rate; /* remember sampling rate */

  265. 

  266.     /* -- allocate memory for one value for each measurement position: -- */

  267.     sp_a = s->sofa.sp_a = av_malloc_array(m_dim, sizeof(float));

  268.     sp_e = s->sofa.sp_e = av_malloc_array(m_dim, sizeof(float));

  269.     sp_r = s->sofa.sp_r = av_malloc_array(m_dim, sizeof(float));

  270.     /* delay and IR values required for each ear and measurement position: */

  271.     data_delay = s->sofa.data_delay = av_calloc(m_dim, 2 * sizeof(int));

  272.     data_ir = s->sofa.data_ir = av_malloc_array(m_dim * n_samples, sizeof(float) * 2);

  273. 

  274.     if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir) {

  275.         /* if memory could not be allocated */

  276.         close_sofa(&s->sofa);

  277.         return AVERROR(ENOMEM);

  278.     }

  279. 

  280.     /* get impulse responses (HRTFs): */

  281.     /* get corresponding ID */

  282.     status = nc_inq_varid(ncid, "Data.IR", &data_ir_id);

  283.     status += nc_get_var_float(ncid, data_ir_id, data_ir); /* read and store IRs */

  284.     if (status != NC_NOERR) {

  285.         av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.IR!\n");

  286.         ret = AVERROR(EINVAL);

  287.         goto error;

  288.     }

  289. 

  290.     /* get source positions of the HRTFs in the SOFA file: */

  291.     status  = nc_inq_varid(ncid, "SourcePosition", &sp_id); /* get corresponding ID */

  292.     status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 0 } ,

  293.                 (size_t[2]){ m_dim, 1}, sp_a); /* read & store azimuth angles */

  294.     status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 1 } ,

  295.                 (size_t[2]){ m_dim, 1}, sp_e); /* read & store elevation angles */

  296.     status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 2 } ,

  297.                 (size_t[2]){ m_dim, 1}, sp_r); /* read & store radii */

  298.     if (status != NC_NOERR) { /* if any source position variable coudn't be read */

  299.         av_log(ctx, AV_LOG_ERROR, "Couldn't read SourcePosition.\n");

  300.         ret = AVERROR(EINVAL);

  301.         goto error;

  302.     }

  303. 

  304.     /* read Data.Delay, check for errors and fit it to data_delay */

  305.     status  = nc_inq_varid(ncid, "Data.Delay", &data_delay_id);

  306.     status += nc_inq_vardimid(ncid, data_delay_id, &data_delay_dim_id[0]);

  307.     status += nc_inq_dimname(ncid, data_delay_dim_id[0], data_delay_dim_name);

  308.     if (status != NC_NOERR) {

  309.         av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay.\n");

  310.         ret = AVERROR(EINVAL);

  311.         goto error;

  312.     }

  313. 

  314.     /* Data.Delay dimension check */

  315.     /* dimension of Data.Delay is [I R]: */

  316.     if (!strncmp(data_delay_dim_name, "I", 2)) {

  317.         /* check 2 characters to assure string is 0-terminated after "I" */

  318.         int delay[2]; /* delays get from SOFA file: */

  319. 

  320.         av_log(ctx, AV_LOG_DEBUG, "Data.Delay has dimension [I R]\n");

  321.         status = nc_get_var_int(ncid, data_delay_id, &delay[0]);

  322.         if (status != NC_NOERR) {

  323.             av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");

  324.             ret = AVERROR(EINVAL);

  325.             goto error;

  326.         }

  327.         int *data_delay_r = data_delay + m_dim;

  328.         for (i = 0; i < m_dim; i++) { /* extend given dimension [I R] to [M R] */

  329.             /* assign constant delay value for all measurements to data_delay fields */

  330.             data_delay[i]   = delay[0];

  331.             data_delay_r[i] = delay[1];

  332.         }

  333.         /* dimension of Data.Delay is [M R] */

  334.     } else if (!strncmp(data_delay_dim_name, "M", 2)) {

  335.         av_log(ctx, AV_LOG_ERROR, "Data.Delay in dimension [M R]\n");

  336.         /* get delays from SOFA file: */

  337.         status = nc_get_var_int(ncid, data_delay_id, data_delay);

  338.         if (status != NC_NOERR) {

  339.             av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n");

  340.             ret = AVERROR(EINVAL);

  341.             goto error;

  342.         }

  343.     } else { /* dimension of Data.Delay is neither [I R] nor [M R] */

  344.         av_log(ctx, AV_LOG_ERROR, "Data.Delay does not have the required dimensions [I R] or [M R].\n");

  345.         ret = AVERROR(EINVAL);

  346.         goto error;

  347.     }

  348. 

  349.     /* save information in SOFA struct: */

  350.     s->sofa.m_dim = m_dim; /* no. measurement positions */

  351.     s->sofa.n_samples = n_samples; /* length on one IR */

  352.     s->sofa.ncid = ncid; /* netCDF ID of SOFA file */

  353.     nc_close(ncid); /* close SOFA file */

  354. 

  355.     return 0;

  356. 

  357. error:

  358.     close_sofa(&s->sofa);

  359.     return ret;

  360. }

  361. 

  362. static int get_speaker_pos(AVFilterContext *ctx,

  363.                            float *speaker_azim, float *speaker_elev)

  364. {

  365.     struct SOFAlizerContext *s = ctx->priv;

  366.     uint64_t channels_layout = ctx->inputs[0]->channel_layout;

  367.     float azim[16] = { 0 };

  368.     float elev[16] = { 0 };

  369.     int m, ch, n_conv = ctx->inputs[0]->channels; /* get no. input channels */

  370. 

  371.     if (n_conv > 16)

  372.         return AVERROR(EINVAL);

  373. 

  374.     s->lfe_channel = -1;

  375. 

  376.     /* set speaker positions according to input channel configuration: */

  377.     for (m = 0, ch = 0; ch < n_conv && m < 64; m++) {

  378.         uint64_t mask = channels_layout & (1 << m);

  379. 

  380.         switch (mask) {

  381.         case AV_CH_FRONT_LEFT:            azim[ch] =  30;      break;

  382.         case AV_CH_FRONT_RIGHT:           azim[ch] = 330;      break;

  383.         case AV_CH_FRONT_CENTER:          azim[ch] =   0;      break;

  384.         case AV_CH_LOW_FREQUENCY:

  385.         case AV_CH_LOW_FREQUENCY_2:       s->lfe_channel = ch; break;

  386.         case AV_CH_BACK_LEFT:             azim[ch] = 150;      break;

  387.         case AV_CH_BACK_RIGHT:            azim[ch] = 210;      break;

  388.         case AV_CH_BACK_CENTER:           azim[ch] = 180;      break;

  389.         case AV_CH_SIDE_LEFT:             azim[ch] =  90;      break;

  390.         case AV_CH_SIDE_RIGHT:            azim[ch] = 270;      break;

  391.         case AV_CH_FRONT_LEFT_OF_CENTER:  azim[ch] =  15;      break;

  392.         case AV_CH_FRONT_RIGHT_OF_CENTER: azim[ch] = 345;      break;

  393.         case AV_CH_TOP_CENTER:            azim[ch] =   0;

  394.                                           elev[ch] =  90;      break;

  395.         case AV_CH_TOP_FRONT_LEFT:        azim[ch] =  30;

  396.                                           elev[ch] =  45;      break;

  397.         case AV_CH_TOP_FRONT_CENTER:      azim[ch] =   0;

  398.                                           elev[ch] =  45;      break;

  399.         case AV_CH_TOP_FRONT_RIGHT:       azim[ch] = 330;

  400.                                           elev[ch] =  45;      break;

  401.         case AV_CH_TOP_BACK_LEFT:         azim[ch] = 150;

  402.                                           elev[ch] =  45;      break;

  403.         case AV_CH_TOP_BACK_RIGHT:        azim[ch] = 210;

  404.                                           elev[ch] =  45;      break;

  405.         case AV_CH_TOP_BACK_CENTER:       azim[ch] = 180;

  406.                                           elev[ch] =  45;      break;

  407.         case AV_CH_WIDE_LEFT:             azim[ch] =  90;      break;

  408.         case AV_CH_WIDE_RIGHT:            azim[ch] = 270;      break;

  409.         case AV_CH_SURROUND_DIRECT_LEFT:  azim[ch] =  90;      break;

  410.         case AV_CH_SURROUND_DIRECT_RIGHT: azim[ch] = 270;      break;

  411.         case AV_CH_STEREO_LEFT:           azim[ch] =  90;      break;

  412.         case AV_CH_STEREO_RIGHT:          azim[ch] = 270;      break;

  413.         case 0:                                                break;

  414.         default:

  415.             return AVERROR(EINVAL);

  416.         }

  417.         if (mask)

  418.             ch++;

  419.     }

  420. 

  421.     memcpy(speaker_azim, azim, n_conv * sizeof(float));

  422.     memcpy(speaker_elev, elev, n_conv * sizeof(float));

  423. 

  424.     return 0;

  425. 

  426. }

  427. 

  428. static int max_delay(struct NCSofa *sofa)

  429. {

  430.     int i, max = 0;

  431. 

  432.     for (i = 0; i < sofa->m_dim * 2; i++) {

  433.         /* search maximum delay in given SOFA file */

  434.         max = FFMAX(max, sofa->data_delay[i]);

  435.     }

  436. 

  437.     return max;

  438. }

  439. 

  440. static int find_m(SOFAlizerContext *s, int azim, int elev, float radius)

  441. {

  442.     /* get source positions and M of currently selected SOFA file */

  443.     float *sp_a = s->sofa.sp_a; /* azimuth angle */

  444.     float *sp_e = s->sofa.sp_e; /* elevation angle */

  445.     float *sp_r = s->sofa.sp_r; /* radius */

  446.     int m_dim = s->sofa.m_dim; /* no. measurements */

  447.     int best_id = 0; /* index m currently closest to desired source pos. */

  448.     float delta = 1000; /* offset between desired and currently best pos. */

  449.     float current;

  450.     int i;

  451. 

  452.     for (i = 0; i < m_dim; i++) {

  453.         /* search through all measurements in currently selected SOFA file */

  454.         /* distance of current to desired source position: */

  455.         current = fabs(sp_a[i] - azim) +

  456.                   fabs(sp_e[i] - elev) +

  457.                   fabs(sp_r[i] - radius);

  458.         if (current <= delta) {

  459.             /* if current distance is smaller than smallest distance so far */

  460.             delta = current;

  461.             best_id = i; /* remember index */

  462.         }

  463.     }

  464. 

  465.     return best_id;

  466. }

  467. 

  468. static int compensate_volume(AVFilterContext *ctx)

  469. {

  470.     struct SOFAlizerContext *s = ctx->priv;

  471.     float compensate;

  472.     float energy = 0;

  473.     float *ir;

  474.     int m;

  475. 

  476.     if (s->sofa.ncid) {

  477.         /* find IR at front center position in the SOFA file (IR closest to 0°,0°,1m) */

  478.         struct NCSofa *sofa = &s->sofa;

  479.         m = find_m(s, 0, 0, 1);

  480.         /* get energy of that IR and compensate volume */

  481.         ir = sofa->data_ir + 2 * m * sofa->n_samples;

  482.         if (sofa->n_samples & 31) {

  483.             energy = avpriv_scalarproduct_float_c(ir, ir, sofa->n_samples);

  484.         } else {

  485.             energy = s->fdsp->scalarproduct_float(ir, ir, sofa->n_samples);

  486.         }

  487.         compensate = 256 / (sofa->n_samples * sqrt(energy));

  488.         av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", compensate);

  489.         ir = sofa->data_ir;

  490.         /* apply volume compensation to IRs */

  491.         s->fdsp->vector_fmul_scalar(ir, ir, compensate, sofa->n_samples * sofa->m_dim * 2);

  492.         emms_c();

  493.     }

  494. 

  495.     return 0;

  496. }

  497. 

  498. typedef struct ThreadData {

  499.     AVFrame *in, *out;

  500.     int *write;

  501.     int **delay;

  502.     float **ir;

  503.     int *n_clippings;

  504.     float **ringbuffer;

  505.     float **temp_src;

  506.     FFTComplex **temp_fft;

  507. } ThreadData;

  508. 

  509. static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)

  510. {

  511.     SOFAlizerContext *s = ctx->priv;

  512.     ThreadData *td = arg;

  513.     AVFrame *in = td->in, *out = td->out;

  514.     int offset = jobnr;

  515.     int *write = &td->write[jobnr];

  516.     const int *const delay = td->delay[jobnr];

  517.     const float *const ir = td->ir[jobnr];

  518.     int *n_clippings = &td->n_clippings[jobnr];

  519.     float *ringbuffer = td->ringbuffer[jobnr];

  520.     float *temp_src = td->temp_src[jobnr];

  521.     const int n_samples = s->sofa.n_samples; /* length of one IR */

  522.     const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */

  523.     float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */

  524.     const int in_channels = s->n_conv; /* number of input channels */

  525.     /* ring buffer length is: longest IR plus max. delay -> next power of 2 */

  526.     const int buffer_length = s->buffer_length;

  527.     /* -1 for AND instead of MODULO (applied to powers of 2): */

  528.     const uint32_t modulo = (uint32_t)buffer_length - 1;

  529.     float *buffer[16]; /* holds ringbuffer for each input channel */

  530.     int wr = *write;

  531.     int read;

  532.     int i, l;

  533. 

  534.     dst += offset;

  535.     for (l = 0; l < in_channels; l++) {

  536.         /* get starting address of ringbuffer for each input channel */

  537.         buffer[l] = ringbuffer + l * buffer_length;

  538.     }

  539. 

  540.     for (i = 0; i < in->nb_samples; i++) {

  541.         const float *temp_ir = ir; /* using same set of IRs for each sample */

  542. 

  543.         *dst = 0;

  544.         for (l = 0; l < in_channels; l++) {

  545.             /* write current input sample to ringbuffer (for each channel) */

  546.             *(buffer[l] + wr) = src[l];

  547.         }

  548. 

  549.         /* loop goes through all channels to be convolved */

  550.         for (l = 0; l < in_channels; l++) {

  551.             const float *const bptr = buffer[l];

  552. 

  553.             if (l == s->lfe_channel) {

  554.                 /* LFE is an input channel but requires no convolution */

  555.                 /* apply gain to LFE signal and add to output buffer */

  556.                 *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;

  557.                 temp_ir += n_samples;

  558.                 continue;

  559.             }

  560. 

  561.             /* current read position in ringbuffer: input sample write position

  562.              * - delay for l-th ch. + diff. betw. IR length and buffer length

  563.              * (mod buffer length) */

  564.             read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo;

  565. 

  566.             if (read + n_samples < buffer_length) {

  567.                 memcpy(temp_src, bptr + read, n_samples * sizeof(*temp_src));

  568.             } else {

  569.                 int len = FFMIN(n_samples - (read % n_samples), buffer_length - read);

  570. 

  571.                 memcpy(temp_src, bptr + read, len * sizeof(*temp_src));

  572.                 memcpy(temp_src + len, bptr, (n_samples - len) * sizeof(*temp_src));

  573.             }

  574. 

  575.             /* multiply signal and IR, and add up the results */

  576.             dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);

  577.             temp_ir += n_samples;

  578.         }

  579. 

  580.         /* clippings counter */

  581.         if (fabs(*dst) > 1)

  582.             *n_clippings += 1;

  583. 

  584.         /* move output buffer pointer by +2 to get to next sample of processed channel: */

  585.         dst += 2;

  586.         src += in_channels;

  587.         wr   = (wr + 1) & modulo; /* update ringbuffer write position */

  588.     }

  589. 

  590.     *write = wr; /* remember write position in ringbuffer for next call */

  591. 

  592.     return 0;

  593. }

  594. 

  595. static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)

  596. {

  597.     SOFAlizerContext *s = ctx->priv;

  598.     ThreadData *td = arg;

  599.     AVFrame *in = td->in, *out = td->out;

  600.     int offset = jobnr;

  601.     int *write = &td->write[jobnr];

  602.     FFTComplex *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */

  603.     int *n_clippings = &td->n_clippings[jobnr];

  604.     float *ringbuffer = td->ringbuffer[jobnr];

  605.     const int n_samples = s->sofa.n_samples; /* length of one IR */

  606.     const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */

  607.     float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */

  608.     const int in_channels = s->n_conv; /* number of input channels */

  609.     /* ring buffer length is: longest IR plus max. delay -> next power of 2 */

  610.     const int buffer_length = s->buffer_length;

  611.     /* -1 for AND instead of MODULO (applied to powers of 2): */

  612.     const uint32_t modulo = (uint32_t)buffer_length - 1;

  613.     FFTComplex *fft_in = s->temp_fft[jobnr]; /* temporary array for FFT input/output data */

  614.     FFTContext *ifft = s->ifft[jobnr];

  615.     FFTContext *fft = s->fft[jobnr];

  616.     const int n_conv = s->n_conv;

  617.     const int n_fft = s->n_fft;

  618.     int wr = *write;

  619.     int n_read;

  620.     int i, j;

  621. 

  622.     dst += offset;

  623. 

  624.     /* find minimum between number of samples and output buffer length:

  625.      * (important, if one IR is longer than the output buffer) */

  626.     n_read = FFMIN(s->sofa.n_samples, in->nb_samples);

  627.     for (j = 0; j < n_read; j++) {

  628.         /* initialize output buf with saved signal from overflow buf */

  629.         dst[2 * j]     = ringbuffer[wr];

  630.         ringbuffer[wr] = 0.0; /* re-set read samples to zero */

  631.         /* update ringbuffer read/write position */

  632.         wr  = (wr + 1) & modulo;

  633.     }

  634. 

  635.     /* initialize rest of output buffer with 0 */

  636.     for (j = n_read; j < in->nb_samples; j++) {

  637.         dst[2 * j] = 0;

  638.     }

  639. 

  640.     for (i = 0; i < n_conv; i++) {

  641.         if (i == s->lfe_channel) { /* LFE */

  642.             for (j = 0; j < in->nb_samples; j++) {

  643.                 /* apply gain to LFE signal and add to output buffer */

  644.                 dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;

  645.             }

  646.             continue;

  647.         }

  648. 

  649.         /* outer loop: go through all input channels to be convolved */

  650.         offset = i * n_fft; /* no. samples already processed */

  651. 

  652.         /* fill FFT input with 0 (we want to zero-pad) */

  653.         memset(fft_in, 0, sizeof(FFTComplex) * n_fft);

  654. 

  655.         for (j = 0; j < in->nb_samples; j++) {

  656.             /* prepare input for FFT */

  657.             /* write all samples of current input channel to FFT input array */

  658.             fft_in[j].re = src[j * in_channels + i];

  659.         }

  660. 

  661.         /* transform input signal of current channel to frequency domain */

  662.         av_fft_permute(fft, fft_in);

  663.         av_fft_calc(fft, fft_in);

  664.         for (j = 0; j < n_fft; j++) {

  665.             const float re = fft_in[j].re;

  666.             const float im = fft_in[j].im;

  667. 

  668.             /* complex multiplication of input signal and HRTFs */

  669.             /* output channel (real): */

  670.             fft_in[j].re = re * (hrtf + offset + j)->re - im * (hrtf + offset + j)->im;

  671.             /* output channel (imag): */

  672.             fft_in[j].im = re * (hrtf + offset + j)->im + im * (hrtf + offset + j)->re;

  673.         }

  674. 

  675.         /* transform output signal of current channel back to time domain */

  676.         av_fft_permute(ifft, fft_in);

  677.         av_fft_calc(ifft, fft_in);

  678. 

  679.         for (j = 0; j < in->nb_samples; j++) {

  680.             /* write output signal of current channel to output buffer */

  681.             dst[2 * j] += fft_in[j].re / (float)n_fft;

  682.         }

  683. 

  684.         for (j = 0; j < n_samples - 1; j++) { /* overflow length is IR length - 1 */

  685.             /* write the rest of output signal to overflow buffer */

  686.             int write_pos = (wr + j) & modulo;

  687. 

  688.             *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re / (float)n_fft;

  689.         }

  690.     }

  691. 

  692.     /* go through all samples of current output buffer: count clippings */

  693.     for (i = 0; i < out->nb_samples; i++) {

  694.         /* clippings counter */

  695.         if (fabs(*dst) > 1) { /* if current output sample > 1 */

  696.             *n_clippings = *n_clippings + 1;

  697.         }

  698. 

  699.         /* move output buffer pointer by +2 to get to next sample of processed channel: */

  700.         dst += 2;

  701.     }

  702. 

  703.     /* remember read/write position in ringbuffer for next call */

  704.     *write = wr;

  705. 

  706.     return 0;

  707. }

  708. 

  709. static int filter_frame(AVFilterLink *inlink, AVFrame *in)

  710. {

  711.     AVFilterContext *ctx = inlink->dst;

  712.     SOFAlizerContext *s = ctx->priv;

  713.     AVFilterLink *outlink = ctx->outputs[0];

  714.     int n_clippings[2] = { 0 };

  715.     ThreadData td;

  716.     AVFrame *out;

  717. 

  718.     out = ff_get_audio_buffer(outlink, in->nb_samples);

  719.     if (!out) {

  720.         av_frame_free(&in);

  721.         return AVERROR(ENOMEM);

  722.     }

  723.     av_frame_copy_props(out, in);

  724. 

  725.     td.in = in; td.out = out; td.write = s->write;

  726.     td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;

  727.     td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;

  728.     td.temp_fft = s->temp_fft;

  729. 

  730.     if (s->type == TIME_DOMAIN) {

  731.         ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2);

  732.     } else {

  733.         ctx->internal->execute(ctx, sofalizer_fast_convolute, &td, NULL, 2);

  734.     }

  735.     emms_c();

  736. 

  737.     /* display error message if clipping occurred */

  738.     if (n_clippings[0] + n_clippings[1] > 0) {

  739.         av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n",

  740.                n_clippings[0] + n_clippings[1], out->nb_samples * 2);

  741.     }

  742. 

  743.     av_frame_free(&in);

  744.     return ff_filter_frame(outlink, out);

  745. }

  746. 

  747. static int query_formats(AVFilterContext *ctx)

  748. {

  749.     struct SOFAlizerContext *s = ctx->priv;

  750.     AVFilterFormats *formats = NULL;

  751.     AVFilterChannelLayouts *layouts = NULL;

  752.     int ret, sample_rates[] = { 48000, -1 };

  753. 

  754.     ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);

  755.     if (ret)

  756.         return ret;

  757.     ret = ff_set_common_formats(ctx, formats);

  758.     if (ret)

  759.         return ret;

  760. 

  761.     layouts = ff_all_channel_layouts();

  762.     if (!layouts)

  763.         return AVERROR(ENOMEM);

  764. 

  765.     ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts);

  766.     if (ret)

  767.         return ret;

  768. 

  769.     layouts = NULL;

  770.     ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO);

  771.     if (ret)

  772.         return ret;

  773. 

  774.     ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts);

  775.     if (ret)

  776.         return ret;

  777. 

  778.     sample_rates[0] = s->sample_rate;

  779.     formats = ff_make_format_list(sample_rates);

  780.     if (!formats)

  781.         return AVERROR(ENOMEM);

  782.     return ff_set_common_samplerates(ctx, formats);

  783. }

  784. 

  785. static int load_data(AVFilterContext *ctx, int azim, int elev, float radius)

  786. {

  787.     struct SOFAlizerContext *s = ctx->priv;

  788.     const int n_samples = s->sofa.n_samples;

  789.     int n_conv = s->n_conv; /* no. channels to convolve */

  790.     int n_fft = s->n_fft;

  791.     int delay_l[16]; /* broadband delay for each IR */

  792.     int delay_r[16];

  793.     int nb_input_channels = ctx->inputs[0]->channels; /* no. input channels */

  794.     float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */

  795.     FFTComplex *data_hrtf_l = NULL;

  796.     FFTComplex *data_hrtf_r = NULL;

  797.     FFTComplex *fft_in_l = NULL;

  798.     FFTComplex *fft_in_r = NULL;

  799.     float *data_ir_l = NULL;

  800.     float *data_ir_r = NULL;

  801.     int offset = 0; /* used for faster pointer arithmetics in for-loop */

  802.     int m[16]; /* measurement index m of IR closest to required source positions */

  803.     int i, j, azim_orig = azim, elev_orig = elev;

  804. 

  805.     if (!s->sofa.ncid) { /* if an invalid SOFA file has been selected */

  806.         av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n");

  807.         return AVERROR_INVALIDDATA;

  808.     }

  809. 

  810.     if (s->type == TIME_DOMAIN) {

  811.         s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));

  812.         s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float));

  813. 

  814.         /* get temporary IR for L and R channel */

  815.         data_ir_l = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_l));

  816.         data_ir_r = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_r));

  817.         if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {

  818.             av_free(data_ir_l);

  819.             av_free(data_ir_r);

  820.             return AVERROR(ENOMEM);

  821.         }

  822.     } else {

  823.         /* get temporary HRTF memory for L and R channel */

  824.         data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv);

  825.         data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv);

  826.         if (!data_hrtf_r || !data_hrtf_l) {

  827.             av_free(data_hrtf_l);

  828.             av_free(data_hrtf_r);

  829.             return AVERROR(ENOMEM);

  830.         }

  831.     }

  832. 

  833.     for (i = 0; i < s->n_conv; i++) {

  834.         /* load and store IRs and corresponding delays */

  835.         azim = (int)(s->speaker_azim[i] + azim_orig) % 360;

  836.         elev = (int)(s->speaker_elev[i] + elev_orig) % 90;

  837.         /* get id of IR closest to desired position */

  838.         m[i] = find_m(s, azim, elev, radius);

  839. 

  840.         /* load the delays associated with the current IRs */

  841.         delay_l[i] = *(s->sofa.data_delay + 2 * m[i]);

  842.         delay_r[i] = *(s->sofa.data_delay + 2 * m[i] + 1);

  843. 

  844.         if (s->type == TIME_DOMAIN) {

  845.             offset = i * n_samples; /* no. samples already written */

  846.             for (j = 0; j < n_samples; j++) {

  847.                 /* load reversed IRs of the specified source position

  848.                  * sample-by-sample for left and right ear; and apply gain */

  849.                 *(data_ir_l + offset + j) = /* left channel */

  850.                 *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j) * gain_lin;

  851.                 *(data_ir_r + offset + j) = /* right channel */

  852.                 *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j  + n_samples) * gain_lin;

  853.             }

  854.         } else {

  855.             fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));

  856.             fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r));

  857.             if (!fft_in_l || !fft_in_r) {

  858.                 av_free(data_hrtf_l);

  859.                 av_free(data_hrtf_r);

  860.                 av_free(fft_in_l);

  861.                 av_free(fft_in_r);

  862.                 return AVERROR(ENOMEM);

  863.             }

  864. 

  865.             offset = i * n_fft; /* no. samples already written */

  866.             for (j = 0; j < n_samples; j++) {

  867.                 /* load non-reversed IRs of the specified source position

  868.                  * sample-by-sample and apply gain,

  869.                  * L channel is loaded to real part, R channel to imag part,

  870.                  * IRs ared shifted by L and R delay */

  871.                 fft_in_l[delay_l[i] + j].re = /* left channel */

  872.                 *(s->sofa.data_ir + 2 * m[i] * n_samples + j) * gain_lin;

  873.                 fft_in_r[delay_r[i] + j].re = /* right channel */

  874.                 *(s->sofa.data_ir + (2 * m[i] + 1) * n_samples + j) * gain_lin;

  875.             }

  876. 

  877.             /* actually transform to frequency domain (IRs -> HRTFs) */

  878.             av_fft_permute(s->fft[0], fft_in_l);

  879.             av_fft_calc(s->fft[0], fft_in_l);

  880.             memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l));

  881.             av_fft_permute(s->fft[0], fft_in_r);

  882.             av_fft_calc(s->fft[0], fft_in_r);

  883.             memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r));

  884.         }

  885. 

  886.         av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n",

  887.                m[i], *(s->sofa.sp_a + m[i]), *(s->sofa.sp_e + m[i]), *(s->sofa.sp_r + m[i]));

  888.     }

  889. 

  890.     if (s->type == TIME_DOMAIN) {

  891.         /* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */

  892.         memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * n_samples);

  893.         memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * n_samples);

  894. 

  895.         av_freep(&data_ir_l); /* free temporary IR memory */

  896.         av_freep(&data_ir_r);

  897.     } else {

  898.         s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));

  899.         s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex));

  900.         if (!s->data_hrtf[0] || !s->data_hrtf[1]) {

  901.             av_freep(&data_hrtf_l);

  902.             av_freep(&data_hrtf_r);

  903.             av_freep(&fft_in_l);

  904.             av_freep(&fft_in_r);

  905.             return AVERROR(ENOMEM); /* memory allocation failed */

  906.         }

  907. 

  908.         memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */

  909.             sizeof(FFTComplex) * n_conv * n_fft); /* filter struct */

  910.         memcpy(s->data_hrtf[1], data_hrtf_r,

  911.             sizeof(FFTComplex) * n_conv * n_fft);

  912. 

  913.         av_freep(&data_hrtf_l); /* free temporary HRTF memory */

  914.         av_freep(&data_hrtf_r);

  915. 

  916.         av_freep(&fft_in_l); /* free temporary FFT memory */

  917.         av_freep(&fft_in_r);

  918.     }

  919. 

  920.     memcpy(s->delay[0], &delay_l[0], sizeof(int) * s->n_conv);

  921.     memcpy(s->delay[1], &delay_r[0], sizeof(int) * s->n_conv);

  922. 

  923.     return 0;

  924. }

  925. 

  926. static av_cold int init(AVFilterContext *ctx)

  927. {

  928.     SOFAlizerContext *s = ctx->priv;

  929.     int ret;

  930. 

  931.     /* load SOFA file, */

  932.     /* initialize file IDs to 0 before attempting to load SOFA files,

  933.      * this assures that in case of error, only the memory of already

  934.      * loaded files is free'd */

  935.     s->sofa.ncid = 0;

  936.     ret = load_sofa(ctx, s->filename, &s->sample_rate);

  937.     if (ret) {

  938.         /* file loading error */

  939.         av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename);

  940.     } else { /* no file loading error, resampling not required */

  941.         av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename);

  942.     }

  943. 

  944.     if (ret) {

  945.         av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n");

  946.         return ret;

  947.     }

  948. 

  949.     s->fdsp = avpriv_float_dsp_alloc(0);

  950.     if (!s->fdsp)

  951.         return AVERROR(ENOMEM);

  952. 

  953.     return 0;

  954. }

  955. 

  956. static int config_input(AVFilterLink *inlink)

  957. {

  958.     AVFilterContext *ctx = inlink->dst;

  959.     SOFAlizerContext *s = ctx->priv;

  960.     int nb_input_channels = inlink->channels; /* no. input channels */

  961.     int n_max_ir = 0;

  962.     int n_current;

  963.     int n_max = 0;

  964.     int ret;

  965. 

  966.     if (s->type == FREQUENCY_DOMAIN) {

  967.         inlink->partial_buf_size =

  968.         inlink->min_samples =

  969.         inlink->max_samples = inlink->sample_rate;

  970.     }

  971. 

  972.     /* gain -3 dB per channel, -6 dB to get LFE on a similar level */

  973.     s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6) / 20 * M_LN10);

  974. 

  975.     s->n_conv = nb_input_channels;

  976. 

  977.     /* get size of ringbuffer (longest IR plus max. delay) */

  978.     /* then choose next power of 2 for performance optimization */

  979.     n_current = s->sofa.n_samples + max_delay(&s->sofa);

  980.     if (n_current > n_max) {

  981.         /* length of longest IR plus max. delay (in all SOFA files) */

  982.         n_max = n_current;

  983.         /* length of longest IR (without delay, in all SOFA files) */

  984.         n_max_ir = s->sofa.n_samples;

  985.     }

  986.     /* buffer length is longest IR plus max. delay -> next power of 2

  987.        (32 - count leading zeros gives required exponent)  */

  988.     s->buffer_length = 1 << (32 - ff_clz(n_max));

  989.     s->n_fft         = 1 << (32 - ff_clz(n_max + inlink->sample_rate));

  990. 

  991.     if (s->type == FREQUENCY_DOMAIN) {

  992.         av_fft_end(s->fft[0]);

  993.         av_fft_end(s->fft[1]);

  994.         s->fft[0] = av_fft_init(log2(s->n_fft), 0);

  995.         s->fft[1] = av_fft_init(log2(s->n_fft), 0);

  996.         av_fft_end(s->ifft[0]);

  997.         av_fft_end(s->ifft[1]);

  998.         s->ifft[0] = av_fft_init(log2(s->n_fft), 1);

  999.         s->ifft[1] = av_fft_init(log2(s->n_fft), 1);

  1000. 

  1001.         if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {

  1002.             av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts.\n");

  1003.             return AVERROR(ENOMEM);

  1004.         }

  1005.     }

  1006. 

  1007.     /* Allocate memory for the impulse responses, delays and the ringbuffers */

  1008.     /* size: (longest IR) * (number of channels to convolute) */

  1009.     s->data_ir[0] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv);

  1010.     s->data_ir[1] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv);

  1011.     /* length:  number of channels to convolute */

  1012.     s->delay[0] = av_malloc_array(s->n_conv, sizeof(float));

  1013.     s->delay[1] = av_malloc_array(s->n_conv, sizeof(float));

  1014.     /* length: (buffer length) * (number of input channels),

  1015.      * OR: buffer length (if frequency domain processing)

  1016.      * calloc zero-initializes the buffer */

  1017. 

  1018.     if (s->type == TIME_DOMAIN) {

  1019.         s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);

  1020.         s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);

  1021.     } else {

  1022.         s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));

  1023.         s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));

  1024.         s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));

  1025.         s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));

  1026.         if (!s->temp_fft[0] || !s->temp_fft[1])

  1027.             return AVERROR(ENOMEM);

  1028.     }

  1029. 

  1030.     /* length: number of channels to convolute */

  1031.     s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim));

  1032.     s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev));

  1033. 

  1034.     /* memory allocation failed: */

  1035.     if (!s->data_ir[0] || !s->data_ir[1] || !s->delay[1] ||

  1036.         !s->delay[0] || !s->ringbuffer[0] || !s->ringbuffer[1] ||

  1037.         !s->speaker_azim || !s->speaker_elev)

  1038.         return AVERROR(ENOMEM);

  1039. 

  1040.     compensate_volume(ctx);

  1041. 

  1042.     /* get speaker positions */

  1043.     if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 0) {

  1044.         av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");

  1045.         return ret;

  1046.     }

  1047. 

  1048.     /* load IRs to data_ir[0] and data_ir[1] for required directions */

  1049.     if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius)) < 0)

  1050.         return ret;

  1051. 

  1052.     av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n",

  1053.         inlink->sample_rate, s->n_conv, nb_input_channels, s->buffer_length);

  1054. 

  1055.     return 0;

  1056. }

  1057. 

  1058. static av_cold void uninit(AVFilterContext *ctx)

  1059. {

  1060.     SOFAlizerContext *s = ctx->priv;

  1061. 

  1062.     if (s->sofa.ncid) {

  1063.         av_freep(&s->sofa.sp_a);

  1064.         av_freep(&s->sofa.sp_e);

  1065.         av_freep(&s->sofa.sp_r);

  1066.         av_freep(&s->sofa.data_delay);

  1067.         av_freep(&s->sofa.data_ir);

  1068.     }

  1069.     av_fft_end(s->ifft[0]);

  1070.     av_fft_end(s->ifft[1]);

  1071.     av_fft_end(s->fft[0]);

  1072.     av_fft_end(s->fft[1]);

  1073.     av_freep(&s->delay[0]);

  1074.     av_freep(&s->delay[1]);

  1075.     av_freep(&s->data_ir[0]);

  1076.     av_freep(&s->data_ir[1]);

  1077.     av_freep(&s->ringbuffer[0]);

  1078.     av_freep(&s->ringbuffer[1]);

  1079.     av_freep(&s->speaker_azim);

  1080.     av_freep(&s->speaker_elev);

  1081.     av_freep(&s->temp_src[0]);

  1082.     av_freep(&s->temp_src[1]);

  1083.     av_freep(&s->temp_fft[0]);

  1084.     av_freep(&s->temp_fft[1]);

  1085.     av_freep(&s->data_hrtf[0]);

  1086.     av_freep(&s->data_hrtf[1]);

  1087.     av_freep(&s->fdsp);

  1088. }

  1089. 

  1090. #define OFFSET(x) offsetof(SOFAlizerContext, x)

  1091. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

  1092. 

  1093. static const AVOption sofalizer_options[] = {

  1094.     { "sofa",      "sofa filename",  OFFSET(filename),  AV_OPT_TYPE_STRING, {.str=NULL},            .flags = FLAGS },

  1095.     { "gain",      "set gain in dB", OFFSET(gain),      AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },

  1096.     { "rotation",  "set rotation"  , OFFSET(rotation),  AV_OPT_TYPE_FLOAT,  {.dbl=0},    -360, 360, .flags = FLAGS },

  1097.     { "elevation", "set elevation",  OFFSET(elevation), AV_OPT_TYPE_FLOAT,  {.dbl=0},     -90,  90, .flags = FLAGS },

  1098.     { "radius",    "set radius",     OFFSET(radius),    AV_OPT_TYPE_FLOAT,  {.dbl=1},       0,   3, .flags = FLAGS },

  1099.     { "type",      "set processing", OFFSET(type),      AV_OPT_TYPE_INT,    {.i64=1},       0,   1, .flags = FLAGS, "type" },

  1100.     { "time",      "time domain",      0,               AV_OPT_TYPE_CONST,  {.i64=0},       0,   0, .flags = FLAGS, "type" },

  1101.     { "freq",      "frequency domain", 0,               AV_OPT_TYPE_CONST,  {.i64=1},       0,   0, .flags = FLAGS, "type" },

  1102.     { NULL }

  1103. };

  1104. 

  1105. AVFILTER_DEFINE_CLASS(sofalizer);

  1106. 

  1107. static const AVFilterPad inputs[] = {

  1108.     {

  1109.         .name         = "default",

  1110.         .type         = AVMEDIA_TYPE_AUDIO,

  1111.         .config_props = config_input,

  1112.         .filter_frame = filter_frame,

  1113.     },

  1114.     { NULL }

  1115. };

  1116. 

  1117. static const AVFilterPad outputs[] = {

  1118.     {

  1119.         .name = "default",

  1120.         .type = AVMEDIA_TYPE_AUDIO,

  1121.     },

  1122.     { NULL }

  1123. };

  1124. 

  1125. AVFilter ff_af_sofalizer = {

  1126.     .name          = "sofalizer",

  1127.     .description   = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."),

  1128.     .priv_size     = sizeof(SOFAlizerContext),

  1129.     .priv_class    = &sofalizer_class,

  1130.     .init          = init,

  1131.     .uninit        = uninit,

  1132.     .query_formats = query_formats,

  1133.     .inputs        = inputs,

  1134.     .outputs       = outputs,

  1135.     .flags         = AVFILTER_FLAG_SLICE_THREADS,

  1136. };