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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/opt.h"

  34. #include "avfilter.h"

  35. #include "internal.h"

  36. #include "audio.h"

  37. 

  38. #define TIME_DOMAIN      0

  39. #define FREQUENCY_DOMAIN 1

  40. 

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

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

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

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

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

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

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

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

  49.     float *sp_r;         /* radii */

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

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

  52. } NCSofa;

  53. 

  54. typedef struct SOFAlizerContext {

  55.     const AVClass *class;

  56. 

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

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

  59. 

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

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

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

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

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

  65. 

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

  67. 

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

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

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

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

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

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

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

  75. 

  76.                                 /* netCDF variables */

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

  78. 

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

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

  81.     float *temp_src[2];

  82.     FFTComplex *temp_fft[2];

  83. 

  84.                          /* control variables */

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

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

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

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

  89.     int type;            /* processing type */

  90. 

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

  92.     FFTComplex *data_hrtf[2];

  93. 

  94.     AVFloatDSPContext *fdsp;

  95. } SOFAlizerContext;

  96. 

  97. static int close_sofa(struct NCSofa *sofa)

  98. {

  99.     av_freep(&sofa->data_delay);

  100.     av_freep(&sofa->sp_a);

  101.     av_freep(&sofa->sp_e);

  102.     av_freep(&sofa->sp_r);

  103.     av_freep(&sofa->data_ir);

  104.     nc_close(sofa->ncid);

  105.     sofa->ncid = 0;

  106. 

  107.     return 0;

  108. }

  109. 

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

  111. {

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

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

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

  115.     char data_delay_dim_name[NC_MAX_NAME];

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

  117.     char *sofa_conventions;

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

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

  120.     char *text;

  121.     unsigned int sample_rate;

  122.     int data_delay_dim_id[2];

  123.     int samplingrate_id;

  124.     int data_delay_id;

  125.     int n_samples;

  126.     int m_dim_id = -1;

  127.     int n_dim_id = -1;

  128.     int data_ir_id;

  129.     size_t att_len;

  130.     int m_dim;

  131.     int *data_delay;

  132.     int sp_id;

  133.     int i, ret;

  134. 

  135.     s->sofa.ncid = 0;

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

  137.     if (status != NC_NOERR) {

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

  139.         return AVERROR(EINVAL);

  140.     }

  141. 

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

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

  144. 

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

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

  147.     if (!dim_length) {

  148.         nc_close(ncid);

  149.         return AVERROR(ENOMEM);

  150.     }

  151. 

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

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

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

  155.             m_dim_id = i;

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

  157.             n_dim_id = i;

  158.     }

  159. 

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

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

  162.         av_freep(&dim_length);

  163.         nc_close(ncid);

  164.         return AVERROR(EINVAL);

  165.     }

  166. 

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

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

  169. 

  170.     av_freep(&dim_length);

  171. 

  172.     /* -- check file type -- */

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

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

  175.     if (status != NC_NOERR) {

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

  177.         nc_close(ncid);

  178.         return AVERROR_INVALIDDATA;

  179.     }

  180. 

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

  182.     text = av_malloc(att_len + 1);

  183.     if (!text) {

  184.         nc_close(ncid);

  185.         return AVERROR(ENOMEM);

  186.     }

  187. 

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

  189.     *(text + att_len) = 0;

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

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

  192.         av_freep(&text);

  193.         nc_close(ncid);

  194.         return AVERROR(EINVAL);

  195.     }

  196.     av_freep(&text);

  197. 

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

  199.     if (status == NC_NOERR) {

  200.         text = av_malloc(att_len + 1);

  201.         if (text) {

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

  203.             *(text + att_len) = 0;

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

  205.             av_freep(&text);

  206.         }

  207.     }

  208. 

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

  210.     if (status == NC_NOERR) {

  211.         text = av_malloc(att_len + 1);

  212.         if (text) {

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

  214.             *(text + att_len) = 0;

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

  216.             av_freep(&text);

  217.         }

  218.     }

  219. 

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

  221.     if (status == NC_NOERR) {

  222.         text = av_malloc(att_len + 1);

  223.         if (text) {

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

  225.             *(text + att_len) = 0;

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

  227.             av_freep(&text);

  228.         }

  229.     }

  230. 

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

  232.     if (status != NC_NOERR) {

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

  234.         nc_close(ncid);

  235.         return AVERROR_INVALIDDATA;

  236.     }

  237. 

  238.     sofa_conventions = av_malloc(att_len + 1);

  239.     if (!sofa_conventions) {

  240.         nc_close(ncid);

  241.         return AVERROR(ENOMEM);

  242.     }

  243. 

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

  245.     *(sofa_conventions + att_len) = 0;

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

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

  248.         av_freep(&sofa_conventions);

  249.         nc_close(ncid);

  250.         return AVERROR(EINVAL);

  251.     }

  252.     av_freep(&sofa_conventions);

  253. 

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

  255.     /* read ID, then value */

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

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

  258.     if (status != NC_NOERR) {

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

  260.         nc_close(ncid);

  261.         return AVERROR(EINVAL);

  262.     }

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

  264. 

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

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

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

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

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

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

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

  272. 

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

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

  275.         close_sofa(&s->sofa);

  276.         return AVERROR(ENOMEM);

  277.     }

  278. 

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

  280.     /* get corresponding ID */

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

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

  283.     if (status != NC_NOERR) {

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

  285.         ret = AVERROR(EINVAL);

  286.         goto error;

  287.     }

  288. 

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

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

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

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

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

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

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

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

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

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

  299.         ret = AVERROR(EINVAL);

  300.         goto error;

  301.     }

  302. 

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

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

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

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

  307.     if (status != NC_NOERR) {

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

  309.         ret = AVERROR(EINVAL);

  310.         goto error;

  311.     }

  312. 

  313.     /* Data.Delay dimension check */

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

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

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

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

  318. 

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

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

  321.         if (status != NC_NOERR) {

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

  323.             ret = AVERROR(EINVAL);

  324.             goto error;

  325.         }

  326.         int *data_delay_r = data_delay + m_dim;

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

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

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

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

  331.         }

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

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

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

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

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

  337.         if (status != NC_NOERR) {

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

  339.             ret = AVERROR(EINVAL);

  340.             goto error;

  341.         }

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

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

  344.         ret = AVERROR(EINVAL);

  345.         goto error;

  346.     }

  347. 

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

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

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

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

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

  353. 

  354.     return 0;

  355. 

  356. error:

  357.     close_sofa(&s->sofa);

  358.     return ret;

  359. }

  360. 

  361. static int get_speaker_pos(AVFilterContext *ctx,

  362.                            float *speaker_azim, float *speaker_elev)

  363. {

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

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

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

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

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

  369. 

  370.     if (n_conv > 16)

  371.         return AVERROR(EINVAL);

  372. 

  373.     s->lfe_channel = -1;

  374. 

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

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

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

  378. 

  379.         switch (mask) {

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

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

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

  383.         case AV_CH_LOW_FREQUENCY:

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

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

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

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

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

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

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

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

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

  393.                                           elev[ch] =  90;      break;

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

  395.                                           elev[ch] =  45;      break;

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

  397.                                           elev[ch] =  45;      break;

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

  399.                                           elev[ch] =  45;      break;

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

  401.                                           elev[ch] =  45;      break;

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

  403.                                           elev[ch] =  45;      break;

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

  405.                                           elev[ch] =  45;      break;

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

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

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

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

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

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

  412.         case 0:                                                break;

  413.         default:

  414.             return AVERROR(EINVAL);

  415.         }

  416.         if (mask)

  417.             ch++;

  418.     }

  419. 

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

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

  422. 

  423.     return 0;

  424. 

  425. }

  426. 

  427. static int max_delay(struct NCSofa *sofa)

  428. {

  429.     int i, max = 0;

  430. 

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

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

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

  434.     }

  435. 

  436.     return max;

  437. }

  438. 

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

  440. {

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

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

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

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

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

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

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

  448.     float current;

  449.     int i;

  450. 

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

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

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

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

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

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

  457.         if (current <= delta) {

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

  459.             delta = current;

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

  461.         }

  462.     }

  463. 

  464.     return best_id;

  465. }

  466. 

  467. static int compensate_volume(AVFilterContext *ctx)

  468. {

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

  470.     float compensate;

  471.     float energy = 0;

  472.     float *ir;

  473.     int m;

  474. 

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

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

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

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

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

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

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

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

  483.         } else {

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

  485.         }

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

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

  488.         ir = sofa->data_ir;

  489.         /* apply volume compensation to IRs */

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

  491.         emms_c();

  492.     }

  493. 

  494.     return 0;

  495. }

  496. 

  497. typedef struct ThreadData {

  498.     AVFrame *in, *out;

  499.     int *write;

  500.     int **delay;

  501.     float **ir;

  502.     int *n_clippings;

  503.     float **ringbuffer;

  504.     float **temp_src;

  505.     FFTComplex **temp_fft;

  506. } ThreadData;

  507. 

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

  509. {

  510.     SOFAlizerContext *s = ctx->priv;

  511.     ThreadData *td = arg;

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

  513.     int offset = jobnr;

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

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

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

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

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

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

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

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

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

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

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

  525.     const int buffer_length = s->buffer_length;

  526.     /* -1 for AND instead of MODULO (applied to powers of 2): */

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

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

  529.     int wr = *write;

  530.     int read;

  531.     int i, l;

  532. 

  533.     dst += offset;

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

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

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

  537.     }

  538. 

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

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

  541. 

  542.         *dst = 0;

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

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

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

  546.         }

  547. 

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

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

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

  551. 

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

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

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

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

  556.                 temp_ir += n_samples;

  557.                 continue;

  558.             }

  559. 

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

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

  562.              * (mod buffer length) */

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

  564. 

  565.             if (read + n_samples < buffer_length) {

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

  567.             } else {

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

  569. 

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

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

  572.             }

  573. 

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

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

  576.             temp_ir += n_samples;

  577.         }

  578. 

  579.         /* clippings counter */

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

  581.             *n_clippings += 1;

  582. 

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

  584.         dst += 2;

  585.         src += in_channels;

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

  587.     }

  588. 

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

  590. 

  591.     return 0;

  592. }

  593. 

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

  595. {

  596.     SOFAlizerContext *s = ctx->priv;

  597.     ThreadData *td = arg;

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

  599.     int offset = jobnr;

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

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

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

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

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

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

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

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

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

  609.     const int buffer_length = s->buffer_length;

  610.     /* -1 for AND instead of MODULO (applied to powers of 2): */

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

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

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

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

  615.     const int n_conv = s->n_conv;

  616.     const int n_fft = s->n_fft;

  617.     int wr = *write;

  618.     int n_read;

  619.     int i, j;

  620. 

  621.     dst += offset;

  622. 

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

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

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

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

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

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

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

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

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

  632.     }

  633. 

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

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

  636.         dst[2 * j] = 0;

  637.     }

  638. 

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

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

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

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

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

  644.             }

  645.             continue;

  646.         }

  647. 

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

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

  650. 

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

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

  653. 

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

  655.             /* prepare input for FFT */

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

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

  658.         }

  659. 

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

  661.         av_fft_permute(fft, fft_in);

  662.         av_fft_calc(fft, fft_in);

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

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

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

  666. 

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

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

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

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

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

  672.         }

  673. 

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

  675.         av_fft_permute(ifft, fft_in);

  676.         av_fft_calc(ifft, fft_in);

  677. 

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

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

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

  681.         }

  682. 

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

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

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

  686. 

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

  688.         }

  689.     }

  690. 

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

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

  693.         /* clippings counter */

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

  695.             *n_clippings = *n_clippings + 1;

  696.         }

  697. 

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

  699.         dst += 2;

  700.     }

  701. 

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

  703.     *write = wr;

  704. 

  705.     return 0;

  706. }

  707. 

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

  709. {

  710.     AVFilterContext *ctx = inlink->dst;

  711.     SOFAlizerContext *s = ctx->priv;

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

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

  714.     ThreadData td;

  715.     AVFrame *out;

  716. 

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

  718.     if (!out) {

  719.         av_frame_free(&in);

  720.         return AVERROR(ENOMEM);

  721.     }

  722.     av_frame_copy_props(out, in);

  723. 

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

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

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

  727.     td.temp_fft = s->temp_fft;

  728. 

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

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

  731.     } else {

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

  733.     }

  734.     emms_c();

  735. 

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

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

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

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

  740.     }

  741. 

  742.     av_frame_free(&in);

  743.     return ff_filter_frame(outlink, out);

  744. }

  745. 

  746. static int query_formats(AVFilterContext *ctx)

  747. {

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

  749.     AVFilterFormats *formats = NULL;

  750.     AVFilterChannelLayouts *layouts = NULL;

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

  752. 

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

  754.     if (ret)

  755.         return ret;

  756.     ret = ff_set_common_formats(ctx, formats);

  757.     if (ret)

  758.         return ret;

  759. 

  760.     layouts = ff_all_channel_layouts();

  761.     if (!layouts)

  762.         return AVERROR(ENOMEM);

  763. 

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

  765.     if (ret)

  766.         return ret;

  767. 

  768.     layouts = NULL;

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

  770.     if (ret)

  771.         return ret;

  772. 

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

  774.     if (ret)

  775.         return ret;

  776. 

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

  778.     formats = ff_make_format_list(sample_rates);

  779.     if (!formats)

  780.         return AVERROR(ENOMEM);

  781.     return ff_set_common_samplerates(ctx, formats);

  782. }

  783. 

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

  785. {

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

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

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

  789.     int n_fft = s->n_fft;

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

  791.     int delay_r[16];

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

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

  794.     FFTComplex *data_hrtf_l = NULL;

  795.     FFTComplex *data_hrtf_r = NULL;

  796.     FFTComplex *fft_in_l = NULL;

  797.     FFTComplex *fft_in_r = NULL;

  798.     float *data_ir_l = NULL;

  799.     float *data_ir_r = NULL;

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

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

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

  803. 

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

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

  806.         return AVERROR_INVALIDDATA;

  807.     }

  808. 

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

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

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

  812. 

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

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

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

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

  817.             av_free(data_ir_l);

  818.             av_free(data_ir_r);

  819.             return AVERROR(ENOMEM);

  820.         }

  821.     } else {

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

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

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

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

  826.             av_free(data_hrtf_l);

  827.             av_free(data_hrtf_r);

  828.             return AVERROR(ENOMEM);

  829.         }

  830.     }

  831. 

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

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

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

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

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

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

  838. 

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

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

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

  842. 

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

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

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

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

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

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

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

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

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

  852.             }

  853.         } else {

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

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

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

  857.                 av_free(data_hrtf_l);

  858.                 av_free(data_hrtf_r);

  859.                 av_free(fft_in_l);

  860.                 av_free(fft_in_r);

  861.                 return AVERROR(ENOMEM);

  862.             }

  863. 

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

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

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

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

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

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

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

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

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

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

  874.             }

  875. 

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

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

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

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

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

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

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

  883.         }

  884. 

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

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

  887.     }

  888. 

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

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

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

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

  893. 

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

  895.         av_freep(&data_ir_r);

  896.     } else {

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

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

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

  900.             av_freep(&data_hrtf_l);

  901.             av_freep(&data_hrtf_r);

  902.             av_freep(&fft_in_l);

  903.             av_freep(&fft_in_r);

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

  905.         }

  906. 

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

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

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

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

  911. 

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

  913.         av_freep(&data_hrtf_r);

  914. 

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

  916.         av_freep(&fft_in_r);

  917.     }

  918. 

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

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

  921. 

  922.     return 0;

  923. }

  924. 

  925. static av_cold int init(AVFilterContext *ctx)

  926. {

  927.     SOFAlizerContext *s = ctx->priv;

  928.     int ret;

  929. 

  930.     /* load SOFA file, */

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

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

  933.      * loaded files is free'd */

  934.     s->sofa.ncid = 0;

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

  936.     if (ret) {

  937.         /* file loading error */

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

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

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

  941.     }

  942. 

  943.     if (ret) {

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

  945.         return ret;

  946.     }

  947. 

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

  949.     if (!s->fdsp)

  950.         return AVERROR(ENOMEM);

  951. 

  952.     return 0;

  953. }

  954. 

  955. static inline unsigned clz(unsigned x)

  956. {

  957.     unsigned i = sizeof(x) * 8;

  958. 

  959.     while (x) {

  960.         x >>= 1;

  961.         i--;

  962.     }

  963. 

  964.     return i;

  965. }

  966. 

  967. static int config_input(AVFilterLink *inlink)

  968. {

  969.     AVFilterContext *ctx = inlink->dst;

  970.     SOFAlizerContext *s = ctx->priv;

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

  972.     int n_max_ir = 0;

  973.     int n_current;

  974.     int n_max = 0;

  975.     int ret;

  976. 

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

  978.         inlink->partial_buf_size =

  979.         inlink->min_samples =

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

  981.     }

  982. 

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

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

  985. 

  986.     s->n_conv = nb_input_channels;

  987. 

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

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

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

  991.     if (n_current > n_max) {

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

  993.         n_max = n_current;

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

  995.         n_max_ir = s->sofa.n_samples;

  996.     }

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

  998.        (32 - count leading zeros gives required exponent)  */

  999.     s->buffer_length = exp2(32 - clz((uint32_t)n_max));

  1000.     s->n_fft         = exp2(32 - clz((uint32_t)(n_max + inlink->sample_rate)));

  1001. 

  1002.     if (s->type == FREQUENCY_DOMAIN) {

  1003.         av_fft_end(s->fft[0]);

  1004.         av_fft_end(s->fft[1]);

  1005.         s->fft[0] = av_fft_init(log2(s->n_fft), 0);

  1006.         s->fft[1] = av_fft_init(log2(s->n_fft), 0);

  1007.         av_fft_end(s->ifft[0]);

  1008.         av_fft_end(s->ifft[1]);

  1009.         s->ifft[0] = av_fft_init(log2(s->n_fft), 1);

  1010.         s->ifft[1] = av_fft_init(log2(s->n_fft), 1);

  1011. 

  1012.         if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {

  1013.             av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts.\n");

  1014.             return AVERROR(ENOMEM);

  1015.         }

  1016.     }

  1017. 

  1018.     /* Allocate memory for the impulse responses, delays and the ringbuffers */

  1019.     /* size: (longest IR) * (number of channels to convolute) */

  1020.     s->data_ir[0] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv);

  1021.     s->data_ir[1] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv);

  1022.     /* length:  number of channels to convolute */

  1023.     s->delay[0] = av_malloc_array(s->n_conv, sizeof(float));

  1024.     s->delay[1] = av_malloc_array(s->n_conv, sizeof(float));

  1025.     /* length: (buffer length) * (number of input channels),

  1026.      * OR: buffer length (if frequency domain processing)

  1027.      * calloc zero-initializes the buffer */

  1028. 

  1029.     if (s->type == TIME_DOMAIN) {

  1030.         s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);

  1031.         s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels);

  1032.     } else {

  1033.         s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float));

  1034.         s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));

  1035.         s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex));

  1036.         s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex));

  1037.         if (!s->temp_fft[0] || !s->temp_fft[1])

  1038.             return AVERROR(ENOMEM);

  1039.     }

  1040. 

  1041.     /* length: number of channels to convolute */

  1042.     s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim));

  1043.     s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev));

  1044. 

  1045.     /* memory allocation failed: */

  1046.     if (!s->data_ir[0] || !s->data_ir[1] || !s->delay[1] ||

  1047.         !s->delay[0] || !s->ringbuffer[0] || !s->ringbuffer[1] ||

  1048.         !s->speaker_azim || !s->speaker_elev)

  1049.         return AVERROR(ENOMEM);

  1050. 

  1051.     compensate_volume(ctx);

  1052. 

  1053.     /* get speaker positions */

  1054.     if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 0) {

  1055.         av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n");

  1056.         return ret;

  1057.     }

  1058. 

  1059.     /* load IRs to data_ir[0] and data_ir[1] for required directions */

  1060.     if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius)) < 0)

  1061.         return ret;

  1062. 

  1063.     av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n",

  1064.         inlink->sample_rate, s->n_conv, nb_input_channels, s->buffer_length);

  1065. 

  1066.     return 0;

  1067. }

  1068. 

  1069. static av_cold void uninit(AVFilterContext *ctx)

  1070. {

  1071.     SOFAlizerContext *s = ctx->priv;

  1072. 

  1073.     if (s->sofa.ncid) {

  1074.         av_freep(&s->sofa.sp_a);

  1075.         av_freep(&s->sofa.sp_e);

  1076.         av_freep(&s->sofa.sp_r);

  1077.         av_freep(&s->sofa.data_delay);

  1078.         av_freep(&s->sofa.data_ir);

  1079.     }

  1080.     av_fft_end(s->ifft[0]);

  1081.     av_fft_end(s->ifft[1]);

  1082.     av_fft_end(s->fft[0]);

  1083.     av_fft_end(s->fft[1]);

  1084.     av_freep(&s->delay[0]);

  1085.     av_freep(&s->delay[1]);

  1086.     av_freep(&s->data_ir[0]);

  1087.     av_freep(&s->data_ir[1]);

  1088.     av_freep(&s->ringbuffer[0]);

  1089.     av_freep(&s->ringbuffer[1]);

  1090.     av_freep(&s->speaker_azim);

  1091.     av_freep(&s->speaker_elev);

  1092.     av_freep(&s->temp_src[0]);

  1093.     av_freep(&s->temp_src[1]);

  1094.     av_freep(&s->temp_fft[0]);

  1095.     av_freep(&s->temp_fft[1]);

  1096.     av_freep(&s->data_hrtf[0]);

  1097.     av_freep(&s->data_hrtf[1]);

  1098.     av_freep(&s->fdsp);

  1099. }

  1100. 

  1101. #define OFFSET(x) offsetof(SOFAlizerContext, x)

  1102. #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

  1103. 

  1104. static const AVOption sofalizer_options[] = {

  1105.     { "sofa",      "sofa filename",  OFFSET(filename),  AV_OPT_TYPE_STRING, {.str=NULL},            .flags = FLAGS },

  1106.     { "gain",      "set gain in dB", OFFSET(gain),      AV_OPT_TYPE_FLOAT,  {.dbl=0},     -20,  40, .flags = FLAGS },

  1107.     { "rotation",  "set rotation"  , OFFSET(rotation),  AV_OPT_TYPE_FLOAT,  {.dbl=0},    -360, 360, .flags = FLAGS },

  1108.     { "elevation", "set elevation",  OFFSET(elevation), AV_OPT_TYPE_FLOAT,  {.dbl=0},     -90,  90, .flags = FLAGS },

  1109.     { "radius",    "set radius",     OFFSET(radius),    AV_OPT_TYPE_FLOAT,  {.dbl=1},       0,   3, .flags = FLAGS },

  1110.     { "type",      "set processing", OFFSET(type),      AV_OPT_TYPE_INT,    {.i64=1},       0,   1, .flags = FLAGS, "type" },

  1111.     { "time",      "time domain",      0,               AV_OPT_TYPE_CONST,  {.i64=0},       0,   0, .flags = FLAGS, "type" },

  1112.     { "freq",      "frequency domain", 0,               AV_OPT_TYPE_CONST,  {.i64=1},       0,   0, .flags = FLAGS, "type" },

  1113.     { NULL }

  1114. };

  1115. 

  1116. AVFILTER_DEFINE_CLASS(sofalizer);

  1117. 

  1118. static const AVFilterPad inputs[] = {

  1119.     {

  1120.         .name         = "default",

  1121.         .type         = AVMEDIA_TYPE_AUDIO,

  1122.         .config_props = config_input,

  1123.         .filter_frame = filter_frame,

  1124.     },

  1125.     { NULL }

  1126. };

  1127. 

  1128. static const AVFilterPad outputs[] = {

  1129.     {

  1130.         .name = "default",

  1131.         .type = AVMEDIA_TYPE_AUDIO,

  1132.     },

  1133.     { NULL }

  1134. };

  1135. 

  1136. AVFilter ff_af_sofalizer = {

  1137.     .name          = "sofalizer",

  1138.     .description   = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."),

  1139.     .priv_size     = sizeof(SOFAlizerContext),

  1140.     .priv_class    = &sofalizer_class,

  1141.     .init          = init,

  1142.     .uninit        = uninit,

  1143.     .query_formats = query_formats,

  1144.     .inputs        = inputs,

  1145.     .outputs       = outputs,

  1146.     .flags         = AVFILTER_FLAG_SLICE_THREADS,

  1147. };