| /***************************************************************************** |
| * sofalizer.c : SOFAlizer filter for virtual binaural acoustics |
| ***************************************************************************** |
| * Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda, |
| * Acoustics Research Institute (ARI), Vienna, Austria |
| * |
| * Authors: Andreas Fuchs <andi.fuchs.mail@gmail.com> |
| * Wolfgang Hrauda <wolfgang.hrauda@gmx.at> |
| * |
| * SOFAlizer project coordinator at ARI, main developer of SOFA: |
| * Piotr Majdak <piotr@majdak.at> |
| * |
| * This program is free software; you can redistribute it and/or modify it |
| * under the terms of the GNU Lesser General Public License as published by |
| * the Free Software Foundation; either version 2.1 of the License, or |
| * (at your option) any later version. |
| * |
| * This program is distributed in the hope that it will be useful, |
| * but WITHOUT ANY WARRANTY; without even the implied warranty of |
| * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
| * GNU Lesser General Public License for more details. |
| * |
| * You should have received a copy of the GNU Lesser General Public License |
| * along with this program; if not, write to the Free Software Foundation, |
| * Inc., 51 Franklin Street, Fifth Floor, Boston MA 02110-1301, USA. |
| *****************************************************************************/ |
| |
| #include <math.h> |
| #include <netcdf.h> |
| |
| #include "libavcodec/avfft.h" |
| #include "libavutil/float_dsp.h" |
| #include "libavutil/intmath.h" |
| #include "libavutil/opt.h" |
| #include "avfilter.h" |
| #include "internal.h" |
| #include "audio.h" |
| |
| #define TIME_DOMAIN 0 |
| #define FREQUENCY_DOMAIN 1 |
| |
| typedef struct NCSofa { /* contains data of one SOFA file */ |
| int ncid; /* netCDF ID of the opened SOFA file */ |
| int n_samples; /* length of one impulse response (IR) */ |
| int m_dim; /* number of measurement positions */ |
| int *data_delay; /* broadband delay of each IR */ |
| /* all measurement positions for each receiver (i.e. ear): */ |
| float *sp_a; /* azimuth angles */ |
| float *sp_e; /* elevation angles */ |
| float *sp_r; /* radii */ |
| /* data at each measurement position for each receiver: */ |
| float *data_ir; /* IRs (time-domain) */ |
| } NCSofa; |
| |
| typedef struct SOFAlizerContext { |
| const AVClass *class; |
| |
| char *filename; /* name of SOFA file */ |
| NCSofa sofa; /* contains data of the SOFA file */ |
| |
| int sample_rate; /* sample rate from SOFA file */ |
| float *speaker_azim; /* azimuth of the virtual loudspeakers */ |
| float *speaker_elev; /* elevation of the virtual loudspeakers */ |
| float gain_lfe; /* gain applied to LFE channel */ |
| int lfe_channel; /* LFE channel position in channel layout */ |
| |
| int n_conv; /* number of channels to convolute */ |
| |
| /* buffer variables (for convolution) */ |
| float *ringbuffer[2]; /* buffers input samples, length of one buffer: */ |
| /* no. input ch. (incl. LFE) x buffer_length */ |
| int write[2]; /* current write position to ringbuffer */ |
| int buffer_length; /* is: longest IR plus max. delay in all SOFA files */ |
| /* then choose next power of 2 */ |
| int n_fft; /* number of samples in one FFT block */ |
| |
| /* netCDF variables */ |
| int *delay[2]; /* broadband delay for each channel/IR to be convolved */ |
| |
| float *data_ir[2]; /* IRs for all channels to be convolved */ |
| /* (this excludes the LFE) */ |
| float *temp_src[2]; |
| FFTComplex *temp_fft[2]; |
| |
| /* control variables */ |
| float gain; /* filter gain (in dB) */ |
| float rotation; /* rotation of virtual loudspeakers (in degrees) */ |
| float elevation; /* elevation of virtual loudspeakers (in deg.) */ |
| float radius; /* distance virtual loudspeakers to listener (in metres) */ |
| int type; /* processing type */ |
| |
| FFTContext *fft[2], *ifft[2]; |
| FFTComplex *data_hrtf[2]; |
| |
| AVFloatDSPContext *fdsp; |
| } SOFAlizerContext; |
| |
| static int close_sofa(struct NCSofa *sofa) |
| { |
| av_freep(&sofa->data_delay); |
| av_freep(&sofa->sp_a); |
| av_freep(&sofa->sp_e); |
| av_freep(&sofa->sp_r); |
| av_freep(&sofa->data_ir); |
| nc_close(sofa->ncid); |
| sofa->ncid = 0; |
| |
| return 0; |
| } |
| |
| static int load_sofa(AVFilterContext *ctx, char *filename, int *samplingrate) |
| { |
| struct SOFAlizerContext *s = ctx->priv; |
| /* variables associated with content of SOFA file: */ |
| int ncid, n_dims, n_vars, n_gatts, n_unlim_dim_id, status; |
| char data_delay_dim_name[NC_MAX_NAME]; |
| float *sp_a, *sp_e, *sp_r, *data_ir; |
| char *sofa_conventions; |
| char dim_name[NC_MAX_NAME]; /* names of netCDF dimensions */ |
| size_t *dim_length; /* lengths of netCDF dimensions */ |
| char *text; |
| unsigned int sample_rate; |
| int data_delay_dim_id[2]; |
| int samplingrate_id; |
| int data_delay_id; |
| int n_samples; |
| int m_dim_id = -1; |
| int n_dim_id = -1; |
| int data_ir_id; |
| size_t att_len; |
| int m_dim; |
| int *data_delay; |
| int sp_id; |
| int i, ret; |
| |
| s->sofa.ncid = 0; |
| status = nc_open(filename, NC_NOWRITE, &ncid); /* open SOFA file read-only */ |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename); |
| return AVERROR(EINVAL); |
| } |
| |
| /* get number of dimensions, vars, global attributes and Id of unlimited dimensions: */ |
| nc_inq(ncid, &n_dims, &n_vars, &n_gatts, &n_unlim_dim_id); |
| |
| /* -- get number of measurements ("M") and length of one IR ("N") -- */ |
| dim_length = av_malloc_array(n_dims, sizeof(*dim_length)); |
| if (!dim_length) { |
| nc_close(ncid); |
| return AVERROR(ENOMEM); |
| } |
| |
| for (i = 0; i < n_dims; i++) { /* go through all dimensions of file */ |
| nc_inq_dim(ncid, i, (char *)&dim_name, &dim_length[i]); /* get dimensions */ |
| if (!strncmp("M", (const char *)&dim_name, 1)) /* get ID of dimension "M" */ |
| m_dim_id = i; |
| if (!strncmp("N", (const char *)&dim_name, 1)) /* get ID of dimension "N" */ |
| n_dim_id = i; |
| } |
| |
| if ((m_dim_id == -1) || (n_dim_id == -1)) { /* dimension "M" or "N" couldn't be found */ |
| av_log(ctx, AV_LOG_ERROR, "Can't find required dimensions in SOFA file.\n"); |
| av_freep(&dim_length); |
| nc_close(ncid); |
| return AVERROR(EINVAL); |
| } |
| |
| n_samples = dim_length[n_dim_id]; /* get length of one IR */ |
| m_dim = dim_length[m_dim_id]; /* get number of measurements */ |
| |
| av_freep(&dim_length); |
| |
| /* -- check file type -- */ |
| /* get length of attritube "Conventions" */ |
| status = nc_inq_attlen(ncid, NC_GLOBAL, "Conventions", &att_len); |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"Conventions\".\n"); |
| nc_close(ncid); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| /* check whether file is SOFA file */ |
| text = av_malloc(att_len + 1); |
| if (!text) { |
| nc_close(ncid); |
| return AVERROR(ENOMEM); |
| } |
| |
| nc_get_att_text(ncid, NC_GLOBAL, "Conventions", text); |
| *(text + att_len) = 0; |
| if (strncmp("SOFA", text, 4)) { |
| av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n"); |
| av_freep(&text); |
| nc_close(ncid); |
| return AVERROR(EINVAL); |
| } |
| av_freep(&text); |
| |
| status = nc_inq_attlen(ncid, NC_GLOBAL, "License", &att_len); |
| if (status == NC_NOERR) { |
| text = av_malloc(att_len + 1); |
| if (text) { |
| nc_get_att_text(ncid, NC_GLOBAL, "License", text); |
| *(text + att_len) = 0; |
| av_log(ctx, AV_LOG_INFO, "SOFA file License: %s\n", text); |
| av_freep(&text); |
| } |
| } |
| |
| status = nc_inq_attlen(ncid, NC_GLOBAL, "SourceDescription", &att_len); |
| if (status == NC_NOERR) { |
| text = av_malloc(att_len + 1); |
| if (text) { |
| nc_get_att_text(ncid, NC_GLOBAL, "SourceDescription", text); |
| *(text + att_len) = 0; |
| av_log(ctx, AV_LOG_INFO, "SOFA file SourceDescription: %s\n", text); |
| av_freep(&text); |
| } |
| } |
| |
| status = nc_inq_attlen(ncid, NC_GLOBAL, "Comment", &att_len); |
| if (status == NC_NOERR) { |
| text = av_malloc(att_len + 1); |
| if (text) { |
| nc_get_att_text(ncid, NC_GLOBAL, "Comment", text); |
| *(text + att_len) = 0; |
| av_log(ctx, AV_LOG_INFO, "SOFA file Comment: %s\n", text); |
| av_freep(&text); |
| } |
| } |
| |
| status = nc_inq_attlen(ncid, NC_GLOBAL, "SOFAConventions", &att_len); |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"SOFAConventions\".\n"); |
| nc_close(ncid); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| sofa_conventions = av_malloc(att_len + 1); |
| if (!sofa_conventions) { |
| nc_close(ncid); |
| return AVERROR(ENOMEM); |
| } |
| |
| nc_get_att_text(ncid, NC_GLOBAL, "SOFAConventions", sofa_conventions); |
| *(sofa_conventions + att_len) = 0; |
| if (strncmp("SimpleFreeFieldHRIR", sofa_conventions, att_len)) { |
| av_log(ctx, AV_LOG_ERROR, "Not a SimpleFreeFieldHRIR file!\n"); |
| av_freep(&sofa_conventions); |
| nc_close(ncid); |
| return AVERROR(EINVAL); |
| } |
| av_freep(&sofa_conventions); |
| |
| /* -- get sampling rate of HRTFs -- */ |
| /* read ID, then value */ |
| status = nc_inq_varid(ncid, "Data.SamplingRate", &samplingrate_id); |
| status += nc_get_var_uint(ncid, samplingrate_id, &sample_rate); |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.SamplingRate.\n"); |
| nc_close(ncid); |
| return AVERROR(EINVAL); |
| } |
| *samplingrate = sample_rate; /* remember sampling rate */ |
| |
| /* -- allocate memory for one value for each measurement position: -- */ |
| sp_a = s->sofa.sp_a = av_malloc_array(m_dim, sizeof(float)); |
| sp_e = s->sofa.sp_e = av_malloc_array(m_dim, sizeof(float)); |
| sp_r = s->sofa.sp_r = av_malloc_array(m_dim, sizeof(float)); |
| /* delay and IR values required for each ear and measurement position: */ |
| data_delay = s->sofa.data_delay = av_calloc(m_dim, 2 * sizeof(int)); |
| data_ir = s->sofa.data_ir = av_malloc_array(m_dim * n_samples, sizeof(float) * 2); |
| |
| if (!data_delay || !sp_a || !sp_e || !sp_r || !data_ir) { |
| /* if memory could not be allocated */ |
| close_sofa(&s->sofa); |
| return AVERROR(ENOMEM); |
| } |
| |
| /* get impulse responses (HRTFs): */ |
| /* get corresponding ID */ |
| status = nc_inq_varid(ncid, "Data.IR", &data_ir_id); |
| status += nc_get_var_float(ncid, data_ir_id, data_ir); /* read and store IRs */ |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.IR!\n"); |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| |
| /* get source positions of the HRTFs in the SOFA file: */ |
| status = nc_inq_varid(ncid, "SourcePosition", &sp_id); /* get corresponding ID */ |
| status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 0 } , |
| (size_t[2]){ m_dim, 1}, sp_a); /* read & store azimuth angles */ |
| status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 1 } , |
| (size_t[2]){ m_dim, 1}, sp_e); /* read & store elevation angles */ |
| status += nc_get_vara_float(ncid, sp_id, (size_t[2]){ 0, 2 } , |
| (size_t[2]){ m_dim, 1}, sp_r); /* read & store radii */ |
| if (status != NC_NOERR) { /* if any source position variable coudn't be read */ |
| av_log(ctx, AV_LOG_ERROR, "Couldn't read SourcePosition.\n"); |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| |
| /* read Data.Delay, check for errors and fit it to data_delay */ |
| status = nc_inq_varid(ncid, "Data.Delay", &data_delay_id); |
| status += nc_inq_vardimid(ncid, data_delay_id, &data_delay_dim_id[0]); |
| status += nc_inq_dimname(ncid, data_delay_dim_id[0], data_delay_dim_name); |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay.\n"); |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| |
| /* Data.Delay dimension check */ |
| /* dimension of Data.Delay is [I R]: */ |
| if (!strncmp(data_delay_dim_name, "I", 2)) { |
| /* check 2 characters to assure string is 0-terminated after "I" */ |
| int delay[2]; /* delays get from SOFA file: */ |
| |
| av_log(ctx, AV_LOG_DEBUG, "Data.Delay has dimension [I R]\n"); |
| status = nc_get_var_int(ncid, data_delay_id, &delay[0]); |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n"); |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| int *data_delay_r = data_delay + m_dim; |
| for (i = 0; i < m_dim; i++) { /* extend given dimension [I R] to [M R] */ |
| /* assign constant delay value for all measurements to data_delay fields */ |
| data_delay[i] = delay[0]; |
| data_delay_r[i] = delay[1]; |
| } |
| /* dimension of Data.Delay is [M R] */ |
| } else if (!strncmp(data_delay_dim_name, "M", 2)) { |
| av_log(ctx, AV_LOG_ERROR, "Data.Delay in dimension [M R]\n"); |
| /* get delays from SOFA file: */ |
| status = nc_get_var_int(ncid, data_delay_id, data_delay); |
| if (status != NC_NOERR) { |
| av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n"); |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| } else { /* dimension of Data.Delay is neither [I R] nor [M R] */ |
| av_log(ctx, AV_LOG_ERROR, "Data.Delay does not have the required dimensions [I R] or [M R].\n"); |
| ret = AVERROR(EINVAL); |
| goto error; |
| } |
| |
| /* save information in SOFA struct: */ |
| s->sofa.m_dim = m_dim; /* no. measurement positions */ |
| s->sofa.n_samples = n_samples; /* length on one IR */ |
| s->sofa.ncid = ncid; /* netCDF ID of SOFA file */ |
| nc_close(ncid); /* close SOFA file */ |
| |
| return 0; |
| |
| error: |
| close_sofa(&s->sofa); |
| return ret; |
| } |
| |
| static int get_speaker_pos(AVFilterContext *ctx, |
| float *speaker_azim, float *speaker_elev) |
| { |
| struct SOFAlizerContext *s = ctx->priv; |
| uint64_t channels_layout = ctx->inputs[0]->channel_layout; |
| float azim[16] = { 0 }; |
| float elev[16] = { 0 }; |
| int m, ch, n_conv = ctx->inputs[0]->channels; /* get no. input channels */ |
| |
| if (n_conv > 16) |
| return AVERROR(EINVAL); |
| |
| s->lfe_channel = -1; |
| |
| /* set speaker positions according to input channel configuration: */ |
| for (m = 0, ch = 0; ch < n_conv && m < 64; m++) { |
| uint64_t mask = channels_layout & (1 << m); |
| |
| switch (mask) { |
| case AV_CH_FRONT_LEFT: azim[ch] = 30; break; |
| case AV_CH_FRONT_RIGHT: azim[ch] = 330; break; |
| case AV_CH_FRONT_CENTER: azim[ch] = 0; break; |
| case AV_CH_LOW_FREQUENCY: |
| case AV_CH_LOW_FREQUENCY_2: s->lfe_channel = ch; break; |
| case AV_CH_BACK_LEFT: azim[ch] = 150; break; |
| case AV_CH_BACK_RIGHT: azim[ch] = 210; break; |
| case AV_CH_BACK_CENTER: azim[ch] = 180; break; |
| case AV_CH_SIDE_LEFT: azim[ch] = 90; break; |
| case AV_CH_SIDE_RIGHT: azim[ch] = 270; break; |
| case AV_CH_FRONT_LEFT_OF_CENTER: azim[ch] = 15; break; |
| case AV_CH_FRONT_RIGHT_OF_CENTER: azim[ch] = 345; break; |
| case AV_CH_TOP_CENTER: azim[ch] = 0; |
| elev[ch] = 90; break; |
| case AV_CH_TOP_FRONT_LEFT: azim[ch] = 30; |
| elev[ch] = 45; break; |
| case AV_CH_TOP_FRONT_CENTER: azim[ch] = 0; |
| elev[ch] = 45; break; |
| case AV_CH_TOP_FRONT_RIGHT: azim[ch] = 330; |
| elev[ch] = 45; break; |
| case AV_CH_TOP_BACK_LEFT: azim[ch] = 150; |
| elev[ch] = 45; break; |
| case AV_CH_TOP_BACK_RIGHT: azim[ch] = 210; |
| elev[ch] = 45; break; |
| case AV_CH_TOP_BACK_CENTER: azim[ch] = 180; |
| elev[ch] = 45; break; |
| case AV_CH_WIDE_LEFT: azim[ch] = 90; break; |
| case AV_CH_WIDE_RIGHT: azim[ch] = 270; break; |
| case AV_CH_SURROUND_DIRECT_LEFT: azim[ch] = 90; break; |
| case AV_CH_SURROUND_DIRECT_RIGHT: azim[ch] = 270; break; |
| case AV_CH_STEREO_LEFT: azim[ch] = 90; break; |
| case AV_CH_STEREO_RIGHT: azim[ch] = 270; break; |
| case 0: break; |
| default: |
| return AVERROR(EINVAL); |
| } |
| if (mask) |
| ch++; |
| } |
| |
| memcpy(speaker_azim, azim, n_conv * sizeof(float)); |
| memcpy(speaker_elev, elev, n_conv * sizeof(float)); |
| |
| return 0; |
| |
| } |
| |
| static int max_delay(struct NCSofa *sofa) |
| { |
| int i, max = 0; |
| |
| for (i = 0; i < sofa->m_dim * 2; i++) { |
| /* search maximum delay in given SOFA file */ |
| max = FFMAX(max, sofa->data_delay[i]); |
| } |
| |
| return max; |
| } |
| |
| static int find_m(SOFAlizerContext *s, int azim, int elev, float radius) |
| { |
| /* get source positions and M of currently selected SOFA file */ |
| float *sp_a = s->sofa.sp_a; /* azimuth angle */ |
| float *sp_e = s->sofa.sp_e; /* elevation angle */ |
| float *sp_r = s->sofa.sp_r; /* radius */ |
| int m_dim = s->sofa.m_dim; /* no. measurements */ |
| int best_id = 0; /* index m currently closest to desired source pos. */ |
| float delta = 1000; /* offset between desired and currently best pos. */ |
| float current; |
| int i; |
| |
| for (i = 0; i < m_dim; i++) { |
| /* search through all measurements in currently selected SOFA file */ |
| /* distance of current to desired source position: */ |
| current = fabs(sp_a[i] - azim) + |
| fabs(sp_e[i] - elev) + |
| fabs(sp_r[i] - radius); |
| if (current <= delta) { |
| /* if current distance is smaller than smallest distance so far */ |
| delta = current; |
| best_id = i; /* remember index */ |
| } |
| } |
| |
| return best_id; |
| } |
| |
| static int compensate_volume(AVFilterContext *ctx) |
| { |
| struct SOFAlizerContext *s = ctx->priv; |
| float compensate; |
| float energy = 0; |
| float *ir; |
| int m; |
| |
| if (s->sofa.ncid) { |
| /* find IR at front center position in the SOFA file (IR closest to 0°,0°,1m) */ |
| struct NCSofa *sofa = &s->sofa; |
| m = find_m(s, 0, 0, 1); |
| /* get energy of that IR and compensate volume */ |
| ir = sofa->data_ir + 2 * m * sofa->n_samples; |
| if (sofa->n_samples & 31) { |
| energy = avpriv_scalarproduct_float_c(ir, ir, sofa->n_samples); |
| } else { |
| energy = s->fdsp->scalarproduct_float(ir, ir, sofa->n_samples); |
| } |
| compensate = 256 / (sofa->n_samples * sqrt(energy)); |
| av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", compensate); |
| ir = sofa->data_ir; |
| /* apply volume compensation to IRs */ |
| s->fdsp->vector_fmul_scalar(ir, ir, compensate, sofa->n_samples * sofa->m_dim * 2); |
| emms_c(); |
| } |
| |
| return 0; |
| } |
| |
| typedef struct ThreadData { |
| AVFrame *in, *out; |
| int *write; |
| int **delay; |
| float **ir; |
| int *n_clippings; |
| float **ringbuffer; |
| float **temp_src; |
| FFTComplex **temp_fft; |
| } ThreadData; |
| |
| static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
| { |
| SOFAlizerContext *s = ctx->priv; |
| ThreadData *td = arg; |
| AVFrame *in = td->in, *out = td->out; |
| int offset = jobnr; |
| int *write = &td->write[jobnr]; |
| const int *const delay = td->delay[jobnr]; |
| const float *const ir = td->ir[jobnr]; |
| int *n_clippings = &td->n_clippings[jobnr]; |
| float *ringbuffer = td->ringbuffer[jobnr]; |
| float *temp_src = td->temp_src[jobnr]; |
| const int n_samples = s->sofa.n_samples; /* length of one IR */ |
| const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */ |
| float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */ |
| const int in_channels = s->n_conv; /* number of input channels */ |
| /* ring buffer length is: longest IR plus max. delay -> next power of 2 */ |
| const int buffer_length = s->buffer_length; |
| /* -1 for AND instead of MODULO (applied to powers of 2): */ |
| const uint32_t modulo = (uint32_t)buffer_length - 1; |
| float *buffer[16]; /* holds ringbuffer for each input channel */ |
| int wr = *write; |
| int read; |
| int i, l; |
| |
| dst += offset; |
| for (l = 0; l < in_channels; l++) { |
| /* get starting address of ringbuffer for each input channel */ |
| buffer[l] = ringbuffer + l * buffer_length; |
| } |
| |
| for (i = 0; i < in->nb_samples; i++) { |
| const float *temp_ir = ir; /* using same set of IRs for each sample */ |
| |
| *dst = 0; |
| for (l = 0; l < in_channels; l++) { |
| /* write current input sample to ringbuffer (for each channel) */ |
| *(buffer[l] + wr) = src[l]; |
| } |
| |
| /* loop goes through all channels to be convolved */ |
| for (l = 0; l < in_channels; l++) { |
| const float *const bptr = buffer[l]; |
| |
| if (l == s->lfe_channel) { |
| /* LFE is an input channel but requires no convolution */ |
| /* apply gain to LFE signal and add to output buffer */ |
| *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe; |
| temp_ir += n_samples; |
| continue; |
| } |
| |
| /* current read position in ringbuffer: input sample write position |
| * - delay for l-th ch. + diff. betw. IR length and buffer length |
| * (mod buffer length) */ |
| read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo; |
| |
| if (read + n_samples < buffer_length) { |
| memcpy(temp_src, bptr + read, n_samples * sizeof(*temp_src)); |
| } else { |
| int len = FFMIN(n_samples - (read % n_samples), buffer_length - read); |
| |
| memcpy(temp_src, bptr + read, len * sizeof(*temp_src)); |
| memcpy(temp_src + len, bptr, (n_samples - len) * sizeof(*temp_src)); |
| } |
| |
| /* multiply signal and IR, and add up the results */ |
| dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples); |
| temp_ir += n_samples; |
| } |
| |
| /* clippings counter */ |
| if (fabs(*dst) > 1) |
| *n_clippings += 1; |
| |
| /* move output buffer pointer by +2 to get to next sample of processed channel: */ |
| dst += 2; |
| src += in_channels; |
| wr = (wr + 1) & modulo; /* update ringbuffer write position */ |
| } |
| |
| *write = wr; /* remember write position in ringbuffer for next call */ |
| |
| return 0; |
| } |
| |
| static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) |
| { |
| SOFAlizerContext *s = ctx->priv; |
| ThreadData *td = arg; |
| AVFrame *in = td->in, *out = td->out; |
| int offset = jobnr; |
| int *write = &td->write[jobnr]; |
| FFTComplex *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */ |
| int *n_clippings = &td->n_clippings[jobnr]; |
| float *ringbuffer = td->ringbuffer[jobnr]; |
| const int n_samples = s->sofa.n_samples; /* length of one IR */ |
| const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */ |
| float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */ |
| const int in_channels = s->n_conv; /* number of input channels */ |
| /* ring buffer length is: longest IR plus max. delay -> next power of 2 */ |
| const int buffer_length = s->buffer_length; |
| /* -1 for AND instead of MODULO (applied to powers of 2): */ |
| const uint32_t modulo = (uint32_t)buffer_length - 1; |
| FFTComplex *fft_in = s->temp_fft[jobnr]; /* temporary array for FFT input/output data */ |
| FFTContext *ifft = s->ifft[jobnr]; |
| FFTContext *fft = s->fft[jobnr]; |
| const int n_conv = s->n_conv; |
| const int n_fft = s->n_fft; |
| int wr = *write; |
| int n_read; |
| int i, j; |
| |
| dst += offset; |
| |
| /* find minimum between number of samples and output buffer length: |
| * (important, if one IR is longer than the output buffer) */ |
| n_read = FFMIN(s->sofa.n_samples, in->nb_samples); |
| for (j = 0; j < n_read; j++) { |
| /* initialize output buf with saved signal from overflow buf */ |
| dst[2 * j] = ringbuffer[wr]; |
| ringbuffer[wr] = 0.0; /* re-set read samples to zero */ |
| /* update ringbuffer read/write position */ |
| wr = (wr + 1) & modulo; |
| } |
| |
| /* initialize rest of output buffer with 0 */ |
| for (j = n_read; j < in->nb_samples; j++) { |
| dst[2 * j] = 0; |
| } |
| |
| for (i = 0; i < n_conv; i++) { |
| if (i == s->lfe_channel) { /* LFE */ |
| for (j = 0; j < in->nb_samples; j++) { |
| /* apply gain to LFE signal and add to output buffer */ |
| dst[2 * j] += src[i + j * in_channels] * s->gain_lfe; |
| } |
| continue; |
| } |
| |
| /* outer loop: go through all input channels to be convolved */ |
| offset = i * n_fft; /* no. samples already processed */ |
| |
| /* fill FFT input with 0 (we want to zero-pad) */ |
| memset(fft_in, 0, sizeof(FFTComplex) * n_fft); |
| |
| for (j = 0; j < in->nb_samples; j++) { |
| /* prepare input for FFT */ |
| /* write all samples of current input channel to FFT input array */ |
| fft_in[j].re = src[j * in_channels + i]; |
| } |
| |
| /* transform input signal of current channel to frequency domain */ |
| av_fft_permute(fft, fft_in); |
| av_fft_calc(fft, fft_in); |
| for (j = 0; j < n_fft; j++) { |
| const float re = fft_in[j].re; |
| const float im = fft_in[j].im; |
| |
| /* complex multiplication of input signal and HRTFs */ |
| /* output channel (real): */ |
| fft_in[j].re = re * (hrtf + offset + j)->re - im * (hrtf + offset + j)->im; |
| /* output channel (imag): */ |
| fft_in[j].im = re * (hrtf + offset + j)->im + im * (hrtf + offset + j)->re; |
| } |
| |
| /* transform output signal of current channel back to time domain */ |
| av_fft_permute(ifft, fft_in); |
| av_fft_calc(ifft, fft_in); |
| |
| for (j = 0; j < in->nb_samples; j++) { |
| /* write output signal of current channel to output buffer */ |
| dst[2 * j] += fft_in[j].re / (float)n_fft; |
| } |
| |
| for (j = 0; j < n_samples - 1; j++) { /* overflow length is IR length - 1 */ |
| /* write the rest of output signal to overflow buffer */ |
| int write_pos = (wr + j) & modulo; |
| |
| *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re / (float)n_fft; |
| } |
| } |
| |
| /* go through all samples of current output buffer: count clippings */ |
| for (i = 0; i < out->nb_samples; i++) { |
| /* clippings counter */ |
| if (fabs(*dst) > 1) { /* if current output sample > 1 */ |
| *n_clippings = *n_clippings + 1; |
| } |
| |
| /* move output buffer pointer by +2 to get to next sample of processed channel: */ |
| dst += 2; |
| } |
| |
| /* remember read/write position in ringbuffer for next call */ |
| *write = wr; |
| |
| return 0; |
| } |
| |
| static int filter_frame(AVFilterLink *inlink, AVFrame *in) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| SOFAlizerContext *s = ctx->priv; |
| AVFilterLink *outlink = ctx->outputs[0]; |
| int n_clippings[2] = { 0 }; |
| ThreadData td; |
| AVFrame *out; |
| |
| out = ff_get_audio_buffer(outlink, in->nb_samples); |
| if (!out) { |
| av_frame_free(&in); |
| return AVERROR(ENOMEM); |
| } |
| av_frame_copy_props(out, in); |
| |
| td.in = in; td.out = out; td.write = s->write; |
| td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings; |
| td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src; |
| td.temp_fft = s->temp_fft; |
| |
| if (s->type == TIME_DOMAIN) { |
| ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2); |
| } else { |
| ctx->internal->execute(ctx, sofalizer_fast_convolute, &td, NULL, 2); |
| } |
| emms_c(); |
| |
| /* display error message if clipping occurred */ |
| if (n_clippings[0] + n_clippings[1] > 0) { |
| av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n", |
| n_clippings[0] + n_clippings[1], out->nb_samples * 2); |
| } |
| |
| av_frame_free(&in); |
| return ff_filter_frame(outlink, out); |
| } |
| |
| static int query_formats(AVFilterContext *ctx) |
| { |
| struct SOFAlizerContext *s = ctx->priv; |
| AVFilterFormats *formats = NULL; |
| AVFilterChannelLayouts *layouts = NULL; |
| int ret, sample_rates[] = { 48000, -1 }; |
| |
| ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT); |
| if (ret) |
| return ret; |
| ret = ff_set_common_formats(ctx, formats); |
| if (ret) |
| return ret; |
| |
| layouts = ff_all_channel_layouts(); |
| if (!layouts) |
| return AVERROR(ENOMEM); |
| |
| ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts); |
| if (ret) |
| return ret; |
| |
| layouts = NULL; |
| ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO); |
| if (ret) |
| return ret; |
| |
| ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts); |
| if (ret) |
| return ret; |
| |
| sample_rates[0] = s->sample_rate; |
| formats = ff_make_format_list(sample_rates); |
| if (!formats) |
| return AVERROR(ENOMEM); |
| return ff_set_common_samplerates(ctx, formats); |
| } |
| |
| static int load_data(AVFilterContext *ctx, int azim, int elev, float radius) |
| { |
| struct SOFAlizerContext *s = ctx->priv; |
| const int n_samples = s->sofa.n_samples; |
| int n_conv = s->n_conv; /* no. channels to convolve */ |
| int n_fft = s->n_fft; |
| int delay_l[16]; /* broadband delay for each IR */ |
| int delay_r[16]; |
| int nb_input_channels = ctx->inputs[0]->channels; /* no. input channels */ |
| float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */ |
| FFTComplex *data_hrtf_l = NULL; |
| FFTComplex *data_hrtf_r = NULL; |
| FFTComplex *fft_in_l = NULL; |
| FFTComplex *fft_in_r = NULL; |
| float *data_ir_l = NULL; |
| float *data_ir_r = NULL; |
| int offset = 0; /* used for faster pointer arithmetics in for-loop */ |
| int m[16]; /* measurement index m of IR closest to required source positions */ |
| int i, j, azim_orig = azim, elev_orig = elev; |
| |
| if (!s->sofa.ncid) { /* if an invalid SOFA file has been selected */ |
| av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n"); |
| return AVERROR_INVALIDDATA; |
| } |
| |
| if (s->type == TIME_DOMAIN) { |
| s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float)); |
| s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float)); |
| |
| /* get temporary IR for L and R channel */ |
| data_ir_l = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_l)); |
| data_ir_r = av_malloc_array(n_conv * n_samples, sizeof(*data_ir_r)); |
| if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) { |
| av_free(data_ir_l); |
| av_free(data_ir_r); |
| return AVERROR(ENOMEM); |
| } |
| } else { |
| /* get temporary HRTF memory for L and R channel */ |
| data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv); |
| data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv); |
| if (!data_hrtf_r || !data_hrtf_l) { |
| av_free(data_hrtf_l); |
| av_free(data_hrtf_r); |
| return AVERROR(ENOMEM); |
| } |
| } |
| |
| for (i = 0; i < s->n_conv; i++) { |
| /* load and store IRs and corresponding delays */ |
| azim = (int)(s->speaker_azim[i] + azim_orig) % 360; |
| elev = (int)(s->speaker_elev[i] + elev_orig) % 90; |
| /* get id of IR closest to desired position */ |
| m[i] = find_m(s, azim, elev, radius); |
| |
| /* load the delays associated with the current IRs */ |
| delay_l[i] = *(s->sofa.data_delay + 2 * m[i]); |
| delay_r[i] = *(s->sofa.data_delay + 2 * m[i] + 1); |
| |
| if (s->type == TIME_DOMAIN) { |
| offset = i * n_samples; /* no. samples already written */ |
| for (j = 0; j < n_samples; j++) { |
| /* load reversed IRs of the specified source position |
| * sample-by-sample for left and right ear; and apply gain */ |
| *(data_ir_l + offset + j) = /* left channel */ |
| *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j) * gain_lin; |
| *(data_ir_r + offset + j) = /* right channel */ |
| *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j + n_samples) * gain_lin; |
| } |
| } else { |
| fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l)); |
| fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r)); |
| if (!fft_in_l || !fft_in_r) { |
| av_free(data_hrtf_l); |
| av_free(data_hrtf_r); |
| av_free(fft_in_l); |
| av_free(fft_in_r); |
| return AVERROR(ENOMEM); |
| } |
| |
| offset = i * n_fft; /* no. samples already written */ |
| for (j = 0; j < n_samples; j++) { |
| /* load non-reversed IRs of the specified source position |
| * sample-by-sample and apply gain, |
| * L channel is loaded to real part, R channel to imag part, |
| * IRs ared shifted by L and R delay */ |
| fft_in_l[delay_l[i] + j].re = /* left channel */ |
| *(s->sofa.data_ir + 2 * m[i] * n_samples + j) * gain_lin; |
| fft_in_r[delay_r[i] + j].re = /* right channel */ |
| *(s->sofa.data_ir + (2 * m[i] + 1) * n_samples + j) * gain_lin; |
| } |
| |
| /* actually transform to frequency domain (IRs -> HRTFs) */ |
| av_fft_permute(s->fft[0], fft_in_l); |
| av_fft_calc(s->fft[0], fft_in_l); |
| memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l)); |
| av_fft_permute(s->fft[0], fft_in_r); |
| av_fft_calc(s->fft[0], fft_in_r); |
| memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r)); |
| } |
| |
| av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n", |
| m[i], *(s->sofa.sp_a + m[i]), *(s->sofa.sp_e + m[i]), *(s->sofa.sp_r + m[i])); |
| } |
| |
| if (s->type == TIME_DOMAIN) { |
| /* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */ |
| memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * n_samples); |
| memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * n_samples); |
| |
| av_freep(&data_ir_l); /* free temporary IR memory */ |
| av_freep(&data_ir_r); |
| } else { |
| s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex)); |
| s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex)); |
| if (!s->data_hrtf[0] || !s->data_hrtf[1]) { |
| av_freep(&data_hrtf_l); |
| av_freep(&data_hrtf_r); |
| av_freep(&fft_in_l); |
| av_freep(&fft_in_r); |
| return AVERROR(ENOMEM); /* memory allocation failed */ |
| } |
| |
| memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */ |
| sizeof(FFTComplex) * n_conv * n_fft); /* filter struct */ |
| memcpy(s->data_hrtf[1], data_hrtf_r, |
| sizeof(FFTComplex) * n_conv * n_fft); |
| |
| av_freep(&data_hrtf_l); /* free temporary HRTF memory */ |
| av_freep(&data_hrtf_r); |
| |
| av_freep(&fft_in_l); /* free temporary FFT memory */ |
| av_freep(&fft_in_r); |
| } |
| |
| memcpy(s->delay[0], &delay_l[0], sizeof(int) * s->n_conv); |
| memcpy(s->delay[1], &delay_r[0], sizeof(int) * s->n_conv); |
| |
| return 0; |
| } |
| |
| static av_cold int init(AVFilterContext *ctx) |
| { |
| SOFAlizerContext *s = ctx->priv; |
| int ret; |
| |
| /* load SOFA file, */ |
| /* initialize file IDs to 0 before attempting to load SOFA files, |
| * this assures that in case of error, only the memory of already |
| * loaded files is free'd */ |
| s->sofa.ncid = 0; |
| ret = load_sofa(ctx, s->filename, &s->sample_rate); |
| if (ret) { |
| /* file loading error */ |
| av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename); |
| } else { /* no file loading error, resampling not required */ |
| av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename); |
| } |
| |
| if (ret) { |
| av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n"); |
| return ret; |
| } |
| |
| s->fdsp = avpriv_float_dsp_alloc(0); |
| if (!s->fdsp) |
| return AVERROR(ENOMEM); |
| |
| return 0; |
| } |
| |
| static int config_input(AVFilterLink *inlink) |
| { |
| AVFilterContext *ctx = inlink->dst; |
| SOFAlizerContext *s = ctx->priv; |
| int nb_input_channels = inlink->channels; /* no. input channels */ |
| int n_max_ir = 0; |
| int n_current; |
| int n_max = 0; |
| int ret; |
| |
| if (s->type == FREQUENCY_DOMAIN) { |
| inlink->partial_buf_size = |
| inlink->min_samples = |
| inlink->max_samples = inlink->sample_rate; |
| } |
| |
| /* gain -3 dB per channel, -6 dB to get LFE on a similar level */ |
| s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6) / 20 * M_LN10); |
| |
| s->n_conv = nb_input_channels; |
| |
| /* get size of ringbuffer (longest IR plus max. delay) */ |
| /* then choose next power of 2 for performance optimization */ |
| n_current = s->sofa.n_samples + max_delay(&s->sofa); |
| if (n_current > n_max) { |
| /* length of longest IR plus max. delay (in all SOFA files) */ |
| n_max = n_current; |
| /* length of longest IR (without delay, in all SOFA files) */ |
| n_max_ir = s->sofa.n_samples; |
| } |
| /* buffer length is longest IR plus max. delay -> next power of 2 |
| (32 - count leading zeros gives required exponent) */ |
| s->buffer_length = 1 << (32 - ff_clz(n_max)); |
| s->n_fft = 1 << (32 - ff_clz(n_max + inlink->sample_rate)); |
| |
| if (s->type == FREQUENCY_DOMAIN) { |
| av_fft_end(s->fft[0]); |
| av_fft_end(s->fft[1]); |
| s->fft[0] = av_fft_init(log2(s->n_fft), 0); |
| s->fft[1] = av_fft_init(log2(s->n_fft), 0); |
| av_fft_end(s->ifft[0]); |
| av_fft_end(s->ifft[1]); |
| s->ifft[0] = av_fft_init(log2(s->n_fft), 1); |
| s->ifft[1] = av_fft_init(log2(s->n_fft), 1); |
| |
| if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) { |
| av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts.\n"); |
| return AVERROR(ENOMEM); |
| } |
| } |
| |
| /* Allocate memory for the impulse responses, delays and the ringbuffers */ |
| /* size: (longest IR) * (number of channels to convolute) */ |
| s->data_ir[0] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv); |
| s->data_ir[1] = av_malloc_array(n_max_ir, sizeof(float) * s->n_conv); |
| /* length: number of channels to convolute */ |
| s->delay[0] = av_malloc_array(s->n_conv, sizeof(float)); |
| s->delay[1] = av_malloc_array(s->n_conv, sizeof(float)); |
| /* length: (buffer length) * (number of input channels), |
| * OR: buffer length (if frequency domain processing) |
| * calloc zero-initializes the buffer */ |
| |
| if (s->type == TIME_DOMAIN) { |
| s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
| s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); |
| } else { |
| s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float)); |
| s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float)); |
| s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); |
| s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); |
| if (!s->temp_fft[0] || !s->temp_fft[1]) |
| return AVERROR(ENOMEM); |
| } |
| |
| /* length: number of channels to convolute */ |
| s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim)); |
| s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev)); |
| |
| /* memory allocation failed: */ |
| if (!s->data_ir[0] || !s->data_ir[1] || !s->delay[1] || |
| !s->delay[0] || !s->ringbuffer[0] || !s->ringbuffer[1] || |
| !s->speaker_azim || !s->speaker_elev) |
| return AVERROR(ENOMEM); |
| |
| compensate_volume(ctx); |
| |
| /* get speaker positions */ |
| if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 0) { |
| av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n"); |
| return ret; |
| } |
| |
| /* load IRs to data_ir[0] and data_ir[1] for required directions */ |
| if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius)) < 0) |
| return ret; |
| |
| av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n", |
| inlink->sample_rate, s->n_conv, nb_input_channels, s->buffer_length); |
| |
| return 0; |
| } |
| |
| static av_cold void uninit(AVFilterContext *ctx) |
| { |
| SOFAlizerContext *s = ctx->priv; |
| |
| if (s->sofa.ncid) { |
| av_freep(&s->sofa.sp_a); |
| av_freep(&s->sofa.sp_e); |
| av_freep(&s->sofa.sp_r); |
| av_freep(&s->sofa.data_delay); |
| av_freep(&s->sofa.data_ir); |
| } |
| av_fft_end(s->ifft[0]); |
| av_fft_end(s->ifft[1]); |
| av_fft_end(s->fft[0]); |
| av_fft_end(s->fft[1]); |
| av_freep(&s->delay[0]); |
| av_freep(&s->delay[1]); |
| av_freep(&s->data_ir[0]); |
| av_freep(&s->data_ir[1]); |
| av_freep(&s->ringbuffer[0]); |
| av_freep(&s->ringbuffer[1]); |
| av_freep(&s->speaker_azim); |
| av_freep(&s->speaker_elev); |
| av_freep(&s->temp_src[0]); |
| av_freep(&s->temp_src[1]); |
| av_freep(&s->temp_fft[0]); |
| av_freep(&s->temp_fft[1]); |
| av_freep(&s->data_hrtf[0]); |
| av_freep(&s->data_hrtf[1]); |
| av_freep(&s->fdsp); |
| } |
| |
| #define OFFSET(x) offsetof(SOFAlizerContext, x) |
| #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM |
| |
| static const AVOption sofalizer_options[] = { |
| { "sofa", "sofa filename", OFFSET(filename), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS }, |
| { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, |
| { "rotation", "set rotation" , OFFSET(rotation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -360, 360, .flags = FLAGS }, |
| { "elevation", "set elevation", OFFSET(elevation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -90, 90, .flags = FLAGS }, |
| { "radius", "set radius", OFFSET(radius), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 3, .flags = FLAGS }, |
| { "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" }, |
| { "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" }, |
| { "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" }, |
| { NULL } |
| }; |
| |
| AVFILTER_DEFINE_CLASS(sofalizer); |
| |
| static const AVFilterPad inputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| .config_props = config_input, |
| .filter_frame = filter_frame, |
| }, |
| { NULL } |
| }; |
| |
| static const AVFilterPad outputs[] = { |
| { |
| .name = "default", |
| .type = AVMEDIA_TYPE_AUDIO, |
| }, |
| { NULL } |
| }; |
| |
| AVFilter ff_af_sofalizer = { |
| .name = "sofalizer", |
| .description = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."), |
| .priv_size = sizeof(SOFAlizerContext), |
| .priv_class = &sofalizer_class, |
| .init = init, |
| .uninit = uninit, |
| .query_formats = query_formats, |
| .inputs = inputs, |
| .outputs = outputs, |
| .flags = AVFILTER_FLAG_SLICE_THREADS, |
| }; |
