/***************************************************************************** * 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/avstring.h" #include "libavutil/channel_layout.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 VirtualSpeaker { uint8_t set; float azim; float elev; } VirtualSpeaker; 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 */ char *speakers_pos; /* custom positions 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 */ VirtualSpeaker vspkrpos[64]; 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_calloc(m_dim * FFALIGN(n_samples, 16), 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: */ int *data_delay_r; 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; } 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 */ av_log(ctx, AV_LOG_DEBUG, "m_dim: %d n_samples %d\n", m_dim, n_samples); return 0; error: close_sofa(&s->sofa); return ret; } static int parse_channel_name(char **arg, int *rchannel) { char buf[8]; int len, i, channel_id = 0; int64_t layout, layout0; /* try to parse a channel name, e.g. "FL" */ if (sscanf(*arg, "%7[A-Z]%n", buf, &len)) { layout0 = layout = av_get_channel_layout(buf); /* channel_id <- first set bit in layout */ for (i = 32; i > 0; i >>= 1) { if (layout >= (int64_t)1 << i) { channel_id += i; layout >>= i; } } /* reject layouts that are not a single channel */ if (channel_id >= 64 || layout0 != (int64_t)1 << channel_id) return AVERROR(EINVAL); *rchannel = channel_id; *arg += len; return 0; } return AVERROR(EINVAL); } static void parse_speaker_pos(AVFilterContext *ctx, int64_t in_channel_layout) { SOFAlizerContext *s = ctx->priv; char *arg, *tokenizer, *p, *args = av_strdup(s->speakers_pos); if (!args) return; p = args; while ((arg = av_strtok(p, "|", &tokenizer))) { float azim, elev; int out_ch_id; p = NULL; if (parse_channel_name(&arg, &out_ch_id)) continue; if (sscanf(arg, "%f %f", &azim, &elev) == 2) { s->vspkrpos[out_ch_id].set = 1; s->vspkrpos[out_ch_id].azim = azim; s->vspkrpos[out_ch_id].elev = elev; } else if (sscanf(arg, "%f", &azim) == 1) { s->vspkrpos[out_ch_id].set = 1; s->vspkrpos[out_ch_id].azim = azim; s->vspkrpos[out_ch_id].elev = 0; } } av_free(args); } 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; if (s->speakers_pos) parse_speaker_pos(ctx, channels_layout); /* set speaker positions according to input channel configuration: */ for (m = 0, ch = 0; ch < n_conv && m < 64; m++) { uint64_t mask = channels_layout & (1ULL << m); switch (mask) { case AV_CH_FRONT_LEFT: azim[ch] = 30; break; case AV_CH_FRONT_RIGHT: azim[ch] = 330; break; case AV_CH_FRONT_CENTER: azim[ch] = 0; break; case AV_CH_LOW_FREQUENCY: case AV_CH_LOW_FREQUENCY_2: s->lfe_channel = ch; break; case AV_CH_BACK_LEFT: azim[ch] = 150; break; case AV_CH_BACK_RIGHT: azim[ch] = 210; break; case AV_CH_BACK_CENTER: azim[ch] = 180; break; case AV_CH_SIDE_LEFT: azim[ch] = 90; break; case AV_CH_SIDE_RIGHT: azim[ch] = 270; break; case AV_CH_FRONT_LEFT_OF_CENTER: azim[ch] = 15; break; case AV_CH_FRONT_RIGHT_OF_CENTER: azim[ch] = 345; break; case AV_CH_TOP_CENTER: azim[ch] = 0; elev[ch] = 90; break; case AV_CH_TOP_FRONT_LEFT: azim[ch] = 30; elev[ch] = 45; break; case AV_CH_TOP_FRONT_CENTER: azim[ch] = 0; elev[ch] = 45; break; case AV_CH_TOP_FRONT_RIGHT: azim[ch] = 330; elev[ch] = 45; break; case AV_CH_TOP_BACK_LEFT: azim[ch] = 150; elev[ch] = 45; break; case AV_CH_TOP_BACK_RIGHT: azim[ch] = 210; elev[ch] = 45; break; case AV_CH_TOP_BACK_CENTER: azim[ch] = 180; elev[ch] = 45; break; case AV_CH_WIDE_LEFT: azim[ch] = 90; break; case AV_CH_WIDE_RIGHT: azim[ch] = 270; break; case AV_CH_SURROUND_DIRECT_LEFT: azim[ch] = 90; break; case AV_CH_SURROUND_DIRECT_RIGHT: azim[ch] = 270; break; case AV_CH_STEREO_LEFT: azim[ch] = 90; break; case AV_CH_STEREO_RIGHT: azim[ch] = 270; break; case 0: break; default: return AVERROR(EINVAL); } if (s->vspkrpos[m].set) { azim[ch] = s->vspkrpos[m].azim; elev[ch] = s->vspkrpos[m].elev; } if (mask) ch++; } memcpy(speaker_azim, azim, n_conv * sizeof(float)); memcpy(speaker_elev, elev, n_conv * sizeof(float)); return 0; } static int max_delay(struct NCSofa *sofa) { int i, max = 0; for (i = 0; i < sofa->m_dim * 2; i++) { /* search maximum delay in given SOFA file */ max = FFMAX(max, sofa->data_delay[i]); } return max; } static int find_m(SOFAlizerContext *s, int azim, int elev, float radius) { /* get source positions and M of currently selected SOFA file */ float *sp_a = s->sofa.sp_a; /* azimuth angle */ float *sp_e = s->sofa.sp_e; /* elevation angle */ float *sp_r = s->sofa.sp_r; /* radius */ int m_dim = s->sofa.m_dim; /* no. measurements */ int best_id = 0; /* index m currently closest to desired source pos. */ float delta = 1000; /* offset between desired and currently best pos. */ float current; int i; for (i = 0; i < m_dim; i++) { /* search through all measurements in currently selected SOFA file */ /* distance of current to desired source position: */ current = fabs(sp_a[i] - azim) + fabs(sp_e[i] - elev) + fabs(sp_r[i] - radius); if (current <= delta) { /* if current distance is smaller than smallest distance so far */ delta = current; best_id = i; /* remember index */ } } return best_id; } static int compensate_volume(AVFilterContext *ctx) { struct SOFAlizerContext *s = ctx->priv; float compensate; float energy = 0; float *ir; int m; if (s->sofa.ncid) { /* find IR at front center position in the SOFA file (IR closest to 0°,0°,1m) */ struct NCSofa *sofa = &s->sofa; m = find_m(s, 0, 0, 1); /* get energy of that IR and compensate volume */ ir = sofa->data_ir + 2 * m * sofa->n_samples; if (sofa->n_samples & 31) { energy = avpriv_scalarproduct_float_c(ir, ir, sofa->n_samples); } else { energy = s->fdsp->scalarproduct_float(ir, ir, sofa->n_samples); } compensate = 256 / (sofa->n_samples * sqrt(energy)); av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", compensate); ir = sofa->data_ir; /* apply volume compensation to IRs */ if (sofa->n_samples & 31) { int i; for (i = 0; i < sofa->n_samples * sofa->m_dim * 2; i++) { ir[i] = ir[i] * compensate; } } else { s->fdsp->vector_fmul_scalar(ir, ir, compensate, sofa->n_samples * sofa->m_dim * 2); emms_c(); } } return 0; } typedef struct ThreadData { AVFrame *in, *out; int *write; int **delay; float **ir; int *n_clippings; float **ringbuffer; float **temp_src; FFTComplex **temp_fft; } ThreadData; static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { SOFAlizerContext *s = ctx->priv; ThreadData *td = arg; AVFrame *in = td->in, *out = td->out; int offset = jobnr; int *write = &td->write[jobnr]; const int *const delay = td->delay[jobnr]; const float *const ir = td->ir[jobnr]; int *n_clippings = &td->n_clippings[jobnr]; float *ringbuffer = td->ringbuffer[jobnr]; float *temp_src = td->temp_src[jobnr]; const int n_samples = s->sofa.n_samples; /* length of one IR */ const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */ float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */ const int in_channels = s->n_conv; /* number of input channels */ /* ring buffer length is: longest IR plus max. delay -> next power of 2 */ const int buffer_length = s->buffer_length; /* -1 for AND instead of MODULO (applied to powers of 2): */ const uint32_t modulo = (uint32_t)buffer_length - 1; float *buffer[16]; /* holds ringbuffer for each input channel */ int wr = *write; int read; int i, l; dst += offset; for (l = 0; l < in_channels; l++) { /* get starting address of ringbuffer for each input channel */ buffer[l] = ringbuffer + l * buffer_length; } for (i = 0; i < in->nb_samples; i++) { const float *temp_ir = ir; /* using same set of IRs for each sample */ *dst = 0; for (l = 0; l < in_channels; l++) { /* write current input sample to ringbuffer (for each channel) */ *(buffer[l] + wr) = src[l]; } /* loop goes through all channels to be convolved */ for (l = 0; l < in_channels; l++) { const float *const bptr = buffer[l]; if (l == s->lfe_channel) { /* LFE is an input channel but requires no convolution */ /* apply gain to LFE signal and add to output buffer */ *dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe; temp_ir += FFALIGN(n_samples, 16); continue; } /* current read position in ringbuffer: input sample write position * - delay for l-th ch. + diff. betw. IR length and buffer length * (mod buffer length) */ read = (wr - *(delay + l) - (n_samples - 1) + buffer_length) & modulo; if (read + n_samples < buffer_length) { memcpy(temp_src, bptr + read, n_samples * sizeof(*temp_src)); } else { int len = FFMIN(n_samples - (read % n_samples), buffer_length - read); memcpy(temp_src, bptr + read, len * sizeof(*temp_src)); memcpy(temp_src + len, bptr, (n_samples - len) * sizeof(*temp_src)); } /* multiply signal and IR, and add up the results */ dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples); temp_ir += FFALIGN(n_samples, 16); } /* clippings counter */ if (fabs(*dst) > 1) *n_clippings += 1; /* move output buffer pointer by +2 to get to next sample of processed channel: */ dst += 2; src += in_channels; wr = (wr + 1) & modulo; /* update ringbuffer write position */ } *write = wr; /* remember write position in ringbuffer for next call */ return 0; } static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs) { SOFAlizerContext *s = ctx->priv; ThreadData *td = arg; AVFrame *in = td->in, *out = td->out; int offset = jobnr; int *write = &td->write[jobnr]; FFTComplex *hrtf = s->data_hrtf[jobnr]; /* get pointers to current HRTF data */ int *n_clippings = &td->n_clippings[jobnr]; float *ringbuffer = td->ringbuffer[jobnr]; const int n_samples = s->sofa.n_samples; /* length of one IR */ const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */ float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */ const int in_channels = s->n_conv; /* number of input channels */ /* ring buffer length is: longest IR plus max. delay -> next power of 2 */ const int buffer_length = s->buffer_length; /* -1 for AND instead of MODULO (applied to powers of 2): */ const uint32_t modulo = (uint32_t)buffer_length - 1; FFTComplex *fft_in = s->temp_fft[jobnr]; /* temporary array for FFT input/output data */ FFTContext *ifft = s->ifft[jobnr]; FFTContext *fft = s->fft[jobnr]; const int n_conv = s->n_conv; const int n_fft = s->n_fft; int wr = *write; int n_read; int i, j; dst += offset; /* find minimum between number of samples and output buffer length: * (important, if one IR is longer than the output buffer) */ n_read = FFMIN(s->sofa.n_samples, in->nb_samples); for (j = 0; j < n_read; j++) { /* initialize output buf with saved signal from overflow buf */ dst[2 * j] = ringbuffer[wr]; ringbuffer[wr] = 0.0; /* re-set read samples to zero */ /* update ringbuffer read/write position */ wr = (wr + 1) & modulo; } /* initialize rest of output buffer with 0 */ for (j = n_read; j < in->nb_samples; j++) { dst[2 * j] = 0; } for (i = 0; i < n_conv; i++) { if (i == s->lfe_channel) { /* LFE */ for (j = 0; j < in->nb_samples; j++) { /* apply gain to LFE signal and add to output buffer */ dst[2 * j] += src[i + j * in_channels] * s->gain_lfe; } continue; } /* outer loop: go through all input channels to be convolved */ offset = i * n_fft; /* no. samples already processed */ /* fill FFT input with 0 (we want to zero-pad) */ memset(fft_in, 0, sizeof(FFTComplex) * n_fft); for (j = 0; j < in->nb_samples; j++) { /* prepare input for FFT */ /* write all samples of current input channel to FFT input array */ fft_in[j].re = src[j * in_channels + i]; } /* transform input signal of current channel to frequency domain */ av_fft_permute(fft, fft_in); av_fft_calc(fft, fft_in); for (j = 0; j < n_fft; j++) { const float re = fft_in[j].re; const float im = fft_in[j].im; /* complex multiplication of input signal and HRTFs */ /* output channel (real): */ fft_in[j].re = re * (hrtf + offset + j)->re - im * (hrtf + offset + j)->im; /* output channel (imag): */ fft_in[j].im = re * (hrtf + offset + j)->im + im * (hrtf + offset + j)->re; } /* transform output signal of current channel back to time domain */ av_fft_permute(ifft, fft_in); av_fft_calc(ifft, fft_in); for (j = 0; j < in->nb_samples; j++) { /* write output signal of current channel to output buffer */ dst[2 * j] += fft_in[j].re / (float)n_fft; } for (j = 0; j < n_samples - 1; j++) { /* overflow length is IR length - 1 */ /* write the rest of output signal to overflow buffer */ int write_pos = (wr + j) & modulo; *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re / (float)n_fft; } } /* go through all samples of current output buffer: count clippings */ for (i = 0; i < out->nb_samples; i++) { /* clippings counter */ if (fabs(*dst) > 1) { /* if current output sample > 1 */ *n_clippings = *n_clippings + 1; } /* move output buffer pointer by +2 to get to next sample of processed channel: */ dst += 2; } /* remember read/write position in ringbuffer for next call */ *write = wr; return 0; } static int filter_frame(AVFilterLink *inlink, AVFrame *in) { AVFilterContext *ctx = inlink->dst; SOFAlizerContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; int n_clippings[2] = { 0 }; ThreadData td; AVFrame *out; out = ff_get_audio_buffer(outlink, in->nb_samples); if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } av_frame_copy_props(out, in); td.in = in; td.out = out; td.write = s->write; td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings; td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src; td.temp_fft = s->temp_fft; if (s->type == TIME_DOMAIN) { ctx->internal->execute(ctx, sofalizer_convolute, &td, NULL, 2); } else { ctx->internal->execute(ctx, sofalizer_fast_convolute, &td, NULL, 2); } emms_c(); /* display error message if clipping occurred */ if (n_clippings[0] + n_clippings[1] > 0) { av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.\n", n_clippings[0] + n_clippings[1], out->nb_samples * 2); } av_frame_free(&in); return ff_filter_frame(outlink, out); } static int query_formats(AVFilterContext *ctx) { struct SOFAlizerContext *s = ctx->priv; AVFilterFormats *formats = NULL; AVFilterChannelLayouts *layouts = NULL; int ret, sample_rates[] = { 48000, -1 }; ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT); if (ret) return ret; ret = ff_set_common_formats(ctx, formats); if (ret) return ret; layouts = ff_all_channel_layouts(); if (!layouts) return AVERROR(ENOMEM); ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->out_channel_layouts); if (ret) return ret; layouts = NULL; ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_STEREO); if (ret) return ret; ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->in_channel_layouts); if (ret) return ret; sample_rates[0] = s->sample_rate; formats = ff_make_format_list(sample_rates); if (!formats) return AVERROR(ENOMEM); return ff_set_common_samplerates(ctx, formats); } static int load_data(AVFilterContext *ctx, int azim, int elev, float radius) { struct SOFAlizerContext *s = ctx->priv; const int n_samples = s->sofa.n_samples; int n_conv = s->n_conv; /* no. channels to convolve */ int n_fft = s->n_fft; int delay_l[16]; /* broadband delay for each IR */ int delay_r[16]; int nb_input_channels = ctx->inputs[0]->channels; /* no. input channels */ float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10); /* gain - 3dB/channel */ FFTComplex *data_hrtf_l = NULL; FFTComplex *data_hrtf_r = NULL; FFTComplex *fft_in_l = NULL; FFTComplex *fft_in_r = NULL; float *data_ir_l = NULL; float *data_ir_r = NULL; int offset = 0; /* used for faster pointer arithmetics in for-loop */ int m[16]; /* measurement index m of IR closest to required source positions */ int i, j, azim_orig = azim, elev_orig = elev; if (!s->sofa.ncid) { /* if an invalid SOFA file has been selected */ av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n"); return AVERROR_INVALIDDATA; } if (s->type == TIME_DOMAIN) { s->temp_src[0] = av_calloc(FFALIGN(n_samples, 16), sizeof(float)); s->temp_src[1] = av_calloc(FFALIGN(n_samples, 16), sizeof(float)); /* get temporary IR for L and R channel */ data_ir_l = av_calloc(n_conv * FFALIGN(n_samples, 16), sizeof(*data_ir_l)); data_ir_r = av_calloc(n_conv * FFALIGN(n_samples, 16), sizeof(*data_ir_r)); if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) { av_free(data_ir_l); av_free(data_ir_r); return AVERROR(ENOMEM); } } else { /* get temporary HRTF memory for L and R channel */ data_hrtf_l = av_malloc_array(n_fft, sizeof(*data_hrtf_l) * n_conv); data_hrtf_r = av_malloc_array(n_fft, sizeof(*data_hrtf_r) * n_conv); if (!data_hrtf_r || !data_hrtf_l) { av_free(data_hrtf_l); av_free(data_hrtf_r); return AVERROR(ENOMEM); } } for (i = 0; i < s->n_conv; i++) { /* load and store IRs and corresponding delays */ azim = (int)(s->speaker_azim[i] + azim_orig) % 360; elev = (int)(s->speaker_elev[i] + elev_orig) % 90; /* get id of IR closest to desired position */ m[i] = find_m(s, azim, elev, radius); /* load the delays associated with the current IRs */ delay_l[i] = *(s->sofa.data_delay + 2 * m[i]); delay_r[i] = *(s->sofa.data_delay + 2 * m[i] + 1); if (s->type == TIME_DOMAIN) { offset = i * FFALIGN(n_samples, 16); /* no. samples already written */ for (j = 0; j < n_samples; j++) { /* load reversed IRs of the specified source position * sample-by-sample for left and right ear; and apply gain */ *(data_ir_l + offset + j) = /* left channel */ *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j) * gain_lin; *(data_ir_r + offset + j) = /* right channel */ *(s->sofa.data_ir + 2 * m[i] * n_samples + n_samples - 1 - j + n_samples) * gain_lin; } } else { fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l)); fft_in_r = av_calloc(n_fft, sizeof(*fft_in_r)); if (!fft_in_l || !fft_in_r) { av_free(data_hrtf_l); av_free(data_hrtf_r); av_free(fft_in_l); av_free(fft_in_r); return AVERROR(ENOMEM); } offset = i * n_fft; /* no. samples already written */ for (j = 0; j < n_samples; j++) { /* load non-reversed IRs of the specified source position * sample-by-sample and apply gain, * L channel is loaded to real part, R channel to imag part, * IRs ared shifted by L and R delay */ fft_in_l[delay_l[i] + j].re = /* left channel */ *(s->sofa.data_ir + 2 * m[i] * n_samples + j) * gain_lin; fft_in_r[delay_r[i] + j].re = /* right channel */ *(s->sofa.data_ir + (2 * m[i] + 1) * n_samples + j) * gain_lin; } /* actually transform to frequency domain (IRs -> HRTFs) */ av_fft_permute(s->fft[0], fft_in_l); av_fft_calc(s->fft[0], fft_in_l); memcpy(data_hrtf_l + offset, fft_in_l, n_fft * sizeof(*fft_in_l)); av_fft_permute(s->fft[0], fft_in_r); av_fft_calc(s->fft[0], fft_in_r); memcpy(data_hrtf_r + offset, fft_in_r, n_fft * sizeof(*fft_in_r)); } av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n", m[i], *(s->sofa.sp_a + m[i]), *(s->sofa.sp_e + m[i]), *(s->sofa.sp_r + m[i])); } if (s->type == TIME_DOMAIN) { /* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */ memcpy(s->data_ir[0], data_ir_l, sizeof(float) * n_conv * FFALIGN(n_samples, 16)); memcpy(s->data_ir[1], data_ir_r, sizeof(float) * n_conv * FFALIGN(n_samples, 16)); av_freep(&data_ir_l); /* free temporary IR memory */ av_freep(&data_ir_r); } else { s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex)); s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(FFTComplex)); if (!s->data_hrtf[0] || !s->data_hrtf[1]) { av_freep(&data_hrtf_l); av_freep(&data_hrtf_r); av_freep(&fft_in_l); av_freep(&fft_in_r); return AVERROR(ENOMEM); /* memory allocation failed */ } memcpy(s->data_hrtf[0], data_hrtf_l, /* copy HRTF data to */ sizeof(FFTComplex) * n_conv * n_fft); /* filter struct */ memcpy(s->data_hrtf[1], data_hrtf_r, sizeof(FFTComplex) * n_conv * n_fft); av_freep(&data_hrtf_l); /* free temporary HRTF memory */ av_freep(&data_hrtf_r); av_freep(&fft_in_l); /* free temporary FFT memory */ av_freep(&fft_in_r); } memcpy(s->delay[0], &delay_l[0], sizeof(int) * s->n_conv); memcpy(s->delay[1], &delay_r[0], sizeof(int) * s->n_conv); return 0; } static av_cold int init(AVFilterContext *ctx) { SOFAlizerContext *s = ctx->priv; int ret; if (!s->filename) { av_log(ctx, AV_LOG_ERROR, "Valid SOFA filename must be set.\n"); return AVERROR(EINVAL); } /* load SOFA file, */ /* initialize file IDs to 0 before attempting to load SOFA files, * this assures that in case of error, only the memory of already * loaded files is free'd */ s->sofa.ncid = 0; ret = load_sofa(ctx, s->filename, &s->sample_rate); if (ret) { /* file loading error */ av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file: '%s'\n", s->filename); } else { /* no file loading error, resampling not required */ av_log(ctx, AV_LOG_DEBUG, "File '%s' loaded.\n", s->filename); } if (ret) { av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify valid SOFA file.\n"); return ret; } s->fdsp = avpriv_float_dsp_alloc(0); if (!s->fdsp) return AVERROR(ENOMEM); return 0; } static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; SOFAlizerContext *s = ctx->priv; int nb_input_channels = inlink->channels; /* no. input channels */ int n_max_ir = 0; int n_current; int n_max = 0; int ret; if (s->type == FREQUENCY_DOMAIN) { inlink->partial_buf_size = inlink->min_samples = inlink->max_samples = inlink->sample_rate; } /* gain -3 dB per channel, -6 dB to get LFE on a similar level */ s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6) / 20 * M_LN10); s->n_conv = nb_input_channels; /* get size of ringbuffer (longest IR plus max. delay) */ /* then choose next power of 2 for performance optimization */ n_current = s->sofa.n_samples + max_delay(&s->sofa); if (n_current > n_max) { /* length of longest IR plus max. delay (in all SOFA files) */ n_max = n_current; /* length of longest IR (without delay, in all SOFA files) */ n_max_ir = s->sofa.n_samples; } /* buffer length is longest IR plus max. delay -> next power of 2 (32 - count leading zeros gives required exponent) */ s->buffer_length = 1 << (32 - ff_clz(n_max)); s->n_fft = 1 << (32 - ff_clz(n_max + inlink->sample_rate)); if (s->type == FREQUENCY_DOMAIN) { av_fft_end(s->fft[0]); av_fft_end(s->fft[1]); s->fft[0] = av_fft_init(log2(s->n_fft), 0); s->fft[1] = av_fft_init(log2(s->n_fft), 0); av_fft_end(s->ifft[0]); av_fft_end(s->ifft[1]); s->ifft[0] = av_fft_init(log2(s->n_fft), 1); s->ifft[1] = av_fft_init(log2(s->n_fft), 1); if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) { av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft); return AVERROR(ENOMEM); } } /* Allocate memory for the impulse responses, delays and the ringbuffers */ /* size: (longest IR) * (number of channels to convolute) */ s->data_ir[0] = av_calloc(FFALIGN(n_max_ir, 16), sizeof(float) * s->n_conv); s->data_ir[1] = av_calloc(FFALIGN(n_max_ir, 16), sizeof(float) * s->n_conv); /* length: number of channels to convolute */ s->delay[0] = av_malloc_array(s->n_conv, sizeof(float)); s->delay[1] = av_malloc_array(s->n_conv, sizeof(float)); /* length: (buffer length) * (number of input channels), * OR: buffer length (if frequency domain processing) * calloc zero-initializes the buffer */ if (s->type == TIME_DOMAIN) { s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float) * nb_input_channels); } else { s->ringbuffer[0] = av_calloc(s->buffer_length, sizeof(float)); s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float)); s->temp_fft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); s->temp_fft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); if (!s->temp_fft[0] || !s->temp_fft[1]) return AVERROR(ENOMEM); } /* length: number of channels to convolute */ s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim)); s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev)); /* memory allocation failed: */ if (!s->data_ir[0] || !s->data_ir[1] || !s->delay[1] || !s->delay[0] || !s->ringbuffer[0] || !s->ringbuffer[1] || !s->speaker_azim || !s->speaker_elev) return AVERROR(ENOMEM); compensate_volume(ctx); /* get speaker positions */ if ((ret = get_speaker_pos(ctx, s->speaker_azim, s->speaker_elev)) < 0) { av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n"); return ret; } /* load IRs to data_ir[0] and data_ir[1] for required directions */ if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius)) < 0) return ret; av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n", inlink->sample_rate, s->n_conv, nb_input_channels, s->buffer_length); return 0; } static av_cold void uninit(AVFilterContext *ctx) { SOFAlizerContext *s = ctx->priv; if (s->sofa.ncid) { av_freep(&s->sofa.sp_a); av_freep(&s->sofa.sp_e); av_freep(&s->sofa.sp_r); av_freep(&s->sofa.data_delay); av_freep(&s->sofa.data_ir); } av_fft_end(s->ifft[0]); av_fft_end(s->ifft[1]); av_fft_end(s->fft[0]); av_fft_end(s->fft[1]); av_freep(&s->delay[0]); av_freep(&s->delay[1]); av_freep(&s->data_ir[0]); av_freep(&s->data_ir[1]); av_freep(&s->ringbuffer[0]); av_freep(&s->ringbuffer[1]); av_freep(&s->speaker_azim); av_freep(&s->speaker_elev); av_freep(&s->temp_src[0]); av_freep(&s->temp_src[1]); av_freep(&s->temp_fft[0]); av_freep(&s->temp_fft[1]); av_freep(&s->data_hrtf[0]); av_freep(&s->data_hrtf[1]); av_freep(&s->fdsp); } #define OFFSET(x) offsetof(SOFAlizerContext, x) #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM static const AVOption sofalizer_options[] = { { "sofa", "sofa filename", OFFSET(filename), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS }, { "gain", "set gain in dB", OFFSET(gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, { "rotation", "set rotation" , OFFSET(rotation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -360, 360, .flags = FLAGS }, { "elevation", "set elevation", OFFSET(elevation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -90, 90, .flags = FLAGS }, { "radius", "set radius", OFFSET(radius), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 3, .flags = FLAGS }, { "type", "set processing", OFFSET(type), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, .flags = FLAGS, "type" }, { "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, "type" }, { "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, "type" }, { "speakers", "set speaker custom positions", OFFSET(speakers_pos), AV_OPT_TYPE_STRING, {.str=0}, 0, 0, .flags = FLAGS }, { NULL } }; AVFILTER_DEFINE_CLASS(sofalizer); static const AVFilterPad inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, .filter_frame = filter_frame, }, { NULL } }; static const AVFilterPad outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, }, { NULL } }; AVFilter ff_af_sofalizer = { .name = "sofalizer", .description = NULL_IF_CONFIG_SMALL("SOFAlizer (Spatially Oriented Format for Acoustics)."), .priv_size = sizeof(SOFAlizerContext), .priv_class = &sofalizer_class, .init = init, .uninit = uninit, .query_formats = query_formats, .inputs = inputs, .outputs = outputs, .flags = AVFILTER_FLAG_SLICE_THREADS, };