Signed-off-by: Paul B Mahol <one...@gmail.com> --- libavfilter/Makefile | 1 + libavfilter/af_sofalizer.c | 816 +++++++++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 3 files changed, 818 insertions(+) create mode 100644 libavfilter/af_sofalizer.c
diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 8884d1d..d7a3f61 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -87,6 +87,7 @@ OBJS-$(CONFIG_SIDECHAINCOMPRESS_FILTER) += af_sidechaincompress.o OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o +OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o diff --git a/libavfilter/af_sofalizer.c b/libavfilter/af_sofalizer.c new file mode 100644 index 0000000..8e4da74 --- /dev/null +++ b/libavfilter/af_sofalizer.c @@ -0,0 +1,816 @@ +/***************************************************************************** + * 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.m...@gmail.com> + * Wolfgang Hrauda <wolfgang.hra...@gmx.at> + * + * SOFAlizer project coordinator at ARI, main developer of SOFA: + * Piotr Majdak <pi...@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 <netcdf.h> + +#include "libavutil/opt.h" +#include "avfilter.h" +#include "internal.h" +#include "audio.h" + +#define N_SOFA 3 /* no. SOFA files loaded (for instant comparison) */ +#define N_POSITIONS 4 /* no. virtual source positions (advanced settings) */ + +typedef struct NCSofa { /* contains data of one SOFA file */ + int i_ncid; /* netCDF ID of the opened SOFA file */ + int i_n_samples; /* length of one impulse response (IR) */ + int i_m_dim; /* number of measurement positions */ + int *pi_data_delay; /* broadband delay of each IR */ + /* all measurement positions for each receiver (i.e. ear): */ + float *pf_sp_a; /* azimuth angles */ + float *pf_sp_e; /* elevation angles */ + float *pf_sp_r; /* radii */ + /* dataat at each measurement position for each receiver: */ + float *pf_data_ir; /* IRs (time-domain) */ +} NCSofa; + +typedef struct SOFAlizerContext { + const AVClass *class; + + char *filename[N_SOFA]; + struct NCSofa sofa[N_SOFA]; /* contains data of the SOFA files */ + + float *pf_speaker_pos; /* positions of the virtual loudspekaers */ + float f_gain_lfe; + + int i_n_conv; /* number of channels to convolute */ + int i_n_clippings_l; + int i_n_clippings_r; + + /* buffer variables (for convolution) */ + float *pf_ringbuffer_l; /* buffers input samples, length of one buffer: */ + float *pf_ringbuffer_r; /* no. input ch. (incl. LFE) x i_buffer_length */ + int i_write_l; /* current write position to ringbuffer */ + int i_write_r; /* current write position to ringbuffer */ + int i_buffer_length; /* is: longest IR plus max. delay in all SOFA files */ + /* then choose next power of 2 */ + int i_n_longest_filter; /* longest IR + max. delay in all SOFA files */ + int i_output_buffer_length; /* remember no. samples in output buffer */ + + /* netCDF variables */ + int i_sofa_id; /* selected SOFA file */ + int *pi_delay_l; /* broadband delay for each channel/IR to be convolved */ + int *pi_delay_r; + float *pf_data_ir_l; /* IRs for all channels to be convolved */ + float *pf_data_ir_r; /* (this excludes the LFE) */ + + /* control variables */ + float f_gain; /* filter gain (in dB) */ + float f_rotation; /* rotation of virtual loudspeakers (in degrees) */ + float f_elevation; /* elevation of virtual loudspeakers (in deg.) */ + float f_radius; /* distance virtual loudspeakers to listener (in metres) */ + int i_switch; /* 0: source positions according to input format plus */ + /* user's rotation and elevation settings, */ + /* 1-4: virtual source pos. defined in advanced settings */ + /* - from advanced settings: virtual source positions: */ + int pi_azimuth_array[N_POSITIONS]; /* azimuth angles (in deg.) */ + int pi_elevation_array[N_POSITIONS]; /* elevation angles (in deg.) */ + + int lfe; /* whether or not the LFE channel is used */ +} SOFAlizerContext; + +static int close_sofa(struct NCSofa *sofa) +{ + av_freep(&sofa->pi_data_delay); + av_freep(&sofa->pf_sp_a); + av_freep(&sofa->pf_sp_e); + av_freep(&sofa->pf_sp_r); + av_freep(&sofa->pf_data_ir); + nc_close(sofa->i_ncid); + sofa->i_ncid = 0; + return 0; +} + +static int load_sofa(AVFilterContext *ctx, char *filename, + int i_sofa_id , int *samplingrate) +{ + struct SOFAlizerContext *s = ctx->priv; + /* variables associated with content of SOFA file: */ + int i_ncid, i_n_dims, i_n_vars, i_n_gatts, i_n_unlim_dim_id, i_status; + unsigned int i_samplingrate; + int i_n_samples = 0; + int i_m_dim = 0; + s->sofa[i_sofa_id].i_ncid = 0; + i_status = nc_open( filename , NC_NOWRITE, &i_ncid); /* open SOFA file read-only */ + if (i_status != NC_NOERR) + { + av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename); + return -1; + } + /* get number of dimensions, vars, global attributes and Id of unlimited dimensions: */ + nc_inq(i_ncid, &i_n_dims, &i_n_vars, &i_n_gatts, &i_n_unlim_dim_id); + + /* -- get number of measurements ("M") and length of one IR ("N") -- */ + char psz_dim_names[i_n_dims][NC_MAX_NAME]; /* names of netCDF dimensions */ + size_t pi_dim_length[i_n_dims]; /* lengths of netCDF dimensions */ + int i_m_dim_id = -1; + int i_n_dim_id = -1; + for( int i = 0; i < i_n_dims; i++ ) /* go through all dimensions of file */ + { + nc_inq_dim( i_ncid, i, psz_dim_names[i], &pi_dim_length[i] ); /* get dimensions */ + if ( !strncmp("M", psz_dim_names[i], 1 ) ) /* get ID of dimension "M" */ + i_m_dim_id = i; + if ( !strncmp("N", psz_dim_names[i], 1 ) ) /* get ID of dimension "N" */ + i_n_dim_id = i; + } + if( ( i_m_dim_id == -1 ) || ( i_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"); + nc_close(i_ncid); + return -1; + } + i_n_samples = pi_dim_length[i_n_dim_id]; /* get number of measurements */ + i_m_dim = pi_dim_length[i_m_dim_id]; /* get length of one IR */ + + /* -- check file type -- */ + size_t i_att_len; /* get length of attritube "Conventions" */ + i_status = nc_inq_attlen(i_ncid, NC_GLOBAL, "Conventions", &i_att_len); + if (i_status != NC_NOERR) + { + av_log(ctx, AV_LOG_ERROR, "Can't get length of attribute \"Conventions\".\n"); + nc_close(i_ncid); + return -1; + } + char psz_conventions[i_att_len + 1]; /* check whether file is SOFA file */ + nc_get_att_text(i_ncid , NC_GLOBAL, "Conventions", psz_conventions); + *( psz_conventions + i_att_len ) = 0; + if ( strncmp( "SOFA" , psz_conventions, 4 ) ) + { + av_log(ctx, AV_LOG_ERROR, "Not a SOFA file!\n"); + nc_close(i_ncid); + return -1; + } + + /* -- check if attribute "SOFAConventions" is "SimpleFreeFieldHRIR": -- */ + nc_inq_attlen (i_ncid, NC_GLOBAL, "SOFAConventions", &i_att_len ); + char psz_sofa_conventions[i_att_len + 1]; + nc_get_att_text(i_ncid, NC_GLOBAL, "SOFAConventions", psz_sofa_conventions); + *( psz_sofa_conventions + i_att_len ) = 0; + if ( strncmp( "SimpleFreeFieldHRIR" , psz_sofa_conventions, i_att_len ) ) + { + av_log(ctx, AV_LOG_ERROR, "Not a SimpleFreeFieldHRIR file!\n"); + nc_close(i_ncid); + return -1; + } + + /* -- get sampling rate of HRTFs -- */ + int i_samplingrate_id; /* read ID, then value */ + i_status = nc_inq_varid( i_ncid, "Data.SamplingRate", &i_samplingrate_id); + i_status += nc_get_var_uint( i_ncid, i_samplingrate_id, &i_samplingrate ); + if (i_status != NC_NOERR) + { + av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.SamplingRate.\n"); + nc_close(i_ncid); + return -1; + } + *samplingrate = i_samplingrate; /* remember sampling rate */ + + /* -- allocate memory for one value for each measurement position: -- */ + float *pf_sp_a = s->sofa[i_sofa_id].pf_sp_a = av_malloc(sizeof(float) * i_m_dim); + float *pf_sp_e = s->sofa[i_sofa_id].pf_sp_e = av_malloc(sizeof(float) * i_m_dim); + float *pf_sp_r = s->sofa[i_sofa_id].pf_sp_r = av_malloc(sizeof(float) * i_m_dim); + /* delay and IR values required for each ear and measurement position: */ + int *pi_data_delay = s->sofa[i_sofa_id].pi_data_delay = + av_calloc (i_m_dim * 2, sizeof(int)); + float *pf_data_ir = s->sofa[i_sofa_id].pf_data_ir = + av_malloc(sizeof(float) * 2 * i_m_dim * i_n_samples); + + if (!pi_data_delay || !pf_sp_a || !pf_sp_e || !pf_sp_r || !pf_data_ir) { + /* if memory could not be allocated */ + close_sofa( &s->sofa[i_sofa_id] ); + return AVERROR(ENOMEM); + } + + /* get impulse responses (HRTFs): */ + int i_data_ir_id; /* get corresponding ID */ + i_status = nc_inq_varid( i_ncid, "Data.IR", &i_data_ir_id); + i_status += nc_get_var_float( i_ncid, i_data_ir_id, pf_data_ir); /* read and store IRs */ + if (i_status != NC_NOERR) { + av_log( ctx, AV_LOG_ERROR, "Couldn't read Data.IR!" ); + goto error; + } + + /* get source positions of the HRTFs in the SOFA file: */ + int i_sp_id; + i_status = nc_inq_varid(i_ncid, "SourcePosition", &i_sp_id); /* get corresponding ID */ + i_status += nc_get_vara_float (i_ncid, i_sp_id, (size_t[2]){ 0 , 0 } , + (size_t[2]){ i_m_dim , 1 } , pf_sp_a ); /* read & store azimuth angles */ + i_status += nc_get_vara_float (i_ncid, i_sp_id, (size_t[2]){ 0 , 1 } , + (size_t[2]){ i_m_dim , 1 } , pf_sp_e ); /* read & store elevation angles */ + i_status += nc_get_vara_float (i_ncid, i_sp_id, (size_t[2]){ 0 , 2 } , + (size_t[2]){ i_m_dim , 1 } , pf_sp_r ); /* read & store radii */ + if (i_status != NC_NOERR) { /* if any source position variable coudn't be read */ + av_log(ctx, AV_LOG_ERROR, "Couldn't read SourcePosition.\n"); + goto error; + } + + /* read Data.Delay, check for errors and fit it to pi_data_delay */ + int i_data_delay_id; + int pi_data_delay_dim_id[2]; + char psz_data_delay_dim_name[NC_MAX_NAME]; + + i_status = nc_inq_varid(i_ncid, "Data.Delay", &i_data_delay_id); + i_status += nc_inq_vardimid ( i_ncid, i_data_delay_id, &pi_data_delay_dim_id[0]); + i_status += nc_inq_dimname ( i_ncid, pi_data_delay_dim_id[0], psz_data_delay_dim_name ); + if (i_status != NC_NOERR) { + av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay.\n"); + goto error; + } + + /* Data.Delay dimension check */ + /* dimension of Data.Delay is [I R]: */ + if (!strncmp(psz_data_delay_dim_name, "I", 2)) { + /* check 2 characters to assure string is 0-terminated after "I" */ + int pi_Delay[2]; /* delays get from SOFA file: */ + av_log ( ctx, AV_LOG_DEBUG, "Data.Delay has dimension [I R]\n" ); + i_status = nc_get_var_int( i_ncid, i_data_delay_id, &pi_Delay[0] ); + if ( i_status != NC_NOERR ) + { + av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n"); + goto error; + } + int *pi_data_delay_r = pi_data_delay + i_m_dim; + for ( int i = 0 ; i < i_m_dim ; i++ ) /* extend given dimension [I R] to [M R] */ + { /* assign constant delay value for all measurements to data_delay fields */ + *( pi_data_delay + i ) = pi_Delay[0]; + *( pi_data_delay_r + i ) = pi_Delay[1]; + } + /* dimension of Data.Delay is [M R] */ + } else if (!strncmp(psz_data_delay_dim_name, "M", 2)) { + av_log( ctx, AV_LOG_ERROR, "Data.Delay in dimension [M R]\n"); + /* get delays from SOFA file: */ + i_status = nc_get_var_int( i_ncid, i_data_delay_id, pi_data_delay ); + if (i_status != NC_NOERR) + { + av_log(ctx, AV_LOG_ERROR, "Couldn't read Data.Delay\n"); + 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"); + goto error; + } + + /* save information in SOFA struct: */ + s->sofa[i_sofa_id].i_m_dim = i_m_dim; /* no. measurement positions */ + s->sofa[i_sofa_id].i_n_samples = i_n_samples; /* length on one IR */ + s->sofa[i_sofa_id].i_ncid = i_ncid; /* netCDF ID of SOFA file */ + nc_close(i_ncid); /* close SOFA file */ + + return 0; + +error: + close_sofa( &s->sofa[i_sofa_id] ); + return -1; +} + +static int get_speaker_pos(AVFilterContext *ctx, float *pf_speaker_pos ) +{ + /* get input channel configuration: */ + uint64_t i_physical_channels = ctx->inputs[0]->channel_layout; + float *pf_pos_temp; + int i_input_nb = ctx->inputs[0]->channels; /* get no. input channels */ + int i_n_conv = i_input_nb; + if (i_physical_channels & AV_CH_LOW_FREQUENCY) { /* if LFE is used */ + /* decrease number of channels to be convolved: */ + i_n_conv = i_input_nb - 1; + } + + /* set speaker positions according to input channel configuration: */ + switch (i_physical_channels) { + case AV_CH_LAYOUT_MONO: + pf_pos_temp = (float[1]){ 0 }; + break; + case AV_CH_LAYOUT_STEREO: + case AV_CH_LAYOUT_2_1: + pf_pos_temp = (float[2]){ 30 , 330 }; + break; + case AV_CH_LAYOUT_SURROUND: + case AV_CH_LAYOUT_3POINT1: + pf_pos_temp = (float[3]){ 30 , 330 , 0 }; + break; + case AV_CH_LAYOUT_4POINT0: + case AV_CH_LAYOUT_4POINT1: pf_pos_temp = (float[4]){ 30 , 330 , 120 , 240 }; + break; + case AV_CH_LAYOUT_5POINT0: + case AV_CH_LAYOUT_5POINT1: + pf_pos_temp = (float[5]){ 30 , 330 , 120 , 240 , 0 }; + break; + case AV_CH_LAYOUT_6POINT0: + pf_pos_temp = (float[6]){ 30 , 330 , 90 , 270 , 150 , 210 }; + break; + case AV_CH_LAYOUT_7POINT0: + case AV_CH_LAYOUT_7POINT1: + pf_pos_temp = (float[7]){ 30 , 330 , 90 , 270 , 150 , 210 , 0 }; + break; + //case AV_CH_LAYOUT_8POINT1: + // pf_pos_temp = (float[8]){ 30 , 330 , 90 , 270 , 150 , 210 , 180 , 0 }; + // break; + default: + return -1; + } + + memcpy(pf_speaker_pos, pf_pos_temp, i_n_conv * sizeof(float)); + + return 0; + +} + +static int MaxDelay ( struct NCSofa *sofa ) +{ + int i_max = 0; + for ( int i = 0; i < ( sofa->i_m_dim * 2 ) ; i++ ) + { /* search maximum delay in given SOFA file */ + if ( *( sofa->pi_data_delay + i ) > i_max ) + i_max = *( sofa->pi_data_delay + i) ; + } + return i_max; +} + +static int find_m(SOFAlizerContext *s, int azim, int elev, float radius) +{ + /* get source positions and M of currently selected SOFA file */ + float *pf_sp_a = s->sofa[s->i_sofa_id].pf_sp_a; /* azimuth angle */ + float *pf_sp_e = s->sofa[s->i_sofa_id].pf_sp_e; /* elevation angle */ + float *pf_sp_r = s->sofa[s->i_sofa_id].pf_sp_r; /* radius */ + int i_m_dim = s->sofa[s->i_sofa_id].i_m_dim; /* no. measurements */ + + int i_best_id = 0; /* index m currently closest to desired source pos. */ + float delta = 1000; /* offset between desired and currently best pos. */ + float f_current; + for ( int i = 0; i < i_m_dim ; i++ ) + { /* search through all measurements in currently selected SOFA file */ + /* distance of current to desired source position: */ + f_current = fabs ( *(pf_sp_a++) - azim ) + + fabs( *(pf_sp_e++) - elev ) + + fabs( *(pf_sp_r++) - radius ); + if ( f_current <= delta ) + { /* if current distance is smaller than smallest distance so far */ + delta = f_current; + i_best_id = i; /* remember index */ + } + } + return i_best_id; +} + +static int compensate_volume(AVFilterContext *ctx) +{ + struct SOFAlizerContext *s = ctx->priv; + float f_energy = 0; + int i_m; + int i_sofa_id_backup = s->i_sofa_id; + float *pf_ir; + float f_compensate; + /* compensate volume for each SOFA file */ + for ( int i = 0 ; i < N_SOFA ; i++ ) { /* go through all SOFA files */ + if( s->sofa[i].i_ncid ) { + /* find IR at front center position in i-th SOFA file (IR closest to 0°,0°,1m) */ + struct NCSofa *p_sofa = &s->sofa[i]; + s->i_sofa_id = i; + i_m = find_m( s, 0, 0, 1 ); + /* get energy of that IR and compensate volume */ + pf_ir = p_sofa->pf_data_ir + 2 * i_m * p_sofa->i_n_samples; + for (int j = 0 ; j < p_sofa->i_n_samples ; j++) { + f_energy += *(pf_ir + j ) * *(pf_ir + j); + } + f_compensate = 256 / (p_sofa->i_n_samples * sqrt(f_energy)); + av_log(ctx, AV_LOG_DEBUG, "Compensate-factor: %f\n", f_compensate); + pf_ir = p_sofa->pf_data_ir; + for (int j = 0; j < ( p_sofa->i_n_samples * p_sofa->i_m_dim * 2 ) ; j++) { + *( pf_ir + j ) *= f_compensate; /* apply volume compensation to IRs */ + } + } + } + + s->i_sofa_id = i_sofa_id_backup; + + return 0; +} + +static void sofalizer_convolute(SOFAlizerContext *s, AVFrame *in, AVFrame *out, int offset, + int *write, int *delay, float *pf_ir, int *i_n_clippings, + float *ringbuffer) +{ + int i_n_samples = s->sofa[s->i_sofa_id].i_n_samples; /* length of one IR */ + const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */ + float *pf_temp_ir; + float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */ + int i_read; + int i_input_nb = in->channels; /* number of input channels */ + /* ring buffer length is: longest IR plus max. delay -> next power of 2 */ + int i_buffer_length = s->i_buffer_length; + /* -1 for AND instead of MODULO (applied to powers of 2): */ + uint32_t i_modulo = (uint32_t) i_buffer_length - 1; + float *pf_ringbuffer[i_input_nb]; /* holds ringbuffer for each input channel */ + + dst += offset; + for (int l = 0 ; l < i_input_nb ; l++ ) { + /* get starting address of ringbuffer for each input channel */ + pf_ringbuffer[l] = ringbuffer + l * i_buffer_length ; + } + int i_write = *write; + + for (int i = 0; i < in->nb_samples; i++) + { /* i is not used as an index in the loop! */ + *(dst) = 0; + for ( int l = 0 ; l < i_input_nb ; l++ ) + { /* write current input sample to ringbuffer (for each channel) */ + *( pf_ringbuffer[l] + i_write ) = *src++; + } + pf_temp_ir = pf_ir; /* using same set of IRs for each sample */ + /* loop goes through all channels to be convolved (excl. LFE): */ + for ( int l = 0 ; l < s->i_n_conv ; l++ ) + { + /* current read position in ringbuffer: input sample write position + * - delay for l-th ch. + diff. betw. IR length and buffer length + * (mod buffer length) */ + i_read = (i_write - *(delay + l) - + (i_n_samples - 1 ) + i_buffer_length ) & i_modulo; + + for (int j = 0; j < i_n_samples; j++) { /* go through samples of IR */ + /* multiply signal and IR, and add up the results */ + *dst += *(pf_ringbuffer[l] + ((i_read++) & i_modulo)) * *(pf_temp_ir++); + } + } + if (s->lfe) { /* LFE */ + /* apply gain to LFE signal and add to output buffer */ + *dst += *( pf_ringbuffer[s->i_n_conv] + i_write ) * s->f_gain_lfe; + } + /* clippings counter */ + if (fabs(*dst) >= 1) + *i_n_clippings += 1; + + /* move output buffer pointer by +2 to get to next sample of processed channel: */ + dst += 2; + i_write = ( i_write + 1 ) & i_modulo; /* update ringbuffer write position */ + } + + *write = i_write; /* remember write position in ringbuffer for next call */ + + return; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + SOFAlizerContext *s = ctx->priv; + AVFilterLink *outlink = ctx->outputs[0]; + 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); + + /* gain -3 dB per channel, -6 dB to get LFE on a similar level */ + s->f_gain_lfe = expf((s->f_gain - 3 * inlink->channels - 6) / 20 * logf(10)); + + sofalizer_convolute(s, in, out, 0, &s->i_write_l, + s->pi_delay_l, s->pf_data_ir_l, + &s->i_n_clippings_l, + s->pf_ringbuffer_l); + sofalizer_convolute(s, in, out, 1, &s->i_write_r, + s->pi_delay_r, s->pf_data_ir_r, + &s->i_n_clippings_r, + s->pf_ringbuffer_r); + + /* display error message if clipping occured */ + if (s->i_n_clippings_l + s->i_n_clippings_r > 0 ) { + av_log(ctx, AV_LOG_WARNING, "%d of %d samples clipped. Please reduce gain.", + s->i_n_clippings_l + s->i_n_clippings_r, out->nb_samples * 2); + } + + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts = NULL; + static int sample_rates[] = { 48000, -1 }; + int ret; + + ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT); + if (ret) + return ret; + ret = ff_set_common_formats(ctx, formats); + if (ret) + return ret; + + ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_5POINT0); + if (ret) + return ret; + ret = ff_add_channel_layout(&layouts, AV_CH_LAYOUT_5POINT1); + if (ret) + return ret; + + 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; + + 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 i_azim, int i_elev, float f_radius) +{ + struct SOFAlizerContext *s = ctx->priv; + if (!s->sofa[s->i_sofa_id].i_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." ); + return -1; + } + const int i_n_samples = s->sofa[s->i_sofa_id].i_n_samples; + int i_n_conv = s->i_n_conv; /* no. channels to convolve (excl. LFE) */ + int pi_delay_l[i_n_conv]; /* broadband delay for each IR */ + int pi_delay_r[i_n_conv]; + int i_input_nb = ctx->inputs[0]->channels; /* no. input channels */ + float f_gain_lin = expf( (s->f_gain - 3 * i_input_nb) / 20 * logf(10) ); /* gain - 3dB/channel */ + + float *pf_data_ir_l = NULL; + float *pf_data_ir_r = NULL; + + /* get temporary IR for L and R channel */ + pf_data_ir_l = av_malloc(sizeof(float) * i_n_conv * i_n_samples); + pf_data_ir_r = av_malloc(sizeof(float) * i_n_conv * i_n_samples); + if (!pf_data_ir_r || !pf_data_ir_l) + return AVERROR(ENOMEM); + + int i_offset = 0; /* used for faster pointer arithmetics in for-loop */ + + int i_m[s->i_n_conv]; /* measurement index m of IR closest to required source positions */ + if (s->i_switch) { /* if switch not 0, use pre-defined virtual source positions */ + i_azim = s->pi_azimuth_array[s->i_switch - 1]; + i_elev = s->pi_elevation_array[s->i_switch -1]; + } + int i_azim_orig = i_azim; + + for ( int i = 0; i < s->i_n_conv; i++ ) { + /* load and store IRs and corresponding delays */ + i_azim = (int)( s->pf_speaker_pos[i] + i_azim_orig ) % 360; + /* get id of IR closest to desired position */ + i_m[i] = find_m( s, i_azim, i_elev, f_radius ); + + /* load the delays associated with the current IRs */ + pi_delay_l[i] = *( s->sofa[s->i_sofa_id].pi_data_delay + 2 * i_m[i] ); + pi_delay_r[i] = *( s->sofa[s->i_sofa_id].pi_data_delay + 2 * i_m[i] + 1); + + i_offset = i * i_n_samples; /* no. samples already written */ + for (int j = 0; j < i_n_samples; j++) { + /* load reversed IRs of the specified source position + * sample-by-sample for left and right ear; and apply gain */ + *( pf_data_ir_l + i_offset + j ) = /* left channel */ + *( s->sofa[s->i_sofa_id].pf_data_ir + + 2 * i_m[i] * i_n_samples + i_n_samples - 1 - j ) * f_gain_lin; + *( pf_data_ir_r + i_offset + j ) = /* right channel */ + *( s->sofa[s->i_sofa_id].pf_data_ir + + 2 * i_m[i] * i_n_samples + i_n_samples - 1 - j + i_n_samples ) * f_gain_lin; + } + + av_log(ctx, AV_LOG_DEBUG, "Index: %d, Azimuth: %f, Elevation: %f, Radius: %f of SOFA file.\n", + i_m[i], *(s->sofa[s->i_sofa_id].pf_sp_a + i_m[i]), + *(s->sofa[s->i_sofa_id].pf_sp_e + i_m[i]), + *(s->sofa[s->i_sofa_id].pf_sp_r + i_m[i]) ); + } + + /* copy IRs and delays to allocated memory in the SOFAlizerContext struct: */ + memcpy(s->pf_data_ir_l, pf_data_ir_l, + sizeof(float) * i_n_conv * i_n_samples); + memcpy(s->pf_data_ir_r, pf_data_ir_r, + sizeof(float) * i_n_conv * i_n_samples); + + av_free(pf_data_ir_l); /* free temporary IR memory */ + av_free(pf_data_ir_r); + + memcpy(s->pi_delay_l, &pi_delay_l[0], sizeof(int) * s->i_n_conv); + memcpy(s->pi_delay_r, &pi_delay_r[0], sizeof(int) * s->i_n_conv); + + return 0; +} + +static inline unsigned clz(unsigned x) +{ + unsigned i = sizeof(x) * 8; + + while (x) { + x >>= 1; + i--; + } + + return i; +} + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + SOFAlizerContext *s = ctx->priv; + int i_samplingrate = 0; /* get sampling rate of audio file/stream: */ + int i_status = 0; /* zero, if no file could be loaded */ + + /* load SOFA files, resample if sampling rate different from audio file */ + for (int i = 0; i < N_SOFA; i++) { + /* 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 ( e.g. in FreeAllSofa() ) */ + s->sofa[i].i_ncid = 0; + } + for (int i = 0; i < N_SOFA; i++ ) { + if ( load_sofa( ctx, s->filename[i], i , &i_samplingrate) != 0 ) + { /* file loading error */ + av_log(ctx, AV_LOG_ERROR, "Error while loading SOFA file %d: '%s'\n", i + 1, s->filename[i] ); + } else { /* no file loading error, resampling not required */ + av_log(ctx, AV_LOG_DEBUG, "File %d: '%s' loaded.\n", i + 1 , s->filename[i] ); + i_status++; /* increase status after successful file loading */ + } + } + if (!i_status) { + av_log(ctx, AV_LOG_ERROR, "No valid SOFA file could be loaded. Please specify at least one valid SOFA file.\n" ); + return -1; + } + + int i_input_nb = inlink->channels; /* no. input channels */ + if (inlink->channel_layout & AV_CH_LOW_FREQUENCY) { /* if LFE is used */ + s->lfe = 1; + s->i_n_conv = i_input_nb - 1 ; /* LFE is an input channel but requires no convolution */ + } else /* if LFE is not used */ { + s->lfe = 0; + s->i_n_conv = i_input_nb ; + } + + /* get size of ringbuffer (longest IR plus max. delay) */ + /* then choose next power of 2 for performance optimization */ + int i_n_max = 0; + int i_n_current; + int i_n_max_ir = 0; + for ( int i = 0 ; i < N_SOFA ; i++ ) + { /* go through all SOFA files and determine the longest IR */ + if ( s->sofa[i].i_ncid != 0 ) + { + i_n_current = s->sofa[i].i_n_samples + MaxDelay ( &s->sofa[i] ); + if ( i_n_current > i_n_max ) + { + /* length of longest IR plus max. delay (in all SOFA files) */ + i_n_max = i_n_current; + /* length of longest IR (without delay, in all SOFA files) */ + i_n_max_ir = s->sofa[i].i_n_samples; + } + } + } + s->i_n_longest_filter = i_n_max; /* longest IR plus max. delay */ + /* buffer length is longest IR plus max. delay -> next power of 2 + (32 - count leading zeros gives required exponent) */ + s->i_buffer_length = exp2(32 - clz((uint32_t)i_n_max)); + + s->i_output_buffer_length = 0; /* initialization */ + + /* Allocate memory for the impulse responses, delays and the ringbuffers */ + /* size: (longest IR) * (number of channels to convolute), without LFE */ + s->pf_data_ir_l = malloc( sizeof(float) * i_n_max_ir * s->i_n_conv ); + s->pf_data_ir_r = malloc( sizeof(float) * i_n_max_ir * s->i_n_conv ); + /* length: number of channels to convolute */ + s->pi_delay_l = malloc ( sizeof( int ) * s->i_n_conv ); + s->pi_delay_r = malloc ( sizeof( int ) * s->i_n_conv ); + /* length: (buffer length) * (number of input channels), + * OR: buffer length (if frequency domain processing) + * calloc zero-initializes the buffer */ + s->pf_ringbuffer_l = av_calloc(s->i_buffer_length * i_input_nb, sizeof(float)); + s->pf_ringbuffer_r = av_calloc(s->i_buffer_length * i_input_nb, sizeof(float)); + /* length: number of channels to convolute */ + s->pf_speaker_pos = malloc( sizeof( float) * s->i_n_conv ); + + /* memory allocation failed: */ + if ( !s->pf_data_ir_l || !s->pf_data_ir_r || !s->pi_delay_l || + !s->pi_delay_r || !s->pf_ringbuffer_l || !s->pf_ringbuffer_r || + !s->pf_speaker_pos ) + return AVERROR(ENOMEM); + + compensate_volume(ctx); + + /* get speaker positions */ + if (get_speaker_pos(ctx, s->pf_speaker_pos)) { + av_log(ctx, AV_LOG_ERROR, "Couldn't get speaker positions. Input channel configuration not supported.\n"); + return -1; + } + /* load IRs to pf_data_ir_l and pf_data_ir_r for required directions */ + /* only load IRs if time-domain convolution is used, + * otherwise, data is loaded on FFT size change */ + if (load_data(ctx, s->f_rotation, s->f_elevation, s->f_radius)) { + return AVERROR(ENOMEM); + } + + av_log(ctx, AV_LOG_DEBUG, "Samplerate: %d Channels to convolute: %d, Length of ringbuffer: %d x %d\n", + inlink->sample_rate, s->i_n_conv, i_input_nb, (int)s->i_buffer_length ); + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + SOFAlizerContext *s = ctx->priv; + + /* go through all SOFA files and free associated memory: */ + for ( int i = 0 ; i < N_SOFA ; i++) { + if (s->sofa[i].i_ncid) { + av_freep(&s->sofa[i].pf_sp_a ); + av_freep(&s->sofa[i].pf_sp_e ); + av_freep(&s->sofa[i].pf_sp_r ); + av_freep(&s->sofa[i].pi_data_delay ); + av_freep(&s->sofa[i].pf_data_ir ); + } + } + av_freep(&s->pi_delay_l ); + av_freep(&s->pi_delay_r ); + av_freep(&s->pf_data_ir_l ); + av_freep(&s->pf_data_ir_r ); + av_freep(&s->pf_ringbuffer_l ); + av_freep(&s->pf_ringbuffer_r ); + av_freep(&s->pf_speaker_pos ); +} + +#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[0]), AV_OPT_TYPE_STRING, {.str=NULL}, .flags = FLAGS }, + { "gain", "set gain in dB", OFFSET(f_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .flags = FLAGS }, + { "rotation", "set rotation" , OFFSET(f_rotation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -360, 360, .flags = FLAGS }, + { "elevation", "set elevation", OFFSET(f_elevation), AV_OPT_TYPE_FLOAT, {.dbl=0}, -90, 90, .flags = FLAGS }, + { "radius", "set radius", OFFSET(f_radius), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 2.1, .flags = FLAGS }, + { "azi1", "set pos 1. azimuth", OFFSET(pi_azimuth_array[0]), AV_OPT_TYPE_INT, {.i64=90}, -180, 180, .flags = FLAGS }, + { "ele1", "set pos 1. elevation", OFFSET(pi_elevation_array[0]), AV_OPT_TYPE_INT, {.i64=0}, -90, 90, .flags = FLAGS }, + { "azi2", "set pos 2. azimuth", OFFSET(pi_azimuth_array[1]), AV_OPT_TYPE_INT, {.i64=180}, -180, 180, .flags = FLAGS }, + { "ele2", "set pos 2. elevation", OFFSET(pi_elevation_array[1]), AV_OPT_TYPE_INT, {.i64=0}, -90, 90, .flags = FLAGS }, + { "azi3", "set pos 3. azimuth", OFFSET(pi_azimuth_array[2]), AV_OPT_TYPE_INT, {.i64=-90}, -180, 180, .flags = FLAGS }, + { "ele3", "set pos 3. elevation", OFFSET(pi_elevation_array[2]), AV_OPT_TYPE_INT, {.i64=0}, -90, 90, .flags = FLAGS }, + { "azi4", "set pos 4. azimuth", OFFSET(pi_azimuth_array[3]), AV_OPT_TYPE_INT, {.i64=0}, -180, 180, .flags = FLAGS }, + { "ele4", "set pos 4. elevation", OFFSET(pi_elevation_array[3]), AV_OPT_TYPE_INT, {.i64=90}, -90, 90, .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, + .uninit = uninit, + .query_formats = query_formats, + .inputs = inputs, + .outputs = outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 0eeef53..131e067 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -109,6 +109,7 @@ void avfilter_register_all(void) REGISTER_FILTER(SIDECHAINGATE, sidechaingate, af); REGISTER_FILTER(SILENCEDETECT, silencedetect, af); REGISTER_FILTER(SILENCEREMOVE, silenceremove, af); + REGISTER_FILTER(SOFALIZER, sofalizer, af); REGISTER_FILTER(STEREOTOOLS, stereotools, af); REGISTER_FILTER(STEREOWIDEN, stereowiden, af); REGISTER_FILTER(TREBLE, treble, af); -- 1.9.1 _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-devel