/***************************************************************************** * 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 * Wolfgang Hrauda * * SOFAlizer project coordinator at ARI, main developer of SOFA: * Piotr Majdak * * 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 #include #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 MySofa { /* contains data of one SOFA file */ struct MYSOFA_HRTF *hrtf; struct MYSOFA_LOOKUP *lookup; struct MYSOFA_NEIGHBORHOOD *neighborhood; int ir_samples; /* length of one impulse response (IR) */ int n_samples; /* ir_samples to next power of 2 */ float *lir, *rir; /* IRs (time-domain) */ float *fir; int max_delay; } MySofa; typedef struct VirtualSpeaker { uint8_t set; float azim; float elev; } VirtualSpeaker; typedef struct SOFAlizerContext { const AVClass *class; char *filename; /* name of SOFA file */ MySofa 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 lfe_gain; /* initial gain for the LFE channel */ 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]; /* Array to hold FFT values */ FFTComplex *temp_afft[2]; /* Array to accumulate FFT values prior to IFFT */ /* 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 */ int framesize; /* size of buffer */ int normalize; /* should all IRs be normalized upon import ? */ int interpolate; /* should wanted IRs be interpolated from neighbors ? */ int minphase; /* should all IRs be minphased upon import ? */ float anglestep; /* neighbor search angle step, in agles */ float radstep; /* neighbor search radius step, in meters */ VirtualSpeaker vspkrpos[64]; FFTContext *fft[2], *ifft[2]; FFTComplex *data_hrtf[2]; AVFloatDSPContext *fdsp; } SOFAlizerContext; static int close_sofa(struct MySofa *sofa) { if (sofa->neighborhood) mysofa_neighborhood_free(sofa->neighborhood); sofa->neighborhood = NULL; if (sofa->lookup) mysofa_lookup_free(sofa->lookup); sofa->lookup = NULL; if (sofa->hrtf) mysofa_free(sofa->hrtf); sofa->hrtf = NULL; av_freep(&sofa->fir); return 0; } static int preload_sofa(AVFilterContext *ctx, char *filename, int *samplingrate) { struct SOFAlizerContext *s = ctx->priv; struct MYSOFA_HRTF *mysofa; char *license; int ret; mysofa = mysofa_load(filename, &ret); s->sofa.hrtf = mysofa; if (ret || !mysofa) { av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename); return AVERROR(EINVAL); } ret = mysofa_check(mysofa); if (ret != MYSOFA_OK) { av_log(ctx, AV_LOG_ERROR, "Selected SOFA file is invalid. Please select valid SOFA file.\n"); return ret; } if (s->normalize) mysofa_loudness(s->sofa.hrtf); if (s->minphase) mysofa_minphase(s->sofa.hrtf, 0.01f); mysofa_tocartesian(s->sofa.hrtf); s->sofa.lookup = mysofa_lookup_init(s->sofa.hrtf); if (s->sofa.lookup == NULL) return AVERROR(EINVAL); if (s->interpolate) s->sofa.neighborhood = mysofa_neighborhood_init_withstepdefine(s->sofa.hrtf, s->sofa.lookup, s->anglestep, s->radstep); s->sofa.fir = av_calloc(s->sofa.hrtf->N * s->sofa.hrtf->R, sizeof(*s->sofa.fir)); if (!s->sofa.fir) return AVERROR(ENOMEM); if (mysofa->DataSamplingRate.elements != 1) return AVERROR(EINVAL); av_log(ctx, AV_LOG_DEBUG, "Original IR length: %d.\n", mysofa->N); *samplingrate = mysofa->DataSamplingRate.values[0]; license = mysofa_getAttribute(mysofa->attributes, (char *)"License"); if (license) av_log(ctx, AV_LOG_INFO, "SOFA license: %s\n", license); return 0; } static int parse_channel_name(char **arg, int *rchannel, char *buf) { int len, i, channel_id = 0; int64_t layout, layout0; /* try to parse a channel name, e.g. "FL" */ if (av_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 >= 1LL << i) { channel_id += i; layout >>= i; } } /* reject layouts that are not a single channel */ if (channel_id >= 64 || layout0 != 1LL << 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))) { char buf[8]; float azim, elev; int out_ch_id; p = NULL; if (parse_channel_name(&arg, &out_ch_id, buf)) { av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf); continue; } if (av_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 (av_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; } typedef struct ThreadData { AVFrame *in, *out; int *write; int **delay; float **ir; int *n_clippings; float **ringbuffer; float **temp_src; FFTComplex **temp_fft; FFTComplex **temp_afft; } 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 ir_samples = s->sofa.ir_samples; /* length of one IR */ const int n_samples = s->sofa.n_samples; const int planar = in->format == AV_SAMPLE_FMT_FLTP; const int mult = 1 + !planar; const float *src = (const float *)in->extended_data[0]; /* get pointer to audio input buffer */ float *dst = (float *)out->extended_data[jobnr * planar]; /* 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; if (!planar) 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] = 0; if (planar) { for (l = 0; l < in_channels; l++) { const float *srcp = (const float *)in->extended_data[l]; /* write current input sample to ringbuffer (for each channel) */ buffer[l][wr] = srcp[i]; } } else { 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[0] += *(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) { memmove(temp_src, bptr + read, n_samples * sizeof(*temp_src)); } else { int len = FFMIN(n_samples - (read % n_samples), buffer_length - read); memmove(temp_src, bptr + read, len * sizeof(*temp_src)); memmove(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, FFALIGN(ir_samples, 32)); temp_ir += n_samples; } /* clippings counter */ if (fabsf(dst[0]) > 1) n_clippings[0]++; /* move output buffer pointer by +2 to get to next sample of processed channel: */ dst += mult; 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 int planar = in->format == AV_SAMPLE_FMT_FLTP; const int mult = 1 + !planar; float *dst = (float *)out->extended_data[jobnr * planar]; /* 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 */ FFTComplex *fft_acc = s->temp_afft[jobnr]; FFTContext *ifft = s->ifft[jobnr]; FFTContext *fft = s->fft[jobnr]; const int n_conv = s->n_conv; const int n_fft = s->n_fft; const float fft_scale = 1.0f / s->n_fft; FFTComplex *hrtf_offset; int wr = *write; int n_read; int i, j; if (!planar) 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[mult * j] = ringbuffer[wr]; ringbuffer[wr] = 0.0f; /* 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[mult * j] = 0; } /* fill FFT accumulation with 0 */ memset(fft_acc, 0, sizeof(FFTComplex) * n_fft); for (i = 0; i < n_conv; i++) { const float *src = (const float *)in->extended_data[i * planar]; /* get pointer to audio input buffer */ if (i == s->lfe_channel) { /* LFE */ if (in->format == AV_SAMPLE_FMT_FLT) { 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; } } else { for (j = 0; j < in->nb_samples; j++) { /* apply gain to LFE signal and add to output buffer */ dst[j] += src[j] * s->gain_lfe; } } continue; } /* outer loop: go through all input channels to be convolved */ offset = i * n_fft; /* no. samples already processed */ hrtf_offset = hrtf + offset; /* fill FFT input with 0 (we want to zero-pad) */ memset(fft_in, 0, sizeof(FFTComplex) * n_fft); if (in->format == AV_SAMPLE_FMT_FLT) { 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]; } } else { 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]; } } /* 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 FFTComplex *hcomplex = hrtf_offset + 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_acc[j].re += re * hcomplex->re - im * hcomplex->im; /* output channel (imag): */ fft_acc[j].im += re * hcomplex->im + im * hcomplex->re; } } /* transform output signal of current channel back to time domain */ av_fft_permute(ifft, fft_acc); av_fft_calc(ifft, fft_acc); for (j = 0; j < in->nb_samples; j++) { /* write output signal of current channel to output buffer */ dst[mult * j] += fft_acc[j].re * fft_scale; } 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_acc[in->nb_samples + j].re * fft_scale; } /* go through all samples of current output buffer: count clippings */ for (i = 0; i < out->nb_samples; i++) { /* clippings counter */ if (fabsf(dst[i * mult]) > 1) { /* if current output sample > 1 */ n_clippings[0]++; } } /* 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; td.temp_afft = s->temp_afft; 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 }; static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_NONE }; formats = ff_make_format_list(sample_fmts); if (!formats) return AVERROR(ENOMEM); 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 getfilter_float(AVFilterContext *ctx, float x, float y, float z, float *left, float *right, float *delay_left, float *delay_right) { struct SOFAlizerContext *s = ctx->priv; float c[3], delays[2]; float *fl, *fr; int nearest; int *neighbors; float *res; c[0] = x, c[1] = y, c[2] = z; nearest = mysofa_lookup(s->sofa.lookup, c); if (nearest < 0) return AVERROR(EINVAL); if (s->interpolate) { neighbors = mysofa_neighborhood(s->sofa.neighborhood, nearest); res = mysofa_interpolate(s->sofa.hrtf, c, nearest, neighbors, s->sofa.fir, delays); } else { if (s->sofa.hrtf->DataDelay.elements > s->sofa.hrtf->R) { delays[0] = s->sofa.hrtf->DataDelay.values[nearest * s->sofa.hrtf->R]; delays[1] = s->sofa.hrtf->DataDelay.values[nearest * s->sofa.hrtf->R + 1]; } else { delays[0] = s->sofa.hrtf->DataDelay.values[0]; delays[1] = s->sofa.hrtf->DataDelay.values[1]; } res = s->sofa.hrtf->DataIR.values + nearest * s->sofa.hrtf->N * s->sofa.hrtf->R; } *delay_left = delays[0]; *delay_right = delays[1]; fl = res; fr = res + s->sofa.hrtf->N; memcpy(left, fl, sizeof(float) * s->sofa.hrtf->N); memcpy(right, fr, sizeof(float) * s->sofa.hrtf->N); return 0; } static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int sample_rate) { struct SOFAlizerContext *s = ctx->priv; int n_samples; int ir_samples; int n_conv = s->n_conv; /* no. channels to convolve */ int n_fft; float delay_l; /* broadband delay for each IR */ float delay_r; 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 i, j, azim_orig = azim, elev_orig = elev; int ret = 0; int n_current; int n_max = 0; av_log(ctx, AV_LOG_DEBUG, "IR length: %d.\n", s->sofa.hrtf->N); s->sofa.ir_samples = s->sofa.hrtf->N; s->sofa.n_samples = 1 << (32 - ff_clz(s->sofa.ir_samples)); n_samples = s->sofa.n_samples; ir_samples = s->sofa.ir_samples; s->data_ir[0] = av_calloc(n_samples, sizeof(float) * s->n_conv); s->data_ir[1] = av_calloc(n_samples, sizeof(float) * s->n_conv); s->delay[0] = av_calloc(s->n_conv, sizeof(int)); s->delay[1] = av_calloc(s->n_conv, sizeof(int)); if (!s->data_ir[0] || !s->data_ir[1] || !s->delay[0] || !s->delay[1]) { ret = AVERROR(ENOMEM); goto fail; } /* get temporary IR for L and R channel */ data_ir_l = av_calloc(n_conv * n_samples, sizeof(*data_ir_l)); data_ir_r = av_calloc(n_conv * n_samples, sizeof(*data_ir_r)); if (!data_ir_r || !data_ir_l) { ret = AVERROR(ENOMEM); goto fail; } if (s->type == TIME_DOMAIN) { s->temp_src[0] = av_calloc(n_samples, sizeof(float)); s->temp_src[1] = av_calloc(n_samples, sizeof(float)); if (!s->temp_src[0] || !s->temp_src[1]) { ret = AVERROR(ENOMEM); goto fail; } } s->speaker_azim = av_calloc(s->n_conv, sizeof(*s->speaker_azim)); s->speaker_elev = av_calloc(s->n_conv, sizeof(*s->speaker_elev)); if (!s->speaker_azim || !s->speaker_elev) { ret = AVERROR(ENOMEM); goto fail; } /* 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"); goto fail; } for (i = 0; i < s->n_conv; i++) { float coordinates[3]; /* 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; coordinates[0] = azim; coordinates[1] = elev; coordinates[2] = radius; mysofa_s2c(coordinates); /* get id of IR closest to desired position */ ret = getfilter_float(ctx, coordinates[0], coordinates[1], coordinates[2], data_ir_l + n_samples * i, data_ir_r + n_samples * i, &delay_l, &delay_r); if (ret < 0) goto fail; s->delay[0][i] = delay_l * sample_rate; s->delay[1][i] = delay_r * sample_rate; s->sofa.max_delay = FFMAX3(s->sofa.max_delay, s->delay[0][i], s->delay[1][i]); } /* get size of ringbuffer (longest IR plus max. delay) */ /* then choose next power of 2 for performance optimization */ n_current = n_samples + s->sofa.max_delay; /* length of longest IR plus max. delay */ n_max = FFMAX(n_max, n_current); /* 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 = n_fft = 1 << (32 - ff_clz(n_max + s->framesize)); 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); ret = AVERROR(ENOMEM); goto fail; } } 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 { /* 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) { ret = AVERROR(ENOMEM); goto fail; } 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)); s->temp_afft[0] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); s->temp_afft[1] = av_malloc_array(s->n_fft, sizeof(FFTComplex)); if (!s->temp_fft[0] || !s->temp_fft[1] || !s->temp_afft[0] || !s->temp_afft[1]) { ret = AVERROR(ENOMEM); goto fail; } } if (!s->ringbuffer[0] || !s->ringbuffer[1]) { ret = AVERROR(ENOMEM); goto fail; } if (s->type == FREQUENCY_DOMAIN) { 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) { ret = AVERROR(ENOMEM); goto fail; } } for (i = 0; i < s->n_conv; i++) { float *lir, *rir; offset = i * n_samples; /* no. samples already written */ lir = data_ir_l + offset; rir = data_ir_r + offset; if (s->type == TIME_DOMAIN) { for (j = 0; j < ir_samples; j++) { /* load reversed IRs of the specified source position * sample-by-sample for left and right ear; and apply gain */ s->data_ir[0][offset + j] = lir[ir_samples - 1 - j] * gain_lin; s->data_ir[1][offset + j] = rir[ir_samples - 1 - j] * gain_lin; } } else { memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l)); memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r)); offset = i * n_fft; /* no. samples already written */ for (j = 0; j < ir_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[s->delay[0][i] + j].re = lir[j] * gain_lin; fft_in_r[s->delay[1][i] + j].re = rir[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)); } } if (s->type == FREQUENCY_DOMAIN) { 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]) { ret = AVERROR(ENOMEM); goto fail; } 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); } fail: av_freep(&data_hrtf_l); /* free temporary HRTF memory */ av_freep(&data_hrtf_r); av_freep(&data_ir_l); /* free temprary IR memory */ av_freep(&data_ir_r); av_freep(&fft_in_l); /* free temporary FFT memory */ av_freep(&fft_in_r); return ret; } 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); } /* preload SOFA file, */ ret = preload_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 ret; if (s->type == FREQUENCY_DOMAIN) { inlink->partial_buf_size = inlink->min_samples = inlink->max_samples = s->framesize; } /* gain -3 dB per channel */ s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10); s->n_conv = inlink->channels; /* load IRs to data_ir[0] and data_ir[1] for required directions */ if ((ret = load_data(ctx, s->rotation, s->elevation, s->radius, inlink->sample_rate)) < 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, inlink->channels, s->buffer_length); return 0; } static av_cold void uninit(AVFilterContext *ctx) { SOFAlizerContext *s = ctx->priv; close_sofa(&s->sofa); av_fft_end(s->ifft[0]); av_fft_end(s->ifft[1]); av_fft_end(s->fft[0]); av_fft_end(s->fft[1]); s->ifft[0] = NULL; s->ifft[1] = NULL; s->fft[0] = NULL; s->fft[1] = NULL; 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_afft[0]); av_freep(&s->temp_afft[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, 5, .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 }, { "lfegain", "set lfe gain", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20,40, .flags = FLAGS }, { "framesize", "set frame size", OFFSET(framesize), AV_OPT_TYPE_INT, {.i64=1024},1024,96000, .flags = FLAGS }, { "normalize", "normalize IRs", OFFSET(normalize), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, .flags = FLAGS }, { "interpolate","interpolate IRs from neighbors", OFFSET(interpolate),AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, .flags = FLAGS }, { "minphase", "minphase IRs", OFFSET(minphase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, .flags = FLAGS }, { "anglestep", "set neighbor search angle step", OFFSET(anglestep), AV_OPT_TYPE_FLOAT, {.dbl=.5}, 0.01, 10, .flags = FLAGS }, { "radstep", "set neighbor search radius step", OFFSET(radstep), AV_OPT_TYPE_FLOAT, {.dbl=.01}, 0.01, 1, .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, };