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-rw-r--r--libavfilter/af_sofalizer.c914
1 files changed, 914 insertions, 0 deletions
diff --git a/libavfilter/af_sofalizer.c b/libavfilter/af_sofalizer.c
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+++ b/libavfilter/af_sofalizer.c
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+/*****************************************************************************
+ * 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 <mysofa.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 MySofa { /* contains data of one SOFA file */
+ struct MYSOFA_EASY *easy;
+ int n_samples; /* length of one impulse response (IR) */
+ float *lir, *rir; /* IRs (time-domain) */
+ 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];
+
+ /* 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 MySofa *sofa)
+{
+ mysofa_close(sofa->easy);
+ sofa->easy = NULL;
+
+ return 0;
+}
+
+static int preload_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ struct MYSOFA_HRTF *mysofa;
+ int ret;
+
+ mysofa = mysofa_load(filename, &ret);
+ if (ret || !mysofa) {
+ av_log(ctx, AV_LOG_ERROR, "Can't find SOFA-file '%s'\n", filename);
+ return AVERROR(EINVAL);
+ }
+
+ if (mysofa->DataSamplingRate.elements != 1)
+ return AVERROR(EINVAL);
+ *samplingrate = mysofa->DataSamplingRate.values[0];
+ s->sofa.n_samples = mysofa->N;
+ mysofa_free(mysofa);
+
+ 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 (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 (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;
+
+}
+
+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] = 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, 32);
+ 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, n_samples);
+ temp_ir += FFALIGN(n_samples, 32);
+ }
+
+ /* clippings counter */
+ if (fabs(dst[0]) > 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;
+ const float fft_scale = 1.0f / s->n_fft;
+ FFTComplex *hrtf_offset;
+ 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 */
+ hrtf_offset = hrtf + offset;
+
+ /* 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 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_in[j].re = re * hcomplex->re - im * hcomplex->im;
+ /* output channel (imag): */
+ fft_in[j].im = re * hcomplex->im + im * hcomplex->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 * 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_in[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 (fabs(*dst) > 1) { /* if current output sample > 1 */
+ n_clippings[0]++;
+ }
+
+ /* 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, int sample_rate)
+{
+ struct SOFAlizerContext *s = ctx->priv;
+ int n_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 filter_length, ret = 0;
+ int n_current;
+ int n_max = 0;
+
+ s->sofa.easy = mysofa_open(s->filename, sample_rate, &filter_length, &ret);
+ if (!s->sofa.easy || ret) { /* 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;
+ }
+
+ n_samples = s->sofa.n_samples;
+
+ s->data_ir[0] = av_calloc(FFALIGN(n_samples, 32), sizeof(float) * s->n_conv);
+ s->data_ir[1] = av_calloc(FFALIGN(n_samples, 32), 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 * FFALIGN(n_samples, 32), sizeof(*data_ir_l));
+ data_ir_r = av_calloc(n_conv * FFALIGN(n_samples, 32), 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(FFALIGN(n_samples, 32), sizeof(float));
+ s->temp_src[1] = av_calloc(FFALIGN(n_samples, 32), 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 */
+ mysofa_getfilter_float(s->sofa.easy, coordinates[0], coordinates[1], coordinates[2],
+ data_ir_l + FFALIGN(n_samples, 32) * i,
+ data_ir_r + FFALIGN(n_samples, 32) * i,
+ &delay_l, &delay_r);
+
+ 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 = s->sofa.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 + 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);
+ 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));
+ if (!s->temp_fft[0] || !s->temp_fft[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 * FFALIGN(n_samples, 32); /* no. samples already written */
+
+ lir = data_ir_l + offset;
+ rir = data_ir_r + offset;
+
+ if (s->type == TIME_DOMAIN) {
+ 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 */
+ s->data_ir[0][offset + j] = lir[n_samples - 1 - j] * gain_lin;
+ s->data_ir[1][offset + j] = rir[n_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 < 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[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 = 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 + 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]);
+ 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 },
+ { "lfegain", "set lfe gain", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -9, 9, .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,
+};
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-21 06:22:25 +0000