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FFmpeg/libavfilter/af_sofalizer.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +02:00

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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 "libavutil/tx.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 "filters.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 */
int nb_samples;
/* 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];
AVComplexFloat *in_fft[2]; /* Array to hold input FFT values */
AVComplexFloat *out_fft[2]; /* Array to hold output FFT values */
AVComplexFloat *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];
AVTXContext *fft[2], *ifft[2];
av_tx_fn tx_fn[2], itx_fn[2];
AVComplexFloat *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(AVFilterContext *ctx, char **arg, int *rchannel)
{
int len, i, channel_id = 0;
int64_t layout, layout0;
char buf[8] = {0};
/* 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) {
av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", buf);
return AVERROR(EINVAL);
}
*rchannel = channel_id;
*arg += len;
return 0;
} else if (av_sscanf(*arg, "%d%n", &channel_id, &len) == 1) {
if (channel_id < 0 || channel_id >= 64) {
av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%d\' as channel number.\n", channel_id);
return AVERROR(EINVAL);
}
*rchannel = channel_id;
*arg += len;
return 0;
}
return AVERROR(EINVAL);
}
static void parse_speaker_pos(AVFilterContext *ctx, int64_t in_channel_layout)
{
SOFAlizerContext *s = ctx->priv;
char *arg, *tokenizer, *p, *args = av_strdup(s->speakers_pos);
if (!args)
return;
p = args;
while ((arg = av_strtok(p, "|", &tokenizer))) {
float azim, elev;
int out_ch_id;
p = NULL;
if (parse_channel_name(ctx, &arg, &out_ch_id)) {
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[64] = { 0 };
float elev[64] = { 0 };
int m, ch, n_conv = ctx->inputs[0]->channels; /* get no. input channels */
if (n_conv < 0 || n_conv > 64)
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;
AVComplexFloat **in_fft;
AVComplexFloat **out_fft;
AVComplexFloat **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[64]; /* 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] - (ir_samples - 1) + buffer_length) & modulo;
if (read + ir_samples < buffer_length) {
memmove(temp_src, bptr + read, ir_samples * sizeof(*temp_src));
} else {
int len = FFMIN(n_samples - (read % ir_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];
AVComplexFloat *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 ir_samples = s->sofa.ir_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;
AVComplexFloat *fft_in = s->in_fft[jobnr]; /* temporary array for FFT input data */
AVComplexFloat *fft_out = s->out_fft[jobnr]; /* temporary array for FFT output data */
AVComplexFloat *fft_acc = s->temp_afft[jobnr];
AVTXContext *ifft = s->ifft[jobnr];
av_tx_fn itx_fn = s->itx_fn[jobnr];
AVTXContext *fft = s->fft[jobnr];
av_tx_fn tx_fn = s->tx_fn[jobnr];
const int n_conv = s->n_conv;
const int n_fft = s->n_fft;
const float fft_scale = 1.0f / s->n_fft;
AVComplexFloat *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(ir_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(AVComplexFloat) * 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(AVComplexFloat) * 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 */
tx_fn(fft, fft_out, fft_in, sizeof(float));
for (j = 0; j < n_fft; j++) {
const AVComplexFloat *hcomplex = hrtf_offset + j;
const float re = fft_out[j].re;
const float im = fft_out[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 */
itx_fn(ifft, fft_out, fft_acc, sizeof(float));
for (j = 0; j < in->nb_samples; j++) {
/* write output signal of current channel to output buffer */
dst[mult * j] += fft_out[j].re * fft_scale;
}
for (j = 0; j < ir_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_out[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.in_fft = s->in_fft;
td.out_fft = s->out_fft;
td.temp_afft = s->temp_afft;
if (s->type == TIME_DOMAIN) {
ff_filter_execute(ctx, sofalizer_convolute, &td, NULL, 2);
} else if (s->type == FREQUENCY_DOMAIN) {
ff_filter_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 activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
SOFAlizerContext *s = ctx->priv;
AVFrame *in;
int ret;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (s->nb_samples)
ret = ff_inlink_consume_samples(inlink, s->nb_samples, s->nb_samples, &in);
else
ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
if (ret > 0)
return filter_frame(inlink, in);
FF_FILTER_FORWARD_STATUS(inlink, outlink);
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static int query_formats(AVFilterContext *ctx)
{
struct SOFAlizerContext *s = ctx->priv;
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
};
ret = ff_set_common_formats_from_list(ctx, sample_fmts);
if (ret)
return ret;
layouts = ff_all_channel_layouts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.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]->incfg.channel_layouts);
if (ret)
return ret;
sample_rates[0] = s->sample_rate;
return ff_set_common_samplerates_from_list(ctx, sample_rates);
}
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 */
AVComplexFloat *data_hrtf_l = NULL;
AVComplexFloat *data_hrtf_r = NULL;
AVComplexFloat *fft_out_l = NULL;
AVComplexFloat *fft_out_r = NULL;
AVComplexFloat *fft_in_l = NULL;
AVComplexFloat *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;
if (s->type == TIME_DOMAIN) {
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);
if (!s->data_ir[0] || !s->data_ir[1]) {
ret = AVERROR(ENOMEM);
goto fail;
}
}
s->delay[0] = av_calloc(s->n_conv, sizeof(int));
s->delay[1] = av_calloc(s->n_conv, sizeof(int));
if (!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) {
float scale;
av_tx_uninit(&s->fft[0]);
av_tx_uninit(&s->fft[1]);
ret = av_tx_init(&s->fft[0], &s->tx_fn[0], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0);
if (ret < 0)
goto fail;
ret = av_tx_init(&s->fft[1], &s->tx_fn[1], AV_TX_FLOAT_FFT, 0, s->n_fft, &scale, 0);
if (ret < 0)
goto fail;
av_tx_uninit(&s->ifft[0]);
av_tx_uninit(&s->ifft[1]);
ret = av_tx_init(&s->ifft[0], &s->itx_fn[0], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0);
if (ret < 0)
goto fail;
ret = av_tx_init(&s->ifft[1], &s->itx_fn[1], AV_TX_FLOAT_FFT, 1, s->n_fft, &scale, 0);
if (ret < 0)
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 if (s->type == FREQUENCY_DOMAIN) {
/* 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->in_fft[0] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
s->in_fft[1] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
s->out_fft[0] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
s->out_fft[1] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
s->temp_afft[0] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
s->temp_afft[1] = av_malloc_array(s->n_fft, sizeof(AVComplexFloat));
if (!s->in_fft[0] || !s->in_fft[1] ||
!s->out_fft[0] || !s->out_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_out_l = av_calloc(n_fft, sizeof(*fft_out_l));
fft_out_r = av_calloc(n_fft, sizeof(*fft_out_r));
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 ||
!fft_out_l || !fft_out_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 if (s->type == FREQUENCY_DOMAIN) {
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 are 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) */
s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(float));
memcpy(data_hrtf_l + offset, fft_out_l, n_fft * sizeof(*fft_out_l));
s->tx_fn[1](s->fft[1], fft_out_r, fft_in_r, sizeof(float));
memcpy(data_hrtf_r + offset, fft_out_r, n_fft * sizeof(*fft_out_r));
}
}
if (s->type == FREQUENCY_DOMAIN) {
s->data_hrtf[0] = av_malloc_array(n_fft * s->n_conv, sizeof(AVComplexFloat));
s->data_hrtf[1] = av_malloc_array(n_fft * s->n_conv, sizeof(AVComplexFloat));
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(AVComplexFloat) * n_conv * n_fft); /* filter struct */
memcpy(s->data_hrtf[1], data_hrtf_r,
sizeof(AVComplexFloat) * 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_out_l); /* free temporary FFT memory */
av_freep(&fft_out_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)
s->nb_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_tx_uninit(&s->ifft[0]);
av_tx_uninit(&s->ifft[1]);
av_tx_uninit(&s->fft[0]);
av_tx_uninit(&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->in_fft[0]);
av_freep(&s->in_fft[1]);
av_freep(&s->out_fft[0]);
av_freep(&s->out_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,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
};
const 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,
.activate = activate,
.uninit = uninit,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SLICE_THREADS,
};
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