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FFmpeg/libavfilter/af_headphone.c
Andreas Rheinhardt 0952f8f909 avfilter/af_headphone: Fix channel assignment 
The documentation of the map argument of the headphone filter states:

"Set mapping of input streams for convolution. The argument is a
’|’-separated list of channel names in order as they are given as
additional stream inputs for filter."

Yet this has not been honoured at all. Instead for the kth given HRIR
channel pair it was checked whether there was a kth mapping and if the
channel position so given was present in the channel layout of the main
input; if so, then the given HRIR channel pair was matched to the kth
channel of the main input. It should actually have been matched to the
channel given by the kth mapping. A consequence of the current algorithm
is that if N additional HRIR channel pairs are given, a permutation of
the first N entries of the mapping does not affect the output at all.

The old code might even set arrays belonging to streams that don't exist
(i.e. whose index is >= the number of channels of the main input
stream); these parts were not read lateron at all. The new code doesn't
do this any longer and therefore the number of elements of some of the
allocated arrays has been reduced (in case the number of mappings was
bigger than the number of channels of the first input stream).

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
2020-09-09 13:47:24 +02:00

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/*
* Copyright (C) 2017 Paul B Mahol
* Copyright (C) 2013-2015 Andreas Fuchs, Wolfgang Hrauda
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <math.h>
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavutil/intmath.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"
#include "audio.h"
#define TIME_DOMAIN 0
#define FREQUENCY_DOMAIN 1
#define HRIR_STEREO 0
#define HRIR_MULTI 1
typedef struct HeadphoneContext {
const AVClass *class;
char *map;
int type;
int lfe_channel;
int have_hrirs;
int eof_hrirs;
int ir_len;
int air_len;
int nb_inputs;
int nb_irs;
float gain;
float lfe_gain, gain_lfe;
float *ringbuffer[2];
int write[2];
int buffer_length;
int n_fft;
int size;
int hrir_fmt;
float *data_ir[2];
float *temp_src[2];
FFTComplex *temp_fft[2];
FFTComplex *temp_afft[2];
FFTContext *fft[2], *ifft[2];
FFTComplex *data_hrtf[2];
AVFloatDSPContext *fdsp;
struct headphone_inputs {
AVFrame *frame;
int ir_len;
int eof;
} *in;
uint64_t mapping[64];
} HeadphoneContext;
static int parse_channel_name(const char *arg, uint64_t *rchannel)
{
uint64_t layout = av_get_channel_layout(arg);
if (av_get_channel_layout_nb_channels(layout) != 1)
return AVERROR(EINVAL);
*rchannel = layout;
return 0;
}
static void parse_map(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
char *arg, *tokenizer, *p;
uint64_t used_channels = 0;
s->lfe_channel = -1;
s->nb_inputs = 1;
p = s->map;
while ((arg = av_strtok(p, "|", &tokenizer))) {
uint64_t out_channel;
p = NULL;
if (parse_channel_name(arg, &out_channel)) {
av_log(ctx, AV_LOG_WARNING, "Failed to parse \'%s\' as channel name.\n", arg);
continue;
}
if (used_channels & out_channel) {
av_log(ctx, AV_LOG_WARNING, "Ignoring duplicate channel '%s'.\n", arg);
continue;
}
used_channels |= out_channel;
s->mapping[s->nb_irs] = out_channel;
s->nb_irs++;
}
if (s->hrir_fmt == HRIR_MULTI)
s->nb_inputs = 2;
else
s->nb_inputs = s->nb_irs + 1;
}
typedef struct ThreadData {
AVFrame *in, *out;
int *write;
float **ir;
int *n_clippings;
float **ringbuffer;
float **temp_src;
FFTComplex **temp_fft;
FFTComplex **temp_afft;
} ThreadData;
static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
HeadphoneContext *s = ctx->priv;
ThreadData *td = arg;
AVFrame *in = td->in, *out = td->out;
int offset = jobnr;
int *write = &td->write[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_len = s->ir_len;
const int air_len = s->air_len;
const float *src = (const float *)in->data[0];
float *dst = (float *)out->data[0];
const int in_channels = in->channels;
const int buffer_length = s->buffer_length;
const uint32_t modulo = (uint32_t)buffer_length - 1;
float *buffer[64];
int wr = *write;
int read;
int i, l;
dst += offset;
for (l = 0; l < in_channels; l++) {
buffer[l] = ringbuffer + l * buffer_length;
}
for (i = 0; i < in->nb_samples; i++) {
const float *temp_ir = ir;
*dst = 0;
for (l = 0; l < in_channels; l++) {
*(buffer[l] + wr) = src[l];
}
for (l = 0; l < in_channels; l++) {
const float *const bptr = buffer[l];
if (l == s->lfe_channel) {
*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
temp_ir += air_len;
continue;
}
read = (wr - (ir_len - 1) + buffer_length) & modulo;
if (read + ir_len < buffer_length) {
memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
} else {
int len = FFMIN(air_len - (read % ir_len), buffer_length - read);
memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
memcpy(temp_src + len, bptr, (air_len - len) * sizeof(*temp_src));
}
dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, FFALIGN(ir_len, 32));
temp_ir += air_len;
}
if (fabsf(dst[0]) > 1)
n_clippings[0]++;
dst += 2;
src += in_channels;
wr = (wr + 1) & modulo;
}
*write = wr;
return 0;
}
static int headphone_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
HeadphoneContext *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];
int *n_clippings = &td->n_clippings[jobnr];
float *ringbuffer = td->ringbuffer[jobnr];
const int ir_len = s->ir_len;
const float *src = (const float *)in->data[0];
float *dst = (float *)out->data[0];
const int in_channels = in->channels;
const int buffer_length = s->buffer_length;
const uint32_t modulo = (uint32_t)buffer_length - 1;
FFTComplex *fft_in = s->temp_fft[jobnr];
FFTComplex *fft_acc = s->temp_afft[jobnr];
FFTContext *ifft = s->ifft[jobnr];
FFTContext *fft = s->fft[jobnr];
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;
n_read = FFMIN(ir_len, in->nb_samples);
for (j = 0; j < n_read; j++) {
dst[2 * j] = ringbuffer[wr];
ringbuffer[wr] = 0.0;
wr = (wr + 1) & modulo;
}
for (j = n_read; j < in->nb_samples; j++) {
dst[2 * j] = 0;
}
memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);
for (i = 0; i < in_channels; i++) {
if (i == s->lfe_channel) {
for (j = 0; j < in->nb_samples; j++) {
dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
}
continue;
}
offset = i * n_fft;
hrtf_offset = hrtf + offset;
memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
for (j = 0; j < in->nb_samples; j++) {
fft_in[j].re = src[j * in_channels + i];
}
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;
fft_acc[j].re += re * hcomplex->re - im * hcomplex->im;
fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
}
}
av_fft_permute(ifft, fft_acc);
av_fft_calc(ifft, fft_acc);
for (j = 0; j < in->nb_samples; j++) {
dst[2 * j] += fft_acc[j].re * fft_scale;
}
for (j = 0; j < ir_len - 1; j++) {
int write_pos = (wr + j) & modulo;
*(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re * fft_scale;
}
for (i = 0; i < out->nb_samples; i++) {
if (fabsf(dst[0]) > 1) {
n_clippings[0]++;
}
dst += 2;
}
*write = wr;
return 0;
}
static int check_ir(AVFilterLink *inlink, int input_number)
{
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
int ir_len, max_ir_len;
ir_len = ff_inlink_queued_samples(inlink);
max_ir_len = 65536;
if (ir_len > max_ir_len) {
av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len);
return AVERROR(EINVAL);
}
s->in[input_number].ir_len = ir_len;
s->ir_len = FFMAX(ir_len, s->ir_len);
return 0;
}
static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
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);
}
out->pts = in->pts;
td.in = in; td.out = out; td.write = s->write;
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, headphone_convolute, &td, NULL, 2);
} else {
ctx->internal->execute(ctx, headphone_fast_convolute, &td, NULL, 2);
}
emms_c();
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 convert_coeffs(AVFilterContext *ctx, AVFilterLink *inlink)
{
struct HeadphoneContext *s = ctx->priv;
const int ir_len = s->ir_len;
int nb_input_channels = ctx->inputs[0]->channels;
float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
FFTComplex *fft_in_l = NULL;
FFTComplex *fft_in_r = NULL;
int offset = 0, ret = 0;
int n_fft;
int i, j, k;
s->air_len = 1 << (32 - ff_clz(ir_len));
if (s->type == TIME_DOMAIN) {
s->air_len = FFALIGN(s->air_len, 32);
}
s->buffer_length = 1 << (32 - ff_clz(s->air_len));
s->n_fft = n_fft = 1 << (32 - ff_clz(ir_len + s->size));
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;
}
s->fft[0] = av_fft_init(av_log2(s->n_fft), 0);
s->fft[1] = av_fft_init(av_log2(s->n_fft), 0);
s->ifft[0] = av_fft_init(av_log2(s->n_fft), 1);
s->ifft[1] = av_fft_init(av_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 {
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_calloc(s->n_fft, sizeof(FFTComplex));
s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
s->temp_afft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
s->temp_afft[1] = av_calloc(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 == TIME_DOMAIN) {
s->temp_src[0] = av_calloc(s->air_len, sizeof(float));
s->temp_src[1] = av_calloc(s->air_len, sizeof(float));
s->data_ir[0] = av_calloc(nb_input_channels * s->air_len, sizeof(*s->data_ir[0]));
s->data_ir[1] = av_calloc(nb_input_channels * s->air_len, sizeof(*s->data_ir[1]));
if (!s->data_ir[0] || !s->data_ir[1] || !s->temp_src[0] || !s->temp_src[1]) {
ret = AVERROR(ENOMEM);
goto fail;
}
} else {
s->data_hrtf[0] = av_calloc(n_fft, sizeof(*s->data_hrtf[0]) * nb_input_channels);
s->data_hrtf[1] = av_calloc(n_fft, sizeof(*s->data_hrtf[1]) * nb_input_channels);
if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
ret = AVERROR(ENOMEM);
goto fail;
}
}
for (i = 0; i < s->nb_inputs - 1; i++) {
int len = s->in[i + 1].ir_len;
float *ptr;
ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &s->in[i + 1].frame);
if (ret < 0)
goto fail;
ptr = (float *)s->in[i + 1].frame->extended_data[0];
if (s->hrir_fmt == HRIR_STEREO) {
int idx = -1;
for (j = 0; j < inlink->channels; j++) {
if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == s->mapping[i]) {
idx = j;
if (s->mapping[i] == AV_CH_LOW_FREQUENCY)
s->lfe_channel = j;
break;
}
}
if (idx == -1)
continue;
if (s->type == TIME_DOMAIN) {
float *data_ir_l = s->data_ir[0] + idx * s->air_len;
float *data_ir_r = s->data_ir[1] + idx * s->air_len;
for (j = 0; j < len; j++) {
data_ir_l[j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
data_ir_r[j] = ptr[len * 2 - j * 2 - 1] * 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 = idx * n_fft;
for (j = 0; j < len; j++) {
fft_in_l[j].re = ptr[j * 2 ] * gain_lin;
fft_in_r[j].re = ptr[j * 2 + 1] * gain_lin;
}
av_fft_permute(s->fft[0], fft_in_l);
av_fft_calc(s->fft[0], fft_in_l);
memcpy(s->data_hrtf[0] + 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(s->data_hrtf[1] + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
}
} else {
int I, N = ctx->inputs[1]->channels;
for (k = 0; k < N / 2; k++) {
int idx = -1;
for (j = 0; j < inlink->channels; j++) {
if ((av_channel_layout_extract_channel(inlink->channel_layout, j)) == s->mapping[k]) {
idx = j;
if (s->mapping[k] == AV_CH_LOW_FREQUENCY)
s->lfe_channel = j;
break;
}
}
if (idx == -1)
continue;
I = k * 2;
if (s->type == TIME_DOMAIN) {
float *data_ir_l = s->data_ir[0] + idx * s->air_len;
float *data_ir_r = s->data_ir[1] + idx * s->air_len;
for (j = 0; j < len; j++) {
data_ir_l[j] = ptr[len * N - j * N - N + I ] * gain_lin;
data_ir_r[j] = ptr[len * N - j * N - N + I + 1] * 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 = idx * n_fft;
for (j = 0; j < len; j++) {
fft_in_l[j].re = ptr[j * N + I ] * gain_lin;
fft_in_r[j].re = ptr[j * N + I + 1] * gain_lin;
}
av_fft_permute(s->fft[0], fft_in_l);
av_fft_calc(s->fft[0], fft_in_l);
memcpy(s->data_hrtf[0] + 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(s->data_hrtf[1] + offset, fft_in_r, n_fft * sizeof(*fft_in_r));
}
}
}
av_frame_free(&s->in[i + 1].frame);
}
s->have_hrirs = 1;
fail:
for (i = 0; i < s->nb_inputs - 1; i++)
av_frame_free(&s->in[i + 1].frame);
av_freep(&fft_in_l);
av_freep(&fft_in_r);
return ret;
}
static int activate(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *in = NULL;
int i, ret;
FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
if (!s->eof_hrirs) {
int eof = 1;
for (i = 1; i < s->nb_inputs; i++) {
if (s->in[i].eof)
continue;
if ((ret = check_ir(ctx->inputs[i], i)) < 0)
return ret;
if (ff_outlink_get_status(ctx->inputs[i]) == AVERROR_EOF) {
if (!ff_inlink_queued_samples(ctx->inputs[i])) {
av_log(ctx, AV_LOG_ERROR, "No samples provided for "
"HRIR stream %d.\n", i - 1);
return AVERROR_INVALIDDATA;
}
s->in[i].eof = 1;
} else {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
ff_inlink_request_frame(ctx->inputs[i]);
eof = 0;
}
}
if (!eof)
return 0;
s->eof_hrirs = 1;
ret = convert_coeffs(ctx, inlink);
if (ret < 0)
return ret;
} else if (!s->have_hrirs)
return AVERROR_EOF;
if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
ret = headphone_frame(s, in, outlink);
if (ret < 0)
return ret;
}
if (ret < 0)
return ret;
FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
if (ff_outlink_frame_wanted(ctx->outputs[0]))
ff_inlink_request_frame(ctx->inputs[0]);
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
struct HeadphoneContext *s = ctx->priv;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
AVFilterChannelLayouts *stereo_layout = NULL;
AVFilterChannelLayouts *hrir_layouts = NULL;
int ret, i;
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]->outcfg.channel_layouts);
if (ret)
return ret;
ret = ff_add_channel_layout(&stereo_layout, AV_CH_LAYOUT_STEREO);
if (ret)
return ret;
ret = ff_channel_layouts_ref(stereo_layout, &ctx->outputs[0]->incfg.channel_layouts);
if (ret)
return ret;
if (s->hrir_fmt == HRIR_MULTI) {
hrir_layouts = ff_all_channel_counts();
if (!hrir_layouts)
return AVERROR(ENOMEM);
ret = ff_channel_layouts_ref(hrir_layouts, &ctx->inputs[1]->outcfg.channel_layouts);
if (ret)
return ret;
} else {
for (i = 1; i < s->nb_inputs; i++) {
ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->outcfg.channel_layouts);
if (ret)
return ret;
}
}
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
if (s->nb_irs < inlink->channels) {
av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->channels);
return AVERROR(EINVAL);
}
return 0;
}
static av_cold int init(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
int i, ret;
AVFilterPad pad = {
.name = "in0",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
};
if ((ret = ff_insert_inpad(ctx, 0, &pad)) < 0)
return ret;
if (!s->map) {
av_log(ctx, AV_LOG_ERROR, "Valid mapping must be set.\n");
return AVERROR(EINVAL);
}
parse_map(ctx);
s->in = av_calloc(s->nb_inputs, sizeof(*s->in));
if (!s->in)
return AVERROR(ENOMEM);
for (i = 1; i < s->nb_inputs; i++) {
char *name = av_asprintf("hrir%d", i - 1);
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
};
if (!name)
return AVERROR(ENOMEM);
if ((ret = ff_insert_inpad(ctx, i, &pad)) < 0) {
av_freep(&pad.name);
return ret;
}
}
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
HeadphoneContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
if (s->hrir_fmt == HRIR_MULTI) {
AVFilterLink *hrir_link = ctx->inputs[1];
if (hrir_link->channels < inlink->channels * 2) {
av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->channels * 2);
return AVERROR(EINVAL);
}
}
s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
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->data_ir[0]);
av_freep(&s->data_ir[1]);
av_freep(&s->ringbuffer[0]);
av_freep(&s->ringbuffer[1]);
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->temp_afft[0]);
av_freep(&s->temp_afft[1]);
av_freep(&s->data_hrtf[0]);
av_freep(&s->data_hrtf[1]);
av_freep(&s->fdsp);
av_freep(&s->in);
for (unsigned i = 1; i < ctx->nb_inputs; i++)
av_freep(&ctx->input_pads[i].name);
}
#define OFFSET(x) offsetof(HeadphoneContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption headphone_options[] = {
{ "map", "set channels convolution mappings", OFFSET(map), 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 },
{ "lfe", "set lfe gain in dB", OFFSET(lfe_gain), AV_OPT_TYPE_FLOAT, {.dbl=0}, -20, 40, .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" },
{ "size", "set frame size", OFFSET(size), AV_OPT_TYPE_INT, {.i64=1024},1024,96000, .flags = FLAGS },
{ "hrir", "set hrir format", OFFSET(hrir_fmt), AV_OPT_TYPE_INT, {.i64=HRIR_STEREO}, 0, 1, .flags = FLAGS, "hrir" },
{ "stereo", "hrir files have exactly 2 channels", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_STEREO}, 0, 0, .flags = FLAGS, "hrir" },
{ "multich", "single multichannel hrir file", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_MULTI}, 0, 0, .flags = FLAGS, "hrir" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(headphone);
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
{ NULL }
};
AVFilter ff_af_headphone = {
.name = "headphone",
.description = NULL_IF_CONFIG_SMALL("Apply headphone binaural spatialization with HRTFs in additional streams."),
.priv_size = sizeof(HeadphoneContext),
.priv_class = &headphone_class,
.init = init,
.uninit = uninit,
.query_formats = query_formats,
.activate = activate,
.inputs = NULL,
.outputs = outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
};
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