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FFmpeg/libavfilter/af_headphone.c
Andreas Rheinhardt 790f793844 avutil/common: Don't auto-include mem.h 
There are lots of files that don't need it: The number of object
files that actually need it went down from 2011 to 884 here.

Keep it for external users in order to not cause breakages.

Also improve the other headers a bit while just at it.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2024-03-31 00:08:43 +01:00

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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/mem.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "avfilter.h"
#include "filters.h"
#include "formats.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_hrir_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];
AVComplexFloat *out_fft[2];
AVComplexFloat *in_fft[2];
AVComplexFloat *temp_afft[2];
AVTXContext *fft[2], *ifft[2];
av_tx_fn tx_fn[2], itx_fn[2];
AVComplexFloat *data_hrtf[2];
float (*scalarproduct_float)(const float *v1, const float *v2, int len);
struct hrir_inputs {
int ir_len;
int eof;
} hrir_in[64];
AVChannelLayout map_channel_layout;
enum AVChannel mapping[64];
uint8_t hrir_map[64];
} HeadphoneContext;
static int parse_channel_name(const char *arg, enum AVChannel *rchannel)
{
int channel = av_channel_from_string(arg);
if (channel < 0 || channel >= 64)
return AVERROR(EINVAL);
*rchannel = channel;
return 0;
}
static void parse_map(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
char *arg, *tokenizer, *p;
uint64_t used_channels = 0;
p = s->map;
while ((arg = av_strtok(p, "|", &tokenizer))) {
enum AVChannel 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 & (1ULL << out_channel)) {
av_log(ctx, AV_LOG_WARNING, "Ignoring duplicate channel '%s'.\n", arg);
continue;
}
used_channels |= (1ULL << out_channel);
s->mapping[s->nb_irs] = out_channel;
s->nb_irs++;
}
av_channel_layout_from_mask(&s->map_channel_layout, used_channels);
if (s->hrir_fmt == HRIR_MULTI)
s->nb_hrir_inputs = 1;
else
s->nb_hrir_inputs = s->nb_irs;
}
typedef struct ThreadData {
AVFrame *in, *out;
int *write;
float **ir;
int *n_clippings;
float **ringbuffer;
float **temp_src;
AVComplexFloat **out_fft;
AVComplexFloat **in_fft;
AVComplexFloat **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->ch_layout.nb_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 *cur_ir = ir;
*dst = 0;
for (l = 0; l < in_channels; l++) {
*(buffer[l] + wr) = src[l];
}
for (l = 0; l < in_channels; cur_ir += air_len, l++) {
const float *const bptr = buffer[l];
if (l == s->lfe_channel) {
*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
continue;
}
read = (wr - (ir_len - 1)) & 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->scalarproduct_float(cur_ir, temp_src, FFALIGN(ir_len, 32));
}
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];
AVComplexFloat *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->ch_layout.nb_channels;
const int buffer_length = s->buffer_length;
const uint32_t modulo = (uint32_t)buffer_length - 1;
AVComplexFloat *fft_out = s->out_fft[jobnr];
AVComplexFloat *fft_in = s->in_fft[jobnr];
AVComplexFloat *fft_acc = s->temp_afft[jobnr];
AVTXContext *ifft = s->ifft[jobnr];
AVTXContext *fft = s->fft[jobnr];
av_tx_fn tx_fn = s->tx_fn[jobnr];
av_tx_fn itx_fn = s->itx_fn[jobnr];
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;
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(AVComplexFloat) * 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 + s->hrir_map[i] * n_fft;
memset(fft_in, 0, sizeof(AVComplexFloat) * n_fft);
for (j = 0; j < in->nb_samples; j++) {
fft_in[j].re = src[j * in_channels + i];
}
tx_fn(fft, fft_out, fft_in, sizeof(*fft_in));
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;
fft_acc[j].re += re * hcomplex->re - im * hcomplex->im;
fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
}
}
itx_fn(ifft, fft_out, fft_acc, sizeof(*fft_acc));
for (j = 0; j < in->nb_samples; j++) {
dst[2 * j] += fft_out[j].re * fft_scale;
if (fabsf(dst[2 * j]) > 1)
n_clippings[0]++;
}
for (j = 0; j < ir_len - 1; j++) {
int write_pos = (wr + j) & modulo;
*(ringbuffer + write_pos) += fft_out[in->nb_samples + j].re * fft_scale;
}
*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->hrir_in[input_number].ir_len = ir_len;
s->ir_len = FFMAX(ir_len, s->ir_len);
if (ff_inlink_check_available_samples(inlink, ir_len + 1) == 1) {
s->hrir_in[input_number].eof = 1;
return 1;
}
if (!s->hrir_in[input_number].eof) {
ff_inlink_request_frame(inlink);
return 0;
}
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.out_fft = s->out_fft;
td.in_fft = s->in_fft;
td.temp_afft = s->temp_afft;
if (s->type == TIME_DOMAIN) {
ff_filter_execute(ctx, headphone_convolute, &td, NULL, 2);
} else {
ff_filter_execute(ctx, headphone_fast_convolute, &td, NULL, 2);
}
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]->ch_layout.nb_channels;
const int nb_hrir_channels = s->nb_hrir_inputs == 1 ? ctx->inputs[1]->ch_layout.nb_channels : s->nb_hrir_inputs * 2;
float gain_lin = expf((s->gain - 3 * nb_input_channels) / 20 * M_LN10);
AVFrame *frame;
int 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) {
float scale = 1.f;
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;
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->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->out_fft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat));
s->out_fft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat));
s->in_fft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat));
s->in_fft[1] = av_calloc(s->n_fft, sizeof(AVComplexFloat));
s->temp_afft[0] = av_calloc(s->n_fft, sizeof(AVComplexFloat));
s->temp_afft[1] = av_calloc(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 == 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_hrir_channels * s->air_len, sizeof(*s->data_ir[0]));
s->data_ir[1] = av_calloc(nb_hrir_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_hrir_channels);
s->data_hrtf[1] = av_calloc(n_fft, sizeof(*s->data_hrtf[1]) * nb_hrir_channels);
if (!s->data_hrtf[0] || !s->data_hrtf[1]) {
ret = AVERROR(ENOMEM);
goto fail;
}
}
for (i = 0; i < s->nb_hrir_inputs; av_frame_free(&frame), i++) {
int len = s->hrir_in[i].ir_len;
float *ptr;
ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &frame);
if (ret < 0)
goto fail;
ptr = (float *)frame->extended_data[0];
if (s->hrir_fmt == HRIR_STEREO) {
int idx = av_channel_layout_index_from_channel(&s->map_channel_layout,
s->mapping[i]);
if (idx < 0)
continue;
s->hrir_map[i] = idx;
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 {
AVComplexFloat *fft_out_l = s->data_hrtf[0] + idx * n_fft;
AVComplexFloat *fft_out_r = s->data_hrtf[1] + idx * n_fft;
AVComplexFloat *fft_in_l = s->in_fft[0];
AVComplexFloat *fft_in_r = s->in_fft[1];
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;
}
s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(*fft_in_l));
s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(*fft_in_r));
}
} else {
int I, N = ctx->inputs[1]->ch_layout.nb_channels;
for (k = 0; k < N / 2; k++) {
int idx = av_channel_layout_index_from_channel(&inlink->ch_layout,
s->mapping[k]);
if (idx < 0)
continue;
s->hrir_map[k] = idx;
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 {
AVComplexFloat *fft_out_l = s->data_hrtf[0] + idx * n_fft;
AVComplexFloat *fft_out_r = s->data_hrtf[1] + idx * n_fft;
AVComplexFloat *fft_in_l = s->in_fft[0];
AVComplexFloat *fft_in_r = s->in_fft[1];
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;
}
s->tx_fn[0](s->fft[0], fft_out_l, fft_in_l, sizeof(*fft_in_l));
s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(*fft_in_r));
}
}
}
}
s->have_hrirs = 1;
fail:
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(outlink, ctx);
if (!s->eof_hrirs) {
int eof = 1;
for (i = 0; i < s->nb_hrir_inputs; i++) {
AVFilterLink *input = ctx->inputs[i + 1];
if (s->hrir_in[i].eof)
continue;
if ((ret = check_ir(input, i)) <= 0)
return ret;
if (s->hrir_in[i].eof) {
if (!ff_inlink_queued_samples(input)) {
av_log(ctx, AV_LOG_ERROR, "No samples provided for "
"HRIR stream %d.\n", i);
return AVERROR_INVALIDDATA;
}
} else {
eof = 0;
}
}
if (!eof) {
ff_filter_set_ready(ctx, 100);
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(inlink, 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(inlink, outlink);
if (ff_outlink_frame_wanted(outlink))
ff_inlink_request_frame(inlink);
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, &(AVChannelLayout)AV_CHANNEL_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_hrir_inputs; i++) {
ret = ff_channel_layouts_ref(stereo_layout, &ctx->inputs[i]->outcfg.channel_layouts);
if (ret)
return ret;
}
}
return ff_set_common_all_samplerates(ctx);
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
if (s->nb_irs < inlink->ch_layout.nb_channels) {
av_log(ctx, AV_LOG_ERROR, "Number of HRIRs must be >= %d.\n", inlink->ch_layout.nb_channels);
return AVERROR(EINVAL);
}
s->lfe_channel = av_channel_layout_index_from_channel(&inlink->ch_layout,
AV_CHAN_LOW_FREQUENCY);
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_append_inpad(ctx, &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);
for (i = 0; i < s->nb_hrir_inputs; i++) {
char *name = av_asprintf("hrir%d", i);
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
};
if (!name)
return AVERROR(ENOMEM);
if ((ret = ff_append_inpad_free_name(ctx, &pad)) < 0)
return ret;
}
if (s->type == TIME_DOMAIN) {
AVFloatDSPContext *fdsp = avpriv_float_dsp_alloc(0);
if (!fdsp)
return AVERROR(ENOMEM);
s->scalarproduct_float = fdsp->scalarproduct_float;
av_free(fdsp);
}
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->ch_layout.nb_channels < inlink->ch_layout.nb_channels * 2) {
av_log(ctx, AV_LOG_ERROR, "Number of channels in HRIR stream must be >= %d.\n", inlink->ch_layout.nb_channels * 2);
return AVERROR(EINVAL);
}
}
s->gain_lfe = expf((s->gain - 3 * inlink->ch_layout.nb_channels + s->lfe_gain) / 20 * M_LN10);
return 0;
}
static av_cold void uninit(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
av_tx_uninit(&s->ifft[0]);
av_tx_uninit(&s->ifft[1]);
av_tx_uninit(&s->fft[0]);
av_tx_uninit(&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->out_fft[0]);
av_freep(&s->out_fft[1]);
av_freep(&s->in_fft[0]);
av_freep(&s->in_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]);
}
#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, .unit = "type" },
{ "time", "time domain", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, .flags = FLAGS, .unit = "type" },
{ "freq", "frequency domain", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, .flags = FLAGS, .unit = "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, .unit = "hrir" },
{ "stereo", "hrir files have exactly 2 channels", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_STEREO}, 0, 0, .flags = FLAGS, .unit = "hrir" },
{ "multich", "single multichannel hrir file", 0, AV_OPT_TYPE_CONST, {.i64=HRIR_MULTI}, 0, 0, .flags = FLAGS, .unit = "hrir" },
{ NULL }
};
AVFILTER_DEFINE_CLASS(headphone);
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const 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,
.activate = activate,
.inputs = NULL,
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
};
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