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afceab7a0a
jellyfin-ffmpeg/libavfilter/af_headphone.c
nyanmisaka afceab7a0a New upstream version 5.1
2022-07-23 10:30:09 +08:00

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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 "libavutil/tx.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_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(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;
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(float));
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);
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);
}
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]->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;
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(float));
s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(float));
}
} 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(float));
s->tx_fn[0](s->fft[0], fft_out_r, fft_in_r, sizeof(float));
}
}
}
}
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(ctx->outputs[0], 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 (ff_outlink_get_status(input) == AVERROR_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;
}
s->hrir_in[i].eof = 1;
} else {
if (ff_outlink_frame_wanted(ctx->outputs[0]))
ff_inlink_request_frame(input);
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, &(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, "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,
},
};
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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