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FFmpeg/libavfilter/af_surround.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
*
* 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 "libavutil/avassert.h"
#include "libavutil/channel_layout.h"
#include "libavutil/mem.h"
#include "libavutil/opt.h"
#include "libavutil/tx.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "internal.h"
#include "formats.h"
#include "window_func.h"
enum SurroundChannel {
SC_FL, SC_FR, SC_FC, SC_LF, SC_BL, SC_BR, SC_BC, SC_SL, SC_SR,
SC_NB,
};
static const int ch_map[SC_NB] = {
[SC_FL] = AV_CHAN_FRONT_LEFT,
[SC_FR] = AV_CHAN_FRONT_RIGHT,
[SC_FC] = AV_CHAN_FRONT_CENTER,
[SC_LF] = AV_CHAN_LOW_FREQUENCY,
[SC_BL] = AV_CHAN_BACK_LEFT,
[SC_BR] = AV_CHAN_BACK_RIGHT,
[SC_BC] = AV_CHAN_BACK_CENTER,
[SC_SL] = AV_CHAN_SIDE_LEFT,
[SC_SR] = AV_CHAN_SIDE_RIGHT,
};
static const int sc_map[16] = {
[AV_CHAN_FRONT_LEFT ] = SC_FL,
[AV_CHAN_FRONT_RIGHT ] = SC_FR,
[AV_CHAN_FRONT_CENTER ] = SC_FC,
[AV_CHAN_LOW_FREQUENCY] = SC_LF,
[AV_CHAN_BACK_LEFT ] = SC_BL,
[AV_CHAN_BACK_RIGHT ] = SC_BR,
[AV_CHAN_BACK_CENTER ] = SC_BC,
[AV_CHAN_SIDE_LEFT ] = SC_SL,
[AV_CHAN_SIDE_RIGHT ] = SC_SR,
};
typedef struct AudioSurroundContext {
const AVClass *class;
AVChannelLayout out_ch_layout;
AVChannelLayout in_ch_layout;
float level_in;
float level_out;
float f_i[SC_NB];
float f_o[SC_NB];
int lfe_mode;
float smooth;
float angle;
float focus;
int win_size;
int win_func;
float win_gain;
float overlap;
float all_x;
float all_y;
float f_x[SC_NB];
float f_y[SC_NB];
float *input_levels;
float *output_levels;
int output_lfe;
int create_lfe;
int lowcutf;
int highcutf;
float lowcut;
float highcut;
int nb_in_channels;
int nb_out_channels;
AVFrame *factors;
AVFrame *sfactors;
AVFrame *input_in;
AVFrame *input;
AVFrame *output;
AVFrame *output_mag;
AVFrame *output_ph;
AVFrame *output_out;
AVFrame *overlap_buffer;
AVFrame *window;
float *x_pos;
float *y_pos;
float *l_phase;
float *r_phase;
float *c_phase;
float *c_mag;
float *lfe_mag;
float *lfe_phase;
float *mag_total;
int rdft_size;
int hop_size;
AVTXContext **rdft, **irdft;
av_tx_fn tx_fn, itx_fn;
float *window_func_lut;
void (*filter)(AVFilterContext *ctx);
void (*upmix)(AVFilterContext *ctx, int ch);
void (*upmix_5_0)(AVFilterContext *ctx,
float c_re, float c_im,
float mag_totall, float mag_totalr,
float fl_phase, float fr_phase,
float bl_phase, float br_phase,
float sl_phase, float sr_phase,
float xl, float yl,
float xr, float yr,
int n);
void (*upmix_5_1)(AVFilterContext *ctx,
float c_re, float c_im,
float lfe_re, float lfe_im,
float mag_totall, float mag_totalr,
float fl_phase, float fr_phase,
float bl_phase, float br_phase,
float sl_phase, float sr_phase,
float xl, float yl,
float xr, float yr,
int n);
} AudioSurroundContext;
static int query_formats(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
int ret;
ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLTP);
if (ret)
return ret;
ret = ff_set_common_formats(ctx, formats);
if (ret)
return ret;
layouts = NULL;
ret = ff_add_channel_layout(&layouts, &s->out_ch_layout);
if (ret)
return ret;
ret = ff_channel_layouts_ref(layouts, &ctx->outputs[0]->incfg.channel_layouts);
if (ret)
return ret;
layouts = NULL;
ret = ff_add_channel_layout(&layouts, &s->in_ch_layout);
if (ret)
return ret;
ret = ff_channel_layouts_ref(layouts, &ctx->inputs[0]->outcfg.channel_layouts);
if (ret)
return ret;
return ff_set_common_all_samplerates(ctx);
}
static void set_input_levels(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
for (int ch = 0; ch < s->nb_in_channels && s->level_in >= 0.f; ch++)
s->input_levels[ch] = s->level_in;
s->level_in = -1.f;
for (int n = 0; n < SC_NB; n++) {
const int ch = av_channel_layout_index_from_channel(&s->in_ch_layout, ch_map[n]);
if (ch >= 0)
s->input_levels[ch] = s->f_i[n];
}
}
static void set_output_levels(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
for (int ch = 0; ch < s->nb_out_channels && s->level_out >= 0.f; ch++)
s->output_levels[ch] = s->level_out;
s->level_out = -1.f;
for (int n = 0; n < SC_NB; n++) {
const int ch = av_channel_layout_index_from_channel(&s->out_ch_layout, ch_map[n]);
if (ch >= 0)
s->output_levels[ch] = s->f_o[n];
}
}
static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioSurroundContext *s = ctx->priv;
int ret;
s->rdft = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->rdft));
if (!s->rdft)
return AVERROR(ENOMEM);
s->nb_in_channels = inlink->ch_layout.nb_channels;
for (int ch = 0; ch < inlink->ch_layout.nb_channels; ch++) {
float scale = 1.f;
ret = av_tx_init(&s->rdft[ch], &s->tx_fn, AV_TX_FLOAT_RDFT,
0, s->win_size, &scale, 0);
if (ret < 0)
return ret;
}
s->input_levels = av_malloc_array(s->nb_in_channels, sizeof(*s->input_levels));
if (!s->input_levels)
return AVERROR(ENOMEM);
set_input_levels(ctx);
s->window = ff_get_audio_buffer(inlink, s->win_size * 2);
if (!s->window)
return AVERROR(ENOMEM);
s->input_in = ff_get_audio_buffer(inlink, s->win_size * 2);
if (!s->input_in)
return AVERROR(ENOMEM);
s->input = ff_get_audio_buffer(inlink, s->win_size + 2);
if (!s->input)
return AVERROR(ENOMEM);
s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->win_size / 2);
s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->win_size / 2);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioSurroundContext *s = ctx->priv;
int ret;
s->irdft = av_calloc(outlink->ch_layout.nb_channels, sizeof(*s->irdft));
if (!s->irdft)
return AVERROR(ENOMEM);
s->nb_out_channels = outlink->ch_layout.nb_channels;
for (int ch = 0; ch < outlink->ch_layout.nb_channels; ch++) {
float iscale = 1.f;
ret = av_tx_init(&s->irdft[ch], &s->itx_fn, AV_TX_FLOAT_RDFT,
1, s->win_size, &iscale, 0);
if (ret < 0)
return ret;
}
s->output_levels = av_malloc_array(s->nb_out_channels, sizeof(*s->output_levels));
if (!s->output_levels)
return AVERROR(ENOMEM);
set_output_levels(ctx);
s->factors = ff_get_audio_buffer(outlink, s->win_size + 2);
s->sfactors = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output_ph = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output_mag = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output_out = ff_get_audio_buffer(outlink, s->win_size + 2);
s->output = ff_get_audio_buffer(outlink, s->win_size + 2);
s->overlap_buffer = ff_get_audio_buffer(outlink, s->win_size * 2);
if (!s->overlap_buffer || !s->output || !s->output_out || !s->output_mag ||
!s->output_ph || !s->factors || !s->sfactors)
return AVERROR(ENOMEM);
s->rdft_size = s->win_size / 2 + 1;
s->x_pos = av_calloc(s->rdft_size, sizeof(*s->x_pos));
s->y_pos = av_calloc(s->rdft_size, sizeof(*s->y_pos));
s->l_phase = av_calloc(s->rdft_size, sizeof(*s->l_phase));
s->r_phase = av_calloc(s->rdft_size, sizeof(*s->r_phase));
s->c_mag = av_calloc(s->rdft_size, sizeof(*s->c_mag));
s->c_phase = av_calloc(s->rdft_size, sizeof(*s->c_phase));
s->mag_total = av_calloc(s->rdft_size, sizeof(*s->mag_total));
s->lfe_mag = av_calloc(s->rdft_size, sizeof(*s->lfe_mag));
s->lfe_phase = av_calloc(s->rdft_size, sizeof(*s->lfe_phase));
if (!s->x_pos || !s->y_pos || !s->l_phase || !s->r_phase || !s->lfe_phase ||
!s->c_phase || !s->mag_total || !s->lfe_mag || !s->c_mag)
return AVERROR(ENOMEM);
return 0;
}
static float sqrf(float x)
{
return x * x;
}
static float r_distance(float a)
{
return fminf(sqrtf(1.f + sqrf(tanf(a))), sqrtf(1.f + sqrf(1.f / tanf(a))));
}
#define MIN_MAG_SUM 0.00000001f
static void angle_transform(float *x, float *y, float angle)
{
float reference, r, a;
if (angle == 90.f)
return;
reference = angle * M_PIf / 180.f;
r = hypotf(*x, *y);
a = atan2f(*x, *y);
r /= r_distance(a);
if (fabsf(a) <= M_PI_4f)
a *= reference / M_PI_2f;
else
a = M_PIf + (-2.f * M_PIf + reference) * (M_PIf - fabsf(a)) * FFDIFFSIGN(a, 0.f) / (3.f * M_PI_2f);
r *= r_distance(a);
*x = av_clipf(sinf(a) * r, -1.f, 1.f);
*y = av_clipf(cosf(a) * r, -1.f, 1.f);
}
static void focus_transform(float *x, float *y, float focus)
{
float a, r, ra;
if (focus == 0.f)
return;
a = atan2f(*x, *y);
ra = r_distance(a);
r = av_clipf(hypotf(*x, *y) / ra, 0.f, 1.f);
r = focus > 0.f ? 1.f - powf(1.f - r, 1.f + focus * 20.f) : powf(r, 1.f - focus * 20.f);
r *= ra;
*x = av_clipf(sinf(a) * r, -1.f, 1.f);
*y = av_clipf(cosf(a) * r, -1.f, 1.f);
}
static void stereo_position(float a, float p, float *x, float *y)
{
av_assert2(a >= -1.f && a <= 1.f);
av_assert2(p >= 0.f && p <= M_PIf);
*x = av_clipf(a+a*fmaxf(0.f, p*p-M_PI_2f), -1.f, 1.f);
*y = av_clipf(cosf(a*M_PI_2f+M_PIf)*cosf(M_PI_2f-p/M_PIf)*M_LN10f+1.f, -1.f, 1.f);
}
static inline void get_lfe(int output_lfe, int n, float lowcut, float highcut,
float *lfe_mag, float c_mag, float *mag_total, int lfe_mode)
{
if (output_lfe && n < highcut) {
*lfe_mag = n < lowcut ? 1.f : .5f*(1.f+cosf(M_PIf*(lowcut-n)/(lowcut-highcut)));
*lfe_mag *= c_mag;
if (lfe_mode)
*mag_total -= *lfe_mag;
} else {
*lfe_mag = 0.f;
}
}
#define TRANSFORM \
dst[2 * n ] = mag * cosf(ph); \
dst[2 * n + 1] = mag * sinf(ph);
static void calculate_factors(AVFilterContext *ctx, int ch, int chan)
{
AudioSurroundContext *s = ctx->priv;
float *factor = (float *)s->factors->extended_data[ch];
const float f_x = s->f_x[sc_map[chan >= 0 ? chan : 0]];
const float f_y = s->f_y[sc_map[chan >= 0 ? chan : 0]];
const int rdft_size = s->rdft_size;
const float *x = s->x_pos;
const float *y = s->y_pos;
switch (chan) {
case AV_CHAN_FRONT_CENTER:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((y[n] + 1.f) * .5f, f_y);
break;
case AV_CHAN_FRONT_LEFT:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf((y[n] + 1.f) * .5f, f_y);
break;
case AV_CHAN_FRONT_RIGHT:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf((y[n] + 1.f) * .5f, f_y);
break;
case AV_CHAN_LOW_FREQUENCY:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((1.f - fabsf(y[n])), f_y);
break;
case AV_CHAN_BACK_CENTER:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(1.f - fabsf(x[n]), f_x) * powf((1.f - y[n]) * .5f, f_y);
break;
case AV_CHAN_BACK_LEFT:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf(1.f - ((y[n] + 1.f) * .5f), f_y);
break;
case AV_CHAN_BACK_RIGHT:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf(1.f - ((y[n] + 1.f) * .5f), f_y);
break;
case AV_CHAN_SIDE_LEFT:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(.5f * ( x[n] + 1.f), f_x) * powf(1.f - fabsf(y[n]), f_y);
break;
case AV_CHAN_SIDE_RIGHT:
for (int n = 0; n < rdft_size; n++)
factor[n] = powf(.5f * (-x[n] + 1.f), f_x) * powf(1.f - fabsf(y[n]), f_y);
break;
default:
for (int n = 0; n < rdft_size; n++)
factor[n] = 1.f;
break;
}
}
static void do_transform(AVFilterContext *ctx, int ch)
{
AudioSurroundContext *s = ctx->priv;
float *sfactor = (float *)s->sfactors->extended_data[ch];
float *factor = (float *)s->factors->extended_data[ch];
float *omag = (float *)s->output_mag->extended_data[ch];
float *oph = (float *)s->output_ph->extended_data[ch];
float *dst = (float *)s->output->extended_data[ch];
const int rdft_size = s->rdft_size;
const float smooth = s->smooth;
if (smooth > 0.f) {
for (int n = 0; n < rdft_size; n++)
sfactor[n] = smooth * factor[n] + (1.f - smooth) * sfactor[n];
factor = sfactor;
}
for (int n = 0; n < rdft_size; n++)
omag[n] *= factor[n];
for (int n = 0; n < rdft_size; n++) {
const float mag = omag[n];
const float ph = oph[n];
TRANSFORM
}
}
static void stereo_copy(AVFilterContext *ctx, int ch, int chan)
{
AudioSurroundContext *s = ctx->priv;
float *omag = (float *)s->output_mag->extended_data[ch];
float *oph = (float *)s->output_ph->extended_data[ch];
const float *mag_total = s->mag_total;
const int rdft_size = s->rdft_size;
const float *c_phase = s->c_phase;
const float *l_phase = s->l_phase;
const float *r_phase = s->r_phase;
const float *lfe_mag = s->lfe_mag;
const float *c_mag = s->c_mag;
switch (chan) {
case AV_CHAN_FRONT_CENTER:
memcpy(omag, c_mag, rdft_size * sizeof(*omag));
break;
case AV_CHAN_LOW_FREQUENCY:
memcpy(omag, lfe_mag, rdft_size * sizeof(*omag));
break;
case AV_CHAN_FRONT_LEFT:
case AV_CHAN_FRONT_RIGHT:
case AV_CHAN_BACK_CENTER:
case AV_CHAN_BACK_LEFT:
case AV_CHAN_BACK_RIGHT:
case AV_CHAN_SIDE_LEFT:
case AV_CHAN_SIDE_RIGHT:
memcpy(omag, mag_total, rdft_size * sizeof(*omag));
break;
default:
break;
}
switch (chan) {
case AV_CHAN_FRONT_CENTER:
case AV_CHAN_LOW_FREQUENCY:
case AV_CHAN_BACK_CENTER:
memcpy(oph, c_phase, rdft_size * sizeof(*oph));
break;
case AV_CHAN_FRONT_LEFT:
case AV_CHAN_BACK_LEFT:
case AV_CHAN_SIDE_LEFT:
memcpy(oph, l_phase, rdft_size * sizeof(*oph));
break;
case AV_CHAN_FRONT_RIGHT:
case AV_CHAN_BACK_RIGHT:
case AV_CHAN_SIDE_RIGHT:
memcpy(oph, r_phase, rdft_size * sizeof(*oph));
break;
default:
break;
}
}
static void stereo_upmix(AVFilterContext *ctx, int ch)
{
AudioSurroundContext *s = ctx->priv;
const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
calculate_factors(ctx, ch, chan);
stereo_copy(ctx, ch, chan);
do_transform(ctx, ch);
}
static void l2_1_upmix(AVFilterContext *ctx, int ch)
{
AudioSurroundContext *s = ctx->priv;
const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
float *omag = (float *)s->output_mag->extended_data[ch];
float *oph = (float *)s->output_ph->extended_data[ch];
const float *mag_total = s->mag_total;
const float *lfe_phase = s->lfe_phase;
const int rdft_size = s->rdft_size;
const float *c_phase = s->c_phase;
const float *l_phase = s->l_phase;
const float *r_phase = s->r_phase;
const float *lfe_mag = s->lfe_mag;
const float *c_mag = s->c_mag;
switch (chan) {
case AV_CHAN_LOW_FREQUENCY:
calculate_factors(ctx, ch, -1);
break;
default:
calculate_factors(ctx, ch, chan);
break;
}
switch (chan) {
case AV_CHAN_FRONT_CENTER:
memcpy(omag, c_mag, rdft_size * sizeof(*omag));
break;
case AV_CHAN_LOW_FREQUENCY:
memcpy(omag, lfe_mag, rdft_size * sizeof(*omag));
break;
case AV_CHAN_FRONT_LEFT:
case AV_CHAN_FRONT_RIGHT:
case AV_CHAN_BACK_CENTER:
case AV_CHAN_BACK_LEFT:
case AV_CHAN_BACK_RIGHT:
case AV_CHAN_SIDE_LEFT:
case AV_CHAN_SIDE_RIGHT:
memcpy(omag, mag_total, rdft_size * sizeof(*omag));
break;
default:
break;
}
switch (chan) {
case AV_CHAN_LOW_FREQUENCY:
memcpy(oph, lfe_phase, rdft_size * sizeof(*oph));
break;
case AV_CHAN_FRONT_CENTER:
case AV_CHAN_BACK_CENTER:
memcpy(oph, c_phase, rdft_size * sizeof(*oph));
break;
case AV_CHAN_FRONT_LEFT:
case AV_CHAN_BACK_LEFT:
case AV_CHAN_SIDE_LEFT:
memcpy(oph, l_phase, rdft_size * sizeof(*oph));
break;
case AV_CHAN_FRONT_RIGHT:
case AV_CHAN_BACK_RIGHT:
case AV_CHAN_SIDE_RIGHT:
memcpy(oph, r_phase, rdft_size * sizeof(*oph));
break;
default:
break;
}
do_transform(ctx, ch);
}
static void surround_upmix(AVFilterContext *ctx, int ch)
{
AudioSurroundContext *s = ctx->priv;
const int chan = av_channel_layout_channel_from_index(&s->out_ch_layout, ch);
switch (chan) {
case AV_CHAN_FRONT_CENTER:
calculate_factors(ctx, ch, -1);
break;
default:
calculate_factors(ctx, ch, chan);
break;
}
stereo_copy(ctx, ch, chan);
do_transform(ctx, ch);
}
static void upmix_7_1_5_0_side(AVFilterContext *ctx,
float c_re, float c_im,
float mag_totall, float mag_totalr,
float fl_phase, float fr_phase,
float bl_phase, float br_phase,
float sl_phase, float sr_phase,
float xl, float yl,
float xr, float yr,
int n)
{
float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
float lfe_mag, c_phase, mag_total = (mag_totall + mag_totalr) * 0.5f;
AudioSurroundContext *s = ctx->priv;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstc = (float *)s->output->extended_data[2];
dstlfe = (float *)s->output->extended_data[3];
dstlb = (float *)s->output->extended_data[4];
dstrb = (float *)s->output->extended_data[5];
dstls = (float *)s->output->extended_data[6];
dstrs = (float *)s->output->extended_data[7];
c_phase = atan2f(c_im, c_re);
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, hypotf(c_re, c_im), &mag_total, s->lfe_mode);
fl_mag = powf(.5f * (xl + 1.f), s->f_x[SC_FL]) * powf((yl + 1.f) * .5f, s->f_y[SC_FL]) * mag_totall;
fr_mag = powf(.5f * (xr + 1.f), s->f_x[SC_FR]) * powf((yr + 1.f) * .5f, s->f_y[SC_FR]) * mag_totalr;
lb_mag = powf(.5f * (-xl + 1.f), s->f_x[SC_BL]) * powf((yl + 1.f) * .5f, s->f_y[SC_BL]) * mag_totall;
rb_mag = powf(.5f * (-xr + 1.f), s->f_x[SC_BR]) * powf((yr + 1.f) * .5f, s->f_y[SC_BR]) * mag_totalr;
ls_mag = powf(1.f - fabsf(xl), s->f_x[SC_SL]) * powf((yl + 1.f) * .5f, s->f_y[SC_SL]) * mag_totall;
rs_mag = powf(1.f - fabsf(xr), s->f_x[SC_SR]) * powf((yr + 1.f) * .5f, s->f_y[SC_SR]) * mag_totalr;
dstl[2 * n ] = fl_mag * cosf(fl_phase);
dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
dstr[2 * n ] = fr_mag * cosf(fr_phase);
dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
dstc[2 * n ] = c_re;
dstc[2 * n + 1] = c_im;
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
dstlb[2 * n ] = lb_mag * cosf(bl_phase);
dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
dstrb[2 * n ] = rb_mag * cosf(br_phase);
dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
dstls[2 * n ] = ls_mag * cosf(sl_phase);
dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
dstrs[2 * n ] = rs_mag * cosf(sr_phase);
dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
}
static void upmix_7_1_5_1(AVFilterContext *ctx,
float c_re, float c_im,
float lfe_re, float lfe_im,
float mag_totall, float mag_totalr,
float fl_phase, float fr_phase,
float bl_phase, float br_phase,
float sl_phase, float sr_phase,
float xl, float yl,
float xr, float yr,
int n)
{
float fl_mag, fr_mag, ls_mag, rs_mag, lb_mag, rb_mag;
float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb, *dstlfe;
AudioSurroundContext *s = ctx->priv;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstc = (float *)s->output->extended_data[2];
dstlfe = (float *)s->output->extended_data[3];
dstlb = (float *)s->output->extended_data[4];
dstrb = (float *)s->output->extended_data[5];
dstls = (float *)s->output->extended_data[6];
dstrs = (float *)s->output->extended_data[7];
fl_mag = powf(.5f * (xl + 1.f), s->f_x[SC_FL]) * powf((yl + 1.f) * .5f, s->f_y[SC_FL]) * mag_totall;
fr_mag = powf(.5f * (xr + 1.f), s->f_x[SC_FR]) * powf((yr + 1.f) * .5f, s->f_y[SC_FR]) * mag_totalr;
lb_mag = powf(.5f * (-xl + 1.f), s->f_x[SC_BL]) * powf((yl + 1.f) * .5f, s->f_y[SC_BL]) * mag_totall;
rb_mag = powf(.5f * (-xr + 1.f), s->f_x[SC_BR]) * powf((yr + 1.f) * .5f, s->f_y[SC_BR]) * mag_totalr;
ls_mag = powf(1.f - fabsf(xl), s->f_x[SC_SL]) * powf((yl + 1.f) * .5f, s->f_y[SC_SL]) * mag_totall;
rs_mag = powf(1.f - fabsf(xr), s->f_x[SC_SR]) * powf((yr + 1.f) * .5f, s->f_y[SC_SR]) * mag_totalr;
dstl[2 * n ] = fl_mag * cosf(fl_phase);
dstl[2 * n + 1] = fl_mag * sinf(fl_phase);
dstr[2 * n ] = fr_mag * cosf(fr_phase);
dstr[2 * n + 1] = fr_mag * sinf(fr_phase);
dstc[2 * n ] = c_re;
dstc[2 * n + 1] = c_im;
dstlfe[2 * n ] = lfe_re;
dstlfe[2 * n + 1] = lfe_im;
dstlb[2 * n ] = lb_mag * cosf(bl_phase);
dstlb[2 * n + 1] = lb_mag * sinf(bl_phase);
dstrb[2 * n ] = rb_mag * cosf(br_phase);
dstrb[2 * n + 1] = rb_mag * sinf(br_phase);
dstls[2 * n ] = ls_mag * cosf(sl_phase);
dstls[2 * n + 1] = ls_mag * sinf(sl_phase);
dstrs[2 * n ] = rs_mag * cosf(sr_phase);
dstrs[2 * n + 1] = rs_mag * sinf(sr_phase);
}
static void filter_stereo(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
const float *srcl = (const float *)s->input->extended_data[0];
const float *srcr = (const float *)s->input->extended_data[1];
const int output_lfe = s->output_lfe && s->create_lfe;
const int rdft_size = s->rdft_size;
const int lfe_mode = s->lfe_mode;
const float highcut = s->highcut;
const float lowcut = s->lowcut;
const float angle = s->angle;
const float focus = s->focus;
float *magtotal = s->mag_total;
float *lfemag = s->lfe_mag;
float *lphase = s->l_phase;
float *rphase = s->r_phase;
float *cphase = s->c_phase;
float *cmag = s->c_mag;
float *xpos = s->x_pos;
float *ypos = s->y_pos;
for (int n = 0; n < rdft_size; n++) {
float l_re = srcl[2 * n], r_re = srcr[2 * n];
float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
float c_phase = atan2f(l_im + r_im, l_re + r_re);
float l_mag = hypotf(l_re, l_im);
float r_mag = hypotf(r_re, r_im);
float mag_total = hypotf(l_mag, r_mag);
float l_phase = atan2f(l_im, l_re);
float r_phase = atan2f(r_im, r_re);
float phase_dif = fabsf(l_phase - r_phase);
float mag_sum = l_mag + r_mag;
float c_mag = mag_sum * 0.5f;
float mag_dif, x, y;
mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum;
mag_dif = (l_mag - r_mag) / mag_sum;
if (phase_dif > M_PIf)
phase_dif = 2.f * M_PIf - phase_dif;
stereo_position(mag_dif, phase_dif, &x, &y);
angle_transform(&x, &y, angle);
focus_transform(&x, &y, focus);
get_lfe(output_lfe, n, lowcut, highcut, &lfemag[n], c_mag, &mag_total, lfe_mode);
xpos[n] = x;
ypos[n] = y;
lphase[n] = l_phase;
rphase[n] = r_phase;
cmag[n] = c_mag;
cphase[n] = c_phase;
magtotal[n] = mag_total;
}
}
static void filter_2_1(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
const float *srcl = (const float *)s->input->extended_data[0];
const float *srcr = (const float *)s->input->extended_data[1];
const float *srclfe = (const float *)s->input->extended_data[2];
const int rdft_size = s->rdft_size;
const float angle = s->angle;
const float focus = s->focus;
float *magtotal = s->mag_total;
float *lfephase = s->lfe_phase;
float *lfemag = s->lfe_mag;
float *lphase = s->l_phase;
float *rphase = s->r_phase;
float *cphase = s->c_phase;
float *cmag = s->c_mag;
float *xpos = s->x_pos;
float *ypos = s->y_pos;
for (int n = 0; n < rdft_size; n++) {
float l_re = srcl[2 * n], r_re = srcr[2 * n];
float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
float c_phase = atan2f(l_im + r_im, l_re + r_re);
float l_mag = hypotf(l_re, l_im);
float r_mag = hypotf(r_re, r_im);
float lfe_mag = hypotf(lfe_re, lfe_im);
float lfe_phase = atan2f(lfe_im, lfe_re);
float mag_total = hypotf(l_mag, r_mag);
float l_phase = atan2f(l_im, l_re);
float r_phase = atan2f(r_im, r_re);
float phase_dif = fabsf(l_phase - r_phase);
float mag_sum = l_mag + r_mag;
float c_mag = mag_sum * 0.5f;
float mag_dif, x, y;
mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum;
mag_dif = (l_mag - r_mag) / mag_sum;
if (phase_dif > M_PIf)
phase_dif = 2.f * M_PIf - phase_dif;
stereo_position(mag_dif, phase_dif, &x, &y);
angle_transform(&x, &y, angle);
focus_transform(&x, &y, focus);
xpos[n] = x;
ypos[n] = y;
lphase[n] = l_phase;
rphase[n] = r_phase;
cmag[n] = c_mag;
cphase[n] = c_phase;
lfemag[n] = lfe_mag;
lfephase[n] = lfe_phase;
magtotal[n] = mag_total;
}
}
static void filter_surround(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
const float *srcl = (const float *)s->input->extended_data[0];
const float *srcr = (const float *)s->input->extended_data[1];
const float *srcc = (const float *)s->input->extended_data[2];
const int output_lfe = s->output_lfe && s->create_lfe;
const int rdft_size = s->rdft_size;
const int lfe_mode = s->lfe_mode;
const float highcut = s->highcut;
const float lowcut = s->lowcut;
const float angle = s->angle;
const float focus = s->focus;
float *magtotal = s->mag_total;
float *lfemag = s->lfe_mag;
float *lphase = s->l_phase;
float *rphase = s->r_phase;
float *cphase = s->c_phase;
float *cmag = s->c_mag;
float *xpos = s->x_pos;
float *ypos = s->y_pos;
for (int n = 0; n < rdft_size; n++) {
float l_re = srcl[2 * n], r_re = srcr[2 * n];
float l_im = srcl[2 * n + 1], r_im = srcr[2 * n + 1];
float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
float c_phase = atan2f(c_im, c_re);
float c_mag = hypotf(c_re, c_im);
float l_mag = hypotf(l_re, l_im);
float r_mag = hypotf(r_re, r_im);
float mag_total = hypotf(l_mag, r_mag);
float l_phase = atan2f(l_im, l_re);
float r_phase = atan2f(r_im, r_re);
float phase_dif = fabsf(l_phase - r_phase);
float mag_sum = l_mag + r_mag;
float mag_dif, x, y;
mag_sum = mag_sum < MIN_MAG_SUM ? 1.f : mag_sum;
mag_dif = (l_mag - r_mag) / mag_sum;
if (phase_dif > M_PIf)
phase_dif = 2.f * M_PIf - phase_dif;
stereo_position(mag_dif, phase_dif, &x, &y);
angle_transform(&x, &y, angle);
focus_transform(&x, &y, focus);
get_lfe(output_lfe, n, lowcut, highcut, &lfemag[n], c_mag, &mag_total, lfe_mode);
xpos[n] = x;
ypos[n] = y;
lphase[n] = l_phase;
rphase[n] = r_phase;
cmag[n] = c_mag;
cphase[n] = c_phase;
magtotal[n] = mag_total;
}
}
static void filter_5_0_side(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
const int rdft_size = s->rdft_size;
float *srcl, *srcr, *srcc, *srcsl, *srcsr;
int n;
srcl = (float *)s->input->extended_data[0];
srcr = (float *)s->input->extended_data[1];
srcc = (float *)s->input->extended_data[2];
srcsl = (float *)s->input->extended_data[3];
srcsr = (float *)s->input->extended_data[4];
for (n = 0; n < rdft_size; n++) {
float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
float fl_mag = hypotf(fl_re, fl_im);
float fr_mag = hypotf(fr_re, fr_im);
float fl_phase = atan2f(fl_im, fl_re);
float fr_phase = atan2f(fr_im, fr_re);
float sl_mag = hypotf(sl_re, sl_im);
float sr_mag = hypotf(sr_re, sr_im);
float sl_phase = atan2f(sl_im, sl_re);
float sr_phase = atan2f(sr_im, sr_re);
float phase_difl = fabsf(fl_phase - sl_phase);
float phase_difr = fabsf(fr_phase - sr_phase);
float magl_sum = fl_mag + sl_mag;
float magr_sum = fr_mag + sr_mag;
float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum;
float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, sr_mag) : (fr_mag - sr_mag) / magr_sum;
float mag_totall = hypotf(fl_mag, sl_mag);
float mag_totalr = hypotf(fr_mag, sr_mag);
float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
float xl, yl;
float xr, yr;
if (phase_difl > M_PIf)
phase_difl = 2.f * M_PIf - phase_difl;
if (phase_difr > M_PIf)
phase_difr = 2.f * M_PIf - phase_difr;
stereo_position(mag_difl, phase_difl, &xl, &yl);
stereo_position(mag_difr, phase_difr, &xr, &yr);
s->upmix_5_0(ctx, c_re, c_im,
mag_totall, mag_totalr,
fl_phase, fr_phase,
bl_phase, br_phase,
sl_phase, sr_phase,
xl, yl, xr, yr, n);
}
}
static void filter_5_1_side(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
const int rdft_size = s->rdft_size;
float *srcl, *srcr, *srcc, *srclfe, *srcsl, *srcsr;
int n;
srcl = (float *)s->input->extended_data[0];
srcr = (float *)s->input->extended_data[1];
srcc = (float *)s->input->extended_data[2];
srclfe = (float *)s->input->extended_data[3];
srcsl = (float *)s->input->extended_data[4];
srcsr = (float *)s->input->extended_data[5];
for (n = 0; n < rdft_size; n++) {
float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
float sl_re = srcsl[2 * n], sl_im = srcsl[2 * n + 1];
float sr_re = srcsr[2 * n], sr_im = srcsr[2 * n + 1];
float fl_mag = hypotf(fl_re, fl_im);
float fr_mag = hypotf(fr_re, fr_im);
float fl_phase = atan2f(fl_im, fl_re);
float fr_phase = atan2f(fr_im, fr_re);
float sl_mag = hypotf(sl_re, sl_im);
float sr_mag = hypotf(sr_re, sr_im);
float sl_phase = atan2f(sl_im, sl_re);
float sr_phase = atan2f(sr_im, sr_re);
float phase_difl = fabsf(fl_phase - sl_phase);
float phase_difr = fabsf(fr_phase - sr_phase);
float magl_sum = fl_mag + sl_mag;
float magr_sum = fr_mag + sr_mag;
float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum;
float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, sr_mag) : (fr_mag - sr_mag) / magr_sum;
float mag_totall = hypotf(fl_mag, sl_mag);
float mag_totalr = hypotf(fr_mag, sr_mag);
float bl_phase = atan2f(fl_im + sl_im, fl_re + sl_re);
float br_phase = atan2f(fr_im + sr_im, fr_re + sr_re);
float xl, yl;
float xr, yr;
if (phase_difl > M_PIf)
phase_difl = 2.f * M_PIf - phase_difl;
if (phase_difr > M_PIf)
phase_difr = 2.f * M_PIf - phase_difr;
stereo_position(mag_difl, phase_difl, &xl, &yl);
stereo_position(mag_difr, phase_difr, &xr, &yr);
s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
mag_totall, mag_totalr,
fl_phase, fr_phase,
bl_phase, br_phase,
sl_phase, sr_phase,
xl, yl, xr, yr, n);
}
}
static void filter_5_1_back(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
const int rdft_size = s->rdft_size;
float *srcl, *srcr, *srcc, *srclfe, *srcbl, *srcbr;
int n;
srcl = (float *)s->input->extended_data[0];
srcr = (float *)s->input->extended_data[1];
srcc = (float *)s->input->extended_data[2];
srclfe = (float *)s->input->extended_data[3];
srcbl = (float *)s->input->extended_data[4];
srcbr = (float *)s->input->extended_data[5];
for (n = 0; n < rdft_size; n++) {
float fl_re = srcl[2 * n], fr_re = srcr[2 * n];
float fl_im = srcl[2 * n + 1], fr_im = srcr[2 * n + 1];
float c_re = srcc[2 * n], c_im = srcc[2 * n + 1];
float lfe_re = srclfe[2 * n], lfe_im = srclfe[2 * n + 1];
float bl_re = srcbl[2 * n], bl_im = srcbl[2 * n + 1];
float br_re = srcbr[2 * n], br_im = srcbr[2 * n + 1];
float fl_mag = hypotf(fl_re, fl_im);
float fr_mag = hypotf(fr_re, fr_im);
float fl_phase = atan2f(fl_im, fl_re);
float fr_phase = atan2f(fr_im, fr_re);
float bl_mag = hypotf(bl_re, bl_im);
float br_mag = hypotf(br_re, br_im);
float bl_phase = atan2f(bl_im, bl_re);
float br_phase = atan2f(br_im, br_re);
float phase_difl = fabsf(fl_phase - bl_phase);
float phase_difr = fabsf(fr_phase - br_phase);
float magl_sum = fl_mag + bl_mag;
float magr_sum = fr_mag + br_mag;
float mag_difl = magl_sum < MIN_MAG_SUM ? FFDIFFSIGN(fl_mag, bl_mag) : (fl_mag - bl_mag) / magl_sum;
float mag_difr = magr_sum < MIN_MAG_SUM ? FFDIFFSIGN(fr_mag, br_mag) : (fr_mag - br_mag) / magr_sum;
float mag_totall = hypotf(fl_mag, bl_mag);
float mag_totalr = hypotf(fr_mag, br_mag);
float sl_phase = atan2f(fl_im + bl_im, fl_re + bl_re);
float sr_phase = atan2f(fr_im + br_im, fr_re + br_re);
float xl, yl;
float xr, yr;
if (phase_difl > M_PIf)
phase_difl = 2.f * M_PIf - phase_difl;
if (phase_difr > M_PIf)
phase_difr = 2.f * M_PIf - phase_difr;
stereo_position(mag_difl, phase_difl, &xl, &yl);
stereo_position(mag_difr, phase_difr, &xr, &yr);
s->upmix_5_1(ctx, c_re, c_im, lfe_re, lfe_im,
mag_totall, mag_totalr,
fl_phase, fr_phase,
bl_phase, br_phase,
sl_phase, sr_phase,
xl, yl, xr, yr, n);
}
}
static void allchannels_spread(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
if (s->all_x >= 0.f)
for (int n = 0; n < SC_NB; n++)
s->f_x[n] = s->all_x;
s->all_x = -1.f;
if (s->all_y >= 0.f)
for (int n = 0; n < SC_NB; n++)
s->f_y[n] = s->all_y;
s->all_y = -1.f;
}
static av_cold int init(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
int64_t in_channel_layout, out_channel_layout;
char in_name[128], out_name[128];
float overlap;
if (s->lowcutf >= s->highcutf) {
av_log(ctx, AV_LOG_ERROR, "Low cut-off '%d' should be less than high cut-off '%d'.\n",
s->lowcutf, s->highcutf);
return AVERROR(EINVAL);
}
in_channel_layout = s->in_ch_layout.order == AV_CHANNEL_ORDER_NATIVE ?
s->in_ch_layout.u.mask : 0;
out_channel_layout = s->out_ch_layout.order == AV_CHANNEL_ORDER_NATIVE ?
s->out_ch_layout.u.mask : 0;
s->create_lfe = av_channel_layout_index_from_channel(&s->out_ch_layout,
AV_CHAN_LOW_FREQUENCY) >= 0;
switch (in_channel_layout) {
case AV_CH_LAYOUT_STEREO:
s->filter = filter_stereo;
s->upmix = stereo_upmix;
break;
case AV_CH_LAYOUT_2POINT1:
s->filter = filter_2_1;
s->upmix = l2_1_upmix;
break;
case AV_CH_LAYOUT_SURROUND:
s->filter = filter_surround;
s->upmix = surround_upmix;
break;
case AV_CH_LAYOUT_5POINT0:
s->filter = filter_5_0_side;
switch (out_channel_layout) {
case AV_CH_LAYOUT_7POINT1:
s->upmix_5_0 = upmix_7_1_5_0_side;
break;
default:
goto fail;
}
break;
case AV_CH_LAYOUT_5POINT1:
s->filter = filter_5_1_side;
switch (out_channel_layout) {
case AV_CH_LAYOUT_7POINT1:
s->upmix_5_1 = upmix_7_1_5_1;
break;
default:
goto fail;
}
break;
case AV_CH_LAYOUT_5POINT1_BACK:
s->filter = filter_5_1_back;
switch (out_channel_layout) {
case AV_CH_LAYOUT_7POINT1:
s->upmix_5_1 = upmix_7_1_5_1;
break;
default:
goto fail;
}
break;
default:
fail:
av_channel_layout_describe(&s->out_ch_layout, out_name, sizeof(out_name));
av_channel_layout_describe(&s->in_ch_layout, in_name, sizeof(in_name));
av_log(ctx, AV_LOG_ERROR, "Unsupported upmix: '%s' -> '%s'.\n",
in_name, out_name);
return AVERROR(EINVAL);
}
s->window_func_lut = av_calloc(s->win_size, sizeof(*s->window_func_lut));
if (!s->window_func_lut)
return AVERROR(ENOMEM);
generate_window_func(s->window_func_lut, s->win_size, s->win_func, &overlap);
if (s->overlap == 1)
s->overlap = overlap;
for (int i = 0; i < s->win_size; i++)
s->window_func_lut[i] = sqrtf(s->window_func_lut[i] / s->win_size);
s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap));
{
float max = 0.f, *temp_lut = av_calloc(s->win_size, sizeof(*temp_lut));
if (!temp_lut)
return AVERROR(ENOMEM);
for (int j = 0; j < s->win_size; j += s->hop_size) {
for (int i = 0; i < s->win_size; i++)
temp_lut[(i + j) % s->win_size] += s->window_func_lut[i];
}
for (int i = 0; i < s->win_size; i++)
max = fmaxf(temp_lut[i], max);
av_freep(&temp_lut);
s->win_gain = 1.f / (max * sqrtf(s->win_size));
}
allchannels_spread(ctx);
return 0;
}
static int fft_channel(AVFilterContext *ctx, AVFrame *in, int ch)
{
AudioSurroundContext *s = ctx->priv;
float *src = (float *)s->input_in->extended_data[ch];
float *win = (float *)s->window->extended_data[ch];
const float *window_func_lut = s->window_func_lut;
const int offset = s->win_size - s->hop_size;
const float level_in = s->input_levels[ch];
const int win_size = s->win_size;
memmove(src, &src[s->hop_size], offset * sizeof(float));
memcpy(&src[offset], in->extended_data[ch], in->nb_samples * sizeof(float));
memset(&src[offset + in->nb_samples], 0, (s->hop_size - in->nb_samples) * sizeof(float));
for (int n = 0; n < win_size; n++)
win[n] = src[n] * window_func_lut[n] * level_in;
s->tx_fn(s->rdft[ch], (float *)s->input->extended_data[ch], win, sizeof(float));
return 0;
}
static int fft_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AVFrame *in = arg;
const int start = (in->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (in->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++)
fft_channel(ctx, in, ch);
return 0;
}
static int ifft_channel(AVFilterContext *ctx, AVFrame *out, int ch)
{
AudioSurroundContext *s = ctx->priv;
const float level_out = s->output_levels[ch] * s->win_gain;
const float *window_func_lut = s->window_func_lut;
const int win_size = s->win_size;
float *dst, *ptr;
dst = (float *)s->output_out->extended_data[ch];
ptr = (float *)s->overlap_buffer->extended_data[ch];
s->itx_fn(s->irdft[ch], dst, (float *)s->output->extended_data[ch], sizeof(AVComplexFloat));
memmove(s->overlap_buffer->extended_data[ch],
s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float),
s->win_size * sizeof(float));
memset(s->overlap_buffer->extended_data[ch] + s->win_size * sizeof(float),
0, s->hop_size * sizeof(float));
for (int n = 0; n < win_size; n++)
ptr[n] += dst[n] * window_func_lut[n] * level_out;
ptr = (float *)s->overlap_buffer->extended_data[ch];
dst = (float *)out->extended_data[ch];
memcpy(dst, ptr, s->hop_size * sizeof(float));
return 0;
}
static int ifft_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
{
AudioSurroundContext *s = ctx->priv;
AVFrame *out = arg;
const int start = (out->ch_layout.nb_channels * jobnr) / nb_jobs;
const int end = (out->ch_layout.nb_channels * (jobnr+1)) / nb_jobs;
for (int ch = start; ch < end; ch++) {
if (s->upmix)
s->upmix(ctx, ch);
ifft_channel(ctx, out, ch);
}
return 0;
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
AVFrame *out;
ff_filter_execute(ctx, fft_channels, in, NULL,
FFMIN(inlink->ch_layout.nb_channels,
ff_filter_get_nb_threads(ctx)));
s->filter(ctx);
out = ff_get_audio_buffer(outlink, s->hop_size);
if (!out)
return AVERROR(ENOMEM);
ff_filter_execute(ctx, ifft_channels, out, NULL,
FFMIN(outlink->ch_layout.nb_channels,
ff_filter_get_nb_threads(ctx)));
av_frame_copy_props(out, in);
out->nb_samples = in->nb_samples;
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];
AudioSurroundContext *s = ctx->priv;
AVFrame *in = NULL;
int ret = 0, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
ret = ff_inlink_consume_samples(inlink, s->hop_size, s->hop_size, &in);
if (ret < 0)
return ret;
if (ret > 0)
ret = filter_frame(inlink, in);
if (ret < 0)
return ret;
if (ff_inlink_queued_samples(inlink) >= s->hop_size) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
ff_outlink_set_status(outlink, status, pts);
return 0;
}
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
av_frame_free(&s->factors);
av_frame_free(&s->sfactors);
av_frame_free(&s->window);
av_frame_free(&s->input_in);
av_frame_free(&s->input);
av_frame_free(&s->output);
av_frame_free(&s->output_ph);
av_frame_free(&s->output_mag);
av_frame_free(&s->output_out);
av_frame_free(&s->overlap_buffer);
for (int ch = 0; ch < s->nb_in_channels; ch++)
av_tx_uninit(&s->rdft[ch]);
for (int ch = 0; ch < s->nb_out_channels; ch++)
av_tx_uninit(&s->irdft[ch]);
av_freep(&s->input_levels);
av_freep(&s->output_levels);
av_freep(&s->rdft);
av_freep(&s->irdft);
av_freep(&s->window_func_lut);
av_freep(&s->x_pos);
av_freep(&s->y_pos);
av_freep(&s->l_phase);
av_freep(&s->r_phase);
av_freep(&s->c_mag);
av_freep(&s->c_phase);
av_freep(&s->mag_total);
av_freep(&s->lfe_mag);
av_freep(&s->lfe_phase);
}
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
AudioSurroundContext *s = ctx->priv;
int ret;
ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
s->hop_size = FFMAX(1, s->win_size * (1. - s->overlap));
allchannels_spread(ctx);
set_input_levels(ctx);
set_output_levels(ctx);
return 0;
}
#define OFFSET(x) offsetof(AudioSurroundContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
#define TFLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
static const AVOption surround_options[] = {
{ "chl_out", "set output channel layout", OFFSET(out_ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str="5.1"}, 0, 0, FLAGS },
{ "chl_in", "set input channel layout", OFFSET(in_ch_layout), AV_OPT_TYPE_CHLAYOUT, {.str="stereo"},0, 0, FLAGS },
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "lfe", "output LFE", OFFSET(output_lfe), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, TFLAGS },
{ "lfe_low", "LFE low cut off", OFFSET(lowcutf), AV_OPT_TYPE_INT, {.i64=128}, 0, 256, FLAGS },
{ "lfe_high", "LFE high cut off", OFFSET(highcutf), AV_OPT_TYPE_INT, {.i64=256}, 0, 512, FLAGS },
{ "lfe_mode", "set LFE channel mode", OFFSET(lfe_mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, TFLAGS, .unit = "lfe_mode" },
{ "add", "just add LFE channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 1, TFLAGS, .unit = "lfe_mode" },
{ "sub", "subtract LFE channel with others", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 1, TFLAGS, .unit = "lfe_mode" },
{ "smooth", "set temporal smoothness strength", OFFSET(smooth), AV_OPT_TYPE_FLOAT, {.dbl=0}, 0, 1, TFLAGS },
{ "angle", "set soundfield transform angle", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl=90}, 0, 360, TFLAGS },
{ "focus", "set soundfield transform focus", OFFSET(focus), AV_OPT_TYPE_FLOAT, {.dbl=0}, -1, 1, TFLAGS },
{ "fc_in", "set front center channel input level", OFFSET(f_i[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "fc_out", "set front center channel output level", OFFSET(f_o[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "fl_in", "set front left channel input level", OFFSET(f_i[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "fl_out", "set front left channel output level", OFFSET(f_o[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "fr_in", "set front right channel input level", OFFSET(f_i[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "fr_out", "set front right channel output level", OFFSET(f_o[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "sl_in", "set side left channel input level", OFFSET(f_i[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "sl_out", "set side left channel output level", OFFSET(f_o[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "sr_in", "set side right channel input level", OFFSET(f_i[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "sr_out", "set side right channel output level", OFFSET(f_o[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "bl_in", "set back left channel input level", OFFSET(f_i[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "bl_out", "set back left channel output level", OFFSET(f_o[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "br_in", "set back right channel input level", OFFSET(f_i[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "br_out", "set back right channel output level", OFFSET(f_o[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "bc_in", "set back center channel input level", OFFSET(f_i[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "bc_out", "set back center channel output level", OFFSET(f_o[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "lfe_in", "set lfe channel input level", OFFSET(f_i[SC_LF]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "lfe_out", "set lfe channel output level", OFFSET(f_o[SC_LF]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, TFLAGS },
{ "allx", "set all channel's x spread", OFFSET(all_x), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, TFLAGS },
{ "ally", "set all channel's y spread", OFFSET(all_y), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, TFLAGS },
{ "fcx", "set front center channel x spread", OFFSET(f_x[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "flx", "set front left channel x spread", OFFSET(f_x[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "frx", "set front right channel x spread", OFFSET(f_x[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "blx", "set back left channel x spread", OFFSET(f_x[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "brx", "set back right channel x spread", OFFSET(f_x[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "slx", "set side left channel x spread", OFFSET(f_x[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "srx", "set side right channel x spread", OFFSET(f_x[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "bcx", "set back center channel x spread", OFFSET(f_x[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "fcy", "set front center channel y spread", OFFSET(f_y[SC_FC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "fly", "set front left channel y spread", OFFSET(f_y[SC_FL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "fry", "set front right channel y spread", OFFSET(f_y[SC_FR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "bly", "set back left channel y spread", OFFSET(f_y[SC_BL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "bry", "set back right channel y spread", OFFSET(f_y[SC_BR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "sly", "set side left channel y spread", OFFSET(f_y[SC_SL]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "sry", "set side right channel y spread", OFFSET(f_y[SC_SR]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "bcy", "set back center channel y spread", OFFSET(f_y[SC_BC]), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, .06, 15, TFLAGS },
{ "win_size", "set window size", OFFSET(win_size), AV_OPT_TYPE_INT, {.i64=4096},1024,65536,FLAGS },
WIN_FUNC_OPTION("win_func", OFFSET(win_func), FLAGS, WFUNC_HANNING),
{ "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, 0, 1, TFLAGS },
{ NULL }
};
AVFILTER_DEFINE_CLASS(surround);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
};
const AVFilter ff_af_surround = {
.name = "surround",
.description = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."),
.priv_size = sizeof(AudioSurroundContext),
.priv_class = &surround_class,
.init = init,
.uninit = uninit,
.activate = activate,
FILTER_INPUTS(inputs),
FILTER_OUTPUTS(outputs),
FILTER_QUERY_FUNC(query_formats),
.flags = AVFILTER_FLAG_SLICE_THREADS,
.process_command = process_command,
};
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