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FFmpeg/libavfilter/af_surround.c
Michael Niedermayer 22ee55a1da
avfilter/af_surround: Check output format 
Fixes: CID1516994 Out-of-bounds access
Fixes: CID1516996 Out-of-bounds access
Fixes: CID1516999 Out-of-bounds access

Sponsored-by: Sovereign Tech Fund
Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
2024-08-11 13:21:12 +02:00

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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 (out_channel_layout) {
case AV_CH_LAYOUT_MONO:
case AV_CH_LAYOUT_STEREO:
case AV_CH_LAYOUT_2POINT1:
case AV_CH_LAYOUT_2_1:
case AV_CH_LAYOUT_2_2:
case AV_CH_LAYOUT_SURROUND:
case AV_CH_LAYOUT_3POINT1:
case AV_CH_LAYOUT_QUAD:
case AV_CH_LAYOUT_4POINT0:
case AV_CH_LAYOUT_4POINT1:
case AV_CH_LAYOUT_5POINT0:
case AV_CH_LAYOUT_5POINT1:
case AV_CH_LAYOUT_5POINT0_BACK:
case AV_CH_LAYOUT_5POINT1_BACK:
case AV_CH_LAYOUT_6POINT0:
case AV_CH_LAYOUT_6POINT1:
case AV_CH_LAYOUT_7POINT0:
case AV_CH_LAYOUT_7POINT1:
case AV_CH_LAYOUT_OCTAGONAL:
break;
default:
goto fail;
}
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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