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FFmpeg/libavfilter/af_surround.c
Andreas Rheinhardt b4f5201967 avfilter: Replace query_formats callback with union of list and callback 
If one looks at the many query_formats callbacks in existence,
one will immediately recognize that there is one type of default
callback for video and a slightly different default callback for
audio: It is "return ff_set_common_formats_from_list(ctx, pix_fmts);"
for video with a filter-specific pix_fmts list. For audio, it is
the same with a filter-specific sample_fmts list together with
ff_set_common_all_samplerates() and ff_set_common_all_channel_counts().

This commit allows to remove the boilerplate query_formats callbacks
by replacing said callback with a union consisting the old callback
and pointers for pixel and sample format arrays. For the not uncommon
case in which these lists only contain a single entry (besides the
sentinel) enum AVPixelFormat and enum AVSampleFormat fields are also
added to the union to store them directly in the AVFilter,
thereby avoiding a relocation.

The state of said union will be contained in a new, dedicated AVFilter
field (the nb_inputs and nb_outputs fields have been shrunk to uint8_t
in order to create a hole for this new field; this is no problem, as
the maximum of all the nb_inputs is four; for nb_outputs it is only
two).

The state's default value coincides with the earlier default of
query_formats being unset, namely that the filter accepts all formats
(and also sample rates and channel counts/layouts for audio)
provided that these properties agree coincide for all inputs and
outputs.

By using different union members for audio and video filters
the type-unsafety of using the same functions for audio and video
lists will furthermore be more confined to formats.c than before.

When the new fields are used, they will also avoid allocations:
Currently something nearly equivalent to ff_default_query_formats()
is called after every successful call to a query_formats callback;
yet in the common case that the newly allocated AVFilterFormats
are not used at all (namely if there are no free links) these newly
allocated AVFilterFormats are freed again without ever being used.
Filters no longer using the callback will not exhibit this any more.

Reviewed-by: Paul B Mahol <onemda@gmail.com>
Reviewed-by: Nicolas George <george@nsup.org>
Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2021-10-05 17:48:25 +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/audio_fifo.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "avfilter.h"
#include "audio.h"
#include "filters.h"
#include "internal.h"
#include "formats.h"
#include "window_func.h"
typedef struct AudioSurroundContext {
const AVClass *class;
char *out_channel_layout_str;
char *in_channel_layout_str;
float level_in;
float level_out;
float fc_in;
float fc_out;
float fl_in;
float fl_out;
float fr_in;
float fr_out;
float sl_in;
float sl_out;
float sr_in;
float sr_out;
float bl_in;
float bl_out;
float br_in;
float br_out;
float bc_in;
float bc_out;
float lfe_in;
float lfe_out;
int lfe_mode;
float angle;
int win_size;
int win_func;
float overlap;
float all_x;
float all_y;
float fc_x;
float fl_x;
float fr_x;
float bl_x;
float br_x;
float sl_x;
float sr_x;
float bc_x;
float fc_y;
float fl_y;
float fr_y;
float bl_y;
float br_y;
float sl_y;
float sr_y;
float bc_y;
float *input_levels;
float *output_levels;
int output_lfe;
int lowcutf;
int highcutf;
float lowcut;
float highcut;
uint64_t out_channel_layout;
uint64_t in_channel_layout;
int nb_in_channels;
int nb_out_channels;
AVFrame *input;
AVFrame *output;
AVFrame *overlap_buffer;
int buf_size;
int hop_size;
AVAudioFifo *fifo;
RDFTContext **rdft, **irdft;
float *window_func_lut;
int64_t pts;
int eof;
void (*filter)(AVFilterContext *ctx);
void (*upmix_stereo)(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n);
void (*upmix_2_1)(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float lfe_im,
float lfe_re,
float x, float y,
int n);
void (*upmix_3_0)(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_mag,
float c_phase,
float mag_total,
float x, float y,
int n);
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_channel_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_channel_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 int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AudioSurroundContext *s = ctx->priv;
int ch;
s->rdft = av_calloc(inlink->channels, sizeof(*s->rdft));
if (!s->rdft)
return AVERROR(ENOMEM);
for (ch = 0; ch < inlink->channels; ch++) {
s->rdft[ch] = av_rdft_init(ff_log2(s->buf_size), DFT_R2C);
if (!s->rdft[ch])
return AVERROR(ENOMEM);
}
s->nb_in_channels = inlink->channels;
s->input_levels = av_malloc_array(s->nb_in_channels, sizeof(*s->input_levels));
if (!s->input_levels)
return AVERROR(ENOMEM);
for (ch = 0; ch < s->nb_in_channels; ch++)
s->input_levels[ch] = s->level_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_FRONT_CENTER);
if (ch >= 0)
s->input_levels[ch] *= s->fc_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_FRONT_LEFT);
if (ch >= 0)
s->input_levels[ch] *= s->fl_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_FRONT_RIGHT);
if (ch >= 0)
s->input_levels[ch] *= s->fr_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_SIDE_LEFT);
if (ch >= 0)
s->input_levels[ch] *= s->sl_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_SIDE_RIGHT);
if (ch >= 0)
s->input_levels[ch] *= s->sr_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_BACK_LEFT);
if (ch >= 0)
s->input_levels[ch] *= s->bl_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_BACK_RIGHT);
if (ch >= 0)
s->input_levels[ch] *= s->br_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_BACK_CENTER);
if (ch >= 0)
s->input_levels[ch] *= s->bc_in;
ch = av_get_channel_layout_channel_index(inlink->channel_layout, AV_CH_LOW_FREQUENCY);
if (ch >= 0)
s->input_levels[ch] *= s->lfe_in;
s->input = ff_get_audio_buffer(inlink, s->buf_size * 2);
if (!s->input)
return AVERROR(ENOMEM);
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->buf_size);
if (!s->fifo)
return AVERROR(ENOMEM);
s->lowcut = 1.f * s->lowcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
s->highcut = 1.f * s->highcutf / (inlink->sample_rate * 0.5) * (s->buf_size / 2);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AudioSurroundContext *s = ctx->priv;
int ch;
s->irdft = av_calloc(outlink->channels, sizeof(*s->irdft));
if (!s->irdft)
return AVERROR(ENOMEM);
for (ch = 0; ch < outlink->channels; ch++) {
s->irdft[ch] = av_rdft_init(ff_log2(s->buf_size), IDFT_C2R);
if (!s->irdft[ch])
return AVERROR(ENOMEM);
}
s->nb_out_channels = outlink->channels;
s->output_levels = av_malloc_array(s->nb_out_channels, sizeof(*s->output_levels));
if (!s->output_levels)
return AVERROR(ENOMEM);
for (ch = 0; ch < s->nb_out_channels; ch++)
s->output_levels[ch] = s->level_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_FRONT_CENTER);
if (ch >= 0)
s->output_levels[ch] *= s->fc_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_FRONT_LEFT);
if (ch >= 0)
s->output_levels[ch] *= s->fl_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_FRONT_RIGHT);
if (ch >= 0)
s->output_levels[ch] *= s->fr_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_SIDE_LEFT);
if (ch >= 0)
s->output_levels[ch] *= s->sl_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_SIDE_RIGHT);
if (ch >= 0)
s->output_levels[ch] *= s->sr_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_BACK_LEFT);
if (ch >= 0)
s->output_levels[ch] *= s->bl_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_BACK_RIGHT);
if (ch >= 0)
s->output_levels[ch] *= s->br_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_BACK_CENTER);
if (ch >= 0)
s->output_levels[ch] *= s->bc_out;
ch = av_get_channel_layout_channel_index(outlink->channel_layout, AV_CH_LOW_FREQUENCY);
if (ch >= 0)
s->output_levels[ch] *= s->lfe_out;
s->output = ff_get_audio_buffer(outlink, s->buf_size * 2);
s->overlap_buffer = ff_get_audio_buffer(outlink, s->buf_size * 2);
if (!s->overlap_buffer || !s->output)
return AVERROR(ENOMEM);
return 0;
}
static void stereo_transform(float *x, float *y, float angle)
{
float reference, r, a;
if (angle == 90.f)
return;
reference = angle * M_PI / 180.f;
r = hypotf(*x, *y);
a = atan2f(*x, *y);
if (fabsf(a) <= M_PI_4)
a *= reference / M_PI_2;
else
a = M_PI + 2 * (-2 * M_PI + reference) * (M_PI - fabsf(a)) * FFDIFFSIGN(a, 0) / (3 * M_PI);
*x = av_clipf(sinf(a) * r, -1, 1);
*y = av_clipf(cosf(a) * r, -1, 1);
}
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_PI);
*x = av_clipf(a+a*FFMAX(0, p*p-M_PI_2), -1, 1);
*y = av_clipf(cosf(a*M_PI_2+M_PI)*cosf(M_PI_2-p/M_PI)*M_LN10+1, -1, 1);
}
static inline void get_lfe(int output_lfe, int n, float lowcut, float highcut,
float *lfe_mag, float *mag_total, int lfe_mode)
{
if (output_lfe && n < highcut) {
*lfe_mag = n < lowcut ? 1.f : .5f*(1.f+cosf(M_PI*(lowcut-n)/(lowcut-highcut)));
*lfe_mag *= *mag_total;
if (lfe_mode)
*mag_total -= *lfe_mag;
} else {
*lfe_mag = 0.f;
}
}
static void upmix_1_0(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float mag, *dst;
dst = (float *)s->output->extended_data[0];
mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
dst[2 * n ] = mag * cosf(c_phase);
dst[2 * n + 1] = mag * sinf(c_phase);
}
static void upmix_stereo(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float l_mag, r_mag, *dstl, *dstr;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
}
static void upmix_2_1(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, l_mag, r_mag, *dstl, *dstr, *dstlfe;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstlfe = (float *)s->output->extended_data[2];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total, s->lfe_mode);
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
}
static void upmix_3_0(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float l_mag, r_mag, c_mag, *dstc, *dstl, *dstr;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstc = (float *)s->output->extended_data[2];
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
}
static void upmix_3_1(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstlfe;
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];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total, s->lfe_mode);
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
}
static void upmix_3_1_surround(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float c_mag,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, l_mag, r_mag, *dstc, *dstl, *dstr, *dstlfe;
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];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &c_mag, s->lfe_mode);
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
}
static void upmix_4_0(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstc = (float *)s->output->extended_data[2];
dstb = (float *)s->output->extended_data[3];
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
b_mag = powf(1.f - fabsf(x), s->bc_x) * powf((1.f - y) * .5f, s->bc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstb[2 * n ] = b_mag * cosf(c_phase);
dstb[2 * n + 1] = b_mag * sinf(c_phase);
}
static void upmix_4_1(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb, *dstlfe;
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];
dstb = (float *)s->output->extended_data[4];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total, s->lfe_mode);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
b_mag = powf(1.f - fabsf(x), s->bc_x) * powf((1.f - y) * .5f, s->bc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstb[2 * n ] = b_mag * cosf(c_phase);
dstb[2 * n + 1] = b_mag * sinf(c_phase);
}
static void upmix_5_0_back(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstc = (float *)s->output->extended_data[2];
dstls = (float *)s->output->extended_data[3];
dstrs = (float *)s->output->extended_data[4];
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
static void upmix_5_1_back(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlfe;
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];
dstls = (float *)s->output->extended_data[4];
dstrs = (float *)s->output->extended_data[5];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total, s->lfe_mode);
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
static void upmix_6_0(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float l_mag, r_mag, ls_mag, rs_mag, c_mag, b_mag, *dstc, *dstb, *dstl, *dstr, *dstls, *dstrs;
dstl = (float *)s->output->extended_data[0];
dstr = (float *)s->output->extended_data[1];
dstc = (float *)s->output->extended_data[2];
dstb = (float *)s->output->extended_data[3];
dstls = (float *)s->output->extended_data[4];
dstrs = (float *)s->output->extended_data[5];
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
b_mag = powf(1.f - fabsf(x), s->bc_x) * powf((1.f - y) * .5f, s->bc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
dstb[2 * n ] = b_mag * cosf(c_phase);
dstb[2 * n + 1] = b_mag * sinf(c_phase);
}
static void upmix_6_1(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, b_mag, *dstc, *dstb, *dstl, *dstr, *dstls, *dstrs, *dstlfe;
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];
dstb = (float *)s->output->extended_data[4];
dstls = (float *)s->output->extended_data[5];
dstrs = (float *)s->output->extended_data[6];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total, s->lfe_mode);
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
b_mag = powf(1.f - fabsf(x), s->bc_x) * powf((1.f - y) * .5f, s->bc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
dstb[2 * n ] = b_mag * cosf(c_phase);
dstb[2 * n + 1] = b_mag * sinf(c_phase);
}
static void upmix_5_1_back_surround(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float c_mag,
float mag_total,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float lfe_mag, l_mag, r_mag, *dstc, *dstl, *dstr, *dstlfe;
float ls_mag, rs_mag, *dstls, *dstrs;
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];
dstls = (float *)s->output->extended_data[4];
dstrs = (float *)s->output->extended_data[5];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &c_mag, s->lfe_mode);
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
static void upmix_5_1_back_2_1(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float lfe_re,
float lfe_im,
float x, float y,
int n)
{
AudioSurroundContext *s = ctx->priv;
float c_mag, l_mag, r_mag, *dstc, *dstl, *dstr, *dstlfe;
float ls_mag, rs_mag, *dstls, *dstrs;
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];
dstls = (float *)s->output->extended_data[4];
dstrs = (float *)s->output->extended_data[5];
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_re;
dstlfe[2 * n + 1] = lfe_im;
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
static void upmix_7_0(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
float l_mag, r_mag, ls_mag, rs_mag, c_mag, lb_mag, rb_mag;
float *dstc, *dstl, *dstr, *dstls, *dstrs, *dstlb, *dstrb;
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];
dstlb = (float *)s->output->extended_data[3];
dstrb = (float *)s->output->extended_data[4];
dstls = (float *)s->output->extended_data[5];
dstrs = (float *)s->output->extended_data[6];
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
lb_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rb_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->sl_x) * powf(1.f - fabsf(y), s->sl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->sr_x) * powf(1.f - fabsf(y), s->sr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlb[2 * n ] = lb_mag * cosf(l_phase);
dstlb[2 * n + 1] = lb_mag * sinf(l_phase);
dstrb[2 * n ] = rb_mag * cosf(r_phase);
dstrb[2 * n + 1] = rb_mag * sinf(r_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
static void upmix_7_1(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
int n)
{
float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_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];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total, s->lfe_mode);
c_mag = powf(1.f - fabsf(x), s->fc_x) * powf((y + 1.f) * .5f, s->fc_y) * mag_total;
l_mag = powf(.5f * ( x + 1.f), s->fl_x) * powf((y + 1.f) * .5f, s->fl_y) * mag_total;
r_mag = powf(.5f * (-x + 1.f), s->fr_x) * powf((y + 1.f) * .5f, s->fr_y) * mag_total;
lb_mag = powf(.5f * ( x + 1.f), s->bl_x) * powf(1.f - ((y + 1.f) * .5f), s->bl_y) * mag_total;
rb_mag = powf(.5f * (-x + 1.f), s->br_x) * powf(1.f - ((y + 1.f) * .5f), s->br_y) * mag_total;
ls_mag = powf(.5f * ( x + 1.f), s->sl_x) * powf(1.f - fabsf(y), s->sl_y) * mag_total;
rs_mag = powf(.5f * (-x + 1.f), s->sr_x) * powf(1.f - fabsf(y), s->sr_y) * mag_total;
dstl[2 * n ] = l_mag * cosf(l_phase);
dstl[2 * n + 1] = l_mag * sinf(l_phase);
dstr[2 * n ] = r_mag * cosf(r_phase);
dstr[2 * n + 1] = r_mag * sinf(r_phase);
dstc[2 * n ] = c_mag * cosf(c_phase);
dstc[2 * n + 1] = c_mag * sinf(c_phase);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
dstlb[2 * n ] = lb_mag * cosf(l_phase);
dstlb[2 * n + 1] = lb_mag * sinf(l_phase);
dstrb[2 * n ] = rb_mag * cosf(r_phase);
dstrb[2 * n + 1] = rb_mag * sinf(r_phase);
dstls[2 * n ] = ls_mag * cosf(l_phase);
dstls[2 * n + 1] = ls_mag * sinf(l_phase);
dstrs[2 * n ] = rs_mag * cosf(r_phase);
dstrs[2 * n + 1] = rs_mag * sinf(r_phase);
}
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) * 0.5;
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, &mag_total, s->lfe_mode);
fl_mag = powf(.5f * (xl + 1.f), s->fl_x) * powf((yl + 1.f) * .5f, s->fl_y) * mag_totall;
fr_mag = powf(.5f * (xr + 1.f), s->fr_x) * powf((yr + 1.f) * .5f, s->fr_y) * mag_totalr;
lb_mag = powf(.5f * (-xl + 1.f), s->bl_x) * powf((yl + 1.f) * .5f, s->bl_y) * mag_totall;
rb_mag = powf(.5f * (-xr + 1.f), s->br_x) * powf((yr + 1.f) * .5f, s->br_y) * mag_totalr;
ls_mag = powf(1.f - fabsf(xl), s->sl_x) * powf((yl + 1.f) * .5f, s->sl_y) * mag_totall;
rs_mag = powf(1.f - fabsf(xr), s->sr_x) * powf((yr + 1.f) * .5f, s->sr_y) * 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->fl_x) * powf((yl + 1.f) * .5f, s->fl_y) * mag_totall;
fr_mag = powf(.5f * (xr + 1.f), s->fr_x) * powf((yr + 1.f) * .5f, s->fr_y) * mag_totalr;
lb_mag = powf(.5f * (-xl + 1.f), s->bl_x) * powf((yl + 1.f) * .5f, s->bl_y) * mag_totall;
rb_mag = powf(.5f * (-xr + 1.f), s->br_x) * powf((yr + 1.f) * .5f, s->br_y) * mag_totalr;
ls_mag = powf(1.f - fabsf(xl), s->sl_x) * powf((yl + 1.f) * .5f, s->sl_y) * mag_totall;
rs_mag = powf(1.f - fabsf(xr), s->sr_x) * powf((yr + 1.f) * .5f, s->sr_y) * 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;
float *srcl, *srcr;
int n;
srcl = (float *)s->input->extended_data[0];
srcr = (float *)s->input->extended_data[1];
for (n = 0; n < s->buf_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 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 = mag_sum < 0.000001 ? FFDIFFSIGN(l_mag, r_mag) : (l_mag - r_mag) / mag_sum;
float mag_total = hypotf(l_mag, r_mag);
float x, y;
if (phase_dif > M_PI)
phase_dif = 2 * M_PI - phase_dif;
stereo_position(mag_dif, phase_dif, &x, &y);
stereo_transform(&x, &y, s->angle);
s->upmix_stereo(ctx, l_phase, r_phase, c_phase, mag_total, x, y, n);
}
}
static void filter_surround(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
float *srcl, *srcr, *srcc;
int n;
srcl = (float *)s->input->extended_data[0];
srcr = (float *)s->input->extended_data[1];
srcc = (float *)s->input->extended_data[2];
for (n = 0; n < s->buf_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_mag = hypotf(c_re, c_im);
float c_phase = atan2f(c_im, c_re);
float l_mag = hypotf(l_re, l_im);
float r_mag = hypotf(r_re, r_im);
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 = mag_sum < 0.000001 ? FFDIFFSIGN(l_mag, r_mag) : (l_mag - r_mag) / mag_sum;
float mag_total = hypotf(l_mag, r_mag);
float x, y;
if (phase_dif > M_PI)
phase_dif = 2 * M_PI - phase_dif;
stereo_position(mag_dif, phase_dif, &x, &y);
stereo_transform(&x, &y, s->angle);
s->upmix_3_0(ctx, l_phase, r_phase, c_phase, c_mag, mag_total, x, y, n);
}
}
static void filter_2_1(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
float *srcl, *srcr, *srclfe;
int n;
srcl = (float *)s->input->extended_data[0];
srcr = (float *)s->input->extended_data[1];
srclfe = (float *)s->input->extended_data[2];
for (n = 0; n < s->buf_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 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 = mag_sum < 0.000001 ? FFDIFFSIGN(l_mag, r_mag) : (l_mag - r_mag) / mag_sum;
float mag_total = hypotf(l_mag, r_mag);
float x, y;
if (phase_dif > M_PI)
phase_dif = 2 * M_PI - phase_dif;
stereo_position(mag_dif, phase_dif, &x, &y);
stereo_transform(&x, &y, s->angle);
s->upmix_2_1(ctx, l_phase, r_phase, c_phase, mag_total, lfe_re, lfe_im, x, y, n);
}
}
static void filter_5_0_side(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
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 < s->buf_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 < 0.000001 ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum;
float mag_difr = magr_sum < 0.000001 ? 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_PI)
phase_difl = 2 * M_PI - phase_difl;
if (phase_difr > M_PI)
phase_difr = 2 * M_PI - 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;
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 < s->buf_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 < 0.000001 ? FFDIFFSIGN(fl_mag, sl_mag) : (fl_mag - sl_mag) / magl_sum;
float mag_difr = magr_sum < 0.000001 ? 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_PI)
phase_difl = 2 * M_PI - phase_difl;
if (phase_difr > M_PI)
phase_difr = 2 * M_PI - 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;
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 < s->buf_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 < 0.000001 ? FFDIFFSIGN(fl_mag, bl_mag) : (fl_mag - bl_mag) / magl_sum;
float mag_difr = magr_sum < 0.000001 ? 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_PI)
phase_difl = 2 * M_PI - phase_difl;
if (phase_difr > M_PI)
phase_difr = 2 * M_PI - 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 av_cold int init(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
float overlap;
int i;
if (!(s->out_channel_layout = av_get_channel_layout(s->out_channel_layout_str))) {
av_log(ctx, AV_LOG_ERROR, "Error parsing output channel layout '%s'.\n",
s->out_channel_layout_str);
return AVERROR(EINVAL);
}
if (!(s->in_channel_layout = av_get_channel_layout(s->in_channel_layout_str))) {
av_log(ctx, AV_LOG_ERROR, "Error parsing input channel layout '%s'.\n",
s->in_channel_layout_str);
return AVERROR(EINVAL);
}
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);
}
switch (s->in_channel_layout) {
case AV_CH_LAYOUT_STEREO:
s->filter = filter_stereo;
switch (s->out_channel_layout) {
case AV_CH_LAYOUT_MONO:
s->upmix_stereo = upmix_1_0;
break;
case AV_CH_LAYOUT_STEREO:
s->upmix_stereo = upmix_stereo;
break;
case AV_CH_LAYOUT_2POINT1:
s->upmix_stereo = upmix_2_1;
break;
case AV_CH_LAYOUT_SURROUND:
s->upmix_stereo = upmix_3_0;
break;
case AV_CH_LAYOUT_3POINT1:
s->upmix_stereo = upmix_3_1;
break;
case AV_CH_LAYOUT_4POINT0:
s->upmix_stereo = upmix_4_0;
break;
case AV_CH_LAYOUT_4POINT1:
s->upmix_stereo = upmix_4_1;
break;
case AV_CH_LAYOUT_5POINT0_BACK:
s->upmix_stereo = upmix_5_0_back;
break;
case AV_CH_LAYOUT_5POINT1_BACK:
s->upmix_stereo = upmix_5_1_back;
break;
case AV_CH_LAYOUT_6POINT0:
s->upmix_stereo = upmix_6_0;
break;
case AV_CH_LAYOUT_6POINT1:
s->upmix_stereo = upmix_6_1;
break;
case AV_CH_LAYOUT_7POINT0:
s->upmix_stereo = upmix_7_0;
break;
case AV_CH_LAYOUT_7POINT1:
s->upmix_stereo = upmix_7_1;
break;
default:
goto fail;
}
break;
case AV_CH_LAYOUT_2POINT1:
s->filter = filter_2_1;
switch (s->out_channel_layout) {
case AV_CH_LAYOUT_5POINT1_BACK:
s->upmix_2_1 = upmix_5_1_back_2_1;
break;
default:
goto fail;
}
break;
case AV_CH_LAYOUT_SURROUND:
s->filter = filter_surround;
switch (s->out_channel_layout) {
case AV_CH_LAYOUT_3POINT1:
s->upmix_3_0 = upmix_3_1_surround;
break;
case AV_CH_LAYOUT_5POINT1_BACK:
s->upmix_3_0 = upmix_5_1_back_surround;
break;
default:
goto fail;
}
break;
case AV_CH_LAYOUT_5POINT0:
s->filter = filter_5_0_side;
switch (s->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 (s->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 (s->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_log(ctx, AV_LOG_ERROR, "Unsupported upmix: '%s' -> '%s'.\n",
s->in_channel_layout_str, s->out_channel_layout_str);
return AVERROR(EINVAL);
}
s->buf_size = 1 << av_log2(s->win_size);
s->pts = AV_NOPTS_VALUE;
s->window_func_lut = av_calloc(s->buf_size, sizeof(*s->window_func_lut));
if (!s->window_func_lut)
return AVERROR(ENOMEM);
generate_window_func(s->window_func_lut, s->buf_size, s->win_func, &overlap);
if (s->overlap == 1)
s->overlap = overlap;
for (i = 0; i < s->buf_size; i++)
s->window_func_lut[i] = sqrtf(s->window_func_lut[i] / s->buf_size);
s->hop_size = s->buf_size * (1. - s->overlap);
if (s->hop_size <= 0)
return AVERROR(EINVAL);
if (s->all_x >= 0.f)
s->fc_x = s->fl_x = s->fr_x = s->bc_x = s->sl_x = s->sr_x = s->bl_x = s->br_x = s->all_x;
if (s->all_y >= 0.f)
s->fc_y = s->fl_y = s->fr_y = s->bc_y = s->sl_y = s->sr_y = s->bl_y = s->br_y = s->all_y;
return 0;
}
static int fft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
AudioSurroundContext *s = ctx->priv;
const float level_in = s->input_levels[ch];
float *dst;
int n;
memset(s->input->extended_data[ch] + s->buf_size * sizeof(float), 0, s->buf_size * sizeof(float));
dst = (float *)s->input->extended_data[ch];
for (n = 0; n < s->buf_size; n++) {
dst[n] *= s->window_func_lut[n] * level_in;
}
av_rdft_calc(s->rdft[ch], (float *)s->input->extended_data[ch]);
return 0;
}
static int ifft_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
{
AudioSurroundContext *s = ctx->priv;
const float level_out = s->output_levels[ch];
AVFrame *out = arg;
float *dst, *ptr;
int n;
av_rdft_calc(s->irdft[ch], (float *)s->output->extended_data[ch]);
dst = (float *)s->output->extended_data[ch];
ptr = (float *)s->overlap_buffer->extended_data[ch];
memmove(s->overlap_buffer->extended_data[ch],
s->overlap_buffer->extended_data[ch] + s->hop_size * sizeof(float),
s->buf_size * sizeof(float));
memset(s->overlap_buffer->extended_data[ch] + s->buf_size * sizeof(float),
0, s->hop_size * sizeof(float));
for (n = 0; n < s->buf_size; n++) {
ptr[n] += dst[n] * s->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 filter_frame(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
AVFrame *out;
int ret;
ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
if (ret < 0)
return ret;
ff_filter_execute(ctx, fft_channel, NULL, NULL, inlink->channels);
s->filter(ctx);
out = ff_get_audio_buffer(outlink, s->hop_size);
if (!out)
return AVERROR(ENOMEM);
ff_filter_execute(ctx, ifft_channel, out, NULL, outlink->channels);
out->pts = s->pts;
if (s->pts != AV_NOPTS_VALUE)
s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
av_audio_fifo_drain(s->fifo, FFMIN(av_audio_fifo_size(s->fifo), s->hop_size));
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);
if (!s->eof && av_audio_fifo_size(s->fifo) < s->buf_size) {
ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
if (ret > 0) {
ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
in->nb_samples);
if (ret >= 0 && s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;
av_frame_free(&in);
if (ret < 0)
return ret;
}
}
if ((av_audio_fifo_size(s->fifo) >= s->buf_size) ||
(av_audio_fifo_size(s->fifo) > 0 && s->eof)) {
ret = filter_frame(inlink);
if (av_audio_fifo_size(s->fifo) >= s->buf_size)
ff_filter_set_ready(ctx, 100);
return ret;
}
if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF) {
s->eof = 1;
if (av_audio_fifo_size(s->fifo) >= 0) {
ff_filter_set_ready(ctx, 100);
return 0;
}
}
}
if (s->eof && av_audio_fifo_size(s->fifo) <= 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}
if (!s->eof)
FF_FILTER_FORWARD_WANTED(outlink, inlink);
return FFERROR_NOT_READY;
}
static av_cold void uninit(AVFilterContext *ctx)
{
AudioSurroundContext *s = ctx->priv;
int ch;
av_frame_free(&s->input);
av_frame_free(&s->output);
av_frame_free(&s->overlap_buffer);
for (ch = 0; ch < s->nb_in_channels; ch++) {
av_rdft_end(s->rdft[ch]);
}
for (ch = 0; ch < s->nb_out_channels; ch++) {
av_rdft_end(s->irdft[ch]);
}
av_freep(&s->input_levels);
av_freep(&s->output_levels);
av_freep(&s->rdft);
av_freep(&s->irdft);
av_audio_fifo_free(s->fifo);
av_freep(&s->window_func_lut);
}
#define OFFSET(x) offsetof(AudioSurroundContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
static const AVOption surround_options[] = {
{ "chl_out", "set output channel layout", OFFSET(out_channel_layout_str), AV_OPT_TYPE_STRING, {.str="5.1"}, 0, 0, FLAGS },
{ "chl_in", "set input channel layout", OFFSET(in_channel_layout_str), AV_OPT_TYPE_STRING, {.str="stereo"},0, 0, FLAGS },
{ "level_in", "set input level", OFFSET(level_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "level_out", "set output level", OFFSET(level_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "lfe", "output LFE", OFFSET(output_lfe), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, FLAGS },
{ "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, FLAGS, "lfe_mode" },
{ "add", "just add LFE channel", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 1, FLAGS, "lfe_mode" },
{ "sub", "substract LFE channel with others", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 1, FLAGS, "lfe_mode" },
{ "angle", "set soundfield transform angle", OFFSET(angle), AV_OPT_TYPE_FLOAT, {.dbl=90}, 0, 360, FLAGS },
{ "fc_in", "set front center channel input level", OFFSET(fc_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "fc_out", "set front center channel output level", OFFSET(fc_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "fl_in", "set front left channel input level", OFFSET(fl_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "fl_out", "set front left channel output level", OFFSET(fl_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "fr_in", "set front right channel input level", OFFSET(fr_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "fr_out", "set front right channel output level", OFFSET(fr_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "sl_in", "set side left channel input level", OFFSET(sl_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "sl_out", "set side left channel output level", OFFSET(sl_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "sr_in", "set side right channel input level", OFFSET(sr_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "sr_out", "set side right channel output level", OFFSET(sr_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "bl_in", "set back left channel input level", OFFSET(bl_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "bl_out", "set back left channel output level", OFFSET(bl_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "br_in", "set back right channel input level", OFFSET(br_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "br_out", "set back right channel output level", OFFSET(br_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "bc_in", "set back center channel input level", OFFSET(bc_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "bc_out", "set back center channel output level", OFFSET(bc_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "lfe_in", "set lfe channel input level", OFFSET(lfe_in), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "lfe_out", "set lfe channel output level", OFFSET(lfe_out), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 10, FLAGS },
{ "allx", "set all channel's x spread", OFFSET(all_x), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, FLAGS },
{ "ally", "set all channel's y spread", OFFSET(all_y), AV_OPT_TYPE_FLOAT, {.dbl=-1}, -1, 15, FLAGS },
{ "fcx", "set front center channel x spread", OFFSET(fc_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "flx", "set front left channel x spread", OFFSET(fl_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "frx", "set front right channel x spread", OFFSET(fr_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "blx", "set back left channel x spread", OFFSET(bl_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "brx", "set back right channel x spread", OFFSET(br_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "slx", "set side left channel x spread", OFFSET(sl_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "srx", "set side right channel x spread", OFFSET(sr_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "bcx", "set back center channel x spread", OFFSET(bc_x), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "fcy", "set front center channel y spread", OFFSET(fc_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "fly", "set front left channel y spread", OFFSET(fl_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "fry", "set front right channel y spread", OFFSET(fr_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "bly", "set back left channel y spread", OFFSET(bl_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "bry", "set back right channel y spread", OFFSET(br_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "sly", "set side left channel y spread", OFFSET(sl_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "sry", "set side right channel y spread", OFFSET(sr_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "bcy", "set back center channel y spread", OFFSET(bc_y), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 15, FLAGS },
{ "win_size", "set window size", OFFSET(win_size), AV_OPT_TYPE_INT, {.i64 = 4096}, 1024, 65536, FLAGS },
{ "win_func", "set window function", OFFSET(win_func), AV_OPT_TYPE_INT, {.i64 = WFUNC_HANNING}, 0, NB_WFUNC-1, FLAGS, "win_func" },
{ "rect", "Rectangular", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_RECT}, 0, 0, FLAGS, "win_func" },
{ "bartlett", "Bartlett", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BARTLETT}, 0, 0, FLAGS, "win_func" },
{ "hann", "Hann", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING}, 0, 0, FLAGS, "win_func" },
{ "hanning", "Hanning", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HANNING}, 0, 0, FLAGS, "win_func" },
{ "hamming", "Hamming", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_HAMMING}, 0, 0, FLAGS, "win_func" },
{ "blackman", "Blackman", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BLACKMAN}, 0, 0, FLAGS, "win_func" },
{ "welch", "Welch", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_WELCH}, 0, 0, FLAGS, "win_func" },
{ "flattop", "Flat-top", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_FLATTOP}, 0, 0, FLAGS, "win_func" },
{ "bharris", "Blackman-Harris", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHARRIS}, 0, 0, FLAGS, "win_func" },
{ "bnuttall", "Blackman-Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BNUTTALL}, 0, 0, FLAGS, "win_func" },
{ "bhann", "Bartlett-Hann", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BHANN}, 0, 0, FLAGS, "win_func" },
{ "sine", "Sine", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_SINE}, 0, 0, FLAGS, "win_func" },
{ "nuttall", "Nuttall", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_NUTTALL}, 0, 0, FLAGS, "win_func" },
{ "lanczos", "Lanczos", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_LANCZOS}, 0, 0, FLAGS, "win_func" },
{ "gauss", "Gauss", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_GAUSS}, 0, 0, FLAGS, "win_func" },
{ "tukey", "Tukey", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_TUKEY}, 0, 0, FLAGS, "win_func" },
{ "dolph", "Dolph-Chebyshev", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_DOLPH}, 0, 0, FLAGS, "win_func" },
{ "cauchy", "Cauchy", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_CAUCHY}, 0, 0, FLAGS, "win_func" },
{ "parzen", "Parzen", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_PARZEN}, 0, 0, FLAGS, "win_func" },
{ "poisson", "Poisson", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_POISSON}, 0, 0, FLAGS, "win_func" },
{ "bohman", "Bohman", 0, AV_OPT_TYPE_CONST, {.i64=WFUNC_BOHMAN}, 0, 0, FLAGS, "win_func" },
{ "overlap", "set window overlap", OFFSET(overlap), AV_OPT_TYPE_FLOAT, {.dbl=0.5}, 0, 1, FLAGS },
{ 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,
};
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