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FFmpeg/libavfilter/af_surround.c
Paul B Mahol 200f497997 avfilter/af_surround: add missing error check 
Signed-off-by: Paul B Mahol <onemda@gmail.com>
2017-11-20 13:54:04 +01:00

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/*
* Copyright (c) 2017 Paul B Mahol
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/audio_fifo.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavcodec/avfft.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.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 lfe_in;
float lfe_out;
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;
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]->in_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]->out_channel_layouts);
if (ret)
return ret;
formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}
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_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_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_position(float a, float p, float *x, float *y)
{
*x = av_clipf(a+FFMAX(0, sinf(p-M_PI_2))*FFDIFFSIGN(a,0), -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)
{
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;
*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 = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * 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 = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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);
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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 = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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);
c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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);
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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)
{
float b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb;
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];
dstb = (float *)s->output->extended_data[3];
c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
b_mag = sqrtf(1.f - fabsf(x)) * ((1.f - y) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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)
{
float lfe_mag, b_mag, l_mag, r_mag, c_mag, *dstc, *dstl, *dstr, *dstb, *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];
dstb = (float *)s->output->extended_data[4];
get_lfe(s->output_lfe, n, s->lowcut, s->highcut, &lfe_mag, &mag_total);
dstlfe[2 * n ] = lfe_mag * cosf(c_phase);
dstlfe[2 * n + 1] = lfe_mag * sinf(c_phase);
c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
b_mag = sqrtf(1.f - fabsf(x)) * ((1.f - y) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * 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)
{
float l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs;
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];
dstls = (float *)s->output->extended_data[3];
dstrs = (float *)s->output->extended_data[4];
c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * 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)
{
float lfe_mag, l_mag, r_mag, ls_mag, rs_mag, c_mag, *dstc, *dstl, *dstr, *dstls, *dstrs, *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];
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);
c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * 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_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);
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * 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 = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * 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 = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
lb_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
rb_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - fabsf(y)) * mag_total;
rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - fabsf(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);
c_mag = sqrtf(1.f - fabsf(x)) * ((y + 1.f) * .5f) * mag_total;
l_mag = sqrtf(.5f * ( x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
r_mag = sqrtf(.5f * (-x + 1.f)) * ((y + 1.f) * .5f) * mag_total;
lb_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
rb_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - ((y + 1.f) * .5f)) * mag_total;
ls_mag = sqrtf(.5f * ( x + 1.f)) * (1.f - fabsf(y)) * mag_total;
rs_mag = sqrtf(.5f * (-x + 1.f)) * (1.f - fabsf(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);
fl_mag = sqrtf(.5f * (xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
fr_mag = sqrtf(.5f * (xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
lb_mag = sqrtf(.5f * (-xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
rb_mag = sqrtf(.5f * (-xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
ls_mag = sqrtf(1.f - fabsf(xl)) * ((yl + 1.f) * .5f) * mag_totall;
rs_mag = sqrtf(1.f - fabsf(xr)) * ((yr + 1.f) * .5f) * 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 = sqrtf(.5f * (xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
fr_mag = sqrtf(.5f * (xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
lb_mag = sqrtf(.5f * (-xl + 1.f)) * ((yl + 1.f) * .5f) * mag_totall;
rb_mag = sqrtf(.5f * (-xr + 1.f)) * ((yr + 1.f) * .5f) * mag_totalr;
ls_mag = sqrtf(1.f - fabsf(xl)) * ((yl + 1.f) * .5f) * mag_totall;
rs_mag = sqrtf(1.f - fabsf(xr)) * ((yr + 1.f) * .5f) * 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_dif = (l_mag - r_mag) / (l_mag + r_mag);
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);
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_dif = (l_mag - r_mag) / (l_mag + r_mag);
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);
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_dif = (l_mag - r_mag) / (l_mag + r_mag);
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);
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 mag_difl = (fl_mag - sl_mag) / (fl_mag + sl_mag);
float mag_difr = (fr_mag - sr_mag) / (fr_mag + sr_mag);
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 mag_difl = (fl_mag - sl_mag) / (fl_mag + sl_mag);
float mag_difr = (fr_mag - sr_mag) / (fr_mag + sr_mag);
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 mag_difl = (fl_mag - bl_mag) / (fl_mag + bl_mag);
float mag_difr = (fr_mag - br_mag) / (fr_mag + br_mag);
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 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_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 = 4096;
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);
for (i = 0; i < s->buf_size; i++)
s->window_func_lut[i] = sqrtf(0.5 * (1 - cosf(2 * M_PI * i / s->buf_size)) / s->buf_size);
overlap = .5;
s->hop_size = s->buf_size * (1. - overlap);
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, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
AudioSurroundContext *s = ctx->priv;
int ret;
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;
while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
AVFrame *out;
ret = av_audio_fifo_peek(s->fifo, (void **)s->input->extended_data, s->buf_size);
if (ret < 0)
return ret;
ctx->internal->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);
ctx->internal->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, s->hop_size);
ret = ff_filter_frame(outlink, out);
if (ret < 0)
return ret;
}
return 0;
}
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 },
{ "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 },
{ "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 },
{ NULL }
};
AVFILTER_DEFINE_CLASS(surround);
static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
},
{ NULL }
};
AVFilter ff_af_surround = {
.name = "surround",
.description = NULL_IF_CONFIG_SMALL("Apply audio surround upmix filter."),
.query_formats = query_formats,
.priv_size = sizeof(AudioSurroundContext),
.priv_class = &surround_class,
.init = init,
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS,
};
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