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avfilter: add audio surround upmixer 

Signed-off-by: Paul B Mahol <onemda@gmail.com>
tags/n3.4
Paul B Mahol 8 years ago
parent
2934a10f2e
commit
dc72d1dde9
6 changed files with 869 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +30
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +835
    -0
      libavfilter/af_surround.c
  5. +1
    -0
      libavfilter/allfilters.c
  6. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

@@ -16,6 +16,7 @@ version <next>:
- spec compliant VP9 muxing support in MP4
- remove the libnut muxer/demuxer wrappers
- remove the libschroedinger encoder/decoder wrappers
- surround audio filter

version 3.3:
- CrystalHD decoder moved to new decode API


+ 30
- 0
doc/filters.texi View File

@@ -3792,6 +3792,36 @@ channels. Default is 0.3.
Set level of input signal of original channel. Default is 0.8.
@end table

@section surround
Apply audio surround upmix filter.

This filter allows to produce multichannel output from stereo audio stream.

The filter accepts the following options:

@table @option
@item chl_out
Set output channel layout. By default, this is @var{5.1}.

See @ref{channel layout syntax,,the Channel Layout section in the ffmpeg-utils(1) manual,ffmpeg-utils}
for the required syntax.

@item level_in
Set input volume level. By default, this is @var{1}.

@item level_out
Set output volume level. By default, this is @var{1}.

@item lfe
Enable LFE channel output if output channel layout has it. By default, this is enabled.

@item lfe_low
Set LFE low cut off frequency. By default, this is @var{128} Hz.

@item lfe_high
Set LFE high cut off frequency. By default, this is @var{256} Hz.
@end table

@section treble

Boost or cut treble (upper) frequencies of the audio using a two-pole


+ 1
- 0
libavfilter/Makefile View File

@@ -108,6 +108,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o
OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
OBJS-$(CONFIG_SURROUND_FILTER) += af_surround.o
OBJS-$(CONFIG_TREBLE_FILTER) += af_biquads.o
OBJS-$(CONFIG_TREMOLO_FILTER) += af_tremolo.o
OBJS-$(CONFIG_VIBRATO_FILTER) += af_vibrato.o generate_wave_table.o


+ 835
- 0
libavfilter/af_surround.c View File

@@ -0,0 +1,835 @@
/*
* 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;
float level_in;
float level_out;
int output_lfe;
int lowcutf;
int highcutf;

float lowcut;
float highcut;

uint64_t out_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 (*upmix)(AVFilterContext *ctx,
float l_phase,
float r_phase,
float c_phase,
float mag_total,
float x, float y,
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, AV_CH_LAYOUT_STEREO);
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 = 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 = 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_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_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 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 channel layout '%s'.\n",
s->out_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->out_channel_layout) {
case AV_CH_LAYOUT_MONO:
s->upmix = upmix_1_0;
break;
case AV_CH_LAYOUT_STEREO:
s->upmix = upmix_stereo;
break;
case AV_CH_LAYOUT_2POINT1:
s->upmix = upmix_2_1;
break;
case AV_CH_LAYOUT_SURROUND:
s->upmix = upmix_3_0;
break;
case AV_CH_LAYOUT_3POINT1:
s->upmix = upmix_3_1;
break;
case AV_CH_LAYOUT_4POINT0:
s->upmix = upmix_4_0;
break;
case AV_CH_LAYOUT_4POINT1:
s->upmix = upmix_4_1;
break;
case AV_CH_LAYOUT_5POINT0_BACK:
s->upmix = upmix_5_0_back;
break;
case AV_CH_LAYOUT_5POINT1_BACK:
s->upmix = upmix_5_1_back;
break;
case AV_CH_LAYOUT_7POINT0:
s->upmix = upmix_7_0;
break;
case AV_CH_LAYOUT_7POINT1:
s->upmix = upmix_7_1;
break;
default:
av_log(ctx, AV_LOG_ERROR, "Unsupported output channel layout '%s'.\n",
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->level_in;
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->level_out;
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;

av_audio_fifo_write(s->fifo, (void **)in->extended_data,
in->nb_samples);

if (s->pts == AV_NOPTS_VALUE)
s->pts = in->pts;

av_frame_free(&in);

while (av_audio_fifo_size(s->fifo) >= s->buf_size) {
float *srcl, *srcr;
AVFrame *out;
int n, ret;

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);

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(ctx, l_phase, r_phase, c_phase, mag_total, x, y, n);
}

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->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 },
{ "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 },
{ 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,
};

+ 1
- 0
libavfilter/allfilters.c View File

@@ -121,6 +121,7 @@ static void register_all(void)
REGISTER_FILTER(SOFALIZER, sofalizer, af);
REGISTER_FILTER(STEREOTOOLS, stereotools, af);
REGISTER_FILTER(STEREOWIDEN, stereowiden, af);
REGISTER_FILTER(SURROUND, surround, af);
REGISTER_FILTER(TREBLE, treble, af);
REGISTER_FILTER(TREMOLO, tremolo, af);
REGISTER_FILTER(VIBRATO, vibrato, af);


+ 1
- 1
libavfilter/version.h View File

@@ -30,7 +30,7 @@
#include "libavutil/version.h"

#define LIBAVFILTER_VERSION_MAJOR 6
#define LIBAVFILTER_VERSION_MINOR 90
#define LIBAVFILTER_VERSION_MINOR 91
#define LIBAVFILTER_VERSION_MICRO 100

#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \


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