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avfilter: add speechnorm filter

tags/n4.4
Paul B Mahol 5 years ago
parent
46e362b765
commit
9f20e5d281
6 changed files with 646 additions and 1 deletions
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  1. +1
    -0
      Changelog
  2. +63
    -0
      doc/filters.texi
  3. +1
    -0
      libavfilter/Makefile
  4. +579
    -0
      libavfilter/af_speechnorm.c
  5. +1
    -0
      libavfilter/allfilters.c
  6. +1
    -1
      libavfilter/version.h

+ 1
- 0
Changelog View File

@@ -45,6 +45,7 @@ version <next>:
- AMV muxer
- NVDEC AV1 hwaccel
- DXVA2/D3D11VA hardware accelerated AV1 decoding
- speechnorm filter


version 4.3:


+ 63
- 0
doc/filters.texi View File

@@ -5276,6 +5276,69 @@ and also with custom gain:
@end example
@end itemize

@section speechnorm
Speech Normalizer.

This filter expands or compresses each half-cycle of audio samples
(local set of samples all above or all below zero and between two nearest zero crossings) depending
on threshold value, so audio reaches target peak value under conditions controlled by below options.

The filter accepts the following options:

@table @option
@item peak, p
Set the expansion target peak value. This specifies the highest allowed absolute amplitude
level for the normalized audio input. Default value is 0.95. Allowed range is from 0.0 to 1.0.

@item expansion, e
Set the maximum expansion factor. Allowed range is from 1.0 to 50.0. Default value is 2.0.
This option controls maximum local half-cycle of samples expansion. The maximum expansion
would be such that local peak value reaches target peak value but never to surpass it and that
ratio between new and previous peak value does not surpass this option value.

@item compression, c
Set the maximum compression factor. Allowed range is from 1.0 to 50.0. Default value is 2.0.
This option controls maximum local half-cycle of samples compression. This option is used
only if @option{threshold} option is set to value greater than 0.0, then in such cases
when local peak is lower or same as value set by @option{threshold} all samples belonging to
that peak's half-cycle will be compressed by current compression factor.

@item threshold, t
Set the threshold value. Default value is 0.0. Allowed range is from 0.0 to 1.0.
This option specifies which half-cycles of samples will be compressed and which will be expanded.
Any half-cycle samples with their local peak value below or same as this option value will be
compressed by current compression factor, otherwise, if greater than threshold value they will be
expanded with expansion factor so that it could reach peak target value but never surpass it.

@item raise, r
Set the expansion raising amount per each half-cycle of samples. Default value is 0.001.
Allowed range is from 0.0 to 1.0. This controls how fast expansion factor is raised per
each new half-cycle until it reaches @option{expansion} value.
Setting this options too high may lead to distortions.

@item fall, f
Set the compression raising amount per each half-cycle of samples. Default value is 0.001.
Allowed range is from 0.0 to 1.0. This controls how fast compression factor is raised per
each new half-cycle until it reaches @option{compression} value.

@item channels, h
Specify which channels to filter, by default all available channels are filtered.

@item invert, i
Enable inverted filtering, by default is disabled. This inverts interpretation of @option{threshold}
option. When enabled any half-cycle of samples with their local peak value below or same as
@option{threshold} option will be expanded otherwise it will be compressed.

@item link, l
Link channels when calculating gain applied to each filtered channel sample, by default is disabled.
When disabled each filtered channel gain calculation is independent, otherwise when this option
is enabled the minimum of all possible gains for each filtered channel is used.
@end table

@subsection Commands

This filter supports the all above options as @ref{commands}.

@section stereotools

This filter has some handy utilities to manage stereo signals, for converting


+ 1
- 0
libavfilter/Makefile View File

@@ -137,6 +137,7 @@ OBJS-$(CONFIG_SIDECHAINGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_SILENCEREMOVE_FILTER) += af_silenceremove.o
OBJS-$(CONFIG_SOFALIZER_FILTER) += af_sofalizer.o
OBJS-$(CONFIG_SPEECHNORM_FILTER) += af_speechnorm.o
OBJS-$(CONFIG_STEREOTOOLS_FILTER) += af_stereotools.o
OBJS-$(CONFIG_STEREOWIDEN_FILTER) += af_stereowiden.o
OBJS-$(CONFIG_SUPEREQUALIZER_FILTER) += af_superequalizer.o


+ 579
- 0
libavfilter/af_speechnorm.c View File

@@ -0,0 +1,579 @@
/*
* Copyright (c) 2020 Paul B Mahol
*
* Speech Normalizer
*
* 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
*/

/**
* @file
* Speech Normalizer
*/

#include <float.h>

#include "libavutil/avassert.h"
#include "libavutil/opt.h"

#define FF_BUFQUEUE_SIZE (1024)
#include "bufferqueue.h"

#include "audio.h"
#include "avfilter.h"
#include "filters.h"
#include "internal.h"

#define MAX_ITEMS 882000
#define MIN_PEAK (1. / 32768.)

typedef struct PeriodItem {
int size;
int type;
double max_peak;
} PeriodItem;

typedef struct ChannelContext {
int state;
int bypass;
PeriodItem pi[MAX_ITEMS];
double gain_state;
double pi_max_peak;
int pi_start;
int pi_end;
int pi_size;
} ChannelContext;

typedef struct SpeechNormalizerContext {
const AVClass *class;

double peak_value;
double max_expansion;
double max_compression;
double threshold_value;
double raise_amount;
double fall_amount;
uint64_t channels;
int invert;
int link;

ChannelContext *cc;
double prev_gain;

int max_period;
int eof;
int64_t pts;

struct FFBufQueue queue;

void (*analyze_channel)(AVFilterContext *ctx, ChannelContext *cc,
const uint8_t *srcp, int nb_samples);
void (*filter_channels[2])(AVFilterContext *ctx,
AVFrame *in, int nb_samples);
} SpeechNormalizerContext;

#define OFFSET(x) offsetof(SpeechNormalizerContext, x)
#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM

static const AVOption speechnorm_options[] = {
{ "peak", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl=0.95}, 0.0, 1.0, FLAGS },
{ "p", "set the peak value", OFFSET(peak_value), AV_OPT_TYPE_DOUBLE, {.dbl=0.95}, 0.0, 1.0, FLAGS },
{ "expansion", "set the max expansion factor", OFFSET(max_expansion), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
{ "e", "set the max expansion factor", OFFSET(max_expansion), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
{ "compression", "set the max compression factor", OFFSET(max_compression), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
{ "c", "set the max compression factor", OFFSET(max_compression), AV_OPT_TYPE_DOUBLE, {.dbl=2.0}, 1.0, 50.0, FLAGS },
{ "threshold", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0.0, 1.0, FLAGS },
{ "t", "set the threshold value", OFFSET(threshold_value), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0.0, 1.0, FLAGS },
{ "raise", "set the expansion raising amount", OFFSET(raise_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
{ "r", "set the expansion raising amount", OFFSET(raise_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
{ "fall", "set the compression raising amount", OFFSET(fall_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
{ "f", "set the compression raising amount", OFFSET(fall_amount), AV_OPT_TYPE_DOUBLE, {.dbl=0.001}, 0.0, 1.0, FLAGS },
{ "channels", "set channels to filter", OFFSET(channels), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS },
{ "h", "set channels to filter", OFFSET(channels), AV_OPT_TYPE_CHANNEL_LAYOUT, {.i64=-1}, INT64_MIN, INT64_MAX, FLAGS },
{ "invert", "set inverted filtering", OFFSET(invert), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{ "i", "set inverted filtering", OFFSET(invert), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{ "link", "set linked channels filtering", OFFSET(link), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{ "l", "set linked channels filtering", OFFSET(link), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, FLAGS },
{ NULL }
};

AVFILTER_DEFINE_CLASS(speechnorm);

static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats;
AVFilterChannelLayouts *layouts;
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;

layouts = ff_all_channel_counts();
if (!layouts)
return AVERROR(ENOMEM);
ret = ff_set_common_channel_layouts(ctx, layouts);
if (ret < 0)
return ret;

formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return ret;

formats = ff_all_samplerates();
if (!formats)
return AVERROR(ENOMEM);
return ff_set_common_samplerates(ctx, formats);
}

static int get_pi_samples(PeriodItem *pi, int start, int end, int remain)
{
int sum;

if (pi[start].type == 0)
return remain;

sum = remain;
while (start != end) {
start++;
if (start >= MAX_ITEMS)
start = 0;
if (pi[start].type == 0)
break;
av_assert0(pi[start].size > 0);
sum += pi[start].size;
}

return sum;
}

static int available_samples(AVFilterContext *ctx)
{
SpeechNormalizerContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
int min_pi_nb_samples;

min_pi_nb_samples = get_pi_samples(s->cc[0].pi, s->cc[0].pi_start, s->cc[0].pi_end, s->cc[0].pi_size);
for (int ch = 1; ch < inlink->channels && min_pi_nb_samples > 0; ch++) {
ChannelContext *cc = &s->cc[ch];

min_pi_nb_samples = FFMIN(min_pi_nb_samples, get_pi_samples(cc->pi, cc->pi_start, cc->pi_end, cc->pi_size));
}

return min_pi_nb_samples;
}

static void consume_pi(ChannelContext *cc, int nb_samples)
{
if (cc->pi_size >= nb_samples) {
cc->pi_size -= nb_samples;
} else {
av_assert0(0);
}
}

static double next_gain(AVFilterContext *ctx, double pi_max_peak, int bypass, double state)
{
SpeechNormalizerContext *s = ctx->priv;
const double expansion = FFMIN(s->max_expansion, s->peak_value / pi_max_peak);
const double compression = 1. / s->max_compression;
const int type = s->invert ? pi_max_peak <= s->threshold_value : pi_max_peak >= s->threshold_value;

if (bypass) {
return 1.;
} else if (type) {
return FFMIN(expansion, state + s->raise_amount);
} else {
return FFMIN(expansion, FFMAX(compression, state - s->fall_amount));
}
}

static void next_pi(AVFilterContext *ctx, ChannelContext *cc, int bypass)
{
av_assert0(cc->pi_size >= 0);
if (cc->pi_size == 0) {
SpeechNormalizerContext *s = ctx->priv;
int start = cc->pi_start;

av_assert0(cc->pi[start].size > 0);
av_assert0(cc->pi[start].type > 0 || s->eof);
cc->pi_size = cc->pi[start].size;
cc->pi_max_peak = cc->pi[start].max_peak;
av_assert0(cc->pi_start != cc->pi_end || s->eof);
start++;
if (start >= MAX_ITEMS)
start = 0;
cc->pi_start = start;
cc->gain_state = next_gain(ctx, cc->pi_max_peak, bypass, cc->gain_state);
}
}

static double min_gain(AVFilterContext *ctx, ChannelContext *cc, int max_size)
{
SpeechNormalizerContext *s = ctx->priv;
double min_gain = s->max_expansion;
double gain_state = cc->gain_state;
int size = cc->pi_size;
int idx = cc->pi_start;

min_gain = FFMIN(min_gain, gain_state);
while (size <= max_size) {
if (idx == cc->pi_end)
break;
gain_state = next_gain(ctx, cc->pi[idx].max_peak, 0, gain_state);
min_gain = FFMIN(min_gain, gain_state);
size += cc->pi[idx].size;
idx++;
if (idx >= MAX_ITEMS)
idx = 0;
}

return min_gain;
}

#define ANALYZE_CHANNEL(name, ptype, zero) \
static void analyze_channel_## name (AVFilterContext *ctx, ChannelContext *cc, \
const uint8_t *srcp, int nb_samples) \
{ \
SpeechNormalizerContext *s = ctx->priv; \
const ptype *src = (const ptype *)srcp; \
int n = 0; \
\
if (cc->state < 0) \
cc->state = src[0] >= zero; \
\
while (n < nb_samples) { \
if ((cc->state != (src[n] >= zero)) || \
(cc->pi[cc->pi_end].size > s->max_period)) { \
double max_peak = cc->pi[cc->pi_end].max_peak; \
int state = cc->state; \
cc->state = src[n] >= zero; \
av_assert0(cc->pi[cc->pi_end].size > 0); \
if (cc->pi[cc->pi_end].max_peak >= MIN_PEAK || \
cc->pi[cc->pi_end].size > s->max_period) { \
cc->pi[cc->pi_end].type = 1; \
cc->pi_end++; \
if (cc->pi_end >= MAX_ITEMS) \
cc->pi_end = 0; \
if (cc->state != state) \
cc->pi[cc->pi_end].max_peak = DBL_MIN; \
else \
cc->pi[cc->pi_end].max_peak = max_peak; \
cc->pi[cc->pi_end].type = 0; \
cc->pi[cc->pi_end].size = 0; \
av_assert0(cc->pi_end != cc->pi_start); \
} \
} \
\
if (cc->state) { \
while (src[n] >= zero) { \
cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, src[n]); \
cc->pi[cc->pi_end].size++; \
n++; \
if (n >= nb_samples) \
break; \
} \
} else { \
while (src[n] < zero) { \
cc->pi[cc->pi_end].max_peak = FFMAX(cc->pi[cc->pi_end].max_peak, -src[n]); \
cc->pi[cc->pi_end].size++; \
n++; \
if (n >= nb_samples) \
break; \
} \
} \
} \
}

ANALYZE_CHANNEL(dbl, double, 0.0)
ANALYZE_CHANNEL(flt, float, 0.f)

#define FILTER_CHANNELS(name, ptype) \
static void filter_channels_## name (AVFilterContext *ctx, \
AVFrame *in, int nb_samples) \
{ \
SpeechNormalizerContext *s = ctx->priv; \
AVFilterLink *inlink = ctx->inputs[0]; \
\
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
ptype *dst = (ptype *)in->extended_data[ch]; \
const int bypass = !(av_channel_layout_extract_channel(inlink->channel_layout, ch) & s->channels); \
int n = 0; \
\
while (n < nb_samples) { \
ptype gain; \
int size; \
\
next_pi(ctx, cc, bypass); \
size = FFMIN(nb_samples - n, cc->pi_size); \
av_assert0(size > 0); \
gain = cc->gain_state; \
consume_pi(cc, size); \
for (int i = n; i < n + size; i++) \
dst[i] *= gain; \
n += size; \
} \
} \
}

FILTER_CHANNELS(dbl, double)
FILTER_CHANNELS(flt, float)

static double lerp(double min, double max, double mix)
{
return min + (max - min) * mix;
}

#define FILTER_LINK_CHANNELS(name, ptype) \
static void filter_link_channels_## name (AVFilterContext *ctx, \
AVFrame *in, int nb_samples) \
{ \
SpeechNormalizerContext *s = ctx->priv; \
AVFilterLink *inlink = ctx->inputs[0]; \
int n = 0; \
\
while (n < nb_samples) { \
int min_size = nb_samples - n; \
int max_size = 1; \
ptype gain = s->max_expansion; \
\
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
\
cc->bypass = !(av_channel_layout_extract_channel(inlink->channel_layout, ch) & s->channels); \
\
next_pi(ctx, cc, cc->bypass); \
min_size = FFMIN(min_size, cc->pi_size); \
max_size = FFMAX(max_size, cc->pi_size); \
} \
\
av_assert0(min_size > 0); \
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
\
if (cc->bypass) \
continue; \
gain = FFMIN(gain, min_gain(ctx, cc, max_size)); \
} \
\
for (int ch = 0; ch < inlink->channels; ch++) { \
ChannelContext *cc = &s->cc[ch]; \
ptype *dst = (ptype *)in->extended_data[ch]; \
\
consume_pi(cc, min_size); \
if (cc->bypass) \
continue; \
\
for (int i = n; i < n + min_size; i++) { \
ptype g = lerp(s->prev_gain, gain, (i - n) / (double)min_size); \
dst[i] *= g; \
} \
} \
\
s->prev_gain = gain; \
n += min_size; \
} \
}

FILTER_LINK_CHANNELS(dbl, double)
FILTER_LINK_CHANNELS(flt, float)

static int filter_frame(AVFilterContext *ctx)
{
SpeechNormalizerContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFilterLink *inlink = ctx->inputs[0];
int ret;

while (s->queue.available > 0) {
int min_pi_nb_samples;
AVFrame *in;

in = ff_bufqueue_peek(&s->queue, 0);
if (!in)
break;

min_pi_nb_samples = available_samples(ctx);
if (min_pi_nb_samples < in->nb_samples && !s->eof)
break;

in = ff_bufqueue_get(&s->queue);

av_frame_make_writable(in);

s->filter_channels[s->link](ctx, in, in->nb_samples);

s->pts = in->pts + in->nb_samples;

return ff_filter_frame(outlink, in);
}

for (int f = 0; f < ff_inlink_queued_frames(inlink); f++) {
AVFrame *in;

ret = ff_inlink_consume_frame(inlink, &in);
if (ret < 0)
return ret;
if (ret == 0)
break;

ff_bufqueue_add(ctx, &s->queue, in);

for (int ch = 0; ch < inlink->channels; ch++) {
ChannelContext *cc = &s->cc[ch];

s->analyze_channel(ctx, cc, in->extended_data[ch], in->nb_samples);
}
}

return 1;
}

static int activate(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
SpeechNormalizerContext *s = ctx->priv;
int ret, status;
int64_t pts;

FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);

ret = filter_frame(ctx);
if (ret <= 0)
return ret;

if (!s->eof && ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF)
s->eof = 1;
}

if (s->eof && ff_inlink_queued_samples(inlink) == 0 &&
s->queue.available == 0) {
ff_outlink_set_status(outlink, AVERROR_EOF, s->pts);
return 0;
}

if (s->queue.available > 0) {
AVFrame *in = ff_bufqueue_peek(&s->queue, 0);
const int nb_samples = available_samples(ctx);

if (nb_samples >= in->nb_samples || s->eof) {
ff_filter_set_ready(ctx, 10);
return 0;
}
}

FF_FILTER_FORWARD_WANTED(outlink, inlink);

return FFERROR_NOT_READY;
}

static int config_input(AVFilterLink *inlink)
{
AVFilterContext *ctx = inlink->dst;
SpeechNormalizerContext *s = ctx->priv;

s->max_period = inlink->sample_rate / 10;

s->prev_gain = 1.;
s->cc = av_calloc(inlink->channels, sizeof(*s->cc));
if (!s->cc)
return AVERROR(ENOMEM);

for (int ch = 0; ch < inlink->channels; ch++) {
ChannelContext *cc = &s->cc[ch];

cc->state = -1;
cc->gain_state = 1.;
}

switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP:
s->analyze_channel = analyze_channel_flt;
s->filter_channels[0] = filter_channels_flt;
s->filter_channels[1] = filter_link_channels_flt;
break;
case AV_SAMPLE_FMT_DBLP:
s->analyze_channel = analyze_channel_dbl;
s->filter_channels[0] = filter_channels_dbl;
s->filter_channels[1] = filter_link_channels_dbl;
break;
default:
av_assert0(0);
}

return 0;
}

static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
char *res, int res_len, int flags)
{
SpeechNormalizerContext *s = ctx->priv;
int link = s->link;
int ret;

ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
if (ret < 0)
return ret;
if (link != s->link)
s->prev_gain = 1.;

return 0;
}

static av_cold void uninit(AVFilterContext *ctx)
{
SpeechNormalizerContext *s = ctx->priv;

ff_bufqueue_discard_all(&s->queue);
av_freep(&s->cc);
}

static const AVFilterPad inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
},
{ NULL }
};

static const AVFilterPad outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};

AVFilter ff_af_speechnorm = {
.name = "speechnorm",
.description = NULL_IF_CONFIG_SMALL("Speech Normalizer."),
.query_formats = query_formats,
.priv_size = sizeof(SpeechNormalizerContext),
.priv_class = &speechnorm_class,
.activate = activate,
.uninit = uninit,
.inputs = inputs,
.outputs = outputs,
.process_command = process_command,
};

+ 1
- 0
libavfilter/allfilters.c View File

@@ -131,6 +131,7 @@ extern AVFilter ff_af_sidechaingate;
extern AVFilter ff_af_silencedetect;
extern AVFilter ff_af_silenceremove;
extern AVFilter ff_af_sofalizer;
extern AVFilter ff_af_speechnorm;
extern AVFilter ff_af_stereotools;
extern AVFilter ff_af_stereowiden;
extern AVFilter ff_af_superequalizer;


+ 1
- 1
libavfilter/version.h View File

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

#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 89
#define LIBAVFILTER_VERSION_MINOR 90
#define LIBAVFILTER_VERSION_MICRO 100




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