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avfilter: add aderivative and aintegral filter
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Signed-off-by: Paul B Mahol <[email protected]>
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richardpl committed May 16, 2018
1 parent 64f59a2 commit 4e816b5
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Showing 6 changed files with 219 additions and 1 deletion.
1 change: 1 addition & 0 deletions Changelog
Original file line number Diff line number Diff line change
Expand Up @@ -6,6 +6,7 @@ version <next>:
- tmix filter
- amplify filter
- fftdnoiz filter
- aderivative and aintegral audio filters


version 4.0:
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6 changes: 6 additions & 0 deletions doc/filters.texi
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Expand Up @@ -585,6 +585,12 @@ adelay=0|500S|700S
@end example
@end itemize

@section aderivative, aintegral

Compute derivative/integral of audio stream.

Applying both filters one after another produces original audio.

@section aecho

Apply echoing to the input audio.
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2 changes: 2 additions & 0 deletions libavfilter/Makefile
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Expand Up @@ -35,6 +35,7 @@ OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o
OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o
OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
Expand All @@ -44,6 +45,7 @@ OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o
OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o
OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
OBJS-$(CONFIG_ALIMITER_FILTER) += af_alimiter.o
OBJS-$(CONFIG_ALLPASS_FILTER) += af_biquads.o
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207 changes: 207 additions & 0 deletions libavfilter/af_aderivative.c
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@@ -0,0 +1,207 @@
/*
* 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 "audio.h"
#include "avfilter.h"
#include "internal.h"

typedef struct ADerivativeContext {
const AVClass *class;
AVFrame *prev;
void (*filter)(void **dst, void **prv, const void **src,
int nb_samples, int channels);
} ADerivativeContext;

static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
static const enum AVSampleFormat derivative_sample_fmts[] = {
AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
static const enum AVSampleFormat integral_sample_fmts[] = {
AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
AV_SAMPLE_FMT_NONE
};
int ret;

formats = ff_make_format_list(strcmp(ctx->filter->name, "aintegral") ?
derivative_sample_fmts : integral_sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret < 0)
return 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_all_samplerates();
return ff_set_common_samplerates(ctx, formats);
}

#define DERIVATIVE(name, type) \
static void aderivative_## name ##p(void **d, void **p, const void **s, \
int nb_samples, int channels) \
{ \
int n, c; \
\
for (c = 0; c < channels; c++) { \
const type *src = s[c]; \
type *dst = d[c]; \
type *prv = p[c]; \
\
for (n = 0; n < nb_samples; n++) { \
const type current = src[n]; \
\
dst[n] = current - prv[0]; \
prv[0] = current; \
} \
} \
}

DERIVATIVE(flt, float)
DERIVATIVE(dbl, double)
DERIVATIVE(s16, int16_t)
DERIVATIVE(s32, int32_t)

#define INTEGRAL(name, type) \
static void aintegral_## name ##p(void **d, void **p, const void **s, \
int nb_samples, int channels) \
{ \
int n, c; \
\
for (c = 0; c < channels; c++) { \
const type *src = s[c]; \
type *dst = d[c]; \
type *prv = p[c]; \
\
for (n = 0; n < nb_samples; n++) { \
const type current = src[n]; \
\
dst[n] = current + prv[0]; \
prv[0] = dst[n]; \
} \
} \
}

INTEGRAL(flt, float)
INTEGRAL(dbl, double)

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

switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break;
case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break;
case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break;
}

if (strcmp(ctx->filter->name, "aintegral"))
return 0;

switch (inlink->format) {
case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break;
case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break;
}

return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
ADerivativeContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);

if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);

if (!s->prev) {
s->prev = ff_get_audio_buffer(inlink, 1);
if (!s->prev) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
}

s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
in->nb_samples, in->channels);

av_frame_free(&in);
return ff_filter_frame(outlink, out);
}

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

av_frame_free(&s->prev);
}

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

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

AVFilter ff_af_aderivative = {
.name = "aderivative",
.description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."),
.query_formats = query_formats,
.priv_size = sizeof(ADerivativeContext),
.uninit = uninit,
.inputs = aderivative_inputs,
.outputs = aderivative_outputs,
};

AVFilter ff_af_aintegral = {
.name = "aintegral",
.description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
.query_formats = query_formats,
.priv_size = sizeof(ADerivativeContext),
.uninit = uninit,
.inputs = aderivative_inputs,
.outputs = aderivative_outputs,
};
2 changes: 2 additions & 0 deletions libavfilter/allfilters.c
Original file line number Diff line number Diff line change
Expand Up @@ -30,6 +30,7 @@ extern AVFilter ff_af_acopy;
extern AVFilter ff_af_acrossfade;
extern AVFilter ff_af_acrusher;
extern AVFilter ff_af_adelay;
extern AVFilter ff_af_aderivative;
extern AVFilter ff_af_aecho;
extern AVFilter ff_af_aemphasis;
extern AVFilter ff_af_aeval;
Expand All @@ -39,6 +40,7 @@ extern AVFilter ff_af_afir;
extern AVFilter ff_af_aformat;
extern AVFilter ff_af_agate;
extern AVFilter ff_af_aiir;
extern AVFilter ff_af_aintegral;
extern AVFilter ff_af_ainterleave;
extern AVFilter ff_af_alimiter;
extern AVFilter ff_af_allpass;
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2 changes: 1 addition & 1 deletion libavfilter/version.h
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Expand Up @@ -30,7 +30,7 @@
#include "libavutil/version.h"

#define LIBAVFILTER_VERSION_MAJOR 7
#define LIBAVFILTER_VERSION_MINOR 22
#define LIBAVFILTER_VERSION_MINOR 23
#define LIBAVFILTER_VERSION_MICRO 100

#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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