mirror of
https://github.com/FFmpeg/FFmpeg.git
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avfilter: add aderivative and aintegral filter
Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
parent
64f59a21b3
commit
4e816b5491
@ -6,6 +6,7 @@ version <next>:
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- tmix filter
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- amplify filter
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- fftdnoiz filter
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- aderivative and aintegral audio filters
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version 4.0:
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@ -585,6 +585,12 @@ adelay=0|500S|700S
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@end example
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@end itemize
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@section aderivative, aintegral
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Compute derivative/integral of audio stream.
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Applying both filters one after another produces original audio.
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@section aecho
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Apply echoing to the input audio.
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@ -35,6 +35,7 @@ OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o
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OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o
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OBJS-$(CONFIG_ACRUSHER_FILTER) += af_acrusher.o
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OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o
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OBJS-$(CONFIG_ADERIVATIVE_FILTER) += af_aderivative.o
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OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o
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OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o
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OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o
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@ -44,6 +45,7 @@ OBJS-$(CONFIG_AFIR_FILTER) += af_afir.o
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OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o
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OBJS-$(CONFIG_AGATE_FILTER) += af_agate.o
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OBJS-$(CONFIG_AIIR_FILTER) += af_aiir.o
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OBJS-$(CONFIG_AINTEGRAL_FILTER) += af_aderivative.o
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OBJS-$(CONFIG_AINTERLEAVE_FILTER) += f_interleave.o
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OBJS-$(CONFIG_ALIMITER_FILTER) += af_alimiter.o
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OBJS-$(CONFIG_ALLPASS_FILTER) += af_biquads.o
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207
libavfilter/af_aderivative.c
Normal file
207
libavfilter/af_aderivative.c
Normal file
@ -0,0 +1,207 @@
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/*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "audio.h"
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#include "avfilter.h"
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#include "internal.h"
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typedef struct ADerivativeContext {
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const AVClass *class;
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AVFrame *prev;
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void (*filter)(void **dst, void **prv, const void **src,
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int nb_samples, int channels);
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} ADerivativeContext;
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layouts = NULL;
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static const enum AVSampleFormat derivative_sample_fmts[] = {
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AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_FLTP,
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AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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static const enum AVSampleFormat integral_sample_fmts[] = {
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AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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formats = ff_make_format_list(strcmp(ctx->filter->name, "aintegral") ?
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derivative_sample_fmts : integral_sample_fmts);
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if (!formats)
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return AVERROR(ENOMEM);
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ret = ff_set_common_formats(ctx, formats);
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if (ret < 0)
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return ret;
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layouts = ff_all_channel_counts();
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if (!layouts)
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return AVERROR(ENOMEM);
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ret = ff_set_common_channel_layouts(ctx, layouts);
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if (ret < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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#define DERIVATIVE(name, type) \
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static void aderivative_## name ##p(void **d, void **p, const void **s, \
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int nb_samples, int channels) \
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{ \
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int n, c; \
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\
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for (c = 0; c < channels; c++) { \
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const type *src = s[c]; \
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type *dst = d[c]; \
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type *prv = p[c]; \
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\
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for (n = 0; n < nb_samples; n++) { \
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const type current = src[n]; \
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\
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dst[n] = current - prv[0]; \
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prv[0] = current; \
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} \
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} \
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}
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DERIVATIVE(flt, float)
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DERIVATIVE(dbl, double)
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DERIVATIVE(s16, int16_t)
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DERIVATIVE(s32, int32_t)
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#define INTEGRAL(name, type) \
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static void aintegral_## name ##p(void **d, void **p, const void **s, \
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int nb_samples, int channels) \
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{ \
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int n, c; \
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\
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for (c = 0; c < channels; c++) { \
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const type *src = s[c]; \
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type *dst = d[c]; \
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type *prv = p[c]; \
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\
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for (n = 0; n < nb_samples; n++) { \
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const type current = src[n]; \
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\
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dst[n] = current + prv[0]; \
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prv[0] = dst[n]; \
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} \
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} \
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}
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INTEGRAL(flt, float)
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INTEGRAL(dbl, double)
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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ADerivativeContext *s = ctx->priv;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLTP: s->filter = aderivative_fltp; break;
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case AV_SAMPLE_FMT_DBLP: s->filter = aderivative_dblp; break;
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case AV_SAMPLE_FMT_S32P: s->filter = aderivative_s32p; break;
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case AV_SAMPLE_FMT_S16P: s->filter = aderivative_s16p; break;
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}
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if (strcmp(ctx->filter->name, "aintegral"))
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return 0;
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switch (inlink->format) {
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case AV_SAMPLE_FMT_FLTP: s->filter = aintegral_fltp; break;
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case AV_SAMPLE_FMT_DBLP: s->filter = aintegral_dblp; break;
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}
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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ADerivativeContext *s = ctx->priv;
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AVFilterLink *outlink = ctx->outputs[0];
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AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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if (!s->prev) {
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s->prev = ff_get_audio_buffer(inlink, 1);
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if (!s->prev) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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}
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s->filter((void **)out->extended_data, (void **)s->prev->extended_data, (const void **)in->extended_data,
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in->nb_samples, in->channels);
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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ADerivativeContext *s = ctx->priv;
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av_frame_free(&s->prev);
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}
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static const AVFilterPad aderivative_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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};
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static const AVFilterPad aderivative_outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_aderivative = {
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.name = "aderivative",
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.description = NULL_IF_CONFIG_SMALL("Compute derivative of input audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(ADerivativeContext),
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.uninit = uninit,
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.inputs = aderivative_inputs,
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.outputs = aderivative_outputs,
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};
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AVFilter ff_af_aintegral = {
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.name = "aintegral",
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.description = NULL_IF_CONFIG_SMALL("Compute integral of input audio."),
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.query_formats = query_formats,
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.priv_size = sizeof(ADerivativeContext),
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.uninit = uninit,
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.inputs = aderivative_inputs,
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.outputs = aderivative_outputs,
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};
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@ -30,6 +30,7 @@ extern AVFilter ff_af_acopy;
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extern AVFilter ff_af_acrossfade;
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extern AVFilter ff_af_acrusher;
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extern AVFilter ff_af_adelay;
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extern AVFilter ff_af_aderivative;
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extern AVFilter ff_af_aecho;
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extern AVFilter ff_af_aemphasis;
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extern AVFilter ff_af_aeval;
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@ -39,6 +40,7 @@ extern AVFilter ff_af_afir;
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extern AVFilter ff_af_aformat;
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extern AVFilter ff_af_agate;
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extern AVFilter ff_af_aiir;
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extern AVFilter ff_af_aintegral;
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extern AVFilter ff_af_ainterleave;
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extern AVFilter ff_af_alimiter;
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extern AVFilter ff_af_allpass;
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@ -30,7 +30,7 @@
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#include "libavutil/version.h"
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#define LIBAVFILTER_VERSION_MAJOR 7
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#define LIBAVFILTER_VERSION_MINOR 22
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#define LIBAVFILTER_VERSION_MINOR 23
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#define LIBAVFILTER_VERSION_MICRO 100
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#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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