| Message ID | 20191002151129.25175-1-onemda@gmail.com |
|---|---|
| State | New |
| Headers | show |
On 10/2/2019 12:11 PM, Paul B Mahol wrote: > Signed-off-by: Paul B Mahol <onemda@gmail.com> > --- > doc/filters.texi | 28 ++++++ > libavfilter/Makefile | 1 + > libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++ > libavfilter/allfilters.c | 1 + > 4 files changed, 218 insertions(+) > create mode 100644 libavfilter/af_acomb.c > > diff --git a/doc/filters.texi b/doc/filters.texi > index e46839bfec..9c50b2e4b2 100644 > --- a/doc/filters.texi > +++ b/doc/filters.texi > @@ -355,6 +355,34 @@ build. > > Below is a description of the currently available audio filters. > > +@section acomb > +Apply comb audio filtering. > + > +Amplifies or attenuates certain frequencies by the superposition of a > +delayed version of the original audio signal onto itself. > + > +@table @option > +@item t > +Set comb filtering type. > + > +It accepts the following values: > +@table @option > +@item f > +set feedforward type > +@item b > +set feedback type > +@end table > + > +@item b0 > +Set direct signal gain. Default is 1. Allowed range is from 0 to 1. > + > +@item xM > +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1. > + > +@item M > +Set delay in number of samples. Default is 10. Allowed range is from 1 to 327680. > +@end table > + > @section acompressor > > A compressor is mainly used to reduce the dynamic range of a signal. > diff --git a/libavfilter/Makefile b/libavfilter/Makefile > index 182fe9df4b..d8a16d6e15 100644 > --- a/libavfilter/Makefile > +++ b/libavfilter/Makefile > @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile > > # audio filters > OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o > +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o > OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o > OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o > OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o > diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c > new file mode 100644 > index 0000000000..3b0730c363 > --- /dev/null > +++ b/libavfilter/af_acomb.c > @@ -0,0 +1,188 @@ > +/* > + * 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/opt.h" > +#include "audio.h" > +#include "avfilter.h" > +#include "internal.h" > + > +typedef struct AudioCombContext { > + const AVClass *class; > + > + double b0, xM; > + int t, M; > + > + int head; > + int tail; > + > + AVFrame *delayframe; > + > + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame *out); > +} AudioCombContext; > + > +#define OFFSET(x) offsetof(AudioCombContext, x) > +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM > + > +static const AVOption acomb_options[] = { > + { "t", "set comb filter type", OFFSET(t), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" }, > + { "f", "feedforward", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" }, > + { "b", "feedback", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" }, > + { "b0", "set direct signal gain", OFFSET(b0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A }, > + { "xM", "set delayed line gain", OFFSET(xM), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A }, > + { "M", "set delay in number of samples", OFFSET(M), AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A }, > + { NULL } > +}; > + > +AVFILTER_DEFINE_CLASS(acomb); > + > +static int query_formats(AVFilterContext *ctx) > +{ > + AVFilterFormats *formats = NULL; > + AVFilterChannelLayouts *layouts = NULL; > + static const enum AVSampleFormat sample_fmts[] = { > + AV_SAMPLE_FMT_FLTP, > + AV_SAMPLE_FMT_DBLP, > + AV_SAMPLE_FMT_NONE > + }; > + int 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; > + > + 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 COMB(name, type, dir, t) \ > +static void acomb_## name ## _ ##dir(AudioCombContext *s, \ > + AVFrame *in, AVFrame *out) \ > +{ \ > + const type b0 = s->b0; \ > + const type xM = s->xM; \ > + const int M = s->M; \ > + int head; \ > + \ > + for (int c = 0; c < in->channels; c++) { \ > + const type *src = (const type *)in->extended_data[c]; \ > + type *delay = (type *)s->delayframe->extended_data[c]; \ > + type *dst = (type *)out->extended_data[c]; \ > + \ > + head = s->head; \ > + for (int n = 0; n < in->nb_samples; n++) { \ > + dst[n] = b0 * src[n] + t * xM * delay[head]; \ > + if (t == 1) \ > + delay[head] = src[n]; \ > + else \ > + delay[head] = dst[n]; \ > + head++; \ > + if (head >= M) \ > + head = 0; \ > + } \ > + } \ > + \ > + s->head = head; \ > +} > + > +COMB(fltp, float, f, 1) > +COMB(dblp, double, f, 1) > +COMB(fltp, float, b, -1) > +COMB(dblp, double, b, -1) > + > +static int config_input(AVFilterLink *inlink) > +{ > + AVFilterContext *ctx = inlink->dst; > + AudioCombContext *s = ctx->priv; > + > + s->delayframe = ff_get_audio_buffer(inlink, s->M); You're leaking s->delayframe every time config_input() is called after the first time. > + if (!s->delayframe) > + return AVERROR(ENOMEM); > + > + switch (inlink->format) { > + case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : acomb_fltp_f; break; > + case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : acomb_dblp_f; break; > + } > + > + return 0; > +} > + > +static int filter_frame(AVFilterLink *inlink, AVFrame *in) > +{ > + AVFilterContext *ctx = inlink->dst; > + AudioCombContext *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); > + > + s->filter(s, in, out); > + > + av_frame_free(&in); > + return ff_filter_frame(outlink, out); > +} > + > +static av_cold void uninit(AVFilterContext *ctx) > +{ > + AudioCombContext *s = ctx->priv; > + > + av_frame_free(&s->delayframe); > +} > + > +static const AVFilterPad acomb_inputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + .filter_frame = filter_frame, > + .config_props = config_input, > + }, > + { NULL } > +}; > + > +static const AVFilterPad acomb_outputs[] = { > + { > + .name = "default", > + .type = AVMEDIA_TYPE_AUDIO, > + }, > + { NULL } > +}; > + > +AVFilter ff_af_acomb = { > + .name = "acomb", > + .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."), > + .query_formats = query_formats, > + .priv_size = sizeof(AudioCombContext), > + .priv_class = &acomb_class, > + .uninit = uninit, > + .inputs = acomb_inputs, > + .outputs = acomb_outputs, > +}; > diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c > index 1a26129069..7417f9656d 100644 > --- a/libavfilter/allfilters.c > +++ b/libavfilter/allfilters.c > @@ -24,6 +24,7 @@ > #include "config.h" > > extern AVFilter ff_af_abench; > +extern AVFilter ff_af_acomb; > extern AVFilter ff_af_acompressor; > extern AVFilter ff_af_acontrast; > extern AVFilter ff_af_acopy; >
On 10/2/19, James Almer <jamrial@gmail.com> wrote: > On 10/2/2019 12:11 PM, Paul B Mahol wrote: >> Signed-off-by: Paul B Mahol <onemda@gmail.com> >> --- >> doc/filters.texi | 28 ++++++ >> libavfilter/Makefile | 1 + >> libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++ >> libavfilter/allfilters.c | 1 + >> 4 files changed, 218 insertions(+) >> create mode 100644 libavfilter/af_acomb.c >> >> diff --git a/doc/filters.texi b/doc/filters.texi >> index e46839bfec..9c50b2e4b2 100644 >> --- a/doc/filters.texi >> +++ b/doc/filters.texi >> @@ -355,6 +355,34 @@ build. >> >> Below is a description of the currently available audio filters. >> >> +@section acomb >> +Apply comb audio filtering. >> + >> +Amplifies or attenuates certain frequencies by the superposition of a >> +delayed version of the original audio signal onto itself. >> + >> +@table @option >> +@item t >> +Set comb filtering type. >> + >> +It accepts the following values: >> +@table @option >> +@item f >> +set feedforward type >> +@item b >> +set feedback type >> +@end table >> + >> +@item b0 >> +Set direct signal gain. Default is 1. Allowed range is from 0 to 1. >> + >> +@item xM >> +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1. >> + >> +@item M >> +Set delay in number of samples. Default is 10. Allowed range is from 1 to >> 327680. >> +@end table >> + >> @section acompressor >> >> A compressor is mainly used to reduce the dynamic range of a signal. >> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >> index 182fe9df4b..d8a16d6e15 100644 >> --- a/libavfilter/Makefile >> +++ b/libavfilter/Makefile >> @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile >> >> # audio filters >> OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o >> +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o >> OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o >> OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o >> OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o >> diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c >> new file mode 100644 >> index 0000000000..3b0730c363 >> --- /dev/null >> +++ b/libavfilter/af_acomb.c >> @@ -0,0 +1,188 @@ >> +/* >> + * 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/opt.h" >> +#include "audio.h" >> +#include "avfilter.h" >> +#include "internal.h" >> + >> +typedef struct AudioCombContext { >> + const AVClass *class; >> + >> + double b0, xM; >> + int t, M; >> + >> + int head; >> + int tail; >> + >> + AVFrame *delayframe; >> + >> + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame >> *out); >> +} AudioCombContext; >> + >> +#define OFFSET(x) offsetof(AudioCombContext, x) >> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >> + >> +static const AVOption acomb_options[] = { >> + { "t", "set comb filter type", OFFSET(t), >> AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" }, >> + { "f", "feedforward", 0, >> AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" }, >> + { "b", "feedback", 0, >> AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" }, >> + { "b0", "set direct signal gain", OFFSET(b0), >> AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A }, >> + { "xM", "set delayed line gain", OFFSET(xM), >> AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A }, >> + { "M", "set delay in number of samples", OFFSET(M), >> AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A }, >> + { NULL } >> +}; >> + >> +AVFILTER_DEFINE_CLASS(acomb); >> + >> +static int query_formats(AVFilterContext *ctx) >> +{ >> + AVFilterFormats *formats = NULL; >> + AVFilterChannelLayouts *layouts = NULL; >> + static const enum AVSampleFormat sample_fmts[] = { >> + AV_SAMPLE_FMT_FLTP, >> + AV_SAMPLE_FMT_DBLP, >> + AV_SAMPLE_FMT_NONE >> + }; >> + int 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; >> + >> + 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 COMB(name, type, dir, t) \ >> +static void acomb_## name ## _ ##dir(AudioCombContext *s, \ >> + AVFrame *in, AVFrame *out) \ >> +{ \ >> + const type b0 = s->b0; \ >> + const type xM = s->xM; \ >> + const int M = s->M; \ >> + int head; \ >> + \ >> + for (int c = 0; c < in->channels; c++) { \ >> + const type *src = (const type *)in->extended_data[c]; \ >> + type *delay = (type *)s->delayframe->extended_data[c]; \ >> + type *dst = (type *)out->extended_data[c]; \ >> + \ >> + head = s->head; \ >> + for (int n = 0; n < in->nb_samples; n++) { \ >> + dst[n] = b0 * src[n] + t * xM * delay[head]; \ >> + if (t == 1) \ >> + delay[head] = src[n]; \ >> + else \ >> + delay[head] = dst[n]; \ >> + head++; \ >> + if (head >= M) \ >> + head = 0; \ >> + } \ >> + } \ >> + \ >> + s->head = head; \ >> +} >> + >> +COMB(fltp, float, f, 1) >> +COMB(dblp, double, f, 1) >> +COMB(fltp, float, b, -1) >> +COMB(dblp, double, b, -1) >> + >> +static int config_input(AVFilterLink *inlink) >> +{ >> + AVFilterContext *ctx = inlink->dst; >> + AudioCombContext *s = ctx->priv; >> + >> + s->delayframe = ff_get_audio_buffer(inlink, s->M); > > You're leaking s->delayframe every time config_input() is called after > the first time. Sorry, but since when its ok to call config_input() multiple times? It was never ok, only filter is allowed to call it by itself. > >> + if (!s->delayframe) >> + return AVERROR(ENOMEM); >> + >> + switch (inlink->format) { >> + case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : >> acomb_fltp_f; break; >> + case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : >> acomb_dblp_f; break; >> + } >> + >> + return 0; >> +} >> + >> +static int filter_frame(AVFilterLink *inlink, AVFrame *in) >> +{ >> + AVFilterContext *ctx = inlink->dst; >> + AudioCombContext *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); >> + >> + s->filter(s, in, out); >> + >> + av_frame_free(&in); >> + return ff_filter_frame(outlink, out); >> +} >> + >> +static av_cold void uninit(AVFilterContext *ctx) >> +{ >> + AudioCombContext *s = ctx->priv; >> + >> + av_frame_free(&s->delayframe); >> +} >> + >> +static const AVFilterPad acomb_inputs[] = { >> + { >> + .name = "default", >> + .type = AVMEDIA_TYPE_AUDIO, >> + .filter_frame = filter_frame, >> + .config_props = config_input, >> + }, >> + { NULL } >> +}; >> + >> +static const AVFilterPad acomb_outputs[] = { >> + { >> + .name = "default", >> + .type = AVMEDIA_TYPE_AUDIO, >> + }, >> + { NULL } >> +}; >> + >> +AVFilter ff_af_acomb = { >> + .name = "acomb", >> + .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."), >> + .query_formats = query_formats, >> + .priv_size = sizeof(AudioCombContext), >> + .priv_class = &acomb_class, >> + .uninit = uninit, >> + .inputs = acomb_inputs, >> + .outputs = acomb_outputs, >> +}; >> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c >> index 1a26129069..7417f9656d 100644 >> --- a/libavfilter/allfilters.c >> +++ b/libavfilter/allfilters.c >> @@ -24,6 +24,7 @@ >> #include "config.h" >> >> extern AVFilter ff_af_abench; >> +extern AVFilter ff_af_acomb; >> extern AVFilter ff_af_acompressor; >> extern AVFilter ff_af_acontrast; >> extern AVFilter ff_af_acopy; >> > > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".
On 10/2/2019 12:37 PM, Paul B Mahol wrote: > On 10/2/19, James Almer <jamrial@gmail.com> wrote: >> On 10/2/2019 12:11 PM, Paul B Mahol wrote: >>> Signed-off-by: Paul B Mahol <onemda@gmail.com> >>> --- >>> doc/filters.texi | 28 ++++++ >>> libavfilter/Makefile | 1 + >>> libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++ >>> libavfilter/allfilters.c | 1 + >>> 4 files changed, 218 insertions(+) >>> create mode 100644 libavfilter/af_acomb.c >>> >>> diff --git a/doc/filters.texi b/doc/filters.texi >>> index e46839bfec..9c50b2e4b2 100644 >>> --- a/doc/filters.texi >>> +++ b/doc/filters.texi >>> @@ -355,6 +355,34 @@ build. >>> >>> Below is a description of the currently available audio filters. >>> >>> +@section acomb >>> +Apply comb audio filtering. >>> + >>> +Amplifies or attenuates certain frequencies by the superposition of a >>> +delayed version of the original audio signal onto itself. >>> + >>> +@table @option >>> +@item t >>> +Set comb filtering type. >>> + >>> +It accepts the following values: >>> +@table @option >>> +@item f >>> +set feedforward type >>> +@item b >>> +set feedback type >>> +@end table >>> + >>> +@item b0 >>> +Set direct signal gain. Default is 1. Allowed range is from 0 to 1. >>> + >>> +@item xM >>> +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1. >>> + >>> +@item M >>> +Set delay in number of samples. Default is 10. Allowed range is from 1 to >>> 327680. >>> +@end table >>> + >>> @section acompressor >>> >>> A compressor is mainly used to reduce the dynamic range of a signal. >>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >>> index 182fe9df4b..d8a16d6e15 100644 >>> --- a/libavfilter/Makefile >>> +++ b/libavfilter/Makefile >>> @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile >>> >>> # audio filters >>> OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o >>> +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o >>> OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o >>> OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o >>> OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o >>> diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c >>> new file mode 100644 >>> index 0000000000..3b0730c363 >>> --- /dev/null >>> +++ b/libavfilter/af_acomb.c >>> @@ -0,0 +1,188 @@ >>> +/* >>> + * 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/opt.h" >>> +#include "audio.h" >>> +#include "avfilter.h" >>> +#include "internal.h" >>> + >>> +typedef struct AudioCombContext { >>> + const AVClass *class; >>> + >>> + double b0, xM; >>> + int t, M; >>> + >>> + int head; >>> + int tail; >>> + >>> + AVFrame *delayframe; >>> + >>> + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame >>> *out); >>> +} AudioCombContext; >>> + >>> +#define OFFSET(x) offsetof(AudioCombContext, x) >>> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >>> + >>> +static const AVOption acomb_options[] = { >>> + { "t", "set comb filter type", OFFSET(t), >>> AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" }, >>> + { "f", "feedforward", 0, >>> AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" }, >>> + { "b", "feedback", 0, >>> AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" }, >>> + { "b0", "set direct signal gain", OFFSET(b0), >>> AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A }, >>> + { "xM", "set delayed line gain", OFFSET(xM), >>> AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A }, >>> + { "M", "set delay in number of samples", OFFSET(M), >>> AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A }, >>> + { NULL } >>> +}; >>> + >>> +AVFILTER_DEFINE_CLASS(acomb); >>> + >>> +static int query_formats(AVFilterContext *ctx) >>> +{ >>> + AVFilterFormats *formats = NULL; >>> + AVFilterChannelLayouts *layouts = NULL; >>> + static const enum AVSampleFormat sample_fmts[] = { >>> + AV_SAMPLE_FMT_FLTP, >>> + AV_SAMPLE_FMT_DBLP, >>> + AV_SAMPLE_FMT_NONE >>> + }; >>> + int 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; >>> + >>> + 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 COMB(name, type, dir, t) \ >>> +static void acomb_## name ## _ ##dir(AudioCombContext *s, \ >>> + AVFrame *in, AVFrame *out) \ >>> +{ \ >>> + const type b0 = s->b0; \ >>> + const type xM = s->xM; \ >>> + const int M = s->M; \ >>> + int head; \ >>> + \ >>> + for (int c = 0; c < in->channels; c++) { \ >>> + const type *src = (const type *)in->extended_data[c]; \ >>> + type *delay = (type *)s->delayframe->extended_data[c]; \ >>> + type *dst = (type *)out->extended_data[c]; \ >>> + \ >>> + head = s->head; \ >>> + for (int n = 0; n < in->nb_samples; n++) { \ >>> + dst[n] = b0 * src[n] + t * xM * delay[head]; \ >>> + if (t == 1) \ >>> + delay[head] = src[n]; \ >>> + else \ >>> + delay[head] = dst[n]; \ >>> + head++; \ >>> + if (head >= M) \ >>> + head = 0; \ >>> + } \ >>> + } \ >>> + \ >>> + s->head = head; \ >>> +} >>> + >>> +COMB(fltp, float, f, 1) >>> +COMB(dblp, double, f, 1) >>> +COMB(fltp, float, b, -1) >>> +COMB(dblp, double, b, -1) >>> + >>> +static int config_input(AVFilterLink *inlink) >>> +{ >>> + AVFilterContext *ctx = inlink->dst; >>> + AudioCombContext *s = ctx->priv; >>> + >>> + s->delayframe = ff_get_audio_buffer(inlink, s->M); >> >> You're leaking s->delayframe every time config_input() is called after >> the first time. > > Sorry, but since when its ok to call config_input() multiple times? > It was never ok, only filter is allowed to call it by itself. I see, so it's an init function and not something called per frame. Disregard what i said, then. I'm not familiar with libavfilter internal workings, which is why i assumed it could happen. > >> >>> + if (!s->delayframe) >>> + return AVERROR(ENOMEM); >>> + >>> + switch (inlink->format) { >>> + case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : >>> acomb_fltp_f; break; >>> + case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : >>> acomb_dblp_f; break; >>> + } >>> + >>> + return 0; >>> +} >>> + >>> +static int filter_frame(AVFilterLink *inlink, AVFrame *in) >>> +{ >>> + AVFilterContext *ctx = inlink->dst; >>> + AudioCombContext *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); >>> + >>> + s->filter(s, in, out); >>> + >>> + av_frame_free(&in); >>> + return ff_filter_frame(outlink, out); >>> +} >>> + >>> +static av_cold void uninit(AVFilterContext *ctx) >>> +{ >>> + AudioCombContext *s = ctx->priv; >>> + >>> + av_frame_free(&s->delayframe); >>> +} >>> + >>> +static const AVFilterPad acomb_inputs[] = { >>> + { >>> + .name = "default", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + .filter_frame = filter_frame, >>> + .config_props = config_input, >>> + }, >>> + { NULL } >>> +}; >>> + >>> +static const AVFilterPad acomb_outputs[] = { >>> + { >>> + .name = "default", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + }, >>> + { NULL } >>> +}; >>> + >>> +AVFilter ff_af_acomb = { >>> + .name = "acomb", >>> + .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."), >>> + .query_formats = query_formats, >>> + .priv_size = sizeof(AudioCombContext), >>> + .priv_class = &acomb_class, >>> + .uninit = uninit, >>> + .inputs = acomb_inputs, >>> + .outputs = acomb_outputs, >>> +}; >>> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c >>> index 1a26129069..7417f9656d 100644 >>> --- a/libavfilter/allfilters.c >>> +++ b/libavfilter/allfilters.c >>> @@ -24,6 +24,7 @@ >>> #include "config.h" >>> >>> extern AVFilter ff_af_abench; >>> +extern AVFilter ff_af_acomb; >>> extern AVFilter ff_af_acompressor; >>> extern AVFilter ff_af_acontrast; >>> extern AVFilter ff_af_acopy; >>> >> >> _______________________________________________ >> ffmpeg-devel mailing list >> ffmpeg-devel@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >> >> To unsubscribe, visit link above, or email >> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >
On 10/2/19, Paul B Mahol <onemda@gmail.com> wrote: > On 10/2/19, James Almer <jamrial@gmail.com> wrote: >> On 10/2/2019 12:11 PM, Paul B Mahol wrote: >>> Signed-off-by: Paul B Mahol <onemda@gmail.com> >>> --- >>> doc/filters.texi | 28 ++++++ >>> libavfilter/Makefile | 1 + >>> libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++ >>> libavfilter/allfilters.c | 1 + >>> 4 files changed, 218 insertions(+) >>> create mode 100644 libavfilter/af_acomb.c >>> >>> diff --git a/doc/filters.texi b/doc/filters.texi >>> index e46839bfec..9c50b2e4b2 100644 >>> --- a/doc/filters.texi >>> +++ b/doc/filters.texi >>> @@ -355,6 +355,34 @@ build. >>> >>> Below is a description of the currently available audio filters. >>> >>> +@section acomb >>> +Apply comb audio filtering. >>> + >>> +Amplifies or attenuates certain frequencies by the superposition of a >>> +delayed version of the original audio signal onto itself. >>> + >>> +@table @option >>> +@item t >>> +Set comb filtering type. >>> + >>> +It accepts the following values: >>> +@table @option >>> +@item f >>> +set feedforward type >>> +@item b >>> +set feedback type >>> +@end table >>> + >>> +@item b0 >>> +Set direct signal gain. Default is 1. Allowed range is from 0 to 1. >>> + >>> +@item xM >>> +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1. >>> + >>> +@item M >>> +Set delay in number of samples. Default is 10. Allowed range is from 1 >>> to >>> 327680. >>> +@end table >>> + >>> @section acompressor >>> >>> A compressor is mainly used to reduce the dynamic range of a signal. >>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >>> index 182fe9df4b..d8a16d6e15 100644 >>> --- a/libavfilter/Makefile >>> +++ b/libavfilter/Makefile >>> @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile >>> >>> # audio filters >>> OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o >>> +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o >>> OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o >>> OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o >>> OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o >>> diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c >>> new file mode 100644 >>> index 0000000000..3b0730c363 >>> --- /dev/null >>> +++ b/libavfilter/af_acomb.c >>> @@ -0,0 +1,188 @@ >>> +/* >>> + * 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/opt.h" >>> +#include "audio.h" >>> +#include "avfilter.h" >>> +#include "internal.h" >>> + >>> +typedef struct AudioCombContext { >>> + const AVClass *class; >>> + >>> + double b0, xM; >>> + int t, M; >>> + >>> + int head; >>> + int tail; >>> + >>> + AVFrame *delayframe; >>> + >>> + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame >>> *out); >>> +} AudioCombContext; >>> + >>> +#define OFFSET(x) offsetof(AudioCombContext, x) >>> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >>> + >>> +static const AVOption acomb_options[] = { >>> + { "t", "set comb filter type", OFFSET(t), >>> AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" }, >>> + { "f", "feedforward", 0, >>> AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" }, >>> + { "b", "feedback", 0, >>> AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" }, >>> + { "b0", "set direct signal gain", OFFSET(b0), >>> AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A }, >>> + { "xM", "set delayed line gain", OFFSET(xM), >>> AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A }, >>> + { "M", "set delay in number of samples", OFFSET(M), >>> AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A }, >>> + { NULL } >>> +}; >>> + >>> +AVFILTER_DEFINE_CLASS(acomb); >>> + >>> +static int query_formats(AVFilterContext *ctx) >>> +{ >>> + AVFilterFormats *formats = NULL; >>> + AVFilterChannelLayouts *layouts = NULL; >>> + static const enum AVSampleFormat sample_fmts[] = { >>> + AV_SAMPLE_FMT_FLTP, >>> + AV_SAMPLE_FMT_DBLP, >>> + AV_SAMPLE_FMT_NONE >>> + }; >>> + int 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; >>> + >>> + 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 COMB(name, type, dir, t) \ >>> +static void acomb_## name ## _ ##dir(AudioCombContext *s, \ >>> + AVFrame *in, AVFrame *out) \ >>> +{ \ >>> + const type b0 = s->b0; \ >>> + const type xM = s->xM; \ >>> + const int M = s->M; \ >>> + int head; \ >>> + \ >>> + for (int c = 0; c < in->channels; c++) { \ >>> + const type *src = (const type *)in->extended_data[c]; \ >>> + type *delay = (type *)s->delayframe->extended_data[c]; \ >>> + type *dst = (type *)out->extended_data[c]; \ >>> + \ >>> + head = s->head; \ >>> + for (int n = 0; n < in->nb_samples; n++) { \ >>> + dst[n] = b0 * src[n] + t * xM * delay[head]; \ >>> + if (t == 1) \ >>> + delay[head] = src[n]; \ >>> + else \ >>> + delay[head] = dst[n]; \ >>> + head++; \ >>> + if (head >= M) \ >>> + head = 0; \ >>> + } \ >>> + } \ >>> + \ >>> + s->head = head; \ >>> +} >>> + >>> +COMB(fltp, float, f, 1) >>> +COMB(dblp, double, f, 1) >>> +COMB(fltp, float, b, -1) >>> +COMB(dblp, double, b, -1) >>> + >>> +static int config_input(AVFilterLink *inlink) >>> +{ >>> + AVFilterContext *ctx = inlink->dst; >>> + AudioCombContext *s = ctx->priv; >>> + >>> + s->delayframe = ff_get_audio_buffer(inlink, s->M); >> >> You're leaking s->delayframe every time config_input() is called after >> the first time. > > Sorry, but since when its ok to call config_input() multiple times? > It was never ok, only filter is allowed to call it by itself. Fixed locally, but note that bunch of other filters may need to be changed too. > >> >>> + if (!s->delayframe) >>> + return AVERROR(ENOMEM); >>> + >>> + switch (inlink->format) { >>> + case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : >>> acomb_fltp_f; break; >>> + case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : >>> acomb_dblp_f; break; >>> + } >>> + >>> + return 0; >>> +} >>> + >>> +static int filter_frame(AVFilterLink *inlink, AVFrame *in) >>> +{ >>> + AVFilterContext *ctx = inlink->dst; >>> + AudioCombContext *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); >>> + >>> + s->filter(s, in, out); >>> + >>> + av_frame_free(&in); >>> + return ff_filter_frame(outlink, out); >>> +} >>> + >>> +static av_cold void uninit(AVFilterContext *ctx) >>> +{ >>> + AudioCombContext *s = ctx->priv; >>> + >>> + av_frame_free(&s->delayframe); >>> +} >>> + >>> +static const AVFilterPad acomb_inputs[] = { >>> + { >>> + .name = "default", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + .filter_frame = filter_frame, >>> + .config_props = config_input, >>> + }, >>> + { NULL } >>> +}; >>> + >>> +static const AVFilterPad acomb_outputs[] = { >>> + { >>> + .name = "default", >>> + .type = AVMEDIA_TYPE_AUDIO, >>> + }, >>> + { NULL } >>> +}; >>> + >>> +AVFilter ff_af_acomb = { >>> + .name = "acomb", >>> + .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."), >>> + .query_formats = query_formats, >>> + .priv_size = sizeof(AudioCombContext), >>> + .priv_class = &acomb_class, >>> + .uninit = uninit, >>> + .inputs = acomb_inputs, >>> + .outputs = acomb_outputs, >>> +}; >>> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c >>> index 1a26129069..7417f9656d 100644 >>> --- a/libavfilter/allfilters.c >>> +++ b/libavfilter/allfilters.c >>> @@ -24,6 +24,7 @@ >>> #include "config.h" >>> >>> extern AVFilter ff_af_abench; >>> +extern AVFilter ff_af_acomb; >>> extern AVFilter ff_af_acompressor; >>> extern AVFilter ff_af_acontrast; >>> extern AVFilter ff_af_acopy; >>> >> >> _______________________________________________ >> ffmpeg-devel mailing list >> ffmpeg-devel@ffmpeg.org >> https://ffmpeg.org/mailman/listinfo/ffmpeg-devel >> >> To unsubscribe, visit link above, or email >> ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >
On 10/2/19, James Almer <jamrial@gmail.com> wrote: > On 10/2/2019 12:37 PM, Paul B Mahol wrote: >> On 10/2/19, James Almer <jamrial@gmail.com> wrote: >>> On 10/2/2019 12:11 PM, Paul B Mahol wrote: >>>> Signed-off-by: Paul B Mahol <onemda@gmail.com> >>>> --- >>>> doc/filters.texi | 28 ++++++ >>>> libavfilter/Makefile | 1 + >>>> libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++ >>>> libavfilter/allfilters.c | 1 + >>>> 4 files changed, 218 insertions(+) >>>> create mode 100644 libavfilter/af_acomb.c >>>> >>>> diff --git a/doc/filters.texi b/doc/filters.texi >>>> index e46839bfec..9c50b2e4b2 100644 >>>> --- a/doc/filters.texi >>>> +++ b/doc/filters.texi >>>> @@ -355,6 +355,34 @@ build. >>>> >>>> Below is a description of the currently available audio filters. >>>> >>>> +@section acomb >>>> +Apply comb audio filtering. >>>> + >>>> +Amplifies or attenuates certain frequencies by the superposition of a >>>> +delayed version of the original audio signal onto itself. >>>> + >>>> +@table @option >>>> +@item t >>>> +Set comb filtering type. >>>> + >>>> +It accepts the following values: >>>> +@table @option >>>> +@item f >>>> +set feedforward type >>>> +@item b >>>> +set feedback type >>>> +@end table >>>> + >>>> +@item b0 >>>> +Set direct signal gain. Default is 1. Allowed range is from 0 to 1. >>>> + >>>> +@item xM >>>> +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1. >>>> + >>>> +@item M >>>> +Set delay in number of samples. Default is 10. Allowed range is from 1 >>>> to >>>> 327680. >>>> +@end table >>>> + >>>> @section acompressor >>>> >>>> A compressor is mainly used to reduce the dynamic range of a signal. >>>> diff --git a/libavfilter/Makefile b/libavfilter/Makefile >>>> index 182fe9df4b..d8a16d6e15 100644 >>>> --- a/libavfilter/Makefile >>>> +++ b/libavfilter/Makefile >>>> @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile >>>> >>>> # audio filters >>>> OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o >>>> +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o >>>> OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o >>>> OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o >>>> OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o >>>> diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c >>>> new file mode 100644 >>>> index 0000000000..3b0730c363 >>>> --- /dev/null >>>> +++ b/libavfilter/af_acomb.c >>>> @@ -0,0 +1,188 @@ >>>> +/* >>>> + * 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/opt.h" >>>> +#include "audio.h" >>>> +#include "avfilter.h" >>>> +#include "internal.h" >>>> + >>>> +typedef struct AudioCombContext { >>>> + const AVClass *class; >>>> + >>>> + double b0, xM; >>>> + int t, M; >>>> + >>>> + int head; >>>> + int tail; >>>> + >>>> + AVFrame *delayframe; >>>> + >>>> + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame >>>> *out); >>>> +} AudioCombContext; >>>> + >>>> +#define OFFSET(x) offsetof(AudioCombContext, x) >>>> +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM >>>> + >>>> +static const AVOption acomb_options[] = { >>>> + { "t", "set comb filter type", OFFSET(t), >>>> AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" }, >>>> + { "f", "feedforward", 0, >>>> AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" }, >>>> + { "b", "feedback", 0, >>>> AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" }, >>>> + { "b0", "set direct signal gain", OFFSET(b0), >>>> AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A }, >>>> + { "xM", "set delayed line gain", OFFSET(xM), >>>> AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A }, >>>> + { "M", "set delay in number of samples", OFFSET(M), >>>> AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A }, >>>> + { NULL } >>>> +}; >>>> + >>>> +AVFILTER_DEFINE_CLASS(acomb); >>>> + >>>> +static int query_formats(AVFilterContext *ctx) >>>> +{ >>>> + AVFilterFormats *formats = NULL; >>>> + AVFilterChannelLayouts *layouts = NULL; >>>> + static const enum AVSampleFormat sample_fmts[] = { >>>> + AV_SAMPLE_FMT_FLTP, >>>> + AV_SAMPLE_FMT_DBLP, >>>> + AV_SAMPLE_FMT_NONE >>>> + }; >>>> + int 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; >>>> + >>>> + 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 COMB(name, type, dir, t) \ >>>> +static void acomb_## name ## _ ##dir(AudioCombContext *s, \ >>>> + AVFrame *in, AVFrame *out) \ >>>> +{ \ >>>> + const type b0 = s->b0; \ >>>> + const type xM = s->xM; \ >>>> + const int M = s->M; \ >>>> + int head; \ >>>> + \ >>>> + for (int c = 0; c < in->channels; c++) { \ >>>> + const type *src = (const type *)in->extended_data[c]; \ >>>> + type *delay = (type *)s->delayframe->extended_data[c]; \ >>>> + type *dst = (type *)out->extended_data[c]; \ >>>> + \ >>>> + head = s->head; \ >>>> + for (int n = 0; n < in->nb_samples; n++) { \ >>>> + dst[n] = b0 * src[n] + t * xM * delay[head]; \ >>>> + if (t == 1) \ >>>> + delay[head] = src[n]; \ >>>> + else \ >>>> + delay[head] = dst[n]; \ >>>> + head++; \ >>>> + if (head >= M) \ >>>> + head = 0; \ >>>> + } \ >>>> + } \ >>>> + \ >>>> + s->head = head; \ >>>> +} >>>> + >>>> +COMB(fltp, float, f, 1) >>>> +COMB(dblp, double, f, 1) >>>> +COMB(fltp, float, b, -1) >>>> +COMB(dblp, double, b, -1) >>>> + >>>> +static int config_input(AVFilterLink *inlink) >>>> +{ >>>> + AVFilterContext *ctx = inlink->dst; >>>> + AudioCombContext *s = ctx->priv; >>>> + >>>> + s->delayframe = ff_get_audio_buffer(inlink, s->M); >>> >>> You're leaking s->delayframe every time config_input() is called after >>> the first time. >> >> Sorry, but since when its ok to call config_input() multiple times? >> It was never ok, only filter is allowed to call it by itself. > > I see, so it's an init function and not something called per frame. > Disregard what i said, then. I'm not familiar with libavfilter internal > workings, which is why i assumed it could happen. It actually happens with astreamselect filter. But that filter is not used much.
diff --git a/doc/filters.texi b/doc/filters.texi index e46839bfec..9c50b2e4b2 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -355,6 +355,34 @@ build. Below is a description of the currently available audio filters. +@section acomb +Apply comb audio filtering. + +Amplifies or attenuates certain frequencies by the superposition of a +delayed version of the original audio signal onto itself. + +@table @option +@item t +Set comb filtering type. + +It accepts the following values: +@table @option +@item f +set feedforward type +@item b +set feedback type +@end table + +@item b0 +Set direct signal gain. Default is 1. Allowed range is from 0 to 1. + +@item xM +Set delayed line gain. Default is 0.5. Allowed range is from 0 to 1. + +@item M +Set delay in number of samples. Default is 10. Allowed range is from 1 to 327680. +@end table + @section acompressor A compressor is mainly used to reduce the dynamic range of a signal. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 182fe9df4b..d8a16d6e15 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -31,6 +31,7 @@ include $(SRC_PATH)/libavfilter/dnn/Makefile # audio filters OBJS-$(CONFIG_ABENCH_FILTER) += f_bench.o +OBJS-$(CONFIG_ACOMB_FILTER) += af_acomb.o OBJS-$(CONFIG_ACOMPRESSOR_FILTER) += af_sidechaincompress.o OBJS-$(CONFIG_ACONTRAST_FILTER) += af_acontrast.o OBJS-$(CONFIG_ACOPY_FILTER) += af_acopy.o diff --git a/libavfilter/af_acomb.c b/libavfilter/af_acomb.c new file mode 100644 index 0000000000..3b0730c363 --- /dev/null +++ b/libavfilter/af_acomb.c @@ -0,0 +1,188 @@ +/* + * 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/opt.h" +#include "audio.h" +#include "avfilter.h" +#include "internal.h" + +typedef struct AudioCombContext { + const AVClass *class; + + double b0, xM; + int t, M; + + int head; + int tail; + + AVFrame *delayframe; + + void (*filter)(struct AudioCombContext *s, AVFrame *in, AVFrame *out); +} AudioCombContext; + +#define OFFSET(x) offsetof(AudioCombContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption acomb_options[] = { + { "t", "set comb filter type", OFFSET(t), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, A, "t" }, + { "f", "feedforward", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "t" }, + { "b", "feedback", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "t" }, + { "b0", "set direct signal gain", OFFSET(b0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 1, A }, + { "xM", "set delayed line gain", OFFSET(xM), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, A }, + { "M", "set delay in number of samples", OFFSET(M), AV_OPT_TYPE_INT, {.i64=10}, 1, 327680, A }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(acomb); + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats = NULL; + AVFilterChannelLayouts *layouts = NULL; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int 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; + + 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 COMB(name, type, dir, t) \ +static void acomb_## name ## _ ##dir(AudioCombContext *s, \ + AVFrame *in, AVFrame *out) \ +{ \ + const type b0 = s->b0; \ + const type xM = s->xM; \ + const int M = s->M; \ + int head; \ + \ + for (int c = 0; c < in->channels; c++) { \ + const type *src = (const type *)in->extended_data[c]; \ + type *delay = (type *)s->delayframe->extended_data[c]; \ + type *dst = (type *)out->extended_data[c]; \ + \ + head = s->head; \ + for (int n = 0; n < in->nb_samples; n++) { \ + dst[n] = b0 * src[n] + t * xM * delay[head]; \ + if (t == 1) \ + delay[head] = src[n]; \ + else \ + delay[head] = dst[n]; \ + head++; \ + if (head >= M) \ + head = 0; \ + } \ + } \ + \ + s->head = head; \ +} + +COMB(fltp, float, f, 1) +COMB(dblp, double, f, 1) +COMB(fltp, float, b, -1) +COMB(dblp, double, b, -1) + +static int config_input(AVFilterLink *inlink) +{ + AVFilterContext *ctx = inlink->dst; + AudioCombContext *s = ctx->priv; + + s->delayframe = ff_get_audio_buffer(inlink, s->M); + if (!s->delayframe) + return AVERROR(ENOMEM); + + switch (inlink->format) { + case AV_SAMPLE_FMT_FLTP: s->filter = s->t ? acomb_fltp_b : acomb_fltp_f; break; + case AV_SAMPLE_FMT_DBLP: s->filter = s->t ? acomb_dblp_b : acomb_dblp_f; break; + } + + return 0; +} + +static int filter_frame(AVFilterLink *inlink, AVFrame *in) +{ + AVFilterContext *ctx = inlink->dst; + AudioCombContext *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); + + s->filter(s, in, out); + + av_frame_free(&in); + return ff_filter_frame(outlink, out); +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioCombContext *s = ctx->priv; + + av_frame_free(&s->delayframe); +} + +static const AVFilterPad acomb_inputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .filter_frame = filter_frame, + .config_props = config_input, + }, + { NULL } +}; + +static const AVFilterPad acomb_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +AVFilter ff_af_acomb = { + .name = "acomb", + .description = NULL_IF_CONFIG_SMALL("Apply comb audio filter."), + .query_formats = query_formats, + .priv_size = sizeof(AudioCombContext), + .priv_class = &acomb_class, + .uninit = uninit, + .inputs = acomb_inputs, + .outputs = acomb_outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 1a26129069..7417f9656d 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -24,6 +24,7 @@ #include "config.h" extern AVFilter ff_af_abench; +extern AVFilter ff_af_acomb; extern AVFilter ff_af_acompressor; extern AVFilter ff_af_acontrast; extern AVFilter ff_af_acopy;
Signed-off-by: Paul B Mahol <onemda@gmail.com> --- doc/filters.texi | 28 ++++++ libavfilter/Makefile | 1 + libavfilter/af_acomb.c | 188 +++++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + 4 files changed, 218 insertions(+) create mode 100644 libavfilter/af_acomb.c