On 10/2/19, James Almer <jamr...@gmail.com> wrote: > On 10/2/2019 12:11 PM, Paul B Mahol wrote: >> Signed-off-by: Paul B Mahol <one...@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-requ...@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-requ...@ffmpeg.org with subject "unsubscribe".