diff --git a/Changelog b/Changelog index 179f63c7d5..5c01e8365e 100644 --- a/Changelog +++ b/Changelog @@ -28,6 +28,7 @@ version : - showcwt multimedia filter - corr video filter - adrc audio filter +- afdelaysrc audio filter version 5.1: diff --git a/doc/filters.texi b/doc/filters.texi index 57088ccc6c..be70a2396b 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -7519,6 +7519,33 @@ aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) | 0.1*sin(2*PI*(360+2.5/2)*t)" @end itemize +@section afdelaysrc + +Generate a fractional delay FIR coefficients. + +The resulting stream can be used with @ref{afir} filter for filtering the audio signal. + +The filter accepts the following options: + +@table @option +@item delay, d +Set the fractional delay. Default is 0. + +@item sample_rate, r +Set the sample rate, default is 44100. + +@item nb_samples, n +Set the number of samples per each frame. Default is 1024. + +@item taps, t +Set the number of filter coefficents in output audio stream. +Default value is 0. + +@item channel_layout, c +Specifies the channel layout, and can be a string representing a channel layout. +The default value of @var{channel_layout} is "stereo". +@end table + @section afirsrc Generate a FIR coefficients using frequency sampling method. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 5783be281d..211ff4daaa 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -171,6 +171,7 @@ OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o OBJS-$(CONFIG_VOLUMEDETECT_FILTER) += af_volumedetect.o OBJS-$(CONFIG_AEVALSRC_FILTER) += aeval.o +OBJS-$(CONFIG_AFDELAYSRC_FILTER) += asrc_afdelaysrc.o OBJS-$(CONFIG_AFIRSRC_FILTER) += asrc_afirsrc.o OBJS-$(CONFIG_ANOISESRC_FILTER) += asrc_anoisesrc.o OBJS-$(CONFIG_ANULLSRC_FILTER) += asrc_anullsrc.o diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 52741b60e4..1ab3c8319a 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -159,6 +159,7 @@ extern const AVFilter ff_af_volume; extern const AVFilter ff_af_volumedetect; extern const AVFilter ff_asrc_aevalsrc; +extern const AVFilter ff_asrc_afdelaysrc; extern const AVFilter ff_asrc_afirsrc; extern const AVFilter ff_asrc_anoisesrc; extern const AVFilter ff_asrc_anullsrc; diff --git a/libavfilter/asrc_afdelaysrc.c b/libavfilter/asrc_afdelaysrc.c new file mode 100644 index 0000000000..2fc57cee4d --- /dev/null +++ b/libavfilter/asrc_afdelaysrc.c @@ -0,0 +1,173 @@ +/* + * Copyright (c) 2023 Paul B Mahol + * + * 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/avassert.h" +#include "libavutil/channel_layout.h" +#include "libavutil/opt.h" + +#include "audio.h" +#include "avfilter.h" +#include "filters.h" +#include "internal.h" + +typedef struct AFDelaySrcContext { + const AVClass *class; + + double delay; + int sample_rate; + int nb_samples; + int nb_taps; + AVChannelLayout chlayout; + char *chlayout_str; + + int64_t pts; +} AFDelaySrcContext; + +static av_cold int init(AVFilterContext *ctx) +{ + AFDelaySrcContext *s = ctx->priv; + int ret; + + ret = ff_parse_channel_layout(&s->chlayout, NULL, s->chlayout_str, ctx); + if (ret < 0) + return ret; + + return 0; +} + +static float sincf(float x) +{ + if (x == 0.f) + return 1.f; + return sinf(M_PI * x) / (M_PI * x); +} + +static int activate(AVFilterContext *ctx) +{ + AVFilterLink *outlink = ctx->outputs[0]; + AFDelaySrcContext *s = ctx->priv; + AVFrame *frame = NULL; + int nb_samples; + float *dst; + + if (!ff_outlink_frame_wanted(outlink)) + return FFERROR_NOT_READY; + + nb_samples = FFMIN(s->nb_samples, s->nb_taps - s->pts); + if (nb_samples <= 0) { + ff_outlink_set_status(outlink, AVERROR_EOF, s->pts); + return 0; + } + + if (!(frame = ff_get_audio_buffer(outlink, nb_samples))) + return AVERROR(ENOMEM); + + dst = (float *)frame->extended_data[0]; + for (int n = 0; n < nb_samples; n++) { + float x = s->pts + n; + dst[n] = sincf(x - s->delay) * cosf(M_PI * (x - s->delay) / s->nb_taps) / sincf((x - s->delay) / s->nb_taps); + } + + for (int ch = 1; ch < frame->ch_layout.nb_channels; ch++) + memcpy(frame->extended_data[ch], dst, sizeof(*dst) * nb_samples); + + frame->pts = s->pts; + s->pts += nb_samples; + + return ff_filter_frame(outlink, frame); +} + +static int query_formats(AVFilterContext *ctx) +{ + AFDelaySrcContext *s = ctx->priv; + AVChannelLayout chlayouts[] = { s->chlayout, { 0 } }; + int sample_rates[] = { s->sample_rate, -1 }; + static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_NONE }; + int ret = ff_set_common_formats_from_list(ctx, sample_fmts); + if (ret < 0) + return ret; + + ret = ff_set_common_channel_layouts_from_list(ctx, chlayouts); + if (ret < 0) + return ret; + + return ff_set_common_samplerates_from_list(ctx, sample_rates); +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AFDelaySrcContext *s = ctx->priv; + + outlink->sample_rate = s->sample_rate; + s->pts = 0; + if (s->nb_taps <= 0) + s->nb_taps = s->delay * 8 + 1; + + return 0; +} + +static const AVFilterPad afdelaysrc_outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, +}; + +static av_cold void uninit(AVFilterContext *ctx) +{ + AFDelaySrcContext *s = ctx->priv; + + av_channel_layout_uninit(&s->chlayout); +} + +#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM +#define OFFSET(x) offsetof(AFDelaySrcContext, x) + +static const AVOption afdelaysrc_options[] = { + { "delay", "set fractional delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE,{.dbl=0}, 0, INT16_MAX, AF }, + { "d", "set fractional delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE,{.dbl=0}, 0, INT16_MAX, AF }, + { "sample_rate", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF }, + { "r", "set sample rate", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64=44100}, 1, INT_MAX, AF }, + { "nb_samples", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF }, + { "n", "set the number of samples per requested frame", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64=1024}, 1, INT_MAX, AF }, + { "taps", "set number of taps for delay filter", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF }, + { "t", "set number of taps for delay filter", OFFSET(nb_taps), AV_OPT_TYPE_INT, {.i64=0}, 0, 32768, AF }, + { "channel_layout", "set channel layout", OFFSET(chlayout_str),AV_OPT_TYPE_STRING,{.str="stereo"},0, 0, AF }, + { "c", "set channel layout", OFFSET(chlayout_str),AV_OPT_TYPE_STRING,{.str="stereo"},0, 0, AF }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(afdelaysrc); + +const AVFilter ff_asrc_afdelaysrc = { + .name = "afdelaysrc", + .description = NULL_IF_CONFIG_SMALL("Generate a Fractional delay FIR coefficients."), + .priv_size = sizeof(AFDelaySrcContext), + .priv_class = &afdelaysrc_class, + .init = init, + .activate = activate, + .uninit = uninit, + .inputs = NULL, + FILTER_OUTPUTS(afdelaysrc_outputs), + FILTER_QUERY_FUNC(query_formats), +}; diff --git a/libavfilter/version.h b/libavfilter/version.h index 9fabc544b5..a56ba3bb6d 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -31,7 +31,7 @@ #include "version_major.h" -#define LIBAVFILTER_VERSION_MINOR 53 +#define LIBAVFILTER_VERSION_MINOR 54 #define LIBAVFILTER_VERSION_MICRO 100