From ecf38be7c7f50a08e5a1f3cd9eea06fc5594d010 Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Wed, 12 Sep 2018 11:12:21 +0200 Subject: [PATCH] avfilter: add amultiply audio filter --- Changelog | 1 + doc/filters.texi | 9 ++ libavfilter/Makefile | 1 + libavfilter/af_amultiply.c | 223 +++++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + libavfilter/version.h | 2 +- 6 files changed, 236 insertions(+), 1 deletion(-) create mode 100644 libavfilter/af_amultiply.c diff --git a/Changelog b/Changelog index 59ea36d08b..5cb3f86f1d 100644 --- a/Changelog +++ b/Changelog @@ -27,6 +27,7 @@ version : - support for AV1 in MP4 - transpose_npp filter - AVS2 video encoder via libxavs2 +- amultiply filter version 4.0: diff --git a/doc/filters.texi b/doc/filters.texi index 860d1eadca..e3ae0b01f0 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -1488,6 +1488,15 @@ Specify weight of each input audio stream as sequence. Each weight is separated by space. By default all inputs have same weight. @end table +@section amultiply + +Multiply first audio stream with second audio stream and store result +in output audio stream. Multiplication is done by multiplying each +sample from first stream with sample at same position from second stream. + +With this element-wise multiplication one can create amplitude fades and +amplitude modulations. + @section anequalizer High-order parametric multiband equalizer for each channel. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index 5b0462692a..f15e520d5d 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -58,6 +58,7 @@ OBJS-$(CONFIG_ALOOP_FILTER) += f_loop.o OBJS-$(CONFIG_AMERGE_FILTER) += af_amerge.o OBJS-$(CONFIG_AMETADATA_FILTER) += f_metadata.o OBJS-$(CONFIG_AMIX_FILTER) += af_amix.o +OBJS-$(CONFIG_AMULTIPLY_FILTER) += af_amultiply.o OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o OBJS-$(CONFIG_ANULL_FILTER) += af_anull.o OBJS-$(CONFIG_APAD_FILTER) += af_apad.o diff --git a/libavfilter/af_amultiply.c b/libavfilter/af_amultiply.c new file mode 100644 index 0000000000..a742f6a9c6 --- /dev/null +++ b/libavfilter/af_amultiply.c @@ -0,0 +1,223 @@ +/* + * Copyright (c) 2018 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/common.h" +#include "libavutil/float_dsp.h" +#include "libavutil/opt.h" + +#define FF_INTERNAL_FIELDS 1 +#include "framequeue.h" + +#include "audio.h" +#include "avfilter.h" +#include "formats.h" +#include "filters.h" +#include "internal.h" + +typedef struct AudioMultiplyContext { + const AVClass *class; + + AVFrame *frames[2]; + int64_t pts; + int planes; + int channels; + int samples_align; + + AVFloatDSPContext *fdsp; +} AudioMultiplyContext; + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP, + AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int 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_make_format_list(sample_fmts); + if (!formats) + return AVERROR(ENOMEM); + ret = ff_set_common_formats(ctx, formats); + if (ret < 0) + return ret; + + formats = ff_all_samplerates(); + if (!formats) + return AVERROR(ENOMEM); + return ff_set_common_samplerates(ctx, formats); +} + +static int activate(AVFilterContext *ctx) +{ + AudioMultiplyContext *s = ctx->priv; + int i, ret, status; + int nb_samples; + int64_t pts; + + FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx); + + nb_samples = FFMIN(ff_framequeue_queued_samples(&ctx->inputs[0]->fifo), + ff_framequeue_queued_samples(&ctx->inputs[1]->fifo)); + for (i = 0; i < ctx->nb_inputs && nb_samples > 0; i++) { + if (s->frames[i]) + continue; + + if (ff_inlink_check_available_samples(ctx->inputs[i], nb_samples) > 0) { + ret = ff_inlink_consume_samples(ctx->inputs[i], nb_samples, nb_samples, &s->frames[i]); + if (ret < 0) + return ret; + } + } + + if (nb_samples > 0 && s->frames[0] && s->frames[1]) { + AVFrame *out; + int plane_samples; + + if (av_sample_fmt_is_planar(ctx->inputs[0]->format)) + plane_samples = FFALIGN(nb_samples, s->samples_align); + else + plane_samples = FFALIGN(nb_samples * s->channels, s->samples_align); + + out = ff_get_audio_buffer(ctx->outputs[0], nb_samples); + if (!out) + return AVERROR(ENOMEM); + + out->pts = s->pts; + s->pts += nb_samples; + + if (av_get_packed_sample_fmt(ctx->inputs[0]->format) == AV_SAMPLE_FMT_FLT) { + for (i = 0; i < s->planes; i++) { + s->fdsp->vector_fmul((float *)out->extended_data[i], + (const float *)s->frames[0]->extended_data[i], + (const float *)s->frames[1]->extended_data[i], + plane_samples); + } + } else { + for (i = 0; i < s->planes; i++) { + s->fdsp->vector_dmul((double *)out->extended_data[i], + (const double *)s->frames[0]->extended_data[i], + (const double *)s->frames[1]->extended_data[i], + plane_samples); + } + } + emms_c(); + + av_frame_free(&s->frames[0]); + av_frame_free(&s->frames[1]); + + ret = ff_filter_frame(ctx->outputs[0], out); + if (ret < 0) + return ret; + } + + if (!nb_samples) { + for (i = 0; i < 2; i++) { + if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) { + ff_outlink_set_status(ctx->outputs[0], status, pts); + return 0; + } + } + } + + if (ff_outlink_frame_wanted(ctx->outputs[0])) { + for (i = 0; i < 2; i++) { + if (ff_framequeue_queued_samples(&ctx->inputs[i]->fifo) > 0) + continue; + ff_inlink_request_frame(ctx->inputs[i]); + return 0; + } + } + return 0; +} + +static int config_output(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioMultiplyContext *s = ctx->priv; + AVFilterLink *inlink = ctx->inputs[0]; + + s->channels = inlink->channels; + s->planes = av_sample_fmt_is_planar(inlink->format) ? inlink->channels : 1; + s->samples_align = 16; + + return 0; +} + +static av_cold int init(AVFilterContext *ctx) +{ + AudioMultiplyContext *s = ctx->priv; + + s->fdsp = avpriv_float_dsp_alloc(0); + if (!s->fdsp) + return AVERROR(ENOMEM); + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioMultiplyContext *s = ctx->priv; + av_freep(&s->fdsp); +} + +static const AVFilterPad inputs[] = { + { + .name = "multiply0", + .type = AVMEDIA_TYPE_AUDIO, + }, + { + .name = "multiply1", + .type = AVMEDIA_TYPE_AUDIO, + }, + { NULL } +}; + +static const AVFilterPad outputs[] = { + { + .name = "default", + .type = AVMEDIA_TYPE_AUDIO, + .config_props = config_output, + }, + { NULL } +}; + +AVFilter ff_af_amultiply = { + .name = "amultiply", + .description = NULL_IF_CONFIG_SMALL("Multiply two audio streams."), + .priv_size = sizeof(AudioMultiplyContext), + .init = init, + .uninit = uninit, + .activate = activate, + .query_formats = query_formats, + .inputs = inputs, + .outputs = outputs, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 10ac52b711..c467064783 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -51,6 +51,7 @@ extern AVFilter ff_af_aloop; extern AVFilter ff_af_amerge; extern AVFilter ff_af_ametadata; extern AVFilter ff_af_amix; +extern AVFilter ff_af_amultiply; extern AVFilter ff_af_anequalizer; extern AVFilter ff_af_anull; extern AVFilter ff_af_apad; diff --git a/libavfilter/version.h b/libavfilter/version.h index 30ccef18ea..2d1316df4b 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,7 +30,7 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 7 -#define LIBAVFILTER_VERSION_MINOR 29 +#define LIBAVFILTER_VERSION_MINOR 30 #define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \