[FFmpeg-cvslog] avfilter: add amultiply audio filter
Paul B Mahol
git at videolan.org
Thu Sep 13 11:36:12 EEST 2018
ffmpeg | branch: master | Paul B Mahol <onemda at gmail.com> | Wed Sep 12 11:12:21 2018 +0200| [ecf38be7c7f50a08e5a1f3cd9eea06fc5594d010] | committer: Paul B Mahol
avfilter: add amultiply audio filter
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=ecf38be7c7f50a08e5a1f3cd9eea06fc5594d010
---
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(-)
diff --git a/Changelog b/Changelog
index 59ea36d08b..5cb3f86f1d 100644
--- a/Changelog
+++ b/Changelog
@@ -27,6 +27,7 @@ version <next>:
- 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
+ at 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, \
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