avfilter: add amultiply audio filter

6 files changed, 236 insertions, 1 deletions

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
 
+@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, \