avfilter: add audio signal to distortion ratio filter

6 files changed, 183 insertions, 1 deletions

diff --git a/Changelog b/Changelog
index 540056d4c3..04ef413192 100644
--- a/Changelog
+++ b/Changelog
@@ -24,6 +24,7 @@ version <next>:
 - amr parser
 - (a)latency filters
 - GEM Raster image decoder
+- asdr audio filter
 
 
 version 4.4:
diff --git a/doc/filters.texi b/doc/filters.texi
index c063316ed3..8b2e85d184 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2556,6 +2556,13 @@ noise removed from input signal.
 
 This filter supports the all above options as @ref{commands}.
 
+@section asdr
+Measure Audio Signal-to-Distortion Ratio.
+
+This filter takes two audio streams for input, and outputs first
+audio stream.
+Results are in dB per channel at end of either input.
+
 @section asetnsamples
 
 Set the number of samples per each output audio frame.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index fabe9c9af0..9ec0963ca8 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -81,6 +81,7 @@ OBJS-$(CONFIG_AREALTIME_FILTER)              += f_realtime.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
 OBJS-$(CONFIG_AREVERSE_FILTER)               += f_reverse.o
 OBJS-$(CONFIG_ARNNDN_FILTER)                 += af_arnndn.o
+OBJS-$(CONFIG_ASDR_FILTER)                   += af_asdr.o
 OBJS-$(CONFIG_ASEGMENT_FILTER)               += f_segment.o
 OBJS-$(CONFIG_ASELECT_FILTER)                += f_select.o
 OBJS-$(CONFIG_ASENDCMD_FILTER)               += f_sendcmd.o
diff --git a/libavfilter/af_asdr.c b/libavfilter/af_asdr.c
new file mode 100644
index 0000000000..72b6dfa78b
--- /dev/null
+++ b/libavfilter/af_asdr.c
@@ -0,0 +1,172 @@
+/*
+ * Copyright (c) 2021 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/channel_layout.h"
+#include "libavutil/common.h"
+#include "libavutil/opt.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "formats.h"
+#include "filters.h"
+#include "internal.h"
+
+typedef struct AudioSDRContext {
+    int channels;
+    int64_t pts;
+    double *sum_u;
+    double *sum_uv;
+
+    AVFrame *cache[2];
+} AudioSDRContext;
+
+static void sdr(AVFilterContext *ctx, const AVFrame *u, const AVFrame *v)
+{
+    AudioSDRContext *s = ctx->priv;
+
+    for (int ch = 0; ch < u->channels; ch++) {
+        const double *const us = (double *)u->extended_data[ch];
+        const double *const vs = (double *)v->extended_data[ch];
+        double sum_uv = s->sum_uv[ch];
+        double sum_u = s->sum_u[ch];
+
+        for (int n = 0; n < u->nb_samples; n++) {
+            sum_u  += us[n] * us[n];
+            sum_uv += (us[n] - vs[n]) * (us[n] - vs[n]);
+        }
+
+        s->sum_uv[ch] = sum_uv;
+        s->sum_u[ch]  = sum_u;
+    }
+}
+
+static int activate(AVFilterContext *ctx)
+{
+    AudioSDRContext *s = ctx->priv;
+    int ret, status;
+    int available;
+    int64_t pts;
+
+    FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
+
+    available = FFMIN(ff_inlink_queued_samples(ctx->inputs[0]), ff_inlink_queued_samples(ctx->inputs[1]));
+    if (available > 0) {
+        AVFrame *out;
+
+        for (int i = 0; i < 2; i++) {
+            ret = ff_inlink_consume_samples(ctx->inputs[i], available, available, &s->cache[i]);
+            if (ret > 0) {
+                if (s->pts == AV_NOPTS_VALUE)
+                    s->pts = s->cache[i]->pts;
+            }
+        }
+
+        sdr(ctx, s->cache[0], s->cache[1]);
+
+        av_frame_free(&s->cache[1]);
+        out = s->cache[0];
+        out->nb_samples = available;
+        out->pts = s->pts;
+        s->pts += available;
+        s->cache[0] = NULL;
+
+        return ff_filter_frame(ctx->outputs[0], out);
+    }
+
+    for (int i = 0; i < 2; i++) {
+        if (ff_inlink_acknowledge_status(ctx->inputs[i], &status, &pts)) {
+            ff_outlink_set_status(ctx->outputs[0], status, s->pts);
+            return 0;
+        }
+    }
+
+    if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+        for (int i = 0; i < 2; i++) {
+            if (ff_inlink_queued_samples(ctx->inputs[i]) > 0)
+                continue;
+            ff_inlink_request_frame(ctx->inputs[i]);
+        }
+        return 0;
+    }
+
+    return FFERROR_NOT_READY;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AVFilterLink *inlink = ctx->inputs[0];
+    AudioSDRContext *s = ctx->priv;
+
+    s->pts = AV_NOPTS_VALUE;
+
+    s->channels = inlink->channels;
+
+    s->sum_u  = av_calloc(outlink->channels, sizeof(*s->sum_u));
+    s->sum_uv = av_calloc(outlink->channels, sizeof(*s->sum_uv));
+    if (!s->sum_u || !s->sum_uv)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AudioSDRContext *s = ctx->priv;
+
+    for (int ch = 0; ch < s->channels; ch++)
+        av_log(ctx, AV_LOG_INFO, "SDR ch%d: %g dB\n", ch, 20. * log10(s->sum_u[ch] / s->sum_uv[ch]));
+
+    av_frame_free(&s->cache[0]);
+    av_frame_free(&s->cache[1]);
+
+    av_freep(&s->sum_u);
+    av_freep(&s->sum_uv);
+}
+
+static const AVFilterPad inputs[] = {
+    {
+        .name = "input0",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    {
+        .name = "input1",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+};
+
+static const AVFilterPad outputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_output,
+    },
+};
+
+const AVFilter ff_af_asdr = {
+    .name           = "asdr",
+    .description    = NULL_IF_CONFIG_SMALL("Measure Audio Signal-to-Distortion Ratio."),
+    .priv_size      = sizeof(AudioSDRContext),
+    .activate       = activate,
+    .uninit         = uninit,
+    FILTER_INPUTS(inputs),
+    FILTER_OUTPUTS(outputs),
+    FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBLP),
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index ad50a8bc5d..a1bf78f776 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -74,6 +74,7 @@ extern const AVFilter ff_af_arealtime;
 extern const AVFilter ff_af_aresample;
 extern const AVFilter ff_af_areverse;
 extern const AVFilter ff_af_arnndn;
+extern const AVFilter ff_af_asdr;
 extern const AVFilter ff_af_asegment;
 extern const AVFilter ff_af_aselect;
 extern const AVFilter ff_af_asendcmd;
diff --git a/libavfilter/version.h b/libavfilter/version.h
index dfbc1bd7c2..e4c25b9225 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
 #include "libavutil/version.h"
 
 #define LIBAVFILTER_VERSION_MAJOR   8
-#define LIBAVFILTER_VERSION_MINOR  11
+#define LIBAVFILTER_VERSION_MINOR  12
 #define LIBAVFILTER_VERSION_MICRO 100