[FFmpeg-devel,3/3] avfilter: add audio downsample filter

Submitted by Paul B Mahol on April 18, 2019, 9:17 p.m.

Details

Message ID 20190418211734.11300-3-onemda@gmail.com
State New
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Commit Message

Paul B Mahol April 18, 2019, 9:17 p.m.
Signed-off-by: Paul B Mahol <onemda@gmail.com>
---
 doc/filters.texi             |  15 +++
 libavfilter/Makefile         |   1 +
 libavfilter/af_adownsample.c | 171 +++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c     |   1 +
 4 files changed, 188 insertions(+)
 create mode 100644 libavfilter/af_adownsample.c

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diff --git a/doc/filters.texi b/doc/filters.texi
index 465eeb4732..d6aac5730d 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -714,6 +714,21 @@  Compute derivative/integral of audio stream.
 
 Applying both filters one after another produces original audio.
 
+@section adownsample, aupsample
+Downsample or upsample audio by integer factor.
+
+For downsampling only the first out of each factor samples is retained,
+the others are discarded
+For upsampling, factor-1 zero-value samples are inserted between each pair of input samples.
+As a result, the original spectrum is replicated into the new frequency space and attenuated.
+
+A description of the accepted parameters follows.
+
+@table @option
+@item factor
+Set factor of downsampling/upsampling. Default is @code{1}.
+@end table
+
 @section aecho
 
 Apply echoing to the input audio.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index a38bc35231..ed36454f9e 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -43,6 +43,7 @@  OBJS-$(CONFIG_ADECLICK_FILTER)               += af_adeclick.o
 OBJS-$(CONFIG_ADECLIP_FILTER)                += af_adeclick.o
 OBJS-$(CONFIG_ADELAY_FILTER)                 += af_adelay.o
 OBJS-$(CONFIG_ADERIVATIVE_FILTER)            += af_aderivative.o
+OBJS-$(CONFIG_ADOWNSAMPLE_FILTER)            += af_adownsample.o
 OBJS-$(CONFIG_AECHO_FILTER)                  += af_aecho.o
 OBJS-$(CONFIG_AEMPHASIS_FILTER)              += af_aemphasis.o
 OBJS-$(CONFIG_AEVAL_FILTER)                  += aeval.o
diff --git a/libavfilter/af_adownsample.c b/libavfilter/af_adownsample.c
new file mode 100644
index 0000000000..1ee661e383
--- /dev/null
+++ b/libavfilter/af_adownsample.c
@@ -0,0 +1,171 @@ 
+/*
+ * Copyright (c) 2019 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/opt.h"
+#include "libavutil/samplefmt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "filters.h"
+#include "internal.h"
+
+typedef struct AudioDownSampleContext {
+    const AVClass *class;
+    int factor;
+
+    int64_t next_pts;
+} AudioDownSampleContext;
+
+#define OFFSET(x) offsetof(AudioDownSampleContext, x)
+#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption adownsample_options[] = {
+    { "factor", "set downsampling factor", OFFSET(factor), AV_OPT_TYPE_INT, {.i64=1}, 1, 64, A },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(adownsample);
+
+static int query_formats(AVFilterContext *ctx)
+{
+    AudioDownSampleContext *s = ctx->priv;
+    AVFilterChannelLayouts *layouts;
+    AVFilterFormats *formats;
+    int sample_rates[] = { 44100, -1 };
+    static const enum AVSampleFormat sample_fmts[] = {
+        AV_SAMPLE_FMT_DBLP,
+        AV_SAMPLE_FMT_NONE
+    };
+    AVFilterFormats *avff;
+    int ret;
+
+    if (!ctx->inputs[0]->in_samplerates ||
+        !ctx->inputs[0]->in_samplerates->nb_formats) {
+        return AVERROR(EAGAIN);
+    }
+
+    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;
+
+    avff = ctx->inputs[0]->in_samplerates;
+    sample_rates[0] = avff->formats[0];
+    if (!ctx->inputs[0]->out_samplerates)
+        if ((ret = ff_formats_ref(ff_make_format_list(sample_rates),
+                                  &ctx->inputs[0]->out_samplerates)) < 0)
+            return ret;
+
+    sample_rates[0] = avff->formats[0] / s->factor;
+    return ff_formats_ref(ff_make_format_list(sample_rates),
+                         &ctx->outputs[0]->in_samplerates);
+}
+
+static int config_input(AVFilterLink *inlink)
+{
+    AVFilterContext *ctx = inlink->dst;
+    AudioDownSampleContext *s = ctx->priv;
+
+    s->next_pts = AV_NOPTS_VALUE;
+
+    return 0;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+    AVFilterLink *inlink = ctx->inputs[0];
+    AVFilterLink *outlink = ctx->outputs[0];
+    AudioDownSampleContext *s = ctx->priv;
+    const int factor = s->factor;
+    AVFrame *in, *out;
+    int nb_samples;
+
+    FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
+
+    nb_samples = ff_inlink_queued_samples(inlink);
+
+    if (nb_samples >= s->factor) {
+        nb_samples = (nb_samples / factor) * factor;
+        ff_inlink_consume_samples(inlink, nb_samples, nb_samples, &in);
+
+        out = ff_get_audio_buffer(outlink, in->nb_samples / s->factor);
+        if (!out) {
+            av_frame_free(&in);
+            return AVERROR(ENOMEM);
+        }
+
+        if (s->next_pts == AV_NOPTS_VALUE)
+            s->next_pts = in->pts;
+
+        for (int c = 0; c < in->channels; c++) {
+            const double *src = (const double *)in->extended_data[c];
+            double *dst = (double *)out->extended_data[c];
+
+            for (int n = 0; n < out->nb_samples; n++)
+                dst[n] = src[n * factor];
+        }
+
+        out->pts = s->next_pts;
+        s->next_pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+        av_frame_free(&in);
+        return ff_filter_frame(outlink, out);
+    }
+
+    FF_FILTER_FORWARD_STATUS(inlink, outlink);
+    FF_FILTER_FORWARD_WANTED(outlink, inlink);
+
+    return FFERROR_NOT_READY;
+}
+
+static const AVFilterPad adownsample_inputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+        .config_props = config_input,
+    },
+    { NULL }
+};
+
+static const AVFilterPad adownsample_outputs[] = {
+    {
+        .name         = "default",
+        .type         = AVMEDIA_TYPE_AUDIO,
+    },
+    { NULL }
+};
+
+AVFilter ff_af_adownsample = {
+    .name          = "adownsample",
+    .description   = NULL_IF_CONFIG_SMALL("Downsample audio by integer factor."),
+    .query_formats = query_formats,
+    .priv_size     = sizeof(AudioDownSampleContext),
+    .priv_class    = &adownsample_class,
+    .activate      = activate,
+    .inputs        = adownsample_inputs,
+    .outputs       = adownsample_outputs,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 29b372a1db..ddff0741a4 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -35,6 +35,7 @@  extern AVFilter ff_af_adeclick;
 extern AVFilter ff_af_adeclip;
 extern AVFilter ff_af_adelay;
 extern AVFilter ff_af_aderivative;
+extern AVFilter ff_af_adownsample;
 extern AVFilter ff_af_aecho;
 extern AVFilter ff_af_aemphasis;
 extern AVFilter ff_af_aeval;