lavfi: add asetnsamples audio filter

This filter changes the number of samples on single output operation.

Based on a patch by Andrey Utkin <andrey.krieger.utkin@gmail.com>.

Changelog

@@ -6,6 +6,7 @@ version next:
 - Scene detection in libavfilter
 - Indeo Audio decoder
 - channelsplit audio filter
+- setnsamples audio filter
 
 
 version 0.11:

doc/filters.texi

@@ -273,6 +273,36 @@ For example, to resample the input audio to 44100Hz:
 aresample=44100
 @end example
 
+@section asetnsamples
+
+Set the number of samples per each output audio frame.
+
+The last output packet may contain a different number of samples, as
+the filter will flush all the remaining samples when the input audio
+signal its end.
+
+The filter accepts parameters as a list of @var{key}=@var{value} pairs,
+separated by ":".
+
+@table @option
+
+@item nb_out_samples, n
+Set the number of frames per each output audio frame. The number is
+intended as the number of samples @emph{per each channel}.
+Default value is 1024.
+
+@item pad, p
+If set to 1, the filter will pad the last audio frame with zeroes, so
+that the last frame will contain the same number of samples as the
+previous ones. Default value is 1.
+@end table
+
+For example, to set the number of per-frame samples to 1234 and
+disable padding for the last frame, use:
+@example
+asetnsamples=n=1234:p=0
+@end example
+
 @section ashowinfo
 
 Show a line containing various information for each input audio frame.

libavfilter/Makefile

@@ -51,6 +51,7 @@ OBJS-$(CONFIG_AMERGE_FILTER)                 += af_amerge.o
 OBJS-$(CONFIG_AMIX_FILTER)                   += af_amix.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
 OBJS-$(CONFIG_ARESAMPLE_FILTER)              += af_aresample.o
+OBJS-$(CONFIG_ASETNSAMPLES_FILTER)           += af_asetnsamples.o
 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o
 OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o
 OBJS-$(CONFIG_ASTREAMSYNC_FILTER)            += af_astreamsync.o

libavfilter/af_asetnsamples.c

+/*
+ * Copyright (c) 2012 Andrey Utkin
+ * Copyright (c) 2012 Stefano Sabatini
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * Filter that changes number of samples on single output operation
+ */
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/avassert.h"
+#include "libavutil/opt.h"
+#include "avfilter.h"
+#include "audio.h"
+#include "formats.h"
+
+typedef struct {
+    const AVClass *class;
+    int nb_out_samples;  ///< how many samples to output
+    AVAudioFifo *fifo;   ///< samples are queued here
+    int64_t next_out_pts;
+    int req_fullfilled;
+    int pad;
+} ASNSContext;
+
+#define OFFSET(x) offsetof(ASNSContext, x)
+
+static const AVOption asns_options[] = {
+{ "pad", "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
+{ "p",   "pad last frame with zeros", OFFSET(pad), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
+{ "nb_out_samples", "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX },
+{ "n",              "set the number of per-frame output samples", OFFSET(nb_out_samples), AV_OPT_TYPE_INT, {.dbl=1024}, 1, INT_MAX },
+{ NULL }
+};
+
+static const AVClass asns_class = {
+    "asetnsamples",
+    av_default_item_name,
+    asns_options
+};
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+    ASNSContext *asns = ctx->priv;
+    int err;
+
+    asns->class = &asns_class;
+    av_opt_set_defaults(asns);
+
+    if ((err = av_set_options_string(asns, args, "=", ":")) < 0) {
+        av_log(ctx, AV_LOG_ERROR, "Error parsing options string: '%s'\n", args);
+        return err;
+    }
+
+    asns->next_out_pts = AV_NOPTS_VALUE;
+    av_log(ctx, AV_LOG_INFO, "nb_out_samples:%d pad:%d\n", asns->nb_out_samples, asns->pad);
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    ASNSContext *asns = ctx->priv;
+    av_audio_fifo_free(asns->fifo);
+}
+
+static int config_props_output(AVFilterLink *outlink)
+{
+    ASNSContext *asns = outlink->src->priv;
+    int nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout);
+
+    asns->fifo = av_audio_fifo_alloc(outlink->format, nb_channels, asns->nb_out_samples);
+    if (!asns->fifo)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static int push_samples(AVFilterLink *outlink)
+{
+    ASNSContext *asns = outlink->src->priv;
+    AVFilterBufferRef *outsamples = NULL;
+    int nb_out_samples, nb_pad_samples;
+
+    if (asns->pad) {
+        nb_out_samples = av_audio_fifo_size(asns->fifo) ? asns->nb_out_samples : 0;
+        nb_pad_samples = nb_out_samples - FFMIN(nb_out_samples, av_audio_fifo_size(asns->fifo));
+    } else {
+        nb_out_samples = FFMIN(asns->nb_out_samples, av_audio_fifo_size(asns->fifo));
+        nb_pad_samples = 0;
+    }
+
+    if (!nb_out_samples)
+        return 0;
+
+    outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_out_samples);
+    av_assert0(outsamples);
+
+    av_audio_fifo_read(asns->fifo,
+                       (void **)outsamples->extended_data, nb_out_samples);
+
+    if (nb_pad_samples)
+        av_samples_set_silence(outsamples->extended_data, nb_out_samples - nb_pad_samples,
+                               nb_pad_samples, av_get_channel_layout_nb_channels(outlink->channel_layout),
+                               outlink->format);
+    outsamples->audio->nb_samples     = nb_out_samples;
+    outsamples->audio->channel_layout = outlink->channel_layout;
+    outsamples->audio->sample_rate    = outlink->sample_rate;
+    outsamples->pts = asns->next_out_pts;
+
+    if (asns->next_out_pts != AV_NOPTS_VALUE)
+        asns->next_out_pts += nb_out_samples;
+
+    ff_filter_samples(outlink, outsamples);
+    asns->req_fullfilled = 1;
+    return nb_out_samples;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+{
+    AVFilterContext *ctx = inlink->dst;
+    ASNSContext *asns = ctx->priv;
+    AVFilterLink *outlink = ctx->outputs[0];
+    int ret;
+    int nb_samples = insamples->audio->nb_samples;
+
+    if (av_audio_fifo_space(asns->fifo) < nb_samples) {
+        av_log(ctx, AV_LOG_DEBUG, "No space for %d samples, stretching audio fifo\n", nb_samples);
+        ret = av_audio_fifo_realloc(asns->fifo, av_audio_fifo_size(asns->fifo) + nb_samples);
+        if (ret < 0) {
+            av_log(ctx, AV_LOG_ERROR,
+                   "Stretching audio fifo failed, discarded %d samples\n", nb_samples);
+            return;
+        }
+    }
+    av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
+    if (asns->next_out_pts == AV_NOPTS_VALUE)
+        asns->next_out_pts = insamples->pts;
+    avfilter_unref_buffer(insamples);
+
+    if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
+        push_samples(outlink);
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+    ASNSContext *asns = outlink->src->priv;
+    AVFilterLink *inlink = outlink->src->inputs[0];
+    int ret;
+
+    asns->req_fullfilled = 0;
+    do {
+        ret = avfilter_request_frame(inlink);
+    } while (!asns->req_fullfilled && ret >= 0);
+
+    if (ret == AVERROR_EOF)
+        while (push_samples(outlink))
+            ;
+
+    return ret;
+}
+
+AVFilter avfilter_af_asetnsamples = {
+    .name           = "asetnsamples",
+    .description    = NULL_IF_CONFIG_SMALL("Set the number of samples for each output audio frames."),
+    .priv_size      = sizeof(ASNSContext),
+    .init           = init,
+    .uninit         = uninit,
+
+    .inputs  = (const AVFilterPad[]) {
+        {
+            .name           = "default",
+            .type           = AVMEDIA_TYPE_AUDIO,
+            .filter_samples = filter_samples,
+            .min_perms      = AV_PERM_READ|AV_PERM_WRITE
+        },
+        { .name = NULL }
+    },
+
+    .outputs = (const AVFilterPad[]) {
+        {
+            .name           = "default",
+            .type           = AVMEDIA_TYPE_AUDIO,
+            .request_frame  = request_frame,
+            .config_props   = config_props_output,
+        },
+        { .name = NULL }
+    },
+};

libavfilter/allfilters.c

@@ -40,6 +40,7 @@ void avfilter_register_all(void)
     REGISTER_FILTER (AMIX,        amix,        af);
     REGISTER_FILTER (ANULL,       anull,       af);
     REGISTER_FILTER (ARESAMPLE,   aresample,   af);
+    REGISTER_FILTER (ASETNSAMPLES, asetnsamples, af);
     REGISTER_FILTER (ASHOWINFO,   ashowinfo,   af);
     REGISTER_FILTER (ASPLIT,      asplit,      af);
     REGISTER_FILTER (ASTREAMSYNC, astreamsync, af);

libavfilter/version.h

@@ -29,7 +29,7 @@
 #include "libavutil/avutil.h"
 
 #define LIBAVFILTER_VERSION_MAJOR  2
-#define LIBAVFILTER_VERSION_MINOR 79
+#define LIBAVFILTER_VERSION_MINOR 80
 #define LIBAVFILTER_VERSION_MICRO 100
 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \