@@ -6,6 +6,7 @@ version next:

 - Scene detection in libavfilter

 - Indeo Audio decoder

 - channelsplit audio filter

+- setnsamples audio filter

 version 0.11:

@@ -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.

@@ -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

new file mode 100644

@@ -0,0 +1,206 @@

+/*

+ * 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 }

+    },

+};

@@ -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);

@@ -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, \