@@ -28,6 +28,7 @@ version <next>:

 - The matroska demuxer can now output proper verbatim ASS packets. It will

   become the default at the next libavformat major bump.

 - decent native animated GIF encoding

+- asetrate filter

 version 1.2:

@@ -900,6 +900,18 @@ disable padding for the last frame, use:

 asetnsamples=n=1234:p=0

 @end example

+@section asetrate

+

+Set the sample rate without altering the PCM data.

+This will result in a change of speed and pitch.

+

+The filter accepts the following options:

+

+@table @option

+@item sample_rate, r

+Set the output sample rate. Default is 44100 Hz.

+@end table

+

 @section ashowinfo

 Show a line containing various information for each input audio frame.

@@ -64,6 +64,7 @@ OBJS-$(CONFIG_ASELECT_FILTER)                += f_select.o

 OBJS-$(CONFIG_ASENDCMD_FILTER)               += f_sendcmd.o

 OBJS-$(CONFIG_ASETNSAMPLES_FILTER)           += af_asetnsamples.o

 OBJS-$(CONFIG_ASETPTS_FILTER)                += f_setpts.o

+OBJS-$(CONFIG_ASETRATE_FILTER)               += af_asetrate.o

 OBJS-$(CONFIG_ASETTB_FILTER)                 += f_settb.o

 OBJS-$(CONFIG_ASHOWINFO_FILTER)              += af_ashowinfo.o

 OBJS-$(CONFIG_ASPLIT_FILTER)                 += split.o

new file mode 100644

@@ -0,0 +1,119 @@

+/*

+ * Copyright (c) 2013 Nicolas George

+ *

+ * 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 "avfilter.h"

+#include "internal.h"

+

+typedef struct {

+    const AVClass *class;

+    int sample_rate;

+    int rescale_pts;

+} ASetRateContext;

+

+#define CONTEXT ASetRateContext

+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

+

+#define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \

+    { name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type,          \

+      { .deffield = def }, min, max, FLAGS, __VA_ARGS__ }

+

+#define OPT_INT(name, field, def, min, max, descr, ...) \

+    OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)

+

+static const AVOption asetrate_options[] = {

+    OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),

+    OPT_INT("r",           sample_rate, 44100, 1, INT_MAX, "set the sample rate"),

+    {NULL},

+};

+

+AVFILTER_DEFINE_CLASS(asetrate);

+

+static av_cold int query_formats(AVFilterContext *ctx)

+{

+    ASetRateContext *sr = ctx->priv;

+    int sample_rates[] = { sr->sample_rate, -1 };

+

+    ff_formats_ref(ff_make_format_list(sample_rates),

+                   &ctx->outputs[0]->in_samplerates);

+    return 0;

+}

+

+static av_cold int config_props(AVFilterLink *outlink)

+{

+    AVFilterContext *ctx = outlink->src;

+    ASetRateContext *sr = ctx->priv;

+    AVFilterLink *inlink = ctx->inputs[0];

+    AVRational intb = ctx->inputs[0]->time_base;

+    int inrate = inlink->sample_rate;

+

+    if (intb.num == 1 && intb.den == inrate) {

+        outlink->time_base.num = 1;

+        outlink->time_base.den = outlink->sample_rate;

+    } else {

+        outlink->time_base = intb;

+        sr->rescale_pts = 1;

+        if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))

+            av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");

+    }

+    return 0;

+}

+

+static int filter_frame(AVFilterLink *inlink, AVFrame *frame)

+{

+    AVFilterContext *ctx = inlink->dst;

+    ASetRateContext *sr = ctx->priv;

+    AVFilterLink *outlink = ctx->outputs[0];

+

+    frame->sample_rate = outlink->sample_rate;

+    if (sr->rescale_pts)

+        frame->pts = av_rescale(frame->pts, inlink->sample_rate,

+                                           outlink->sample_rate);

+    return ff_filter_frame(outlink, frame);

+}

+

+static const AVFilterPad asetrate_inputs[] = {

+    {

+        .name         = "default",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .filter_frame = filter_frame,

+    },

+    { NULL }

+};

+

+static const AVFilterPad asetrate_outputs[] = {

+    {

+        .name         = "default",

+        .type         = AVMEDIA_TYPE_AUDIO,

+        .config_props = config_props,

+    },

+    { NULL }

+};

+

+AVFilter avfilter_af_asetrate = {

+    .name          = "asetrate",

+    .description   = NULL_IF_CONFIG_SMALL("Change the sample rate without "

+                                          "altering the data."),

+    .query_formats = query_formats,

+    .priv_size     = sizeof(ASetRateContext),

+    .inputs        = asetrate_inputs,

+    .outputs       = asetrate_outputs,

+    .priv_class    = &asetrate_class,

+};

@@ -62,6 +62,7 @@ void avfilter_register_all(void)

     REGISTER_FILTER(ASENDCMD,       asendcmd,       af);

     REGISTER_FILTER(ASETNSAMPLES,   asetnsamples,   af);

     REGISTER_FILTER(ASETPTS,        asetpts,        af);

+    REGISTER_FILTER(ASETRATE,       asetrate,       af);

     REGISTER_FILTER(ASETTB,         asettb,         af);

     REGISTER_FILTER(ASHOWINFO,      ashowinfo,      af);

     REGISTER_FILTER(ASPLIT,         asplit,         af);

@@ -29,7 +29,7 @@

 #include "libavutil/avutil.h"

 #define LIBAVFILTER_VERSION_MAJOR  3

-#define LIBAVFILTER_VERSION_MINOR  58

+#define LIBAVFILTER_VERSION_MINOR  59

 #define LIBAVFILTER_VERSION_MICRO 100

 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \