lavfi: add asetrate filter.

Changelog

@@ -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:

doc/filters.texi

@@ -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.

libavfilter/Makefile

@@ -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

libavfilter/af_asetrate.c

@@ -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,
+};

libavfilter/allfilters.c

@@ -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);

libavfilter/version.h

@@ -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, \