1 files changed, 126 insertions, 0 deletions
diff --git a/libavfilter/af_aresample.c b/libavfilter/af_aresample.c
new file mode 100644
index 0000000000..23980d1556
--- /dev/null
+++ b/libavfilter/af_aresample.c
@@ -0,0 +1,126 @@
+/*
+ * Copyright (c) 2011 Stefano Sabatini
+ * Copyright (c) 2011 Mina Nagy Zaki
+ *
+ * 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
+ * resampling audio filter
+ */
+
+#include "libswresample/swresample.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct {
+    int out_rate;
+    double ratio;
+    struct SwrContext *swr;
+} AResampleContext;
+
+static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
+{
+    AResampleContext *aresample = ctx->priv;
+    int ret;
+
+    if (args) {
+        if ((ret = ff_parse_sample_rate(&aresample->out_rate, args, ctx)) < 0)
+            return ret;
+    } else {
+        aresample->out_rate = -1;
+    }
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AResampleContext *aresample = ctx->priv;
+    swr_free(&aresample->swr);
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    int ret;
+    AVFilterContext *ctx = outlink->src;
+    AVFilterLink *inlink = ctx->inputs[0];
+    AResampleContext *aresample = ctx->priv;
+
+    if (aresample->out_rate == -1)
+        aresample->out_rate = outlink->sample_rate;
+    else
+        outlink->sample_rate = aresample->out_rate;
+    outlink->time_base = (AVRational) {1, aresample->out_rate};
+
+    //TODO: make the resampling parameters (filter size, phrase shift, linear, cutoff) configurable
+    aresample->swr = swr_alloc_set_opts(aresample->swr,
+                                        inlink->channel_layout, inlink->format, aresample->out_rate,
+                                        inlink->channel_layout, inlink->format, inlink->sample_rate,
+                                        0, ctx);
+    if (!aresample->swr)
+        return AVERROR(ENOMEM);
+    ret = swr_init(aresample->swr);
+    if (ret < 0)
+        return ret;
+
+    aresample->ratio = (double)outlink->sample_rate / inlink->sample_rate;
+
+    av_log(ctx, AV_LOG_INFO, "r:%"PRId64"Hz -> r:%"PRId64"Hz\n",
+           inlink->sample_rate, outlink->sample_rate);
+    return 0;
+}
+
+static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
+{
+    AResampleContext *aresample = inlink->dst->priv;
+    const int n_in  = insamplesref->audio->nb_samples;
+    int n_out       = n_in * aresample->ratio;
+    AVFilterLink *const outlink = inlink->dst->outputs[0];
+    AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
+
+    n_out = swr_convert(aresample->swr, outsamplesref->data, n_out,
+                                 (void *)insamplesref->data, n_in);
+
+    avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
+    outsamplesref->audio->sample_rate = outlink->sample_rate;
+    outsamplesref->audio->nb_samples  = n_out;
+    outsamplesref->pts = insamplesref->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
+        av_rescale(outlink->sample_rate, insamplesref->pts, inlink ->sample_rate);
+
+    ff_filter_samples(outlink, outsamplesref);
+    avfilter_unref_buffer(insamplesref);
+}
+
+AVFilter avfilter_af_aresample = {
+    .name          = "aresample",
+    .description   = NULL_IF_CONFIG_SMALL("Resample audio data."),
+    .init          = init,
+    .uninit        = uninit,
+    .priv_size     = sizeof(AResampleContext),
+
+    .inputs    = (const AVFilterPad[]) {{ .name      = "default",
+                                    .type            = AVMEDIA_TYPE_AUDIO,
+                                    .filter_samples  = filter_samples,
+                                    .min_perms       = AV_PERM_READ, },
+                                  { .name = NULL}},
+    .outputs   = (const AVFilterPad[]) {{ .name      = "default",
+                                    .config_props    = config_output,
+                                    .type            = AVMEDIA_TYPE_AUDIO, },
+                                  { .name = NULL}},
+};