configure: add decimate filter dependency on avcodec

[ffmpeg] / libavfilter / af_amerge.c

diff --git a/libavfilter/af_amerge.c b/libavfilter/af_amerge.c
index 802188d028ed0c585b36f4ce71e2a2b03f9709e2..1b29e88757432b5dc4c3498ec19aa38eef58deb5 100644 (file)
--- a/libavfilter/af_amerge.c
+++ b/libavfilter/af_amerge.c
@@ -11,7 +11,7 @@
  * 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 General Public License for more details.
+ * 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
@@ -46,10 +46,11 @@ typedef struct {
 } AMergeContext;
 
 #define OFFSET(x) offsetof(AMergeContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
 
 static const AVOption amerge_options[] = {
     { "inputs", "specify the number of inputs", OFFSET(nb_inputs),
-      AV_OPT_TYPE_INT, { .dbl = 2 }, 2, SWR_CH_MAX },
+      AV_OPT_TYPE_INT, { .dbl = 2 }, 2, SWR_CH_MAX, FLAGS },
     {0}
 };
 
@@ -157,7 +158,7 @@ static int config_output(AVFilterLink *outlink)
     }
     av_bprintf(&bp, " -> out:");
     av_bprint_channel_layout(&bp, -1, ctx->outputs[0]->channel_layout);
-    av_log(ctx, AV_LOG_INFO, "%s\n", bp.str);
+    av_log(ctx, AV_LOG_VERBOSE, "%s\n", bp.str);
 
     return 0;
 }
@@ -212,7 +213,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
     }
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
 {
     AVFilterContext *ctx = inlink->dst;
     AMergeContext *am = ctx->priv;
@@ -232,7 +233,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
     for (i = 1; i < am->nb_inputs; i++)
         nb_samples = FFMIN(nb_samples, am->in[i].nb_samples);
     if (!nb_samples)
-        return;
+        return 0;
 
     outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
     outs = outbuf->data[0];
@@ -241,13 +242,13 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
         ins[i] = inbuf[i]->data[0] +
                  am->in[i].pos * am->in[i].nb_ch * am->bps;
     }
+    avfilter_copy_buffer_ref_props(outbuf, inbuf[0]);
     outbuf->pts = inbuf[0]->pts == AV_NOPTS_VALUE ? AV_NOPTS_VALUE :
                   inbuf[0]->pts +
                   av_rescale_q(am->in[0].pos,
                                (AVRational){ 1, ctx->inputs[0]->sample_rate },
                                ctx->outputs[0]->time_base);
 
-    avfilter_copy_buffer_ref_props(outbuf, inbuf[0]);
     outbuf->audio->nb_samples     = nb_samples;
     outbuf->audio->channel_layout = outlink->channel_layout;
 
@@ -285,7 +286,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
             }
         }
     }
-    ff_filter_samples(ctx->outputs[0], outbuf);
+    return ff_filter_samples(ctx->outputs[0], outbuf);
 }
 
 static av_cold int init(AVFilterContext *ctx, const char *args)
@@ -334,4 +335,5 @@ AVFilter avfilter_af_amerge = {
           .request_frame    = request_frame, },
         { .name = NULL }
     },
+    .priv_class = &amerge_class,
 };