Merge remote-tracking branch 'qatar/master'

* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>

1 files changed, 13 insertions, 9 deletions

diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
index 6dad3db0d0..7f83750fa1 100644
--- a/libavfilter/af_amix.c
+++ b/libavfilter/af_amix.c
@@ -305,9 +305,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
     if (s->next_pts != AV_NOPTS_VALUE)
         s->next_pts += nb_samples;
 
-    ff_filter_samples(outlink, out_buf);
-
-    return 0;
+    return ff_filter_samples(outlink, out_buf);
 }
 
 /**
@@ -448,31 +446,37 @@ static int request_frame(AVFilterLink *outlink)
     return output_frame(outlink, available_samples);
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     MixContext       *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
-    int i;
+    int i, ret = 0;
 
     for (i = 0; i < ctx->nb_inputs; i++)
         if (ctx->inputs[i] == inlink)
             break;
     if (i >= ctx->nb_inputs) {
         av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
-        return;
+        ret = AVERROR(EINVAL);
+        goto fail;
     }
 
     if (i == 0) {
         int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
                                    outlink->time_base);
-        frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+        ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+        if (ret < 0)
+            goto fail;
     }
 
-    av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
-                        buf->audio->nb_samples);
+    ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+                              buf->audio->nb_samples);
 
+fail:
     avfilter_unref_buffer(buf);
+
+    return ret;
 }
 
 static int init(AVFilterContext *ctx, const char *args)