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, 14 insertions, 12 deletions

diff --git a/libavfilter/audio.c b/libavfilter/audio.c
index 0ebec3c2d0..f3eebbfdae 100644
--- a/libavfilter/audio.c
+++ b/libavfilter/audio.c
@@ -150,19 +150,19 @@ fail:
     return NULL;
 }
 
-static void default_filter_samples(AVFilterLink *link,
-                                   AVFilterBufferRef *samplesref)
+static int default_filter_samples(AVFilterLink *link,
+                                  AVFilterBufferRef *samplesref)
 {
-    ff_filter_samples(link->dst->outputs[0], samplesref);
+    return ff_filter_samples(link->dst->outputs[0], samplesref);
 }
 
-void ff_filter_samples_framed(AVFilterLink *link,
-                              AVFilterBufferRef *samplesref)
+int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
 {
-    void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+    int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
     AVFilterPad *dst = link->dstpad;
     int64_t pts;
     AVFilterBufferRef *buf_out;
+    int ret;
 
     FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
 
@@ -193,21 +193,22 @@ void ff_filter_samples_framed(AVFilterLink *link,
 
     link->cur_buf = buf_out;
     pts = buf_out->pts;
-    filter_samples(link, buf_out);
+    ret = filter_samples(link, buf_out);
     ff_update_link_current_pts(link, pts);
+    return ret;
 }
 
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 {
     int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
     AVFilterBufferRef *pbuf = link->partial_buf;
     int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
+    int ret = 0;
 
     if (!link->min_samples ||
         (!pbuf &&
          insamples >= link->min_samples && insamples <= link->max_samples)) {
-        ff_filter_samples_framed(link, samplesref);
-        return;
+        return ff_filter_samples_framed(link, samplesref);
     }
     /* Handle framing (min_samples, max_samples) */
     while (insamples) {
@@ -218,7 +219,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
             if (!pbuf) {
                 av_log(link->dst, AV_LOG_WARNING,
                        "Samples dropped due to memory allocation failure.\n");
-                return;
+                return 0;
             }
             avfilter_copy_buffer_ref_props(pbuf, samplesref);
             pbuf->pts = samplesref->pts +
@@ -234,10 +235,11 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
         insamples               -= nb_samples;
         pbuf->audio->nb_samples += nb_samples;
         if (pbuf->audio->nb_samples >= link->min_samples) {
-            ff_filter_samples_framed(link, pbuf);
+            ret = ff_filter_samples_framed(link, pbuf);
             pbuf = NULL;
         }
     }
     avfilter_unref_buffer(samplesref);
     link->partial_buf = pbuf;
+    return ret;
 }