Use buffer size from audiobackend in output code.

1 files changed, 5 insertions, 2 deletions

diff --git a/src/inputstreamer.cc b/src/inputstreamer.cc
index c7e5986..3eef76a 100644
--- a/src/inputstreamer.cc
+++ b/src/inputstreamer.cc
@@ -142,13 +142,16 @@ void InputStreamer::run()
     lrtp_unpack(lrtp, packet, packetsize);
     int n = 0;
     int ret;
-    char frame[512 * 1024]; // 512kbyte should be enough for even the larges
+    char frame[512 * 1024 * 4]; // 512kbyte should be enough for even the larges
                             //  JPEG frames...
     unsigned int csrc;
     unsigned int ts;
     while((ret = lrtp_dequeue_frame(lrtp, frame, sizeof(frame), &csrc, &ts))
               != 0) {
-      if(ret < 0) printf("I:lrtp_dequeue_frame: %d\n", ret);
+      if(ret < 0) {
+        printf("I:lrtp_dequeue_frame: %d (frame skipped)\n", ret);
+        continue;
+      }
       if(csrc == CSRC_V) {
         // Video frame
         Frame f(frame, ret);