From 6d1bc935a6982f045298dc074f0867c2431c3d24 Mon Sep 17 00:00:00 2001 From: Bent Bisballe Nyeng Date: Wed, 1 Oct 2014 18:56:24 +0200 Subject: Use buffer size from audiobackend in output code. --- src/inputstreamer.cc | 7 +++++-- 1 file changed, 5 insertions(+), 2 deletions(-) (limited to 'src/inputstreamer.cc') 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); -- cgit v1.2.3