aboutsummaryrefslogtreecommitdiffstats
diff options
context:
space:
mode:
authorJeff Morriss <jeff.morriss.ws@gmail.com>2016-06-17 20:41:31 -0400
committerMichael Mann <mmann78@netscape.net>2016-06-23 03:30:39 +0000
commitfea6e738bb843eea7667873b7fc3c7f35ae2b32a (patch)
tree288b2bd03056f4362f8b333aa982b19e51380845
parentbcae998048190ce70ae9d0cbc05dc61e9e66c3b4 (diff)
RTP player: increase the maximum number of silence frames to 30 minutes worth.
The BadAlloc X11 crash I reported in bug 4119 (which is why the limit was as low as it was) has long since been fixed thanks to bug 2630/I71e1bd2f9a62792db06ce887e2bbe7a96d110e0a so we can now deal with more silence frames. Change-Id: I0127381e71e497560e0f23af04f9d96af1ed6335 Ping-Bug: 5902 Ping-Bug: 4119 Ping-Bug: 2270 Reviewed-on: https://code.wireshark.org/review/16003 Petri-Dish: Michael Mann <mmann78@netscape.net> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
-rw-r--r--ui/gtk/rtp_player.c1
-rw-r--r--ui/qt/rtp_audio_stream.cpp2
-rw-r--r--ui/rtp_stream.h2
3 files changed, 3 insertions, 2 deletions
diff --git a/ui/gtk/rtp_player.c b/ui/gtk/rtp_player.c
index bcbd2c9f41..178e9f8a35 100644
--- a/ui/gtk/rtp_player.c
+++ b/ui/gtk/rtp_player.c
@@ -641,7 +641,6 @@ decode_rtp_stream(rtp_stream_info_t *rsi, gpointer ptr)
/* Fix for bug 4119/5902: don't insert too many silence frames.
* XXX - is there a better thing to do here?
*/
-#define MAX_SILENCE_FRAMES 240000
if (silence_frames > MAX_SILENCE_FRAMES)
silence_frames = MAX_SILENCE_FRAMES;
diff --git a/ui/qt/rtp_audio_stream.cpp b/ui/qt/rtp_audio_stream.cpp
index 7bd072ef52..7154d8df7a 100644
--- a/ui/qt/rtp_audio_stream.cpp
+++ b/ui/qt/rtp_audio_stream.cpp
@@ -175,7 +175,7 @@ static const int sample_bytes_ = sizeof(SAMPLE) / sizeof(char);
/* Fix for bug 4119/5902: don't insert too many silence frames.
* XXX - is there a better thing to do here?
*/
-static const int max_silence_samples_ = 240000;
+static const int max_silence_samples_ = MAX_SILENCE_FRAMES;
void RtpAudioStream::decode()
{
if (rtp_packets_.size() < 1) return;
diff --git a/ui/rtp_stream.h b/ui/rtp_stream.h
index 746fb694d0..f4b46dd63f 100644
--- a/ui/rtp_stream.h
+++ b/ui/rtp_stream.h
@@ -169,6 +169,8 @@ gboolean rtp_stream_info_is_reverse(const rtp_stream_info_t *stream_a, rtp_strea
*/
void rtpstream_mark(rtpstream_tapinfo_t *tapinfo, capture_file *cap_file, rtp_stream_info_t* stream_fwd, rtp_stream_info_t* stream_rev);
+#define MAX_SILENCE_FRAMES 14400000
+
#ifdef __cplusplus
}
#endif /* __cplusplus */