when the gtk version is too old (for the 2,14,0 case).
Remove all old access methods that were guarded by 2,14,0 tests.
Feel free to do the same for newer guards :-)
svn path=/trunk/; revision=38147
Use event callback function user_data arg instead of using a property;
Replace an 'if(...) {... exit(10);}' by a g_assert;
Remove several unneeded 'if(...) {... exit(10);}' groups of code;
Do whitespace cleanuip.
svn path=/trunk/; revision=37159
from a double to a time_t. I removed nstime_to_secs() and grab the
seconds portion of the nstime (which is a time_t), since that's all the
precision needed in the code right now anyway.
svn path=/trunk/; revision=35293
call's packet flow along the x-axis. Add " s" to the end of each number
to give the user an idea those numbers are seconds.
svn path=/trunk/; revision=35278
Flag when a packet has been dropped by the jitter buffer in the audio player,
by showing:
D dropped packet
W wrong timestamp
S silence added by wireshark
To show when audio 'glitches' may have come from the processing the received
packets through the jitter buffer.
svn path=/trunk/; revision=35192
https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=4119 :
Never insert more than 1000 silence frames (e.g., if the sequence number jumps
massively). There may be a better way, but at least now we won't crash.
Leave a comment in the code indicating this.
svn path=/trunk/; revision=32304
Therefore opening the default stream may fail in the precense of usable devices, on other Host API's.
If the default stream fails to open iterate among the Host API's to find one with a default device.
svn path=/trunk/; revision=31318
When audio samples have to be dropped or silence samples inserted to reflect
the timestamp there is no indication of these problems on the display.
I propose that such problems be indicated on the waveform display by the use of
amber coloration and that the number of incorrect timestamps be listed
svn path=/trunk/; revision=28451
Added a new checkbox for the RTP player to use the RTP timing instead of the
arriving time of the packets. This is useful when the RTP is being tunneled
(e.g. EtherIP), so the original timing is missing.
In some cases when encapsulating the original IP/RTP over EtherIP, the
encapsulated data comes in chunks of 1sec, so it is useful to use the RTP
timing to play it. In these cases we can not simulate jitter buffer, so this is
disabled when checking the new "Use RTP timestamp" option.
svn path=/trunk/; revision=25293
This patch provides a new RTP Player preferences dialog. It allows one to
select the maximum number of visable channels in the RTP Player window. The
default is four (4) channels which is the maximum number of usable channels
that the RTP Player can display and still have access to the bottom row buttons
on a 1024*768 resolution display. Specifying a value less than 1 or greater than
10 will be result in the RTP Player displaying the default 4 channels.
svn path=/trunk/; revision=24112