path: root/ui/rtp_stream.h
AgeCommit message (Collapse)AuthorFilesLines
2016-12-18SIP/SDP, RTP: Dissectors shows information about ED-137 related states of ↵Jiri Novak1-0/+1
radio in info column/VoIP call flow Based on EUROCAE ED-137B specification: ED-137B, Part 1: RADIO, INTEROPERABILITY STANDARDS FOR VOIP ATM COMPONENTS https://boutique.eurocae.net/eshop/catalog/index.php Bug: 13252 Change-Id: Ifab1aaf47e3405fcd46309167237f11ce2d7e2ff Reviewed-on: https://code.wireshark.org/review/19302 Petri-Dish: Michael Mann <mmann78@netscape.net> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2016-06-23RTP player: increase the maximum number of silence frames to 30 minutes worth.Jeff Morriss1-0/+2
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>
2015-11-01Disable RTP player debug logs that were presumably left activated by mistakePascal Quantin1-1/+1
Change-Id: Ieeca052bba14735447cdd6e53de8ed7cda69a27f Reviewed-on: https://code.wireshark.org/review/11480 Reviewed-by: Pascal Quantin <pascal.quantin@gmail.com>
2015-10-27Add jitter logic to RtpAudioStream.Gerald Combs1-1/+1
Copy the jitter logic from rtp_player.c to rtp_audio_stream.cpp. This still isn't correct but the RTP player should now be complete enough to start looking at the bug list at the top of rtp_player_dialog.cpp. Disable timing and jitter controls while we're playing while we're here. Fixes bug 11635. Bug: 11635 Change-Id: Ie583ade522702cbe1bbcea4475a535caa1d74fa2 Reviewed-on: https://code.wireshark.org/review/11295 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-21Split RTP player tapping, decoding, and plotting.Gerald Combs1-1/+1
In RtpAudioStream split tapping+decoding into separate member functions. Store RTP payloads in memory. In RtpPlayerDialog split tapping+plotting. This more closely resembles what we're doing in the GTK+ UI and paves the way for jitter support and other changes. Change-Id: I244c225cec8930545622e6582b7be35ebe45b237 Reviewed-on: https://code.wireshark.org/review/11195 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-02Qt: Initial RTP playback.Gerald Combs1-5/+14
Note the "initial". This is woefully incomplete. See the "to do" lists below and in the code. This differs a bit from the GTK+ version in that you specify one or more streams to be decoded. Instead of showing waveforms in individual widgets, add them all to a single QCustomPlot. This conserves screen real estate and lets us more easily take advantage of the QCP API. It also looks better IMHO. Change a bunch of checks for QtMultimediaWidgets to QtMultimedia. We probably won't use the widgets until we make 5.0 our minimum Qt version and plain old QtMultimedia lets us support Qt 4 more easily (in theory at least). Add resampling code from libspeex. I initially used this to resample each packet to match the preferred rate of our output device, but this resulted in poorer audio quality than expected. Leave it in and use to create visual samples for QCP and to match rates any time the rate changes. The latter is currently untested. Add some debugging macros. Note that both the RTP player and RTP analysis dialogs decode audio data using different code. Note that voip_calls_packet and voip_calls_init_tap appear to be dead code. To do: - Add silence frames where needed. - Implement the jitter buffer. - Implement the playback timing controls. - Tapping / scanning streams might be too slow. Change-Id: I20dd3b66d3df53c9b1f3501262dc01458849f6b4 Bug: 9007 Reviewed-on: https://code.wireshark.org/review/10458 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-10-01Fix crashes related to RTP Streams analysisPeter Wu1-0/+2
The data that describes RTP streams become invalid when packets are re-dissected. This results in a crash in GTK when the "RTP Analyse" option is used and and a crash in Qt when the display filter is changed while the RTP Streams dialog is open. Fix this by adding a tap_reset callback (modelled after mcaststream) to the RTP tap listener that allows the GTK+ and Qt dialogs to clear the displayed list of RTP streams. Bug: 10016 Change-Id: I7478678db63d7ac8110c44c163844e9f66fad9e9 Reviewed-on: https://code.wireshark.org/review/10728 Reviewed-by: Peter Wu <peter@lekensteyn.nl> Petri-Dish: Peter Wu <peter@lekensteyn.nl> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Michael Mann <mmann78@netscape.net>
2015-09-03Move IAX2 analysis to the ui directory.Gerald Combs1-1/+1
Rename ui/gtk/iax2_analysis.h to ui/tap-iax2-analysis.h. Move iax2_packet_analyse to ui/tap-iax2-analysis.c. Rename rtp_analysis.h to tap-rtp-analysis.h to match IAX2. Change-Id: Ice7e9ad0d7bf62d631850089c880ec09a3e101dd Reviewed-on: https://code.wireshark.org/review/10375 Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-08-18UDP multicast stream dialog.Gerald Combs1-1/+0
Add the UDP multicast stream dialog. Abuse TapParameterDialog a bit more so that we can edit parameters. Remove some unused struct members and an unused function. Change-Id: I962c70344e792f0959527e4bcba8a20bd7e8acf9 Reviewed-on: https://code.wireshark.org/review/10084 Petri-Dish: Gerald Combs <gerald@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-07-13RTP updates.Gerald Combs1-17/+0
Merge rtp_sample_header_t into rtp_sample_t. That's the only place it was used. Note that rtp_sample_t is used for writing rtpdump files. Move the rtp_sample_t definition to tap-rtp-common.c. Rename it to rtpdump_info_t. Make rtp_write_sample static. Change-Id: I04e7428f634efa87a98e5d6c82a354f94ab1765d Reviewed-on: https://code.wireshark.org/review/9629 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-05-20Start exposing the filter field of a tap listener to the RTP GUI APIs.Michael Mann1-2/+2
A tap listener has the ability to apply a filter (typically the display filter). Add a parameter to RTP GUI API functions to allow them to pass in a filter. Bug: 996 Change-Id: Ib184dfb023be5d1d24a0d842b4039311426b5293 Reviewed-on: https://code.wireshark.org/review/8468 Petri-Dish: Michael Mann <mmann78@netscape.net> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Anders Broman <a.broman58@gmail.com>
2015-02-07Fix RTP crash on RTP analysis attemptPeter Wu1-1/+2
The tap listener was handling rtpstream_tapinfo_t* types while other users was expecting a GList* instead. Fix this and avoid future confusion by replacing void* pointers. Ping-Bug: 10714 Change-Id: I66f62eaaed4a529714264bbf4e7ad1e72b46ce5a Reviewed-on: https://code.wireshark.org/review/6997 Reviewed-by: Gerald Combs <gerald@wireshark.org>
2015-01-30Qt: Add the RTP Streams dialog.Gerald Combs1-14/+38
Add keyboard shortcuts. Note that not all of the buttons made it from GTK+. Add a "Go to setup frame" option. Move rtp_streams.c from ui/gtk to ui. Add a help URL for RTP analysis (which needs to be split into streams + analysis). Fix RTP stream packet marking. Change-Id: Ifb8192ff701a933422509233d76461a46e459f4f Reviewed-on: https://code.wireshark.org/review/6852 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-12-20Consolidate RTP stream structs.Gerald Combs1-13/+19
Consolidate the three different RTP stream structs in ui/rtp_stream.h, ui/gtk/rtp_player.c, and ui/voip_calls.c into one. Make the member names a bit more consistent. Document what each GList contains. Use nstime_t for timestamps since that's what we get from the frame data. Use g_new0 to initialize our structs. Change-Id: I2b3f8f2051394a6a98a5c7bc49c117f07161d031 Reviewed-on: https://code.wireshark.org/review/5843 Petri-Dish: Gerald Combs <gerald@wireshark.org> Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org> Reviewed-by: Gerald Combs <gerald@wireshark.org>
2014-10-14Always put editor-modelines at the end of the file ...Bill Meier1-2/+2
... to ensure that there are no potential issues with respect to editors limiting the number of lines scanned at the end of the file when checking for editor modelines. Change-Id: Ic85cbb108bb5159d6ec4116fea11f5eebb4e44a4 Reviewed-on: https://code.wireshark.org/review/4688 Reviewed-by: Bill Meier <wmeier@newsguy.com>
2014-10-12Add editor modelines; Adjust whitespace as needed.Bill Meier1-33/+45
Change-Id: I4da7b335d905dbca10bbce03aa88e1cdeeb1f8ad Reviewed-on: https://code.wireshark.org/review/4626 Reviewed-by: Bill Meier <wmeier@newsguy.com>
2014-03-04Remove all $Id$ from top of fileAlexis La Goutte1-2/+0
(Using sed : sed -i '/^ \* \$Id\$/,+1 d') Fix manually some typo (in export_object_dicom.c and crc16-plain.c) Change-Id: I4c1ae68d1c4afeace8cb195b53c715cf9e1227a8 Reviewed-on: https://code.wireshark.org/review/497 Reviewed-by: Anders Broman <a.broman58@gmail.com>
2013-12-23From Ville Skyttä: Spelling FixesBill Meier1-1/+1
https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=9591 svn path=/trunk/; revision=54387
2013-11-03Bluetooth: AVDTP: Add support for Content Protection type SCMS-T (and some ↵Michael Mann1-1/+1
minor cleanup). Bug 7893 (https://bugs.wireshark.org/bugzilla/show_bug.cgi?id=7893) From Michal Labedzki svn path=/trunk/; revision=53065
2013-10-24Make things compile again.Jörg Mayer1-0/+155
svn path=/trunk/; revision=52828