Remove the editor modeline blocks from the source files in ui that use 4
space indentation by running
perl -i -p0e 's{ \n+ /[ *\n]+ editor \s+ modelines .* shiftwidth= .* \*/ \s+ } {\n}gsix' $( ag -l shiftwidth=4 $( ag -g '\.(c|cpp|h|m|mm)') )
This gives us one source of indentation truth for these files, and it
*shouldn't* affect anyone since
- These files match the default in our top-level .editorconfig.
- The one notable editor that's likely to be used on these files and
*doesn't* support EditorConfig (Qt Creator) defaults to 4 space
indentation.
Retap and UI response are much faster when many RTP streams are
processed. RTP Streams/Analyse 1000+, RTP Player 500+.
Changes:
- RTP streams are searched with hash, not by iterating over list.
- UI operations do not redraw screen after every change, just after all
changes. UI is locked when rereading packets.
- Sample list during RTP decoding is stored in memory so wireshark uses
just half of opened files for audio decoding than before.
- Analysis window checkbox area is limited in height
- Dialogs shows shows count of streams, count of selected streams and
count of unmuted streams
- Documentation extended with chapter about RTP decoding parameters
- Documentation extended with performance estimates
Changes:
- all waveforms has common scale therefore louder/quiter signal is visible
- when stream/streams are deleted from view, Y axis is rescaled and
waveforms are rearranged to reuse empty space
If a header declares a function, or anything else requiring the extern
"C" decoration, have it wrap the declaration itself; don't rely on the
header itself being included inside extern "C".
VoIP Calls dialog and RTP Streams dialog has now option to apply display
filter dialog during processing packets.
Filter checkbox is activated during dialog open when display filter is active.
New field apply_display_filter had to be added to voip_calls_tapinfo_t and
_rtpstream_tapinfo/rtpstream_tapinfo_t structures.
Patch adds ability to set start of audio play by double clicking on waveform.
Patch fixes unreported issue with placing waveform at incorrect place when switched relative/absolute time mode (check/uncheck Time of Day).
Change-Id: Ib8ce24aea870e2443e033afbb6d6e9fbcf222431
Reviewed-on: https://code.wireshark.org/review/35621
Petri-Dish: Anders Broman <a.broman58@gmail.com>
Tested-by: Petri Dish Buildbot
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Change all wireshark.org URLs to use https.
Fix some broken links while we're at it.
Change-Id: I161bf8eeca43b8027605acea666032da86f5ea1c
Reviewed-on: https://code.wireshark.org/review/34089
Reviewed-by: Guy Harris <guy@alum.mit.edu>
*rtp_stream* -> rtpstream to follow common name
Change-Id: I381bc1cdb8206c5cfe67e94dd7fb1a5cb25f9c16
Reviewed-on: https://code.wireshark.org/review/28394
Petri-Dish: Anders Broman <a.broman58@gmail.com>
Tested-by: Petri Dish Buildbot
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Changes:
- rtpstream_id_t is introduced and its related functions. It encapsulates comparsion of two rtpstreams.
- dest_* renamed to dst_*
- src_port and dst_port are 16bits only.
- sharkd_session.c use common id functions
- IAX2 part related to RTP updated to common *id* function
Change-Id: Id38728a4e5d80363480c7ce42ff9c6eaad069686
Reviewed-on: https://code.wireshark.org/review/28340
Petri-Dish: Anders Broman <a.broman58@gmail.com>
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Changes:
- rtpstream_packet renamed to rtpstream_packet_cb to follow *_cb pattern
- variables/types used in iax2_analysis_dialog were created as copy of *rtp* ones, but names were left as *rtp* -> *iax2*
- struct _rtp_stream_info replaced with rtp_stream_info_t
- there was tap-rtp-analysis.h, but no tap-rtp-analysis.c - related content was moved from tap-rtp-common.c
- *rtp_stream* functions renamed to *rtpstream*
- renamed rtp_stream_info_t to rtpstream_info_t to follow *rtpstream* pattern.
- renamed ui/rtp_stream.c rtpstream_draw -> rtpstream_draw_cb
Change-Id: Ib11ff5367cc464ea1b0c73432bc50b0eb9cd203e
Reviewed-on: https://code.wireshark.org/review/28299
Reviewed-by: Anders Broman <a.broman58@gmail.com>
The first is deprecated, as per https://spdx.org/licenses/.
Change-Id: I8e21e1d32d09b8b94b93a2dc9fbdde5ffeba6bed
Reviewed-on: https://code.wireshark.org/review/25661
Petri-Dish: Anders Broman <a.broman58@gmail.com>
Petri-Dish: Dario Lombardo <lomato@gmail.com>
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Change-Id: I6b05399395bcc35e59b73b4030ba4a05711a7b1a
Reviewed-on: https://code.wireshark.org/review/25565
Petri-Dish: Michael Mann <mmann78@netscape.net>
Reviewed-by: Michael Mann <mmann78@netscape.net>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
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>
... 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>
(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>
João Valverde