- RTP analysis and player dialogs are modal whereas they were not modal in GTK
UI
Only fixed calling up the RTP analysis window being modal when called from the
RTP streams dialog as it is modeless when called from the menu in the main
window with a stream packet selected.
- Action 'Next problem packet' triggers an infinite loop when there are no
'problem packets' and first packet is selected when calling the action.
- Crosshairs not implemented in RTP player/crosshairs context menu item not
working.
Context menu item commented out.
bug: 13500
Change-Id: I0ba954f895b85605462c316e4426280ec4d437b2
Reviewed-on: https://code.wireshark.org/review/20660
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Remove unnecessary Q_DECLARE_METATYPE macros and replace calls
to QVariant conversions with VariantPointer where necessary
Change-Id: Ia4690590095f930bf94644197de7fa30b00ee7ec
Reviewed-on: https://code.wireshark.org/review/19611
Petri-Dish: Roland Knall <rknall@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Roland Knall <rknall@gmail.com>
QComboBox::setCurrentText() method is available in Qt5.x.
Older versions code won't compile with it.
Bug: 13235
Change-Id: Ia2e2713fefe0f2be01a0b77ff1ac39c9162fd0d1
Reviewed-on: https://code.wireshark.org/review/19219
Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Peter Wu <peter@lekensteyn.nl>
The CONSTANT attribute indicates that the same value will be returned
every time. That isn't the case here so remove it.
Change-Id: Ie7451e6aabcb4fa1a6960762d96ad190f32b3d7a
Reviewed-on: https://code.wireshark.org/review/19130
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Michael Mann <mmann78@netscape.net>
Add a combobox for selecting the output device and populate it with our
available devices. Let the user know if our output format isn't
supported.
Ping-Bug: 13105
Change-Id: I299c7d0f191bb66d93896338036000e2c377781f
Reviewed-on: https://code.wireshark.org/review/19046
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Reviewed-by: Peter Wu <peter@lekensteyn.nl>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Integrate the Spandsp library for G.722 and G.726 support. Adds support
for G.722 and all eight variants of G.726.
Note: this also fixes a crash in Qt (buffer overrun, reading too much
data) caused by confusion of the larger output buffer (resample_buff)
with the smaller input buffer (decode_buff). It was not triggered before
because the sample rate was always 8k, but with the addition of the new
codecs, a different sample rate became possible (16k).
Fix also a crash which occurs when the RTP_STREAM_DEBUG macro is enabled
and the VOIP Calls dialog is opened (the begin frame, start_fd, is not
yet known and therfore a NULL dereference could occur).
Passes testing (plays normally without bad RTP timing errors) with
SampleCaptures files: sip-rtp-g722.pcap and sip-rtp-g726.pcap. Tested
with cmake (Qt), autotools (Qt and GTK+) with ASAN enabled.
Bug: 5619
Change-Id: I5661908d193927bba50901079119eeff0c04991f
Reviewed-on: https://code.wireshark.org/review/18939
Petri-Dish: Peter Wu <peter@lekensteyn.nl>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Alexis La Goutte <alexis.lagoutte@gmail.com>
Reviewed-by: Peter Wu <peter@lekensteyn.nl>
make sure that generated include files are picked up only from the
directories set by -I (or /I), not from the current directory
if we use #include "file.h", Visual Studio searches for file.h in the
same diretory as the source file that includes file.h
if we do an out-of-tree build with cmake and the source directory
contains files from an in-tree build (done with autotools), we might end
up including the wrong file
Change-Id: Iaaed2626258b6ff0c12485fe3f436bd03bbb5adf
Reviewed-on: https://code.wireshark.org/review/15873
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Make sure audio_stream_ is non-NULL before we try to use it. Delete
audio_stream_ more gracefully and add a note about mutexes on OS X and
Windows.
Bug: 12166
Change-Id: I12e76c49e631bc1de813c5c7d82c7d928c71237e
Reviewed-on: https://code.wireshark.org/review/15759
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Add GeometryStateDialog class to handle load and save dialog geometry.
The QDialog class name will be used as window name. For shared
classes the UAT name or the statistics title or abbr will be used.
Change-Id: I5a019598307fb3861518f41e733de834788184d8
Reviewed-on: https://code.wireshark.org/review/14139
Reviewed-by: Stig Bjørlykke <stig@bjorlykke.org>
When dragging the UI, this somehow causes a great lag. Then by
spam-clicking on the Stop button, a double free seems to occur.
Fix this by moving the audio cleanup to the outputStateChanged callback
as documented at https://doc.qt.io/qt-5/qaudiooutput.html. Note that
calling stop() in the IdleState also triggers a change event, resulting
in the desired cleanup.
Stop streams before the dialog is closed (via accept/reject). This
*cannot* be done in the destrutor of RtpPlayerDialog because destructing
QAudioOutput processes events from the event queue, resulting in
preature destruction of other objects... crash.
Change-Id: I6bfb33c9396e9bc1ffd346519d22390a97b6bdaf
Reviewed-on: https://code.wireshark.org/review/11894
Petri-Dish: Peter Wu <peter@lekensteyn.nl>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
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>
It looks like QPushButton gets included via ui_rtp_player_dialog.h in Qt
5 but not in Qt 4. Make sure we include it explicitly whether or not
QT_MULTIMEDIA_LIB is defined.
Change-Id: I8203a1cc6f7b9beef0f749b93836a75885f85edd
Reviewed-on: https://code.wireshark.org/review/10962
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Rename the "Play Call" button to "Play Streams". Move the button
creation code to a common routine. Use it to add a "Play Streams" button
to the RTP Stream Analysis, similar to the GTK+ UI.
Don't restrict RTP to IPv[46] as suggested by Michal. I don't have any
RTP-over-Bluetooth captures so I can't test this directly.
Change-Id: I4703cac1d5bf5b3ff0255d36da2c5164feb0547d
Reviewed-on: https://code.wireshark.org/review/10888
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Change-Id: I6298b3de5f0a1cb988014ff16082eaf8c2a3c3c0
Reviewed-on: https://code.wireshark.org/review/10786
Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Michael Mann <mmann78@netscape.net>
Change-Id: I28ae077be535f32ef81ac370d6782033f219017d
Reviewed-on: https://code.wireshark.org/review/10777
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: Anders Broman <a.broman58@gmail.com>
Reviewed-by: Michael Mann <mmann78@netscape.net>
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>
Erik de Jong