Changes:
- audio rates are fixed, I selected common ones
- fixed signal names - they didn't matched expected autogenerated ones
- first play notification delay on Qt6 compensated
RTP dialogs can stay opened, therefore calls of its functions are
protected by locks. There was issue that same mutex was used during
construction of the dialog and calling functions. It created possible
deadlock.
Change separates lock used for dialog creation and lock for function calls.
When function call lock is locked, new calls are ignored and warning is
printed to STDERR. Showing a dialog with warning looks too intrusive to me.
Fixes#18025
Wireshark successfully compiles on Windows with Qt6.2 with following
cmake options:
-DUSE_qt6=ON -DDISABLE_ERROR=ON
QCustomPlot QT 6.2 patch is taken from QCustomPlot forum post by miccs.
In order to be able to defer solving all Qt6 API differences at once
I tried to reactivate the QT_MULTIMEDIA_LIB feature. I managed to fix
most problems but one problem remains in both Qt5 and Qt6 builds.
Without Qt[56]Multimedia, the following error exceeds my non-existing
C++ knowledge:
jmayer/work/wireshark/git/ui/qt/rtp_player_dialog.cpp:154:18: error: out-of-line definition of 'RtpPlayerDialog' does not match any declaration in 'RtpPlayerDialog'
RtpPlayerDialog::RtpPlayerDialog(QWidget &parent, CaptureFile &cf, bool capture_running) :
^~~~~~~~~~~~~~~
Singletons moved from main_window to each class's static open<NameOfClass>
method:
- RtpPlayerDialog
- RtpStreamDialog
- VoipCallsDialog
- RtpAnalysisDialog
Fixed issue with selecting RTP stream in sequence dialog. When user
selected a stream and moved mouse to Rtp Player button and pressed it,
incorrect RTP stream was sent to it.
When button is pressed or triggered by shortcut, it opens same
window as before.
User can click small arrow next to button and it open menu with all
new actions e.g. Set/Add/Remove for RTP Player.
Documentation updated.
Code is NOT able to do VAD (Voice Activity Detection) so audio silence
(sequence of equal samples) nor noise are not recognized as silence. Just
missing RTP (Confort Noise, interupted RTP, ...) and muted streams are
recognized as silence for this feature.
User can control duration of shortest silence to skip.
Updated documentation.
Audio for play is now decoded and stored without silence parts.
Changes:
- ui/qt/utils/rtp_audio_file.cpp created to handle silence skipping
- ui/qt/rtp_audio_stream.cpp refactored to support it
- Fixed issue with exporting streams: File synchronized export was missing
leading silence.
- No line is shown in waveform graph if there is silence
Changes:
- RTP Player added to Telephony/RTP menu.
- When openning RTP Analysis or RTP Player from RTP menu, just selected
stream is added. When Ctrl is hold during opening, reverse stream is
searched and added too.
- RTP Player: Added tool to select/deselect all inaudible streams
- RTP Player: Added Prepare Filter button
- RTP Player: Added Analyze button
- RTP Analysis: Added Prepare Filter button
- documentation updated
Code changes:
- RTP Player::rescanPacket() is not fired multiple times during rate change and during dialog creation
- Error shown in RTP player is cleared after every new decode of streams
- RTP Player handles case when Qt do not emit stop stream event
- "Select" menu code unified between dialogs>
- RTP Player: Audio routing menu unified
- buttons are connected to actions by signals()
- Analyze dialog is called by list of rtpstream_id, not rtpstream_info
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:
- Fixed issue with hanging the player. The issue is in Qt - internal
Mutex is locked when Qt calls outputStateChanged() and you can't call
any other action on the audio object
- Fixed issue when play marker stream was running forever
- Removed !1855 because it introduces delay on play on Windows platform
Features:
- saves multiple streams (all selected and unmuted)
- saves streams same way they are played (jitter buffer, sampling, ...)
- only streams with audio (play rate >0) are exported
- streams with play rate == 0 are silently ignored even selected for
export
- all exported streams must use same play rate (user can change it
before save)
Features:
- saves multiple streams (all selected and unmuted)
- saves streams same way they are played (jitter buffer, sampling, ...)
- only streams with audio (play rate >0) are exported
- streams with play rate == 0 are silently ignored even selected for
export
- all exported streams must use same play rate (user can change it
before save)
Tool allows a user to replay at specific rate when there is any issue
with autodetected rate by payloads.
Offered rates are provided by selected audio device.
Changes:
- refactored main_dialog handling of telephony dialogs
- RTP Player dialog is nonmodal now and can be left open
- it is possible to issue three actions on RTP Player dialog from other
dialogs (other dialog have selected set of RTP streams before action)
- replace - removes existing streams from RTP dialog and shows new set
- add - adds new set to existing list in RTP dialog
- remove - remove streams in set from list in RTP dialog
- Sequence Dialog:
- was modified to hold rtpstream_info_t for RTP streams
- added Play button
- VoIP features (RTP Play button, select/deselect RTP stream) are
disabled after creation and must be enabled. It handles that RTP
Play button is not shown e.g. in TCP sequence show
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
Added shortcuts:
- Mute/Unmute/Invert Mute
- Play/Pause/Stop
Removed outdated help in tooltip text. It makes no sense to use it, all
actions are available in context menu.
Changes:
- It is possible to select multiple streams in list and in graph
- Select All/None/Invert implemented in list of streams and in graph
- Indication of "Selected" stream redesigned in graph
- Mouse hovering shows related row/wave
- All operations adapted to multiselection
Remove the editor modeline blocks from most of the source files in ui/qt
by running
perl -i -p0e 's{ \n+ /[ *\n]+ editor \s+ modelines .* shiftwidth= .* \*/ \s+ } {\n}gsix' $( ag -g '\.(cpp|h)' )
then cleaning up the remaining files by hand.
This *shouldn't* affect anyone since
- All of the source files in ui/qt use 4 space indentation, which
matches 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.
Functional changes:
Audio routing information is now stored in audio stream and not in table. It
is handled by separate utils/rtp_audio_routing.cpp class which is able to
convert it between mono/stereo etc.
There is new utils/rtp_audio_routing_filter.cpp class which is able to
route mono audio stream to any audio channel.
Sample file separated from audio stream file - sample file is generated only
during recap. So when we need new waveform, we just use existing sample file
and no recap required - it is much faster.
Audio stream exports just mono audio. Mono audio is then expanded to stereo
and correct channel by AudioRoutingFilter during play as required. So when
audio routing is changed, no recap and no audio export is required.
When audio stream is muted, no audio is produced nor played.
Most of signals between RtpPlayerDialog and RtpAudioStream were removed.
Start/Pause/Stop is processed in RtpPlayDialog (just for non muted streams).
Play possition is not received from every playing stream but from independent
silence stream.
Added Mute/Unmute function.
Optimalization:
When audio routing is changed, just graphs are updated. No retap nor audio
decoding is required.
When TOD is changed, just graphs are updated. No retap nor audio decoding is
required.
When many streams are analyzed, some streams can be found as useless during
replay. Patch adds option to remove it from list in RTP player.
Once stream is removed, it can't be returned except close/reopen RTP player
with new selection.
When there is many waveforms, it is not clear which one is selected in list.
It is important for selecting packets from graph.
Patch emphasis selected waveform. In addition, it allows to select waveform
by clicking on it.
Reason for #16452 is that RTP Player dialog is destroyed before tap
finishes when Escape key pressed. The only solution I found is to check
whether dialog does exists when tap finishes. If so, data are processed
and shown. If dialog was destroyed in meanwhile, no data processing is done.
Column 'Play' added to player. Double click on a stream in the column changes
audio routing for the stream.
When soundcard supports only one channel, there are Mute/Play option. When
soundcard supports two or more channels, there are Mute/L/L+R/R options.
Muted channel is drawn with dotted line.
Change-Id: If120c902195da46f98a1663c589f20c6a1da0ba7
Reviewed-on: https://code.wireshark.org/review/35687
Petri-Dish: Alexis La Goutte <alexis.lagoutte@gmail.com>
Tested-by: Petri Dish Buildbot
Reviewed-by: Anders Broman <a.broman58@gmail.com>
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>
Favor asynchronous show() as it does not create new event loop.
Change-Id: I01982806f87705f04138f15ae8eb084f1d4f9b2c
Reviewed-on: https://code.wireshark.org/review/34677
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot
Reviewed-by: Anders Broman <a.broman58@gmail.com>
Upgrade the internal version of QCustomPlot to 2.0.1
Change-Id: I1eb372d8e6a2f6c1bbdde4c74596785bf2d405c2
Reviewed-on: https://code.wireshark.org/review/17980
Reviewed-by: Roland Knall <rknall@gmail.com>
Petri-Dish: Roland Knall <rknall@gmail.com>
Tested-by: Petri Dish Buildbot
Reviewed-by: Anders Broman <a.broman58@gmail.com>
This allows taps that can fail to report an error and fail; a failed
tap's packet routine won't be called again, so they don't have to keep
track of whether they've failed themselves.
We make the return value from the packet routine an enum.
Don't have a separate type for the per-packet routine for "follow" taps;
they're expected to act like tap packet routines, so just use the type
for tap packet routines.
One tap packet routine returned -1; that's not a valid return value, and
wasn't one before this change (the return value was a boolean), so
presume the intent was "don't redraw".
Another tap routine's early return, without doing any work, returned
TRUE; this is presumably an error (no work done, no need to redraw), so
presumably it should be "don't redraw".
Clean up some white space while we're at it.
Change-Id: Ia7d2b717b2cace4b13c2b886e699aa4d79cc82c8
Reviewed-on: https://code.wireshark.org/review/31283
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_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>
Move */ to a separate line below the SPDX identifier.
Change-Id: Id1032215449cfccae0933147b45e04b65e0b727f
Reviewed-on: https://code.wireshark.org/review/27211
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>
In the RTP player dialog, list the default audio device first, ensure
it's selected by default and ensure that the list items are unique.
According to
http://code.qt.io/cgit/qt/qtmultimedia.git/tree/src/plugins/windowsaudio/qwindowsaudiodeviceinfo.cpp?h=5.9
the default device on Windows uses the special WAVE_MAPPER id, which
appears to support various sample rates even when the underlying
hardware doesn't.
Ensuring the names are unique fixes an issue I'm seeing on a test
machine here.
When decoding, check to see if our sample rate is supported by our
output device and adjust accordingly.
Bug: 13906
Change-Id: Iddc0beb2459bfac42276ff29d227c2619b0a8d90
Reviewed-on: https://code.wireshark.org/review/22756
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Jirka Novak