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:
In nearly all cases decoding match content of capture. The exception is #2270,
where timestamps do not match recorded time which causes discrepancy in
decoding.
Decoding of audio correctly follows different soundcard rates.
RTP Player shows first sample rate in each stream in place of rate of playing.
Fixed incorrect time axis calculation
Fixes#16837Fixes#4960Fixes#2270
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
Visual waveform is derived from decoded audio. When audio is decoded
incorrectly, waveform now shows it.
E.g. on issue 14401 is now audio play aligned with waveform, but it
exhibits that decoded audio is incorrect - about two times longer than
pcap!
Changes:
- samplefile_ renamed to sample_file_
- tempfile_ is renamed to temp_file_
- decode() is separated to decodeAudio and decodeVisual
- Frame info stores frame len and frame_num for every frame. We must hold
it per frame as it may change in time. Info is stored in separate temp file
as waveform samples.
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.
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>
*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>
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>
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>
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>
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>
Jirka Novak