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
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.
Change improves Wireshark ability to save rtp streams. It allows a user
to save any supported codec with 8 kHz rate. In real, it means G.711 and
G.729 for now.
There is no hardcoded codec limitation during save anymore. If code detects
unsupported codec or rate during save, it replaces samples with silence and
reports it. Therefore any added codec in future will be supported.
Note to RTP saving:
RTP streams (there can be up to two of them for save) can contain multiple
codecs in each direction - some of it can be supported and some
unsupported. What should be exported then?
Till my patch save do not run and a user received nothing even part of stream
was OK/encoded with supported codec.
Therefore I managed the code to start with export and do its best.
Unknown codec/part is replaced with silence and user is warned after
export. Therefore a user will get:
a) audio - when all codecs are supported (no warning)
b) mix audio/silence - when some codecs are supported (warning)
c) only silence - when no codec is supported (warning)
BTW same output user sees/gets in RTP player for years.
Change-Id: Id938d419f5841af46d2d2d3ddfaf1ec9a0235bcc
Reviewed-on: https://code.wireshark.org/review/35105
Petri-Dish: Roland Knall <rknall@gmail.com>
Tested-by: Petri Dish Buildbot
Reviewed-by: Roland Knall <rknall@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>
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>
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>
Small bugs were introduced when copy/pasting the code from GTK UI:
- arrive_offset is stored in seconds and not milliseconds
- some tests regarding the current playback mode were wrong
Change-Id: I21fb82ba8ff6c8defa7df90c815c040e9e074aaa
Reviewed-on: https://code.wireshark.org/review/13885
Petri-Dish: Pascal Quantin <pascal.quantin@gmail.com>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Pascal Quantin <pascal.quantin@gmail.com>
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>
Moshe Kaplan