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>
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: 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>
Gerald Combs