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>
The data that describes RTP streams become invalid when packets are
re-dissected. This results in a crash in GTK when the "RTP Analyse"
option is used and and a crash in Qt when the display filter is changed
while the RTP Streams dialog is open.
Fix this by adding a tap_reset callback (modelled after mcaststream) to
the RTP tap listener that allows the GTK+ and Qt dialogs to clear the
displayed list of RTP streams.
Bug: 10016
Change-Id: I7478678db63d7ac8110c44c163844e9f66fad9e9
Reviewed-on: https://code.wireshark.org/review/10728
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: Michael Mann <mmann78@netscape.net>
Rename ui/gtk/iax2_analysis.h to ui/tap-iax2-analysis.h. Move
iax2_packet_analyse to ui/tap-iax2-analysis.c.
Rename rtp_analysis.h to tap-rtp-analysis.h to match IAX2.
Change-Id: Ice7e9ad0d7bf62d631850089c880ec09a3e101dd
Reviewed-on: https://code.wireshark.org/review/10375
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Add the UDP multicast stream dialog. Abuse TapParameterDialog a bit more
so that we can edit parameters.
Remove some unused struct members and an unused function.
Change-Id: I962c70344e792f0959527e4bcba8a20bd7e8acf9
Reviewed-on: https://code.wireshark.org/review/10084
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Merge rtp_sample_header_t into rtp_sample_t. That's the only place it
was used. Note that rtp_sample_t is used for writing rtpdump files.
Move the rtp_sample_t definition to tap-rtp-common.c. Rename it to
rtpdump_info_t. Make rtp_write_sample static.
Change-Id: I04e7428f634efa87a98e5d6c82a354f94ab1765d
Reviewed-on: https://code.wireshark.org/review/9629
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
A tap listener has the ability to apply a filter (typically the display filter). Add a parameter to RTP GUI API functions to allow them to pass in a filter.
Bug: 996
Change-Id: Ib184dfb023be5d1d24a0d842b4039311426b5293
Reviewed-on: https://code.wireshark.org/review/8468
Petri-Dish: Michael Mann <mmann78@netscape.net>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Anders Broman <a.broman58@gmail.com>
The tap listener was handling rtpstream_tapinfo_t* types while other
users was expecting a GList* instead. Fix this and avoid future
confusion by replacing void* pointers.
Ping-Bug: 10714
Change-Id: I66f62eaaed4a529714264bbf4e7ad1e72b46ce5a
Reviewed-on: https://code.wireshark.org/review/6997
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Add keyboard shortcuts. Note that not all of the buttons made it from
GTK+. Add a "Go to setup frame" option.
Move rtp_streams.c from ui/gtk to ui.
Add a help URL for RTP analysis (which needs to be split into streams +
analysis).
Fix RTP stream packet marking.
Change-Id: Ifb8192ff701a933422509233d76461a46e459f4f
Reviewed-on: https://code.wireshark.org/review/6852
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
Consolidate the three different RTP stream structs in ui/rtp_stream.h,
ui/gtk/rtp_player.c, and ui/voip_calls.c into one. Make the member names
a bit more consistent. Document what each GList contains. Use nstime_t
for timestamps since that's what we get from the frame data. Use g_new0
to initialize our structs.
Change-Id: I2b3f8f2051394a6a98a5c7bc49c117f07161d031
Reviewed-on: https://code.wireshark.org/review/5843
Petri-Dish: Gerald Combs <gerald@wireshark.org>
Tested-by: Petri Dish Buildbot <buildbot-no-reply@wireshark.org>
Reviewed-by: Gerald Combs <gerald@wireshark.org>
... to ensure that there are no potential issues with respect to
editors limiting the number of lines scanned at the end of the file
when checking for editor modelines.
Change-Id: Ic85cbb108bb5159d6ec4116fea11f5eebb4e44a4
Reviewed-on: https://code.wireshark.org/review/4688
Reviewed-by: Bill Meier <wmeier@newsguy.com>
(Using sed : sed -i '/^ \* \$Id\$/,+1 d')
Fix manually some typo (in export_object_dicom.c and crc16-plain.c)
Change-Id: I4c1ae68d1c4afeace8cb195b53c715cf9e1227a8
Reviewed-on: https://code.wireshark.org/review/497
Reviewed-by: Anders Broman <a.broman58@gmail.com>