Qt: Initial RTP playback.
authorGerald Combs <gerald@wireshark.org>
Sat, 13 Dec 2014 00:51:40 +0000 (16:51 -0800)
committerGerald Combs <gerald@wireshark.org>
Fri, 2 Oct 2015 18:26:05 +0000 (18:26 +0000)
commit3687d393040a40655d84e3e03417a474032bad86
tree55f208b60abb59c5812bae2407a9b36dfdd2f09a
parentfd5eafa50a77bc319a240727600be38307e54f86
Qt: Initial RTP playback.

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>
41 files changed:
CMakeLists.txt
Makefile.nmake
acinclude.m4
cmakeconfig.h.in
codecs/CMakeLists.txt
codecs/Makefile.common
codecs/Makefile.nmake
codecs/speex/README.txt [new file with mode: 0644]
codecs/speex/arch.h [new file with mode: 0644]
codecs/speex/resample.c [new file with mode: 0644]
codecs/speex/speex_resampler.h [new file with mode: 0644]
codecs/speex/stack_alloc.h [new file with mode: 0644]
configure.ac
docbook/wsug_src/WSUG_chapter_telephony.asciidoc
epan/dissectors/packet-rtp.h
ui/gtk/rtp_analysis.c
ui/gtk/rtp_player.c
ui/gtk/voip_calls_dlg.c
ui/help_url.c
ui/help_url.h
ui/qt/CMakeLists.txt
ui/qt/Makefile.am
ui/qt/Makefile.common
ui/qt/Wireshark.pro
ui/qt/packet_dialog.cpp
ui/qt/rtp_audio_stream.cpp [new file with mode: 0644]
ui/qt/rtp_audio_stream.h [new file with mode: 0644]
ui/qt/rtp_player_dialog.cpp [new file with mode: 0644]
ui/qt/rtp_player_dialog.h [new file with mode: 0644]
ui/qt/rtp_player_dialog.ui [new file with mode: 0644]
ui/qt/voip_calls_dialog.cpp
ui/qt/voip_calls_dialog.h
ui/qt/voip_calls_dialog.ui
ui/rtp_media.h
ui/rtp_stream.c
ui/rtp_stream.h
ui/tap-sequence-analysis.c
ui/tap-sequence-analysis.h
ui/voip_calls.c
ui/voip_calls.h
vagrant_provision.sh