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.
Added API description clarifies when bytes/samples are used. New variable names
proposed and all existing codecs are adapted to it. Change is just renaming...
Change-Id: I75dba64a49eb3f4369ec7160cb793dda4b44c810
Reviewed-on: https://code.wireshark.org/review/35576
Reviewed-by: Michael Mann <mmann78@netscape.net>
Petri-Dish: Michael Mann <mmann78@netscape.net>
Tested-by: Petri Dish Buildbot
Reviewed-by: Pascal Quantin <pascal@wireshark.org>
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>
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>
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>
Gerald Combs