Specifically, I'm moving
safe_compare.h
safe_conversions.h
safe_minmax.h
They shouldn't be part of the API, and moving them to an appropriate
subdirectory of rtc_base/ is a good way to keep track of that.
BUG=webrtc:8445
Change-Id: I458531aeb30bcf4291c4bec3bf22a2fffbf054ff
Reviewed-on: https://webrtc-review.googlesource.com/20860
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20829}
This is pretty useful if you need to debug an SCTP issue. SCTP goes
over DTLS, which is encrypted, which means you need a private key
in order to decrypt a normal packet capture. We don't log this key,
for understandable reasons. So the alternative is to turn on SCTP
verbose logging, then turn the text log into a pcap file.
NOTRY=True
TBR=zhihuang@webrtc.org
Bug: None
Change-Id: If3380d7953ea829b3ae9945326d3c820ce7cc6a3
Reviewed-on: https://webrtc-review.googlesource.com/14561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20380}
This is the same thing we're doing for usrsctp. Before this CL, the
first SrtpSession to call SetKey would initialize libsrtp, and
ChannelManager's destructor would deinitialize it. This works for an
application that only uses one instance of ChannelManager simultaneously
(or one instance of PeerConnectionFactory), but not one that uses
multiple.
Now, libsrtp is effectively reference-counted, with the first
SrtpSession to take a reference initializing it, and the last to remove
its reference deinitializing it.
This issue was discovered recently due to a change that resulted in
using srtp_update. Without using that method, the issue went unnoticed;
maybe srtp_protect/srtp_unprotect don't require initialization?
Bug: webrtc:8388
Change-Id: If1329360f0b469e454810e62e9b5acfbd4cba100
Reviewed-on: https://webrtc-review.googlesource.com/9000
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20262}
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
