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webrtc
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src.git
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f181137b05e4b899fa3f15afafc4f27e683d83cc
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pc
/
srtp_filter.cc
a4d8737
Format almost everything.
by Jonas Olsson
· 6 years ago
10542f2
(4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
by Steve Anton
· 6 years ago
1c05765
(3) Rename files to snake_case: move the files
by Steve Anton
· 6 years ago
[Renamed from pc/srtpfilter.cc]
3e70781
[Cleanup] Add missing #include. Remove useless ones. IWYU part 2.
by Yves Gerey
· 6 years ago
26968ba
Delete unused utf8 conversion utilities
by Niels Möller
· 6 years ago
a76af0c
Move base64.h to the proper location.
by Artem Titov
· 7 years ago
66cadcc
Replace rtc::Optional with absl::optional in pc
by Danil Chapovalov
· 7 years ago
665174f
Reformat the WebRTC code base
by Yves Gerey
· 7 years ago
5b32f23
Securely clear memory containing key information / passwords before freeing.
by Joachim Bauch
· 7 years ago
e818b6e
Create the JsepTransportController and JsepTransport2.
by Zhi Huang
· 7 years ago
45cc890
Assorted logging pedantry
by Jonas Olsson
· 7 years ago
36f8f3e
Optional: Use nullopt and implicit construction in /pc
by Oskar Sundbom
· 7 years ago
675513b
Stop using LOG macros in favor of RTC_ prefixed macros.
by Mirko Bonadei
· 7 years ago
b140b9f
Keep count of libsrtp clients, and only deinitialize when it goes to 0.
by Taylor Brandstetter
· 7 years ago
cf990f5
Reland: Completed the functionalities of SrtpTransport.
by Zhi Huang
· 8 years ago
eb23e17
Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ )
by zhihuang
· 8 years ago
92ea95e
Fixing WebRTC after moving from src/webrtc to src/
by Mirko Bonadei
· 8 years ago
bb54720
Moving src/webrtc into src/.
by Mirko Bonadei
· 8 years ago
[Renamed from webrtc/pc/srtpfilter.cc]
e683c68
Completed the functionalities of SrtpTransport.
by zhihuang
· 8 years ago
4dde3df
Move SrtpSession and tests to their own files.
by zstein
· 8 years ago
c20978e
Rename webrtc/base -> webrtc/rtc_base
by Edward Lemur
· 8 years ago
a80c16a
Revert "Update includes for webrtc/{base => rtc_base} rename (2/3)"
by Henrik Kjellander
· 8 years ago
c3771cc
Update includes for webrtc/{base => rtc_base} rename (2/3)
by kjellander
· 8 years ago
5869f50
Support encrypted RTP extensions (RFC 6904)
by jbauch
· 8 years ago
03fa534
Support getting external HMAC auth context with libsrtp 2.1.0.
by jbauch
· 8 years ago
af99b6d
Delete SignalSrtpError.
by nisse
· 8 years ago
eaa9c1d
Remove HAVE_SRTP define and unmaintained code.
by jbauch
· 8 years ago
dfcab72
Reland: Improve testing of SRTP external auth code paths.
by jbauch
· 8 years ago
d81f121
Revert of Improve testing of SRTP external auth code paths. (patchset #2 id:20001 of https://codereview.webrtc.org/2722423003/ )
by jbauch
· 8 years ago
ac170d5
Improve testing of SRTP external auth code paths.
by jbauch
· 8 years ago
d48f488
Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.
by jbauch
· 8 years ago
7d25426
Delete unneeded includes of base/common.h.
by nisse
· 8 years ago
79e0588
Set actual transport overhead in rtp_rtcp
by michaelt
· 8 years ago
0d8ade5
Remove remnants of libsrtp1
by mattdr
· 8 years ago
8ff52cc
Remove useless debugging code
by mattdr
· 8 years ago
51f2919
Update WebRTC to build against libsrtp 2.0
by mattdr
· 9 years ago
cb56065
Add support for GCM cipher suites from RFC 7714.
by jbauch
· 9 years ago
37bb54e
Reland: Remove global list of SRTP sessions.
by Joachim Bauch
· 9 years ago
82d7862
Change default timestamp to 64 bits in all webrtc directories.
by Honghai Zhang
· 9 years ago
7bc7c06
Revert of Remove the rtc_relative_path GYP variable and similar defines (patchset #1 id:1 of https://codereview.webrtc.org/1903553003/ )
by kjellander
· 9 years ago
e19cf59
Remove the rtc_relative_path GYP variable and similar defines
by kjellander
· 9 years ago
b252856
Remove all uses of the HAVE_CONFIG_H define.
by Henrik Kjellander
· 9 years ago
65c7f67
Fix license headers in webrtc/pc
by kjellander
· 9 years ago
9b8df25
Move talk/session/media -> webrtc/pc
by kjellander@webrtc.org
· 9 years ago
[Renamed (99%) from talk/session/media/srtpfilter.cc]
a96e2d7
Move talk/media to webrtc/media
by kjellander
· 9 years ago
77fa59d
Fix build break in google3 import caused by https://codereview.webrtc.org/1532543003
by guoweis
· 9 years ago
4638331
DTLS-SRTP set up is bypassed when the channel has been writable.
by guoweis
· 9 years ago
46ad542
Revert of "Create rtc::AtomicInt POD struct." (patchset #3 id:40001 of https://codereview.webrtc.org/1498953002/ )
by pbos
· 9 years ago
84f0970
Reland of "Create rtc::AtomicInt POD struct."
by Peter Boström
· 9 years ago
521ed7b
Reland Convert internal representation of Srtp cryptos from string to int
by Guo-wei Shieh
· 9 years ago
318166b
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
by guoweis
· 9 years ago
2764e10
Convert internal representation of Srtp cryptos from string to int.
by guoweis
· 9 years ago
cbe9f51
Revert of Remove global list of SRTP sessions. (patchset #4 id:60001 of https://codereview.webrtc.org/1416093010/ )
by phoglund
· 9 years ago
9cafd97
Remove global list of SRTP sessions.
by jbauch
· 9 years ago
ff134eb
talk: Use NDEBUG macro.
by tfarina
· 9 years ago
0c4e06b
Use suffixed {uint,int}{8,16,32,64}_t types.
by Peter Boström
· 9 years ago
456696a
Reland Change WebRTC SslCipher to be exposed as number only
by Guo-wei Shieh
· 10 years ago
27dc29b
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
by guoweis
· 10 years ago
4fe3c9a
Change WebRTC SslCipher to be exposed as number only.
by guoweis
· 10 years ago
e70028e
Protect access to shared list of SRTP sessions.
by Joachim Bauch
· 10 years ago
fec2c6d
Prevent potential double-free if srtp_create fails.
by Joachim Bauch
· 10 years ago
469c2c0
Make Config::default_value leak instead of having an exit-time destructor.
by Andrew MacDonald
· 10 years ago
9478437
rtc::Buffer improvements
by Karl Wiberg
· 10 years ago
4b3c0d6
Use WebRTC API to convert byteorder in srtpfilter.
by Jiayang Liu
· 10 years ago
a197a5e
Update libsrtp includes in preparation of roll into Chromium.
by jiayl@webrtc.org
· 10 years ago
a9cf079
Rename external_hmac_ctx_t to ExternalHmacContext.
by pbos@webrtc.org
· 10 years ago
2e7ee4b
Fix the SrtpFilter crash caused by two local offers.
by pthatcher@webrtc.org
· 10 years ago
a09a999
(Auto)update libjingle 73222930-> 73226398
by buildbot@webrtc.org
· 11 years ago
d4e598d
(Auto)update libjingle 72097588-> 72159069
by buildbot@webrtc.org
· 11 years ago
0537634
(Auto)update libjingle 62713454-> 62865357
by henrike@webrtc.org
· 11 years ago
371243d
Remove std:: prefixes from C functions in talk/.
by pbos@webrtc.org
· 11 years ago
d43aa9d
Update libjingle 61901702->61966318
by henrike@webrtc.org
· 11 years ago
a7b9818
Unrevert 5590 "description"(=(Auto)update libjingle 61834300->61901702).
by henrike@webrtc.org
· 11 years ago
ef22151
Revert 5590 "description"
by xians@webrtc.org
· 11 years ago
2643805
description
by henrike@webrtc.org
· 11 years ago
9dba525
* Update libjingle to 50389769.
by wu@webrtc.org
· 12 years ago
28e2075
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk
by henrike@webrtc.org
· 12 years ago