| 3542013f587f0858fb24fa8e554ec3c01a323da8 |
|
14-Jan-2016 |
sprang <sprang@webrtc.org> |
Revert of Update with new default boringssl no-aes cipher suites. Re-enable tests. (patchset #3 id:40001 of https://codereview.webrtc.org/1550773002/ )
Reason for revert:
We're getting boringssl version conflicts. Reverting for now.
Original issue's description:
> Update with new default boringssl no-aes cipher suites. Re-enable tests.
>
> This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
>
> BUG=webrtc:5381
> R=davidben@webrtc.org, henrika@webrtc.org
>
> Committed: https://crrev.com/31c8d2eac5aec977f584ab0ae5a1d457d674f101
> Cr-Commit-Position: refs/heads/master@{#11250}
TBR=davidben@webrtc.org,henrika@webrtc.org,torbjorng@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5381
Review URL: https://codereview.webrtc.org/1586183002
Cr-Commit-Position: refs/heads/master@{#11253}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 31c8d2eac5aec977f584ab0ae5a1d457d674f101 |
|
14-Jan-2016 |
Torbjorn Granlund <torbjorng@google.com> |
Update with new default boringssl no-aes cipher suites. Re-enable tests.
This undoes https://codereview.webrtc.org/1533253002 (except the DEPS part).
BUG=webrtc:5381
R=davidben@webrtc.org, henrika@webrtc.org
Review URL: https://codereview.webrtc.org/1550773002 .
Cr-Commit-Position: refs/heads/master@{#11250}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| e2976c87f7ba627fa1e1246f0ccfb34b4b9f3a73 |
|
04-Jan-2016 |
Peter Boström <pbos@webrtc.org> |
Remove DISABLED_ON_ macros.
Macro incorrectly displays DISABLED_ON_ANDROID in test names for
parameterized tests under --gtest_list_tests, causing tests to be
disabled on all platforms since they contain the DISABLED_ prefix rather
than their expanded variants.
This expands the macro variants to inline if they're disabled or not,
and removes building some tests under configurations where they should
fail, instead of building them but disabling them by default.
The change also removes gtest_disable.h as an unused include from many
other files.
BUG=webrtc:5387, webrtc:5400
R=kjellander@webrtc.org, phoglund@webrtc.org
TBR=henrik.lundin@webrtc.org
Review URL: https://codereview.webrtc.org/1547343002 .
Cr-Commit-Position: refs/heads/master@{#11150}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 2f042f26a3d0c062c43dc553058a286bd4dd8f19 |
|
20-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Roll chromium_revision 1b6c421..db567a8 (365999:366304)
I had to disable some Dtls12Both tests failing under MSan (see bug).
Notice those errors started happening in the range of
https://boringssl.googlesource.com/boringssl.git/+log/afd565f..9f897b2
while this CL brings in an even newer BoringSSL (that still has the same problem).
Change log: https://chromium.googlesource.com/chromium/src/+log/1b6c421..db567a8
Full diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8
Changed dependencies:
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/afd565f..afe57cb
* src/third_party/libyuv: https://chromium.googlesource.com/libyuv/libyuv.git/+log/1019e45..1ccbf8f
* src/third_party/nss: https://chromium.googlesource.com/chromium/deps/nss.git/+log/a676aa0..aee1b12
DEPS diff: https://chromium.googlesource.com/chromium/src/+/1b6c421..db567a8/DEPS
No update to Clang.
NOTRY=True
BUG=webrtc:5381
TBR=torbjorng@webrtc.org, pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1533253002
Cr-Commit-Position: refs/heads/master@{#11095}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 46c9cc01901ecd3af0191872f9660b710d5fe757 |
|
01-Dec-2015 |
Torbjorn Granlund <torbjorng@google.com> |
Provide method for returning certificate expiration time stamp.
We convert ASN1 time via std::tm to int64_t representing milliseconds-since-epoch. We do not use time_t since that cannot store milliseconds, and expires for 32-bit platforms in 2038 also for seconds.
Conversion via std::tm might might seem silly, but actually doesn't add any complexity.
One would expect tm -> seconds-since-epoch to already exist on the standard library. There is mktime, but it uses localtime (and sets an environment variable, and has the 2038 problem).
The ASN1 TIME parsing is limited to what is required by RFC 5280.
BUG=webrtc:5150
R=hbos@webrtc.org, nisse@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1468273004 .
Cr-Commit-Position: refs/heads/master@{#10854}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 7593aad1634c2ab88351a56eca108cb6af9a274e |
|
19-Nov-2015 |
torbjorng <torbjorng@webrtc.org> |
Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes.
BUG=
Review URL: https://codereview.webrtc.org/1459153002
Cr-Commit-Position: refs/heads/master@{#10719}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| e488a0dbe4114ce51feeaf663ad4e2a6bd4b9a2b |
|
19-Nov-2015 |
jbauch <jbauch@webrtc.org> |
Fix DTLS packet boundary handling in SSLStreamAdapterTests.
The tests were not honoring packet boundaries, thus causing failures
in tests with dropped/broken packets. This CL fixes this and also
re-enables the tests.
R=torbjorng@webrtc.org,pthatcher@webrtc.org,tommi@webrtc.org,juberti@webrtc.org
BUG=webrtc:5005,webrtc:5188
Review URL: https://codereview.webrtc.org/1440193002
Cr-Commit-Position: refs/heads/master@{#10709}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 521ed7bf022c4e30574d7970c2be5be46567f4cd |
|
19-Nov-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Convert internal representation of Srtp cryptos from string to int
TBR=pthatcher@webrtc.org
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1458023002 .
Cr-Commit-Position: refs/heads/master@{#10703}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 318166bed75dcbc00a7b79f715f9953aff9ffbc7 |
|
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
Reason for revert:
Broke chromium fyi build.
Original issue's description:
> Convert internal representation of Srtp cryptos from string to int.
>
> Note that the coversion from int to string happens in 3 places
> 1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
> 2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
> 3) stats collection also needs external names.
>
> External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
> Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
>
> The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
>
> BUG=webrtc:5043
>
> Committed: https://crrev.com/2764e1027a08a5543e04b854a27a520801faf6eb
> Cr-Commit-Position: refs/heads/master@{#10701}
TBR=juberti@webrtc.org,pthatcher@webrtc.org,juberti@google.com
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1455233005
Cr-Commit-Position: refs/heads/master@{#10702}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 2764e1027a08a5543e04b854a27a520801faf6eb |
|
19-Nov-2015 |
guoweis <guoweis@webrtc.org> |
Convert internal representation of Srtp cryptos from string to int.
Note that the coversion from int to string happens in 3 places
1) SDP layer from int to external names. mediasession.cc GetSupportedSuiteNames.
2) for SSL_CTX_set_tlsext_use_srtp(), converting from int to internal names.
3) stats collection also needs external names.
External names are like AES_CM_128_HMAC_SHA1_80, specified in sslstreamadapter.cc.
Internal names are like SRTP_AES128_CM_SHA1_80, specified in opensslstreamadapter.cc.
The conversion from string to int happens in one place only, SDP layer, SrtpFilter::ApplyParams().
BUG=webrtc:5043
Review URL: https://codereview.webrtc.org/1416673006
Cr-Commit-Position: refs/heads/master@{#10701}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 4e572470a3f181a043f9f9b98cc0153b7195b9f5 |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Provide RSA2048 as per RFC
Original CL here:
https://codereview.webrtc.org/1329493005
That CL is in patch set #1 of this CL.
This CL resolves a method collision in Chrome.
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1394223002
Cr-Commit-Position: refs/heads/master@{#10222}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 335204c550e9570d356d0d6264475ac40c7f92f6 |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
Revert of Provide RSA2048 as per RFC (patchset #9 id:200001 of https://codereview.webrtc.org/1329493005/ )
Reason for revert:
Breaks chrome.
Original issue's description:
> provide RSA2048 as per RFC
>
> BUG=webrtc:4972
>
> Committed: https://crrev.com/0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e
> Cr-Commit-Position: refs/heads/master@{#10209}
TBR=hbos@webrtc.org,juberti@google.com,jbauch@webrtc.org,henrikg@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1397703002
Cr-Commit-Position: refs/heads/master@{#10210}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 0df3eb03c9a6a8299d7e18c8c314ca58c2f0681e |
|
08-Oct-2015 |
torbjorng <torbjorng@webrtc.org> |
provide RSA2048 as per RFC
BUG=webrtc:4972
Review URL: https://codereview.webrtc.org/1329493005
Cr-Commit-Position: refs/heads/master@{#10209}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.
BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org
Review URL: https://codereview.webrtc.org/1362503003 .
Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 6caafbe5b6b777b309a6eb90a02cf54d5106fb9b |
|
05-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Convert uint16_t to int for WebRTC cipher/crypto suite.
This is a follow up CL on https://codereview.webrtc.org/1337673002
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1377733004 .
Cr-Commit-Position: refs/heads/master@{#10175}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 456696a9c1bbd586701dcca3e4b2695e419a10ba |
|
01-Oct-2015 |
Guo-wei Shieh <guoweis@webrtc.org> |
Reland Change WebRTC SslCipher to be exposed as number only
This is to revert the change of https://codereview.webrtc.org/1380603005/
TBR=pthatcher@webrtc.org
BUG=523033
Review URL: https://codereview.webrtc.org/1375543003 .
Cr-Commit-Position: refs/heads/master@{#10126}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 27dc29b0df23eed5034f28d4d5f66ea0bb425d6c |
|
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Revert of Change WebRTC SslCipher to be exposed as number only. (patchset #20 id:750001 of https://codereview.webrtc.org/1337673002/ )
Reason for revert:
This broke chromium.fyi bot.
Original issue's description:
> Change WebRTC SslCipher to be exposed as number only.
>
> This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
>
> For SRTP, currently it's still string internally but is reported as IANA number.
>
> This is used by the ongoing CL https://codereview.chromium.org/1335023002.
>
> BUG=523033
>
> Committed: https://crrev.com/4fe3c9a77386598db9abd1f0d6983aefee9cc943
> Cr-Commit-Position: refs/heads/master@{#10124}
TBR=juberti@webrtc.org,rsleevi@chromium.org,pthatcher@webrtc.org,davidben@chromium.org,juberti@google.com,davidben@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=523033
Review URL: https://codereview.webrtc.org/1380603005
Cr-Commit-Position: refs/heads/master@{#10125}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 4fe3c9a77386598db9abd1f0d6983aefee9cc943 |
|
01-Oct-2015 |
guoweis <guoweis@webrtc.org> |
Change WebRTC SslCipher to be exposed as number only.
This makes the SSL exposed as uint16_t which is the IANA value. GetRfcSslCipherName is introduced to handle the conversion to names from ID. IANA value will be used for UMA reporting. Names will still be used for WebRTC stats reporting.
For SRTP, currently it's still string internally but is reported as IANA number.
This is used by the ongoing CL https://codereview.chromium.org/1335023002.
BUG=523033
Review URL: https://codereview.webrtc.org/1337673002
Cr-Commit-Position: refs/heads/master@{#10124}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 07d09364b003e6738a02d9940aebab5d3814da6d |
|
22-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
Purge nss files and dependencies.
This replaces https://codereview.webrtc.org/1313233005
which was reverted after triggering Chromium issues.
The only difference is that we're cleaned up dependencies
on use_openssl from the gyp file.
Since https://codereview.chromium.org/1358913003 landed,
this CL should cause no Chromium issues.
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1351503004
Cr-Commit-Position: refs/heads/master@{#10019}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| eecbab7cd54814b189fe1b5fdf2ba2afa4df4fbf |
|
16-Sep-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
Roll chromium_revision a28d8d5..5482f56 (346100:347609)
Recent changes (https://codereview.chromium.org/1311013010) introduces a
dependency on WebKit (Blink) in Chromium, which forces us to start pulling
down that as well (+6GB). However Blink is about to be merged into the
Chromium repo soon anyway, so the size increase is inevitable.
Luckily, this can be removed in the next roll, if we roll past
http://crrev.com/348812
The ijar dependency was introduced in https://codereview.chromium.org/1323053003 (#347208)
Relevant changes:
* src/third_party/boringssl/src: 12fe1b2..ac8302a
* src/third_party/libvpx: a208eca..0304cef
* src/third_party/libyuv: 3c4f573..0bc626a
* src/tools/gyp: 6ee91ad..5d01a8c
Details: https://chromium.googlesource.com/chromium/src/+/a28d8d5..5482f56/DEPS
Clang version was not updated in this roll.
R=torbjorng@webrtc.org
TBR=marpan@webrtc.org
BUG=webrtc:5005, chromium:530112
Review URL: https://codereview.webrtc.org/1305043008 .
Cr-Commit-Position: refs/heads/master@{#9956}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 9eb1365939683cc5462a5359344148efb7d84f97 |
|
05-Sep-2015 |
deadbeef <deadbeef@webrtc.org> |
Revert of purge nss files and dependencies (patchset #1 id:1 of https://codereview.webrtc.org/1313233005/ )
Reason for revert:
It looks like this broke the FYI bots. I tried updating libjingle_nacl.gyp, but the IOS build still failed because in Chrome it's configured to use NSS. See https://codereview.chromium.org/1316863012/.
Original issue's description:
> purge nss files and dependencies
>
> BUG=webrtc:4497
>
> Committed: https://crrev.com/5647a2cf3db888195c928a1259d98f72f6ecbc15
> Cr-Commit-Position: refs/heads/master@{#9862}
TBR=tommi@webrtc.org,kjellander@webrtc.org,torbjorng@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1311843006
Cr-Commit-Position: refs/heads/master@{#9867}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 5647a2cf3db888195c928a1259d98f72f6ecbc15 |
|
04-Sep-2015 |
torbjorng <torbjorng@webrtc.org> |
purge nss files and dependencies
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1313233005
Cr-Commit-Position: refs/heads/master@{#9862}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| b6d4ec418504fd947c6f96829c73180e9487e203 |
|
17-Aug-2015 |
Torbjorn Granlund <torbjorng@google.com> |
Support generation of EC keys using P256 curve and support ECDSA certs.
This CL started life here: https://webrtc-codereview.appspot.com/51189004
BUG=webrtc:4685, webrtc:4686
R=hbos@webrtc.org, juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1189583002 .
Cr-Commit-Position: refs/heads/master@{#9718}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 831c5585c7d2b4c4442e3c1255332f1c23b6a983 |
|
20-May-2015 |
Joachim Bauch <jbauch@webrtc.org> |
Allow setting maximum protocol version for SSL stream adapters.
This CL adds an API to SSL stream adapters to set the maximum allowed
protocol version and with that implements support for DTLS 1.2.
With DTLS 1.2 the default cipher changes in the unittests as follows.
BoringSSL
TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA -> TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
NSS
TLS_RSA_WITH_AES_128_CBC_SHA -> TLS_RSA_WITH_AES_128_GCM_SHA256
BUG=chromium:428343
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/50989004
Cr-Commit-Position: refs/heads/master@{#9232}
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 3ee4fe5a940128cbfe76c8609a56c69c2aeb0175 |
|
11-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Re-land: Add API to get negotiated SSL ciphers
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
The previously approved CL https://webrtc-codereview.appspot.com/26009004/ was reverted in https://webrtc-codereview.appspot.com/40689004/ due to compilation issues while rolling into Chromium.
As the new method has landed in Chromium in https://crrev.com/bc321c76ace6e1d5a03440e554ccb207159802ec, this should be safe to land here now.
BUG=3976
R=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/37209004
Cr-Commit-Position: refs/heads/master@{#8343}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8343 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 2bf0e90c9d152c2b4377f710d03b1eded427c9ef |
|
07-Feb-2015 |
tommi@webrtc.org <tommi@webrtc.org> |
Revert 8275 "This CL adds an API to the SSL stream adapters and ..."
I'm reverting the patch due to compilation issues. It would be great if we could make sure Chromium is ready for the patch before we land it in WebRTC.
As is, we're trying to roll webrtc into Chromium and we can't (this is not the only reason though). I might reland this after the roll, depending on how that goes though.
Here's an example failure:
e:\b\build\slave\win_gn\build\src\jingle\glue\channel_socket_adapter_unittest.cc(77) : error C2259: 'jingle_glue::MockTransportChannel' : cannot instantiate abstract class
due to following members:
'bool cricket::TransportChannel::GetSslCipher(std::string *)' : is abstract
e:\b\build\slave\win_gn\build\src\third_party\webrtc\p2p\base\transportchannel.h(107) : see declaration of 'cricket::TransportChannel::GetSslCipher'
ninja: build stopped: subcommand failed.
> This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
>
> BUG=3976
> R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/26009004
TBR=pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40689004
Cr-Commit-Position: refs/heads/master@{#8282}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8282 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 1d11c8202bd19b5dc07902107bae1d3d71575e67 |
|
06-Feb-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
This CL adds an API to the SSL stream adapters and transport channels to get the SSL cipher that was negotiated with the remote peer.
BUG=3976
R=davidben@chromium.org, juberti@webrtc.org, pthatcher@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26009004
Cr-Commit-Position: refs/heads/master@{#8275}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8275 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|
| 127ca3f8e5df9cd9c8a77dbf243ca5d99fbe7d96 |
|
09-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Disable TestDTLSConnectWithSmallMtu on all platforms.
Other bots elsewhere are breaking on this test, my money is on that this
might be due to different SSL versions being used on the different bots.
This test fails on at least a couple of bots that has use_openssl=1.
R=kjellander@webrtc.org
TBR=henrike@webrtc.org
BUG=3910
Review URL: https://webrtc-codereview.appspot.com/25839004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7403 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| 34f2a9ea7245bac103fececfa53e92359680467a |
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28-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Initialize SSL in unittest_main.cc.
Instead of having each test individually initialize and tear down SSL
move this to unittest_main.cc so that all tests are properly
initialized and new tests "don't have to think about it".
R=pthatcher@webrtc.org
BUG=
Review URL: https://webrtc-codereview.appspot.com/30549004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7316 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| f1d751c7dede158bc9770e4d7c4cb07191ffdf3f |
|
25-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
BUG=crbug/414211
R=juberti@webrtc.org
Committed: https://code.google.com/p/webrtc/source/detail?r=7293
Review URL: https://webrtc-codereview.appspot.com/22739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7301 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| 37e1846d732815dfaa1ef1e5f84e4bcc52eb66e4 |
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25-Sep-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Revert "Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup." (rev 7293).
Breaks windows bot as it was already showing on the try jobs on the
BUG=crbug/414211
R=jiayl@webrtc.org,juberti@webrtc.org
TBR=jiayl@webrtc.org,juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/26599004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7294 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| fe1eafb71a8c3be8fa0e2a100955bffebfba6947 |
|
24-Sep-2014 |
jiayl@webrtc.org <jiayl@webrtc.org> |
Call SSL_shutdown in OpenSSLStreamAdapter::Cleanup.
BUG=crbug/414211
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/22739004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7293 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| fded02c164ea4cc3d28d7f30ac9ce9d94d76ef7a |
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19-Sep-2014 |
henrike@webrtc.org <henrike@webrtc.org> |
base: disabled several base tests on Mac so that rtc_unittests can be turned back on
BUG=N/A
R=kjellander@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/30449004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7240 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| f048872e915a3ee229044ec4bc541f6cbf9e4de1 |
|
13-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.
BUG=N/A
R=niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17479005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| e9a604accd54ab14dbf98f99ccdcf3ae1c54d27c |
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13-May-2014 |
perkj@webrtc.org <perkj@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.
http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457
> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
>
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
>
> Review URL: https://webrtc-codereview.appspot.com/12199004
TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/14479004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
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| 2c7d1b39b9374d2bc9bda4755fd4813db66a135c |
|
12-May-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/base/sslstreamadapter_unittest.cc
|