458ffd574b
Use SSL_CTX_set_max_proto_version instead of SSL_CTX_set_max_version.
...
These functions are identical. BoringSSL added these APIs, then OpenSSL
1.1.0 added similar ones but with slightly longer names. We're
standardizing on the OpenSSL names to avoid API skew.
This is a cherry-pick of https://codereview.webrtc.org/2550423004 .
Test: mma
Change-Id: Iffe4416bd300d71bbaa94c7c127ae89d6286cd7b
2017-04-21 19:47:12 -04:00
3542013f58
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}
2016-01-14 17:14:06 +00:00
31c8d2eac5
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}
2016-01-14 14:18:02 +00:00
a7446d2a50
Change DTLS default from 1.0 to 1.2 for webrtc.
...
This changes for standalone webrtc applications.
BUG=
R=pthatcher@webrtc.org
Review URL: https://codereview.webrtc.org/1548733002 .
Cr-Commit-Position: refs/heads/master@{#11211}
2016-01-11 23:27:12 +00:00
efb047d2dd
Compilation failed with openssl.
...
Missing a cast.
BUG=webrtc:5365
Review URL: https://codereview.webrtc.org/1529043003
Cr-Commit-Position: refs/heads/master@{#11074}
2015-12-17 21:45:03 +00:00
521ed7bf02
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}
2015-11-19 03:42:00 +00:00
318166bed7
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}
2015-11-19 03:03:46 +00:00
2764e1027a
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}
2015-11-19 02:02:40 +00:00
c073615d56
Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc.
...
In preparation for implementing the standardized variant of CHACHA20_POLY1305
(it changed slightly in the standardization process),
TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305 and TLS1_CK_ECDHE_ECDSA_CHACHA20_POLY1305
were renamed to have an _OLD suffix with compatibility unsuffixed #defines
temporarily available.
Update references to include the _OLD suffixed ones. Once we've cycled through
the few consumers of the unsuffixed names (just WebRTC and QUIC), the unsuffixed
names can refer to the to-be-implemented standardized variant and eventually
the draft version will be removed.
(This has no effect on upstream OpenSSL compatibility as OpenSSL never defined
these symbols to begin with. Though probably they will once standardization is
done.)
BUG=none
Review URL: https://codereview.webrtc.org/1412803010
Cr-Commit-Position: refs/heads/master@{#10681}
2015-11-17 20:58:17 +00:00
a41ab9326c
Switch usage of _DEBUG macro to NDEBUG.
...
http://stackoverflow.com/a/29253284/5237416
BUG=None
R=tommi@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1429513004
Cr-Commit-Position: refs/heads/master@{#10468}
2015-10-30 23:08:54 +00:00
0c4e06b4c6
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}
2015-10-07 10:23:32 +00:00
6caafbe5b6
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}
2015-10-05 19:43:33 +00:00
42b4faa28a
Fix a build issue when use external OpenSSL.
...
R=juberti@google.com
TBR=juberti@webrtc.org
BUG=webrtc:5049
Review URL: https://codereview.webrtc.org/1378353005 .
Cr-Commit-Position: refs/heads/master@{#10159}
2015-10-05 03:02:52 +00:00
456696a9c1
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}
2015-10-01 04:49:02 +00:00
27dc29b0df
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}
2015-10-01 02:23:15 +00:00
4fe3c9a773
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}
2015-10-01 01:49:17 +00:00
07d09364b0
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}
2015-09-22 18:58:13 +00:00
9eb1365939
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}
2015-09-05 11:39:24 +00:00
5647a2cf3d
purge nss files and dependencies
...
BUG=webrtc:4497
Review URL: https://codereview.webrtc.org/1313233005
Cr-Commit-Position: refs/heads/master@{#9862}
2015-09-04 15:12:00 +00:00
b6d4ec4185
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}
2015-08-17 12:09:10 +00:00
f4baca50bb
Set mtu for DTLS to 1280
...
Set mtu for DTLS to 1280, otherwise it defaults to 256 in recent
versions of openssl.
R=juberti@webrtc.org
Review URL: https://codereview.webrtc.org/1174483002 .
Cr-Commit-Position: refs/heads/master@{#9407}
2015-06-10 07:46:05 +00:00
a398020cba
SSL_set_read_ahead no longer needed with BoringSSL.
...
In https://boringssl-review.googlesource.com/#/c/4820/ BoringSSL
deprecated "SSL_set_read_ahead". That commit has rolled into
Chromium/WebRTC, so calling the function is no longer needed.
BUG=chromium:447431
R=juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/55559004
Cr-Commit-Position: refs/heads/master@{#9355}
2015-06-02 21:07:50 +00:00
831c5585c7
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}
2015-05-20 10:48:24 +00:00
5ca688b3da
Enable read-ahead on OpenSSL DTLS stream adapters.
...
Prevent multiple BIO reads when reading header and body but read from
internal OpenSSL buffer where possible.
BUG=chromium:447431
R=davidben@chromium.org , juberti@google.com
Review URL: https://webrtc-codereview.appspot.com/46319004
Cr-Commit-Position: refs/heads/master@{#9230}
2015-05-20 08:40:03 +00:00
f7bb6e723b
Use new API from BoringSSL to get RFC name of cipher.
...
This CL uses the new API "SSL_CIPHER_get_rfc_name" from BoringSSL to
get the RFC-compliant cipher name instead of having a custom hardcoded
list of cipher names.
BUG=none
R=juberti@webrtc.org , tommi@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/40829004
Cr-Commit-Position: refs/heads/master@{#8541}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8541 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-28 01:41:49 +00:00
3ee4fe5a94
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
2015-02-11 22:35:30 +00:00
2bf0e90c9d
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
2015-02-07 11:13:18 +00:00
1d11c8202b
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
2015-02-06 19:47:39 +00:00
36d5c3cb44
Leave BIO_METHOD non-const.
...
This breaks building against OpenSSL upstream, which is still supported on iOS.
This reverts part of https://webrtc-codereview.appspot.com/34649004 .
BUG=none
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36879004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8132 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-22 23:06:17 +00:00
c10eceab6e
Always tag SRTP_PROTECTION_PROFILE and BIO_METHOD as const.
...
The BIO_METHODs ought to be const so they can go into rodata; BoringSSL makes
BIO_new take a const BIO_METHOD *, so there's no need for it to be non-const.
Also set SRTP_PROTECTION_PROFILE as const so we can constify those within
BoringSSL (https://boringssl-review.googlesource.com/#/c/2720/ )
BUG=none
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/34649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8013 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-01-07 17:59:28 +00:00
d89b69aade
Fix WebRTC Win64 + BoringSSL build.
...
There were many size_t to int conversions. RAND_poll and RAND_seed no longer do
anything in BoringSSL, so fix that one by removing it. Use a checked_cast for
the remaining ones.
BUG=chromium:429039
R=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/28909004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7655 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-11-06 17:23:09 +00:00
f1d751c7de
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
2014-09-25 16:38:46 +00:00
37e1846d73
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
2014-09-25 07:30:14 +00:00
fe1eafb71a
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
2014-09-24 21:13:39 +00:00
c569a49a3d
Unit tests for SSLAdapter
...
R=juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17309004
Patch from Manish Jethani <manish.jethani@gmail.com >.
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7269 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-23 05:56:44 +00:00
11c6bde474
Specify an ECDH group for ECDHE.
...
By default, OpenSSL cannot negotiate ECDHE cipher suites as a server because it
doesn't know what curve to use.
BUG=chromium:406458
TEST=Download Firefox nightly build from 2014-08-12.
https://ftp.mozilla.org/pub/mozilla.org/firefox/nightly/2014-08-12-mozilla-central-debug/
Point Firefox to https://apprtc.appspot.com
Point Chrome on Android to the URL Firefox redirects to (it'll say ?r=NUMBERS at the end)
After tapping through various permissions prompts on either side, the call goes through.
R=agl@chromium.org , henrike@webrtc.org , jiayl@webrtc.org , juberti@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/18269004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7002 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-08-28 16:14:38 +00:00
4e5f65a4c6
Rebase webrtc/base with r6345 version of talk/base:
...
cd webrtc/base
svn diff -r 6249:6300 http://webrtc.googlecode.com/svn/trunk/talk/base >
6300.diff
patch -p0 -i 6300.diff
ls genericslot* | xargs rm
cp ../../talk/base/sigslottester* .
manual edits of sigslottester* to get rid of talk and talk_base.
BUG=3379
TBR=jiayang
Review URL: https://webrtc-codereview.appspot.com/19649004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6347 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-06-05 20:40:11 +00:00
14abcc7322
libvpx's UNUSED macro conflicts with webrtc/base's. Added missing include of assert.h. Globally defined function "Unused" in talk/base and its copy (webrtc/base) is causing a conflict.
...
libvpx macro (UNUSED) can be found here:
http://src.chromium.org/viewvc/chrome/trunk/deps/third_party/libvpx/source/libvpx/vpx/vpx_codec.h
BUG=N/A
R=andrew@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/17489004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@6185 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-16 16:54:44 +00:00
f048872e91
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
2014-05-13 18:00:26 +00:00
e9a604accd
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
2014-05-13 08:15:48 +00:00
2c7d1b39b9
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
2014-05-12 18:03:09 +00:00
