History log of /external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
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/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_unittest.cc
893505d0fb41a840be5e4a44a1250dba83d79bf5 08-Jan-2016 Taylor Brandstetter <deadbeef@webrtc.org> Adding unit test to ensure TURN server priorities are unique.

BUG=webrtc:5209
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1570563002 .

Cr-Commit-Position: refs/heads/master@{#11177}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
0c7e9f540b282d60b94081f601a1694054d8646e 29-Dec-2015 Taylor Brandstetter <deadbeef@webrtc.org> Removing webrtc::PortAllocatorFactoryInterface.

ICE servers are now passed directly into PortAllocator,
making PortAllocatorFactoryInterface redundant. This CL also
moves SetNetworkIgnoreMask to PortAllocator.

R=phoglund@webrtc.org, pthatcher@webrtc.org, tkchin@webrtc.org

Review URL: https://codereview.webrtc.org/1520963002 .

Cr-Commit-Position: refs/heads/master@{#11139}
/external/webrtc/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_unittest.cc
bd7d8f7e2b824a887aa12236cb6185d446d7da61 19-Dec-2015 deadbeef <deadbeef@webrtc.org> Adding a MediaStream parameter to createSender.

This will allow an app to create senders with the same stream id,
without SDP munging.

Review URL: https://codereview.webrtc.org/1538673002

Cr-Commit-Position: refs/heads/master@{#11092}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
4638331fd8857b263bb65f12dbf5e1f7005e1a9a 18-Dec-2015 guoweis <guoweis@webrtc.org> DTLS-SRTP set up is bypassed when the channel has been writable.

This regression was introduced by CL 1505573002 to support remote fingerprint update. What happened is that during PrAnswer, we incorrectly do not apply bundle. However, the channel has become writable at that time. When Answer comes, we still reset the srtp_filter but since the channel has been writable, the new SRTP context has never been applied.

We're making sure that we could always apply SRTP context even when channel has been writable. We'll address the issue that bundle should apply even in PrAnswer in a different CL.

BUG=568734

Review URL: https://codereview.webrtc.org/1532543003

Cr-Commit-Position: refs/heads/master@{#11075}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
0eb15ed7b806125774bd13fb214aeb403e2c6857 17-Dec-2015 kwiberg <kwiberg@webrtc.org> Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

We can now use std::move instead!

This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.

Review URL: https://codereview.webrtc.org/1460043002

Cr-Commit-Position: refs/heads/master@{#11064}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
158879305bf5910c0b9e3630a073324a048b59ef 15-Dec-2015 deadbeef <deadbeef@webrtc.org> Fixing flaky LocalP2PTestSctpDataChannel test.

SCTP data channels are closed asynchronously in-band, unlike RTP
data channels, so the test must be slightly modified.

TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1527833003

Cr-Commit-Position: refs/heads/master@{#11017}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
c9be00797edf9a12ff88c81bb56194c74dcacf7f 15-Dec-2015 deadbeef <deadbeef@webrtc.org> Fixing and re-enabling some flaky PeerConnection tests.

BUG=webrtc:3362

Review URL: https://codereview.webrtc.org/1512763003

Cr-Commit-Position: refs/heads/master@{#11016}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
7c73bdbd82956729ee2274318a451a481164f0c6 11-Dec-2015 deadbeef <deadbeef@webrtc.org> Renaming JsepPeerConnectionP2PTestClient back to P2PTestConductor.

Updating blacklists as well.

Review URL: https://codereview.webrtc.org/1508683004

Cr-Commit-Position: refs/heads/master@{#10980}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
cf890bc58eb28d5f1f6ce3f90d4e541983042369 07-Dec-2015 Peter Boström <pbos@webrtc.org> Roll gtest-parallel.

Brings in fixes that save log output to disk instead of piping them
through Python. Should fix problem where output from tests stall for
more than 10 seconds.

Also enabling JsepPeerConnectionP2PTestClient on all platforms again.

BUG=webrtc:5231
R=kjellander@webrtc.org

Review URL: https://codereview.webrtc.org/1509463002 .

Cr-Commit-Position: refs/heads/master@{#10917}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
1218d7ad2fac035376914bd0649fe99e657b33d3 05-Dec-2015 Guo-wei Shieh <guoweis@webrtc.org> Allow remote fingerprint update during a call

Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.

TBR=pthatcher@webrtc.org
BUG=webrtc:3618

This is a reland of https://codereview.webrtc.org/1453523002

Review URL: https://codereview.webrtc.org/1505573002 .

Cr-Commit-Position: refs/heads/master@{#10903}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
86aaa4be8de8f49f91faeefbfd1a23f312898dd2 05-Dec-2015 Guo-wei Shieh <guoweis@webrtc.org> Revert "Allow remote fingerprint update during a call"

This reverts commit 9c38c2d33fa6d794704d53b18f39d5235439fe63.

This commit somehow is different from what I have in my local copy. Revert and will recommit.

TBR=pthatcher@webrtc.org
BUG=3618

Review URL: https://codereview.webrtc.org/1494373004 .

Cr-Commit-Position: refs/heads/master@{#10902}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
9c38c2d33fa6d794704d53b18f39d5235439fe63 05-Dec-2015 Guo-wei Shieh <guoweis@webrtc.org> Allow remote fingerprint update during a call

Changes include the following
1. modify FakeDtlsIdentityStore to support alternate certificate so we could have a different fingerprint in test case.
2. dtlstransportchannel can accept a new fingerprint and trigger DTLS handshake.
3. #2 will trigger new signal on the media side to reset SRTP context. Only reset SRTP context when we are using DTLS (not SDES).
4. Test cases for caller or callee are transfees.

BUG=webrtc:3618
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1453523002 .

Cr-Commit-Position: refs/heads/master@{#10901}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
9cf0c3d4ddfab865dcf924155cc81b763c919a53 04-Dec-2015 Ivo Creusen <ivoc@webrtc.org> Removes MAYBE_ from several test case names in JsepPeerConnectionP2PTestClient.

BUG=webrtc:5231
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://codereview.webrtc.org/1495853002 .

Cr-Commit-Position: refs/heads/master@{#10887}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
fac0655fd7fe0b40ef50dc5b7f11ea44d72cec6c 25-Nov-2015 deadbeef <deadbeef@webrtc.org> Reland of Adding the ability to create an RtpSender without a track.
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )

Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.

Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}

Review URL: https://codereview.webrtc.org/1468113002

Cr-Commit-Position: refs/heads/master@{#10790}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
b5cb19b37c361a263a9cec2e2fb356d16520afd1 24-Nov-2015 deadbeef <deadbeef@webrtc.org> Fixing direction attribute in answer for non-RTP protocols.

"non-RTP protocols" refers to SCTP data channels. Because
there are no streams for SCTP data channels, the answer was being
set to RECVONLY.

BUG=webrtc:5228

Review URL: https://codereview.webrtc.org/1473013002

Cr-Commit-Position: refs/heads/master@{#10762}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
5def7b9fdea0d027bca3df734d86fb877a83bdbf 20-Nov-2015 deadbeef <deadbeef@webrtc.org> Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )

Reason for revert:
Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection.

Original issue's description:
> Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
>
> Reason for revert:
> Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
>
> Original issue's description:
> > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
> >
> > Reason for revert:
> > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
> >
> > Original issue's description:
> > > Adding the ability to create an RtpSender without a track.
> > >
> > > This CL also changes AddStream to immediately create a sender, rather
> > > than waiting until the track is seen in SDP. And the PeerConnection now
> > > builds the list of "send streams" from the list of senders, rather than
> > > the collection of local media streams.
> > >
> > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > > Cr-Commit-Position: refs/heads/master@{#10414}
> >
> > TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> >
> > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> > Cr-Commit-Position: refs/heads/master@{#10417}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae
> Cr-Commit-Position: refs/heads/master@{#10730}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1460323002

Cr-Commit-Position: refs/heads/master@{#10732}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
6834fa10f142bf5e2275142acb834898911d09ae 20-Nov-2015 deadbeef <deadbeef@webrtc.org> Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )

Reason for revert:
Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.

Original issue's description:
> Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
>
> Reason for revert:
> Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
>
> Original issue's description:
> > Adding the ability to create an RtpSender without a track.
> >
> > This CL also changes AddStream to immediately create a sender, rather
> > than waiting until the track is seen in SDP. And the PeerConnection now
> > builds the list of "send streams" from the list of senders, rather than
> > the collection of local media streams.
> >
> > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > Cr-Commit-Position: refs/heads/master@{#10414}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> Cr-Commit-Position: refs/heads/master@{#10417}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1413983004

Cr-Commit-Position: refs/heads/master@{#10730}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
191c1f9d5bc1a4adbcaf87fe93214c54b3530dc8 19-Nov-2015 ivoc <ivoc@webrtc.org> Disable all JsepPeerConnectionP2PTestClient tests on Mac due to flakiness on Mac Debug bots.

NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:5231

Review URL: https://codereview.webrtc.org/1462933002

Cr-Commit-Position: refs/heads/master@{#10716}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
1503867850670447624c8227aea26b038454295b 19-Nov-2015 ivoc <ivoc@webrtc.org> Disabled several JsepPeerConnectionP2PTestClient tests on Mac, due to flakiness on Debug Mac trybots.

NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:5231

Review URL: https://codereview.webrtc.org/1459883002

Cr-Commit-Position: refs/heads/master@{#10710}
/external/webrtc/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_unittest.cc
faac497af560ece34301343eb40377fd5503f7a0 13-Nov-2015 deadbeef <deadbeef@webrtc.org> Fix for scenario where m-line is revived after being set to port 0.

When this is detected, we'll now "reconfigure" the senders and
receivers, which will reconnect the capturers/renderers to the
underlying streams which have been recreated.

BUG=webrtc:2136

Review URL: https://codereview.webrtc.org/1428243005

Cr-Commit-Position: refs/heads/master@{#10628}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
8f46c63f6f764254892f4111b54aa1cc8f32eeeb 26-Oct-2015 deadbeef <deadbeef@webrtc.org> Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )

Reason for revert:
Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.

Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}

TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true

Review URL: https://codereview.webrtc.org/1426443007

Cr-Commit-Position: refs/heads/master@{#10417}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
ac9d92ccbe2b29590c53f702e11dc625820480d5 26-Oct-2015 deadbeef <deadbeef@webrtc.org> Adding the ability to create an RtpSender without a track.

This CL also changes AddStream to immediately create a sender, rather
than waiting until the track is seen in SDP. And the PeerConnection now
builds the list of "send streams" from the list of senders, rather than
the collection of local media streams.

Review URL: https://codereview.webrtc.org/1413713003

Cr-Commit-Position: refs/heads/master@{#10414}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
cbc9507755e730a7f8d81ab3d8cf6efb6678f2ae 16-Oct-2015 deadbeef <deadbeef@webrtc.org> Temporarily rename P2PTestConductor.

Need to do this because some build bots were relying on the previous
name, in order to skip tests that were expected to time out.

TBR=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1412553002

Cr-Commit-Position: refs/heads/master@{#10295}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
af1b59cf271854177915342692a78ec0aba61ccd 15-Oct-2015 deadbeef <deadbeef@webrtc.org> Cleaning up peerconnection_unittest.

Merging the PeerConnectionTestClientBase and JsepTestClient classes,
since there's no real logical distinction. This should make it slightly
less painful to write new PeerConnection tests.

Review URL: https://codereview.webrtc.org/1393223005

Cr-Commit-Position: refs/heads/master@{#10292}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
0a6c4ca942f3a25c15c7af64a9515d381c34cd9c 06-Oct-2015 deadbeef <deadbeef@webrtc.org> Catching more errors when parsing ICE server URLs.

Every malformed URL should now produce an error message in JS, rather than
silently failing and possibly printing a warning message to the console (and
possibly crashing).

Also added some unit tests, and made "ParseIceServers" public.

BUG=445002

Review URL: https://codereview.webrtc.org/1344143002

Cr-Commit-Position: refs/heads/master@{#10186}
/external/webrtc/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_unittest.cc
ee8c6d327357ecd2e17edede8d15f6e3893409a8 13-Aug-2015 deadbeef <deadbeef@webrtc.org> In PeerConnectionTestWrapper, put audio input on a separate thread.

This will prevent it from blocking network input when it falls behind,
which is happening when running with ThreadSanitizer.

BUG=webrtc:4663

Review URL: https://codereview.webrtc.org/1236023010

Cr-Commit-Position: refs/heads/master@{#9707}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
5e56c5927e097f095aef2e9f7be49fd3d59221e1 11-Aug-2015 Henrik Boström <hbos@webrtc.org> DtlsIdentityStoreInterface added and the implementation is called DtlsIdentityStoreImpl (previously named without the -Impl bit and without an interface).

DtlsIdentityStoreImpl is updated to take KeyType into account, something which will be relevant after this CL lands:
https://codereview.webrtc.org/1189583002

The DtlsIdentityService[Interface] classes are about to be removed (to be removed when Chromium no longer implements and uses the interface). This was an unnecessary layer of complexity. The FakeIdentityService is now instead a FakeDtlsIdentityStore.
Where a service was previously passed around, a store is now passed around.

Identity generation is now commonly performed using DtlsIdentityStoreInterface. Previously, if a service was not specified, WebRtcSessionDescriptionFactory could fall back on its own generation code. Now, a store has to be provided for generation to occur.

For more information about the steps being taken to land this without breaking Chromium, see referenced bug.

BUG=webrtc:4899
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1176383004 .

Cr-Commit-Position: refs/heads/master@{#9696}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
ac8869ec5a606e0a0ab71e70937c8fbf403630ce 03-Jul-2015 jbauch <jbauch@webrtc.org> Report metrics about negotiated ciphers.

This CL adds an API to the metrics observer interface to report negotiated
ciphers for WebRTC sessions. This can be used from Chromium for UMA metrics
later to get an idea which cipher suites are used by clients (e.g. compare
the use of DTLS 1.0 / 1.2).

BUG=428343

Review URL: https://codereview.webrtc.org/1156143005

Cr-Commit-Position: refs/heads/master@{#9537}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
be24c94c95056e4f0a22039f25f2fa8a27be6b66 23-Jun-2015 jbauch <jbauch@webrtc.org> Set / verify stats report timestamps.

This CL updates the track report timestamps which were fixed at "0" before
and updates the timestamps in reports for local audio tracks.

Also the timestamps are checked in various tests to make sure no "0" is
returned.

Original CL is at https://webrtc-codereview.appspot.com/51829004/

BUG=webrtc:4316
TBR=hta@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1204493002

Cr-Commit-Position: refs/heads/master@{#9485}
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
04e5b498278c633bc3c49da43d08c15b1e75ebc0 29-May-2015 Joachim Bauch <jbauch@webrtc.org> Make maximum SSL version configurable through PeerConnectionFactory::Options

This can be used to activate DTLS 1.2 through a command-line flag from Chromium
later.

BUG=chromium:428343
R=jiayl@webrtc.org, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/54509004

Cr-Commit-Position: refs/heads/master@{#9328}
/external/webrtc/talk/app/webrtc/peerconnection_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/talk/app/webrtc/peerconnection_unittest.cc
61e00b0bcab899a32f14c1e2e0f4b7f316cc1f03 04-Mar-2015 jiayl@webrtc.org <jiayl@webrtc.org> Create a in-memory DTLS identity store that keeps a free identity generated in the background.

BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Committed: https://code.google.com/p/webrtc/source/detail?r=8581

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8605}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8605 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
7bea1ffe772e837d96f8faa5c9dd06e531b95379 04-Mar-2015 pthatcher@webrtc.org <pthatcher@webrtc.org> Expose negotiated ciphers through stats API.

Use the new internal API to expose the negotiated SRTP/SSL ciphers
through the stats API.
This is a follow-up to https://webrtc-codereview.appspot.com/37209004.

BUG=3976
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35169004

Cr-Commit-Position: refs/heads/master@{#8584}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8584 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
be77872d2ce7a5faf15d3794635456ee81a5ced1 04-Mar-2015 jiayl@webrtc.org <jiayl@webrtc.org> Revert "Create a in-memory DTLS identity store that keeps a free identity generated in the background."

Breaking Chromium FYI.

TBR=pthatcher@webrtc.org

This reverts commit 369f68255ffd3d6f3e449e0defeae820cefd4f29.

BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004


Review URL: https://webrtc-codereview.appspot.com/47389004

Cr-Commit-Position: refs/heads/master@{#8583}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8583 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
369f68255ffd3d6f3e449e0defeae820cefd4f29 04-Mar-2015 jiayl@webrtc.org <jiayl@webrtc.org> Create a in-memory DTLS identity store that keeps a free identity generated in the background.

BUG=4241
R=pthatcher@webrtc.org

Committed: https://code.google.com/p/webrtc/source/detail?r=8576

Review URL: https://webrtc-codereview.appspot.com/37889004

Cr-Commit-Position: refs/heads/master@{#8581}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8581 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
40fdb8ab9669ee22b2723d154afeeebd44a08b5d 13-Feb-2015 solenberg@webrtc.org <solenberg@webrtc.org> Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway.

BUG=3871
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41879004

Cr-Commit-Position: refs/heads/master@{#8359}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8359 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
503c33666ff7c382b540296755793ddab8d4b909 12-Feb-2015 solenberg@webrtc.org <solenberg@webrtc.org> Re-enabling LocalP2PTestAnswerVideo and LocalP2PTestAnswerAudio test cases in peerconnection_unittest.

BUG=2288
R=perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39919004

Cr-Commit-Position: refs/heads/master@{#8350}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8350 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
5f93d0a140515e3b8cdd1b9a4c6f5871144e5dee 20-Jan-2015 jlmiller@webrtc.org <jlmiller@webrtc.org> Update libjingle license statements at top of talk files for consistency

BUG=2133
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39559004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8105 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
9eacb8cc5911eb38d7f31d0cfe07bde981d33316 02-Jan-2015 pbos@webrtc.org <pbos@webrtc.org> Make P2PTestConductor use VirtualSocketServer.

Permits running JsepPeerConnectionP2PTestClient in parallel.

TBR=juberti@webrtc.org
BUG=2598
TEST=third_party/gtest-parallel/gtest-parallel -w 128 -r 100 out/Debug/libjingle_peerconnection_unittest --gtest_filter=JsepPeerConnectionP2PTestClient.*

Review URL: https://webrtc-codereview.appspot.com/37459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7988 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
c2dd5ee2c05b466949fedae3fcfac63838104392 04-Nov-2014 perkj@webrtc.org <perkj@webrtc.org> Prepare for removal of PeerConnectionObserver::OnError.
Prepare for removal of constraints to PeerConnection::AddStream.

OnError has never been implemented and has been removed from the spec.
Also, constraints to PeerConnection::AddStream has also been removed from the spec and have never been implemented.

R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7605 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
269fb4bc90b79bebbb8311da0110ccd6803fd0a8 28-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
28100cb38896fe298b6df11ffd31838d9faf5b8a 18-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."

BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
d1ba6d9cbfc44618d2c553ff7851948c730ae37b 15-Oct-2014 henrike@webrtc.org <henrike@webrtc.org> Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.

BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
626624061e0de73b346027660383d7ec006ae3b8 29-Sep-2014 asapersson@webrtc.org <asapersson@webrtc.org> Disable flaky tests:
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined

BUG=3871
R=henrike@webrtc.org, kjellander@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7323 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
34f2a9ea7245bac103fececfa53e92359680467a 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/talk/app/webrtc/peerconnection_unittest.cc
3987b6de506a7e72a5bdfdf8c8ad9964705c5a28 24-Sep-2014 jiayl@webrtc.org <jiayl@webrtc.org> Fix a problem in Thread::Send.
Previously if thread A->Send is called on thread B, B->ReceiveSends will be called, which enables an arbitrary thread to invoke calls on B while B is wait for A->Send to return. This caused mutliple problems like issue 3559, 3579.
The fix is to limit B->ReceiveSends to only process requests from A.
Also disallow the worker thread invoking other threads.

BUG=3559
R=juberti@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/15089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7290 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
000d86792d6fb57948ce60b3a3e9c8f34768f46c 15-Sep-2014 pbos@webrtc.org <pbos@webrtc.org> Make BW checks > 0 in peerconnection_unittest.cc.

These checks (> 40k) fail on LSan FYI bots and the purpose of them seem
to be that we're getting non-zero BW reported.

R=stefan@webrtc.org
TBR=jiayl@webrtc.org, solenberg@webrtc.org
BUG=3817,chromium:375154

Review URL: https://webrtc-codereview.appspot.com/29479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7183 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
ceb956b29dc28ffac03450240ce6a5741989a762 04-Sep-2014 pbos@webrtc.org <pbos@webrtc.org> Abort Negotiate() if DoCreateOffer() fails.

Addressing crash in test.

R=jiayl@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/19239004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7066 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
00f11f5e2445d5ede48c394e8478308812bbbb71 27-Aug-2014 solenberg@webrtc.org <solenberg@webrtc.org> - Make local constant non-static.
- Remove spammy log line.

BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6987 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
e9bfed0648656a22b41a9357e50a57d3c2d17e14 25-Aug-2014 kjellander@webrtc.org <kjellander@webrtc.org> Move constant so it is not stripped out for TSAN bots.

BUG=
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6971 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
6556a59db1efca3a4796d17de72420b0ddbcb29e 25-Aug-2014 solenberg@webrtc.org <solenberg@webrtc.org> As expected, r6569 (https://code.google.com/p/webrtc/source/detail?r=6965) caused memcheck bots to complain. Adding expections for that, in line with outher peerconnection tests.

Also, caused some issues with other peerconnection_unittest tests, so changed the design of those.

BUG=
R=kjellander@webrtc.org, perkj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22019004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6968 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
b4c7b09c1352174ecc1faf8c0cd93c66028a0485 25-Aug-2014 buildbot@webrtc.org <buildbot@webrtc.org> (Auto)update libjingle 73927775-> 74032598

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
a09a99950ec40aef6421e4ba35eee7196b7a6e68 13-Aug-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 73222930-> 73226398

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
e7d47a1473e885a57986dcdbf06e7e1d25226ca6 05-Aug-2014 jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Maintain the order of the m-lines in CreateOffer and CreateAnswer.
The order in the offer follows the order in the current local description.
The order in the answer follows the order in the current offer.

BUG=2395
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12799004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6828 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
d4e598d57aed714a599444a7eab5e8fdde52a950 29-Jul-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 72097588-> 72159069

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
6c6f33b5bb934602896cdf06c9397fca1b9f6bdf 12-Jun-2014 jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Fix the flaky RTP DataChannel test.

BUG=2891
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/18519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6418 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
044bdacfefa860715e84663d4df651e8f4984469 03-Jun-2014 pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Remove kMaxWaitForStatsMs from tsanv2 compilation.

As some tests are #ifdef'd out on THREAD_SANITIZER this constant
triggers an unused-const-variable warning which breaks the build.

BUG=1205,3220
TBR=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6308 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
688ed699e0a95e91777a15f5b507139af627f11b 14-May-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 67017551-> 67023528

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6158 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
3e01e0b16cbde481241b9bcfdbbdd591cd920b99 13-May-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 66867790-> 66887616

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6128 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
9c16c39e613ebc5cdfa8ca5818a62ef5c3b18bd7 01-May-2014 jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Sets the SCTP port codec in the native SessionDescription.
Previously it's only set when a SDP string is parsed into SessionDescription, causing failuring for native client.

BUG=3141
R=juberti@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6036 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
8f88f20af239805155f1540d7da53e106dd195d7 16-Apr-2014 jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Expand the test max wait time from 1000ms to 2000ms.
The createOffer/createAnswer methods sometimes times out due to slow identity generation under memcheck.

BUG=2838
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5920 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
61c1b8ea32d1801384151286ad8bd4eeccacf34b 09-Apr-2014 buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> (Auto)update libjingle 64585415-> 64594651

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5870 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
b90991dade9139e5c14c3b616a9eff07b9d6fdda 04-Mar-2014 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update libjingle 62472237->62550414

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5640 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
db41b4dbcdeb9a3b71b8de274db8654f3e51c99c 03-Mar-2014 jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Remove the deprecated GetStats method from PeerConnectionInterface.

R=fischman@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
385857dfd414dcc1fb4941218b52417808349030 14-Feb-2014 mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update talk to 61549749.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5549 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
9cf037b83184374230c6825e4aa407cdafaba434 07-Feb-2014 sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update libjingle to 61168196

R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5502 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
a576faf82a692c9422dcdc3278394ed25e6ee4f7 29-Jan-2014 jiayl@webrtc.org <jiayl@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Enable SCTP and use OPENSSL on Anroid and NSS on other platforms.
It includes unit test fixes to properly initialize SSL if DTLS or SSL random number generator is used in the tests.
The private key and certificate constant strings used in some tests are updated to be compatible with NSS.
A few potentially overflow type conversions caused compiling warning on Windows and they are fixed by importing and using Chromium's checked_cast, which aborts the program if overflow occurs.
It also fixes a leak in nssstreamadapter.cc by releasing the PRFileDesc* in StreamClose.

BUG=2253
R=fischman@webrtc.org, juberti@google.com, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/4679005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5459 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
97077a3ab27259164eb121034b6e0ebe9ba592df 25-Oct-2013 wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update libjingle to 55618622.
Update libyuv to r826.

TEST=try bots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
50bc5538525960d4a5346dbc6c4669e258eea28e 21-Oct-2013 mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Reenable DTLS renegotiation unittest in libjingle.

This test is failing on memcheck bots. After investigation problem per
say is not in this particular unittest and rather is in suite. So this test
is added to memcheck exclude list.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5011 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
d1cfa7149e5997010bdc8106dc2df3ff76367075 16-Oct-2013 kjellander@webrtc.org <kjellander@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> TSan v2 suppressions and exclusions for libjingle tests.

Add suppressions for libjingle tests so they pass under TSan v2.
Disable the following tests for TSan v2 (only) since they're failing:
* StunServerTest.TestGood
* JsepPeerConnectionP2PTestClient.*

See build logs at:
http://build.chromium.org/p/client.webrtc.fyi/builders/Linux%20Tsan%20v2/
for more details.

BUG=1205,2078,2079,2080,2517
TEST=Ran a successful run of each test locally on Linux using:
GYP_DEFINES='tsan=1 linux_use_tcmalloc=0 release_extra_cflags="-gline-tables-only"' gclient runhooks
ninja -C out/Release
For each test, run standing in trunk/:
TSAN_OPTIONS="suppressions=tools/valgrind-webrtc/tsan_v2/suppressions.txt print_suppressions=1 report_signal_unsafe=0 report_thread_leaks=0 history_size=7" out/Release/[libjingle_testname]
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2411004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4977 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
6fa456f92826024921d578c7bf076e7ea2414198 15-Oct-2013 mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Disabling the DTLS renegotiation test case for PeerConnection.
Currently it's failing on Linux memcheck, most likely due to timing issues.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2394006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4962 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
19f27e6a24f877fc2b0409a94b02d5f40ba3dc8c 13-Oct-2013 mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update talk to 54527154.

TBR=wu

Review URL: https://webrtc-codereview.appspot.com/2389004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4954 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
da79008ab4e2d1f652199ea2f927892291e28f5e 17-Sep-2013 stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Disabling crashing or flaky tests in peerconnection_unittest.

R=kjellander@webrtc.org
TBR=wu@webrtc.org
TESTS=trybots
BUG=2378

Review URL: https://webrtc-codereview.appspot.com/2227004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4767 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
a59696b2a5f0c138d4176249bac223ad6c4316d5 14-Sep-2013 sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update libjingle to 52300956

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2213004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4744 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
c0b1a280ab8eaebeccf5317230c3bb826454020b 23-Aug-2013 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Some tests were not disabled correctly as it should be DISABLED_* not DISABLE_*.

TBR=wu@webrtc.org

BUG=N/A

Review URL: https://webrtc-codereview.appspot.com/2095005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4602 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
61b262c427d5747825d3582086786fab68d12a09 22-Aug-2013 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Disable tests according to issues: 1205,2272,2288,2290,2291

BUG=1205,2272,2288,2290,2291
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2069005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
28654cbc2256230c978f41cbaf550bc2e9c2f2db 22-Jul-2013 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update talk folder to revision=49713299.

TBR=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1848004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
723d683ecbe6a934885a60712c66ca2c01700a51 12-Jul-2013 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository.

TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1797004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc
28e20752806a492f5a6a5d343c02f9556f39b1cd 10-Jul-2013 henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> Adds trunk/talk folder of revision 359 from libjingles google code to
trunk/talk


git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/app/webrtc/peerconnection_unittest.cc