- add DTLS1.3 ciphers (without KeyType)
- remove code in dtls_transport.cc that tries to parse DTLS packet
- cleanup some test
- start on test for packet loss during dtls handshake (more to come!)
After this patch is submitted, it is possible
to set max version = dtls1.3 and it will active
but DON'T do it yet.
BUG=webrtc:383141571
Change-Id: I6f9a120c53415ccee7a560ea83bd0c2636702997
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371300
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43595}
by going from cartesian product of combinations to an explicit
minimal list.
BUG=webrtc:375552698
Change-Id: I99b9afd7376f19abde54dafd8917954617d8c255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367504
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43373}
and remove the testing of the nondefault key size from the "server" parameters to speed up tests
BUG=webrtc:375552698
Change-Id: Ibc1bd491300964aa45826b98962ed3e56c6d4974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366941
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Christoffer Dewerin <jansson@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43321}
and use uint8_t instead of unsigned char. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/365274
BUG=webrtc:357776213
Change-Id: Ibc97e5cc85316ba69b4133b7f3c42e3afbdd7abd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365540
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43263}
This is a reland of commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d
with more backward compat which also fixes the off-by-one issue which caused wrong SRTP keys to be extracted.
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I5d43dc23f90ef630834fb400751979fcc5e18203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43225}
This reverts commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d.
Reason for revert: breaks downstream compilation
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I03ffcda3d6821718f355b243ce78a9c54b4036f3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365062
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43202}
and simplify the interface used as this is only used for exporting
SRTP keys and passing arcane OpenSSL arguments around does not make
much sense.
BUG=webrtc:357776213
Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43198}
since 1024 is already deprecated by OpenSSL and causes "too small key"
issues on systems enforcing a minimum size. Similar issue here:
https://github.com/nodejs/node/pull/44498
The minimum key size is not yet changed from 1024, this will require more effort for deprecation.
BUG=webrtc:364338811
Change-Id: Id4b24a2c289ec5e3f112288d32b8ac697ba1cfed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361128
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43110}
improving the error message from PEM parsing and adding a few DCHECKs
Tested locally with OpenSSL 3.x
BUG=webrtc:42225468
Change-Id: Ia2ff1e5826f486060db73bee979e2703fc6c5823
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358441
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42776}
since OpenSSL 3.x deprecated SHA1 there and SHA256 has been the default
in Chrome for a decade. Test all variants with a reduced test suite.
BUG=webrtc:42225468
Change-Id: I728bfd953c3248d6a7804c55ab71009fcc701a45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358820
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42765}
since we do not need two adapters with TLS and DTLS modes.
SSLAdapter is the TLS adapter,
SSLStreamAdapter is the DTLS adapter.
BUG=webrtc:353750117
Change-Id: I223917c71c88437339380e1f196dcf3c0e2021c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42713}
this is flipping
WebRTC-PermuteTlsClientHello
to a killswitch in the SSLStreamAdapter used for DTLS.
BUG=webrtc:42225803
Change-Id: I942851c474ec5e723c5b6c9f6206e7eafbe80ea4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42676}
keeping around the DTLS 1.0 constant for unit tests.
BUG=webrtc:40644300
Change-Id: I6d0c3ba1f434bbf3ef1a1b812aeef26943dcf646
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352530
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42471}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
In its stead, there's now a SetEventCallback() method.
Bug: webrtc:11943
Change-Id: If936d6e1e23e8a584f06feb123ecf2d450ea4145
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/319040
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42187}
This is a step towards removing StreamInterface::SignalEvent.
Downstream dependency will need to be updated to call FireEvent()
before further changes can land in webrtc.
Bug: webrtc:11943
Change-Id: Ia7d3f1c43fda52b7cf5bfa082aef3f462553cd67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347884
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42143}
This makes it easier to remove use of sigslot for SignalEvent
since the tests were written in a way that could set more than one
event handlers to the same callback method, which places unnecessary
requirements on the definition of the callback object. I.e. the
sigslot can't be replaced with a simple (single) std::function - which
would be consistent with how the event callback is used elsewhere
in the code.
Bug: webrtc:11943
Change-Id: I7e596295b1b534d4d49334449b1e01535eedf06d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344723
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42072}
which has been enabled by default since M84. This was still available
under an enterprise policy which is gone since M121:
https://chromiumdash.appspot.com/commit/39d28bb7657b482f1fdcab81ca88371d8914809b
BUG=webrtc:10261,chromium:1132854
Change-Id: Icd534342b60799b7862bc3e7edda6825de7ae976
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317360
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41145}
Speculative fix; replaces nullpointer following with test failure.
Bug: b/307520538
Change-Id: I294d5747c852a8cde43a0acad973dca12712c69e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326000
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41077}
These tests were failing when building WebRTC against OpenSSL instead of
BoringSSL. The reason is that OpenSSLStreamAdapter::SSLVerifyCallback in
the BoringSSL mode returns the full cert_chain by calling
SSL_get0_peer_certificates. This API does not exist in OpenSSL, instead
only a single certificate is fetched via X509_STORE_CTX_get0_cert.
ifdef out the parts of the test that assert on cert[1] and cert[2].
An alternative but more involved way to fix these tests could be to use
X509_STORE_CTX_get1_chain to fetch the full chain on the OpenSSL path.
Bug: webrtc:15153
Change-Id: I1ede6a3c5a63d4afd2de849f5e44fcd67592aa3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40022}
This cleans up the last vestiges of the old interface for rtc::StreamInterface
and will cause builds to refer to the old functions to fail.
Bug: webrtc:14632
Change-Id: I569b16677754d7f9e08449e273672a59a86e6498
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283844
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38784}
Updates the code to use the new interfaces
Bug: webrtc:14632
Change-Id: I33b2a25b5968de0251e3cbc84076afc013ecef6e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282680
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38601}
A few usages in ssl_stream_adapter_unittests are converted to make
sure the aliases are usable.
Next steps are:
- Change all usages inside WebRTC to the new form
- Deprecate the old API
- Remove the old API
Pipewire failures believed to be unrelated, so No-try.
No-try: true
Bug: webrtc:14632
Change-Id: I618551e61a05d53e524e97483d3c7cef59b88a25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282221
Reviewed-by: Tomas Gunnarsson <tommi@google.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38577}
which signals a permanent connection failure to the application
BUG=webrtc:13999
Change-Id: I7ba25db4aa9035583558a613db97561c48796c76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260100
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#36700}
This is part of a large-scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL converts the majority of "const std::string&"s in function
parameters under rtc_base/ to absl::string_view.
Bug: webrtc:13579
Change-Id: I2b1e3776aa42326aa405f76bb324a2d233b21dca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254081
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Anders Lilienthal <andersc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36239}
Uppercase constants are more likely to conflict with macros (for
example rtc::SRTP_AES128_CM_SHA1_80 and OpenSSL SRTP_AES128_CM_SHA1_80).
This CL renames some constants and follows the C++ style guide.
Bug: webrtc:12997
Change-Id: I2398232568b352f88afed571a9b698040bb81c30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226564
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34553}
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.
Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.
No-Try: True
Bug: webrtc:11410
Change-Id: I86ddb361b94ad85b15ebb8743490de83632ca53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196941
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32818}
This reverts commit 72f638a9a279e7abb5534fa66a0ade2cf18ec1a7.
Reason for revert: downstream build failures
Original change's description:
> Use CRYPTO_BUFFER APIs instead of X509 when building with BoringSSL.
>
> Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
> security gains, and will provide binary size improvements as well once
> the default list of built-in certificates can be removed; the code
> dealing with them still depends on the X509 API.
>
> Implemented by splitting openssl_identity and openssl_certificate
> into BoringSSL and vanilla OpenSSL implementations.
>
> Bug: webrtc:11410
> Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: David Benjamin <davidben@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32811}
TBR=deadbeef@webrtc.org,mbonadei@webrtc.org,davidben@webrtc.org,hta@webrtc.org
Change-Id: Ib5e55cb5798a2f3d25a4460f5311d2e650d3fa82
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11410
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196742
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32812}
Using CRYPTO_BUFFERs instead of legacy X509 objects offers memory and
security gains, and will provide binary size improvements as well once
the default list of built-in certificates can be removed; the code
dealing with them still depends on the X509 API.
Implemented by splitting openssl_identity and openssl_certificate
into BoringSSL and vanilla OpenSSL implementations.
Bug: webrtc:11410
Change-Id: Idc043462faac5e4ab1b75bedab2057197f80aba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174120
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: David Benjamin <davidben@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32811}
Now this requires explicitly passing the
WebRTC-LegacyTlsProtocols/Enabled/ field trial flag or an override.
Bug: webrtc:10261
Change-Id: Ib880bcc50cec0a21dcaa4784c228cacb020e5568
NOKEYCHECK: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190282
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32488}
This includes relying on related types such as MessageData and
PostEvent functionality inside the StreamInterface itself.
This affects mostly tests but OpenSSLStreamAdapter
requires special attention.
Bug: webrtc:11988
Change-Id: Ib5c895f1bdf77bb49e3162bd49718f8a98812d91
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185505
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32290}
Also remove test code that can cause leaks into production.
Add sequence checkers.
Bug: webrtc:11988
Change-Id: I67b4cec6ee77d73ccffbbc88c9081ebb3c3cc423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185503
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32228}
This reverts commit af1f8655b2cb69af382396ea642eb0a2bf04bb4d
Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.
Original change's description:
> Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC."
>
> This reverts commit 7276b974b78ea4f409d8738b1b6f1515f7a8968e.
>
> Reason for revert: Changing to a later Chrome release.
>
> Original change's description:
> > Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
> >
> > This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> > is part of a larger effort at Google to remove old TLS protocols:
> > https://security.googleblog.com/2018/10/modernizing-transport-security.html
> >
> > For the M74 timeline I have added a disabled by default field trial
> > WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> > as consumers move away from these legacy cipher protocols but it will be off
> > in Chrome.
> >
> > This is compliant with the webrtc-security-arch specification which states:
> >
> > All Implementations MUST implement DTLS 1.2 with the
> > TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> > curve [FIPS186]. Earlier drafts of this specification required DTLS
> > 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> > at the time of this writing some implementations do not support DTLS
> > 1.2; endpoints which support only DTLS 1.2 might encounter
> > interoperability issues. The DTLS-SRTP protection profile
> > SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> > Implementations MUST favor cipher suites which support (Perfect
> > Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> > over non-AEAD cipher suites.
> >
> > Bug: webrtc:10261
> > Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: David Benjamin <davidben@webrtc.org>
> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27006}
>
> TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10261
> Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27403}
Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
