default connection address is a hostname candidate.
Using a FQDN in the c= line has caused an inter-op issue with Firefox
when hostname candidates are the only candidates gathered when forming
the media sections. To address this issue, we use 0.0.0.0:9 when a
hostname candidate would be used to populate the c= and the m= lines.
The SDP grammar related to ICE candidates has been moved out of RFC8445,
and is currently defined in draft-ietf-mmusic-ice-sip-sdp. A FQDN
address must not be used in the connection address attribute per the
latest draft, if the ICE agent generates local candidates. Also, the
wildcard addresses (0.0.0.0 or ::) with port 9 are given the exception
as the connection address that will not result in an ICE mismatch. We
thus adopt the aforementioned solution after combining these
considerations.
Bug: chromium:927309, chromium:982108
Change-Id: I3df2db0f154276da39f99650289cf81baa677e74
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145280
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28547}
This documents in the API what is already true in the
implementation - that SessionDescription will eventually
delete MediaDescription objects passed to it.
The old API is preserved for backwards compatibility, but
marked as RTC_DEPRECATED.
Bug: webrtc:10701
Change-Id: I9a822b20cf3e58c5945fa51dbf6082960a332de8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139880
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28144}
Semi-automatically created with:
git grep -l " testing::" | xargs sed -i "s/ testing::/ ::testing::/g"
git grep -l "(testing::" | xargs sed -i "s/(testing::/(::testing::/g"
git cl format
After this, two .cc files failed to compile and I have fixed them
manually.
Bug: webrtc:10523
Change-Id: I4741d3bcedc831b6c5fdc04485678617eb4ce031
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132018
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27526}
Googletest recently started replacing the term Test Case by Test Suite.
From now on, the preferred API is TestSuite*; the older TestCase* API
will be slowly deprecated.
This CL moves WebRTC to the new set of APIs.
More info in [1].
This CL has been generated with this script:
declare -A items
items[TYPED_TEST_CASE]=TYPED_TEST_SUITE
items[TYPED_TEST_CASE_P]=TYPED_TEST_SUITE_P
items[REGISTER_TYPED_TEST_CASE_P]=REGISTER_TYPED_TEST_SUITE_P
items[INSTANTIATE_TYPED_TEST_CASE_P]=INSTANTIATE_TYPED_TEST_SUITE_P
items[INSTANTIATE_TEST_CASE_P]=INSTANTIATE_TEST_SUITE_P
for i in "${!items[@]}"
do
git ls-files | xargs sed -i "s/\b$i\b/${items[$i]}/g"
done
git cl format
[1] - https://github.com/google/googletest/blob/master/googletest/docs/primer.md#beware-of-the-nomenclature
Bug: None
Change-Id: I5ae191e3046caf347aeee01554d5743548ab0e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/118701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26494}
