The address and the related address of local candidates are sanitized
accordingly when the mDNS concealment of local IPs is enabled. Also,
remote hostname candidates created from signaling are sanitized in stats
as well. A couple of unit tests are revised to reflect the desired
behavior of AsyncResolverInterface so that when a hostname candidate is
resolved, the hostname is kept in the candidate address.
Bug: webrtc:9605, chromium:914452
Change-Id: Iad9ad04ce4e50304e44cf04b15b97a7ae2dec960
Reviewed-on: https://webrtc-review.googlesource.com/c/113643
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Jeroen de Borst <jeroendb@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25996}
This is a propagation of upstream chromium change needed to
resume DEPS autorolls into WebRTC.
Original comment from upstream change:
> This change is made in preparation for an ErrorProne
> check to catch this at compile time. See bug for details.
Bug: chromium:771683
Change-Id: I56aed15f73a633dcadae7ece6c645cd3596f9257
Reviewed-on: https://webrtc-review.googlesource.com/c/113505
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25951}
So that users can add dependencies on them, and not break when a bunch
of headers move out of rtc_base:rtc_base.
Bug: webrtc:9987
Change-Id: Iecd5dd903cb8b97cb6f051e3a0cb6df7f8ba22b3
Reviewed-on: https://webrtc-review.googlesource.com/c/113425
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25923}
Since not all fields are compared on NetworkRoute structs, the ==
operator overload doesn't really make the code easier to read. In fact
the feature that it only compares a subset of the fields is only used
once, at the other places, all fields are compared.
Removing the overload makes it more clear what is compared at each call
site.
Bug: webrtc:9883
Change-Id: I74f7eb32b602aa33fd282a815b71a172ae3f6a8b
Reviewed-on: https://webrtc-review.googlesource.com/c/113001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25891}
This includes a refactoring of jseptransport to store a refcounted
object instead of a std::unique_ptr to the cricket::DtlsTransport.
Bug: chromium:907849
Change-Id: Ib557ce72c2e6ce8af297c2b8deb7ec3a103d6d31
Reviewed-on: https://webrtc-review.googlesource.com/c/111920
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25831}
Windows UWP allows an application to be built that targets
across all Windows 10 based systems and the Windows store.
Change-Id: I69694bb7e83fb01ad6db2438b065b55738cf01fd
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/110570
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25814}
std::is_trivially_* is not available on certain old STL
implementations. Using absl implementation will allow
maximized compatibility.
Bug: webrtc:10054
Change-Id: I17ed0fff44328b3d7c51d14e8c4470f1df0e66ad
Reviewed-on: https://webrtc-review.googlesource.com/c/111728
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25788}
WinUWP cannot use the win task queue as post/peek message event loop
is not available. A replacement version written using stdlib compatible
with WinUWP is added as an alternative.
Change-Id: Ie9d6e6f11f395d1815d8f04633772a0c597ed30a
Bug: webrtc:10046
Reviewed-on: https://webrtc-review.googlesource.com/c/108520
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25781}
This CL decouples //rtc_base:rtc_base_tests_utils from gunit by
moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h
(which depends on gunit helpers) to their own build target.
It also removes some unused dependencies in the WebRTC build graph.
Bug: None
Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623
Reviewed-on: https://webrtc-review.googlesource.com/c/111861
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25769}
Our macros always pass __FILE__ and __LINE__ as parameters, so the impact was limited. However, doing the correct thing is obviously preferable to doing the wrong thing, so let's fix it.
Bug: webrtc:10003
Change-Id: Id2529c4bd8c7e90a8f0ac3ffa713dbe305ba66d8
Reviewed-on: https://webrtc-review.googlesource.com/c/111244
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25712}
rtc::scoped_refprt is used in WebRTC api/ code so it makes sense to
move it to api/ and remove exceptions from api/DEPS.
Bug: webrtc:9887
Change-Id: If58c387e5fdfacd8fc1830b4bd79fa1a73942cc9
Reviewed-on: https://webrtc-review.googlesource.com/c/111252
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25698}
This reduces code duplication and ensures common behavior
between the unit classes.
Bug: webrtc:9709
Change-Id: I9529ef10b3f538355f53250a2b67c6b4e250cce8
Reviewed-on: https://webrtc-review.googlesource.com/c/110901
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25690}
Those alias do not save much typing, but may cause conflicts, specially the one in the header
Bug: None
Change-Id: Ifb17f639e528aaff72861ff55dcd7a96a229715d
Reviewed-on: https://webrtc-review.googlesource.com/c/110784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25628}
cricket::Port::SignalPortComplete.
The mDNS name registration is asynchronously executed by the mDNS
responder, and a host candidate with an mDNS name is only gathered after
this completes. SignalPortComplete however is currently done
synchronously by UDPPort, and any candidate gathered by a UDPPort after
this signal is fired would be discarded.
Bug: webrtc:9964, webrtc:9605
Change-Id: If8aaf193ef26c06bd118e6418b62ba0de5e87e3c
Reviewed-on: https://webrtc-review.googlesource.com/c/109541
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Zach Stein <zstein@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25534}
Replaced by a int64_t representing time in us. To aid transition of
downstream code, rtc::PacketTime is made an alias for int64_t.
Bug: webrtc:9584
Change-Id: Ic3a5ee87d6de2aad7712894906dab074f1443df9
Reviewed-on: https://webrtc-review.googlesource.com/c/91860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25503}
This utility class is needed in rtcp_rtp. Instead of reimplementing it
again, the existing class is moved to rtc_base, cleaned from unused
features and extended as required for the new usage.
Bug: webrtc:9914
Change-Id: I3b0d83d08d8fa5e1384b4721a93c6a90781948fd
Reviewed-on: https://webrtc-review.googlesource.com/c/109081
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25498}
The replacements are absl::EqualsIgnoreCase and
absl::StartsWithIgnoreCase. Also delete the alias
RtpUtility::StringCompare.
Bug: webrtc:6424
Change-Id: I4bed71540264450f85137ad0c2564125c5c6213f
Reviewed-on: https://webrtc-review.googlesource.com/c/109006
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25481}
The frame time deltas are now capped based on the current noise.
This has been tested in various conditions using both screen content
and typical mobile video settings, to produce delays that are not overly
high screen content, and simultaneously not negatively affect mobile
calls on really bad network that may have high natural jitter.
Bug: webrtc:9898
Change-Id: I51ad279af156aba1b5cc75ae203334a34bce9d48
Reviewed-on: https://webrtc-review.googlesource.com/c/107349
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25469}
This is a preparation for deleting rtc::PacketTime. Next step, after
downstream code has been updated to not access the |timestamp| member,
is to make rtc::PacketTime an alias for int64_t.
Also delete the unused member rtc::PacketTime::not_before.
Bug: webrtc:9584
Change-Id: Iba9d2d55047d69565ad62b1beb525591fd432ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/108860
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25468}
to Mdns.*.
MdnsResponderInterface now explicitly requires the reference counting
of created names to allow the coexistence of multiple users of the same
responder where one user would not remove identical names created by
others.
MDns.* is also renamed to Mdns.* per the style guide.
TBR=aleloi@webrtc.org
Bug: webrtc:9605
Change-Id: I047fc41f34de8d4e97c980409a7f373769c4c252
Reviewed-on: https://webrtc-review.googlesource.com/c/101921
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25458}
Moved methods: GetReadData, ConsumeReadData, GetWriteBuffer,
ConsumeWriteBuffer, GetWriteRemaining.
These methods represented an optional interface for reading and
writing streams, intended to optimize certain use cases. However,
it was implemented only in the FifoBuffer subclass, and the few
users of that class all have a concrete FifoBuffer, and hence
don't need the methods on the abstract StreamInterface.
Bug: webrtc:6424
Change-Id: I6de74d1a9205fcb7037ad84e24679d4a27c1d219
Reviewed-on: https://webrtc-review.googlesource.com/c/108621
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25446}
This change just wraps the openssl key derivation functions in a simple
interface in a similar way to how we do it for messagedigest.h so we aren't
coupled to openssl in the core implementation.
Bug: webrtc:9917
Change-Id: I8556bd6e38b7da34d93abbe29415c3366f6532ba
Reviewed-on: https://webrtc-review.googlesource.com/c/107981
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25440}
Refactor only remaining user, IsDefaultRoute (helper function
called from BasicNetworkManager::IsIgnoredNetwork) to use a
FILE* and fgets instead.
Bug: webrtc:6424
Change-Id: I57652f664b9a6965c19575c1b5d7f7de24f2ed44
Reviewed-on: https://webrtc-review.googlesource.com/c/108089
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25433}
Problem found while refactoring usage in examples/turnserver/.
Bug: webrtc:6424
Change-Id: Ib1d54055c5914136b5bf165d48ab7d19520ff967
Reviewed-on: https://webrtc-review.googlesource.com/c/108302
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25412}
anonymous namespace.
There is some really scary code in this function that I did not refactor in
this change. I believe the ASN parsing code should be removed completely
and have attached TODOs to do this once we have a correct test suite to validate
the functionality. I am almost certain openssl has functions that do this
better.
Bug: webrtc:9860
Change-Id: Ice06079eb1e5b10bdb2ee45ae45cbfb2ce8f6f13
Reviewed-on: https://webrtc-review.googlesource.com/c/108206
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25404}
This is some of the older code in the code base and is using raw gotos. This
first pass of the file just does some basic refactorings to make the code more
readable.
Bug: webrtc:9860
Change-Id: Ic7b8dc51fe4b43af77c44dd725877bd0f4d47aec
Reviewed-on: https://webrtc-review.googlesource.com/c/108202
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25403}
Just a simple rename change to update these functions to be in compliance with
the WebRTC/Chromium style guide.
Bug: webrtc:9860
Change-Id: I5bc831754c80b7b00bd1e5e0b3905e55f5d22b0c
Reviewed-on: https://webrtc-review.googlesource.com/c/108204
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25402}
OpenSSL implementations are all final implementations of their more abstract
SSL variants. This should be both documented and enforced by the use of the
final keyword to indicate to future WebRTC contributors that this is the
intended depth of inheritance and it shouldn't be extended again. Hopefully
this minor change will help keep the code simpler to maintain going forward.
Bug: webrtc:9860
Change-Id: Ie22de722214e3b209c3d7727a93ac819c112434e
Reviewed-on: https://webrtc-review.googlesource.com/c/108203
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25401}
We have several places in the SSL APIs where we will poke holes through the API
surface with boolean flags to enable scenarios like disabling authentication.
This isn't an ideal approach because it is error prone and confusing to the
API user. Instead authentication should be dependency injected with a default
secure component and a fake can be created for testing.
For now this CL just cleans up the left over unused test flags and renames the
remaining ones with a ForTesting postfix to make it very clear they shouldn't
be used in any production code.
Bug: webrtc:9860
Change-Id: I31f55cf85097bacb9cd895c16a6fad3773cd1c2b
Reviewed-on: https://webrtc-review.googlesource.com/c/107786
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25377}
