Reasons:
- the code is no longer used in Chrome
- it is conceptually weird for WebRTC to have JSON parsing in its API
- there are concerns around the reliability of the underlying JSON library
Additionally, this CL removes the rtc_json "poisonous" attribute: the scheme is incompatible and redundant with testonly.
Bug: webrtc:1493351
Change-Id: I0b621b0e3f183df7315919d9c89242fbe387928f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325062
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41014}
Required logic to query the min value of a SampleCounter along with some
additions to the existing test cases
Bug: webrtc:15580
Change-Id: I46afb30ad130f17f9e68ebc794b6935187bb2479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323900
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40956}
Convert most field trials used in PCLF tests.
Change-Id: I26c0c4b1164bb0870aae1a488942cde888cb459d
Bug: webrtc:10335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322703
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40909}
Checking in sending classes avoids using global field trial string in favor of the injected one.
In addition to that RateLimiter looks wrong layer for check that field trial:
checking inside RateLimiter class might be surprising if it is used for limiting something else than RTX bitrate.
evaluating field trial for each retransmitting packet might be expensive
Bug: webrtc:15184, webrtc:10335
Change-Id: I87bae3522bbd9692629d4f9b6caa119be03f2bd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322720
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40908}
This removes internal usage of AsyncResolver, including from
PhysicalSocketServer.
PhysicalSocketServer was also run through IWYU.
Bug: webrtc:12598
Change-Id: I18aa6fb60e4a40face4afa0446a161379470680d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322721
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40900}
Its predecessor is being used directly by at least one Chromium function.
Bug: webrtc:12598
Change-Id: I0c521f03cf6664036a48d5d45dcacaa74ae8582c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/322800
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40896}
Also removing has_slots depdency from OpenSSLStreamAdapter and moving
it to the OpenSSLStreamAdapter subclass where it's still needed.
Bug: webrtc:11943
Change-Id: Ibcae5ea1efff146d78b32bb0eca63d7f44ed08c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318885
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40702}
Remove internal use of SignalSSLHandshakeError and prepare removal of
sigslot dependency from SSLStreamAdapter.
Bug: webrtc:11943
Change-Id: I9768e2e31529945620bdd8d0d285042bb2388b7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318881
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40695}
This time, hit the BUILD files too (where possible).
Bug: webrtc:11943
Change-Id: Ic8f2d77e1ba66f740efe0ef73b1ea6051356dedc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40654}
Also run iwyu on async_udp_socket.cc
Bug: webrtc:11943
Change-Id: I4ca0f468d27be08fa869fde791aec51cf0029047
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317940
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40640}
The revised version should work in more network configurations.
Submitted with no-try to unbreak the build.
No-try: true
Bug: b/297247924, webrtc:12598
Change-Id: I4b4bc586af1ec2393dc257b9cebf06fd71268131
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40614}
This should replace the wrapping async DNS resolver used
for default resolution.
Bug: webrtc:12598
Change-Id: Ic65ecd17da7a5695d0003178aeb30824a707ec78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316928
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40607}
This is part of the long term plan to stop using pointer + length
to pass around buffers.
Bug: webrtc:14870
Change-Id: Ibaf5258fd326b56132b9b5a8a6b1563a763ef2f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40512}
This is a reland of commit 86cfe50c0e
The fix was to add a backwards compatible #include + build dep.
They will be removed once Chromium is migrated.
Original change's description:
> Extract HasIPv4Enabled/HasIPv6Enabled.
>
> Bug: b/292167110
> Change-Id: Idafa4ef23e87951bdd0276c29dee3e7f8be68476
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312580
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#40478}
Bug: b/292167110
Change-Id: I9797f52adf15aba57e114d0a1efec0f757ead278
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313264
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40491}
Similar to earlier CL https://webrtc-review.googlesource.com/c/src/+/169683. While this is a rather specific use case and solution, currently these macros are using a mix of styles ever since https://webrtc-review.googlesource.com/c/src/+/184934. By switching to ::rtc we can ensure that the right namespace is used.
Bug: webrtc:11400
Change-Id: Id06f09b5224a399bd6b43bf0237422b24b5adfb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40490}
With the intent to migrate all usages of the RateStatistics and RateTracker to these two new classes and thus encourage strong types over raw ints
Bug: webrtc:13756
Change-Id: I6d98024e903e75c41b2929509f601bb32d15259d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/312460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40451}
This CL adds the Shipped field (and may update the
License File field) in Chromium READMEs. Changes were
automatically created, so if you disagree with any of
them (e.g. a package is used only for testing purposes
and is not shipped), comment the suggested change and
why.
See the LSC doc at go/lsc-chrome-metadata.
Bug: b:285450740
Change-Id: If4955c6f6e7b58e0c99469fc45ed5b9e8f30a32b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311720
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Anne Redulla <aredulla@google.com>
Cr-Commit-Position: refs/heads/main@{#40424}
To allow use more efficient move instead of copy when wrapped into std::vector
Bug: webrtc:15263
Change-Id: Ie085e3ae41fc4e49b350e430a6dea4767777bbf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40403}
This CL addresses the review comments for
https://webrtc-review.googlesource.com/c/src/+/261221
in the downstream cherry-pick: https://crrev.com/c/4660950.
* Always use size_t{} for casting.
* Remove unneeded size_t casts.
* Avoid using __x as it is reserved for the compiler.
Bug: b:217226507
Change-Id: I13c57cb69d7db066ac9a6dbd15b7f6de54abb613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311360
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Li-Yu Yu <aaronyu@google.com>
Cr-Commit-Position: refs/heads/main@{#40395}
making it return actual microseconds instead of being limited to
millisecond resolution.
This uses GetSystemTimeAsFileTime
https://learn.microsoft.com/en-us/windows/win32/api/sysinfoapi/nf-sysinfoapi-getsystemtimeasfiletime
which returns a timestamp in multiples of 100ns since January 1st 1601.
BUG=webrtc:15212
Change-Id: I293868d8f683289a9dbcf6eccb910bd9c6694e57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306440
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40360}
The logcat is spammed with warnings from timestamp aligner.
This CL make a workaround to use the timestamp aligner to get capture times,
without generating the warnings.
Bug: webrtc:14970
Change-Id: Idab4b298e0484a57841a214db9440f9ac6faaa4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296324
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#39486}
This feature has had positive impact in downstream experiments, so we should enable it by default. It will be kept around as a kill switch for a while though.
Bug: webrtc:15260
Change-Id: Ibfd25f5be124f65cd4360ae76f7022bb46f65301
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/309781
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40327}
Referencing this type directly is not allowed when building
with the macOS 14.0 SDK.
Other usages in webrtc follow this inline pattern too so
going with this instead of "auto" which also works.
Change-Id: I84a0ba9c78e83843bc73c71c642caca75750f127
Bug: chromium:1454356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40313}
This is useful in environments where OpenSSL may not be available.
Bug: webrtc:15240
Change-Id: I7ba29e44bd1d25231df13ee79dacc74f260ded67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308600
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#40293}
The flexibility offered by the GN `rtc_jsoncpp_root` should be enough
to wire a different version of jsoncpp.
Bug: b/281848049
Change-Id: I8af39fec2406e27c3af7b7ec1949a4762dce762f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304861
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40045}
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}
As pointed out in issue webrtc:15146 this Mac/iOS specific variable,
makes the SequenceChecker behave incorrectly on those platforms.
The variable was introduced in a CL that merged the previous checker
classes, ThreadChecker and SequencedTaskChecker, but curiously neither
one of them had such a variable. So I'm not exactly sure what problem
was being solved. Hence I'm wondering if we actually need it.
Reference: https://webrtc-review.googlesource.com/c/src/+/129721
Bug: webrtc:15146
Change-Id: Ia7a9eb17b993c4f8a1e8204c658bf0b3dbdaa1e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304401
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40019}
This CL introduces a new feature enabling video packet send batches.
The feature is enabled via
PeerConnectionInterface
::RTCConfiguration
::MediaConfig
::enable_send_packet_batching.
PacketOptions have been augmented with attribute "batchable" (set for
all video packets) and attribute "last_packet_in_batch" which gives
injected AsyncPacketSockets a chance to understand when a batch begins
and ends.
When the feature is on, packets are collected in RtpSenderEgress. On
reception of OnBatchComplete from PacingController, RtpSenderEgress
sends the collected batch, setting "last_packet_in_batch" to true
in the last packet.
Bug: chromium:1439830
Change-Id: I1846b9d4a8a0efd227d617691213a2e048bdc8a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/303720
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40012}
The fixes have been default enabled, so clean up dead code.
Bug: webrtc:14334
Change-Id: I4967d5ad451ac333c54294fc14bea6c7ba1445e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301180
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#39949}
Fully launched on mobile and approved for chrome launch.
Removed tests for feature enablement with field trial.
Bug: webrtc:14334
Change-Id: I7ca7183ff618835fef8c820cfd52863e1c7fa25e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301163
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#39879}
This CL is partly a test to see if there's an impact on binary size:
- Not a big difference for binaries (decrease): -776b to -4Kb
- For libraries (libwebrtc.a) it actually increases the size: +40Kb
Secondarily this CL is basically to introduce this pattern to the
code base. In terms of LOC, this makes things slightly more compact.
From:
class Foo {
public:
Foo() {
checker_.Detach();
}
private:
SequenceChecker checker_;
};
To:
class Foo {
public:
Foo() = default;
private:
SequenceChecker checker_{SequenceChecker::kDetached};
};
Bug: none
Change-Id: I59fc34ccea10847e13455a349851ce9a0af458e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39664}
This cl per default enable the experiment WebRTC-SCM-Timestamp but
leaves the wiring in place for now to explictly allow disabling it.
Bug: webrtc:5773, webrtc:14066
Change-Id: I6118eef73384791ab4d1377e35d36435dc4fa0e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298442
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39624}
The motivation for this is to not have to implement this pattern:
foo.h:
class Foo {
public:
Foo();
private:
SequenceChecker checker_;
};
foo.cc:
Foo::Foo() {
checker_.Detach();
}
And instead be able to do this inline in the .h file:
class Foo {
public:
Foo();
private:
SequenceChecker checker_{SequenceChecker::kDetached};
};
Bug: none
Change-Id: Idd7ca82d15c2f77f3aaccf26f1943a49f4b40661
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298445
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39616}
Fuchsia's libc provides `select` and `poll` but not `epoll`.
This CL adds a `WaitPoll` method, which is modeled after `WaitSelect` but uses `poll`. The pre-existing `WaitPoll` method was renamed to `WaitPollOneDispatcher`.
TESTED="2p video call on Fuchsia. WaitPoll is faster compared to
WaitSelect, primarily because WaitSelect pessimistically calls
getsockopt(SO_ERROR) on each fd, while WaitPoll does so only on fds that
have entered an error state."
Original author: tombergan@google.com
Bug: None
Change-Id: I83cc824fca40d691fd93712c1c933ff21b3f877c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296826
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tom Bergan <tombergan@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39564}
Add a WebRTC-specific arg that can be used to control use of targets
that rely on //third_party/google_benchmarks, so the .gni in that
directory can eventually be removed.
Bug: chromium:1404759
Change-Id: I2a9422fae119ca13eb50028d962fc0a671b5fb33
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39553}
This presumably worked before because some libc++ header included it
transitively, but that's no longer the case.
Bug: chromium:1423839
Change-Id: I6ed1c3474c1bfa02084a665c0b9e249484ac50d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297420
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Hans Wennborg <hans@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39544}
The logcat is spammed with warnings from timestamp aligner.
This CL make a workaround to use the timestamp aligner to get capture times,
without generating the warnings.
Bug: webrtc:14970
Change-Id: Idab4b298e0484a57841a214db9440f9ac6faaa4d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296324
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#39486}
This CL migrates unit tests to the new TaskQueueBase interface.
Bug: chromium:1416199
Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39434}
Previously, field_trial::FindFullName would be called before
field_trial::FieldTrialsAllowedInScopeForTesting was initialized because
member variables will be initialized after the initializer list.
Bug: webrtc:14154
Change-Id: I69098259a21a55a268d5b45949c3ed02e82ae001
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295541
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39420}
This prepares TaskQueueBase sub classes to be able to migrate to
the location and traits-based API. It re-introduces a Location class
into the webrtc namespace, which is meant to be overridden by Chromium.
Bug: chromium:1416199
Change-Id: I712c7806a71b3b99b2a2bf95e555b357c21c15ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294381
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39400}
CFStringFromCString is useful, however if we have a static C string, we can use the defined macro CFSTR to create a CFString at runtime instead.
Bug: None
Change-Id: I54b3f590b3ab07858409af27b817013c98556ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293060
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Seija K. <doremylover123@gmail.com>
Cr-Commit-Position: refs/heads/main@{#39341}
This library is not expected to be used outside of WebRTC.
Bug: None
Change-Id: I35e864eccc2e4101255332372e0c27bb0c492586
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293341
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39322}
Call operators do not improve code clarity, and usage was moderate.
Bug: None
Change-Id: I8d86bd7d435ce88e99f4abee8ab95a336d47dc22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/292960
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39294}
Remove integration with socket server of the current thread
Network thread that uses PhysicalSocketServer shouldn't be allowed to do blocking calls
Other threads that use NullSocketServer do not need to process any messages while blocking
Bug: webrtc:14856
Change-Id: I56865b86e0992e60376ecefe163ff6b23911edca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291527
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39225}
This should help to debug problems with DNS resolution of turn server
name. E.g. DNS server returns fake IP address to block turn server.
Similar change to stun_port.cc were done in
https://webrtc-review.googlesource.com/c/src/+/215926
Bug: None
Change-Id: If1cc410f4cd0f89620d3678aabf05a0f1b22a393
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290992
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Yury Yarashevich <yura.yaroshevich@gmail.com>
Cr-Commit-Position: refs/heads/main@{#39152}
Implement GetProcessCpuTimeNanos and GetThreadCpuTimeNanos for Fuchsia.
This is needed for the tests call_perf_tests and
video_pc_full_stack_tests on Fuchsia.
Bug: fuchsia:115601
Change-Id: Idd10db93d4087d10896ae3fde6abbc37176f625e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290920
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sarah Pham <smpham@google.com>
Cr-Commit-Position: refs/heads/main@{#39119}
There is now only one sequence number unwrapper so this is redundant.
Bug: webrtc:13982
Change-Id: I210378e069366eb21dea0051f83d7431e4177057
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290892
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39085}
Some of the new targets are otherwise not visible by external users.
Bug: webrtc:9838
Change-Id: I7b92803692de64e0f93c9ec582c9fe615fda5e65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290844
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39068}
Some of the new targets are otherwise not visible by external users.
Bug: webrtc:9838
Change-Id: Idc585f45eeeb937802d5a898be57cf3d887fe142
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290730
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39059}
This target belongs only to WebRTC and shouldn't be defined there.
Bug: webrtc:9838
Change-Id: Id44573ed4649170820a5a61b68d6077784da8549
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290722
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39041}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This is in prep for the migration of all unwrappers to
SequenceNumberUnwrapper as a standard implementation.
This moves the SeqNumUnwapper to its own header and adds 2 methods to
SeqNumUnwrapper which are defined by other unwrappers:
* PeekUnwrap
* Reset
It also adds two implementations for RtpTimestamps and
RtpSequenceNumbers.
Bug: webrtc:13982
Change-Id: I5baefb2de1db92fe1bb600760bd63b71e9310eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288742
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39030}
ThreadManager, i.e. list of all thread is only used for debugging
purpose now, such uses do not enter the loop recursively
Bug: webrtc:11567
Change-Id: I7deb86e0245950c92fe27485285fd119baf6707a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/289461
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38986}
UnitBase own function to divide with rounding overflows when dividend is close to the max int64_t
Bug: b/262999013
Change-Id: I5b0c3f4408165a0f03690cab80bd098e506fc984
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288521
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38943}
This tests all of the known integer unwrappers in WebRTC
* rtc::TimestampWrapAroundHandler
* webrtc::Unwrapper<T>
* webrtc::SequenceNumberUnwrapper<T>
* dcsctp::UnwrappedSequenceNumber
The conformance tests show differences between in behavior between the
unwrappers.
This change also adds const accessors for the internal values of dcsctp::UnwrappedSequenceNumber.
Bug: webrtc:13982
Change-Id: Ia1b483d7ef5ceb43fa4d93013a76cd0251a58c22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287620
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38900}
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.
Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
Recently, Chromium added -fsanitize=alignment for `is_ubsan=true`,
and due to this, unaligned memory access was found in several places,
and chromium roll is currently blocked.
Modify unaligned memory access in rtc_base/byte_order.h to use memcpy.
Since libaom and libsrtp perform unaligned memory accesses, add them
to the suppression list.
Also, remove any mention of yasm from the UBSAN suppression list,
as yasm is no longer used.
Bug: chromium:1057551
Change-Id: I4961b66831750f4fa7b6de0b80b2052fe6ef27c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286200
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38806}
This new name emphasizes that the field trial keys are only allowed
within the current scope. We already have test::ScopedFieldTrials that
can be used to ensure that the global field trials string itself is
isolated.
Bug: webrtc:14705
Change-Id: I8b66bbd9c11d97985292c334d2d3496a047074a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284862
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38796}
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}
and set the underlying type to unknown for mac-based detection.
BUG=webrtc:13097,webrtc:13172
Co-authored by: amir.pirogovsky@microsoft.com
Change-Id: I177214f9eb5c2990faa65262f77b3d0ce9d5eab7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284500
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38709}
After changing base functions to a CHECK instead of an =0, these
are no longer needed.
Bug: webrtc:14632
Change-Id: If3f1a62905cf433486f4974b2153c9210d1e045b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283542
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38643}
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}
This is to ensure Epoch is the same if transport switch to TCP or another transport.
First packet received will always be timestamped with rtc::TimeMicros.
Other packet timstamps will use the kernel timestamp as an offset from the first packet timestamp.
For BWE, it is important that there is not a large time base diff if transport change.
This change is protected by the experiment WebRTC-SCM-Timestamp.
Bug: webrtc:14066
Change-Id: Iaeb49831e7019e21601bc90895ac56003a54e206
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281000
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38587}
Using ::recvmsg ensure packet timestamp can then be read directly when reading the buffer
instead of a separate system call and should also work on Ios/Mac.
The same experiment field trial flag will be "WebRTC-SCM-Timestamp/enabled/" and is also planned to be used for fixing webrtc:14066
Bug: webrtc:5773, webrtc:14066
Change-Id: I8a3749e87c686aa18fcee947472c1b602a0f63c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279280
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38585}
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}
A bug in the id being searched for inside OpenSSLStreamAdapter::SslCipherSuiteToName prevented the lookup from ever succeeding.
This resulted in this stat being unavailable when calling PeerConnection::GetStats(). To fix the problem, look for (0x03000000L | cipher_suite) which matches what the BoringSSL codepath is doing.
Bug: webrtc:14596
Change-Id: Ic36d77dbc4c2378fbde1e2f21a9f5bd735b36741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38460}
For now, the run-time check will only be enabled if the
rtc_strict_field_trials GN arg is set.
In order to allow testing with imaginary field trial keys, two test
helpers have been added. It's a bit awkward to test these since the
field trial string is already global, hence the helpers are also
modifying global state. Tests must make sure this global state is reset
between runs. Things won't be an issue anymore when [1] has removed the
global string.
[1] https://crbug.com/webrtc/10335
Bug: webrtc:14154
Change-Id: Ida44cc817079d7177325e2228cf1f1d242b799e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276269
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38447}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
This reverts commit 8cd7b0a7ba.
The assumption in AndroidNetworkMonitor that an interface name
is unique has turned out to be incorrect :( for some (weird) devices,
i.e ccmni0.
It is unclear if it is a permanent setup or a transient state.
This cl/ changes the impl. to cope with that, the last
OnNetworkConnected_n "owns" the interface name, and when
OnNetworkDisconnected_n runs, we check if we're "owner"
and maybe set a new "owner" (if we're not "owner" we do nothing).
New testcases added.
I also
1) change NetworkMonitorInterface to return a struct
with all the information that is requested with interface name
as key.
2) Change Network.cc adding (debug) assertions that network
properties can't change inside a loop (in one thread).
Original change's description:
> Revert "Reset all maps in AndroidNetworkMonitor Start()/Stop()"
>
> This reverts commit 02293096f9.
>
> Reason for revert: mysterious crashes in android_network_monitor.cc
>
> Original change's description:
> > Reset all maps in AndroidNetworkMonitor Start()/Stop()
> >
> > This cl/ fixes another race condition with the recent additions
> > to NetworkMonitorAutoDetect (getAllNetworksFromCache).
> >
> > The getAllNetworksFromCache-feature uses the by the Android team
> > preferred way of enumerating networks, i.e to register network listeners.
> >
> > Th recent fix to add IsAdapterAvailable, https://webrtc-review.googlesource.com/c/src/+/257400
> > contained a bug in that the adapter_type_by_name_ map was not
> > reset either on disconnect or Start/Stop.
> >
> > This cl/ addresses that including unit test.
> > It also de-obfuscates NetworkMonitor so that it always
> > calls NotifyOfActiveNetworkList on startMonitoring even
> > if list.size() == 0. This should not matter but makes
> > code easier to understand.
> >
> > Bug: webrtc:13741
> > Change-Id: I438b877eebf769a8b2e7292b697ef1c0a349b24f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258721
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#36530}
>
> Bug: webrtc:13741
> Change-Id: I36fbf63f658d3e8048e13959cbebfbd14df12b14
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264146
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37016}
Bug: webrtc:13741
Change-Id: Ib4eb072b775e493b564528f0be94c685b70ec20f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264421
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37056}
See a full explanation of the problem on this blog [1] post about changing
std::sort in LLVM and relative issues uncovered.
The CompareNetwork function was violating the 4th rule of "strict weak
ordering" (Transitivity of incomparability: x == y and y == z imply x == z, where x == y means x < y and y < x are both false).
[1] - https://danlark.org/2022/04/20/changing-stdsort-at-googles-scale-and-beyond/
Bug: None
Change-Id: I7e893f0a30da31403766284823f75c45c4db91c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/251681
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36711}
This is a reland of commit 7679e9bf07
Breakage in chrome/services/sharing/ (not built as part of webrtc
presubmit) should be fixed with
https://chromium-review.googlesource.com/c/chromium/src/+/3604642
Original change's description:
> Delete deprecated versions of MergeNetworkList
>
> Bug: webrtc:13869
> Change-Id: I6b888ba14ca664a1f28de2fb59b7d1343cb18bd8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259300
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36611}
Bug: webrtc:13869
Change-Id: I4cf0fe0f2310eabb2d0b32c6dec5f8aef64c7712
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259869
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36704}
