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 86cfe50c0e3549544ca4a7ec097feac44f0e8437
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}
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}
