A similar function was defined in rtc_base/openssl_adapter. Moving it from net/dcsctp/common/ to rtc_base/strings/. I'm planning to use StrJoin in a video codec test (a follow-up change).
Bug: webrtc:14852
Change-Id: Ie657c03e7f9fb52c189c127af6f66ec505b512ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327322
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41166}
It's useful for other parts of WebRTC and there is no real reason why
it should be located in net/dcsctp.
Bug: None
Change-Id: Iccaed4e943e21ddaea8603182d693114b2da9f6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232606
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35055}
dcSCTP library users can set their custom
g_handover_state_transformer_for_test that can serialize and
deserialize the state. All dcSCTP handover tests call
g_handover_state_transformer_for_test. If some part of the state is
serialized incorrectly or is forgotten, high chance that it will
fail the tests.
Bug: webrtc:13154
Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35035}
Those were found when trying to build within Chromium's codebase.
Bug: webrtc:12614
Change-Id: Ic3f7a266ad4b5d816a693645e1e909fc39d513c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217220
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33896}
Adding fuzzers to the build made "gn gen --check" discover a lot
of dependency errors between various components of dcSCTP.
Bug: webrtc:12614
Change-Id: I0b2dd7321aec2624da417f413c727bd11b4743e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215003
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33705}
There are numerous identifiers and sequences in SCTP, all of them being
unsigned 16 or 32-bit integers.
* Stream identifiers
* Payload Protocol Identifier (PPID)
* Stream Sequence Numbers (SSN)
* Message Identifiers (MID)
* Fragment Sequence Numbers (FSN)
* Transmission Sequence Numbers (TSN)
The first two of these are publicly exposed in the API, and the
remaining ones are never exposed to the client and are all part of SCTP
protocol.
Then there are some more not as common sequence numbers, and some
booleans. Not all will be in internal_types.h - it depends on if they
can be scoped to a specific component instead. And not all types will
likely become strong types.
The unwrapped sequence numbers have been renamed to not cause conflicts
and the current UnwrappedSequenceNumber class doesn't support wrapping
strongly typed integers as it can't reach into the type of the
underlying integer. That's something to explore later.
Bug: webrtc:12614
Change-Id: I4e0016be26d5d4826783d6e0962044f56cbfa97d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213422
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33620}
These are quite generic utilities that are used by multiple modules
within dcSCTP. Some would be good to have in rtc_base and are simple
replicas of utilities available in abseil.
Bug: webrtc:12614
Change-Id: I9914286ced7317a34628a71697da9149d6d19d38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/213190
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33609}
