8 Commits

Author SHA1 Message Date
Florent Castelli
0af55ba60d Remove //rtc_base:logging from public deps
Bug: webrtc:8603
Change-Id: I2704da8618f88032adac7ae9eb2a0f47fce4a836
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257908
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36443}
2022-04-05 10:31:19 +00:00
Florent Castelli
e17d111f4a dcsctp: Remove dependency on //rtc_base
It's not used and pulls a lot of dependencies.

Bug: None
Change-Id: I8fd41b1f5793b281fddb83891d63b6e3eca5235f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257902
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: 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@{#36426}
2022-04-04 13:28:06 +00:00
Henrik Boström
b951dc6f4c Allow specifying delayed task precision of dcsctp::Timer.
Context: The timer precision of PostDelayedTask() is about to be lowered
to include up to 17 ms leeway. In order not to break use cases that
require high precision timers, PostDelayedHighPrecisionTask() will
continue to have the same precision that PostDelayedTask() has today.
webrtc::TaskQueueBase has an enum (kLow, kHigh) to decide which
precision to use when calling PostDelayedTaskWithPrecision().

See go/postdelayedtask-precision-in-webrtc for motivation and a table of
delayed task use cases in WebRTC that are "high" or "low" precision.

Most timers in DCSCTP are believed to only be needing low precision (see
table), but the delayed_ack_timer_ of DataTracker[1] is an example of a
use case that is likely to break if the timer precision is lowered (if
ACK is sent too late, retransmissions may occur). So this is considered
a high precision use case.

This CL makes it possible to specify the precision of dcsctp::Timer.
In a follow-up CL we will update delayed_ack_timer_ to kHigh precision.

[1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/net/dcsctp/rx/data_tracker.cc;l=340

Bug: webrtc:13604
Change-Id: I8eec5ce37044096978b5dd1985fbb00bc0d8fb7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249081
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35809}
2022-01-26 18:40:24 +00:00
Victor Boivie
4b7024b572 Revert "dcsctp: Use rtc::CopyOnWriteBuffer"
This reverts commit 2db59a6584eca54245794a0e657ca9ded9e6707f.

Reason for revert: Causes msan-issue in crc32c, reading uninitialized
memory.

Bug: webrtc:12943, chromium:1275559
Change-Id: I05f1012d896aeaca86c4562e0df15fa7ea326d60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35461}
2021-12-02 12:33:46 +00:00
Victor Boivie
2db59a6584 dcsctp: Use rtc::CopyOnWriteBuffer
This avoids copying the payload at all. Future CL will change the
transport.

In performance tests, memcpy was visible in the performance profiles
prior to this change.

Bug: webrtc:12943
Change-Id: I507a1a316165db748e73cf0d58c1be62cc76a2d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236346
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35428}
2021-11-29 11:53:19 +00:00
Victor Boivie
236ac50628 dcsctp: Add public API for BufferedAmountLow
This adds native support for the RTCDataChannel properties:
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/bufferedAmount
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/bufferedAmountLowThreshold

And the RTCDataChannel event:
https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel/onbufferedamountlow

The old callback, NotifyOutgoingMessageBufferEmpty, is deprecated as it
didn't work very well. It will not be triggered and will be removed
as soon as all users of it are gone. There is a new callback,
OnTotalBufferedAmountLow, that serves the same purpose but also allows
setting an arbitrary limit when it should be triggered (See
DcSctpOptions::total_buffered_amount_low_threshold).

Bug: webrtc:12794
Change-Id: Ic1c92f174eff8a1acda0b5fd3dcc45bd1cfa2704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219691
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34144}
2021-05-27 15:27:27 +00:00
Victor Boivie
92bd9020af dcsctp: Restrict fuzzing input length
Restricting the fizzing input length according to the instructions at
https://chromium.googlesource.com/chromium/src/testing/libfuzzer/+/HEAD/getting_started.md#common-tricks

Without this limit, it finds inputs that are unreasonably large (160kB+)
that just make the ASAN built fuzzer hit the default timeout of 60s.

Bug: webrtc:12614
Change-Id: I1417f22698fba8d9bd2c56f8c3d51850b8f00f54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219161
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34034}
2021-05-18 09:22:37 +00:00
Victor Boivie
a30362cc75 dcsctp: Add socket fuzzer helper
The socket fuzzer is build as a structure-aware fuzzer where the full
public API is exercised as well as receival of SCTP packets with random
sequences of valid chunks.

It begins by putting the socket in a defined starting state and then,
based on the fuzzing data, performs a sequence of operations on the
socket such as receiving packets, sending data, resetting streams or
expiring timers.

This is the first iteration, and when running it a while and analyzing
code coverage, it will be modified to perform better. It could probably
be a little more random.

Bug: webrtc:12614
Change-Id: I50d6ffaecef5722be5cf666fee2f0de7d15cc2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218500
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33998}
2021-05-12 15:06:17 +00:00