Add GUARDED_BY annotation on members claimed to be protected by the
lock, and add missing lock operations. Also mark a few members const.
Bug: webrtc:11567, webrtc:2079
Change-Id: I8f12ca7627df0c24e07fa2ae24a387c6a0ed76cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208224
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34340}
This reverts commit 06540166ca97028454adea48cec9bf109b771ddc.
Reason for revert: breaks downstream test.
Original change's description:
> Port: migrate to TaskQueue.
>
> Port uses legacy rtc::Thread message handling. In order
> to cancel callbacks it uses rtc::Thread::Clear() which uses locks and
> necessitates looping through all currently queued (unbounded) messages
> in the thread. In particular, these Clear calls are common during
> negotiation and the probability of having a lot of queued messages is
> high due to a long-running network thread function invoked on the
> network thread.
>
> Fix this by migrating Port to task queues.
>
>
> Bug: webrtc:12840, webrtc:9702
> Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34338}
TBR=hta@webrtc.org,handellm@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I014ef9267d224c10595cfa1c12899eabe0093306
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12840, webrtc:9702
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223062
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34339}
Port uses legacy rtc::Thread message handling. In order
to cancel callbacks it uses rtc::Thread::Clear() which uses locks and
necessitates looping through all currently queued (unbounded) messages
in the thread. In particular, these Clear calls are common during
negotiation and the probability of having a lot of queued messages is
high due to a long-running network thread function invoked on the
network thread.
Fix this by migrating Port to task queues.
Bug: webrtc:12840, webrtc:9702
Change-Id: I6c6fb83323899b56091f0857a1c2d15d19199002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221370
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34338}
The factory allows us to isolate the implementation from users who only
need to depend directly on the public folder now.
Bug: webrtc:12614
Change-Id: Ied09cf772ed427eaf17a7b5705f587da57405640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220939
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34330}
While in the COOKIE ECHO state, there is a TCB and there might be data
in the send buffer, and RFC4960 allows the COOKIE ECHO chunk to bundle
additional DATA chunks in the same packet, but there mustn't be more
than one such packet sent, and that packet must have a COOKIE ECHO chunk
as the first chunk in it.
When the COOKIE ACK chunk has been received, the socket is allowed to
send multiple packets.
Previously, this was state managed by the socket and not the TCB, as
the socket is responsible for moving between the different states. And
when the COOKIE ECHO chunk was sent, the TCB was instructed to only send
a single packet by the socket.
However, if there were retransmissions or anything else that could
result in calling TransmissionControlBlock::SendBufferedChunks, it would
do as instructed and send those, even if the socket was in a state where
that wasn't allowed.
When the peer was dcSCTP, this didn't cause any issues as dcSCTP tries
to be tolerant in what it receives (but strict in what it sends, except
for when there are bugs). When the peer was usrsctp, it would send an
ABORT for each received packet that didn't have a COOKIE ECHO as the
first chunk, and then restart the handshake (sending an INIT). So this
resulted in a longer handshake, but the connection would eventually be
correctly established and any DATA chunks that resulted in the ABORTs
would've been retransmitted.
By making the TCB aware of that particular state, and to make it
responsible for creating the SCTP packet with the COOKIE ECHO chunk
first, and also to only send a single packet when it is in that state,
there will not be any way to bypass this limitation.
Also, while not explicitly mentioned in the RFC, the retransmission
timer will not affect resending any outstanding DATA chunks that were
bundled together with the COOKIE ECHO chunk, as then there would be two
timers that both would drive resending COOKIE ECHO and DATA chunks.
Bug: webrtc:12880
Change-Id: I76f215a03cceab5bafe9f16eb4775f3dc68a6f05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222645
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34329}
Integrate ClippingPredictorEvaluator in AgcManagerDirect adding the
possibility to run the predictor without affecting the analog gain
adjustment process.
The evaluator is used to compute precision, recall and F1 score.
F1 score and the measured clipping prediction intervals are logged as
`WebRTC.Audio.Agc.ClippingPredictor.F1Score` and `.PredictionInterval`
histograms respectively.
Bug: webrtc:12774
Change-Id: I708dcda9321f92d5bd17ec4c36ebce1165ead57f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221921
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34327}
This is a reland of e6ee8fab7eac915b2b6abc9b71b6d33ad086f3d1
Original change's description:
> Deprecate microsecond timestamps in RTC event log.
>
> (Microsecond timestamps are only used in the legacy wire-format,
> and the clocks only have microsecond resolution on some platforms.)
>
> Also convert structs on the parsing side to use a Timestamp instead
> of a uint64_t to represent the log time.
>
> Bug: webrtc:11933
> Change-Id: Ide5a0217d99f13f2e243115b163f13e0525648c7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219467
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34097}
Bug: webrtc:11933
Change-Id: I295be966ee96b50719ceb4690dad7e7ce958dbac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221361
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34321}
This builds on a few other CLs that avoid recreating the audio receive
streams on config changes and removes redundant config state in
WebRtcAudioReceiveStream, constructs the embedded receive stream in the
initializer list and keeps it const.
Bug: webrtc:11993
Change-Id: Iad28e0170bee6bf1e08713a89af7c81435b4265e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222100
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34317}
it is not required because during construction members can be set on
wrong thread, and in some corner cases it may even cause a crash.
Bug: none
Change-Id: I37d7f2a7772b6ab5e574077d3f53bca2529f9ae1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222651
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34315}
This reverts commit 10814873c584df17e560462adcc2b72e488ada91.
Reason for revert: Breaks down stream project
Original change's description:
> Avoid video stream allocation on configuration change after timeout.
>
> This is to prevent the video stream to get in a state where it is
> allocated but there is no activity.
>
> Bug: b/189842675
> Change-Id: I0793bd4cbf2a4faed92cf811550437ae75742102
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221618
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34276}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: b/189842675
Change-Id: If930360000f5ca290100920a4f215358b8c3e644
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222652
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34314}
Add logging of DTLS handshake retransmission,
either when it happens or when it fails.
Note that is only for the handshake messages,
which are retransmitted with exponential back off.
This patch aim to help rare DTLS hanging problems.
BUG=None
Change-Id: Iae808190711dd80dd8a43ff22757dd69919d63ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222647
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34304}
The previous approach was that the caller was responsible for ensuring
that any buffer passed in to the Bounded IO wrappers, and that any
offset from where sub-readers were created were valid. The called would
always do a validation of the data and return proper error messages
if they were not.
This didn't pan out. https://crbug.com/1216758 found an overflow that
fooled the validation logic and the fuzzer could read out-of-bounds,
although it would always crash in that particular case.
There was already bounds checking, but under DCHECKs. This CL changes
that so that any bounds checking is done with CHECKS, as would've been
done in Rust. It's better to crash than to read arbitrary memory.
Bug: chromium:1216758
Change-Id: I89b52f0758495b5fe46f926c142870a263b96314
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221743
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34303}
This method has been deprecated since 2018-07:
https://webrtc-review.googlesource.com/c/src/+/88368/
It is never called by WebRTC itself.
Custom `VideoDecoderFactory` implementations overriding this method can
switch to the overload accepting a `VideoCodecInfo` object.
This is also adding a `toString()` implementation to `VideoCodecInfo`,
to make logging of the value more useful.
Bug: webrtc:7925
Change-Id: I70ec07a0cd4ffa07d165c9851e393439fcc5870b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221960
Commit-Queue: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34302}
Also applies sanitizing to prflx candidates, not just local ones.
Also add tests for the port allocator Sanitize function.
Bug: chromium:1218346
Change-Id: Ifbc7843cd6e289c09ca72b6ec610a34bbbf7e04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222581
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34292}
