An upcoming change will remove PeerConnection's involvement in
the RTCP packet path and keep the flow consistent with the RTP packet
path.
Bug: webrtc:11993
Change-Id: I7ce1bbe8bbc352a49310e2e55b55ca5d8d927935
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217389
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33927}
This was caught in an integration test which had stricter assertions
than the FakeTimeout which is used in unit tests, so the first thing was
to add the same assertions to the FakeTimeout.
The issue is that when a Timer triggers, and if it's set to
automatically restart (possibly with an exponential backoff), the
`is_running_` field was set to true while the timer callback was called
to allow the client to know that the timer is in fact running, but the
timer was actually not started until the callback returned. Which made
sense, as the callback can with its return value override the duration,
which should affect the backoff algorithm.
The problem was when a timer was manually started within the callback.
As the Timer itself thought that it was already running, it first would
Stop() the underlying Timeout, then Start(). But calling Stop() on a
timeout that is not running is illegal, which set of assertions.
So the solution is to don't lie; Don't say that a timer is running when
it's not. Make sure that the timer is running when the timer callback is
triggered, which makes it consistent at all times. That may result in
unnecessary timeout invocations (stopping and starting), but that's not
too expensive.
Bug: webrtc:12614
Change-Id: I7b4447ccd88bd43d181e158f0d29b0770c8a3fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217522
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33926}
In real life, when a Timeout expires, the caller is supposed to call
DcSctpSocket::HandleTimeout directly, as the Timeout that just expired
is stopped (it just expired), but the Timer still believes it's running.
The system is not in a consistent state.
In tests, all timeouts were evaluated at the same time, which, if two
timeouts expired at the same time, would put them both as "not running",
and with their timers believing they were running. So if you would do
any operation on a timer whose timeout had just expired, the timeout
would assert saying that "you can't stop a stopped timeout" or similar.
This isn't relevant in non-test scenarios.
Solved by expiring timeouts one by one.
Bug: webrtc:12614
Change-Id: I79d006f4d3e96854d77cec3eb0080aa23b8569cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217560
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33925}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
Also take the opportunity to declare factory_ const.
(Bug reference is where I noticed the possibility; it is unlikely
to fix the bug.)
Bug: chromium:1205343
Change-Id: I6078f170cf68d94314ee184bdfd2dc6f4ffc1e71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217385
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33918}
The channel classes have stored the negotiated headers but a more
natural place to store them is in the RtpTransceiver class where
RtpHeaderExtension state is managed as well as the implementation of
the only method that depends on the stored state,
HeaderExtensionsNegotiated().
Also adding a TODO for further improvements where we're unnecessarily
storing state in the channel classes for the purposes of the transports.
Bug: webrtc:12726
Change-Id: If36668e3e49782ddeada23ebed126ee2c4935b8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216691
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33917}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
VideoCodecInitializer::SetupCodec never sets startBitrate,
so SetAv1SvcConfig shouldn't use it.
Bug: webrtc:12720
Change-Id: I04835dc27368f32c19132d93c72364173d7050fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217382
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33915}
As INFO is aliased to LS_INFO, this didn't trigger any warnings or
compilation errors.
Bug: None
Change-Id: I1ed30c435d9ee6ea1b51d85a375d70135d3475e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216689
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33912}
Also fix similar annotation on NackModule to have effect
(when defining an alias with C++ using, ABSL_DEPRECATED should appear
on the left hand side).
Bug: webrtc:12339
Change-Id: Id80a20bf2c56a826777b8a40e06ac5c65e4f8db7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33905}
This is about doing the best with what we have. As delayed tasks can't
be cancelled, and dcSCTP timers will almost always be stopped or
restarted, and will generally only expire on packet loss.
This implementation will post a delayed task whenever a Timeout is
started. Whenever it's stopped or restarted, it will keep the scheduled
delay task running (there's no alternative), but it will also not start
a new delayed task on subsequent starts/restarts. Instead, it will wait
until the original delayed task has triggered, and will then - if the
timer is still running, which it probably isn't - post a new delayed
task with the remainder of the the duration.
There is special handling for when a shorter duration is requested, as
that can't re-use the scheduled task, but that shouldn't be very common.
Bug: webrtc:12614
Change-Id: I7f3269cabf84f80dae3b8a528243414a93d50fc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217223
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33904}
This is related to upcoming changes whereby it will be enforced that
calls to SetInterface(<valid ptr>) and SetInterface(nullptr) be matched
up correctly.
Bug: webrtc:11993
Change-Id: Ic022f9487a7ab297adaced8e620e2384e055673b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217241
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33903}
This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb.
Reason for reland:
Relanding the original change but without the modification for
VideoSendStream::GetStats. Essentially there's a TODO there to fix
the downstream issue, which seems to be benign.
Original change's description:
> Revert "Remove Invoke from VideoChannel::FillBitrateInfo."
>
> This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
>
> Reason for revert: Speculative revert (breaks downstream project).
>
> Original change's description:
> > Remove Invoke from VideoChannel::FillBitrateInfo.
> >
> > The method is relied upon by StatsCollector where it was called from the
> > signaling thread in a loop. Now there's at most one invoke (not N).
> >
> > Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> > VideoSendStream. Updating all related tests that fetched stats from
> > the wrong context.
> >
> > Bug: webrtc:12726
> > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33894}
>
> TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:12726
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33898}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12726
Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33902}
This should avoid the situation where WebRTC's GN check is green and
Chromium (which turns it ON for //third_party/webrtc) fails.
Bug: webrtc:12614
Change-Id: Id4c06ac57e9faa07c5e43491a61fbc093c68a40d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217221
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33900}
The Invoke() isn't necessary as the method is always called on the
correct thread (now enforced with a DCHECK).
Bug: webrtc:12726
Change-Id: I53bbdabc5d6de1316e7cf478d8912e19dd0e45e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216690
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33899}
This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14.
Reason for revert: Speculative revert (breaks downstream project).
Original change's description:
> Remove Invoke from VideoChannel::FillBitrateInfo.
>
> The method is relied upon by StatsCollector where it was called from the
> signaling thread in a loop. Now there's at most one invoke (not N).
>
> Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
> VideoSendStream. Updating all related tests that fetched stats from
> the wrong context.
>
> Bug: webrtc:12726
> Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33894}
TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I2520957cdb33492d187f04320c7416788fd0f820
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33898}
libstdc++ does not allow incomplete type for T2 with std::pair<T1,T2>,
which is used by std::unordered_map. Include full definition of
TmmbrInformation, RrtrInformation and LastFirStatus.
Bug: chromium:957519
Change-Id: I00cad6d5e5a782791f5f64b4e38d7738b2c5ae87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217180
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33897}
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}
New tests (transport sequence number plus newly added abs send time) now
test more of production code and less of rtp_sender_unittest.cc test
fixture code.
Bug: webrtc:11340
Change-Id: I8ec0022c3d18467a4144ce984996af1a452760dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216327
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33895}
The method is relied upon by StatsCollector where it was called from the
signaling thread in a loop. Now there's at most one invoke (not N).
Uncommenting thread checks and removing TODOs in SendStatisticsProxy,
VideoSendStream. Updating all related tests that fetched stats from
the wrong context.
Bug: webrtc:12726
Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33894}
This completes the basic implementation of the dcSCTP library. There
are a few remaining commits to e.g. add compatibility tests and
benchmarks, as well as more support for e.g. RFC8260, but those are not
strictly vital for evaluation of the library.
The Socket contains the connection establishment and teardown sequences
as well as the general chunk dispatcher.
Bug: webrtc:12614
Change-Id: I313b6c8f4accc144e3bb88ddba22269ebb8eb3cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214342
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33890}
We don't support audio simulcast, so we should reject the layers
early during an addTransceiver() call.
Bug: webrtc:12719
Change-Id: Ieeb92c66de741e9b11943e0173a6f2e052926f13
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216685
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33886}
