This reverts commit 8a18e5b3c9.
Reason for revert: Removing the problematic DCHECK.
Original change's description:
> Revert "Remove AudioReceiveStream::Reconfigure() method."
>
> This reverts commit e2561e17e2.
>
> Reason for revert: Speculative revert: breaks an downstream project
>
> Original change's description:
> > Remove AudioReceiveStream::Reconfigure() method.
> >
> > Instead, adding specific setters that are needed at runtime:
> > * SetDepacketizerToDecoderFrameTransformer
> > * SetDecoderMap
> > * SetUseTransportCcAndNackHistory
> >
> > The whole config struct is big and much of the state it holds, needs to
> > be considered const. For that reason the Reconfigure() method is too
> > broad of an interface since it overwrites the whole config struct
> > and doesn't actually handle all the potential config changes that might
> > occur when the config changes.
> >
> > Bug: webrtc:11993
> > Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#34252}
>
> TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
>
> Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:11993
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Andrey Logvin <landrey@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34253}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:11993
Change-Id: I0d3bf9abdcdc8d3f9259d014e6074a5e6b6cc73c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221747
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34255}
This reverts commit e2561e17e2.
Reason for revert: Speculative revert: breaks an downstream project
Original change's description:
> Remove AudioReceiveStream::Reconfigure() method.
>
> Instead, adding specific setters that are needed at runtime:
> * SetDepacketizerToDecoderFrameTransformer
> * SetDecoderMap
> * SetUseTransportCcAndNackHistory
>
> The whole config struct is big and much of the state it holds, needs to
> be considered const. For that reason the Reconfigure() method is too
> broad of an interface since it overwrites the whole config struct
> and doesn't actually handle all the potential config changes that might
> occur when the config changes.
>
> Bug: webrtc:11993
> Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#34252}
TBR=saza@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com
Change-Id: I15ca2d8ee5fd612e13dc1f4b3bfb9c885c21dc66
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221746
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34253}
Instead, adding specific setters that are needed at runtime:
* SetDepacketizerToDecoderFrameTransformer
* SetDecoderMap
* SetUseTransportCcAndNackHistory
The whole config struct is big and much of the state it holds, needs to
be considered const. For that reason the Reconfigure() method is too
broad of an interface since it overwrites the whole config struct
and doesn't actually handle all the potential config changes that might
occur when the config changes.
Bug: webrtc:11993
Change-Id: Ia5311978f56b2e136781467e44f0d18039f0bb2d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221363
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34252}
* Make VideoSendStream and VideoSendStreamImpl construction non-blocking.
* Move ownership of the rtp video sender to VideoSendStream.
* Most state is constructed in initializer lists.
* More state is now const (including VideoSendStreamImpl ptr)
* Adding thread checks to classes that appear to have had a race before
E.g. RtpTransportControllerSend. The change in threading now actually
fixes an issue we weren't aware of.
* Moved from using weak_ptr to safety flag and made some PostTask calls
cancellable that could potentially have been problematic. Initalizing
the flag without thread synchronization is also simpler.
This should speed up renegotiation significantly when there are
multiple channels. A follow-up change will improve SetSend as well
which is another costly step during renegotiation.
Bug: webrtc:12840
Change-Id: If4b28da5a085643ce132c7cfcf80a62cd1a625c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/221105
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34224}
Call send statistic updates are initiated on the send
transport sequence which forced calls to PostTask to the
worker thread which keeps several related attributes
protected by it.
Change this by:
* Using std::atomics for three attributes where synchronization
doesn't really matter and which can be accessed on either
context.
* Introducing a thread-compatible internal class which keeps
the statistics protected by the send transport sequence, and
emits UMA statistics on destruction.
The change also achieves the following trivial changes:
* The call origin time is now tracked by a proper
webrtc::Timestamp.
* The explicit use of the |send_transport_queue_| was replaced by
a more relaxed sequence checker.
Bug: webrtc:11993
Change-Id: I428a4d98b5fd2fd31222f62e597a9d61a3d4899f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220931
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34187}
This way we can have custom implementation of RtpTransportControllerSendInterface and pass it properly to Call.
Call relies on RtpTransportControllerSendInterface already so this is natural way to customize RTP related classes.
If there is custom factory present in dependencies it will be used, otherwise default factory will be used.
Intention behind this change is to have ability to have custom QoS with custom parameters.
Bug: webrtc:12778
Change-Id: I5b88957025621ef4bcd63eaa98c218ad213da9c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217769
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/master@{#34181}
This is in preparation for actually doing this initialization
differently in the Call class. This CL takes the registration
steps that are inherently network thread associated and makes
them separate from the ctor/dtor.
Inject Call* instead of worker_thread(), which will simplify upcoming
work that needs to access the network_thread() as well.
This is related to:
https://webrtc-review.googlesource.com/c/src/+/220608https://webrtc-review.googlesource.com/c/src/+/220609
Bug: webrtc:11993
Change-Id: I72769fd61de84967d9a645750c40d01660a2716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220764
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34172}
This is in preparation for actually doing this initialization
differently in the Call class. This CL takes the registration
steps that are inherently network thread associated and makes
them separate from the ctor/dtor.
Bug: webrtc:11993
Change-Id: Ice2e16c108e0c302157534a3aa2b46738aaa7a93
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220608
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34163}
This is part of moving the thread hops from the network thread to
worker (required by Call) into Call itself so that we can eventually
remove them.
Bug: webrtc:11993
Change-Id: Ib3ccdd6c75a3848daae2e3ce6c9a55d9617c2f50
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220604
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34160}
This change collects the receive stats that Call
maintains into a new thread-compatible
internal class which can easily be switched to
the network thread.
Bug: webrtc:11993
Change-Id: I9fa9a7f057149789aa327e5ba8a8cb3379762272
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220760
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34158}
This is currently unused and since we ultimately don't want the delivery
of packets to be async at this stage (but rather stay on the network
thread), we don't need it.
Bug: webrtc:11993
Change-Id: I6809026b6901c8ecfacd961e98ddf79aaa16d0bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220601
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34152}
This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.
Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
Previously this value was rounded up to a millisecond value.
This change is complementary to another change:
https://webrtc-review.googlesource.com/c/src/+/216398
Bug: webrtc:12722
Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33928}
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}
* Adds a OnPacketSent callback to MediaChannel, which matches with
MediaChannel::NetworkInterface::SendPacket.
* Moves the OnPacketSent handling to the media channel implementations
(video/voice) and removes the PeerConnection/SdpOfferAnswerHandler
layer from the call path.
* Call::OnSentPacket is called directly from the channels on the network
thread. This eliminates a PostTask to the worker thread for every
audio/video network packet.
* Remove sigslot dependency from MediaChannel (and derived).
Bug: webrtc:11993
Change-Id: I1f79a7aa60f05d47e1882f9be1c9323ea8fac5f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215403
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33777}
Classes associated with the Call instance, need access to these threads
and/or awareness, for checking for thread correctness.
Bug: webrtc:11993
Change-Id: I93bcee0657875f211be2ec959b96f818fa9fd8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215584
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33772}
This is for better compatibility with thread annotations and how the
histogram data is collected. In the dtor we can make assumptions about
the state of the object, but that context is lost in member methods
even though they're only called from the destructor (and therefore
thread annotations can't "know" that the object is being destructed
inside those calls).
Bug: webrtc:11993
Change-Id: I8b698cc3340fb0db49430da6f7a9b9a02cabf0c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/208200
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33295}
To avoid unnecessary repeating tasks, CallStats' timer is started only
upon Call::EnsureStarted().
Bug: chromium:1152887
Change-Id: I1015315f42127bf510affc3d22c930b20eac8bba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/206880
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33232}
This is a reland of d48a2b14e7
The diff of the reland (what caused the tsan error) can be seen
by diffing patch sets 2 and 3. Essentially I missed keeping the calls
to the transport controller on the worker thread. Note to self to add
thread/sequence checks to that code so that we won't have to rely on
tsan :)
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
Bug: webrtc:11993, webrtc:12430
Change-Id: I4fccaa418d22c2087a55bbb3ddbb25fac3b4dfcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33153}
This reverts commit d48a2b14e7.
Reason for revert: TSan tests started to fail constantly after this CL (it looks like it is flaky and the CQ was lucky to get green). See https://ci.chromium.org/ui/p/webrtc/builders/ci/Linux%20Tsan%20v2/25042/overview.
Original change's description:
> Prepare to avoid hops to worker for network events.
>
> This moves the thread hop for network events, from BaseChannel and
> into Call. The reason for this is to move the control over those hops
> (including DeliverPacket[Async]) into the same class where the state
> is held that is affected by those hops. Once that's done, we can start
> moving the relevant network state over to the network thread and
> eventually remove the hops.
>
> I'm also adding several TODOs for tracking future steps and give
> developers a heads up.
>
> Bug: webrtc:11993
> Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33138}
TBR=nisse@webrtc.org,tommi@webrtc.org
Change-Id: Id87cf9cbcc8ed58e74d755a110f0ef9dd980e298
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11993
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205525
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33145}
This moves the thread hop for network events, from BaseChannel and
into Call. The reason for this is to move the control over those hops
(including DeliverPacket[Async]) into the same class where the state
is held that is affected by those hops. Once that's done, we can start
moving the relevant network state over to the network thread and
eventually remove the hops.
I'm also adding several TODOs for tracking future steps and give
developers a heads up.
Bug: webrtc:11993
Change-Id: Ice7ee3b5b6893532df52039324293979196d341d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204800
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33138}
This will allow for transitioning PacketReceiver callbacks and
network related callbacks from being posted over to the worker thread
and instead can stay on the network thread along with related state.
Bug: webrtc:11993
Change-Id: I38df462d4dee064015c490f2b8f809cb47f23cf1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202039
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33116}
This change includes a basic first implementation of the method in Call.
The implementation assumes being called on the network thread
and simply posts to the worker thread which is what currently happens
inside the BaseChannel class (but I'm that moving out of there).
Bug: webrtc:11993
Change-Id: Id94934808f655071ea9dc87ab20d2a4d8ca7e61e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202255
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33033}
This CL adds a string to the resulting WebRTC library (trying to make
sure the version string will be there no matter how WebRTC is packaged).
This CL should be followed by some process to regularly and
automatically update the version string.
No-Try: True
No-Presubmit: True
Bug: webrtc:12159
Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32825}
This is a reland of f5e261aaf6
This CL disables RTC_NO_UNIQUE_ADDRESS on MSan builds since
there have been some issues.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
Bug: webrtc:11495, webrtc:12218
Change-Id: I4e6c7cc37d3daffad2407c9a2acfa897fa5b426a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189968
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32668}
This reverts commit f5e261aaf6.
Reason for revert: Breaks downstream projects.
Original change's description:
> Introduce RTC_NO_UNIQUE_ADDRESS.
>
> This macro introduces the possibility to suggest the compiler that a
> data member doesn't need an address different from other non static
> data members.
>
> The usage of a macro is to maintain portability since at the moment
> the attribute [[no_unique_address]] is only supported by clang
> with at least -std=c++11 but it should be supported by all the
> compilers starting from C++20.
>
> Bug: webrtc:11495
> Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32246}
TBR=mbonadei@webrtc.org,kwiberg@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11495
Change-Id: Ice318d1b11ca3dff09c190187a0b0a32ca945fe3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32335}
This macro introduces the possibility to suggest the compiler that a
data member doesn't need an address different from other non static
data members.
The usage of a macro is to maintain portability since at the moment
the attribute [[no_unique_address]] is only supported by clang
with at least -std=c++11 but it should be supported by all the
compilers starting from C++20.
Bug: webrtc:11495
Change-Id: I9f12b67b4422a2749649eaa6b004a67d5fd572d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173331
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32246}
This is a reland of
https://webrtc-review.googlesource.com/c/src/+/174261
Patchset 1 contains the old cl (plus a merge conflict fix).
Later patchets are bufixes: A PeerConnection can be created without a
Call instance (in the case of DataChannel only), so we can't always
use that to fetch the current trials.
Old CL descritpion:
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
// Since re-land is otherwise unchanged, setting previous reviewers as TBR
TBR=kthelgason@webrtc.org,mbonadei@webrtc.org,stefan@webrtc.org,srte@webrtc.org
Bug: webrtc:11926
Change-Id: I57a9e8c3454f226f77fb93215bcac83da65034b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185003
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32163}
This replaces field_trial:: -based functions from system_wrappers.
Field trials are still used as fallback, but injectable trials are now
possible.
Bug: webrtc:11926
Change-Id: I70f28c4fbabf6d9e55052342000e38612b46682c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174261
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32129}
This avoids the pacer thread waking up at 5ms interval if a
PeerConnection is created without actually using media.
The TaskQueuePacedSender solves the problem too, this CL is mostly a
safeguard in case we still find issues when turning it on...
Can be turned off by setting field trial "WebRTC-LazyPacerStart" to
"Disabled".
Bug: webrtc:10809
Change-Id: I8501106e608eccb14487576f24bdceaf3f324d80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183982
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32101}
Because a single Resource only has a single ResourceListener, injected
Resources only gets wired up to the stream's ResourceAdaptationProcessor
that was last to call SetResourceListener. This could potentially lead
to the relevant stream not adapting based on the injected resource
because it got wired up to the wrong stream's processor.
This CL fixes this issue by introducing BroadcastResourceListener. By
listening to 1 resource (the injected one), it can spawn N "adapter"
resources that mirror's the injected resource's usage signal, allowing
all ResourceAdaptationProcessor's to react to the signal.
This is wired up in Call, and tests are updated to verify the signal
gets through.
Bug: chromium:1101263, webrtc:11720
Change-Id: I8a37284cb9a68f08ca1bdb1ee050b7144c451297
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178386
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31612}
This CL adds AddAdaptationResource to Call and
AddAdaptationResource/GetAdaptationResources method to relevant
VideoSendStream and VideoStreamEncoder interfaces and implementations.
Unittests are added to ensure that resources can be added to the Call
both before and after the creation of a VideoSendStream and that the
resources always gets added to the streams.
In a follow-up CL, we will continue to plumb the resources all the way
to PeerConnectionInterface, and an integration test will then be added
to ensure that injected resources are capable of triggering adaptation.
Bug: webrtc:11525
Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31499}
Just use the task queue it represents instead.
Also rename the accessor to not confuse it with the 'network thread'.
Bug: none
Change-Id: Ic6c61652768ca3ff60dc0a2acc4850350feeb98e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176226
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31375}
...from the Call class.
With minimal shifting of ownership of a few variables, we can
maintain their state fully on the worker thread, where they're used.
This change also removes the dependency on RWLockWrapper.
Bug: webrtc:11612
Change-Id: Ia14be5bd6b50bd0b32d04f078b1e283080c00a19
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176122
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31360}
This lock isn't needed anymore which has been clarified with recent
refactoring. Updated thread guards and related comments.
Change-Id: Ia7a1e59c45b67597264e73158654e4dffe6c4fc5
Bug: webrtc:11610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176120
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31351}
This reduces the number of threads allocated per PeerConnection when
more than one PC is needed.
Bug: webrtc:11598
Change-Id: I3c1fd71705f90c4b4bbb1bc3f0f659c94016e69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31347}
I named the network sequence checker 'worker' some time ago.
It represents the network TQ in RtpTransportControllerSendInterface
though, so should rather be called 'network'.
Bug: none
Change-Id: If82d7528b8cfcc180e1515b2fad63ed01610ec32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175340
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31291}
Call is instantiated on what we traditionally call the 'worker thread'
in PeerConnection terms. Call statistics are however gathered, processed
and reported in a number of different ways, which results in a lot of
locking, which is also unpredictable due to the those actions themselves
contending with other parts of the system.
Designating the worker thread as the general owner of the stats, helps
us keeps things regular and avoids loading unrelated task queues/threads
with reporting things like histograms or locking up due to a call to
GetStats().
This is a reland of remaining changes from https://webrtc-review.googlesource.com/c/src/+/172847:
This applies the changes from the above CL to the forked files and
switches call.cc over to using the forked implementation.
Bug: webrtc:11489
Change-Id: I93ad560500806ddd0e6df1448b1bcf5a1aae7583
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31186}
from LS_INFO to LS_VERBOSE.
By default, unit tests run with logging at info level.
A random run today produced more than 70.000 lines of
output. This CL would reduce that by approximately 15.000.
Bug: none
Change-Id: Ie62708cebf109510a2443aa5ab5c4e645ffc6707
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161950
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30077}
This check just makes it more clear what the expectations are.
Pululating trials was made mandatory in an earlier CL, but if you don't
populate this field it will trigger a DCHECK at lower layer where we're
actually trying to parse an experiment. That is confusing and
misleading.
Bug: None
Change-Id: I1f520841a5a3b911048c8ee6d309eb7bb179e037
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161301
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30005}
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.
Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
Injecting both a custom NetEqFactory and an AudioDecoderFactory is not
supported, in that case the AudioDecoderFactory should be wrapped inside
the NetEqFactory.
Bug: webrtc:11005
Change-Id: I4e311eb1bfa03c91bca587d70540e81829f881c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158720
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29673}
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.
Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
This flag became unused in https://codereview.webrtc.org/2789843002;
it was set, but the setting had no effect.
Bug: webrtc:7135
Change-Id: I012a7c3600bc7a371c7a589695823b30ed5647a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29192}
These methods were defined, and called, but not doing anything.
Bug: None
Change-Id: I9955843a6bd86e4a583b0213ddb6b3b42e2ab815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150792
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29020}
There are a few reasons for making this test only:
* The code is only used by tests and utilities.
* The pure interface has only a single implementation so an interface isn't really needed.
(a followup change could remove it altogether)
* The implementation always incorporates locking regardless of how the class gets used.
See e.g. previous use in the Packet class.
* The implementation is a layer on top of RtpUtility::RtpHeaderParser which is
sufficient for most production cases.
Change-Id: Ide6d50567cf8ae5127a2eb04cceeb10cf317ec36
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150658
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29010}
Deprecated the field BitrateAllocationUpdate::link_capacity since it is only
used by the Opus codec in order to smooth the target bitrate, which is
equivalent to the stable_target_bitrate field.
The unused field trial WebRTC-Bwe-StableBandwidthEstimate is also removed.
Bug: webrtc:10126
Change-Id: Ic4a8a9ca4202136d011b91dc23c3a27cfd00d975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149839
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28941}
This only fixes the name string you get when you query the threads, the
functionality is not changes.
Bug: None
Change-Id: I29408cf38e0e41faa127a70a010d37a980bb24ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/149167
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28875}
I've updated all the tests that previously were calling this method on
the wrong thread, so we can enable this check now.
I've also landed some changes that simplify the threading model in this
class and subsequently I've removed some locks and can remove some more
in this CL.
Added some comments about future improvements for GetStats() to reduce
synchronization.
Simplified CallStats::OnRttUpdate() to have one fewer async methods.
Bug: webrtc:10847
Change-Id: I48e6809172142cc4be4385b7d4aa2affb52a963a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148588
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28821}
There aren't any tests for this and the code isn't currently
active except for the fact that it adds complexity to the Call
class, synchronization into the active code path and makes future
improvements to the class more complex or impossible.
Bug: webrtc:9719
Change-Id: Ia41af0b2186b8a36ca70a07858990b6af7f3a5c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148078
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28807}
The PacedSender is being reworked and will need an interface so we can
inject different implementations of it.
This CL introduces a new RtpPacketPacer interface inside the pacing
module. This interface handles the details of _how_ packets should be
paced, such as pacing rates/account for audio/max queue length etc.
The RtpPacketSender interface exposed from the rtp_rtcp module handles
only the actual sending of packets.
Some minor cleanups are included here.
Bug: webrtc:10809
Change-Id: I150b1a6262306d99e3f9d5f0b4afdb16a50e5ad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145212
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28699}
This interface is intended to only handle packet-sending parts of the
paced sender.
See https://webrtc-review.googlesource.com/c/src/+/145212 for context
Bug: webrtc:10809
Change-Id: I93f0b40e1865665c2d436db67021350a0ed0687b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/145216
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28662}
This removes PeerConnectionInterface::SetBitrateAllocationStrategy()
plus a ton of now-dead code.
Bug: webrtc:10556
Change-Id: Icfae3bdd011588552934d9db4df16000847db7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133169
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28523}
This moves the conversion from RtpPacketReceived to ReceivedPacket to
Call rather than RtpTransportController. This prepares for reusing the
struct for receive side network state estimation.
Bug: webrtc:10742
Change-Id: I9581438bc912ef4bb635a5d9a6dea488cf871d48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141872
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28284}
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.
TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.
Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
Add base class NetworkPredictor and NetworkPredictorFactory in /api, make it possible to inject customized NetworkPredictor in PeerConnectionFactory level. The NetworkPredictor object will be pass down to GoogCCNetworkControl and DelayBasedBwe.
Bug: webrtc:10492
Change-Id: Iceeadbe1c9388b11ce4ac01ee56554cb0bf64d04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130201
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27543}
A reasonable amount of incoming packets could generate feedback
for millions of packets.
Bug: chromium:949020
Change-Id: I7f3e6b75b683af5b2732c472cc92c6788540486b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131333
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27481}
This change introduces new logic to allow the injection of the FrameDecryptor
into an arbitrary already running VideoReceiveStream without resetting it. It
does this by taking advantage of the BufferedFrameDecryptor which will
forcefully be created regardless of whether a FrameDecryptor is passed in
during construction of the VideoReceiver if the
crypto_option.require_frame_encryption is true. By allowing the
BufferedFrameDecryptor to swap out which FrameDecryptor it uses this allows the
Receiver to switch decryptors without resetting the stream.
This is intended to mostly be used when you set your FrameDecryptor at a point
post creation for the first time.
Bug: webrtc:10416
Change-Id: If656b2acc447e2e77537cfa394729e5c3a8b660a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130361
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27458}
to decouple it from other optional parameters
and with plan to make it mandatory
Bug: webrtc:10284
Change-Id: I71c1d3d9eaf09d00b99b0bc4c811ab173ea5f01f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130473
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27385}
This reverts commit 90705cbc41.
Reason for revert: failed to compile due to conflict with another recent change
Original change's description:
> Move TaskQueueFactory from Call::Create parameter to CallConfig
>
> to decouple it from other optional parameters
> and with plan to make it mandatory
>
> Bug: webrtc:10284
> Change-Id: I1224abd90d8e06e0ee2d2baaa6d0fd54f8caad2b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130470
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27382}
TBR=danilchap@webrtc.org,nisse@webrtc.org
Change-Id: Ibe70f191d35f72e0373e49e5300d765b88d02db0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130472
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27383}
to decouple it from other optional parameters
and with plan to make it mandatory
Bug: webrtc:10284
Change-Id: I1224abd90d8e06e0ee2d2baaa6d0fd54f8caad2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130470
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27382}
(so far SetBitrate did not do anything for media transport)
Bug: webrtc:9719
Change-Id: I48e669341ffe6c9e4697ff9146c314be7796a209
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127980
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27169}
WebRTC video engine now configures bitrate on media transport
correctly.
Bug: webrtc:9719
Change-Id: I85884cd76644b7eca3763cec8ce9e31b5b64db27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127941
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27167}
Add a feature (gated by field trial) that stores
packets with unknown ssrc in a circular buffer
and replays them once a receive stream with matching
ssrc is created.
This improves situation where media is incoming
but signaling or SetFrameDecryptor is slow.
BUG=webrtc:10405
Change-Id: I7c7b2f4bd96c942c09e96db0cdae4ce5efef2541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127543
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27159}
These are manual edits please verify there are no typos.
Feel free to auto-submit if there are no issues.
Bug: webrtc:10410
Change-Id: I08ff36bd689fa7c3716c8e7017bd571cc9f09f35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127642
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27125}
Before this change the encoder tasks runs on a shared worker queue.
That makes the destruction require synchronization to avoid races.
By keeping a separate encode queue to ChannelSend, we can safely
destruct the object without worrying for left over tasks, as they
will be stopped when the task queue is destroyed.
For TaskQueue implementations using one thread per TaskQueue this
will increase the thread count by the number of AudioSendStreams,
which typically is just one.
This is partly a reland of 9b9344742b
Original change's description:
> Removes lock from ChannelSend.
>
> The lock isn't really needed as encoder_queue_is_active_ can be checked
> on the task queue to provide synchronization.
>
> There is one behavioral change due to this: We will not cancel any currently
> pending encoding tasks when we stop sending, they will be allowed to finish.
>
> Bug: webrtc:10365
> Change-Id: I2b4897dde8d49bc7ee5d2d69694616aee8aaea38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125096
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26963}
Bug: webrtc:10365
Change-Id: Iafb84e25d90ec8639359be80fad65763d08e5719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125740
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27038}
This prepares for making AudioSendStream use its own task queue. In the
future more of the functionality that depends on running on the task
queue is planned to be moved directly into RtpTransportControllerSend.
They should instead get it from the transport controller. This affects
the media transport tests which previously assumed that the transport
controller could be missing. However, this is not something that is used
in production, so this is an improvement of the tests as they will
behave more like production code.
Bug: webrtc:9883
Change-Id: Ie32f4c2f6433ec37ac16a08d531ceb690ea9c0b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126000
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27010}
There are two RTT values reported to GoogCC. They come from the same
source initially but one is calculated and smoothed in the video call stats.
However, there's not really any technical reasons why this value should
be received via the stats, this has just been maintained for legacy reasons.
Experiments shows no real difference between the modes, therefore the
stats-reported RTT is removed in this CL as a cleanup.
Bug: None
Change-Id: If1462d6c91570ffb883ecef2ba034f04a571c9b5
Reviewed-on: https://webrtc-review.googlesource.com/c/123883
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26833}
Bug: webrtc:9719
Change-Id: I90bd1d9858c259d7339420c574ad83d6fb18299c
Reviewed-on: https://webrtc-review.googlesource.com/c/118946
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26426}
By enabling this trial, we can also remove reporting of packet
feedback status from send streams that was used before.
Bug: webrtc:9718
Change-Id: I3e7c4656b0ac6592a834617e044f23a072454181
Reviewed-on: https://webrtc-review.googlesource.com/c/118281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26363}
Currently the stats callback is registered too early.
For now we ignore media transport for these callbacks (it was ignored
already), and we will introduce changes to media transport in the
future.
Bug: webrtc:9719
Bug: chromium:906998
Bug: chromium:906533
Change-Id: I24c0265d46ec2eb35743de6cd96a11d8c41fefbe
Reviewed-on: https://webrtc-review.googlesource.com/c/114904
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26062}
It never saw much use, and is blocking refactoring.
Histograms.xml-side cleanup:
https://chromium-review.googlesource.com/c/chromium/src/+/1344141
Bug: webrtc:7882
Change-Id: I112232a573fcd218dc7a51bfcdd7898847d14f18
Reviewed-on: https://webrtc-review.googlesource.com/c/111506
Commit-Queue: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25780}
The target bandwidth is a more stable target rate as it does not follow
the variation in the control signal directly. It's intended to be used to
configure the audio frame length.
Bug: webrtc:9718
Change-Id: Idcc83ba0fef90e0ead2926d18ba6893a2b0f085f
Reviewed-on: https://webrtc-review.googlesource.com/c/107729
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25718}
Add TargetRateObservers for media transport in the call object.
Bug: webrtc:9719
Change-Id: I5448d05359cf09b8cd2a678b2ac876aa8f8970e7
Reviewed-on: https://webrtc-review.googlesource.com/c/110622
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Commit-Queue: Peter Slatala <psla@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25662}
It was previously logged in Call, but streams are not always created
with the full configuration, which caused header extensions to be
missing from the log.
Bug: webrtc:9885
Change-Id: I86c0424004c4629ebab0f6b155b83fb90e15b131
Reviewed-on: https://webrtc-review.googlesource.com/c/108601
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25483}
This also moves the packet feedback tracking to RtpVideoSender.
Bug: webrtc:9517
Change-Id: Ifb1ff85051730108a0b0d1dd30f6f8595ad2af6e
Reviewed-on: https://webrtc-review.googlesource.com/c/95920
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25019}
When bitrate is allocated to streams that does not have packet feedback,
the allocated bitrate should be included in the estimate. This was
previously only implemented for the old congestion controller and not
for the new task queue based version.
To make the behavior more robust, the responsibility for tracking this
is moved to BitrateAllocator where it's handled consistently for
multiple streams without feedback.
Bug: webrtc:9586, webrtc:8243
Change-Id: I8af7fec23e1bdc08cc61cf1b4ff10461c3711fb0
Reviewed-on: https://webrtc-review.googlesource.com/102681
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24905}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I74cb86c29cebb69dd22083718f1446f18f705cd4
Reviewed-on: https://webrtc-review.googlesource.com/95883
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24483}
Replaced by a int64_t representing time in us.
Bug: webtrc:9584
Change-Id: I0505c020ef741ad940203ec300e8adb103856dda
Reviewed-on: https://webrtc-review.googlesource.com/91840
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24204}
- Move PayloadRouter to RtpTransportControllerInterface.
- Move RetransmissionLimiter inside RtpTransportControllerSend from
VideoSendStreamImpl.
- Move video RTP specifics into PayloadRouter, in particular ownership
of the RTP modules.
- PayloadRouter now contains all video specific RTP code, and will be
renamed in a follow-up to VideoRtpSender.
- Introduce VideoRtpSenderInterface.
Bug: webrtc:9517
Change-Id: I1c7b293fa6f9c320286c80533b3c584498034a38
Reviewed-on: https://webrtc-review.googlesource.com/88240
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24009}
Running clang-format with chromium's style guide.
The goal is n-fold:
* providing consistency and readability (that's what code guidelines are for)
* preventing noise with presubmit checks and git cl format
* building on the previous point: making it easier to automatically fix format issues
* you name it
Please consider using git-hyper-blame to ignore this commit.
Bug: webrtc:9340
Change-Id: I694567c4cdf8cee2860958cfe82bfaf25848bb87
Reviewed-on: https://webrtc-review.googlesource.com/81185
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23660}
This is a no-op change because rtc::Optional is an alias to absl::optional
This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+
find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;
git cl format
Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
This is a kitchen-sink header, some pieces should be moved to
byteorder.h, the rest likely deleted.
Delete most includes of basictypes.h. In leaf headers,
include stddef.h and stdint.h explicitly where needed.
Bug: webrtc:6853
Change-Id: Ibc809936a8f94d418e4eb650da1e89c1b9142073
Reviewed-on: https://webrtc-review.googlesource.com/77721
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23333}
This prepares for allowing injection of a network controller.
Bug: webrtc:9155
Change-Id: I5624f47738db9c5cd4750eac76cb6289e06a7aa3
Reviewed-on: https://webrtc-review.googlesource.com/73100
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23188}
Moving ownership of worker task queue in Call to
RtpTransportControllerSend. This CL also ensures that the task queue
is not destroyed until the process thread running
SendSideCongestionController is stopped.
The worker queue should be owned by RtpTransportControllerSend since
it is mainly used for rtp and transport related tasks such as bitrate
allocation and signaling network state.
Bug: webrtc:9232
Change-Id: I211edf1a3b9f9b2572875d5584cb788cb2449ef9
Reviewed-on: https://webrtc-review.googlesource.com/63023
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23119}
This reverts commit 04dd176862.
Reason for revert: Regression in ramp up perf tests.
Original change's description:
> Reland "Move rtp-specific config out of EncoderSettings."
>
> This is a reland of bc900cb1d1
>
> Original change's description:
> > Move rtp-specific config out of EncoderSettings.
> >
> > In VideoSendStream::Config, move payload_name and payload_type from
> > EncoderSettings to Rtp.
> >
> > EncoderSettings now contains configuration for VideoStreamEncoder only,
> > and should perhaps be renamed in a follow up cl. It's no longer
> > passed as an argument to VideoCodecInitializer::SetupCodec.
> >
> > The latter then needs a different way to know the codec type,
> > which is provided by a new codec_type member in VideoEncoderConfig.
> >
> > Bug: webrtc:8830
> > Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> > Reviewed-on: https://webrtc-review.googlesource.com/62062
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22532}
>
> Bug: webrtc:8830
> Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
> Reviewed-on: https://webrtc-review.googlesource.com/63721
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22595}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Bug: webrtc:8830,chromium:827080
Change-Id: Iaaf146de91ec5c0d741b8efdf143f7e173084fef
Reviewed-on: https://webrtc-review.googlesource.com/65520
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22677}
Reduce synchronization in the class significantly and not hold a lock
while calling out to external implementations.
* Rewrite tests to use a real ProcessThread.
* Update some code to use C++ 11 constructs & library features.
Bug: webrtc:9064
Change-Id: I240a819efb6ef8197da3f2edf7acf068d2a27e8b
Reviewed-on: https://webrtc-review.googlesource.com/64521
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22649}
This is a reland of bc900cb1d1
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
Bug: webrtc:8830
Change-Id: If88ef7d57cdaa4fae3c7b2a97ea5a6e1b833e019
Reviewed-on: https://webrtc-review.googlesource.com/63721
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22595}
This moves it from an API directory (system_wrappers/include/) to a
non-API directory, which is exactly what we want for utilities like
this.
BUG=webrtc:8445
NOPRESUBMIT=true
Change-Id: Ie2879aca5fc1667e4222499d2a8fc2bba9ae2425
Reviewed-on: https://webrtc-review.googlesource.com/21328
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22587}
This CL adds reporting of per packet feedback availability from Call
via RtpTransportControllerSend to SendSideCongestionController.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I20b3dbb4a027c46476bc2d2bc875374bff05609a
Reviewed-on: https://webrtc-review.googlesource.com/63220
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22566}
The receive time calculator combines the packet time stamps received
from the socket interface with the system clock used in WebRTC. This
means that the packet timestamps are set in the WebRTC clock timebase
and that large jumps in the time stamps from the socket will not affect
the reported receive time stamps.
Bug: None
Change-Id: I293925c41919829524a115bb9377027bf0a797fb
Reviewed-on: https://webrtc-review.googlesource.com/61862
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22540}
This reverts commit bc900cb1d1.
Reason for revert: Broke downstream projects.
Original change's description:
> Move rtp-specific config out of EncoderSettings.
>
> In VideoSendStream::Config, move payload_name and payload_type from
> EncoderSettings to Rtp.
>
> EncoderSettings now contains configuration for VideoStreamEncoder only,
> and should perhaps be renamed in a follow up cl. It's no longer
> passed as an argument to VideoCodecInitializer::SetupCodec.
>
> The latter then needs a different way to know the codec type,
> which is provided by a new codec_type member in VideoEncoderConfig.
>
> Bug: webrtc:8830
> Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
> Reviewed-on: https://webrtc-review.googlesource.com/62062
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22532}
TBR=brandtr@webrtc.org,magjed@webrtc.org,nisse@webrtc.org,stefan@webrtc.org
Change-Id: I01f06c1fcf21eb2cd40dca7d4f268614200ee490
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8830
Reviewed-on: https://webrtc-review.googlesource.com/63720
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22537}
This CL adds a boolean indicating availability of per packet feedback
to the OnAllocationLimitsChanged callback on the
BitrateAllocator::LimitObserver interface.
This is part of a series of CLs tracking the transport feedback status
of the streams known to BitrateAllocator and reporting the status to
the congestion controller.
Bug: webrtc:8415
Change-Id: I5bd6e5796733da312556f2f681ff06d49ea2becc
Reviewed-on: https://webrtc-review.googlesource.com/63201
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22533}
In VideoSendStream::Config, move payload_name and payload_type from
EncoderSettings to Rtp.
EncoderSettings now contains configuration for VideoStreamEncoder only,
and should perhaps be renamed in a follow up cl. It's no longer
passed as an argument to VideoCodecInitializer::SetupCodec.
The latter then needs a different way to know the codec type,
which is provided by a new codec_type member in VideoEncoderConfig.
Bug: webrtc:8830
Change-Id: Ifcc691aef1ee6a95e43c0452c5e630d92a511cd6
Reviewed-on: https://webrtc-review.googlesource.com/62062
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22532}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
TBR=stefan@webrtc.org, philipel@webrtc.org
Originally reviewed on: https://webrtc-review.googlesource.com/33013
Bug: webrtc:8910
Change-Id: I162dde5fa20a260b41e5187fcf30b49f5e6fb0e0
Reviewed-on: https://webrtc-review.googlesource.com/61782
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22430}
This reverts commit 31a12c557d.
Reason for revert: Breaks downstream project.
Original change's description:
> Add ability to emulate degraded network in Call via field trial
>
> This is especially useful in Chrome, allowing use to emulate network
> conditions in incoming or outgoing media without the need for platform
> specific tools or hacks. It also doesn't interfere with the rest of the
> network traffic.
>
> Also includes some refactorings.
>
> Bug: webrtc:8910
> Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
> Reviewed-on: https://webrtc-review.googlesource.com/33013
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22418}
TBR=sprang@webrtc.org,stefan@webrtc.org,philipel@webrtc.org
Change-Id: I22bda6da01c2ff5abd6f408c5ee9e4fba21294f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8910
Reviewed-on: https://webrtc-review.googlesource.com/61700
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22419}
This is especially useful in Chrome, allowing use to emulate network
conditions in incoming or outgoing media without the need for platform
specific tools or hacks. It also doesn't interfere with the rest of the
network traffic.
Also includes some refactorings.
Bug: webrtc:8910
Change-Id: I2656a2d4218acbe7f8ffd669de19a02275735438
Reviewed-on: https://webrtc-review.googlesource.com/33013
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22418}
Removing the Synchronous call AvailableBandwidth from the
RtpTransportControllerSend interface. The bandwidth estimate is
provided trough a new interface that communicates with a struct
making it easier to add parameters in the future.
This prepares for removing locking behavior in
SendSideCongestionController that exists just to support this feature.
To keep backwards compatibility with the old
SendSideCongestionController, the struct TargetTransferRate
is constructed in RtpTransportControllerSend. This step can be
removed in the future when the old SendSideCongestionController
is deprecated.
Bug: webrtc:8415
Change-Id: I06f64a89848157de412901c989650d1ecf35246b
Reviewed-on: https://webrtc-review.googlesource.com/60800
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22387}
SSCC was accessing the pacer just to report values back to
RtpTransportControllerSend which already owns the pacer.
This CL moves those access methods.
To make RtpTransportControllerSend simpler, Call is made
responsible to keep track of network status used only as a
condition for report the pacer queuing delay.
Bug: webrtc:8415
Change-Id: I306bc9fcd3d8dcc7a637d51f2629ececebd48cad
Reviewed-on: https://webrtc-review.googlesource.com/60483
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22331}
Ownership of the retransmission rate limiter for video is moved
from send side congestion controller to Call. This is to reduce the
interface on the rtp transport controller send.
Bug: webrtc:8415
Change-Id: Ie9c7317400a9eb61a3c8325b9e527844ffc13769
Reviewed-on: https://webrtc-review.googlesource.com/58745
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22254}
Injecting the retransmission rate limiter used in video send stream
directly rather than using the transport controller reference.
This prepares for removing ownership of the retransmission rate limiter
from the congestion controller.
Bug: webrtc:8415
Change-Id: Iee8af53e62f407ee430625008f2d2b0cabb1f369
Reviewed-on: https://webrtc-review.googlesource.com/58800
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22251}
