This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation.
This poison extends and thus replaces default task queue poison.
Bug: webrtc:15656
Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41247}
Add test helper to inject fake media engine for those tests.
Bug: webrtc:15574
Change-Id: Iae4282d2d3b9804548ccadf58797f39508f07c6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325880
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41075}
Update most of the webrtc tests to use EnableMediaWithDefaults instead of SetMediaEngineDefaults
Bug: webrtc:15574
Change-Id: I489a09e4ea3479dc26829ee0c1235e67bcbca7c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325485
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41059}
instead of requiring to pass in call_factory and media_engine
webrtc users should set media_factory member and media dependencies into PeerConnectionFactoryDependencies
Bug: webrtc:15574
Change-Id: I2dc584fe7afa41c9f170bdc51533396155cdcb06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41049}
Speculative fix. Writing the test for it is more complex.
Bug: chromium:1483874
Change-Id: Icfaf1524b0499c609010753e0b6f3cadbd0e98f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321480
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40820}
making the UsedId generator the source of truth.
BUG=webrtc:12197
Change-Id: I4318a1366f8b2e20ea5ae264232437a9006c5103
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/321120
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40802}
which is not a generic SDP serializer but only deals with the
simulcast SDP.
BUG=None
Change-Id: I6bed6ada28ad5b96f07fd7670ad3d635bd4bc732
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320280
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40761}
If there happens to be an asymmetry between local and remote codecs we
shouldn't validate that there's a 1:1 packetization mapping for every
codec. It's sufficient to check that there's at least one matching
packetization per codec.
Bug: webrtc:15473
Change-Id: Ib4fc8fdd54bb4dccf96f0c802746c848e2deed83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320440
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Auto-Submit: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40760}
This time, hit the BUILD files too (where possible).
Bug: webrtc:11943
Change-Id: Ic8f2d77e1ba66f740efe0ef73b1ea6051356dedc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/318100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40654}
The implementation covers the latest specification, but does not
support mixed-codec simulcast at the moment.
Changing codec for audio and video is supported.
Bug: webrtc:15064
Change-Id: I09082f39e2a7d54dd4a663a8a57bf9df5a851690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311663
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40616}
There are now multiple ways to configure VP9 L1Tx:
- Legacy API: configure legacy SVC and disable encodings, this gets
interpreted as disabling spatial layers (non-standard API hack).
- Standard API: configure scalability_mode. This can be done either
with a single encoding or multiple encodings. As long as only one
encoding is active we get a single L1Tx ssrc, same as legacy API.
Due to a bug, the ApplySpatialLayerBitrateLimits() logic which tweaks
bitrates was only applied in the legacy API code path, not the standard
API code path, despite both code paths configuring L1Tx.
The issue is that IsSimulcastOrMultipleSpatialLayers() was checking if
`number_of_streams == 1`. This is true in legacy code path but not
standard code path. The fix is to look at
`numberOfSimulcastStreams == 1` instead, which is set to the correct
value regardless of code path used.
This CL adds comments documenting the difference between
`number_of_streams` and `numberOfSimulcastStreams` to reduce the risk
of more mistakes like this in the future.
Bug: chromium:1455039, b:279161263
Change-Id: I69789b68cc5d45ef1b3becd310687c8dec8e7c87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/308722
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40287}
This makes the handling somewhat more uniform, and is the same
for both video and audio channels.
Bug: webrtc:13931
Change-Id: I26605c56e069e8a34e03708d45eb27a6b7492130
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40107}
This is in preparation of using the state that SourceTracker manages
for more things than only getContributingSources. Audio levels reported
via getStats(), aren't consistent with levels reported via getCS.
Since more operations will be derived from the ST owned data, moving
the management of it away from the audio thread, reduces the potential
of contention.
Bug: webrtc:14029, webrtc:7517, webrtc:15119
Change-Id: I553f7e473316a1c61eeb43ded905a18242a04424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302280
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39943}
One problem with the existing Send() method is that it has a return
value that is problematic for a fully async implementation.
A second problem with Send() is that the return value is bool and not
RTCError (webrtc:13289), which is why OnSendComplete() uses RTCError.
Also, start deprecating `bool Send()` in favor of `void SendAsync()` and
adding `network_safety_` flag for posting async operations to the
network thread. This flag also takes over from the
`connected_to_transport_` which can now be removed.
Bug: webrtc:11547, webrtc:13289
Change-Id: I87bbc7e9b964a52684bdfe0e6ebc5230be254e8b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299760
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39817}
This is a pure rename+move of PeerConnectionSimulcastMediaFlowTests.
The reason for renaming is to reflect that a) this is an integration
test, not a unit test, and b) not all of the tests use simulcast (some
use a single encoding, i.e. singlecast or SVC).
Shared helper functions between PeerConnectionEncodingsIntegrationTest
and PeerConnectionSimulcastTests are placed in a test-only util file.
# Already pass, no need to wait for chromium bots for webrtc testonly CL
NOTRY=True
Bug: webrtc:15063
Change-Id: Iec90d1a7ab712be1395c7644723422c8c6179974
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300540
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39799}
This reverts commit 7f16fcda0fd5bb625584b71311dd37b54c096136.
Reason for reland: Re-landing after addressing issues in downstream
code and hardening the ObserverAdapter from situations where attempted
usage of data channel proxies could occur after shutting down the
peer connection and terminating the network thread.
Original change's description:
> Revert "[DataChannel] Send and receive packets on the network thread."
>
> This reverts commit fe53fec24e02d2d644220f913c3f9ae596bbb2d9.
>
> Reason for revert: Speculative revert, may be breaking downstream project
>
> Original change's description:
> > [DataChannel] Send and receive packets on the network thread.
> >
> > This updates sctp channels, including work that happens between the
> > data channel controller and the transport, to run on the network
> > thread. Previously all network traffic related to data channels was
> > routed through the signaling thread before going to either the network
> > thread or the caller's thread (e.g. js thread in chrome). Now the
> > calls can go straight from the network thread to the JS thread with
> > enabling a special flag on the observer (see below) and similarly
> > calls to send data, involve 2 threads instead of 3.
> >
> > * Custom data channel observer adapter implementation that
> > maintains compatibility with existing observer implementations in
> > that notifications are delivered on the signaling thread.
> > The adapter can be explicitly disabled for implementations that
> > want to optimize the callback path and promise to not block the
> > network thread.
> > * Remove the signaling thread copy of data channels in the controller.
> > * Remove several PostTask operations that were needed to keep things
> > in sync (but the need has gone away).
> > * Update tests for the controller to consistently call
> > TeardownDataChannelTransport_n to match with production.
> > * Update stats collectors (current and legacy) to fetch the data
> > channel stats on the network thread where they're maintained.
> > * Remove the AsyncChannelCloseTeardown test since the async teardown
> > step has gone away.
> > * Remove `sid_s` in the channel code since we only need the network
> > state now.
> > * For the custom observer support (with and without data adapter) and
> > maintain compatibility with existing implementations, added a new
> > proxy macro that allows an implementation to selectively provide
> > its own implementation without being proxied. This is used for
> > registering/unregistering a data channel observer.
> > * Update the data channel proxy to map most methods to the network
> > thread, avoiding the interim jump to the signaling thread.
> > * Update a plethora of thread checkers from signaling to network.
> >
> > Bug: webrtc:11547
> > Change-Id: Ib4cff1482e31c46008e187189a79e967389bc518
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299142
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39760}
>
> Bug: webrtc:11547
> Change-Id: Id0d65594bf727ccea5c49093c942b09714d101ad
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300341
> Auto-Submit: Andrey Logvin <landrey@webrtc.org>
> Owners-Override: Andrey Logvin <landrey@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39764}
Bug: webrtc:11547
Change-Id: I47dfa7e7168be0cd2faab4f8f3ebf110c3728af5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300360
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39786}
With this cl, packets that are discarded in RtpTransport now notifies Call, so that
they can be part of BWE even if they are dropped.
These packets have been recevied on the transport, and has bin decrypted
and parsed and thus can be accounted for.
The un demuxable packets are forwarded to Call similarly how RTCP packets are forwarded.
Bug: webrtc:14928
Change-Id: Ia53349c7b316c4442a3c7aac085a85ec4f4ab9ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299262
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39727}
Make DataChannelController's AddSctpDataStream and
RemoveSctpDataStream be required to be called on the network thread.
This moves blocking calls within those methods over to the
SctpDataChannel class instead.
For production code there's no functional change in this CL. However, this CL:
1) Introduces an actual dedicated network thread to
DataChannelController and SctpDataChannel tests.
2) Removes two data_channel_transport() checks inside DCC that
were being done on the wrong thread (signaling) and
3) introduces a network calling block to SctpDataChannel, where more
network thread related work needs to be done and can be bundled.
(to be done in follow-up CLs).
Bug: webrtc:11547
Change-Id: I6787ac395e61d4a25ae3a74a123e3357cbb46b54
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298052
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39688}
This reverts commit 2d3b294e49027607c80766c50f1c3c8d7d4b38b9.
Reason for revert: The CL was believed to make AV1 always available
but it turned out that the import bots still failed due to not
having AV1, so it is better to use the built in factories than
to make custom test-only ones.
Original change's description:
> Ensure AV1 is always available in PeerConnectionSimulcastTests.
>
> Unblocks a WebRTC import where a bot without AV1 support would
> otherwise have been running and failing during setting codec
> preferences.
>
> # Non-chromium bots passed, no need to wait for chromium to land.
> # Want to unblock importer.
> NOTRY=True
>
> Bug: webrtc:15005
> Change-Id: I93c6a0ce5591a057c3a0ee49f6dbaef3676c0e1d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298021
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Jeremy Leconte <jleconte@google.com>
> Cr-Commit-Position: refs/heads/main@{#39592}
Bug: webrtc:15005
Change-Id: I8f0850852edb0d0234000b2d956e2648a9adf904
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298120
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39596}
Unblocks a WebRTC import where a bot without AV1 support would
otherwise have been running and failing during setting codec
preferences.
# Non-chromium bots passed, no need to wait for chromium to land.
# Want to unblock importer.
NOTRY=True
Bug: webrtc:15005
Change-Id: I93c6a0ce5591a057c3a0ee49f6dbaef3676c0e1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39592}
The goal of the VP9 simulcast project is that when `scalability_mode`
is set, multiple encodings are always interpreted as simulcast, even
if VP9 or AV1 is used. This CL makes this so, but only if the flag
"WebRTC-AllowDisablingLegacyScalability" is "/Enabled/". This allows us
to make "SendingThreeEncodings_VP9_Simulcast" EXPECT VP9 simulcast.
When we are ready to ship we will remove the need to use the field
trial, but before we ship this we'll want to revisit if
SvcRateAllocator can be updated to support simulcast. (Today if we use
SvcRateAllocator when VP9 simulcast is used, all encodings except the
first one get bitrate=0, causing the test to fail because media is not
flowing on all layers.) For now, a TODO is added.
Bug: webrtc:14884
Change-Id: Ie20ae748b0c0405162f3a1b015ab94956ef83dae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297340
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39552}
Instead there are direct member variables for the various relevant
states, some weren't needed, some can be const but the `id` member
in particular needs special handling and can't be const.
For dealing with the stream id, we now have SctpSid. A class that does range validation, checks thread safety, handles the special `-1` case (for what's essentially an unsigned 16 bit int). Using a special type
for this also has the effect that range checking happens more
consistently (although I'm not modifying the structs in api/).
With upcoming steps of avoiding thread hops, the ID may need to
migrate to the network thread, which the thread checks will help with.
Along the way, update SctpSidAllocator to use flat_set instead of std::set and moving some of the sctp data channel code to the cc file
to help with more accurately tracking code coverage.
Bug: webrtc:11547
Change-Id: Iea6e7647ab8f93052044c5afbcc449115206b4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296444
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39539}
Instead, just use the posted task to release the reference to a
pending data channel object.
Bug: none
Change-Id: I34f0bfd604cab88587a892eaa218856c890fc907
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296767
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39527}
This test (and binary) will be used to verify that the field trial
for enabling split-MediaChannel works in both "off" and "on" modes,
so that it can be run as a field trial. It is intended to be deleted
once the conversion to split-MediaChannel is complete.
Bug: webrtc:13931
Change-Id: If62d19be9b2f205067b86dc859946279442fdd58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296322
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39493}
RTCStatsCollector internally keeps track of open data channels but
does not need (or want) to interact directly with those channels,
hence uintptr_t was used instead of pointers to the channel objects.
This changes that to use void* to avoid having to do the cast.
This is a follow-up action item to
https://webrtc-review.googlesource.com/c/src/+/295781
This CL also changes the container type:
std::set -> webrtc::flat_set
Bug: webrtc:12689
Change-Id: I13d3f4a41ef83dab38411193187e872b9d6d3cff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295871
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39468}
DataChannelController used WeakPtr to clear outstanding references
upon destruction - except for the case of SctpDataChannel where we
had a pointer+flag for the same purpose. This change updates
SctpDataChannel and FakeDataChannelController to use a consistent
approach.
Bug: none
Change-Id: I0248471c241365a2c0de76afbb37302115650194
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39464}
It turns out that there were several sigslot instances across data
channel, pc and stats classes that in practice only served as means
to update two counters in RTCStatsCollector. There's already a
notification path that's suitable.
This also fixes a case where the PC instance sat in the middle
of notifications from datachannels to the datachannel controller.
Bug: webrtc:11943
Change-Id: Ic60b76021584019f82085f6651230fe2fe82d465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39456}
Previous tests only asserted that O/A succeeded and that the number of
encodings was as expected. This test goes further and also asserts that
bytesSent eventually becomes non-zero (after an initial ramp-up time).
Let's get testing straight before we add VP9 simulcast support.
Bug: webrtc:14885, webrtc:14884
Change-Id: Idccce66698a077264fa0df2c448c8474d2439aea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291960
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39271}
This is done by allowing implementations of AudioDeviceModule to
implement the GetStats() method. The default implementation returns
nullopt, in which case RTCAudioPlayoutStats will not be visible in the
stats.
Bug: webrtc:14653
Change-Id: I8e4aa6f1b8fcfa47a30f633d28a4013191752e20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290563
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Fredrik Hernqvist <fhernqvist@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39115}
With this cl, a packet is only parsed once in RtpTransport::DemuxPacket and the metadata is reused.
Extensions are still identified twice- one for demuxing based on mid. The second time in Channel::OnReceivedPacket in order to use extensions specific to that mid.
Bug: webrtc:7135, webrtc:14795
Change-Id: I50e3814af92ca4378f148876b20a54bcfac1e146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290540
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39058}
