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}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This helps in figuring out which dependencies exist, and gets closer
to obeying the "one target per .cc file" rule.
Test failures seem unrelated, so using No-Try.
No-Try: true
Bug: webrtc:14775
Change-Id: Id25466c8b8fe628d05c819cf7c69ae6d8421c6cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38910}
This is in the webrtc-stats spec at
https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-scalabilitymode.
This adds the scalability mode to CodecSpecificInfo which is used to
plumb the modes for each simulcast layer.
TBR=orphis@webrtc.org
Tested: Compiled into Chrome and confirmed the scalability mode set for AV1, VP9, VP8 and H264 software encoders in chrome://webrtc-internals.
Bug: webrtc:14730
Change-Id: I71ceba8f6485a4f4a73e0856031b8d5f16f913f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285085
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38847}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
ScalabilityMode should be validated against the currently
allowed codecs or the currently used codec.
Bug: webrtc:11607
Change-Id: Id2e6cbfad4f089de450150e1203657ed316e2f29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277403
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38433}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
`cached_certificates_by_transport_` is used on the network thread, but
can be cleared from the signaling thread. To fix the race where clear
happens at the same time as stats collecting, a mutex is added.
This mutex should very rarely be contended in practise since
ClearCachedStatsReport() typically only happen during renegotiation
(e.g. when someone joins/leaves) and getStats only happens once per
second or less (typically).
NOTRY=Everything passes except unrelated purple bot
Bug: webrtc:14510
Change-Id: Iaf539a5cc8c87184fa0a87b9c889a13b941a9ad1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38262}
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed
Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
callback are know at construction time and only need some synchronization at destruction time. In this case such synchronization can be done with cheaper/simpler WeakPtr concept.
Asynchronous call to SetCertificate is no longer needed thanks to
previous removal of sigslot in
https://webrtc-review.googlesource.com/c/src/+/192362
Bug: webrtc:11943
Change-Id: Icadbcb4f83be9ed4b8f53a72beaef8573f2c9356
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272402
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37868}
