Bug: chromium:1431897
Change-Id: Ib871dc22d2cf93180d7aa05016e34ffec944d73e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301040
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/main@{#39830}
This component is mostly "glue" and is heavily tested in the
socket tests, but not the ToString method, which results in
getting "low test coverage" warnings.
So for the sake of it, add a test that verifies that it works.
Bug: None
Change-Id: Id2b75e2798f334452be50631ef1ff15f53fe4157
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300441
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39826}
Fix Issue 15059: The target bitrate was mistakenly set to be the maximal
bitrate when initializing the libaom encoder.
Bug: webrtc:15059
Change-Id: I38498d4cce7b0a9c26736d9f1096178dd2e1fef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300004
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39822}
This reverts commit dd557fdb1e300068c62c870d9dc5273b48c7b79d.
Reason for revert: Looks like the Chromium FYI builders are failing.
Original change's description:
> [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream
>
> This remove use of MaybeWorkerThread* rtp_transport_queue_ from
> AudioSendStream. The worker queue is alwauys assumed ot be used where
> rtp_transport_queue_ was used.
>
> Bug: webrtc:14502
> Change-Id: Ia516ce7340d712671e0ecb301bba9d66e7216973
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300400
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39816}
Bug: webrtc:14502
Change-Id: I0547548032756fc579b76b6bb362f576aa06b8f7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301020
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39820}
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 remove use of MaybeWorkerThread* rtp_transport_queue_ from
AudioSendStream. The worker queue is alwauys assumed ot be used where
rtp_transport_queue_ was used.
Bug: webrtc:14502
Change-Id: Ia516ce7340d712671e0ecb301bba9d66e7216973
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300400
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39816}
Removes the only remaining dependency on sequence number in NetEq
except for the NackTracker (which arguably doesn't belong in NetEq).
This fixes a potential issue where FEC is not perfectly aligned with
the original packet boundaries, causing both the FEC and the original
packet to be decoded.
Bug: webrtc:13322
Change-Id: I3abec9ebfc194976fca42d5f4f4ed4ee136f44ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300560
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39815}
Cleanup and remove usage of MaybeWorkerThread from VideoSendStream.
VideoSendStream is now created and lives on the worker thread.
Bug: webrtc:14502
Change-Id: I81ccf6b9fc6e8889db81b09bd4a75a3831a003e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300842
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39814}
This remove use of MaybeWorkerThread from TaskQueuePacedSender. Instead,
the TaskQueue used when creating the TaskQueuePacedSender is used for
pacing. That is, the "worker thread".
Bug: webrtc:14502
Change-Id: I504f8e634653af6493e609db6e42b07d488fd699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300820
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39813}
Previously this test only ran on VP9, now it runs for all codecs.
Bug: webrtc:15080
Change-Id: Id61a261cef3463a22062e3d313dc2725e051773d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300861
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39812}
NetEq packet source input doesn't have any other uses than rtp dump,
so remove that layer.
Bug: None
Change-Id: I667bb4aead9f0f2fe8a1c0d6d911a4670ded67e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300542
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39810}
As is, send() might return false while error() would indicate OK.
Bug: none
Change-Id: Ia303701148e86e1bcaf70cc54e689a3ff7f5a184
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300822
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39809}
This CL introduces PeerConnectionEncodingsIntegrationParameterizedTest,
which is based on PeerConnectionEncodingsIntegrationTest but covers all
codecs using INSTANTIATE_TEST_SUITE_P (VP8, VP9, H264, AV1).
This applies to all standard paths, which in the case of VP9 and AV1
requires opting in to it by specifying scalabilityMode and
scaleResolutionDownBy. They are also limited to L1Tx because the other
codecs don't support SVC.
The VP9-only tests continue to run as TEST_F with
PeerConnectionEncodingsIntegrationTest.
Bug: webrtc:15079
Change-Id: I3429c90f2f79ff60adad0b33975bccdda31ce6d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300900
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39808}
There was no check for null in the code that prepends config buffer to key frame buffer. It is not a requirement for codec to deliver config buffer. Some codecs probably do not do that.
Bug: none
Change-Id: Id9c57efc5d1de5f569fa773313da4db3cd8b074c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299900
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39807}
This can happen when there are audio SSRCs in the event log without any
associated events.
Issue was introduced in
https://webrtc-review.googlesource.com/c/src/+/300300
Bug: None
Change-Id: Ib0e009095bf67633812d937aa5a9e65e2cd8958a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300743
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39806}
This extends the RTP header extension API usage to generating answers.
Also re-adds unit tests removed by the revert.
BUG=chromium:1051821
Change-Id: Ib754284e9a77cb49e22bea7072c475d240f2563b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298740
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39800}
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}
third_party/webrtc/files/stable/webrtc/sdk/objc/native/api/video_capturer.mm
calls `webrtc::CreateVideoTrackSourceProxy()`, which is defined in
third_party/webrtc/files/stable/webrtc/pc/video_track_source_proxy.cc.
Some pending changes to the Apple related rules will expose this missing
dependency, so adding the missing dependency to not have downstream users
break.
Bug: b/276754006
Change-Id: I278872123f5351614c6e3affbdceffdb7e0f969c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300625
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39798}
We shouldn't treat VP9 simulcast {active,inactive,inactive} different
from VP9 singlecast when it comes to bitrates, so the condition
`config.simulcast_layers.size() <= 1` is updated to
`video_codec.numberOfSimulcastStreams <= 1` which holds true in the
"single active stream" case as well.
This is consistent with existing logic, such as the fact that we use
`SvcRateAllocator` instead of `SimulcastRateAllocator` when
`numberOfSimulcastStreams <= 1`.
Bug: webrtc:15061
Change-Id: I67fc78b9c0f97f1d607c91bbc689b06c3fd5cb48
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298920
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39791}
This is to make sure that thread checks on the signaling thread inside
the ObserverAdapter, don't dereference the `channel_` object which
may have gone away.
(using No-try: true since the internal bots are behind)
No-try: True
Bug: webrtc:11547
Change-Id: I8f1dbf266cfc3f69fea8598a5db9baf82e4db0af
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300601
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39787}
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}
We've only seen heap-use-after-free issues when the test continues to
run after EXPECT_TRUE_WAIT failures. This may speculatively reduce the
risk of flakes by aborting the test as soon as a failure happens.
Ideally the peer connections would all close due to going out of scope
making frame encoding after this point an impossibility.
Bug: webrtc:15018
Change-Id: I69d8bcf0f76e3bfb591d2ea81b9e9f68b1f11ffe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300481
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39782}
