The test is flaking during shutdown in general, but this is especially
apparent on ASAN bots that don't retry. It is related to
heap-use-after-free and is believef to be a test-only problem.
Bug: webrtc:15018
Change-Id: I593c18dbbe45605404b17af74fd7101f118db7bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299003
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39661}
This makes "WebRTC-AllowDisablingLegacyScalability" enabled-by-default,
meaning any app can opt-in to spec-compliant simulcast when
scalabilityMode is specified.
The opt-in criteria is also made more restricitve: you now have to
specify both scalabilityMode and scaleResolutionDownBy to get simulcast,
otherwise you continue to get legacy "single stream" path.
The reason for this is not to cause any surprises in use cases like
[{scalabilityMode:"L1T1", active:true}, {active:false}, {active:false}]
In cases like this where scaleResolutionDownBy is not specified, it
defaults to 4:2:1 if simulcast is used but the legacy path caps it to
one stream, meaning full resolution. By restricing simulcast only to
cases that set scaleResolutionDownBy, we remove the risk of an app
getting a different resolution than expected due to opt-in.
Bug: webrtc:14884, webrtc:15005
Change-Id: I5efb87af60afaeb1e3ff76698d887aaa1f9d63a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298922
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39660}
Various "if streams == 1" cases are updated to "if
IsSinglecastOrAllNonFirstLayersInactive()" in order not to cause subtle
differences between VP9 {active} and VP9 {active,inactive,inactive}.
This CL also affects a line that conditionally sets
`simulcastStream[0].active = codec_active` so it seemed fitting to
improve the test coverage of "if all streams are inactive, don't send".
Bug: webrtc:15028
Change-Id: I8872dc8be0f2dfc1d8914bdba5e6433f9ba8cbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298881
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39656}
This reverts commit be03c0971863c9e6807fcbdb4175754e8242a652.
Causes regression in web projects that
1/ add a stopped-by-default extension in SRD
2/ call createAnswer
3/ munge the stopped-by-default extension back in SLD
4/ create a subsequent offer and expect the extension to be present
BUG=chromium:1051821
Change-Id: If77f77c2c0ac19625f502fb8a07facd151475807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298744
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39655}
Attempting to ship "WebRTC-AllowDisablingLegacyScalability" revealed a
DCHECK that happens when negotiating 3 VP9 streams prior to the
setParameters() call:
1. By default, `scalability_mode` is missing, so those 3 streams
defaulted to legacy SVC, meaning only a single stream is used.
2. Then, setParameters() was called to make
`encodings[0].scalability_mode = "L2T2_KEY"` and
`encodings[1-2].active = false`. The inactive streams were just
dummies and never expected to exist.
Without simulcast support this is OK, because both 1) and 2) are
interpreted to have a single stream. But with simulcast support, 1) is
interpreted as single stream and 2) as three streams (1 active, 2
inactive). This should be roughly the same setup, but our code treats
them differently.
The DCHECK crash was a mismatch in number of streams in one of the
layers.
The fix is to re-create the streams when the number of streams change
for this reason. The new test revealed other issues and fixes too:
- Support for multiple spatial layers (e.g. "L2T2_KEY") when multiple
encodings exist but only one encoding is active.
- Allow inactive layers not to have a scalability mode set.
A laundry list (https://crbug.com/webrtc/15028) has been created to
update known places doing "if streams == 1" that need to do "if
active streams == 1" instead.
Credit:
The RecreateWebRtcStream() fix is based on eshr@'s POC from
https://webrtc-review.googlesource.com/c/src/+/298565.
Bug: webrtc:15016
Change-Id: I909a3f83a4ef53562894549ade0a870b208cec7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298443
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/main@{#39651}
This allows the SslRole to be queried from the network thread which
will simplify some code paths and avoid thread hopping.
The next steps will be to remove GetSctpSslRole and only query the
DTLS role on the network thread and start combining other operations.
Bug: webrtc:11547
Change-Id: I222dc838fc5ee274a294c8d81d38b5a4ea8fea1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298302
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39642}
after the limitation to 32 TURN servers shipped in M110
BUG=webrtc:13195
Change-Id: I247e5b164188751d94eb9f4fb93aadf1dd645d2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298308
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39634}
This is a small tweak to explicitly remove this second construction
step from SctpDataChannel (async call to OnTransportReady) and move
it over to DataChannelController, which is where OnTransportReady()
is called from otherwise.
Bug: webrtc:11547
Change-Id: Ie86fa85cbb79b405248f88b47d5920c7f163dba6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297921
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39627}
so the size and order corresponds to the local capabilities.
The direction may differ.
BUG=chromium:1051821
Change-Id: Icf5312237b8ed137f822c9f7dd35f70a01d2df99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298043
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39623}
This avoids a couple of layers of error code conversion, reduces
dependency on cricket error types and allows us to preserve error
information from dcsctp. Along the way remove SendDataResult.
Bug: none
Change-Id: I1ad18a8f0b2fb181745b19c49f36f270708720c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298305
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39619}
Prior to this CL we would EXPECT_TRUE_WAIT until
HasOutboundRtpExpectedResolutions() confirmed that we achieved the
maximum expected resolution on all simulcast layers. This was meant to
catch bugs in case the wrong layers were configured with the wrong
layer resolutions.
The problem is that if CPU or BW adaptation kicks in, all layers get
downscaled by some factor and the test may not always recover in time,
e.g. if running on slow slow bots.
This CL relaxes the expectation only to fail if the resolution
exceeds what we expect, not if they are smaller. This is not as air
tight but it should still catch most bugs of interest and reduce
flakiness.
This was reported in comment https://crbug.com/webrtc/15018#c14 but
note that this CL does not attempt to fix the other ASAN issue.
Bug: webrtc:15018
Change-Id: I3305bdade5d1626b09aa5c67217bdedb22cdd876
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298563
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39615}
This reverts commit 75990b9a8f98ea2d597a31472fb778ec4d55f698.
Reason for revert: Breaks downstream, a use case of having three VP9
encodings, scalability mode only specified on the first layer
(L2T2_KEY) and the other two layers not having a scalability mode but
also being active=false appears to trigger a DCHECK in
call/rtp_video_sender.cc:501. More investigation needed
Original change's description:
> Ship ability to opt-in to VP9/AV1 simulcast.
>
> With this unflagging, an app can opt-in to simulcast when using multiple
> encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This
> ensures backwards-compat with apps relying on 3 encodings to mean SVC
> who traditionally have not specified scalabilityMode.
>
> It fixes the spec/API bug of asking for simulcast and not getting
> simulcast. The field trial exists only as a kill-switch with a TODO to
> remove it.
>
> This ships initial support, however note that the VP9/AV1 simulcast uses
> SimulcastRateAllocator (just like VP8/H264 simulcast). This rate
> allocator uses more kbps than SvcRateAllocator. This should be revisited
> to avoid significant higher bitrates, for example when comparing VP9
> simulcast to VP9 SVC.
>
> Shipping the ability for apps to opt-in makes it easier to exercise
> these new code paths and allows initial feedback from developers, but
> due to the high bitrate (= same bitrate as VP8/H264 simulcast today)
> many apps may find that VP9 SVC is still more beneficial for BW reasons.
>
> Bug: webrtc:14884, webrtc:15005
> Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39601}
Bug: webrtc:14884, webrtc:15005
Change-Id: Ic8f77e6a2971f493d6cd8c23faecd435058a8847
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298440
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39605}
These log statements may have been useful when the initial code was
being written but now it's essentially dead code except for when
debugging while working on the code (and then, enabling the log
statements is simple).
Low-Coverage-Reason: CL modifies VERBOSE log lines that aren't currently covered.
Bug: none
Change-Id: Id9a45fe53574d39ff3feba08c596e0ac4ce294fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297760
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39604}
With this unflagging, an app can opt-in to simulcast when using multiple
encodings by specifying RTCRtpEncodingParameters.scalabilityMode. This
ensures backwards-compat with apps relying on 3 encodings to mean SVC
who traditionally have not specified scalabilityMode.
It fixes the spec/API bug of asking for simulcast and not getting
simulcast. The field trial exists only as a kill-switch with a TODO to
remove it.
This ships initial support, however note that the VP9/AV1 simulcast uses
SimulcastRateAllocator (just like VP8/H264 simulcast). This rate
allocator uses more kbps than SvcRateAllocator. This should be revisited
to avoid significant higher bitrates, for example when comparing VP9
simulcast to VP9 SVC.
Shipping the ability for apps to opt-in makes it easier to exercise
these new code paths and allows initial feedback from developers, but
due to the high bitrate (= same bitrate as VP8/H264 simulcast today)
many apps may find that VP9 SVC is still more beneficial for BW reasons.
Bug: webrtc:14884, webrtc:15005
Change-Id: I748aae1adb47acc8a6b79b5852cff6aa47a46f5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298046
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39601}
- Patch Set 1: Re-land of 297982.
- Patch Set 2: Skip test (return early) if AV1 is not available.
Original CL description (297982):
This is something we get "for free" with the
"WebRTC-AllowDisablingLegacyScalability" field trial that has been
wired up to support VP9 simulcast.
This test works and passes, however the ramp-up time is pretty bad.
- VP9 simulcast takes approximately 4 seconds to ramp up.
- VP9 SVC takes approximately 16 seconds to ramp up.
- AV1 simulcast takes approximately 22 seconds to ramp up.
A TODO is added (webrtc:15006) and the test is given extra timeout,
a full minute to get bytes flowing on all layers.
Despite ramp-up being bad, it's important to test that AV1 simulcast
is in fact working to avoid regressions due to obsolete assumptions
about which codec do or do not support simulcast. AV1 simulcast is an
opt-in feature so there is no harm in the API not being perfect yet.
Bug: webrtc:15005, webrtc:15006
Change-Id: Ie8ec9f17c709ef93525e4ea5feb7c95496062add
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298050
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39597}
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}
This reverts commit c8ab6c449c84f17fabf8da58456d396bdb5da762.
Reason for revert: new test fails to run upstream
Original change's description:
> Exercise AV1 simulcast paths in tests.
>
> This is something we get "for free" with the
> "WebRTC-AllowDisablingLegacyScalability" field trial that has been
> wired up to support VP9 simulcast.
>
> This test works and passes, however the ramp-up time is pretty bad.
> - VP9 simulcast takes approximately 4 seconds to ramp up.
> - VP9 SVC takes approximately 16 seconds to ramp up.
> - AV1 simulcast takes approximately 22 seconds to ramp up.
>
> A TODO is added (webrtc:15006) and the test is given extra timeout,
> a full minute to get bytes flowing on all layers.
>
> Despite ramp-up being bad, it's important to test that AV1 simulcast
> is in fact working to avoid regressions due to obsolete assumptions
> about which codec do or do not support simulcast. AV1 simulcast is an
> opt-in feature so there is no harm in the API not being perfect yet.
>
> Bug: webrtc:15005, webrtc:15006
> Change-Id: If0158d172647f0462bd6db802406249d93e01871
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297982
> 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@{#39586}
Bug: webrtc:15005, webrtc:15006
Change-Id: I7da6df8bb51219e7d0acfd3b62b4ec08e25bfdc7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298049
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39595}
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}
This is something we get "for free" with the
"WebRTC-AllowDisablingLegacyScalability" field trial that has been
wired up to support VP9 simulcast.
This test works and passes, however the ramp-up time is pretty bad.
- VP9 simulcast takes approximately 4 seconds to ramp up.
- VP9 SVC takes approximately 16 seconds to ramp up.
- AV1 simulcast takes approximately 22 seconds to ramp up.
A TODO is added (webrtc:15006) and the test is given extra timeout,
a full minute to get bytes flowing on all layers.
Despite ramp-up being bad, it's important to test that AV1 simulcast
is in fact working to avoid regressions due to obsolete assumptions
about which codec do or do not support simulcast. AV1 simulcast is an
opt-in feature so there is no harm in the API not being perfect yet.
Bug: webrtc:15005, webrtc:15006
Change-Id: If0158d172647f0462bd6db802406249d93e01871
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297982
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@{#39586}
Following https://webrtc-review.googlesource.com/c/src/+/297100
it seems that sctp_data_channels_ gets modified while we're iterating
through it. This temporary fix creates a copy of the array and iterates
through the copy instead of sctp_data_channels_. A follow-up CL (or CLs)
will provide more clarity, testing and regression guards.
Bug: webrtc:15004
Change-Id: I0cb5dfb6829d36b51328875c8c9cfa392ff393a7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297981
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39584}
Since AddTrack now has an implicit init_encodings value, it will also
have a StableState saved when associating a transceiver.
That state may not have a saved mid and mline_index, and so on a
rollback, it could blindly reset the mid and mline_index of an
associated transceiver.
This is wrong, the mid and mline_index of associated transceivers
should only be updated when the StableState objects actually
have one saved.
Bug: chromium:1424238
Change-Id: I8e80a04cd072d90200ca7643de892c0ef29b1f1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297920
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39577}
All tests do this already except for RTCStatsCollectorTest.
Bug: none
Change-Id: I318f45a2c79b3d07ca6c92902ebb4f0622ec3200
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297862
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39576}
This removes the behavior of not requesting datachannel if the first
datachannel is closed before the offer is created.
Bug: chromium:1423562
Change-Id: I90eab0f908507e65d9ee3dff51842ee6d61a8aa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297860
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39570}
This struct only contains two member variables now and there isn't
much value added by having it.
Low-Coverage-Reason: No change in coverage, CL modifies uncovered RTC_LOG lines.
Bug: none
Change-Id: I924d450f4c8f8e49b1cfeabaebee9fd5235a90cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297360
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39563}
This includes:
* SignalDataChannelTransportWritable_s
* SignalDataChannelTransportReceivedData_s
* SignalDataChannelTransportChannelClosing_s
* Removing sigslot::has_slots<> inheritance from SctpDataChannel
Instead, we use the existing sctp_data_channels_ vector of channels
known to the DCC to deliver the callbacks.
Bug: webrtc:11943, webrtc:11547
Change-Id: I7935d7505856eedf04981b8ba665ef8419166c1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/297100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39557}
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}
This removes one sigslot and also simplifies the teardown procedure
of a data channel when the channel is closed by the transport.
In this case we no longer need an additional async teardown task that
releases the last remaining reference to the channel.
Bug: webrtc:11943, webrtc:11547
Change-Id: I1c170349a6cbb3cb3c5a47d284e3a3d416c92b11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39551}
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}
The test was assuming that after all thee layers have bytesSent > 0 we
would have fully ramped up to the expected resolutions. But there are
reasons why this may not be true, such as if adaptation kicks in.
This CL attempts to de-flake by using kLongTimeoutForRampUp when
checking the resolutions as well.
// Just increasing a timeout...
NOTRY=True
Bug: webrtc:14884
Change-Id: I5ef57ec3e3cc99552c9ae32a6fdf07889ff06ee1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296883
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39534}
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}
If a data channel object was closed (via calling Close()) right after
construction and before attaching to a transport, it would never
transition to the `kClosed` state. This addresses that corner case,
which caused a DCHECK to trigger but might also cause a situation
whereby more than one DC instance existed for a given sctp sid.
Bug: chromium:1421534
Change-Id: Id757c0528f929f2e2daa5343236d7f62e309f6cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296341
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39513}
following the updates from
https://github.com/w3c/webrtc-extensions/pull/142
BUG=chromium:1051821
Change-Id: I2d561bad1ddffb412bdd7e66cf62a3cb5fc73791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296480
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@google.com>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39500}
to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.
The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.
BUG=webrtc:14973
Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
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@{#39497}
This reverts commit 18c869bc36b342cd4a79947067e52a93a04a7808.
Reason for revert: Added a field trial that allows landing the code without affecting performance in prod.
This CL also incorporates subsequent CLs that also had to be reverted.
Original change's description:
> Revert "Use two MediaChannels for 2 directions."
>
> This reverts commit 8981a6fac3d665beac4a58b9453e6c39988a024f.
>
> Reason for revert: Quality regression detected.
>
> Original change's description:
> > Use two MediaChannels for 2 directions.
> >
> > This CL separates the two directions of MediaChannel into two separate objects that do not couple with each other.
> >
> > The notable API change is that receiver local SSRC now has to be set explicitly - before, it was done implicitly when the send-side MediaChannel had a stream added to it.
> >
> > Bug: webrtc:13931
> > Change-Id: I83c2e3c8e79f89872d5adda1bc2899f7049748b3
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288400
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39340}
>
> No-Try: true
> Bug: webrtc:13931
> Change-Id: I791997ad9eff75c3ac9cd2e4bbacf5bc6c3a3a79
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295663
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39445}
Bug: webrtc:13931
Change-Id: I1318910a685188e2b846c9040e1efc04c2c894ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296080
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39494}
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}
