When SetCodecPreferences was used, the media session was adding codecs
from a list that didn't have corrected payload type mappings. As a
result, it's possible to generate offers or answers that use the same
payload type for audio and video codecs, which is a clear violation.
Bug: webrtc:12169
Change-Id: Ib7be73b4b3b4c57b8d2f374dba8b039c7a3df5a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231620
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34961}
The experiment WebRTC-PreStreamDecoders (aka Lazy decoder creation) has
investigated the benefit of only creating a subset of all decoders
during negotiation and the remaining decoders on demand.
This CL changes the default value to only create one decoder during
negotiation. This frees up hardware resources and reduces the SDP
negotiation time.
Bug: chromium:1202042
Change-Id: I6e2206839162aa857fcc948ccd53d0ff91cbdeaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231643
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34959}
Following Congestion avoidance and control by V. Jacobson at
https://dl.acm.org/doi/10.1145/52324.52356, use integer math instead
of floating point. Not that it matters, but it results in some code size
savings, and is more efficient. Due to not using floating point math,
some golden values in test cases were rounded a bit differently.
Bug: webrtc:12614
Change-Id: I0b7d54b8fd9ce7156e6b2582437ef5720f8838ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231229
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34956}
This is done by adding a reorder optimizer that estimates the probability of receiving reordered packets.
The optimal delay is decided by balancing the cost of increasing the delay against the probability of missing a reordered packet, resulting in a loss. This balance is decided using the `ms_per_loss_percent` parameter.
The usage and parameters can be controlled via field trial.
Bug: webrtc:10178
Change-Id: Ic484df0412af35610e74b3a6070f2bac7a926a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231541
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34954}
Setting the rtp header extensions on the packet delivery thread
(currently worker, soon to be network), is now possible without
taking the hit of deleting and recreating the receive stream (and
rtp receiver and related state).
Bug: webrtc:11993
Change-Id: I9bbe306844a25d85d79cd216092ead66eaf68960
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/223741
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34953}
The HandleXr method has output arguments that are not set when an RTCP
report cannot be parsed. We should give these a sensible default value
to avoid accessing uninitialized memory
Bug: chromium:1247182
Change-Id: I6c54260aef3834643c41b96c0709489522d82533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231237
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34943}
To focus on the ability to predict clipping, the clipping predictor
evaluator doesn't increment the true positive count anymore when a
prediction is simultaneously observed with a detection.
Note that `WebRTC.Audio.Agc.ClippingPredictor.F1Score` is still used
to log the F1 score - i.e., the histogram hasn't been renamed.
Bug: webrtc:12774
Change-Id: Ia987e568a6df2a3ddba7fa1b5697d6feda22d20c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231233
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34942}
BitstreamReader itself uses idea of Read function that always succeed,
and a separate function to check for errors.
Thus extra layer in the DependencyDescriptorReader is not needed.
Bug: None
Change-Id: Ie58861f2cbecc02a5a1a9538232494b4442c9afd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231226
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34940}
Split out `RelativeArrivalDelayTracker` and `DelayOptimizer` logic.
This is in preparation for adding another `DelayOptimizer` specialized in handling reordered packets.
Bug: webrtc:10178
Change-Id: Id3c1746d91980b171fa524f9b2b71cf11fc75f64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231224
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34938}
Evaluate the clipping predictor whenever injected but keep using the
predictions only when allowed.
Bug: webrtc:12774
Change-Id: I9e8930a528d1d514d52b821a28b6c8ad0c3aeb5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231137
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34937}
Move Precision, Recall and F1-score computation from `AgcManagerDirect`
to a separate function that can be tested.
Bug: webrtc:12774
Change-Id: Iba20f153a72b7f957bf938e0642055d421045c02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231228
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34933}
In benchmarks, each log statement represent 2% of the CPU usage. RTC_LOG
is not very expensive, but not free either, and it's called for every
received and sent packet.
Bug: webrtc:12943
Change-Id: Id65baafb5e494091a3a7604687718fdd4f477d86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231223
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34929}
The previous limits were taken from Oracles SCTP stack[1] as they were
more up-to-date than the suggested ones in RFC4960. However, after
having evaluated them for a while, it's evident that they are a bit too
aggressive and likely have their origin from a wired LAN network.
Let's do a re-take. These values have been taken from Solaris TCP
stack[2]. They are even less aggressive than Linux defaults. This can be
iterated even more, and is always possible to override by the client.
It's generally the increase of rto_min that is helping here, as the
delayed SACK and RTT jitter require that the RTO.min is quite much
higher than the delayed SACK timeout of the peer (which isn't in control
by us, but one can assume it's 200ms or less). And with a too low
RTO.min, it's increased risk of getting spurious retransmissions and
decreasing the congestion window.
[1] https://docs.oracle.com/cd/E93309_01/docs.466/SIGTRAN/GUID-2136614F-4BED-407C-87B0-7EE10E0FF534.htm
[2] https://docs.oracle.com/cd/E19120-01/open.solaris/819-2724/chapter4-69/index.html
Bug: webrtc:12943
Change-Id: I9678ac4396286a55c251c5f57589379da70fd27d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231139
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34927}
According to the RANDR 1.5 spec:
https://cgit.freedesktop.org/xorg/proto/randrproto/tree/randrproto.txt
RRSetMonitor and RRDeleteMonitor requests will generate a
ConfigureNotify event on the root window of the screen.
They do not appear to generate any RRScreenChangeNotify or other
similar event. So this CL causes ScreenCapturerX11's monitor list to be
updated on ConfigureNotify events. It is needed, for example, when
using a commandline such as "xrandr --setmonitor ..." to add monitors.
Bug: None
Change-Id: I1948a8b96800721409472ac6264c935abe169ec3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230882
Commit-Queue: Joe Downing <joedow@chromium.org>
Reviewed-by: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/main@{#34919}
Both suppressions are related to the same lock and can be triggered
from 2 different API calls.
Since those calls are causing a significant amount of flaky tests,
I opted to suppress the issue for now, until we can either find a solution
upstream or replace usrsctp with DcSCTP.
Bug: webrtc:12823
Change-Id: I0ae49c94810264f94838b97781080cf5e6e4b24c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231136
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34916}
This reverts commit 677195d3eb6a5f0bc1d31d794a5190ba281c0335.
Reason for revert: Broke WebRTC to Chrome rolls:
https://crrev.com/c/3141000
example: https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/790256/overview
The error is similar to the failure on previous attempt to land this CL. See: https://crrev.com/c/3135220, and crash https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/787945/overview
Original change's description:
> Reland "PipeWire capturer: implement proper DMA-BUFs support""
>
> This is a reland of f2177f6612079ccce9c320ea7e77bc934c684f5c
>
> Original change's description:
> > PipeWire capturer: implement proper DMA-BUFs support
> >
> > Currently both KWin (KDE) and Mutter (GNOME) window managers don't
> > use DMA-BUFs by default, but only when client asks specifically for
> > them (KWin) or when experimental DMA-BUF support is enabled (Mutter).
> > While current implementation works just fine on integrated graphics
> > cards, it causes issues on dedicated GPUs (AMD and NVidia) where the
> > code either crashes or screensharing is slow and unusable.
> >
> > To fix this, DMA-BUFs has to be opened using OpenGL context and not
> > being directly mmaped(). This implementation requires to use DMA-BUF
> > modifiers, as they are now mandatory for DMA-BUFs usage.
> >
> > Documentation for this behavior can be found here:
> > https://gitlab.freedesktop.org/pipewire/pipewire/-/blob/master/doc/dma-buf.dox
> >
> > Bug: chromium:1233417, webrtc:13137
> > Change-Id: I0cecf16d6bb0f576954b9e8f071cab526f7baf2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227022
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34889}
>
> Bug: chromium:1233417, webrtc:13137
> Change-Id: I7d5763dd5db708cee20a31e559b26db0287f40d6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230946
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34903}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: chromium:1233417, webrtc:13137
Change-Id: I64e2ce864f69e6097aba65ade04af7166e407409
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231135
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34915}
Update users to pass in the appropriate rtc::SocketFactory, instead of
relying on BasicPacketSocketFactory using the rtc::SocketServer
associated with the thread the constructor runs on.
Bug: webrtc:13145
Change-Id: I74eca1ce2c5885c14372a797f6374825b1bc1873
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231134
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34914}
By allowing the max timer backoff duration to be limited, a socket can
fast recover in case of intermittent network issues. Before this CL, the
exponential backoff algorithm could result in very long retry durations
(in the order of minutes), when connection has been lost or been flaky
for a long while.
Note that limiting the maximum backoff duration might require
compensating the maximum retransmission limit to avoid closing the
socket prematurely due to reaching the maximum retransmission limit much
faster than previously.
Bug: webrtc:13129
Change-Id: Ib94030d666433e3fa1a2c8ef69750a1afab8ef94
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230702
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34913}
