Moves OnSendSideDelayUpdated and OnSendPacketUpdated out from
rtp_sender_unittest and into rtp_sender_egress_unittest and
rtp_rtcp_impl2_unittest. The former test now only tests the logic for
updating send-side-delay stats. The latter is now on a proper
RtpRtcp-level and also verifies that frame timestamps makes it to the
egress (as assumed by the first test).
Bug: webrtc:11340
Change-Id: I784042ad91eb66a4d1eebdbbc625f9522528bfb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218502
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33996}
This removes PacketRouter inheritance from RemoteBitrateObserver and TransportFeedbackSenderInterface.
Call binds methods for sending REMB and transport feedback messages from RemoteCongestionController to PacketRouter.
This is needed until the RTCPTranseiver is used instead of the RTP modules.
Bug: webrtc:12693
Change-Id: I7088de497cd6d1e15c98788ff3e6b0a2c8897ea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215965
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33993}
- The FEC receiver tracks maximum of 48 media packets at a time, and packet reordering can delay the FEC packet from its protected media packets by more than 48 sequences.
- Such FEC packets do not get purged until much later when newer FEC packets with much higher sequence mark them as old.
- Until that happens, they sit in the receiver queue, wasting CPU cycles.
- If the receiver maintains a larger queue size for the media packets, it increases possibility of having all media packets in the queue, thereby organically purging the FEC packet.
- More importantly, this also increases the efficacy of FEC decode for such packet, since media packets now remain relevant for longer and aid in lost packet recovery.
Bug: webrtc:12656
Change-Id: Id0058df9a23ea31839decf2c37e0670a54c947fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215882
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33989}
When VideoFrameType for svc upper layer is kVideoFrameDelta for key pic,
the svc unittest will fail due to the wrong frame type for the super
frame of first key picture.
Bug: None
Change-Id: Iff026aaecb73890d3c45d2c88c9654a12d6fe3bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216461
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Zhaoliang Ma <zhaoliang.ma@intel.com>
Cr-Commit-Position: refs/heads/master@{#33986}
Historically the PacketBuffer used a callback for assembled frames, and because of that RtpPacketInfos were piped through it even though they didn't have anything to do with the PacketBuffer.
With this CL RtpPacketInfos are stored in the RtpVideoStreamReceiver(2) instead.
Bug: webrtc:12579
Change-Id: Ia6285b59e135910eee7234b89b23425bb0fc0d2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215320
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33980}
Before this CL, the RemoteEstimatorProxy used a std::map to track which
arrival time a packet with a certain sequence number was received at.
While this works, it's fairly slow as most manipulations and insertions
are O(log(N)) and there were quite many of them.
By taking advantage that sequence numbers generally are received in
sequence, recording a packet is now amortized O(1). Also other
operations such as creating the periodic feedback reports, are also
much faster as it previously was done by searching quite a few times
in that map.
In highly loaded Media Servers, RemoteEstimatorProxy's usage of
std::map attributes to around 0.52% CPU.
Bug: webrtc:12689
Change-Id: I3dd58105f9fbfb111f176833cd4aa6b040c0e01d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217388
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33979}
This reverts commit 793bac569fdf1be16cbf24d7871d20d00bbec81b.
Reason for revert: rare compilation error fixed
Original change's description:
> Revert "Refactor the PlatformThread API."
>
> This reverts commit c89fdd716c4c8af608017c76f75bf27e4c3d602e.
>
> Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
> See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
>
> Original change's description:
> > Refactor the PlatformThread API.
> >
> > PlatformThread's API is using old style function pointers, causes
> > casting, is unintuitive and forces artificial call sequences, and
> > is additionally possible to misuse in release mode.
> >
> > Fix this by an API face lift:
> > 1. The class is turned into a handle, which can be empty.
> > 2. The only way of getting a non-empty PlatformThread is by calling
> > SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> > code reader.
> > 3. Handles can be Finalized, which works differently for joinable and
> > detached threads:
> > a) Handles for detached threads are simply closed where applicable.
> > b) Joinable threads are joined before handles are closed.
> > 4. The destructor finalizes handles. No explicit call is needed.
> >
> > Fixed: webrtc:12727
> > Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> > Commit-Queue: Markus Handell <handellm@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#33923}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=handellm@webrtc.org
>
> Bug: webrtc:12727
> Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
> Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
> Reviewed-by: Markus Handell <handellm@webrtc.org>
> Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33936}
# Not skipping CQ checks because this is a reland.
Bug: webrtc:12727
Change-Id: Ifd6f44eac72fed84474277a1be03eb84d2f4376e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217881
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33950}
As it will be heavily modified in a follow-up change, it's refactored
as a separate commit to make commits small:
* Extracted code to separate AddPacket and CullOldPackets methods
* BuildFeedbackPacket now returns a packet. In the next iteration, it
might not, so it needs to be able to decide when to increment the
packet sequence number.
* Documented some existing fields.
The follow-up is in change I3dd58105f9fbfb111f176833cd4aa6b040c0e01d.
Bug: webrtc:12689
Change-Id: I5ad0aee5a7da008f9e209f7c13bf299c12f9d1f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217581
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33942}
This reverts commit c89fdd716c4c8af608017c76f75bf27e4c3d602e.
Reason for revert: Causes rare compilation error on win-libfuzzer-asan trybot.
See https://ci.chromium.org/p/chromium/builders/try/win-libfuzzer-asan-rel/713745?
Original change's description:
> Refactor the PlatformThread API.
>
> PlatformThread's API is using old style function pointers, causes
> casting, is unintuitive and forces artificial call sequences, and
> is additionally possible to misuse in release mode.
>
> Fix this by an API face lift:
> 1. The class is turned into a handle, which can be empty.
> 2. The only way of getting a non-empty PlatformThread is by calling
> SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
> code reader.
> 3. Handles can be Finalized, which works differently for joinable and
> detached threads:
> a) Handles for detached threads are simply closed where applicable.
> b) Joinable threads are joined before handles are closed.
> 4. The destructor finalizes handles. No explicit call is needed.
>
> Fixed: webrtc:12727
> Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#33923}
# Not skipping CQ checks because original CL landed > 1 day ago.
TBR=handellm@webrtc.org
Bug: webrtc:12727
Change-Id: Ic0146be8866f6dd3ad9c364fb8646650b8e07419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217583
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33936}
Previously this value was rounded up to a millisecond value.
This change is complementary to another change:
https://webrtc-review.googlesource.com/c/src/+/216398
Bug: webrtc:12722
Change-Id: I0fd2baceb4608132615fb6ad241ec863e343edb1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217521
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33928}
PlatformThread's API is using old style function pointers, causes
casting, is unintuitive and forces artificial call sequences, and
is additionally possible to misuse in release mode.
Fix this by an API face lift:
1. The class is turned into a handle, which can be empty.
2. The only way of getting a non-empty PlatformThread is by calling
SpawnJoinable or SpawnDetached, clearly conveying the semantics to the
code reader.
3. Handles can be Finalized, which works differently for joinable and
detached threads:
a) Handles for detached threads are simply closed where applicable.
b) Joinable threads are joined before handles are closed.
4. The destructor finalizes handles. No explicit call is needed.
Fixed: webrtc:12727
Change-Id: Id00a0464edf4fc9e552b6a1fbb5d2e1280e88811
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215075
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33923}
Before this CL, timestamps of received packets were rounded
to the nearest millisecond and stored as int64_t. Due to the
rounding it sometimes happened that timestamps later in the
pipeline that are not rounded seem to occur even before the
video frame was received.
Change-Id: I92d8f3540b23baae2d4a1dc6a7cb3f58bcdaad18
Bug: webrtc:12722
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216398
Reviewed-by: Chen Xing <chxg@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33916}
VideoCodecInitializer::SetupCodec never sets startBitrate,
so SetAv1SvcConfig shouldn't use it.
Bug: webrtc:12720
Change-Id: I04835dc27368f32c19132d93c72364173d7050fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217382
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33915}
We want to make this information available to RtpTransportControllerSend.
Bug: None
Change-Id: Id9237fe1a1fe65834cf2ac79155bc4502744e4db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216683
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33914}
potentially allowing distances of more than 2.
BUG=webrtc:11640
Change-Id: I0d8c831218285d57cf07f0a8e5829810afd4ab3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188383
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33913}
Also fix similar annotation on NackModule to have effect
(when defining an alias with C++ using, ABSL_DEPRECATED should appear
on the left hand side).
Bug: webrtc:12339
Change-Id: Id80a20bf2c56a826777b8a40e06ac5c65e4f8db7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217242
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33905}
libstdc++ does not allow incomplete type for T2 with std::pair<T1,T2>,
which is used by std::unordered_map. Include full definition of
TmmbrInformation, RrtrInformation and LastFirStatus.
Bug: chromium:957519
Change-Id: I00cad6d5e5a782791f5f64b4e38d7738b2c5ae87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217180
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33897}
New tests (transport sequence number plus newly added abs send time) now
test more of production code and less of rtp_sender_unittest.cc test
fixture code.
Bug: webrtc:11340
Change-Id: I8ec0022c3d18467a4144ce984996af1a452760dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216327
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33895}
Add the option to run the adaptive digital controller of AGC2 without
side-effects - i.e., no gain applied.
Tested: adapation verified during a video call in chromium
Bug: webrtc:7494
Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33875}
In RTPSenderVideoFrameTransformerDelegate::TransformFrame(), the encoder
queue might still be null when a frame queued by a previous delegate
arrives. This could happen in the context of renegotiation that results
in a codec reset. In this case, the frame should be dropped.
Bug: webrtc:12691
Change-Id: Ib738ce31738cffc7e01053dbc82237f457fc2286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216393
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33866}
In highly loaded media servers, ReceiveStatisticsImpl's use of std::map
attributes to ~0.32% CPU. It needs to be able to iterate through the
statisticians in order when reporting, but that is considered to be rare
compared to how often they are looked up. So this commit adds a separate
sorted set for just keeping track of the SSRCs, and letting the map of
SSRC to Statisticians, be unordered.
Bug: webrtc:12689
Change-Id: I69fe41d96bca31b2e8d669b58b5c7afabceaa6a6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216385
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33864}
The test assumed a certain order in report blocks, which can have
changed with tasks to use unordered collections. This commit makes
the test more robust.
Bug: webrtc:12689
Change-Id: Ie0087dcb7dc955d70aa39208848bb99fd2f1750b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216386
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33863}
This is essentially replacing `new rtc::RefCountedObject` with
`rtc::make_ref_counted` in many files. In a couple of places I
made minor tweaks to make things compile such as adding parenthesis
when they were missing.
Bug: webrtc:12701
Change-Id: I3828dbf3ee0eb0232f3a47067474484ac2f4aed2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215973
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33852}
In highly loaded media servers, RTCPReceiver's use of std::map
attributes to ~0.5% CPU. It's mostly ::find and the [] operator, and
they are all keyed by SSRC, which is an unordered data type. This makes
these maps suitable as unordered maps, as they have constant time
complexity for lookups.
Bug: webrtc:12689
Change-Id: I7b305e233fcbed0e452632946ab0de5ee66f8dda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216321
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33850}
In highly loaded media servers, RTCPReceiver's use of std::set
attributes to ~0.87% CPU. It's mostly ::find and the [] operator and the
assignment operator.
* Removed locking of a mutex in `TriggerCallbacksFromRtcpPacket``
as it copied members that were already const.
* Switched the use of std::set for the list of registered local SSRCs
to an absl::InlinedVector, as the set is very small and it's not
expected that any more complicated container would be faster than a
linear search within a cache line.
Bug: webrtc:12689
Change-Id: I734578c22eeca2d9ba89fef77ecc689b72624567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216322
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33849}
In highly loaded Media Servers, RoundRobinPacketQueue's usage of
std::map attributes to around 0.3% CPU. In profiling, it has been found
that it's the `streams_` map that is the culprit and std::map::find
being the function that is used most.
By using an unordered map, lookups should be faster. This will be
evaluated and if this commit doesn't show results, it will be reverted.
Bug: webrtc:12689
Change-Id: Ia368f1184130298cfbb9064528828435aa7a5c66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216320
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33848}
In Chromium, the I420ABufferInterface implementation uses the default
CropAndScale() implementation which converts to I420 in the process.
This should be OK, because we do not encode the alpha channel anyway,
so having WebRTC scaling ignore the alpha channel might even be a good
thing. Unfortunatety, an if statement in the LibvpxVp8Encoder did not
consider I420A and I420 to be the same, resulting in dropping perfectly
valid frames.
This CL fixes that by considering I420A and I420 "compatible" in a
comparison helper function. The problem only happens in this encoder,
so only this encoder needs to be fixed.
Bug: chromium:1203206
Change-Id: Iec434d4ada897c79e09914cac823148fd5b05e57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216323
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33845}
The extracts and refactors some test code from rtp_sender_unittest.cc
and puts it in a new target intended to only test RtpSenderEgress, and
do it as pure unit test, rather than the unholy
not-quite-unit-not-quite-integration-test thingy we have today.
Only a first test case is actually ported with this CL, but it's a
start...
Bug: webrtc:11340
Change-Id: Ie2cdde63a00a6ff6eba7b8d443eeb76ce2a527c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216180
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33841}
This changes add two new options to the DesktopCaptureOptions class so
consumers can opt in to using the WGC capturer. The capturer is still
behind the RTC_ENABLE_WIN_WGC build flag which is off by default, so
these options will have no affect until that flag is enabled.
Bug: webrtc:11760
Change-Id: Ib7166f3bb335f29aeff8cb5d2bebea2c06c14d4c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215243
Commit-Queue: Austin Orion <auorion@microsoft.com>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Cr-Commit-Position: refs/heads/master@{#33837}
The eventual implementation of changing the status will be async so the
return value isn't that useful and was in fact only being used to log
a warning if an error occured.
This change is to facilitate upcoming changes related to media engine.
Bug: webrtc:11993
Change-Id: Ia7f85a9ea18b2648b511fa356918cf32a201461f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215975
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33825}
This change disables native cursor capture in WgcCapturerWin to better
support the existing idiom of wrapping a capturer in a
DesktopAndCursorComposer. That means we also need to set the top_left
property of output DesktopFrames, so I've also implemented GetTopLeft in
WgcCaptureSource to facilitate this. I've also added a few unit tests
for WgcCaptureSource.
Bug: webrtc:12654
Change-Id: I5c9988a6f8548b584451b073ac29fbb482e09e2e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215102
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33821}
Previously, windows captured by WgcCapturerWin may have been occluded
by other windows with no functionality to bring them to the top of the
z order. This change implements FocusOnSelectedSource for
WgcCapturerWin to afford consumers this functionality, and match the
experience offered by current capturers.
Bug: webrtc:12664
Change-Id: I8bc067ade90fba0be66a9be57d3429ba54ba1ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215241
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#33809}
Fix simulcast svc controller to reuse dropped frame configuration,
same as full svc and k-svc controllers do.
This fuzzer reminded the issue was still there.
This is a reland of https://webrtc-review.googlesource.com/c/src/+/212281
Bug: webrtc:11999
Change-Id: Id3b2cd6c7e0923adfffb4e04c35ed2d6faca6704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215921
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33802}
Logic for throttling how often REMB messages are sent is added to ReceiveSideCongestionController as well as a new method SetMaxDesiredReceiveBitrate. These are based on the logic in PacketRouter. The logic for throttling REMB and setting the max REMB will be removed from PacketRouter in a follow up cl.
The purpose is to eventually decouple PacketRouter from sending RTCP messages when RtcpTransceiver is used.
Bug: webrtc:12693
Change-Id: I9fb5cbcd14bb17d977e76d329a906fc0a9abc276
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215685
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33801}
The change introduces support for detachable PlatformThreads, for which
the Stop() call doesn't wait until the thread has finished executing.
The change also introduces rtc::ThreadAttributes that carries priority
and detachability thread attributes. It additionally refactors all
known use to use the new semantics.
Bug: b:181572711, webrtc:12659
Change-Id: Id96e87c2a0dafabc8047767d241fd5da4505d14c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214704
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33796}
- Received FEC packets are purged from the queue if:
1. All media packets protected by the FEC packet are received.
2. All media packets protected by the FEC packet are recovered.
3. Newer FEC packet(s) with sequence number '0x3fff' larger than an old FEC packet is received.
- When FEC packets get separated from their protected media packets by more than 48, none of the first conditions ever delete that FEC packet, no matter how old/ irrelevant it gets.
- Under specific circumstances, the new FEC packet (condition 3) is not received before the media sequence number space wraps around, and incorrectly activates the old FEC packet, resulting in FEC decode for the wrong packet.
- This change purges such old FEC packets in time before the media sequence numbers wrap around.
Bug: webrtc:12656
Change-Id: I6ddf5382638c8c7e9a65724b2544dfbbc4803342
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215100
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33795}
On a 32bit system, this reduces the allocation size of the flag
down from 12 bytes to 8, and removes the need for a vtable (the extra
4 bytes are the vtable pointer).
The downside is that this change makes the binary layout of the
flag, less compatible with RefCountedObject<> based reference counting
objects and thus we don't immediately get the benefits of identical
COMDAT folding and subsequently there's a slight binary size increase.
With wider use, the binary size benefits will come.
Bug: none
Change-Id: I04129771790a3258d6accaf0ab1258b7a798a55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215681
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33793}
The ERLE is used to estimate residual echo for echo suppression. The
ERLE is reduced during far-end offset to avoid echo leakage. When there
is a strong near-end present this can cause unnecessary transparency loss.
This change adds an ERLE estimation that does not compensate for onsets and
uses it for residual echo estimation when the suppressor considers the near-end to be dominant.
Bug: webrtc:12686
Change-Id: Ida78eeacf1f95c6e62403f86ba3f2ff055898a84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215323
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33786}
