If configured, attempt to negotiate "zero checksum
acceptable" capability, which will make the outgoing
packets have a fixed checksum of zero. Received
packets will not be verified for a correct checksum
if it's zero.
Also includes some boilerplate:
- Setting capability in state cookie
- Adding capability to handover state
- Adding metric to know if the feature is used
This feature is not enabled by default, as it will be
first evaluated with an A/B experiment before making
it the default.
When the feature is enabled, lower CPU consumption for
both receiving and sending packets is expected. How
much depends on the architecture, as some architectures
have better support for generating CRC32 in hardware
than others.
Bug: webrtc:14997
Change-Id: If23f73e87220c7d42bd4f9a92772cda16bc18fcb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299076
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39920}
This removes all "_WAIT" operations in the tests and all uses
of kTimeout for loop+poll check for values.
On my linux box, this also reduces the time it takes to run all
./rtc_media_unittests (in parallel) from about 3500ms, to ~50ms.
(longest running test was WebRtcVideoChannelBaseTest.GetStats)
Bug: none
Change-Id: If544aa10cb0281cb5e5e51fb654db5f45de871da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302343
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39918}
This reverts commit 45eae346930aedbf4624d054e0633c94b397b8ec.
It was found not to be the root cause of the performance
regression, so it's safe to reland.
Bug: webrtc:14997
Change-Id: I67c90752875bf4071cbdd5adfa462a37f4d4ceab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302162
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39910}
This reverts commit 2080dacfb7946daf79ecd3f69efbd0c9e08b9be2.
Reason for revert: This CL is causing a lot of flakiness on iOS bots
https://ci.chromium.org/p/webrtc/builders/ci/iOS%20Debug%20%28simulator%29
Original change's description:
> For AV1, disable error resilience on upper temporal layers. Error resilience is no longer required for upper temporal layers. Disabling error resilience on the upper layers leads to a ~2% PSNR BD-rate gain.
>
> Bug: webrtc:15106
> Change-Id: Id92d51defbd26c1a77e3c9fe19607e9db4a3e7c1
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302001
> Reviewed-by: Marco Paniconi <marpan@webrtc.org>
> Commit-Queue: Michael Horowitz <mhoro@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39900}
Bug: webrtc:15106
Change-Id: I24515280113ed6681c9766026ec24d689035c031
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301983
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Owners-Override: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#39903}
https://webrtc-review.googlesource.com/c/src/+/240680 made encoder aware of stride and slice height of input buffer but calculation of buffer size passed to queueInputBuffer() was not updated.
Bug: webrtc:13427
Change-Id: Iba8687f56eda148ac67b331d35c45317a4ec5c59
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301321
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39895}
The deadlock happens when the WebRtcCombinedNetworkAndWorkerThread
experiment is running because the worker thread doing the PostTask is
the same thread as the network thread. When using BlockingCall instead
this method will avoid the PostTask and just execute in-line instead
if the experiment is running and otherwise do what the old path did.
As per webrtc:15099, we do not want to increase uses of rtc::Thread in
general, and adding more block-invokes in is also discouraged
(webrtc:12649) so instead of adding new methods to TaskQueueBase we
simply do a static_cast<rtc::Thread*>.
When WebRtcCombinedNetworkAndWorkerThread has launched the blocking
call can be removed because then we're on a single thread always.
Bug: webrtc:15098
Change-Id: I6dcc09bcf6ee0ad12e4beffef3b206989265540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301880
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39894}
Instead of crashing with a CHECK fail when an insertable stream of a
Video RTPReceiver is given a frame from an RTPSender's insertable
stream, construct a reasonable analogous receive frame and pass it
through to be decoded.
A small step towards removing the split we have between Sender and
Receiver implementations of TransformableFrameInterface which just
confuses users of the API.
Bug: chromium:1250638
Change-Id: I02e0f1d9d35c16dc12718927c5200ff7cf4407e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301181
Reviewed-by: Palak Agarwal <agpalak@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Tony Herre <herre@google.com>
Cr-Commit-Position: refs/heads/main@{#39888}
The same observer implementation was being used for both client and
server but the role is different (sender vs receiver), so I split
the functionality up into two separate classes.
Bug: webrtc:11547
Change-Id: Ia60ab96fb86b4ff61fa7bff5f30d59b6fe0f9746
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300742
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39886}
This removes frequent output (typically 3 times in a row):
"RED codec red is missing an associated payload type."
Bug: none
Change-Id: Ie7e0f344209cb01f9730960a6fec9d5987eaadfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301720
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39885}
This reverts commit 298313534df2420e079ffc6fc9c6019d01d29a88.
Changes from the original commit:
* Call OnTransportClosed() from TeardownDataChannelTransport_n()
(same as before the original commit)
* Not call OnTransportClosed() from OnTransportChanged() when its
called with nullptr (also preserving the behaviour from before
the original commit).
Original change's description:
> Revert "Add param to DCC::SetupDataChannelTransport_n, simplify DCC* setup code."
>
> This reverts commit 2ec6a6c57830e06f601607c1b9473ad821b57e07.
>
> Reason for revert: It breaks WPT tests (e.g. https://ci.chromium.org/ui/p/chromium/builders/try/linux-rel/1361972/overview) blocking the roll into Chromium.
>
> Original change's description:
> > Add param to DCC::SetupDataChannelTransport_n, simplify DCC* setup code.
> >
> > * DCC = DataChannelController.
> >
> > * Consolidate steps to set the mid and transport name. They're now
> > set at the same time and without a separate PostTask.
> > * Transport sink is now consistently set in DCC
> > * Order of notifications for setting up the transport is now the same
> > regardless of the first time the transport is being set or if it's
> > being replaced.
> > * Made set_data_channel_transport() private.
> >
> > Bug: webrtc:11547
> > Change-Id: I39e89c6e269e6f06d55981d7944678bf23c8817a
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300562
> > Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#39859}
>
> Bug: webrtc:11547
> Change-Id: I0d8d7453b71be80fbf1b7eba7d161336e29de091
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301360
> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Cr-Commit-Position: refs/heads/main@{#39864}
Bug: webrtc:11547
Change-Id: I8ebbc3d3a12786dff2096350a77e03e98466ff00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301702
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39884}
This helper class is no longer used.
Bug: webrtc:14502
Change-Id: I7940de762ebb9a7c6d04927603f249f5b0061051
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39880}
Fully launched on mobile and approved for chrome launch.
Removed tests for feature enablement with field trial.
Bug: webrtc:14334
Change-Id: I7ca7183ff618835fef8c820cfd52863e1c7fa25e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301163
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Cr-Commit-Position: refs/heads/main@{#39879}
This reverts commit bd46bb7660fbc9f31799196058f7a1957f50aa31.
Reason for revert: There is a slight performance degradation
pointing to this CL, so revert this to be able to confirm if
it is the culprit.
Original change's description:
> dcsctp: Support zero checksum packets
>
> If configured, the packet parser will allow packets with
> a set checksum of zero. In that case, the correct checksum
> will not even be calculated, avoiding a CPU intensive
> calculation.
>
> Also, if specified when building a packet, the checksum can
> be opted to be not calculated and written to the packet.
> This is to be used when draft-tuexen-tsvwg-sctp-zero-checksum
> has been negotiated, except for some packets during association
> establishment.
>
> This is mainly a preparation CL and follow-up CL will enable
> this feature.
>
> Low-Coverage-Reason: Affects debug logging code not run in tests
> Bug: webrtc:14997
> Change-Id: I3207ac3a626df039ee2990403c2edd6429f17617
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298481
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Victor Boivie <boivie@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39737}
Bug: webrtc:14997
Change-Id: Ie22267abb4bcd25d5af07875eb933c51ed5be853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301580
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39878}
This move further clarifies that the file and its class are deprecated. It also cleans up the modules/video_coding root folder a bit.
No functional changes are intended.
Bug: webrtc:14876
Change-Id: Ic3ac439b3dd3492e6c9c85869efa80a6708658ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301521
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39876}
Before this change we first released output frame buffer in the code path which prepends config buffer to a keyframe and then called getOutputFormat(index). getOutputFormat(index) throws an exception if index points to a released buffer. This change rearranges calls such that getOutputFormat(index) always executed before releaseOutputBuffer(index).
Bug: webrtc:15015
Change-Id: Ia64f5d8e7483aeeb316d1a71c0cb79233f4bbee9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301364
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39874}
native_test_jni_onload depends on base_jni which depends on modules/audio_processing:api. This requires to include audio_device_java in pure video targets like video_codec_perf_tests.
Bug: webrtc:14852
Change-Id: I5e7b102fd730801562695bf3f4d5170ec8e59b58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/301363
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39873}
MaybeWorkerThread* GetWorkerQueue() and is removed.
Instead all work is expected to be done on the taskqueue used when
creating the RtpTransportControllerSend.
Bug: webrtc:14502
Change-Id: Iedc30efb8de7592611d6d3c5b5c6cd33c17a60c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300867
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39872}
