A call to GetScalabilityMode was added for logging purpose and causes an expectation failure for tests using 4 temporal layers.
Plan is to remove the old GetScalabilityMode and keep only the one that returns an optional.
Change-Id: I0e37a496bb621d9754d6572ef5838b58193aa183
Bug: b/327381318
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341520
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#41838}
Left in target are just .cc files with .h files used externally.
Bug: webrtc:14775
Change-Id: I264f69bb29147fc0f8db877e3def8b21ed42181d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341420
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41835}
describing video codecs with their parameters as static members of SdpVideoFormat:
static const SdpVideoFormat VP8();
static const SdpVideoFormat H264();
static const SdpVideoFormat VP9Profile0();
static const SdpVideoFormat VP9Profile1();
static const SdpVideoFormat VP9Profile2();
static const SdpVideoFormat VP9Profile3();
static const SdpVideoFormat AV1Profile0();
static const SdpVideoFormat AV1Profile1();
This removes the need to craft instances of these by hand.
BUG=webrtc:15703
Change-Id: I2171e08b48ec98f18424f53f3b5d6d148130532e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337441
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41833}
Announce that we support SPA_DATA_DmaBuf and tell PipeWire not to map
memory for us so we can handle it ourself, similar like we do in case of
screen sharing. This fixes an issue when a camera is already in use by
gstreamer (pipewiresrc), where DMABufs are used, and we try to share
same camera and get no content, as PipeWire doesn't want to mmap DMABuf
memory for us and we get NULL data pointers.
Firefox bug: https://bugzilla.mozilla.org/show_bug.cgi?id=1876895
Bug: webrtc:15654
Change-Id: I788d8d12b2fcd5588329d7265e45b479f74bb628
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338921
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41826}
Instead of using PacketTransportInternal::SignalReadPacket.
Bug: webrtc:15368
Change-Id: Icdc2d7f85df6db944f0ba0232891e6c5a8986a66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340440
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41823}
The previous default size was 256kB.
The increase reduces packet loss at very high/bursty receive rates.
Bug: chromium:41485050
Change-Id: I2cf24b14e704bfd855701461afd3060ac078df70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340340
Auto-Submit: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41820}
adds a unit test for
https://webrtc-review.googlesource.com/c/src/+/340322
which is a single m-line variant of the original
fiddle that does not require renegotiation
BUG=chromium:326493639
Change-Id: Icc5ebb1dda6502b00828a77e13b9f5fc865d34c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340500
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41818}
Marking capturer as failed will indicate consumers will not be getting
any new frames by sending back ERROR_PERMANENT and let them know that
screencast can be stopped from their side. This will make screencast to
stop when a window we share is closed or when screencast is closed from
system tray.
Bug: chromium:40276865
Change-Id: Ia2c13461bd3126cab9c4838b8aa6840578562e9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339560
Commit-Queue: Jan Grulich <grulja@gmail.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#41817}
This exports neccessary API as dependency for h.265 parameter sets tracker to be submitted at CL:5307256.
Bug: webrtc:13485
Change-Id: I042599472f17d12ece4fa862c3715497502a5d76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340004
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41814}
This reverts commit d99499abbae94793a02944a1f28f7015816447f5.
Reason for revert: Breaks downstream projects and I can also repro locally when running the rtc_unittest test target (it does however pass in isolation indicating test cleanup/setup needs to be fixed)
Original change's description:
> p2p: separate ICE tie breaker and foundation seed
>
> BUG=webrtc:14626
>
> Change-Id: I189a708192c9cef0b50c3fcbe798b30376d3b547
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/338982
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41806}
Bug: webrtc:14626
Change-Id: If45f8a33395c562c9388b3d3748e8566efa87ecb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341081
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Christoffer Dewerin <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Christoffer Dewerin <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41812}
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.
Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
This is a reland of commit 1cce1d7ddcbde3a3648007b5a131bd0c2638724b
after updating the WPT that broke on Mac.
Original change's description:
> Make setCodecPreferences only look at receive codecs
>
> which is what is noted in JSEP:
> https://www.rfc-editor.org/rfc/rfc8829.html#name-setcodecpreferences
>
> Some W3C spec modifications are required since the W3C specification
> currently takes into account send codecs as well.
>
> Spec issue:
> https://github.com/w3c/webrtc-pc/issues/2888
> Spec PR:
> https://github.com/w3c/webrtc-pc/pull/2926
>
> setCodecPreferences continues to modify the codecs in an offer.
>
> Also rename RtpSender::SetCodecPreferences to RtpSender::SetSendCodecs for consistent semantics.
>
> BUG=webrtc:15396
>
> Change-Id: I1e8fbe77cb2670575578a777ed1336567a1e4031
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328780
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41719}
Bug: webrtc:15396
Change-Id: I0c7b17f00de02286f176b500460e17980b83b35b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/339541
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41807}
which needs to be added to the remote codecs a=fmtp:
This also forces SimulcastCastEncoderAdapter to avoid issues with codecs that have native simulcast capability but do require synchronized keyframes.
This parameter allows for large-scale experimentation and A/B testing
whether the new behavior has advantages. It is to be considered
transitional and may be removed again in the future.
BUG=webrtc:10107
Change-Id: I81f496c987b2fed7ff3089efb746e7e89e89c033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/333560
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41805}
Moved to cc file to fix link issue when linking with dynamic library
(crd).
Bug: webrtc:15368
Change-Id: I51cefcd439fda93d1135fcffa75198ab680e8583
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340302
Reviewed-by: Jonas Oreland <jonaso@google.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41801}
Goal is to make PeerConnectionInterface methods pure virtual.
This is a split of cl https://webrtc-review.googlesource.com/c/src/+/340143 in order to be able to fix Chromium test RTCPeerConnectionHandlerTest.OnRenegotiationNeeded
Bug: none
Change-Id: I5eac4d9a96c1b594c9e2b3505ef2466046065dc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340481
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41798}
Plan is to replace SignalReadPacket with this new method and have clients register as listeners.
rtc::ReceivedPacket is ammended with Descryption information so that receivers can know how to treat a received packet.
This will replace the current "flag". Also see https://webrtc-review.googlesource.com/c/src/+/340301/3
Bug: webrtc:15368
Change-Id: I9ea1f34e8b1e923d67c2e92e36a22b3dd10dbd73
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340181
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41795}
A fingerprint should not be required for m-lines which are rejected.
BUG=chromium:326493639,webrtc:11066
Change-Id: I7428c91a144ca46650e13d72868f160652a98339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340322
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@{#41794}
