Those never existed, were likely a copy-paste error in the spec
that we somehow inherited.
Bug: webrtc:11607
Change-Id: Ib4a038f061123e879f1099656273f6392f092213
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273485
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37953}
AV1 tests seem to be running fine now that we have the dependency
descriptor enabled, so remove the need for the RTP header extension
as it doesn't allow discarding frames.
Bug: webrtc:11607
Change-Id: Ifd0670ab61a5b69d0570f65ba30c352a31376992
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273488
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37952}
This reverts commit 021512b76a872b04e803d61f46c740ed363d641b.
Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.
Original change's description:
> rtpsender interface: make pure virtual again
>
> after providing default implementations in Chromium tests
>
> BUG=None
>
> Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37941}
Bug: None
Change-Id: I40f27c36819365fadae32032521f7e11184bee62
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
Owners-Override: Andrey Logvin <landrey@google.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37947}
This tests all existing codecs (VP8, VP9) with the depdendency
descriptor and adds the AV1 tests that requires it as well.
Placeholders for missing modes have been added for both VP9 and AV1.
Bug: webrtc:11607
Change-Id: Ie900bddc54ccbf4dcc466f3a7a6c8241906a243a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272807
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37906}
This simplifies the logic of what simulcast layers to signal, and avoids
situations where the upper layers get confused about which layers exist.
Bug: chromium:1350245
Change-Id: I9edeb93cbb30e872c4d3f3429a85a1fccf17996a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37905}
We should have done this a long time ago.
Let's do the same for stats_types.h in a separate CL because that file
is part of the api/ folder and needs some special care (typedefs and
temporarily include helper to avoid breaking downstream projects).
Bug: webrtc:14180
Change-Id: Id9c71ebd53dd97dd238bdf7527c36d7cf0e91f85
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37426}
This CL removes the use of the rtc::Thread::socketserver() method
in one place.
Bug: webrtc:13145
Change-Id: I1a1b2501450788263d5280c43e4328ade46f4146
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263320
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37340}
The tests check that the various scalability mode are supported
and the frames are marked properly by the encoder with their
spatial and temporal index.
The same information is then checked on the receiving side.
A new member is added on EncodedImage to store the temporal index,
and is filled by the encoders and retreived by the ref finder
objects on the decoding side.
Bug: webrtc:11607
Change-Id: I7522f6a6fc5402244cab0c4c64b544ce09bc5204
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260189
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37303}
https://w3c.github.io/webrtc-stats/#guidelines-for-getstats-results-caching-throttling
"When the state of the RTCPeerConnection visibly changes as a result of an API call, a promise resolving or an event firing, subsequent new getStats() calls must return up-to-date dictionaries for the affected objects."
BUG=webrtc:14190
Change-Id: I4560be22795f30e0369d573bda0100e490efb57b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265870
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37255}
Adds a function to PeerConnectionIntegrationBaseTest to stop and destroy
the caller and callee objects. This should take care of dangling pointers.
Before this change, the affected test would crash randomly - typically
detected within a few minutes of a gtest-repeat=-1 run.
After this change, it has not crashed in 15 minutes of running.
Bug: webrtc:12592
Change-Id: I9980f8974015bf2b2104fcb83c2ca0d677d03c3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264555
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37096}
This is a reland of commit ad6807805d12e48f11c3a68b4befaf8d7c23e8b5
Original change's description:
> sdp: reject duplicate codecs with the same id but different name or clockrate
>
> since something like
> rtpmap:96 VP8/90000
> rtpmap:96 VP9/90000
> or
> rtpmap:97 ISAC/32000
> rtpmap:97 ISAC/16000
> is wrong. Note that fmtp or rtcp-fb are not taken into account.
> Also note that sending invalid static payload types now throws an error.
>
> Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
>
> BUG=None
>
> Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#37028}
Bug: webrtc:14140
Change-Id: I63a37aacea6b9e0a9d7570b8422849275eb69aae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264544
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37066}
This reverts commit ad6807805d12e48f11c3a68b4befaf8d7c23e8b5.
Reason for revert: Speculative revert due to consistent Mac browser
test failures preventing WebRTC from rolling int Chromium:
https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/10410/overview
"Failed to parse SessionDescription. a=rtpmap:103 ISAC/16000 Duplicate payload type with conflicting codec name, clock rate or number of channels."
Original change's description:
> sdp: reject duplicate codecs with the same id but different name or clockrate
>
> since something like
> rtpmap:96 VP8/90000
> rtpmap:96 VP9/90000
> or
> rtpmap:97 ISAC/32000
> rtpmap:97 ISAC/16000
> is wrong. Note that fmtp or rtcp-fb are not taken into account.
> Also note that sending invalid static payload types now throws an error.
>
> Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
>
> BUG=None
>
> Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
> Cr-Commit-Position: refs/heads/main@{#37028}
Bug: None
Change-Id: Ic9c06c9309bb68bd94bfdd2e30ffd6ff96f6812b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37064}
This extends AlwaysValidPointer to avoid creating a unique_ptr inside it.
Bug: webrtc:13145
Change-Id: I73a4f18d0a7037b57f575b04b134e4f7eadceb79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263240
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37048}
since something like
rtpmap:96 VP8/90000
rtpmap:96 VP9/90000
or
rtpmap:97 ISAC/32000
rtpmap:97 ISAC/16000
is wrong. Note that fmtp or rtcp-fb are not taken into account.
Also note that sending invalid static payload types now throws an error.
Drive-by: replace "RtpMap" with "Rtpmap" for consistency.
BUG=None
Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#37028}
This is part of the project to delete the class entirely.
The CL also adds an "use_rtx" parameter to the function for listing
video codecs, rather than filtering those away afterwards.
Bug: webrtc:13931
Change-Id: I96b9b18c694a1c0986ccf22face76ef4c704d372
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262666
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36963}
This is a preparatory step in deleting the ChannelManager class.
Also delete some declarations whose implementation was previously removed.
Bug: webrtc:13931
Change-Id: I8764c00fa696932e79fcfe17550ef2490d6a1ed1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262804
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36923}
Since the lifetime of an SctpDataChannel is not strictly controlled
by its controller, the controller might go away before the channel
does. This CL guards against this.
Bug: webrtc:13931
Change-Id: I07046fe896d1a66bf89287429beb0587382a13a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36852}
This is in pursuit of an issue with another CL, but large enough
to be worth submitting separately.
Bug: webrtc:13931
Change-Id: If470488f092f8640d3a773922f6f0d22765b9e97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261728
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36833}
This reverts commit c48ad732d6eb69f14dd6d44f801d62997cef2c2f.
Reason for revert: breaks downstream project
Original change's description:
> Don't create channel_manager when media_engine is not set
>
> Also remove a bunch of functions in ChannelManager that were just
> forwarding to MediaEngineInterface.
>
> Bug: webrtc:13931
> Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36801}
Bug: webrtc:13931
Change-Id: I1e260a2489547bd9483b50e043c28d2805b0fa5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261660
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Owners-Override: Artem Titov <titovartem@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#36811}
Also remove a bunch of functions in ChannelManager that were just
forwarding to MediaEngineInterface.
Bug: webrtc:13931
Change-Id: Ia38591fd22c665cace16d032f5c1e384e413cded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261304
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36801}
Repeatedly open and close data channels on a peer connection
to check that the channels are properly negotiated and SCTP
stream IDs properly recycled.
Bug: webrtc:13994, chromium:1320194
Change-Id: I244911abb5abaf0a290de07a0d790cd1edffe8cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260984
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36780}
If media_engine is not passed in init parameters, the PC can't handle
media, but can be used for datachannels. This CL adds testing that
datachannels work without media engine, and adds failure returns
to PeerConnection APIs that manipulate media when media engine is
not present.
Bug: webrtc:13931
Change-Id: Iecdf17a0a0bb89e0ad39eb74d6ed077303b875c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261246
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36778}
This ensures that only the compilation units that actually need
ChannelManager details can see it.
Bug: webrtc:13931
Change-Id: Iddd37580c0ceceba5b7095e84b981e6a525b2800
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36762}
This also hides the existence of the classes VideoChannel and
VoiceChannel from anything that does not include "channel.h".
Bug: webrtc:13931
Change-Id: I080a692b6acfd5d2d0401ec20d59c3a684eddb05
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260944
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36746}
