So applications don't need to create and inject their own instance of
BasicPortAllocator, just to change these settings.
Bug: webrtc:13145
Change-Id: I08ac8658b4c0ef87019fa579be9195a8a6b50feb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239643
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35476}
This patch adds reporting of relay protocol,
i.e how a client connect to the turn server.
This is added in the old stats api...cause there
are clients still using it.
Bug: none
Change-Id: Iac7fe3e3de0ba42d5897c304ebbae368edf498fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239640
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35469}
the maximum used in practice is multiopus with
6 or 8 channels. 24 is the maximum number of channels
supported in the audio decoder.
BUG=chromium:1265806
Change-Id: Iba8e3185a1f235b846fed9c154e66fb3983664ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@nvidia.com>
Cr-Commit-Position: refs/heads/main@{#35440}
This was the behaviour before https://webrtc-review.googlesource.com/c/src/+/218605,
and is currently relied upon by Chrome to mute received audio tracks
by default, until they should be played out.
Bug: chromium:1272566
Change-Id: I8a288a287e7c01392f4af1db5b083e8d7ee7b2a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238665
Commit-Queue: Tony Herre <toprice@chromium.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35401}
This is a partial revert of commit f9e502d935fa9ca8d9cc7540680021b93d285f28.
Reason for revert: Functionality turns out to be needed by some partners for some months more.
Original change's description:
> Remove enable_dtls_srtp option
>
> This is part of the removal of support for SDES.
>
> Bug: webrtc:11066
> Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35262}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:11066, chromium:1271469
Change-Id: I79a90f025e53816789b391bc52a0e896b9be87e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238170
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35378}
they should be assertions and just decrease the readability of the actual test.
BUG=None
Change-Id: Ib6b23246fe03c8058e721cd63e887c0cd92a32e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237806
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35358}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
This makes relay candidates identifiable in scenarios
where the TURN server is behind another entity and the peer
sees a different ip address:
client -> turn -> relay address -> third party relay + address -> peer
In those cases, the relay candidate will become peer-reflexive
since the peer sends the third party relay's address in the xor-mapped
address and it is currently not easily possible to determine this is a
relay candidate anymore.
BUG=webrtc:13392
Change-Id: I6787339d0abdc735f8a43f636a676cccd8cadcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237561
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35346}
this could happen if setCodecPreferences is used to prefer
red over opus as it is done for red+opus.
BUG=webrtc:13287
Change-Id: I3d61cd8f1a364572bc531a75dcc239c3919138cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237800
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35344}
which is the maxium length allowed by one-byte header extensions
BUG=webrtc:12517
Change-Id: I003105d3566a34b5b7affb84ffe69b7705973ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35333}
which was not checking this candidate
BUG=webrtc:7063
Change-Id: Ic2b9a27cfa842bcacd434d171a919515aafb2312
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237560
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35332}
Also switch to "LT(meaured, limit)" format as this is easier to read.
There have been bot runs that exceeded the old limit.
Bug: webrtc:13354
Change-Id: I1c19c98e1c1777177e7dc14ae4679765c1c40550
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237342
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35317}
close to the codec they relate to as it is done by video_engine.
This does not change functionality but improves readability of the SDP
BUG=webrtc:13287
Change-Id: I9b03cd3131eaa932ffed3fb4e66cbf55faedcdd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235665
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35311}
According to https://w3c.github.io/webrtc-pc/#datachannel-send it should
return an error, definitely not close the data channel.
While we should probably return an RTCError will better information, this
would break the API and will be done later.
Bug: webrtc:13289
Change-Id: I90baf012440fbe2a38a826cf50b50b2b668fd7ff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35306}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
Changes one preexisting enum-to-string function to use the new format.
Also changes the RTC_LOG macros that created collisions with ToString,
for tidiness, and documents the recommended function form.
Bug: webrtc:13272
Change-Id: Ic8bb54ed31402ba32675b142d796cf276ee78df5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235722
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35296}
This field is unused within WebRTC, and doesn't seem to
be essential for any existing customers.
If this works well, it will be deprecated and removed.
Bug: none
Change-Id: I96d7485e4d094abfa6a8c3d1e6855834c13dedd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189680
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35263}
This is part of the removal of support for SDES.
Bug: webrtc:11066
Change-Id: I448d0e0032672c04c87b00550ab4b9d792071a0b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234864
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35262}
As suggested in another review.
Also add one more guard, and some commentary.
Bug: None
Change-Id: I9b84453ff2533fe01d157fe84f07405d352e1dc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235820
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35245}
These seem to have been forgotten when modifying
sdp_offer_answer.h. It's nice to be consistent.
Bug: none
Change-Id: Iffc4acbc48c0052141e029dcff4faebedbb22784
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235726
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35243}
The queued_send_data_ packet queue contains the actual data and has an
efficient byte_count() accessor. It removes the need to do some manual
accounting on the side.
Bug: webrtc:13288
Change-Id: Ie6bc39c344186160c630bcf337631614c6d9ee10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235372
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35227}
Previous limits was only in a comment and users had no way to query it
from the API.
Bug: webrtc:13289
Change-Id: I6187dd9f9482bc3e457909c5e703ef1553d8ef15
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235378
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35224}
for created and connected connections. This will allow making
an informed choice for
https://webrtc-review.googlesource.com/c/src/+/234867
BUG=webrtc:13265
Change-Id: Ica3bb7695f53403e481ab1ea2a78fa2719fe44a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234867
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35203}
This change
- adds new type VideoTrackSourceConstraints expressing min/max FPS
constraints.
- adds new method VideoTrackSourceInterface::ProcessConstraints.
- adds new method VideoSinkInterface<>::OnConstraintsChanged.
- updates AdaptedVideoTrackSource and VideoBroadcaster to forward
the constraints to sinks.
- adds several unit tests for the added functionality.
- and finally, implements OnConstraintsChanged in VideoStreamEncoder.
Chromium will be updated in coming CLs to supply constraints set
through the MediaStream module.
go/rtc-0hz-present
Bug: chromium:1255737
No-Try: true
Change-Id: Iffef239217269c332a1aaa902ddeae2440929e22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235040
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35197}
First CL to try to understand the extent of the cleanup needed in
order to remove -Wno-shadow and follow Chromium on enabling this
diagnostic.
Bug: webrtc:13219
Change-Id: Ie699762da50fe3dbc08b1fd92220962d4b7da86b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35134}
Used by QueryDefaultLocalAddress, instead of relying on the update
thread's associated socket server.
This is not the only use of rtc::Thread::socketserver() in the
BasicNetworkManager class. It also interacts with the thread's
socket server to call set_network_binder. That is unchanged by this cl,
perhaps those calls can be moved to the caller of StartNetworkMonitor and
StopNetworkMonitor.
Bug: webrtc:13145
Change-Id: If109c2dcb0e74b183e10bb3db7a5aefcc95d1a8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232613
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35118}
along the lines of RTX handling but with limited support for missing
fmtp lines because of video/red.
BUG=webrtc:13178
Change-Id: Ia866c0e857da6da2ef1e4b81b51f90f534c7bb83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231948
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35107}
In callers where it's non-trivial to explicitly pass the right
SocketFactory, pull the call to rtc::Thread::socketserver() into the
caller, with a TODO comment.
Bug: webrtc:13145
Change-Id: I029d3adca385d822180e089f016c3778e0d4fd0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231227
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35063}
This makes it possible to invoke methods on the transceiver object
from any thread.
Also makes a few of the mock observer objects thread-safe, to allow
testing when the main thread is not the signaling thread.
Bug: webrtc:13183
Change-Id: Ic97efef71a21c3075700a028103061032f8d2bcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232120
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35010}
* Replace "AV1X" with "AV1";
* Keep mapping of "AV1X" payload name to kVideoCodecAv1 to not break
support of injectable "AV1X".
Bug: webrtc:13166
Change-Id: I9a50481209209f3857bbf28f4ed529ee6972377e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34987}
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}
This reverts commit 2c41cbae37cac548a1133589b9d2c2e8614fa6cb.
Reason for revert: The breaking test in Chromium has been temporarily disabled in https://chromium-review.googlesource.com/c/chromium/src/+/3139794/2.
Original change's description:
> Revert "Wire up non-sender RTT for audio, and implement related standardized stats."
>
> This reverts commit fb0dca6c055cbf9e43af665d3c437eba6f43372e.
>
> Reason for revert: Speculative revert due to failing stats test in chromium. Possibly because the chromium test expected the metrics to not be supported, and now they are. Reverting just to unblock the webrtc roll into chromium.
>
> Original change's description:
> > Wire up non-sender RTT for audio, and implement related standardized stats.
> >
> > The implemented stats are:
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-totalroundtriptime
> > - https://www.w3.org/TR/webrtc-stats/#dom-rtcremoteoutboundrtpstreamstats-roundtriptimemeasurements
> >
> > Bug: webrtc:12951, webrtc:12714
> > Change-Id: Ia362d5c4b0456140e32da79d40edc06ab9ce2a2c
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226956
> > Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#34861}
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> TBR=hta,hbos,minyue
>
> Bug: webrtc:12951, webrtc:12714
> Change-Id: If07ad63286eea9cdde88271e61cc28f4b268b290
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231001
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Olga Sharonova <olka@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#34897}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:12951, webrtc:12714
Change-Id: I786b06933d85bdffc5e879bf52436bb3469b7f3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#34930}
