Access to `internal_dtls_transport_` only occurs on the network thread
and doesn't require locking. Access to `info_` still requires a lock
but writing to it only occurs on the network thread. If reading from
the network thread is needed, that could be done without requiring
the lock.
The scope of holding the lock is much smaller now.
Bug: none
Change-Id: Ic284df04196dfcf8b77c66a48e484ca6893de050
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325283
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41387}
by also taking into account any a=msid: line in addition to
msid-semantic. Also document issues with msid-semantic generation and unify support determination by removing the msid_supported flag.
BUG=webrtc:10421
Change-Id: Icea554ebd1998f2b526846457029eff6854a772a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329760
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41386}
which come from the a=fmtp:<pt> lines in the SDP and were used as either
std::map<std::string, std:string>
with three aliases,
cricket::CodecParameterMap
SdpAudioFormat::Parameters
SdpVideoFormat::Parameters
Use webrtc::CodecParameterMap in all places.
BUG=None
Change-Id: If47692bde7347834c349c6539b43309d8770e67b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41375}
instead of throwing an error when trying to pick a send codec.
BUG=webrtc:15145,webrtc:4957
Change-Id: I056b145c093348576e1aeaf5def50d5414f2de70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330122
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41360}
This adds neccessary checks for SDP negotiation with HEVC.
Test: Manually apply the CL on Chromium and enable HEVC HW encoder,
and add HEVC profiles in rtc video decoder/encoder factory, H265 is
negotiated in SDP with correct FMTP lines added.
Bug: webrtc:13485
Change-Id: I5557b20b646cc96c5acb578521204fe10df0dcf0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330202
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#41357}
as preparation for H265 work.
BUG=webrtc:15703
Change-Id: Ib6e0afa5ccbb8172a70d4e4eb876639559070fd6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329981
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41350}
This is temporary and should be re-enabled as soon as the test is
fixed.
Bug: webrtc:15722
Change-Id: I9d262c9931a19bc9c33f7f93e9e275d39fab403c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330561
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41348}
This allow exernal applications to control how many packets can be sent relative current BWE.
This is a partial revert of https://webrtc-review.googlesource.com/c/src/+/311102
Bug: chromium:1354491
Change-Id: Ia236aaacc468ddac12341efa555041bb2dfdde62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330580
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41343}
This reverts commit 33c7edd58ad0edc71939b9372fff3ab563c1f4a7.
Reason for revert: Breaks downstream project
Original change's description:
> Enable DD and VLA header extensions by default for Simulcast/SVC
>
> When Simulcast (more than one encoding) or SVC (a scalability mode
> other than the default L1T1) is used, enable the AV1 Dependency
> Descriptor and the video-layer-allocations RTP header extensions by
> default.
>
> The RTP header extensions API can be used to disable them if needed.
>
> BUG=webrtc:15378
>
> Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Cr-Commit-Position: refs/heads/main@{#41332}
Bug: webrtc:15378
Change-Id: I6b5f71f321d30a510db3bd180deaa57732f9349b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330540
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41341}
When Simulcast (more than one encoding) or SVC (a scalability mode
other than the default L1T1) is used, enable the AV1 Dependency
Descriptor and the video-layer-allocations RTP header extensions by
default.
The RTP header extensions API can be used to disable them if needed.
BUG=webrtc:15378
Change-Id: I587ac32c9d681461496a136f6950b007e72da86d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41332}
Now that Chromium has migrated to the new name[1], "decode_metronome",
we can delete the variable with the old name, "metronome".
[1] https://chromium-review.googlesource.com/c/chromium/src/+/5093942
Bug: webrtc:15704
Change-Id: I50fef88a692d83e37af10956b2e12389fa601662
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330300
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41331}
ConntectionContext now keeps and expose field trials as part of the
Environment, and do not need to be aware about field trials specifically
Bug: webrtc:15656
Change-Id: Ib78694a65a9ca7c8bf273eeaf9334323ddb841c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329420
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41328}
This function isn't used anymore.
Bug: webrtc:9987
Change-Id: I37f1c86cc4802950347db302e8a9207b9dd370bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330261
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41327}
Replace CallFactory class with a factory function
Bug: webrtc:15574
Change-Id: Ib1d8cff8d7550da3af01693a7bc117a7bd342258
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330000
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41321}
In preparation for experimentally supporting different types of
metronomes and metronome use cases we'd like to rename for clarity.
This is the first step, which introduces the new name and prefers it if
it is set, but keeps the old name for backwards compat reasons.
Once Chromium has migrated to the new name, we can delete the old name.
Bug: webrtc:15704
Change-Id: I23077bf2415ebb2b2338320c9a14e3bd17d3abb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/330020
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41319}
To be submitted after downstream usage has been removed, but no earlier than December 1, 2023.
Bug: webrtc:12598
Change-Id: Id9acbac591c48c0c5883fe8f06cf6a68471b70f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323004
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41290}
Now that it is used through the environment both for creating MediaEngine and for creating Calls.
Bug: webrtc:15656
Change-Id: Ib95ee46fe08d9d1ed1ef96bd67189e98052599ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329202
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41286}
This reverts commit 1ae700a9233ed647e1b4080c0fcb48f61a0cca0a.
Reason for revert: Potential root cause of crbug.com/1504351
Original change's description:
> Make Codec::Matches also consider packetization
>
> If it's not considered it can lead to payload IDs erroneously being
> reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
>
> Bug: webrtc:15473
> Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41153}
Bug: webrtc:15473 chromium:1504351
Change-Id: I87fb671d76c3b17beb65124603cc040bb9bf4fa5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329201
Commit-Queue: Tony Herre <herre@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41285}
This allows both the signal and the callback to be used.
Bug: webrtc:11943
Change-Id: I89460126d415520295c7e7d4ee440156a6e9e5ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329140
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41282}
This is done as a preparation to move RtcEventLog ownership into Environment where destruction happens later, when all users of the Environment are deleted.
Bug: webrtc:15656
Change-Id: I2a72c74f1fabb1e25c5200aa47a5d61e4b3d9cd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41272}
PacingController per default use a burst interval of 40ms. The behaviour can still be overriden by using the method SetSendBurstInterval.
Bug: chromium:1354491
Change-Id: Ie3513109e88e9832dff47380c482ed6d943a2f2b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/311102
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41254}
with an one-user callback.
Bug: webrtc:11943
Change-Id: Ia61c7811f0058fa7238d47ef13fadfd547f052ce
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328900
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41250}
with a callback function.
Bug: webrtc:11943
Change-Id: Ieed740a36f86be6dd45d6a495cc4fd023ea98477
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328862
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41249}
This poison guards against accidental use of EnvironmentFactory and thus ensures low level WebRTC class would use utilities from propagated environment instead of accidentally using a default implementation.
This poison extends and thus replaces default task queue poison.
Bug: webrtc:15656
Change-Id: I577bef8af08b9c7dd649ad5a2284eb236e6f4a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328380
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41247}
and unify algorithms a bit more.
BUG=webrtc:15214
Change-Id: Ie9903f3e56d25b1dc026367e8ae6817275faa07b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328442
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41244}
since the offer/answer rules do not depend on the media type for
the most part. Also make use of recently introduced Codec types.
BUG=webrtc:15214
Change-Id: Ieae27247a8910c3fcaa9609dca0297985907f86a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327740
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@{#41221}
StunServer is updated to ensure registring for receiving packet from the socket is happening on the same thread as where the packets are recevied.
Bug: webrtc:15368, webrtc:11943
Change-Id: I94cc3a47278d5489de7f170c8d43015d1551c437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328120
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41219}
at time of connect. This may allow deprecating SHA-1 which
is no longer used by browsers and not supported by the JS
API.
BUG=None
Change-Id: Iae1d800a61d46e0dcdb622ccb009acc6fb7db53f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327540
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41187}
since the algorithms are the same, the codec-specifics have been
separated out in the previous CL.
Manual rebase of https://webrtc-review.googlesource.com/c/src/+/326101
BUG=webrtc:15214
Change-Id: Ic3effe63f3af2ccc22f326d1685f6f55afa2db52
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327420
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41181}
If it's not considered it can lead to payload IDs erroneously being
reused if the SDP is munged, see https://crbug.com/webrtc/15473#c10.
Bug: webrtc:15473
Change-Id: I195a06d556e8a57dbeeb946effc4e0f27cc930b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326522
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41153}
- Adds namespaces which allows removing redundant using declarations and
static keywords.
- Adds some new using declarations and make sure they are all fully
qualified.
- Use std::tuple instead of testing::tuple.
Bug: None
Change-Id: If2ca3c0b94efc8a96da8d2dada0585370e85a7e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/326520
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41151}
This test drives the new tools_webrtc/remove_extra_namespace.py tool.
Bug: None
Change-Id: I9b590aa1213e4cace2d64d555f4dafd893f03606
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327021
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41141}
These functions had no callers and no tests.
Under YAGNI principles, they need to be deleted.
Bug: None
Change-Id: I8b5d74678b804ef2be70409d05a5237f1637eaea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/327024
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41134}
This uses libSRTPs srtp_remove_stream()
https://github.com/cisco/libsrtp/blob/main/include/srtp.h#L597
method to remove SSRCs from the libSRTP session when they are removed
from the RTP demuxer. This works even when the stream was added
automatically via the ssrc_any_inbound mechanism.
Only streams for inbound SSRCs that were added explicitly via SDP negotiation are removed.
Guarded by WebRTC-SrtpRemoveReceiveStream field trial.
BUG=webrtc:15604
Change-Id: I655bde5f8ddf26ac91395ef54bd1b3c598813380
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/324720
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#41105}
