This reverts commit 7dea26d8bb0fbb2f6fe25e74d2baac9293e413a8.
Reason for revert: breaks downstream
Original change's description:
> Move resources to resources/BUILD.gn.
>
> iOS bundle all resources in the same folder and some conflicts can arise from that.
> Having all resources in the same file makes it easier to reason about it.
>
> Change-Id: I37f420dfbd265ec644804e9d4c96515c83d2a992
> Bug: b/397385850
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377821
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43944}
Bug: b/397385850
Change-Id: I80788590498fc24709c95a6a9580fdad65860f8c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378280
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
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Cr-Commit-Position: refs/heads/main@{#43947}
iOS bundle all resources in the same folder and some conflicts can arise from that.
Having all resources in the same file makes it easier to reason about it.
Change-Id: I37f420dfbd265ec644804e9d4c96515c83d2a992
Bug: b/397385850
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/377821
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43944}
This CL fixes two issues with the old way targetBitrate was reported:
1. The target is per encoder, i.e. per SSRC, but the old way to report
it was per sender and was approximately the sum of all encodings'
targetBitrate in most cases.
2. The old value did not come directly from the VideoBitrateAllocation
and tended to be greater than the sum of all targets (don't know
why).
We know the old value was wrong and the new value correct because
the actual bytes produced by the encoder closely matches the configured
target, which wasn't always the case with the old metric implementation.
Tested with unit tests and manually in Chrome by going to
https://henbos.github.io/codec-quality/src/index.html and ensuring
target ~= actual bytes produced. It also matches the debug logging of
video_stream_encoder.cc.
Bug: webrtc:42225524, chromium:392424845
Change-Id: I7a6f69e053ebc3fd972c2c4b7712750e721c0acc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/376460
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43854}
Lists of codecs have a lot of cross references (RTX/APT and the like).
We should introduce functionality to verify that those linkages are correct
before modifying the handling of these.
This CL introduces the CodecList class, which can be extended to do
that verification. It is used by pc/media_session.cc, but inter-module
APIs are not changed in this version (they will be later).
Bug: webrtc:360058654
Change-Id: Ifd6313d0289cfa090e51ac28bc775265d18fe6f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371600
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43582}
This reverts commit e046787a5a80a9d292b3aec7e946644e025a2b95.
Reason for revert: Revised codec matching to fix issue.
Changes also back out some changes that should not have been
included (using PayloadTypePicker for codec list merging).
Original change's description:
> Revert "Use PayloadTypePicker for video PT assignment"
>
> This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
>
> Reason for revert: Broke internal tests.
>
> Original change's description:
> > Use PayloadTypePicker for video PT assignment
> >
> > This includes changes that change the order of codecs.
> > It is preparatory to doing late assignment of video PTs.
> >
> > Bug: webrtc:360058654
> > Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#43489}
>
> Bug: webrtc:360058654
> Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43490}
Bug: webrtc:360058654
Change-Id: I66b3b6bd657c66f8860c5e67a504266d7707f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370380
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43554}
dtls_transport will when detecting a new fingerprint
(e.g by usage of pranswer) signal DtlsTransportState::kNew.
When this happen, the dtls crypto state is lost, and
sctp should reconnect, srtp does this automatically
in current code base.
The existing behavior in dcsctp is that it will detect
peer sending an init, and reconnect. But any messages sent
between the dtls restart and the message arriving from the
peer will be lost.
This patch changes so that this case is gracefully handled by
a) letting dcsctp_transport listen to dtls state
this is big part of patch and involves changing the type of
the underlying dtransport from rtc::PacketTransportInternal to cricket::DtlsTransportInternal. If requested, I can put this
into a separate patch...
b) if a dtls restart happens, delete and restart socket.
Testcase that fails before patch and works after is attached.
Bonus: And include-what-you-use on patch
Bug: b/375327137
Change-Id: Ib78488ae75fd8aeb50d121adf464a33dabbf95e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367202
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43546}
removing the webrtc need for having sources in it.
BUG=webrtc:42226155
Change-Id: I40fbde9064f4fa629c7c6b0cf99f23ab1726da75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370820
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43540}
Set use_default_launcher=false in rtc_test on android
Bug: webrtc:42223878
Change-Id: If05da40b420d5da8f9e0f39560eb07380ebada14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368921
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43505}
This reverts commit e5048949b0fcc275264e24f3b2a4c658fcc84aa3.
Reason for revert: Broke internal tests.
Original change's description:
> Use PayloadTypePicker for video PT assignment
>
> This includes changes that change the order of codecs.
> It is preparatory to doing late assignment of video PTs.
>
> Bug: webrtc:360058654
> Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43489}
Bug: webrtc:360058654
Change-Id: I5c94a7bafa49bdf17f665480398707155e458d26
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370240
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
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Cr-Commit-Position: refs/heads/main@{#43490}
This includes changes that change the order of codecs.
It is preparatory to doing late assignment of video PTs.
Bug: webrtc:360058654
Change-Id: Id5ddaf94d4b9557c0502a373e42635108d8fdf26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366400
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43489}
This is a reland of commit 775639e930f14a619974944594b40c633cc574a3
Original change's description:
> Set default scalability mode for H.265 to L1T1.
>
> H.265 does not have software fallback, and it may have issue supporting
> more than 1 temporal layers on some devices. Set default to L1T1 when
> scalability is not configured, or if a scalability mode is reported as
> not supported by encoder.
>
> Bug: chromium:41480904
> Change-Id: I53895c45ec821d65774ffe2db5f418184e3fb02a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367835
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
> Cr-Commit-Position: refs/heads/main@{#43389}
Bug: chromium:41480904
Change-Id: Idedf6249130bd01dd31261672c624b88c3f4c1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368261
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43412}
This seems to have no effect on tests, so it appears that these were
not used after all.
The goal is to make transport-cc a media-section-level attribute.
Bug: webrtc:378698658
Change-Id: Ia20ca5b91472b02db30f911ad1a1892cf36cd682
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368440
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43411}
The std::lcm and std::gcd functions are part of the C++ standard
library. The existing functions are marked as deprecated rather than
deleted in the case of possible third party uses.
#rtc_cleanup
Bug: webrtc:377205743
Change-Id: I174e663f152d750c984a35dc7136bc18dc01bc8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367440
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@google.com>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43368}
Inactive encoders are included in the string when they are paused due to
bitrate allocation being 0 for that simulcast layer.
#rtc_ktlo
Bug: webrtc:376804631
Change-Id: I4234b452b60fee58981907380df41962fda5bf40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367660
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43367}
The split shows that some places don't need it at all. Most other
places will depend on both send and receive stream targets.
Bug: webrtc:373151158
Change-Id: I788136a2ee84180c16345a7929b7f7bf3f97507b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365460
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43230}
This is in preparation for making a matcher that checks the parameters
when all payload types come from the same number space.
Bug: webrtc:360058654
Change-Id: Ibcf4fee8d882eb0fa7f83faf0278bc6757761e18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365361
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43223}
Use the same code in PayloadTypePicker as in Codec.Matches()
Bug: webrtc:360058654
Change-Id: I549ed24860648cfdb6a173a19773daf01db827b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365102
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43217}
This CL is a pure move; later CLs will try to increase consistency
between the functions.
Bug: webrtc:360058654
Change-Id: I6662b3d35f8e2dab60c2778a4755454fe3029fe2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365100
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43210}
The `requested_resolution` API must not change aspect ratio, example:
- Frame is 60x30
- Requested is 30x30
- We expect 30x15 (not 30x30!) as to maintain aspect ratio.
This bug was previously fixed by making VideoAdapter unaware of the
requested resolution behind a flag: this seemed OK since the
VideoStreamEncoder ultimately decides the resolution, whether or not
the incoming frame is adapted.
But this is not desired for some non-Chrome use cases. This CL attempts
to make both Chrome and non-Chrome use cases happy by implementing the
aspect ratio preserving restriction inside VideoAdapter too.
This allows us to get rid of the "use_standard_requested_resolution"
flag and change the "VideoStreamEncoderResolutionTest" TEST_P to
TEST_F.
Bug: webrtc:366067962, webrtc:366284861
Change-Id: I1dfd10963274c5fdfd18d0f4443b2f209d2e9a4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362720
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43037}
This allows detecting if it has been set reliably.
0 is a valid payload type.
Bug: webrtc:360058654
Change-Id: Ic3646abe20d0247592145ad27549fa46ddb7ec90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43016}
This entails passing in a PayloadTypeSuggester as a dependency. PT suggesting is still done according to the old method, but with new code.
Bug: webrtc:360058654
Change-Id: I12a7d2aa6aa482fb62ff3dfb34b9761ebb7dddef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42989}
Since media/ and pc/ both have to use this, and both
depend on call/, this seems to be the right place to put it.
Also factor out the interface that media will use in a separate
interface class.
Bug: webrtc:360058654
Change-Id: I34acbecc618f23e19542ce4b0110d0e8ed9e55ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361281
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42933}
Will be later used to conditionally enable mixed codec simulcast
with a field trial.
Bug: webrtc:42220378
Change-Id: I527a488c04cd2b5a9f4ec703504b67943e966ab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361403
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42929}
The new type PriorityValue is a strong 16-bit integer matching RFC 8831
requirements that can be built from a Priority enum.
The value is now propagated and used by the SCTP transport, but enabling
the feature still requires a field trial for now.
Bug: webrtc:42225365
Change-Id: I56c9f48744c70999a8c2d01415a08a0b6761df4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357941
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42695}
Libvpx works without this, so the existing tests pass. However, other
encoder implementations (like rtc_video_encoder in Chrome) look at
different fields and get confused about the configuration.
Test: Integration tests with Chrome and windows hardware encoders.
Bug: webrtc:348342168
Change-Id: Id0d96cff34eb34c7e019a24255623f3aeeca5772
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355500
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42555}
and files that broke when I fixed the first set.
Bug: webrtc:42226242
Change-Id: I321cd63537ab3002098c7bdecd889a6fc5a1eb25
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353421
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42429}
This is a reland of commit 47bfe39ecfe45b2f94c616ace97949003d9e87b4
Original change's description:
> Split digest methods from ssl target into digest target
>
> in an attempt to break up the monolithic ssl target.
>
> BUG=None
>
> Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#42249}
Bug: webrtc:339300437
Change-Id: I31bb79bbc6cc55a2634176f95ec67de195974e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42304}
in an attempt to break up the monolithic ssl target.
BUG=None
Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42249}
To pass field trials to EncoderStreamFactory in FakeVideoSendStream and thus reduce dependency on the global field trial.
Bug: webrtc:10335
Change-Id: Iad32881c2d9158fe1d77f1b71f8d606374ea111e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346340
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42023}
This CL removes the send buffers (but not the receive buffer) from
SctpDataChannel and increases the send buffer in DcSctpSocket instead.
The reasons are:
1) Simplify the code. This additional buffering was strictly needed
before we migrated away from usrsctp, as that send buffer was very
limited in size (by design). But with the migration to dcSCTP, it's
no longer needed, so it just adds complexity.
2) Make `RTCDataChannel::bufferedAmount` correct. Before this CL, it
represented just the data buffered in SctpDataChannel, and not the
data accepted by the SCTP socket, but not yet put on the wire. This
makes it hard for clients to know when a message has ever been sent.
3) Better handle draining data on data channel close. While this is not
implemented in dcSCTP, having a single buffer makes this easier to
add.
While most of this CL is straightforward, the handling of bufferedAmount
in the signaling thread (in RTCDataChannel in Blink), is a bit special.
The number returned by `RTCDataChannel::bufferedAmount` is not what the
true value is inside the SCTP socket, but an eventual consistent view
of that value. When a message is sent, the value is incremented and:
- Before this change: When a message was put on the SCTP socket, the
view's value was decremented. Which made the view reflect what was
buffered outside the SCTP socket, and that buffering is now gone.
- After this change: SctpDataChannel will track what RTCDataChannel
will think it is, and provide updates to that number as we are
notified that it's reduced - by setting a "low threshold" callback
trigger.
A bonus with the new behavior is that it will be eventually consistent
and auto-heal also in error conditions - when messages are dropped due
to errors (bad input, bad state, etc). Previously, the bufferedAmount
value could drift away from the correct value on errors.
Note that a big chunk of unit tests were removed with this CL, as those
tested how the buffering behaved. Now, there is no buffering, so the
removed test cases represent a simpler interface.
This CL has been extensively tested with data channel benchmarks that
use the bufferedAmount thresholds (in Javascript).
Bug: chromium:40072842
Change-Id: I1a6a4af6b6e1116832f5028f989ce9f44683d229
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343361
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41945}
Also remove all dependencies on rtc_media_base except for a few
that are suspected of being linker directives.
Bug: webrtc:14775
Change-Id: Ic0daf88b5422047d3ed7079ee6af9e689853310c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341461
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41886}
The DcSctpTransport will soon use field trials to conditionally enable
some options.
And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.
Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
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}
