by storing
[[LastCreatedOffer]] / [[LastCreatedAnswer]]
which are similar to the W3C equivalent but as
description objects instead of serialized SDP strings.
While rejecting all SDP munging is not feasible, this lets us
measure and reject certain modifications gradually.
Chromium metrics CL:
https://chromium-review.googlesource.com/c/chromium/src/+/6089633
This is measured at three points during the lifetime of a peerconnection:
* for the first SLD call
* when the connection is first established
* when the connection was established and is being closed
Note that the "first" SDP munging detected is returned which may hide that something uses more than one modification.
BUG=chromium:40567530
Change-Id: I964e3ee6e75f73b777d90556fac8691a6f3dc27f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370680
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43741}
This reverts commit 9572b2fa5850da6d319b9efb5ee36290e2895f7f.
Reason for revert: Breaks internal tests
Original change's description:
> srtp: spanify Protect + Unprotect
>
> Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.
>
> Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.
>
> BUG=webrtc:357776213
> No-Iwyu: missing include is a private libsrtp header
>
> Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Cr-Commit-Position: refs/heads/main@{#43601}
Bug: webrtc:357776213
Change-Id: I5c36ecc2fd9ab672f61cd6b15398452cbd5e98a8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372200
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43608}
Makes SrtpSession and SrtpTransport use rtc::CopyOnWriteBuffer for the Protect and Unprotect operations instead of passing around void pointers.
Also updates the unit tests to use CopyOnWriteBuffer instead of char arrays with a fixed length.
BUG=webrtc:357776213
No-Iwyu: missing include is a private libsrtp header
Change-Id: I02a22ceb4e183e93c4ebd8c0a9c931404e0e32f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358442
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43601}
Demonstrates use of matchers and WaitUntil to have tests that are more
understandable during failure.
Drive by changes,
* Remove the `const` on RTCStats.id_ as to allow for the implicit copy
constructor.
* Add [[nodiscard]] to WaitUntil as it is not useful without checking
the return value.
Bug: webrtc:381524905
Change-Id: I379910ce0fc8d9d81c96b8f164aa5a040637c85a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370802
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43599}
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}
In particular that avoids lifetime issues with the field trials passed into peerconnection, as now PC takes field trials object by unique_ptr and thus fully manages its lifetime.
Bug: webrtc:42220378
Change-Id: Ia863e9703b5c76ae1866d0ff995b83286c0b947e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/371480
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43576}
This allows other tests using RTC stats to get pretty printing as well.
Bug: webrtc:381524905
Change-Id: Ib1eb9e1dad36b89e5b1c2ec687fcfeb308f82939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/370761
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43575}
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}
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 replaces the WaitUntilCondition function that was used in the
peer_connection_encodings_integrationtest previously. Along with that it
adds tests and improved error message printing.
As a drive-by, matchers were added for RTCError as these are the return
type of this utility function.
Bug: webrtc:381524905
Change-Id: If7ff18692396d3996b5b289f2d2c92520226003e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/369980
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43494}
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}
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}
The RTCP mode is a send property for both send and receive channels. Send properties should be configured based on what peers support/prefer, which is described by the remote description (content).
Bug: webrtc:340041654
Change-Id: I18cd59e98aecfbbd8f4919b98381836184c10d77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368980
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Lundqvist <tomasl@google.com>
Cr-Commit-Position: refs/heads/main@{#43449}
On a sendrecv m-line, the offered level-id represents the maximum that
can be both sent and received; on a sendonly m-line, the offered
level-id represents the maximum that can be sent; on a recvonly m-line,
the offered level-id represents the maximum that can be received.
Also according to RFC 7798 section 5, the highest level indicated by the
answer is either equal to or lower than that in the offer
Bug: chromium:41480904
Change-Id: I1729c8edc3aed0c00c41cea96204abafc37c002b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367322
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Cr-Commit-Position: refs/heads/main@{#43425}
The gn target for rtc_pc_unittests cleared the "configs" that is by
default set for rtc_test. Restore it back so we get RTC_ENABLE_H265
macro when rtc_use_h265 is configured.
BUG: chromium:41480904
Change-Id: If172482776e5be2ad99d976db12dcfa556ee8a24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368183
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Jianlin Qiu <jianlin.qiu@intel.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43403}
This will turn on RFC 8888 feedback messages if "ack ccfb" is negotiated.
This should eliminate the need for the "force" flag in the field trial.
Bug: webrtc:42225697
Change-Id: Iec7a894c244a417a8499200861550a33f89966a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367400
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43398}
Note: This still doesn't enable CCFB - it just completes the signalling.
Bug: webrtc:42225697
Change-Id: I2dfd346075f2adcc438588f592c8f735f4101c89
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367260
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43348}
To stress there is no intention to use each instance more than once.
Bug: webrtc:369904700
Change-Id: Id53ad804f39f8ee596ec0b45ff15393009fdfab0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366640
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43324}
with the exception of the legacy stats collector unittest
BUG=None
Change-Id: I1ef28ab2052b1194ec788fa69606418d42d5a433
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366020
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@{#43258}
This interface allows to delegate construction of AudioProcessing to
the PeerConnectionFactory where it can provide propagated field trials
Bug: webrtc:369904700
Change-Id: Ie05cd771e4a869fa5f43173e127256800ae0727f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365320
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43233}
which shows that a DtlsSrtpTransport can send and receive
from the SrtpTransport which extracts the key from its DTLS transport.
The SrtpTransport takes its keys from the DtlsSrtpTransport which
(by the way of encryption and decryption) ensures both sides agree
on the keys to use
BUG=webrtc:357776213
Change-Id: I605c6ae660eab5a53bef69bcf84d7e70a34d7be1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365274
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43231}
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 adds payload types to the codecs at the time when offer
is being generated, if they are unassigned at that point.
Bug: webrtc:360058654
Change-Id: I231ed057ebaf7fb0fffaf6ff5d600b064ba21f5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362282
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43033}
* IWYU export <sys/socket.h> from rtc_base/net_helpers.h.
* Add a presubmit check to ensures that <sys/socket.h> is included through net_helpers.h (expect if there is a IWYU pragma or a no-presubmit-check).
* Clean up existing includes of <sys/socket.h>
Change-Id: I4bc6cce045c046287f8f74f89edfc9321293b274
Bug: b/236227627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362082
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#42996}
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}
which broke since libsrtp included openssl/srtp.h instead of
its own srtp.h due to the order of include directories
BUG=webrtc:42234521
Change-Id: Idc5cba2114febd1e0835d201b6c23424a88e62d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360705
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42913}
- Modify munger to take (mutable)
std::unique_ptr<SessionDescriptionInterface> rather than
cricket::SessionDescription (that latter is embedded in the former)
- For all pranswer test cases, do a final SetRemoteDescription(kAnswer) and
check that signaling_state == stable
Add new test cases:
1) A test case that only applies it as prAnswer on caller (callee is stable)
2) A test case that "scramble" sdb between prAnswer and Anser.
Bug: None
Change-Id: Ifedd92ade01ae781a2e59d0569133c486c7093fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360781
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42891}
Normally (scaleResolutionDownBy) restrictions are applied at the source
which changes the input frame size which triggers reconfiguration with
appropriate scaling factors.
But when requested_resolution is used, encoder settings are by
definition not relative to the input frame size. In order for
restrictions to have an effect, they are applied inside
ReconfigureEncoder(): you get the minimum between the requested
resolution and the restricted resolution.
ReconfigureEncoder() happens when you SetParameters(), but the bug
here is that we don't do it again once the restrictions are updated.
So if restrictions are 540p when you ask for 720p, you get 540p and
after restrictions change to unlimited you're still stuck in 540p.
The fix is to also trigger ReconfigureEncoder() inside
OnVideoSourceRestrictionsUpdated() when the restricted resolution is
changing and a requested_resolution is configured.
To ensure reconfiguring the encoder "on the fly" like this does not
reset initial frame dropping logic, InitialFrameDropper caring about
input frame size changing is made conditional on not using
requested_resolution.
# Slow purple bots failing but they are not affected by this change.
NOTRY=True
Bug: webrtc:361477261
Change-Id: I1389aa16cf408b0d14e0b5b6f68c2442db955be9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360200
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42882}
Before this cl, ReadyToSend signaled false if sending a packet failed and transport->GetError() returns ECONN.
ECONN may be reported by the TCP connection (TcpConnection) if the remote closed the connection. TcpConnection will attempt to reconnect and should change the writable state if it fail.
Changing the state in the context of sending packets may cause recursive
calls and seems to cause problems with incorrect states.
It is simpler if RtpTransport::SendPacket ignore these failures and
upper layers treat these lost packets similar to if the packets had been
lost via UDP.
For safety, this change can be reverted by field trial WebRTC-SetReadyToSendFalseIfSendFail/Enabled/.
Bug: webrtc:361124449 b/359989715
Change-Id: I8e7016dfb4301862286215c4512aa8ac03a16685
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/360120
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42868}
e.g all files in the api/test folder not including subdirectories
Bug: webrtc:42226242
Change-Id: I18d74a18f8feec41eb252faa9acfffd1d6f45ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/359420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#42773}
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}
since it contains helpers mostly related to cryptographically secure random numbers and strings.
BUG=webrtc:339300437
Change-Id: I10db939534b25dc792ac1600a4721d1b84521880
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42441}
