This stores usage for all cases, making it easier to discover
abusive usages on unexpected patterns.
Bug: None
Change-Id: I62c9b07498e811ac04c221f57cfbc02312aaaacc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368902
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43442}
This is preparatory to ensuring that transport-cc gets turned off when
RFC8888 ccfb is negotiated.
Bug: webrtc:378698658
Change-Id: Ie76677bd6aa046701562bbd93d8489858488f863
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/368543
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43426}
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}
which launched a while back.
BUG=webrtc:40644399,webrtc:364825888
Change-Id: Ied1d76d8ab2cbb395e09c08f6354d99b4e082cef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367840
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@{#43383}
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}
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}
- Avoid redundant get() when dereferencing smartpointers
- Use const ref instead of copy for RtpExtension
- Use `.empty()` instead of `.size() == 0`
- Remove some unused using declarations
Bug: None
Change-Id: I0dfdc0dfdf165f153c9ba119c115cd492e9599fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43334}
These functions seem to have been unused except for tests.
It seems to have been added in 2017.
Bug: None
Change-Id: I01983f4b72456b1df27ec2d346014e0de1b5cae7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366943
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43332}
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}
It works in the same way as the first packet received callback and can be used for latency measurements.
One important detail is that RTCP and probe packets are excluded from triggering the callback.
Bug: b/375148360
Change-Id: I5f99b565f96b622e864669cf227be5534aab0fc7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366644
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43309}
This is a reland of commit 82617ac51e7825db53451818f4d1ad52b69761fd
The reason for the revert was a downstream use of
`rtc::VideoSinkWants::requested_resolution`, so in this reland we don't
rename this field, it's fine just to rename the one in
RtpEncodingParameters for now.
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
NOTRY=True
Bug: webrtc:375048799
Change-Id: Ic4ee156c1d50aa36070a8d84059870791dcbbe5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366660
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43304}
This reverts commit 82617ac51e7825db53451818f4d1ad52b69761fd.
Reason for revert: Break downstream projects
Original change's description:
> Rename `requested_resolution` to `scale_resolution_down_to`.
>
> This is a pure refactor/rename CL without any changes in behavior.
>
> This field is called scaleResolutionDownTo in the spec and JavaScript.
> Let's make C++ match to avoid confusion.
>
> In order not to break downstream during the transition a variable with
> the old name being a pure reference to the renamed attribute is added.
> This means we have to add custom constructors, but we can change this
> back to "= default" when the transition is completed, which should only
> be a couple of CLs away.
>
> Bug: webrtc:375048799
> Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43300}
Bug: webrtc:375048799
Change-Id: Ie41723a39420e12e7b5b681d3d00ccd14f66b4b1
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366642
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#43301}
This is a pure refactor/rename CL without any changes in behavior.
This field is called scaleResolutionDownTo in the spec and JavaScript.
Let's make C++ match to avoid confusion.
In order not to break downstream during the transition a variable with
the old name being a pure reference to the renamed attribute is added.
This means we have to add custom constructors, but we can change this
back to "= default" when the transition is completed, which should only
be a couple of CLs away.
Bug: webrtc:375048799
Change-Id: If755102ccd79d46020ce5b33acd3dfcc29799e47
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366560
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43300}
This fixes the bug where scaleResolutionDownTo must be specified even
on inactive encodings (scaleResolutionDownTo is the JavaScript name for
what is called requested_resolution inside WebRTC).
Bug: chromium:375048792
Change-Id: I3206ef7de09eaba24a5b4305d888ec4904617e58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366522
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43292}
The const-ref result of .str() must be copied into the returned
value, whereas the result of .Release() can be moved.
Bug: webrtc:374845009
Change-Id: I3abc98be30ce9947127c7664f5ffa6846b772ea2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43288}
This is similar to MatchesRtpCodec but not an exact match of parameters, unspecified parameters are treated as default. Use IsSameRtpCodec for comparison when codec is configured via encodings.
Bug: b:299588022
Change-Id: I0ea800e50af6f5666e3e867a928e15b0aa044635
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365142
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43272}
and use uint8_t instead of unsigned char. Follow-up from
https://webrtc-review.googlesource.com/c/src/+/365274
BUG=webrtc:357776213
Change-Id: Ibc97e5cc85316ba69b4133b7f3c42e3afbdd7abd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365540
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43263}
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}
We should use the Timestamp type, rather then int64, to store timestamps. In https://webrtc-review.googlesource.com/c/src/+/365001/ an additional int64 timestamp was added (last_sender_report_timestamp_ms).
This CL fixes the new timestamp, as well as other similar timestamps in MediaReceiverInfo (last_sender_report_utc_timestamp_ms and last_sender_report_remote_utc_timestamp_ms).
Bug: webrtc:372393493
Change-Id: I0e473730e85a69ec595b421e2c3db920364008eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365641
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43248}
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 is a reland of commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d
with more backward compat which also fixes the off-by-one issue which caused wrong SRTP keys to be extracted.
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I5d43dc23f90ef630834fb400751979fcc5e18203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#43225}
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}
Add an environment clock timestamp to SenderReportStats and make it visible in rtc_stats_collector.cc. This make it possible to use the pc->GetConfiguration().stats_timestamp_with_environment_clock() flag to decide which timestamp to use when creating a RTCRemoteOutboundRtpStreamStats object.
This CL is the third (and possible the last) of a series of CLs that aim to replace the UTC timestamps in RTCStats objects to Environment clock timestamps. The other CLs where https://webrtc-review.googlesource.com/c/src/+/363946 and https://webrtc-review.googlesource.com/c/src/+/364782.
When Chromium and Google internal uses of RTCStats are updated to set the stats_timestamp_with_environment_clock configuration, the flag can be deleted.
Bug: chromium:369369568
Change-Id: Ic0b07d7b012505267bd6516f19a9ba90df4cafab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365001
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43206}
This reverts commit 65ae3245f9380e46b1d755f3f452ba63ab6cdf8d.
Reason for revert: breaks downstream compilation
Original change's description:
> Spanify SRTP key export
>
> and simplify the interface used as this is only used for exporting
> SRTP keys and passing arcane OpenSSL arguments around does not make
> much sense.
>
> BUG=webrtc:357776213
>
> Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
> Commit-Queue: Philipp Hancke <phancke@meta.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Florent Castelli <orphis@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#43198}
Bug: webrtc:357776213
Change-Id: I03ffcda3d6821718f355b243ce78a9c54b4036f3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/365062
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Jeremy Leconte <jleconte@webrtc.org>
Owners-Override: Jeremy Leconte <jleconte@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43202}
and simplify the interface used as this is only used for exporting
SRTP keys and passing arcane OpenSSL arguments around does not make
much sense.
BUG=webrtc:357776213
Change-Id: I9e5a94fe368b77975e48b6dd5ab6a2d2575d6382
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364521
Commit-Queue: Philipp Hancke <phancke@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43198}
I missed one timestamp in https://webrtc-review.googlesource.com/c/src/+/363946, meaning that the config flag that was added do not yet work for all timestamps in RTCStats objects. The RTCRemoteOutboundRtpStreamStats still has UTC timestamps even if the config flag is set.
I will solve this by saving both an UTC (existing) and env (to be added) timestamp, and then let rtc_stats_collector choose timestamp based on the value of the config flag (just like RTCRemoteInboundRtpStreamStats is done in the 363946 commit).
Before adding the new env_ timestamp I want to make this change. I rename the existing timestamp to show what epoch it uses (NTP or UTC). This will later make it clear which timestamp is which.
So this CL will make no logical change, just renaming members.
I only need to rename the last_sender_report_timestamp_ms, but opted to rename the remote timestamp as well, to be consistent with the naming convention I add in this CL.
Bug: chromium:369369568
Change-Id: Icfe7cf274995b39799e1478a1bb8cdf5134f0b16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364782
Commit-Queue: Olov Brändström <brandstrom@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43194}
According to RFC 6904 a header extension MAY be offered both encrypted and unecrypted. In the case when encryption is enabled the encrypted version SHOULD be used and vice versa. However, this is under the assumption that both peers actually offer the same extension header both encrypted and unecrypted. With this PR we tighten the negotiation rules to the encryption option SHOULD be the same both in the sender and receiver in order to not drop the extension. Especially, see test `TestOfferAnswerPreferEncryptedRtpHeaderExtensionsWhenEncryptionEnabled` and `TestOfferAnswerPreferEncryptedRtpHeaderExtensionsWhenEncryptionDisabled`.
Bug: chromium:40623740
Change-Id: I68c65a776fcf7be97aaf60a797594c4361a06800
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363940
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emil Vardar (xWF) <vardar@google.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43191}
More test coverage for previously fixed bug
https://crbug.com/webrtc/369654168.
Two tests are added:
1. LibvpxVp9Encoder unit test that 4:2:1 720p can be reconfigured to
singlecast (which is what happens for encodings[0] in the bug).
2. Integration test that 4:2:1 720p can change to 180p,360p,540p.
This is the exact same test as was added in [1] but using
requested_resolution instead of scale_resolution_down_by.
[1] https://webrtc-review.googlesource.com/c/src/+/363941
Bug: webrtc:369654168
Change-Id: I83456b9254c1c6f647586d340d0fe5864b5515c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/364200
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43185}
Add media config for using environment monotonic timestamps (i.e. not UTC) in RTCStats constructor, and implemented the usage of the flag.
Bug: chromium:369369568
Change-Id: Ia93d048742c28af201164fe7b2152b791bb6d0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363946
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Olov Brändström <brandstrom@google.com>
Cr-Commit-Position: refs/heads/main@{#43156}
