42cf83cc61
Update WebRTC code version (2022-05-19T04:04:47).
...
Bug: None
Change-Id: I8eff6cd60b9416f81b32a1a51ca56286fedb7919
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262736
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2022-05-19 05:49:10 +00:00
0359ba2225
stats: add frame assembly time stats
...
implements a total frame assembly time statistic that measures the
cumulative time between the arrival of the first packet of a frame
(the lowest reception time) and the time all packets of the frame have
been received (i.e. the highest reception time)
This is similar to totalProcessingDelay
https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame.
This statistic is useful for evaluating mechanisms like NACK and FEC
and gives some insight into the behavior of the pacer sending the
packets.
Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added.
Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as
totalAssemblyTime of type double
Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received.
Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible.
This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket.
framesAssembledFromMultiplePacket of type unsigned long
Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet.
For such frames the totalAssemblyTime is incremented.
BUG=webrtc:13986
Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Commit-Queue: Harald Alvestrand <hta@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36922}
2022-05-18 09:16:10 +00:00
0601db9a48
Rename ReceiveStream to ReceiveStreamInterface
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Bug: webrtc:7484
Change-Id: I41176a66b8399f6c8cf568630f2808eb95cf6247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262767
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org >
Commit-Queue: Niels Moller <nisse@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36917}
2022-05-18 07:26:50 +00:00
79dc0a223e
Update WebRTC code version (2022-05-18T04:01:44).
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Bug: None
Change-Id: Ib10b8b8236ab5905ed6c524d058a39a356eebebd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262732
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2022-05-18 05:16:30 +00:00
88af20356f
Use ProbeClusterConfig in BitrateProber from GoogCC
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Instead of using field trials in BitrateProber for probe duration, use values provided in ProbeClusterConfig from GoogCC.
Field trials are instead read in ProbeController.
To avoid having to do a thread jump for every ProbeClusterConfig, RtpPacketPacer interface is changed to RtpPacketPacer::CreateProbeClusters(std::vector<ProbeClusterConfig>
Deprecates field trial "WebRTC-Bwe-ProbingConfiguration"
Change-Id: I3991e4b54770601855a3af2d6a16678f11d41c31
Bug: webrtc:14027
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261265
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Per Kjellander <perkj@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36911}
2022-05-17 12:29:25 +00:00
641a1b11b6
Adopt absl::string_view in call/
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Bug: webrtc:13579
Change-Id: Ib616eb3372da341fafb55c23038182751b9da5a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262780
Reviewed-by: Erik Språng <sprang@webrtc.org >
Commit-Queue: Ali Tofigh <alito@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36910}
2022-05-17 12:00:45 +00:00
1331c1821c
Reland: Update local_ssrc without needing to recreate video streams.
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This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
This is a reland of commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca
with an additional fix in Patchset 2. Another problem turned out to be
in RTCPReceiver, which is fixed in:
https://webrtc-review.googlesource.com/c/src/+/262663
Bug: webrtc:11993
Change-Id: I63c7cf62a6dd50f88b491fea3ba866697552ef5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262665
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36907}
2022-05-17 10:59:54 +00:00
2e521f5a25
Update WebRTC code version (2022-05-17T04:04:40).
...
Bug: None
Change-Id: I323023722ba3d5d467439a57cc23cb9f16b08b95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262724
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2022-05-17 05:17:46 +00:00
c92ee5f3c3
Revert "Update local_ssrc without needing to recreate video streams."
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This reverts commit 16a8b25d809e4d4982f9fc4b4e973acd506b8bca.
Reason for revert: Checking if this is blocking the Chromium autoroller.
Original change's description:
> Update local_ssrc without needing to recreate video streams.
>
> This is comparable to this change done previously for for audio streams:
> https://webrtc-review.googlesource.com/c/src/+/222042
>
> Bug: webrtc:11993
> Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
> Cr-Commit-Position: refs/heads/main@{#36876}
Bug: webrtc:11993
Change-Id: I3a8d2f6a7e89b6784754d8e891a4e01479807c2d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262422
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
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Cr-Commit-Position: refs/heads/main@{#36892}
2022-05-13 22:30:44 +00:00
f3f3a61167
Remove legacy PacedSender.
...
The new TaskQueuePacedSender has been default-on in code since M97, and
there are no further usages of it that I can find. Let's clean this up!
The PacingController and associated tests will be cleaned up in a
follow-up cl.
Bug: webrtc:10809
Change-Id: I0cb888602939add953415977ee79ff0b3878fea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258025
Reviewed-by: Per Kjellander <perkj@webrtc.org >
Commit-Queue: Erik Språng <sprang@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36890}
2022-05-13 20:31:06 +00:00
16a8b25d80
Update local_ssrc without needing to recreate video streams.
...
This is comparable to this change done previously for for audio streams:
https://webrtc-review.googlesource.com/c/src/+/222042
Bug: webrtc:11993
Change-Id: Ic953f816a8f7c56d1c3dc9a16d85bef3696a663d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261960
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36876}
2022-05-13 10:08:54 +00:00
5ef44ac686
Update WebRTC code version (2022-05-13T04:03:44).
...
Bug: None
Change-Id: Ib73994a64c962eb9c05114bbf3e9bebd7979db0e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262265
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2022-05-13 04:53:33 +00:00
b3a99f6eb6
Update WebRTC code version (2022-05-12T04:02:41).
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Bug: None
Change-Id: Ied41819bce6f0224dce4596d26e4f621c7923dd9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262083
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2022-05-12 05:36:23 +00:00
a59311cd6f
Update WebRTC code version (2022-05-11T04:01:49).
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Bug: None
Change-Id: I38d87bceb2115a41682307f856c2c6a7d2bd13d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262003
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2022-05-11 06:30:13 +00:00
cf4ed1516e
Add GetRtpExtensionMap to ReceiveStream and remove GetRtpExtensions.
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GetRtpExtensions() is still used in one corner case for audio receive
streams, so GetRtpExtensions has migrated to AudioReceiveStream.
Updated FlexfecReceiveStream config management (incl. pass by value) and
now store an RtpHeaderExtensionMap in FlexfecReceiveStreamImpl.
Call GetRtpExtensionMap() from call.cc instead of constructing one on
the fly for each rtp packet (for video packets at least).
Bug: webrtc:11993
Change-Id: Id90ec5d43ea368f58edd6f17cb39d8c54aec641f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261800
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36839}
2022-05-10 13:50:31 +00:00
89d20a6889
Update WebRTC code version (2022-05-10T04:02:02).
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Bug: None
Change-Id: Iaa9e04d325236c4dd68e3f8d1f1873814f874a5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261844
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2022-05-10 05:12:30 +00:00
363e812f2d
Remove the VideoReceiveStream2::rtp() accessor.
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Instead offer accessors for the specific config values from the struct
that are needed at different times. The remote_ssrc and rtx_ssrc
properties maybe accessed from any thread, other properties have
stricter requiremets.
Bug: webrtc:11993
Change-Id: I3ff8527b13452c773fae1b2574f1e3fd2583b481
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261319
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Cr-Commit-Position: refs/heads/main@{#36823}
2022-05-09 20:25:29 +00:00
7a15ff3f14
Add a transport_cc() getter and remove rtp_config().
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Bug: webrtc:11993
Change-Id: Ie435a702c91b4d3827e528083f474e378fc75cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261318
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Reviewed-by: Niels Moller <nisse@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36822}
2022-05-09 20:21:14 +00:00
6be3e788f5
Add getter for rtp header extensions for receiver classes.
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This is to avoid accessing the array via the config struct.
Moving forward we might want to consider using the RtpHeaderExtensionMap
instead of a std::vector of RtpExtension.
Bug: webrtc:11993
Change-Id: I8469dbbd9bb95a69f87b5912bfc4bf8b8f603beb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261317
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36820}
2022-05-09 16:59:19 +00:00
cb7c7366d0
Separate reading remote_ssrc from using the rtp_config() getter.
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`remote_ssrc` can be considered const while some other state represented
by rtp_config() can not and also is tied to a specific thread.
Separating access to these variables, makes moving things around easier.
Bug: webrtc:11993
Change-Id: I70aa000daab6174a401e01dca163213174e8f284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261316
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
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Cr-Commit-Position: refs/heads/main@{#36818}
2022-05-09 14:55:00 +00:00
cc50b04c02
Remove config() getter from AudioReceiveStream().
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This reduces the surface of externally accessible state that belongs
to the class, which makes it easier to control what state belongs to
what thread. In this CL enforcing remote_ssrc() to be conceptually const
and sync_group to conceptually belong to the packet delivery thread.
Bug: webrtc:11993
Change-Id: I7de9366dc0c2bf451b5c58595c2d073b4016f2e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261450
Reviewed-by: Niels Moller <nisse@webrtc.org >
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36813}
2022-05-09 11:21:44 +00:00
89e017dd72
Update WebRTC code version (2022-05-07T04:01:40).
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Bug: None
Change-Id: I5332e8a36c357feedcfe2aa7630f85429e43ed18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261442
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2022-05-07 06:30:52 +00:00
09a92ff3de
Update WebRTC code version (2022-05-06T04:02:20).
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Bug: None
Change-Id: I78653dd5b5c88eb310635e315c08878f5c693a3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261380
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2022-05-06 05:42:30 +00:00
d852d9c536
Update WebRTC code version (2022-05-05T04:02:39).
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Bug: None
Change-Id: I3da3e7eb522fddf512ecc751f48223a30871f536
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261145
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2022-05-05 05:54:08 +00:00
0dc2b7095f
Update WebRTC code version (2022-05-04T04:04:14).
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Bug: None
Change-Id: Ibbd8db1953d8406f864679b45563c644f2ef3fdd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261026
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2022-05-04 05:16:37 +00:00
777c48b553
Update WebRTC code version (2022-05-03T04:02:12).
...
Bug: None
Change-Id: Ibd8c9420236e6132eeb39dd6a96503ad14e9a678
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260908
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2022-05-03 05:23:26 +00:00
a16a6a6341
stats: implement inbound-rtp totalProcessingDelay for video
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https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay
BUG=webrtc:13984
Change-Id: Ifd821bd8553add46218f09a11366096d62f5d09f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259768
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36732}
2022-05-02 10:56:22 +00:00
c706220cbd
Update WebRTC code version (2022-05-02T04:04:05).
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Bug: None
Change-Id: Ia7ceabc10b15f378ced28b8a83ecbdd26fe31039
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260821
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2022-05-02 04:57:13 +00:00
249382e79d
Update WebRTC code version (2022-04-30T04:02:31).
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Bug: None
Change-Id: I779009d5e864e6cca54f7ae060c05161870d8091
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260542
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2022-04-30 05:51:30 +00:00
c61c9783f5
Update WebRTC code version (2022-04-28T04:04:23).
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Bug: None
Change-Id: Iffe5715813f546971b26f8fe95f384a7134c7b29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260206
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2022-04-28 05:53:03 +00:00
df209e797b
Avoid more usage of implicit conversion from scoped_refptr<T> to T*
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Update api/, call/, examples/ and rtc_tools/.
Bug: webrtc:13464
Change-Id: I7b0008cca68c579e89b45527a45300d1e67c3483
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260000
Reviewed-by: Henrik Boström <hbos@webrtc.org >
Reviewed-by: Magnus Jedvert <magjed@webrtc.org >
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2022-04-27 13:18:30 +00:00
0f6d406d84
Update WebRTC code version (2022-04-27T04:03:53).
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Bug: None
Change-Id: I890088b0f16c7640ba630a069f02b015960b51df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260121
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Cr-Commit-Position: refs/heads/main@{#36666}
2022-04-27 08:51:40 +00:00
632e0590ff
Update WebRTC code version (2022-04-26T04:02:56).
...
Bug: None
Change-Id: Icd5abef1712d815db40f9a293b1acda2c7289eb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260080
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Cr-Commit-Position: refs/heads/main@{#36648}
2022-04-26 05:28:26 +00:00
c3e6e3a3e8
Remove dependency on rtc_base_approved from most targets
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Bug: webrtc:9838
Change-Id: Ibd0199803597eff48ca139a5cecdc3209c62c5d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259873
Auto-Submit: Florent Castelli <orphis@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36643}
2022-04-25 12:15:30 +00:00
4e297525f2
Update WebRTC code version (2022-04-25T04:02:30).
...
Bug: None
Change-Id: I1e991c26e5c7582ade33f8895688c8f146bcd459
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259980
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36639}
2022-04-25 06:04:30 +00:00
a30aef3dea
Move event_tracer out of rtc_base_approved
...
Bug: webrtc:9838
Change-Id: Ic3c424729b5edd3e378c4195afe33ae5c88ad491
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259312
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36637}
2022-04-24 14:47:40 +00:00
ed4aadc0a2
Move copy_on_write_buffer out of rtc_base_approved
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Bug: webrtc:9838
Change-Id: Ib9a9cd3bc868e716315594c436df7e2cce0d9a5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259311
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36636}
2022-04-24 09:51:50 +00:00
3abc630f26
Update WebRTC code version (2022-04-24T04:03:42).
...
Bug: None
Change-Id: I6f710e1d701a50e37d88951365b18d03fad07437
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259920
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Cr-Commit-Position: refs/heads/main@{#36635}
2022-04-24 05:06:59 +00:00
58dfdd221a
Update WebRTC code version (2022-04-23T04:02:41).
...
Bug: None
Change-Id: If75c7063e137ec72968bcb19d22c432056196569
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259900
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Cr-Commit-Position: refs/heads/main@{#36630}
2022-04-23 05:11:19 +00:00
8d7595d2ec
Update WebRTC code version (2022-04-22T04:03:29).
...
Bug: None
Change-Id: I0bbd922445f39d149ba47ff26a1f1ad726342ab7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259808
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Cr-Commit-Position: refs/heads/main@{#36610}
2022-04-22 04:53:25 +00:00
f4db351625
Move race_checker out of rtc_base_approved
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Bug: webrtc:9838
Change-Id: If180abcca1ef598314de3aed70e4a6eb04f062d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258770
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36607}
2022-04-21 14:40:06 +00:00
4587852740
Update WebRTC code version (2022-04-21T04:02:19).
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Bug: None
Change-Id: I88267ce0cb5fcf2fd724ea591150283ab1885959
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259720
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Cr-Commit-Position: refs/heads/main@{#36595}
2022-04-21 05:22:44 +00:00
831e24e6b2
Update WebRTC code version (2022-04-20T04:02:12).
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Bug: None
Change-Id: I07b92d60728c72c456646a52e420d589f828f22f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259459
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Cr-Commit-Position: refs/heads/main@{#36577}
2022-04-20 05:20:47 +00:00
aa6d05d268
Move location out of rtc_base_approved
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Bug: webrtc:9838
Change-Id: I3510f3d01df532cb228a6cc14be830f6ab81a8e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258764
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36576}
2022-04-19 21:57:48 +00:00
71337f387e
Move random out of rtc_base_approved
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Bug: webrtc:9838
Change-Id: I64a5ef18c19d446139354d04aa6cb2a76d18aad0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258762
Reviewed-by: Harald Alvestrand <hta@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36572}
2022-04-19 14:00:47 +00:00
d1fd9b5e73
Update WebRTC code version (2022-04-19T04:02:48).
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Bug: None
Change-Id: I7f1b7a2dc0e79b9cedf158e928d714eaccfd6ada
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259188
Bot-Commit: webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com <webrtc-version-updater@webrtc-ci.iam.gserviceaccount.com >
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Cr-Commit-Position: refs/heads/main@{#36567}
2022-04-19 05:03:52 +00:00
45a0599978
Remove platform_thread from //rtc_base:rtc_base_approved public_deps
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While the target has a restricted visibility, since it was in rtc_base_approved
public deps, a lot of targets were able to bypass the visibility check.
So we remove the visibility restrictions and use the dependency explicitely
everywhere instead.
Bug: webrtc:8603
Change-Id: I94a03fdf7f94c54ab72081a58dd648e2cca73d17
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258944
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org >
Commit-Queue: Florent Castelli <orphis@webrtc.org >
Cr-Commit-Position: refs/heads/main@{#36566}
2022-04-18 23:12:52 +00:00
29ff2a8779
Update WebRTC code version (2022-04-18T04:02:53).
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Bug: None
Change-Id: If96ba206ca879964043d66928fcd1a4be8cdb80d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259118
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Cr-Commit-Position: refs/heads/main@{#36562}
2022-04-18 05:15:01 +00:00
a62136ac74
Update WebRTC code version (2022-04-17T04:03:50).
...
Bug: None
Change-Id: I8a0314c05e284f1c4c8a120a48f1ceeb10daa5b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259120
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Cr-Commit-Position: refs/heads/main@{#36560}
2022-04-17 05:23:56 +00:00
a72988d198
Update WebRTC code version (2022-04-16T04:06:08).
...
Bug: None
Change-Id: I1dd8201210d3894a7c14cff1f02905e288c1ce5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/259082
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Cr-Commit-Position: refs/heads/main@{#36559}
2022-04-16 05:16:35 +00:00
