88 Commits

Author SHA1 Message Date
Boris Tsirkin
105172eaa3 Format /audio folder
Formatting done via:

git ls-files | grep -E '^audio\/.*\.(h|cc)' | xargs clang-format -i

No-Iwyu: Includes didn't change and it isn't related to formatting
Bug: webrtc:42225392
Change-Id: I28fc4e2ef9929ae2ff0d473d71020392218298a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373602
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43673}
2025-01-07 23:43:38 -08:00
Dor Hen
3fa21c89c0 Comment unused variables in implemented functions 13\n
Increased the number of errors the automation is fixing to 150 from
75 in this commit.

Bug: webrtc:370878648
Change-Id: If6e6a5f40db7eb54c27c1a85fb7031838e478c70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366205
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Cr-Commit-Position: refs/heads/main@{#43337}
2024-10-31 07:44:25 +00:00
Dor Hen
b52416eccf Comment unused variables in implemented functions 8\n
Bug: webrtc:370878648
Change-Id: If66e079ff5e455b5c3c483c4c42ef7b38bd34307
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/366262
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Dor Hen <dorhen@meta.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43316}
2024-10-28 12:05:18 +00:00
Olov Brändström
b9c4c242d4 rename timestamps to show epoch
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}
2024-10-08 16:27:58 +00:00
Henrik Lundin
1131c26b25 Move default_neteq_factory to api/neteq and make it publicly visible
Bug: webrtc:14867
Change-Id: I30eefba754a3aae28ffa761f706f5655a2de657d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43092}
2024-09-27 08:34:56 +00:00
Henrik Lundin
8487d3248b Remove all use of AcmReceiver from WebRTC
The class itself and its unit test remains, for now, but will be removed
later.

Bug: webrtc:14867
Change-Id: I36cec8fca7913663f63c53622ed2760e5e048c2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/362580
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43023}
2024-09-16 08:49:25 +00:00
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Danil Chapovalov
24366b0b4c Propagate Environment to audio RtpRtcp modules
Bug: webrtc:362762208
Change-Id: I5be383dd709958cbefb06fe489c96e5ba6891bef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361143
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42904}
2024-09-02 08:57:49 +00:00
Danil Chapovalov
33582ea42f Pass Environment instead of just clock to AcmReceiver at construction
Passing Environment would allow to propage field trials with it further to NetEq and AudioDecoders

Bug: webrtc:356878416
Change-Id: Ic68420df3b157ed341146207a2c45cb49e59a931
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358501
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42728}
2024-08-06 08:28:23 +00:00
Danil Chapovalov
943828b7ff Pass Clock and TaskQueueFactory as Environment in voip audio channel
To make Environment available for creating AudioDecoders to use propagated field trials

Bug: webrtc:356878416
Change-Id: Ic2371f038b75402bbd007c948f43c60cc6cca8a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/358400
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42718}
2024-08-02 10:16:09 +00:00
Danil Chapovalov
1030eaaffe Provide Environment to create an audio encoder in tests
Bug: webrtc:343086059
Change-Id: I73a48770ae67e529eb5065e957ea6420dea44975
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/354881
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42542}
2024-06-26 12:54:36 +00:00
Danil Chapovalov
578905e7ca Provide Environment to create audio encoders in both prod code paths
Bug: webrtc:343086059
Change-Id: I4a3e48dcafe99c47f7c9847c5c3994c9c49807c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/355002
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42508}
2024-06-18 12:31:27 +00:00
Danil Chapovalov
03ebfdf044 Create Environment for VoipCore
To make Environment available for creating AudioEncoders in follow ups

Bug: webrtc:343086059
Change-Id: I0965155915caeee28964ce8406045beeabaa0185
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353741
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42460}
2024-06-11 10:49:19 +00:00
Harald Alvestrand
c74412b304 Deprecate rtc::RefCountInterface
and move usages to webrtc::RefCountInterface

This CL also moves more stuff to webrtc:: and adds backwards
compatible aliases for them.

Bug: webrtc:42225969
Change-Id: Iefb8542cff793bd8aa46bef8f2f3c66a1e979d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353720
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42446}
2024-06-07 09:47:26 +00:00
Lionel Koenig
5889cf5888 Propagate arrival time inside NetEq
Bug: webrtc:341266986
Change-Id: I0fdd14e3fc5b09cbc9369497501f399464964211
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352920
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42414}
2024-05-30 14:21:42 +00:00
Lionel Koenig Gélas
61dc3ac202 Revert "Propagate arrival time inside NetEq"
This reverts commit 0a23279e33e48c88cc1336128f10090564df61af.

Reason for revert: Breaks internal Google builds.

Original change's description:
> Propagate arrival time inside NetEq
>
> Bug: webrtc:341266986
> Change-Id: I1532ba2329272d6ca1602924f4e9ee61b19ad890
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352201
> Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42405}

Bug: webrtc:341266986
Change-Id: I92c12df3d1c3f6584f2ead3d965d78988a7b5405
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352822
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Lionel Koenig Gélas <lionelk@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42410}
2024-05-30 11:06:43 +00:00
Lionel Koenig
0a23279e33 Propagate arrival time inside NetEq
Bug: webrtc:341266986
Change-Id: I1532ba2329272d6ca1602924f4e9ee61b19ad890
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/352201
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42405}
2024-05-29 15:36:12 +00:00
Florent Castelli
99c519b3fd Mass removal of absl_deps in all BUILD.gn files
Bug: webrtc:341803749
Change-Id: Id73844ba8d63b9f2f2c9391d8d8116ad0864c36d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/351540
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42372}
2024-05-23 15:09:46 +00:00
Florent Castelli
f4673f97ed Move webrtc::AudioDeviceModule include to api/ folder
Bug: webrtc:15874
Change-Id: I5bdb19d5e710838b41e6ca283d406c9f1f21286b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/348060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42137}
2024-04-22 08:56:31 +00:00
Florent Castelli
0afde7614d Move webrtc::AudioProcessing include to api/ folder
Bug: webrtc:15874
Change-Id: Ie8a6e031c0f0505cfe238f7d252c47e9c34408d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347983
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42128}
2024-04-20 07:02:50 +00:00
Danil Chapovalov
b1799b0814 Cleanup usage of the rtc::TaskQueue in audio/
Bug: webrtc:14169
Change-Id: I91f158ce072cb1109ec2d8f9e9c8f6a530aa02cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/335080
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41559}
2024-01-18 12:24:14 +00:00
Alfred E. Heggestad
c951d1b0f6 audio: fix some typos
Bug: None
Change-Id: I255a23a893d008dc58c3c9cb3facf61419c88c72
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/320620
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#40779}
2023-09-21 05:42:29 +00:00
Danil Chapovalov
4c556219e5 Cleanup RTPSenderAudio::SendAudio
Combine all parameters into single struct so that it is easier to add and remove optional parameters
Use Timestamp type instad of plain int to represent capture time
Use rtc::ArrayView instead of pointer+size to represent payload
Merge passing audio level into send function.

Bug: webrtc:13757, webrtc:14870
Change-Id: I0386b710eb99b864334d61235add9abcde9bc69d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317442
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40688}
2023-09-04 11:27:42 +00:00
Harald Alvestrand
d43af9172b Remove internal overrides using old SendRtp and SendRtcp interfaces.
This CL takes away all usages except for Android code.

Low-Coverage-Reason: Refactoring old code
Bug: webrtc:15410
Change-Id: I66bed6a4a2787b4177a82e599b48623ca67cd235
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315940
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40554}
2023-08-15 13:20:21 +00:00
Per K
48c44e3543 Ensure RtpSenderEgress run on worker queue
VoipCore still use RtpSenderEgress::NonPacedPacketSender, therefore
packets sent using NonPacedPacketSender::EnqueuePackets are proxied
to the worker thead.
When NonPacedPacketSender is used, the Pacer already guarantee that packets are sent on the worker queue.

Lock is removed from RtpSenderEgress and instead calls must be made on
the worker thread.


Bug: webrtc:15209
Change-Id: Iaf03377ad8a037ecedbbe588a4c1e8e4eadacd81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/306960
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40252}
2023-06-09 13:40:35 +00:00
Danil Chapovalov
8095d02884 Add RtpRtcpInterface::LastRtt function to replace RtpRtcpInterface::RTT
RtpRtcpInterface::RTT follows discouraged style of using return values,
uses raw integers to represent time delta,
and returns values that no code uses (min, max, average RTT)

added LastRtt function addresses all these stylistic issues.

Bug: webrtc:13757
Change-Id: Iaf947dd1b7139026f2beb991e69634c606c6b608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304520
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40028}
2023-05-09 14:54:50 +00:00
Danil Chapovalov
00ff2bb36a Cleanup usasge of ReportBlockData::report_block accessor in audio
This reduces dependency on the struct RTCPReportBlock and would allow to
delete it in favor of class ReportBlockData

Bug: None
Change-Id: I751c7fae1b0285eccdff6e0fe85c8e1ea7d7362c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304280
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39992}
2023-05-05 13:57:30 +00:00
Danil Chapovalov
ec2670e631 Cleanup ReportBlockData class: use Timestamp and TimeDelta
Bug: webrtc:13757
Change-Id: Ic3ddb05413f58cedd12bf0f32c852befb9bd40f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/300940
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39841}
2023-04-13 08:51:12 +00:00
Markus Handell
a1ceae206b Implement support for Chrome task origin tracing. #3.5/4
This CL migrates unit tests to the new TaskQueueBase interface.

Bug: chromium:1416199
Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39434}
2023-03-01 11:11:37 +00:00
Danil Chapovalov
9f397217e1 Delete RtpRtcpInterface::RemoteNtp as redundant to GetSenderReportStats
Bug: None
Change-Id: I8d5ed723ce29231f805e6819156a16ba275f8e3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295321
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39415}
2023-02-28 13:55:27 +00:00
Henrik Lundin
84f75699c6 Break apart AudioCodingModule and AcmReceiver
This change makes AudioCodingModule a pure sender and AcmReceiver a pure
receiver.

The Config struct is in practice no longer used by AudioCodingModule,
so a new definition is included in AcmReceiver. The old definition
remains in AudioCodingModule while downstream clients are being
updated.

Bug: webrtc:14867
Change-Id: If0d0b4214c5aa278cf6c85c5b62c6da644de20e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291533
Reviewed-by: Tomas Lundqvist <tomasl@google.com>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39244}
2023-02-01 16:09:26 +00:00
Harald Alvestrand
1f206b841e Use ArrayView in the IncomingRtcpPacket function.
The lowest level and some of the highest levels of this function are
already using ArrayView. Make this consistent throughout.
Use deprecation for the old API rather than deleting it, since upstream
may be using it.

Bug: webrtc:14870
Change-Id: If5e1a6e9802ecf7e8e3ec27befb5167ca9985517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291706
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39241}
2023-02-01 12:19:03 +00:00
Evan Shrubsole
8267cf31ce [Unwrap] Use RtpTimestampUnwrapper in audio_ingress
Bug: webrtc:13982
Change-Id: I748f8e9d5497eac3335b8a9397199ddf24ecc8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288746
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39035}
2023-01-09 14:47:22 +00:00
Danil Chapovalov
9e09a1f327 Replace Thread::Invoke with Thread::BlockingCall
BlockingCall doesn't take rtc::Location parameter and thus most of the dependencies on location can be removed

Bug: webrtc:11318
Change-Id: I91a17e342dd9a9e3e2c8f7fbe267474c98a8d0e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/274620
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38045}
2022-09-09 10:44:17 +00:00
Markus Handell
2cfc1af78a Update rtc::Event::Wait call sites to use TimeDelta.
Bug: webrtc:14366
Change-Id: I949c1d26f030696b18153afef977633c9a5bd4cf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272003
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37835}
2022-08-19 10:07:28 +00:00
Danil Chapovalov
c05a1be5b4 Migrate remaining webrtc usage of TaskQueueBase to absl::AnyInvocable
Bug: webrtc:14245
Change-Id: I8de2c23da5fbdfc0b1efbbe07fb6e8de744424a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268191
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37565}
2022-07-20 08:15:08 +00:00
Danil Chapovalov
0fd2ed516b Delete ProcessThread and related Module interface
Bug: webrtc:7219
Change-Id: Id71430a24b21e591494557cf54419d2bc8b3f8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267400
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37416}
2022-07-04 10:20:35 +00:00
Niels Möller
83830f316e Delete TestListener and top-level thread wrapping.
Instead use rtc::AutoThread in tests that need that.

Bug: webrtc:9714
Change-Id: I1f33b1b2d321770d062504dd9ef86d66a345dd42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254681
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36950}
2022-05-20 15:21:21 +00:00
Mirko Bonadei
dab50c6fe8 Revert "Use ADM internal state for init state check."
This reverts commit 0e2221eb2f02ed950f4fd9c7fea40b382ea0a0c8.

Reason for revert: Speculative revert, breaks downstream.

Original change's description:
> Use ADM internal state for init state check.
>
> When ADM is terminated and its state requires reinitialized, VoipCore::initialized_ field will falsely skip required reinitializing.
>
> Bug: webrtc:14054
> Change-Id: Ibeb4987a7e9763b8e40926acc4d7eaabde7a3478
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261924
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Tim Na <natim@google.com>
> Commit-Queue: Tim Na <natim@google.com>
> Cr-Commit-Position: refs/heads/main@{#36893}

Bug: webrtc:14054
Change-Id: I1fa0a1ff440b9619aba60ec25970ce88a67739db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262660
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36896}
2022-05-16 14:08:04 +00:00
Tim Na
0e2221eb2f Use ADM internal state for init state check.
When ADM is terminated and its state requires reinitialized, VoipCore::initialized_ field will falsely skip required reinitializing.

Bug: webrtc:14054
Change-Id: Ibeb4987a7e9763b8e40926acc4d7eaabde7a3478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261924
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tim Na <natim@google.com>
Commit-Queue: Tim Na <natim@google.com>
Cr-Commit-Position: refs/heads/main@{#36893}
2022-05-14 02:17:45 +00:00
Danil Chapovalov
edcb25b623 Migrate RemoteNtpTimeEstimator to more precise time representations
Reland of https://webrtc-review.googlesource.com/c/src/+/261311

Bug: webrtc:13757
Change-Id: I34a58100b8fadfe3dbea9ffce71829b7670daad8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261726
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36838}
2022-05-10 13:38:31 +00:00
Danil Chapovalov
853a407273 Revert "Migrate RemoteNtpTimeEstimator to more precise time representations"
This reverts commit a154a15c978a0eae133d957dcad3581fd5f98c7b.

Reason for revert: breaks downstream tests

Original change's description:
> Migrate RemoteNtpTimeEstimator to more precise time representations
>
> Bug: webrtc:13757
> Change-Id: I880ab3cc6e4f72da587ae42ddca051332907c07f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261311
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#36817}

Bug: webrtc:13757
Change-Id: Id21edb1378e6e944b24955396250ddc33fa70663
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261722
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36819}
2022-05-09 15:48:29 +00:00
Danil Chapovalov
a154a15c97 Migrate RemoteNtpTimeEstimator to more precise time representations
Bug: webrtc:13757
Change-Id: I880ab3cc6e4f72da587ae42ddca051332907c07f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261311
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36817}
2022-05-09 14:32:19 +00:00
Florent Castelli
c3e6e3a3e8 Remove dependency on rtc_base_approved from most targets
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
Florent Castelli
aa6d05d268 Move location out of rtc_base_approved
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
Niels Möller
7336422fe3 Delete some unneeded references to ProcessThread.
Bug: None
Change-Id: I77528df2a8bd2d461440cf59ada8229e732a1e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242370
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35613}
2022-01-03 15:36:02 +00:00
Tim Na
1d73243466 Use a new instance of RTP stack for each test.
- Reusing RTP stack may have contributed to some flakiness as
  the previous state could have persisted to new test being performed.

Bug: webrtc:13241
Change-Id: Idf70b56bd3377bc99321fddf7191d7a72c37b085
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237540
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35336}
2021-11-11 17:33:03 +00:00
Artem Titov
b0ea637ec2 Use backticks not vertical bars to denote variables in comments for /audio
Bug: webrtc:12338
Change-Id: Ief89269aa39d0cb6749a1c6cc995ce8830ca327f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226942
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34564}
2021-07-27 15:36:40 +00:00
Markus Handell
eb61b7f620 ModuleRtcRtcpImpl2: remove Module inheritance.
This change achieves an Idle Wakeup savings of 200 Hz.

ModuleRtcRtcpImpl2 had Process() logic only active if TMMBR() is
enabled in RtcpSender, which it never is. Hence the Module
inheritance could be removed. The change removes all known
dependencies of the module inheritance, and any related mentions
of ProcessThread.

Fixed: webrtc:11581
Change-Id: I440942f07187fdb9ac18186dab088633969b340e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/222604
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34358}
2021-06-22 14:51:04 +00:00
Tomas Gunnarsson
c1d589146b Replace new rtc::RefCountedObject with rtc::make_ref_counted in a few files
Bug: webrtc:12701
Change-Id: Ie50225374f811424faf20caf4cf454b2fd1c4dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215930
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33818}
2021-04-23 12:04:39 +00:00