Commit graph

16 commits

Author SHA1 Message Date
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
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
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
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
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
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
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
Tim Na
1cbf21e157 ChannelStatistics RTT test case around remote SSRC change.
Bug: webrtc:11989
Change-Id: I7211ffa83ad381b6367d78e553cd5943dca515f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201880
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33039}
2021-01-19 19:07:47 +00:00
Tim Na
507eacfd35 Reland "ChannelStatistics used for RTP stats in VoipStatistics."
This is a reland of 444e04be69

Reason for reland: resolved the breaks from downstream project

Original change's description:
> ChannelStatistics used for RTP stats in VoipStatistics.
>
> - Added local and remote RTP statistics query API.
> - Change includes simplifying remote SSRC change handling
>   via received RTP and RTCP packets.
>
> Bug: webrtc:11989
> Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tim Na <natim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32954}

Bug: webrtc:11989
Change-Id: I88620a9f1c037b512821cac9d556905149666870
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201481
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32966}
2021-01-13 16:57:22 +00:00
Alex Loiko
37827c9058 Revert "ChannelStatistics used for RTP stats in VoipStatistics."
This reverts commit 444e04be69.

Reason for revert: breaks downstream project

Original change's description:
> ChannelStatistics used for RTP stats in VoipStatistics.
>
> - Added local and remote RTP statistics query API.
> - Change includes simplifying remote SSRC change handling
>   via received RTP and RTCP packets.
>
> Bug: webrtc:11989
> Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Tim Na <natim@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32954}

TBR=mbonadei@webrtc.org,saza@webrtc.org,hta@webrtc.org,natim@webrtc.org

Change-Id: I5ce6a698c1216c7d56e32fce3308c16daac852f4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11989
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201460
Reviewed-by: Alex Loiko <aleloi@google.com>
Commit-Queue: Alex Loiko <aleloi@google.com>
Cr-Commit-Position: refs/heads/master@{#32956}
2021-01-12 21:35:19 +00:00
Tim Na
444e04be69 ChannelStatistics used for RTP stats in VoipStatistics.
- Added local and remote RTP statistics query API.
- Change includes simplifying remote SSRC change handling
  via received RTP and RTCP packets.

Bug: webrtc:11989
Change-Id: Ia3ee62c1191baaedc67e033ea3c661d8c9301abc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/199060
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32954}
2021-01-12 18:55:41 +00:00
Tim Na
cd4203bf72 Adding total duration and more test cases to VoipStatistics.
- Introduced IngressStatistics to cover total_duration which
comes from AudioLevel.

Bug: webrtc:11989
Change-Id: Iba52d3722b5fe6286b048ab5690e32a4f75e972a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190940
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32538}
2020-11-03 07:15:42 +00:00
Tim Na
f4347f7bac VoipStatistics subAPI and implementation.
- Adding an interface that fetches lifetime NetEq statistics struct.

Bug: webrtc:11989
Change-Id: I871455bccdd53a33dd260f744e03ec81d29fbfd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190200
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32516}
2020-10-28 21:59:05 +00:00
Tim Na
c0df5fc25b VoIP API implementation on top of AudioIngress/Egress
This is one last CL that includes the rest of VoIP API implementation.

Bug: webrtc:11251
Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31168}
2020-05-05 19:55:29 +00:00