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23 Commits

Author SHA1 Message Date
Harald Alvestrand
97597c0f51 Remove usage of INFO alias for LS_INFO in log messages 
Bug: webrtc:13362
Change-Id: Ifda893861a036a85c045cd366f9eab33c62ebde0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237221
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35310}
 2021-11-04 13:46:17 +00:00
Artem Titov
0e61fdd27c Use backticks not vertical bars to denote variables in comments for /api 
Bug: webrtc:12338
Change-Id: Ib97b2c3d64dbd895f261ffa76a2e885bd934a87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226940
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34554}
 2021-07-26 18:27:34 +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
76bbc98d72 Adding MockVoipEngine for downstream project's tests 
Bug: webrtc:11989
Change-Id: Ie9cfe11a0c2b041457de66c3e3a6cdcd6179e4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201900
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33093}
 2021-01-28 21:06:16 +00:00
Danil Chapovalov
111a3712e7 Delete unused.h include from api as unused 
Bug: None
Change-Id: Ib62bdc296fdff8ecb3eb5b22d5eda1ef8fc35284
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202036
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33026}
 2021-01-18 13:34:17 +00:00
Danil Chapovalov
098da17f35 Reland "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" 
This is a reland of 8c2250eddc7263036397179a0794b9b17d7afb38

Original change's description:
> Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
>
> Bug: webrtc:12336
> Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32978}

Bug: webrtc:12336
Change-Id: I1cd017d45c1578528dec4532345950e9823f4a63
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201732
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33003}
 2021-01-15 17:59:05 +00:00
Danil Chapovalov
4319b1695e Revert "Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code" 
This reverts commit 8c2250eddc7263036397179a0794b9b17d7afb38.

Reason for revert: breaks downstream project

Original change's description:
> Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code
>
> Bug: webrtc:12336
> Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32978}

TBR=danilchap@webrtc.org,mbonadei@webrtc.org,crodbro@webrtc.org

Change-Id: I5c9d419785254878a825865808b56841cd30b9b5
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12336
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201731
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32979}
 2021-01-14 15:02:47 +00:00
Danil Chapovalov
8c2250eddc Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in c++ code 
Bug: webrtc:12336
Change-Id: If76f00d0883b5c8a90d3ef5554f5e22384b3fb58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197620
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32978}
 2021-01-14 14:32:26 +00:00
Tim Na
507eacfd35 Reland "ChannelStatistics used for RTP stats in VoipStatistics." 
This is a reland of 444e04be6988fbdcc039d775481ac22481ff9ff4

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 444e04be6988fbdcc039d775481ac22481ff9ff4.

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
9325d343e5 Enforcing return type handling on VoIP API. 
- This CL also affects some return type handling in Android Voip demo
app due to changes in return type handling.

Bug: webrtc:12193
Change-Id: Id76faf7c871476ed1f2d08fb587211ae234ae8d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196625
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32821}
 2020-12-11 20:38:15 +00:00
Alex Konradi
c20baf6067 Remove nesting of Naggy/Strict/NiceMock 
This will soon become a compile-time error. Fix class hierarchies that
wrap StrictMock in a NiceMock or vice-versa by removing redundant
wrappings and removing inheritance from Nice/StrictMock and fixing the
call sites as appropriate.

Bug: b/173702213
Change-Id: Ic90b1f270c180f7308f40e52e358a8f6a6baad86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196461
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32783}
 2020-12-07 08:19:50 +00:00
Tim Na
b223cb60e9 Defining API result types on VoIP API 
Bug: webrtc:12193
Change-Id: I6f5ffd82cc838e6982257781f225f9d8159e6b82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193720
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32656}
 2020-11-20 18:02:05 +00:00
Tim Na
a58cae3eae VoipVolumeControl subAPI for VoIP API 
- mute/unmute API.
- speech level/energy/duration API.

Bug: webrtc:12111
Change-Id: I54757b9874d15d59a145f2ca70801ee9ef0f4430
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191060
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32607}
 2020-11-13 19:27:12 +00:00
Tim Na
254ad1b914 Delay VoipCore initialization. 
Starting from Android N, mobile app may not be able to access
microphone while in background where it fails the call.
In order to mitigate the issue, delay the ADM initialization
as late as possible.

Bug: webrtc:12120
Change-Id: I0fbf0300299b6c53413dfaaf88f748edc0a06bc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191100
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32598}
 2020-11-12 18:05:19 +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
Jason Long
a53472940e DTMF Event Sub-API on VoIP API 
Added VoipDtmf in VoipEngine as a sub-API to provide DTMF related interfaces; also added relevant unit tests.

Bug: webrtc:11802
Change-Id: Ie9832aebe075a48ae1207be142361b73646673ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180225
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31974}
 2020-08-20 17:10:02 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release. 
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
 2020-06-08 12:59:40 +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
Tim Na
ccefde95b3 VoIP interfaces API enhancement (continuation of 169000) 
Bug: webrtc:11251
Change-Id: Iecde33b86856b14db5abade3301a842d5007568d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169034
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30675}
 2020-03-03 18:19:54 +00:00
Tim Na
c63bf10790 VoIP interface headers in api/voip directory. This separates the implementation that will come in audio/voip. 
Bug: webrtc:11251
Change-Id: I26b6915d3ad6bb5a50f9898a6866889867fd53f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169000
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30594}
 2020-02-24 15:23:19 +00:00