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13 Commits

Author SHA1 Message Date
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
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
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
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