Commit Graph

10 Commits

Author SHA1 Message Date
d15a575ec3 Use SequenceChecker from public API
Bug: webrtc:12419
Change-Id: I00cca16a0ec70246156ba00b97aa7ae5ccbf5364
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205323
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33220}
2021-02-10 15:04:55 +00:00
c8421c4c3e Replace rtc::ThreadChecker with webrtc::SequenceChecker
Bug: webrtc:12419
Change-Id: I825c014cc1c4b1dcba5ef300409d44859e971144
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/205002
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33136}
2021-02-02 14:56:27 +00:00
26ce03e469 Locating input audio level before TaskQueue.
- TaskQueue needs to be destroyed at last to avoid thread race condition.

Bug: webrtc:12111
Change-Id: Ibfc96e2ebd71a2aa8d1ac8c83038d256bac0e600
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193780
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32624}
2020-11-17 20:53:36 +00:00
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
dba1f945cf Added Error Checking in Ingress/Egress and Extra Unit Tests
Added error checking in AudioIngress and AudioEgress to detect situations where codecs have not been set; added additional unit tests for VoipCore

Bug: webrtc:11251
Change-Id: Ibd57e518892c76e2865b844ba866e380a565dd6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180229
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31874}
2020-08-06 20:48:13 +00:00
6287280d64 Migrate audio/ to use webrtc::Mutex
Bug: webrtc:11567
Change-Id: Ic6a753f09aafb508690f4b8dadd4c99433fcfeb6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176741
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31635}
2020-07-06 14:21:38 +00:00
f25761d798 Remove dependency from RtpRtcp on the Module interface.
The 'Module' part of the implementation must not be
called via the RtpRtcp interface, but is rather a part of
the contract with ProcessThread. That in turn is an
implementation detail for how timers are currently implemented
in the default implementation.

Along the way I'm deprecating away the factory function which
was inside the interface and tied it to one specific implementation.
Instead, I'm moving that to the implementation itself and down the
line, we don't have to go through it if we just want to create an
instance of the class.

The key change is in rtp_rtcp.h and the new rtp_rtcp_interface.h
header file (things moved from rtp_rtcp.h), the rest falls from that.

Change-Id: I294f13e947b9e3e4e649400ee94a11a81e8071ce
Bug: webrtc:11581
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176419
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31440}
2020-06-04 08:11:21 +00:00
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
11f92bc81b Audio ingress implementation for voip api.
This is based on channel_receive.cc implementation where non-audio
related logics are trimmed off for smaller footprint in size.

Bug: webrtc:11251
Change-Id: I743c9f93f509fa6fcc12981fa73a6f01ce38348c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172821
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31117}
2020-04-21 20:19:37 +00:00
8ab3c77c01 Audio egress implementation for initial voip api in api/voip.
For simplicity and flexibility on audio only API, it deemed
to be better to trim off all audio unrelated logic to serve
the purpose.

Bug: webrtc:11251
Change-Id: I40e3eba2714c171f7c98b158303a7b3f744ceb78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169462
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30922}
2020-03-27 18:45:43 +00:00