Commit graph

12 commits

Author SHA1 Message Date
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
2412602b68 Using absl::optional for round trip time return type handling.
No-Try: True
Bug: webrtc:11989
Change-Id: If2ed9b83468c03b82b372e64d8012e5786295476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32827}
2020-12-15 08:38:12 +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
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
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
Markus Handell
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
Tomas Gunnarsson
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
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
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