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17 Commits

Author SHA1 Message Date
stefan
b521aa704f Clean up abs-send-time for audio. 
BUG=None

Review-Url: https://codereview.webrtc.org/2455013003
Cr-Commit-Position: refs/heads/master@{#14870}
 2016-11-01 10:17:18 +00:00
kwiberg
0ccff57024 VoERTP_RTCP: Remove GetREDStatus and SetREDStatus 
They always fail, because RED isn't supported.

BUG=webrtc:5922

Review-Url: https://codereview.webrtc.org/2055753002
Cr-Commit-Position: refs/heads/master@{#13743}
 2016-08-15 10:34:52 +00:00
Peter Boström
c3f4dbc40b Remove rtp_rtcp/ dump functionality. 
Removes RTP dumping from VideoEngine and VoiceEngine.

BUG=1695
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47179004

Cr-Commit-Position: refs/heads/master@{#9234}
 2015-05-20 12:10:56 +00:00
Peter Boström
300eeb68f5 Remove VideoEngine interfaces. 
Removes ViE interfaces, _impl.cc files, managers (such as
ViEChannelManager and ViEInputManager) as well as ViESharedData.

Interfaces necessary to implement observers have been moved to a
corresponding header (such as vie_channel.h).

BUG=1695, 4491
R=mflodman@webrtc.org, solenberg@webrtc.org
TBR=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/55379004

Cr-Commit-Position: refs/heads/master@{#9179}
 2015-05-12 14:51:08 +00:00
Jelena Marusic
0d266054ac VoE: apply new style guide on VoE interfaces and their implementations 
Changes:
1. Ran clang-format on VoE interfaces and their implementations.
2. Replaced virtual with override in derived classes.

R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/49239004

Cr-Commit-Position: refs/heads/master@{#9130}
 2015-05-04 12:15:41 +00:00
stefan@webrtc.org
4ef438e2de Remove the send-side cname getter APIs from voice and video engine. 
These APIs aren't being used, and introduces deadlocks when using GetStats() in the new Call api. Having getters for cname at the send-side is pointless, as it's always the user who sets the cname.

R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16899004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6659 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-11 09:55:30 +00:00
minyue@webrtc.org
c1a40a7b68 This CL is to adding feedback of packet loss rate to encoder in voice engine. A direct reason for doing it is to make use of Opus FEC, which can adapt itself to changes in the packet loss rate. 
This CL is going to be combined with another CL in ACM, which is to be landed.

TEST=passed_try_bots
BUG=
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13449004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-28 09:52:06 +00:00
henrika@webrtc.org
66021e0fa2 Removes parts of the webrtc::VoERTP_RTCP sub API as part of a clean-up operation where the goal is to remove unused APIs. 
BUG=3206
R=niklas.enbom@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13489005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6100 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-12 08:53:27 +00:00
wu@webrtc.org
93fd25c20c * Add webrtc::VoERTP_RTCP::SetReceiveAudioLevelIndicationStatus. 
* Cast rtp header extension to int in log in rtp_utility.cc.

BUG=3237
TEST=try bots
R=stefan@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5975 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-04-24 20:33:08 +00:00
solenberg@webrtc.org
b1f5010075 VoE changes to allow forwarding of packets from VoE to ViE BWE. 
BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5757 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-24 10:38:25 +00:00
wu@webrtc.org
ebdb0e3ad0 Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. 
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.

This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.

BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-03-06 23:49:08 +00:00
henrika@webrtc.org
b7a91fa95a Removes VoERTP_RTCP::InsertExtraRTPPacket. 
Reasons for removing:

- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.

BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/8819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-02-19 08:58:08 +00:00
niklas.enbom@webrtc.org
b35d2e3abc Add dummy audio NACK APIs 
R=pwestin@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1579006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4151 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-31 21:13:52 +00:00
pbos@webrtc.org
956aa7e087 Include files from webrtc/.. paths in voice_engine/ 
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1434005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4079 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-21 13:52:32 +00:00
pbos@webrtc.org
9213521ea9 Remove const for plain data types in voice_engine/ 
BUG=1644
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1463004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-14 08:31:39 +00:00
roosa@google.com
0870f02cdb Add API to retreive last received RTP timestamp to VoiceEngine. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/969016

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3271 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 21:31:41 +00:00
andrew@webrtc.org
14b43beb7c Move src/ -> webrtc/ 
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/915006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-10-22 18:19:23 +00:00