6f6ba6edee
Fix issues with sequence number wrap-around in jitter statistics
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Wrap-arounds in sequence numbers (and in timestamps) were not always
treated correctly. This is fixed by introducing two helper functions
for correct comparisons, and by casting to the right word size.
Also added a new member variable to AutomodeInst_t. The new member keeps
track of when the first packet has been registered in the automode code.
This was previously done implicitly (and not very good) using the fact
that the lastSeqNo and lastTimestamp members were initialized to zero.
Two new unit tests were added as part of this CL.
NetEqDecodingTest.SequenceNumberWrap was failing before the fixes were
made; now it is ok.
BUG=2654
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/4139004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5150 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-21 13:17:29 +00:00
401ef361ac
Added configuration of max delay to ACM and NetEq
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R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1964004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4499 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-08-06 21:01:36 +00:00
fee739c224
Risk of division by zero.
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bug=b9338699
R=tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1634004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4223 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-06-12 20:10:06 +00:00
e46c8d3875
API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay.
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TEST=unit-test, manual, trybots.
R=henrik.lundin@webrtc.org , henrika@webrtc.org , mflodman@webrtc.org , mikhal@webrtc.org , stefan@webrtc.org , tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1384005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@4087 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-05-22 20:39:43 +00:00
0946a56023
WebRtc_Word32 => int32_t etc. in audio_coding/
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BUG=314
Review URL: https://webrtc-codereview.appspot.com/1271006
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3789 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-04-09 00:28:06 +00:00
6388c3e2fd
Implement initial delay. This CL allows clients of VoE to set an initial delay. Playout of audio is delayed and the extra playout delay is maintained during the call. While packets are buffered (in NetEq) to acheive the desired delay. ACM will playout silence (zeros). Initial delay has to be set before any packet is pushed into ACM.
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TEST=ACM unit test is added, also a manual integration test is writen.
Review URL: https://webrtc-codereview.appspot.com/1097009
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3506 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-02-12 21:42:18 +00:00
cd2f1356ee
Revert 3405
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TBR=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1074004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3407 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-24 22:05:30 +00:00
05e7bfeeea
Mainly hlundin's patch.
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Review URL: https://webrtc-codereview.appspot.com/1052004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3405 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-01-24 18:53:43 +00:00
14b43beb7c
Move src/ -> webrtc/
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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
