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webrtc/modules/audio_coding
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Jakob Ivarsson 37b5662a5c Remove zero lower bound of estimated inter-arrival time. 
When we offset the measured inter-arrival time due to packet loss, it will sometimes be less than zero. This is the correct value to use when calculating the relative packet arrival delay.

Bug: webrtc:10333
Change-Id: I14a68563a379fa0b9444684304362503a6f1bfca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127547
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27164}
2019-03-18 16:52:01 +00:00
..
acm2 Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
audio_network_adaptor webrtc: Remove semicolons. 2019-02-20 16:02:59 +00:00
codecs Fix -Wextra-semi warnings. 2019-02-25 09:22:51 +00:00
include Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
neteq Remove zero lower bound of estimated inter-arrival time. 2019-03-18 16:52:01 +00:00
test Prepare for splitting FrameType into AudioFrameType and VideoFrameType 2019-03-07 10:12:57 +00:00
audio_coding.gni Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
BUILD.gn Mark neteq_rtpplay as publicly visible. 2019-03-18 14:51:50 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00