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webrtc_m130/webrtc/modules/audio_coding
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ossu 6b6c88f184 NetEq jitter calculation now done in uint64_t.
The timestamps are 32 bit and can (conceivably) be spaced far enough
apart for the calculation, which is done in Q4, to overflow.

BUG=chromium:653268

Review-Url: https://codereview.webrtc.org/2460393002
Cr-Commit-Position: refs/heads/master@{#14856}
2016-10-31 15:59:34 +00:00
..
acm2
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
audio_network_adaptor
Simplifying audio network adaptor by moving receiver frame length range to ctor.
2016-10-24 16:19:22 +00:00
codecs
Using AudioOption to enable audio network adaptor.
2016-10-31 11:08:37 +00:00
include
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
2016-10-12 18:04:16 +00:00
neteq
NetEq jitter calculation now done in uint64_t.
2016-10-31 15:59:34 +00:00
test
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
audio_coding_tests.gypi
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
audio_coding.gni
GN conversion of audio_decoder_unittests
2016-08-01 14:49:50 +00:00
audio_coding.gypi
Stop using old AudioCodingModule::RegisterReceiveCodec overloads
2016-10-24 20:47:16 +00:00
BUILD.gn
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
DEPS
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
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