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webrtc_m130/webrtc/modules/audio_coding
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Henrik Lundin b1629cf5d6 Avoid overflow in NetEq's TimeStretch::SpeechDetection
BUG=chromium:675193
R=kwiberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2718943004 .
Cr-Commit-Position: refs/heads/master@{#16902}
2017-02-28 13:58:30 +00:00
..
acm2
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
audio_network_adaptor
Change frame length on very low bandwidth.
2017-02-22 15:35:05 +00:00
codecs
Change frame length on very low bandwidth.
2017-02-22 15:35:05 +00:00
include
Remove VoEVideoSync interface.
2017-02-15 08:42:31 +00:00
neteq
Avoid overflow in NetEq's TimeStretch::SpeechDetection
2017-02-28 13:58:30 +00:00
test
Disable flaky tests on iOS
2017-02-27 06:10:14 +00:00
audio_coding.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
BUILD.gn
Move AudioDecoder and related stuff to the api/ directory
2017-02-10 16:15:44 +00:00
DEPS
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
OWNERS
Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 15:52:14 +00:00
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