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webrtc_m130/webrtc/modules/audio_coding
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Stefan Holmer 1acbd68718 Move RtpExtension to api/ directory and config.h/.cc to call/.
BUG=webrtc:5876
R=deadbeef@webrtc.org, solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/3004723002 .
Cr-Commit-Position: refs/heads/master@{#19639}
2017-09-01 13:29:30 +00:00
..
acm2
Add stats totalSamplesReceived and concealedSamples
2017-08-25 00:50:42 +00:00
audio_network_adaptor
Decouple FEC controller's decision from the decision making for frame length.
2017-08-31 12:08:56 +00:00
codecs
Remove dead code
2017-08-26 05:22:42 +00:00
include
Rename webrtc/base -> webrtc/rtc_base
2017-07-06 19:11:40 +00:00
neteq
Replace remaining gflags usages with rtc_base/flags
2017-08-31 10:21:39 +00:00
test
Replace remaining gflags usages with rtc_base/flags
2017-08-31 10:21:39 +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 RtpExtension to api/ directory and config.h/.cc to call/.
2017-09-01 13:29:30 +00:00
DEPS
Recently we moved webrtc/base to webrtc/rtc_base, so these
2017-08-30 11:45:58 +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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