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webrtc_m130/modules/audio_coding
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Mirko Bonadei 7edc49cb31 Mark neteq_rtpplay as publicly visible. 
Bug: None
Change-Id: I051c7c23851ab15345c8e0f0322458d4f9a7e187
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128123
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27162}
2019-03-18 14:51:50 +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
Increase the default maximum jitter buffer size to 200 packets.
2019-03-15 10:25:11 +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
…
OWNERS
Make ivoc owner of audio_coding.
2018-10-15 15:08:28 +00:00