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webrtc/modules/audio_coding
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Minyue Li f7789c6e89 Limiting increment in timestamps with neteq simulation. 
Bug: None
Change-Id: I9a0688bcf1c887793b5c94ea023b025aed7366a5
Reviewed-on: https://webrtc-review.googlesource.com/74840
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23733}
2018-06-26 08:07:38 +00:00
..
acm2 Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-06-21 12:36:44 +00:00
audio_network_adaptor Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
codecs Extract third party part of g722 codec into separate target 2018-06-25 11:30:59 +00:00
include Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
neteq Limiting increment in timestamps with neteq simulation. 2018-06-26 08:07:38 +00:00
test Revert "NetEq: Deprecate playout modes Fax, Off and Streaming" 2018-06-21 12:36:44 +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 Extract third party part of g722 codec into separate target 2018-06-25 11:30:59 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00