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webrtc/modules/audio_coding
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Jakob Ivarsson 647d5e6d91 Increase the default maximum jitter buffer size to 200 packets. 
Bug: webrtc:10415
Change-Id: Iec5a5a263c11d92a23290c1c2de053fe9e5d5839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128082
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27142}
2019-03-15 10:25:11 +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 Reland "NetEQ RTP Play: Optionally write output audio file" 2019-03-13 15:33:29 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Make ivoc owner of audio_coding. 2018-10-15 15:08:28 +00:00