Ying Wang 0e1a558fb3 Allowing 40ms audio frame length. ... Currently 20ms, 60ms and 120ms frame length are supported. The motivation is to better adapt audio bit rate to network conditions with more frame length choices. This is continuation of https://webrtc-review.googlesource.com/c/src/+/146206, since crodbro is out of office, I created this commit for continuing the code review. Bug: webrtc:10820 Change-Id: I0e35e91b524f63686bfdf767b7a95c51aeb24716 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146780 Reviewed-by: Minyue Li <minyue@webrtc.org> Reviewed-by: Alex Narest <alexnarest@webrtc.org> Reviewed-by: Bjorn Mellem <mellem@webrtc.org> Commit-Queue: Ying Wang <yinwa@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28882}		2019-08-16 20:24:18 +00:00
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acm2	Roll chromium_revision 8f0166a59b..f0fd984a31 (685582:685691)	2019-08-12 15:53:01 +00:00
audio_network_adaptor	Allowing 40ms audio frame length.	2019-08-16 20:24:18 +00:00
codecs	Make ANA frame length controller more robust to encoder frame lengths.	2019-08-16 10:55:39 +00:00
include	Delete NACK-related methods from AudioCodingModule	2019-08-12 09:41:10 +00:00
neteq	Avoid copying of vectors in RtpPacketInfos.	2019-08-14 15:46:02 +00:00
test	Delete unused Opus-specific methods of AudioCodingModule	2019-08-09 07:06:36 +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	Delete deprecated rtc_event_log header	2019-08-07 10:58:17 +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