Minyue Li e8fbc5d702 Refactor WebRtcOpus_PacketHasFec. ... WebRtcOpus_PacketHasFec was written long time ago. see http://webrtc-codereview.appspot.com/7539004. When revisiting, I notice that adding more comments should help. Code style should be improved a bit too. Bug: webrtc:10772 Change-Id: If4d60b210e6235b4f787608047e88efc949f6838 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144056 Commit-Queue: Minyue Li <minyue@webrtc.org> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28479}		2019-07-04 07:51:52 +00:00
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acm2	Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker.	2019-07-01 15:56:40 +00:00
audio_network_adaptor	Stop DCHECK which occurs in ANA BitrateController when overhead is zero.	2019-04-27 00:20:37 +00:00
codecs	Refactor WebRtcOpus_PacketHasFec.	2019-07-04 07:51:52 +00:00
include	Expose new audio stats on the API	2019-05-03 10:10:15 +00:00
neteq	Improve buffer level estimation with DTX and add CNG time stretching.	2019-07-03 15:12:09 +00:00
test	WebRTC Opus C interface: Add support for non-48 kHz decode sample rate	2019-05-29 10:33:03 +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	Improve buffer level estimation with DTX and add CNG time stretching.	2019-07-03 15:12:09 +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