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webrtc/modules/audio_coding
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Alex Luebs eeb2765f6c Implement Opus bandwidth adjustment behind a FieldTrial 
Bug: webrtc:8522
Change-Id: I3a32ebfecd27ff74b507c2cee9e16aab17153442
Reviewed-on: https://webrtc-review.googlesource.com/22210
Commit-Queue: Alejandro Luebs <aluebs@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20799}
2017-11-20 20:04:19 +00:00
..
acm2 Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
audio_network_adaptor Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
codecs Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
include Remove AudioCodingModule::IncomingPayload 2017-09-29 14:23:27 +00:00
neteq neteq_rtpplay: Add buffer size (target and current) to print-out 2017-11-20 08:07:30 +00:00
test Avoid flagging Opus DTX frames as speech. 2017-11-20 14:53:40 +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 Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +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