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

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