webrtc/modules/audio_coding
Niels Möller 48b32b748e Delete support for enabling adaptive isac mode
This appears unused. If deleted, other code related to isac bandwidth
estimation becomes unused and may be deleted in followup cls.

Bug: webrtc:10098
Change-Id: Ifeac2e90de895b12c337ea28cc33704350b9abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29252}
2019-09-20 10:41:09 +00:00
..
acm2 Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
audio_network_adaptor Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
codecs Delete support for enabling adaptive isac mode 2019-09-20 10:41:09 +00:00
include Delete unused method AudioCodingModule::GetDecodingCallStatistics 2019-09-04 10:08:16 +00:00
neteq Delete support for enabling adaptive isac mode 2019-09-20 10:41:09 +00:00
test Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +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 support for enabling adaptive isac mode 2019-09-20 10:41:09 +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