webrtc/modules/audio_coding
Niels Möller 544dfb5a97 Delete isac GetBandwidthInfo/SetBandwidthInfo
Bug: webrtc:10098
Change-Id: I4a56cdc6d081b15a1fc52cba2051783daf4e5ae3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153881
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29256}
2019-09-20 13:53:52 +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 isac GetBandwidthInfo/SetBandwidthInfo 2019-09-20 13:53:52 +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 isac GetBandwidthInfo/SetBandwidthInfo 2019-09-20 13:53:52 +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