webrtc/modules/audio_coding
Sebastian Jansson f34116e356 Replacing bandwidth adaptation trial with stable target in Opus encoder.
This also means that the NetworkEstimate::bandwidth can be deprecated
as it's currently just a copy of the target_rate.

Bug: webrtc:10981
Change-Id: I1bc57b98480bd77ce052736b19d630c775428546
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153669
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29288}
2019-09-24 16:35:02 +00:00
..
acm2 Delete AudioDecoder method IncomingPacket 2019-09-24 08:30:24 +00:00
audio_network_adaptor Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
codecs Replacing bandwidth adaptation trial with stable target in Opus encoder. 2019-09-24 16:35:02 +00:00
include Delete unused method AudioCodingModule::GetDecodingCallStatistics 2019-09-04 10:08:16 +00:00
neteq Support 2 byte payload size DTX packets in NetEq simulation. 2019-09-24 15:18:05 +00:00
test Include module_common_types.h only where needed 2019-09-24 08:22:38 +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 Include module_common_types.h only where needed 2019-09-24 08:22:38 +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