webrtc/modules/audio_coding
Sebastian Jansson 62aee9379c Adds trial to calculate audio overhead based on available data.
This adds the ability to disable legacy overhead calculation so we'll
use the available data on per packet over head and frame length range
to set the min and max total  allocatable bitrate.

Bug: webrtc:11001
Change-Id: I2a94499433e15bad11a08f81fe7f1dfc27982cdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155175
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29368}
2019-10-02 13:42:15 +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 Adds trial to calculate audio overhead based on available data. 2019-10-02 13:42:15 +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