webrtc/modules/audio_coding
Gustaf Ullberg b9fc6508c0 Add min and max allowed bitrate in Opus bitrate tests
Instead of checking for an exact bitrate check that the bitrate is between
the min and max values.
Also relax a threshold in a bandwith adaptation test.

Bug: webrtc:9280
Change-Id: I465d785a53759f73242198ee1ccd7da1a26c48b7
Reviewed-on: https://webrtc-review.googlesource.com/78041
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23330}
2018-05-21 16:41:35 +00:00
..
acm2 Add min and max allowed bitrate in Opus bitrate tests 2018-05-21 16:41:35 +00:00
audio_network_adaptor Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
codecs Add min and max allowed bitrate in Opus bitrate tests 2018-05-21 16:41:35 +00:00
include Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq NetEq: Fixing an overflow bug in expand.cc 2018-05-21 13:39:25 +00:00
test Remove incompatiblities with absl::optional in audio_coding 2018-04-17 12:05:13 +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 Break out the part of the iSAC codec that's used for Voice Activity Detection 2018-05-04 08:53:34 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00