webrtc/modules/audio_coding
Minyue Li 1272dade56 Reduce log level of Opus bitrate.
Bug: None
Change-Id: Iab815dbbc12bf1ca2c1cc87acb0765e2ccade591
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157895
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29570}
2019-10-22 12:02:09 +00:00
..
acm2 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate 2019-10-11 08:34:53 +00:00
audio_network_adaptor Use std::make_unique instead of absl::make_unique. 2019-09-17 15:47:29 +00:00
codecs Reduce log level of Opus bitrate. 2019-10-22 12:02:09 +00:00
include Delete unused method AudioCodingModule::GetDecodingCallStatistics 2019-09-04 10:08:16 +00:00
neteq Change failing rtc::dchecked_cast to rtc::saturated_cast. 2019-10-21 12:06:52 +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 Move rtc_base/ignore_wundef.h to its own target. 2019-10-19 10:50:36 +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