webrtc/modules/audio_coding
Mirko Bonadei 5b86f0a24b Stop using ByteSize (deprecated) to get the size of a proto message.
The method ByteSize has been deprecated [1], this CL switches to
ByteSizeLong.

[1] - https://cs.chromium.org/chromium/src/third_party/protobuf/src/google/protobuf/message_lite.h?l=252&rcl=ac47edd22c481fcfe119769d6b7abf365abea8fa

Bug: None
Change-Id: I1ba622df52f47719a5beda6d230cb603a0163d43
Reviewed-on: https://webrtc-review.googlesource.com/27021
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20952}
2017-11-30 14:27:50 +00:00
..
acm2 Add new UMA metric for NetEq target buffer delay 2017-11-29 08:56:29 +00:00
audio_network_adaptor Stop using ByteSize (deprecated) to get the size of a proto message. 2017-11-30 14:27:50 +00:00
codecs Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
include Add new UMA metric for NetEq target buffer delay 2017-11-29 08:56:29 +00:00
neteq Fix circular dependency in rtc_event_log. 2017-11-29 10:46:19 +00:00
test Avoid flagging Opus DTX frames as speech. 2017-11-20 14:53:40 +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 Fix circular dependency in rtc_event_log. 2017-11-29 10:46:19 +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