Danil Chapovalov a86cef7e2c Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding ... Bug: webrtc:12336 Change-Id: Icae229b957c2bfcc410788179a504c576cfde151 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201736 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32995}		2021-01-15 10:58:20 +00:00
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cng	Reland "Rename FATAL() into RTC_FATAL()."	2020-11-18 20:49:08 +00:00
g711	Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.	2020-03-25 22:19:21 +00:00
g722	Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.	2020-03-25 22:19:21 +00:00
ilbc	Replace RTC_WARN_UNUSED_RESULT with ABSL_MUST_USE_RESULT in audio_coding	2021-01-15 10:58:20 +00:00
isac	Reland "Reland "Default enable WebRTC-SendSideBwe-WithOverhead.""	2020-11-02 11:05:56 +00:00
opus	Revert "opus: take SILK vad result into account for voice detection"	2020-11-04 07:29:48 +00:00
pcm16b	Format almost everything.	2019-07-08 13:45:15 +00:00
red	red: do not generate packets which are > 1200 bytes	2020-08-04 09:53:47 +00:00
tools	Add RTC_ prefix to non-standard format specifier macro "PRIdNS"	2019-08-07 13:36:05 +00:00
audio_decoder.h	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
audio_encoder.h	Fixing WebRTC after moving from src/webrtc to src/	2017-09-15 05:02:56 +00:00
builtin_audio_decoder_factory_unittest.cc	Format almost everything.	2019-07-08 13:45:15 +00:00
builtin_audio_encoder_factory_unittest.cc	Format almost everything.	2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame.cc	[Cleanup] Add missing #include. Remove useless ones.	2018-10-23 11:32:56 +00:00
legacy_encoded_audio_frame.h	Format almost everything.	2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame_unittest.cc	Stop using Googletest legacy APIs.	2019-01-31 13:23:33 +00:00