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a86cef7e2ccaa1ff4db0d26754ea70634c94dd70
platform-external-webrtc/modules/audio_coding/codecs
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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
..
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
…
audio_encoder.h
…
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_unittest.cc
Stop using Googletest legacy APIs.
2019-01-31 13:23:33 +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