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webrtc/modules/audio_coding/codecs
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Mirko Bonadei 9653d26f8e Rename FATAL() into RTC_FATAL(). 
No-Try: True
Bug: webrtc:8454
Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32620}
2020-11-17 16:12:40 +00:00
..
cng Rename FATAL() into RTC_FATAL(). 2020-11-17 16:12:40 +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 Introduce RTC_CHECK_NOTREACHED(), an always-checking RTC_NOTREACHED() 2020-11-09 10:47:55 +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