Jakob Ivarsson 918eb19303 Fix crash when Opus maxptime < 20ms. ... A follow up cl will be created to better handle nullopt frame length range in AudioSendStream. Note that maxptime is still not used for setting the frame length (only ptime is). Bug: chromium:1109337 Change-Id: Id21fd8c76a6c4a0c85719a955116f8d16001a3d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284501 Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38702}		2022-11-22 01:21:24 +00:00
..
cng	Reland "Remove RTC_DISALLOW_COPY_AND_ASSIGN usages completely"	2022-01-27 12:55:44 +00:00
g711	Make header files self contained.	2022-10-08 08:38:36 +00:00
g722	Make header files self contained.	2022-10-08 08:38:36 +00:00
ilbc	Fix chromium roll: remove variables that are written but not accessed	2022-04-06 09:48:20 +00:00
isac	Reland "[ACM] iSAC audio codec removed"	2022-11-17 12:52:35 +00:00
opus	Fix crash when Opus maxptime < 20ms.	2022-11-22 01:21:24 +00:00
pcm16b	Adding fuzzer for PCM16b decoder and fixing a fuzzer problem	2022-02-15 15:59:01 +00:00
red	Adopt absl::string_view in modules/audio_coding/	2022-07-20 13:34:23 +00:00
tools	Delete rtc_base/format_macros.h	2022-05-09 12:03:21 +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	Reland "[ACM] iSAC audio codec removed"	2022-11-17 12:52:35 +00:00
builtin_audio_encoder_factory_unittest.cc	Set the maximum number of audio channels to 24	2021-11-16 17:01:54 +00:00
legacy_encoded_audio_frame.cc	Use backticks not vertical bars to denote variables in comments for /modules/audio_coding	2021-08-02 10:45:40 +00:00
legacy_encoded_audio_frame.h	Format almost everything.	2019-07-08 13:45:15 +00:00
legacy_encoded_audio_frame_unittest.cc	Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED	2021-11-15 21:44:59 +00:00