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webrtc/modules/audio_coding
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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
..
acm2 Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
audio_network_adaptor Adopt absl::string_view in modules/audio_coding/ 2022-07-20 13:34:23 +00:00
codecs Fix crash when Opus maxptime < 20ms. 2022-11-22 01:21:24 +00:00
g3doc Update links to point at main branch 2021-07-22 16:41:26 +00:00
include Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
neteq Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
test Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00