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webrtc/modules/audio_coding
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Philipp Hancke 82e5f91a2b audio: fix handling of RED packets where the primary encoding is too large 
by falling back to the primary encoding. This can happen with
opus stereo packets at the maximum bitrate which results in
1276 encoded bytes.

BUG=chromium:1470261

Change-Id: I3fd9bb30773963a519bbb5da44fe71db5dec2bd7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/315141
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40524}
2023-08-08 13:40:26 +00:00
..
acm2 Delete RTPHeader::payload_type_frequency as unused 2023-03-09 16:32:22 +00:00
audio_network_adaptor Remove mentions of already deleted field trials 2023-03-01 15:53:37 +00:00
codecs audio: fix handling of RED packets where the primary encoding is too large 2023-08-08 13:40:26 +00:00
g3doc Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
include Format /modules 2023-04-20 02:02:45 +00:00
neteq Make packet info optional and only set for primary packets in NetEq. 2023-06-07 18:17:03 +00:00
test Format /modules 2023-04-20 02:02:45 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Refactor NetEq rtp dump input. 2023-04-11 14:32:35 +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