webrtc/modules/audio_coding
Jim Gustafson 281e582847 Add function to check if packet represents speech
The original code assumed that one packet contains one frame, which is not
true anymore since multi-frame packets and DTX are now supported.

Includes an updated reference to signalapp/opus so that DTX frames are not
padded.
2023-05-12 09:01:29 -07:00
..
acm2 Delete deprecated Create method and config from AudioCodingModule 2023-02-02 17:06:29 +00:00
audio_network_adaptor Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
codecs Add function to check if packet represents speech 2023-05-12 09:01:29 -07:00
g3doc Reland "Migrate WebRTC documentation to new renderer" 2023-01-31 09:30:04 +00:00
include Delete deprecated Create method and config from AudioCodingModule 2023-02-02 17:06:29 +00:00
neteq Merge branch 'm112' into 5615 2023-04-27 12:45:13 -04:00
test Break apart AudioCodingModule and AcmReceiver 2023-02-01 16:09:26 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Use opus fork from signalapp/opus@webrtc 2023-05-12 08:49:19 -07:00
DEPS
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00