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webrtc_m130/audio/voip
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Tim Na 2412602b68 Using absl::optional for round trip time return type handling. 
No-Try: True
Bug: webrtc:11989
Change-Id: If2ed9b83468c03b82b372e64d8012e5786295476
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197060
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32827}
2020-12-15 08:38:12 +00:00
..
test
Enforcing return type handling on VoIP API.
2020-12-11 20:38:15 +00:00
audio_channel.cc
VoipVolumeControl subAPI for VoIP API
2020-11-13 19:27:12 +00:00
audio_channel.h
VoipVolumeControl subAPI for VoIP API
2020-11-13 19:27:12 +00:00
audio_egress.cc
VoipVolumeControl subAPI for VoIP API
2020-11-13 19:27:12 +00:00
audio_egress.h
Locating input audio level before TaskQueue.
2020-11-17 20:53:36 +00:00
audio_ingress.cc
Using absl::optional for round trip time return type handling.
2020-12-15 08:38:12 +00:00
audio_ingress.h
Using absl::optional for round trip time return type handling.
2020-12-15 08:38:12 +00:00
BUILD.gn
Using absl::optional for round trip time return type handling.
2020-12-15 08:38:12 +00:00
voip_core.cc
Defining API result types on VoIP API
2020-11-20 18:02:05 +00:00
voip_core.h
Defining API result types on VoIP API
2020-11-20 18:02:05 +00:00