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Use Opus for VoIP

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Server/client VoIP protocol is handled by adding new cvars
cl_voipProtocol and sv_voipProtocol, sv_voip and cl_voip
are used to auto set/clear them. All users need to touch
are cl/sv_voip as 0 or 1 just like before.

Old Speex VoIP packets in demos are skipped.
New VoIP packets are skipped in demos if sv_voipProtocol
doesn't match cl_voipProtocol.

Notable difference between usage of speex and opus codecs,
when using Speex client would be sent 80ms at a time.
Using Opus, 60ms is sent at a time. This was changed because
the Opus codec supports encoding up to 60ms at a time.
(Simpler to send only one codec frame in a packet.)
This commit is contained in:
Zack Middleton 2013-12-10 21:14:13 -06:00
parent fe619680f8
commit 615b73288f
13 changed files with 167 additions and 240 deletions
Download patch file Download diff file Expand all files Collapse all files

2
code/server/sv_snapshot.c
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@ -547,7 +547,7 @@ static void SV_WriteVoipToClient(client_t *cl, msg_t *msg)
if (totalbytes > (msg->maxsize - msg->cursize) / 2)
break;
MSG_WriteByte(msg, svc_voip);
MSG_WriteByte(msg, svc_voipOpus);
MSG_WriteShort(msg, packet->sender);
MSG_WriteByte(msg, (byte) packet->generation);
MSG_WriteLong(msg, packet->sequence);