[asterisk-commits] oej: branch group/pinefool-poor-mans-plc-1.4 r383505 - in /team/group/pinefoo...

[SVN commits to the Asterisk project]

Author: oej
Date: Thu Mar 21 05:28:49 2013
New Revision: 383505

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=383505
Log:
Make PLC configurable in sip.conf (backport from 1.8 strangely enough)

Modified:
    team/group/pinefool-poor-mans-plc-1.4/channels/chan_sip.c
    team/group/pinefool-poor-mans-plc-1.4/configs/sip.conf.sample

Modified: team/group/pinefool-poor-mans-plc-1.4/channels/chan_sip.c
URL: http://svnview.digium.com/svn/asterisk/team/group/pinefool-poor-mans-plc-1.4/channels/chan_sip.c?view=diff&rev=383505&r1=383504&r2=383505
==============================================================================
--- team/group/pinefool-poor-mans-plc-1.4/channels/chan_sip.c (original)
+++ team/group/pinefool-poor-mans-plc-1.4/channels/chan_sip.c Thu Mar 21 05:28:49 2013
@@ -797,6 +797,7 @@
 #define SIP_PAGE2_RTAUTOCLEAR		(1 << 2)
 #define SIP_PAGE2_RT_FROMCONTACT 	(1 << 4)
 #define SIP_PAGE2_RTSAVE_SYSNAME 	(1 << 5)
+#define SIP_PAGE2_POORMANSPLC		(1 << 6)
 /* Space for addition of other realtime flags in the future */
 #define SIP_PAGE2_STATECHANGEQUEUE	(1 << 9)	/*!< D: Unsent state pending change exists */
 #define SIP_PAGE2_IGNOREREGEXPIRE	(1 << 10)
@@ -829,7 +830,7 @@
 #define SIP_PAGE2_FLAGS_TO_COPY \
 	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
 	SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | \
-	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_FORWARD_LOOP_DETECTED)
+	SIP_PAGE2_UDPTL_DESTINATION | SIP_PAGE2_FORWARD_LOOP_DETECTED | SIP_PAGE2_POORMANSPLC)
 
 /* SIP packet flags */
 #define SIP_PKT_DEBUG		(1 << 0)	/*!< Debug this packet */
@@ -4900,6 +4901,7 @@
 			free(p);
 			return NULL;
 		}
+		ast_rtp_set_plc(p->rtp, ast_test_flag(&p->flags[1], SIP_DTMF) == SIP_PAGE2_POORMANSPLC);
 		ast_rtp_setdtmf(p->rtp, ast_test_flag(&p->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
 		ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
 		ast_rtp_settos(p->rtp, global_tos_audio);
@@ -18281,6 +18283,9 @@
 	} else if (!strcasecmp(v->name, "forwardloopdetected")) {
 		ast_set_flag(&mask[1], SIP_PAGE2_FORWARD_LOOP_DETECTED);
 		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_FORWARD_LOOP_DETECTED);
+	} else if (!strcasecmp(v->name, "plc")) {
+		ast_set_flag(&mask[1], SIP_PAGE2_POORMANSPLC);
+		ast_set2_flag(&flags[1], ast_true(v->value), SIP_PAGE2_POORMANSPLC);
 	} else
 		res = 0;
 

Modified: team/group/pinefool-poor-mans-plc-1.4/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/group/pinefool-poor-mans-plc-1.4/configs/sip.conf.sample?view=diff&rev=383505&r1=383504&r2=383505
==============================================================================
--- team/group/pinefool-poor-mans-plc-1.4/configs/sip.conf.sample (original)
+++ team/group/pinefool-poor-mans-plc-1.4/configs/sip.conf.sample Thu Mar 21 05:28:49 2013
@@ -489,6 +489,9 @@
                                 ; it may be a mandatory requirement for some
                                 ; destinations which do not have a prior
                                 ; account relationship with your server. 
+
+; plc=yes			; Enable Poor Man's Packet Loss Concealment in RTP/Audio channels
+				; Settable in device-level too.
 
 ;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
 ; jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a