asterisk-opus: add variant for asterisk 16 412/head
authorAndre Heider <a.heider@gmail.com>
Fri, 12 Oct 2018 10:41:59 +0000 (12:41 +0200)
committerAndre Heider <a.heider@gmail.com>
Thu, 2 May 2019 08:53:12 +0000 (10:53 +0200)
Signed-off-by: Andre Heider <a.heider@gmail.com>
net/asterisk-opus/Makefile

index 36f4abd7940d150b633feb43dfcf753b420fadd8..f357ab13c94e18a8dc20565eedc2dcdd51321986 100644 (file)
@@ -53,6 +53,12 @@ $(call Package/$(PKG_NAME)/Default)
   VARIANT:=asterisk15
 endef
 
+define Package/asterisk16-codec-opus
+$(call Package/$(PKG_NAME)/Default)
+  DEPENDS+=asterisk16
+  VARIANT:=asterisk16
+endef
+
 define Package/description/Default
   Opus is the default audio codec in WebRTC. WebRTC is available in
   Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
@@ -69,6 +75,7 @@ endef
 
 Package/asterisk13-codec-opus/description = $(Package/description/Default)
 Package/asterisk15-codec-opus/description = $(Package/description/Default)
+Package/asterisk16-codec-opus/description = $(Package/description/Default)
 
 ifeq ($(BUILD_VARIANT),asterisk13)
 TARGET_CFLAGS+=-I$(STAGING_DIR)/usr/include/asterisk-13/include
@@ -78,6 +85,10 @@ ifeq ($(BUILD_VARIANT),asterisk15)
 TARGET_CFLAGS+=-I$(STAGING_DIR)/usr/include/asterisk-15/include
 endif
 
+ifeq ($(BUILD_VARIANT),asterisk16)
+TARGET_CFLAGS+=-I$(STAGING_DIR)/usr/include/asterisk-16/include
+endif
+
 define Package/Install/Default
        $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules
        $(INSTALL_BIN) $(PKG_BUILD_DIR)/codecs/codec_opus_open_source.so \
@@ -86,9 +97,11 @@ endef
 
 Package/asterisk13-codec-opus/install = $(Package/Install/Default)
 Package/asterisk15-codec-opus/install = $(Package/Install/Default)
+Package/asterisk16-codec-opus/install = $(Package/Install/Default)
 
 define Build/Configure
 endef
 
 $(eval $(call BuildPackage,asterisk13-codec-opus))
 $(eval $(call BuildPackage,asterisk15-codec-opus))
+$(eval $(call BuildPackage,asterisk16-codec-opus))