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
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$(STAGING_DIR)/usr/include/asterisk-13/include
TARGET_CFLAGS+=-I$(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 \
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))