From: Sebastian Kemper <sebastian_ml@gmx.net>
Date: Sun, 1 Nov 2020 10:56:02 +0000 (+0100)
Subject: asterisk-opus: update for Asterisk 18 LTS
X-Git-Url: http://git.lede-project.org./?a=commitdiff_plain;h=af20cf00a63352676a7bc89c7db36ad05fdb06b6;p=feed%2Ftelephony.git

asterisk-opus: update for Asterisk 18 LTS

Removes VARIANT-related bits as we only have one Asterisk version.

Signed-off-by: Sebastian Kemper <sebastian_ml@gmx.net>
---

diff --git a/net/asterisk-opus/Makefile b/net/asterisk-opus/Makefile
index c0cf5c0..4d67b20 100644
--- a/net/asterisk-opus/Makefile
+++ b/net/asterisk-opus/Makefile
@@ -8,46 +8,34 @@
 include $(TOPDIR)/rules.mk
 
 PKG_NAME:=asterisk-opus
-PKG_VERSION:=20171009
 PKG_RELEASE:=1
 
-PKG_SOURCE:=$(PKG_NAME)-$(PKG_VERSION).tar.xz
 PKG_SOURCE_URL:=https://github.com/traud/asterisk-opus.git
-PKG_SOURCE_SUBDIR:=$(PKG_NAME)-$(PKG_VERSION)
+PKG_SOURCE_DATE:=2017-10-09
 PKG_SOURCE_VERSION:=83e1b458c77e0e287adeca494eeb79edb077b0ff
-PKG_MIRROR_HASH:=89265ee63bc3cb02ef9e75087f782147db1dec53433f9d6cfc08c7229cd61881
+PKG_MIRROR_HASH:=c71b859db7518cdafff1650e629c5901b290fe68f8af54ef1afd57bc9f15b122
 PKG_SOURCE_PROTO:=git
 
-PKG_BUILD_DIR=$(BUILD_DIR)/$(PKG_NAME)-$(PKG_VERSION)
-
 PKG_LICENSE:=GPL-2.0
 PKG_LICENSE_FILES:=LICENSE
 PKG_MAINTAINER:=Jiri Slachta <jiri@slachta.eu>
 
-PKG_BUILD_DIR:=$(BUILD_DIR)/$(PKG_NAME)-$(BUILD_VARIANT)/$(PKG_NAME)-$(PKG_VERSION)
-
 include $(INCLUDE_DIR)/package.mk
 
 TARGET_CFLAGS += \
 	-DAST_MODULE_SELF_SYM=__internal_codec_opus_open_source_self \
 	$(FPIC)
 
-define Package/$(PKG_NAME)/Default
+define Package/asterisk-codec-opus
   SUBMENU:=Telephony
   SECTION:=net
   CATEGORY:=Network
   TITLE:=Opus codec support
   URL:=https://github.com/traud/asterisk-opus
-  DEPENDS:=+libopus
-endef
-
-define Package/asterisk16-codec-opus
-$(call Package/$(PKG_NAME)/Default)
-  DEPENDS+=asterisk16
-  VARIANT:=asterisk16
+  DEPENDS:=asterisk +libopus
 endef
 
-define Package/description/Default
+define Package/asterisk-codec-opus/description
   Opus is the default audio codec in WebRTC. WebRTC is available in
   Asterisk via SIP over WebSockets (WSS). Nevertheless, Opus can be used
   for other transports (UDP, TCP, TLS) as well. Opus supersedes previous
@@ -61,21 +49,13 @@ define Package/description/Default
   well.
 endef
 
-Package/asterisk16-codec-opus/description = $(Package/description/Default)
-
-ifeq ($(BUILD_VARIANT),asterisk16)
-TARGET_CFLAGS+=-I$(STAGING_DIR)/usr/include/asterisk-16/include
-endif
-
-define Package/Install/Default
+define Package/asterisk-codec-opus/install
 	$(INSTALL_DIR) $(1)/usr/lib/asterisk/modules
 	$(INSTALL_BIN) $(PKG_BUILD_DIR)/codecs/codec_opus_open_source.so \
 		$(1)/usr/lib/asterisk/modules
 endef
 
-Package/asterisk16-codec-opus/install = $(Package/Install/Default)
-
 define Build/Configure
 endef
 
-$(eval $(call BuildPackage,asterisk16-codec-opus))
+$(eval $(call BuildPackage,asterisk-codec-opus))