TARGET_CFLAGS += \
$(FPIC)
-define Package/asterisk-codec-opus
+define Package/asterisk-opus/Default
SUBMENU:=Telephony
SECTION:=net
CATEGORY:=Network
- TITLE:=Opus codec support
URL:=https://github.com/traud/asterisk-opus
DEPENDS:=asterisk +libopus
endef
+define Package/asterisk-codec-opus
+$(call Package/asterisk-opus/Default)
+ TITLE:=Opus codec support
+endef
+
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
$(1)/usr/lib/asterisk/modules
endef
+define Package/asterisk-format-ogg-opus
+$(call Package/asterisk-opus/Default)
+ TITLE:=OGG/Opus audio support
+ DEPENDS+=+libopusfile +libopusenc
+endef
+
+define Package/asterisk-format-ogg-opus/description
+ Reading and writing audio files in the OGG/Opus format.
+endef
+
+define Package/asterisk-format-ogg-opus/install
+ $(INSTALL_DIR) $(1)/usr/lib/asterisk/modules
+ $(INSTALL_BIN) $(PKG_BUILD_DIR)/formats/format_ogg_opus_open_source.so \
+ $(1)/usr/lib/asterisk/modules
+endef
+
define Build/Configure
endef
$(eval $(call BuildPackage,asterisk-codec-opus))
+$(eval $(call BuildPackage,asterisk-format-ogg-opus))
ASTMODDIR=$(libdir)/asterisk/modules
-MODULES=codec_opus_open_source format_ogg_opus_open_source format_vp8 res_format_attr_opus
-+MODULES=codec_opus_open_source
++MODULES=codec_opus_open_source format_ogg_opus_open_source
.SUFFIXES: .c .so
+@@ -38,7 +38,7 @@ codec_opus_open_source: DEFS+=-DAST_MODU
+ -DAST_MODULE_SELF_SYM=__internal_codec_opus_open_source_self
+ codec_opus_open_source: codecs/codec_opus_open_source.so
+
+-format_ogg_opus_open_source: CPATH+=-I/usr/include/opus
++format_ogg_opus_open_source: CPATH+=-I$(STAGING_DIR)/usr/include/opus
+ format_ogg_opus_open_source: LIBS+=-lopus -lopusfile
+ format_ogg_opus_open_source: DEFS+=-DAST_MODULE=\"format_ogg_opus_open_source\" \
+ -DAST_MODULE_SELF_SYM=__internal_format_ogg_opus_open_source_self