From baffecda1678d68ab6d16514ae1903d2d4dc1e5d Mon Sep 17 00:00:00 2001 From: Peter Seiderer Date: Sat, 16 Mar 2019 23:14:40 +0100 Subject: [PATCH] package/gst1-plugins-bad: add webrtc option - remove old webrtc Config.in.legacy entry introduced by [1] (misnamed webrtc option was introduced with 2017.02, renamed to webrtcdsp for 2017.08 and although backported to 2017.02.4) [1] https://git.buildroot.net/buildroot/commit/?id=4c06d2490a07f0b88f42c56c7409899fd2f5608a Signed-off-by: Peter Seiderer Signed-off-by: Thomas Petazzoni --- Config.in.legacy | 10 ---------- package/gstreamer1/gst1-plugins-bad/Config.in | 11 +++++++++++ .../gstreamer1/gst1-plugins-bad/gst1-plugins-bad.mk | 7 +++++++ 3 files changed, 18 insertions(+), 10 deletions(-) diff --git a/Config.in.legacy b/Config.in.legacy index 2030b51e1d..0a0bce41f9 100644 --- a/Config.in.legacy +++ b/Config.in.legacy @@ -1685,16 +1685,6 @@ config BR2_PACKAGE_GST1_PLUGINS_UGLY_PLUGIN_MAD bool "mad (*.mp3 audio) removed" select BR2_LEGACY -config BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBRTC - bool "gst1-plugins-bad webrtc renamed to webrtcdsp" - select BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBRTCDSP - select BR2_LEGACY - help - The WebRTC plugin in GStreamer 1.x has always been named - webrtcdsp, but was wrongly introduced in Buildroot under the - name webrtc. Therefore, we have renamed the option to match - the actual name of the GStreamer plugin. - config BR2_STRIP_none bool "Strip command 'none' has been removed" select BR2_LEGACY diff --git a/package/gstreamer1/gst1-plugins-bad/Config.in b/package/gstreamer1/gst1-plugins-bad/Config.in index 869f0a9d45..6830902f3a 100644 --- a/package/gstreamer1/gst1-plugins-bad/Config.in +++ b/package/gstreamer1/gst1-plugins-bad/Config.in @@ -558,6 +558,17 @@ config BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBP help Webp image format plugin +config BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBRTC + bool "webrtc" + depends on !BR2_STATIC_LIBS # libnice -> gnutls + select BR2_PACKAGE_GST1_PLUGINS_BASE # libgstsdp + select BR2_PACKAGE_LIBNICE + help + WebRTC plugins (webrtcbin - a bin for webrtc connections) + +comment "webrtc needs a toolchain w/ dynamic library" + depends on BR2_STATIC_LIBS + config BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBRTCDSP bool "webrtcdsp" # All depends from webrtc-audio-processing diff --git a/package/gstreamer1/gst1-plugins-bad/gst1-plugins-bad.mk b/package/gstreamer1/gst1-plugins-bad/gst1-plugins-bad.mk index f5b081f972..e4e7661ac4 100644 --- a/package/gstreamer1/gst1-plugins-bad/gst1-plugins-bad.mk +++ b/package/gstreamer1/gst1-plugins-bad/gst1-plugins-bad.mk @@ -703,6 +703,13 @@ else GST1_PLUGINS_BAD_CONF_OPTS += --disable-webp endif +ifeq ($(BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBRTC),y) +GST1_PLUGINS_BAD_CONF_OPTS += --enable-webrtc +GST1_PLUGINS_BAD_DEPENDENCIES += gst1-plugins-base libnice +else +GST1_PLUGINS_BAD_CONF_OPTS += --disable-webrtc +endif + ifeq ($(BR2_PACKAGE_GST1_PLUGINS_BAD_PLUGIN_WEBRTCDSP),y) GST1_PLUGINS_BAD_CONF_OPTS += --enable-webrtcdsp GST1_PLUGINS_BAD_DEPENDENCIES += webrtc-audio-processing -- 2.30.2