* [Buildroot] [PATCH v2] package/gst1-plugins-bad: add webrtc option
@ 2019-03-16 22:14 Peter Seiderer
2019-03-18 21:19 ` Thomas Petazzoni
0 siblings, 1 reply; 2+ messages in thread
From: Peter Seiderer @ 2019-03-16 22:14 UTC (permalink / raw)
To: buildroot
- 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 <ps.report@gmx.net>
---
Changes v1 -> v2:
- renamed option form webrtcbin to webrtc
- remove legacy webrtc Config.in.legacy entry
---
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
--
2.21.0
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