gstreamer/ext/webrtcdsp/Makefile.am
George Kiagiadakis d299c27892 webrtcdsp: add support for using F32/non-interleaved buffers
This is the native format that is in use by the webrtc audio processing
library internally, so this avoids internal {de,}interleaving and
format conversion (S16->F32 and back)

https://bugzilla.gnome.org/show_bug.cgi?id=793605
2018-08-03 13:20:12 +03:00

21 lines
628 B
Makefile

plugin_LTLIBRARIES = libgstwebrtcdsp.la
libgstwebrtcdsp_la_SOURCES = gstwebrtcdsp.cpp gstwebrtcechoprobe.cpp
libgstwebrtcdsp_la_CXXFLAGS = \
-std=c++11 \
$(GST_CXXFLAGS) \
$(GST_BASE_CFLAGS) \
$(GST_PLUGINS_BASE_CFLAGS) \
-I$(top_srcdir)/gst-libs \
-I$(top_builddir)/gst-libs \
$(WEBRTCDSP_CFLAGS)
libgstwebrtcdsp_la_LIBADD = \
-lgstaudio-$(GST_API_VERSION) \
$(top_builddir)/gst-libs/gst/audio/libgstbadaudio-$(GST_API_VERSION).la \
$(GST_LIBS) $(GST_BASE_LIBS) \
$(GST_PLUGINS_BASE_LIBS) \
$(WEBRTCDSP_LIBS)
libgstwebrtcdsp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
noinst_HEADERS = gstwebrtcdsp.h gstwebrtcechoprobe.h