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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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d299c27892
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
20 lines
628 B
Makefile
20 lines
628 B
Makefile
plugin_LTLIBRARIES = libgstwebrtcdsp.la
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libgstwebrtcdsp_la_SOURCES = gstwebrtcdsp.cpp gstwebrtcechoprobe.cpp
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libgstwebrtcdsp_la_CXXFLAGS = \
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-std=c++11 \
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$(GST_CXXFLAGS) \
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$(GST_BASE_CFLAGS) \
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$(GST_PLUGINS_BASE_CFLAGS) \
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-I$(top_srcdir)/gst-libs \
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-I$(top_builddir)/gst-libs \
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$(WEBRTCDSP_CFLAGS)
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libgstwebrtcdsp_la_LIBADD = \
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-lgstaudio-$(GST_API_VERSION) \
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$(top_builddir)/gst-libs/gst/audio/libgstbadaudio-$(GST_API_VERSION).la \
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$(GST_LIBS) $(GST_BASE_LIBS) \
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$(GST_PLUGINS_BASE_LIBS) \
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$(WEBRTCDSP_LIBS)
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libgstwebrtcdsp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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noinst_HEADERS = gstwebrtcdsp.h gstwebrtcechoprobe.h
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