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
632 B
Meson

webrtc_sources = [
'gstwebrtcdsp.cpp',
'gstwebrtcechoprobe.cpp'
]
webrtc_dep = dependency('webrtc-audio-processing', version : ['>= 0.2', '< 0.4'],
required : get_option('webrtcdsp'))
if webrtc_dep.found()
gstwebrtcdsp = library('gstwebrtcdsp',
webrtc_sources,
cpp_args : gst_plugins_bad_args,
link_args : noseh_link_args,
include_directories : [configinc],
dependencies : [gstbase_dep, gstaudio_dep, gstbadaudio_dep, webrtc_dep],
install : true,
install_dir : plugins_install_dir,
)
pkgconfig.generate(gstwebrtcdsp, install_dir : plugins_pkgconfig_install_dir)
endif