gstreamer/subprojects/gst-plugins-bad/ext/webrtcdsp/meson.build
Arun Raghavan d5755744c3 webrtcdsp: Update code for webrtc-audio-processing-1
Updated API usage appropriately, and now we have a versioned package to
track breaking vs. non-breaking updates.

Deprecates a number of properties (and we have to plug in our own values
for related enums which are now gone):

  * echo-suprression-level
  * experimental-agc
  * extended-filter
  * delay-agnostic
  * voice-detection-frame-size-ms
  * voice-detection-likelihood

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2943>
2023-06-01 09:34:37 +00:00

26 lines
839 B
Meson

webrtc_sources = [
'gstwebrtcdsp.cpp',
'gstwebrtcechoprobe.cpp',
'gstwebrtcdspplugin.cpp'
]
webrtc_dep = dependency('webrtc-audio-processing-1', version : ['>= 1.0'],
required : get_option('webrtcdsp'))
if not gnustl_dep.found() and get_option('webrtcdsp').enabled()
error('webrtcdsp plugin enabled but could not find gnustl dep for Android c++ support')
endif
if webrtc_dep.found() and gnustl_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, gnustl_dep],
install : true,
install_dir : plugins_install_dir,
override_options : ['cpp_std=c++17'],
)
plugins += [gstwebrtcdsp]
endif