mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-01-05 06:58:49 +00:00
d5755744c3
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>
26 lines
839 B
Meson
26 lines
839 B
Meson
webrtc_sources = [
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'gstwebrtcdsp.cpp',
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'gstwebrtcechoprobe.cpp',
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'gstwebrtcdspplugin.cpp'
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]
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webrtc_dep = dependency('webrtc-audio-processing-1', version : ['>= 1.0'],
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required : get_option('webrtcdsp'))
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if not gnustl_dep.found() and get_option('webrtcdsp').enabled()
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error('webrtcdsp plugin enabled but could not find gnustl dep for Android c++ support')
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endif
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if webrtc_dep.found() and gnustl_dep.found()
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gstwebrtcdsp = library('gstwebrtcdsp',
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webrtc_sources,
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cpp_args : gst_plugins_bad_args,
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link_args : noseh_link_args,
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include_directories : [configinc],
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dependencies : [gstbase_dep, gstaudio_dep, gstbadaudio_dep, webrtc_dep, gnustl_dep],
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install : true,
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install_dir : plugins_install_dir,
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override_options : ['cpp_std=c++17'],
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)
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plugins += [gstwebrtcdsp]
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endif
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