| webrtc_sources = [ |
| 'gstwebrtcdsp.cpp', |
| 'gstwebrtcechoprobe.cpp' |
| ] |
| |
| webrtc_dep = dependency('webrtc-audio-processing', version : '>= 0.2', required : false) |
| webrtc_max_dep = dependency('webrtc-audio-processing', version : '>= 0.4', required : false) |
| |
| if (webrtc_max_dep.found()) |
| message('WebRTC Audio Processing library is not API stable,' |
| + ' we cannot support newer version ' + webrtc_max_dep.version() |
| + ' (we only support 0.2 and 0.3)') |
| elif (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, webrtc_dep], |
| install : true, |
| install_dir : plugins_install_dir, |
| ) |
| endif |
