meson: rename the option from 'webrtc-audio-processing' to 'aec'

Change-Id: I69cc9da36dfbb1258f8f84e9351a6c1c29c2a722

meson.build

@@ -80,9 +80,6 @@ conf.set10('HAVE_PULSE', deplibpulse.found())
 depjack = dependency('jack', required: get_option('jack'))
 conf.set10('HAVE_JACK', depjack.found())
 
-depwebrtcap = dependency('webrtc-audio-processing', required: get_option('webrtc-audio-processing'))
-conf.set10('HAVE_WEBRTC_AP', depwebrtcap.found())
-
 depportaudio = dependency('portaudio-2.0', required: get_option('portaudio'))
 conf.set10('HAVE_PORTAUDIO', depportaudio.found())
 
@@ -107,6 +104,9 @@ conf.set10('HAVE_LIBNATPMP', depnatpmp.found())
 depopenssl = dependency('openssl', required: get_option('name_service'))
 conf.set10('HAVE_RINGNS', depopenssl.found())
 
+depwebrtcap = dependency('webrtc-audio-processing', required: get_option('aec'))
+conf.set10('HAVE_WEBRTC_AP', depwebrtcap.found())
+
 if get_option('video')
     conf.set('ENABLE_VIDEO', true)
     if host_machine.system() == 'linux' and meson.get_compiler('cpp').get_define('__ANDROID__') != '1'

meson_options.txt

@@ -11,7 +11,7 @@ option('jack', type: 'feature', value: 'auto', description: 'Enable support for
 option('portaudio', type: 'feature', value: 'auto', description: 'Enable support for PortAudio')
 option('upnp', type: 'feature', value: 'auto', description: 'Enable support for UPnP')
 option('natpmp', type: 'feature', value: 'auto', description: 'Enable support for NAT-PMP')
-option('webrtc-audio-processing', type: 'feature', value: 'auto', description: 'Enable support for WebRtc audio processing')
+option('aec', type: 'feature', value: 'auto', description: 'Enable Acoustic Echo Cancellation')
 
 option('natpmp_prefix', type: 'string', value: '', description: 'Override a system directory to search for the library "natpmp"')
 option('tests', type: 'boolean', value: false, description: 'Build tests')

src/meson.build

@@ -32,6 +32,7 @@ libjami_sources = files(
     'jamidht/p2p.cpp',
     'jamidht/server_account_manager.cpp',
     'jamidht/sips_transport_ice.cpp',
+    'media/audio/echo-cancel/null_echo_canceller.cpp',
     'media/audio/sound/audiofile.cpp',
     'media/audio/sound/dtmf.cpp',
     'media/audio/sound/dtmfgenerator.cpp',
@@ -45,9 +46,6 @@ libjami_sources = files(
     'media/audio/audio_receive_thread.cpp',
     'media/audio/audio_rtp_session.cpp',
     'media/audio/audio_sender.cpp',
-    'media/audio/echo-cancel/echo_canceller.h',
-    'media/audio/echo-cancel/null_echo_canceller.h',
-    'media/audio/echo-cancel/null_echo_canceller.cpp',
     'media/audio/dcblocker.cpp',
     'media/audio/dsp.cpp',
     'media/audio/resampler.cpp',
@@ -162,14 +160,6 @@ if conf.get('HAVE_PULSE') == 1
     libjami_dependencies += deplibpulse
 endif
 
-if conf.get('HAVE_WEBRTC_AP') == 1
-    libjami_sources += files(
-        'media/audio/echo-cancel/webrtc_echo_canceller.h',
-        'media/audio/echo-cancel/webrtc_echo_canceller.cpp'
-    )
-    libjami_dependencies += depwebrtcap
-endif
-
 if conf.get('HAVE_JACK') == 1
     libjami_sources += files(
         'media/audio/jack/jacklayer.cpp'
@@ -220,6 +210,13 @@ if conf.get('HAVE_RINGNS') == 1
     libjami_dependencies += depopenssl
 endif
 
+if conf.get('HAVE_WEBRTC_AP') == 1
+    libjami_sources += files(
+        'media/audio/echo-cancel/webrtc_echo_canceller.cpp'
+    )
+    libjami_dependencies += depwebrtcap
+endif
+
 if conf.get('ENABLE_VIDEO')
     libjami_sources += files(
         'client/videomanager.cpp',