webrtc: Replace dsp and echoprobe by gst webrtcdsp and webrtcechoprobe

.github/workflows/build.yml

@@ -22,7 +22,7 @@ jobs:
       - run: sudo apt-get update
       - run: sudo apt-get remove libunwind-14-dev
       - run: sudo apt-get install -y build-essential gettext libadwaita-1-dev libcanberra-dev libgcrypt20-dev libgee-0.8-dev libgpgme-dev libgstreamer-plugins-base1.0-dev libgstreamer1.0-dev libgtk-4-dev libnice-dev libnotify-dev libqrencode-dev libsignal-protocol-c-dev libsoup2.4-dev libsqlite3-dev libsrtp2-dev libwebrtc-audio-processing-dev meson valac
-      - run: meson setup build -Dcrypto-backend=auto -Dplugin-ice=enabled -Duse-soup2=true
+      - run: meson setup build -Duse-soup2=true -Dplugin-rtp-webrtc-audio-processing=disabled
       - run: meson compile -C build
       - run: meson test -C build
   build-flatpak:

CMakeLists.txt

@@ -12,6 +12,13 @@ endif ()
 
 include(CTest)
 
+option(PLUGIN_RTP_WEBRTC_AUDIO_PROCESSING "Use WebRTC audio processing" ON)
+
+# https://gitlab.kitware.com/cmake/cmake/-/issues/19804
+if (WIN32)
+    list(APPEND CMAKE_FIND_LIBRARY_SUFFIXES .dll)
+endif()
+
 # Prepare Plugins
 set(DEFAULT_PLUGINS omemo;openpgp;http-files;ice;rtp)
 if (WIN32)

cmake/FindGstWebrtcDsp.cmake

@@ -0,0 +1,11 @@
+find_library(GstWebrtcDsp_LIBRARY gstwebrtcdsp PATH_SUFFIXES gstreamer-1.0)
+
+if(GstWebrtcDsp_LIBRARY_FOUND)
+    find_package(Gst)
+    set(GstWebrtcDsp_VERSION ${Gst_VERSION})
+endif()
+
+include(FindPackageHandleStandardArgs)
+find_package_handle_standard_args(GstWebrtcDsp
+    REQUIRED_VARS GstWebrtcDsp_LIBRARY
+    VERSION_VAR GstWebrtcDsp_VERSION)

meson.build

@@ -58,6 +58,8 @@ else
     libsoup_version = '3.0'
 endif
 
+cc = meson.get_compiler('c')
+
 dep_gdk_pixbuf = dependency('gdk-pixbuf-2.0')
 dep_gee = dependency('gee-0.8')
 dep_gio = dependency('gio-2.0')
@@ -81,9 +83,12 @@ dep_libsrtp2 = dependency('libsrtp2', disabler: true, required: plugin_crypto)
 dep_libsignal_protocol_c = dependency('libsignal-protocol-c', version: ['>=2.3.2', '<2.3.4'], disabler: true, required: get_option('plugin-omemo'))
 dep_libsoup = dependency('libsoup-@0@'.format(libsoup_version), disabler: true, required: get_option('plugin-http-files'))
 dep_nice = dependency('nice', version: '>=0.1.15', disabler: true, required: get_option('plugin-ice'))
-dep_m = meson.get_compiler('c').find_library('m', required: false)
+dep_m = cc.find_library('m', required: false)
 dep_sqlite3 = dependency('sqlite3', version: '>=3.24')
-dep_webrtc_audio_processing = dependency('webrtc-audio-processing', version: ['>=0.2', '<0.4'], required: get_option('plugin-rtp-webrtc-audio-processing'))
+
+dep_gstreamer_bad = dependency('gstreamer-plugins-bad-1.0', disabler: true, required: get_option('plugin-rtp-webrtc-audio-processing'))
+gstpluginsdir = dep_gstreamer_bad.get_variable('pluginsdir')
+dep_webrtcdsp = cc.find_library('gstwebrtcdsp', dirs: gstpluginsdir, disabler: true, required: get_option('plugin-rtp-webrtc-audio-processing'))
 
 prog_git = find_program('git', required: false)
 prog_python = python.find_installation()

plugins/rtp/CMakeLists.txt

@@ -1,5 +1,13 @@
 find_package(GLib ${GLib_GLOBAL_VERSION} REQUIRED)
-find_package(WebRTCAudioProcessing 0.2)
+
+set(RTP_DEFINITIONS)
+set(RTP_EXTRA_OPTIONS)
+
+if(PLUGIN_RTP_WEBRTC_AUDIO_PROCESSING)
+    set(EXTRA_RTP_PACKAGES GstWebrtcDsp)
+    list(APPEND RTP_DEFINITIONS WITH_VOICE_PROCESSOR)
+endif()
+
 find_packages(RTP_PACKAGES REQUIRED
     Gee
     GLib
@@ -11,11 +19,9 @@ find_packages(RTP_PACKAGES REQUIRED
     GstAudio
     GstRtp
     GstVideo
+    ${EXTRA_RTP_PACKAGES}
 )
 
-set(RTP_DEFINITIONS)
-set(RTP_EXTRA_OPTIONS)
-
 if(GstRtp_VERSION VERSION_GREATER_EQUAL "1.16")
     set(RTP_DEFINITIONS ${RTP_DEFINITIONS} GST_1_16)
 endif()
@@ -52,20 +58,6 @@ if(RTP_ENABLE_MSDK)
     set(RTP_DEFINITIONS ${RTP_DEFINITIONS} ENABLE_MSDK)
 endif()
 
-if(WebRTCAudioProcessing_VERSION GREATER "0.4")
-    message(STATUS "Ignoring WebRTCAudioProcessing, only versions < 0.4 supported so far")
-    unset(WebRTCAudioProcessing_FOUND)
-endif()
-
-if(WebRTCAudioProcessing_FOUND)
-    set(RTP_DEFINITIONS ${RTP_DEFINITIONS} WITH_VOICE_PROCESSOR)
-    set(RTP_VOICE_PROCESSOR_VALA src/voice_processor.vala)
-    set(RTP_VOICE_PROCESSOR_CXX src/voice_processor_native.cpp)
-    set(RTP_VOICE_PROCESSOR_LIB webrtc-audio-processing)
-else()
-    message(STATUS "WebRTCAudioProcessing not found, build without voice pre-processing!")
-endif()
-
 vala_precompile(RTP_VALA_C
 SOURCES
     src/codec_util.vala

plugins/rtp/meson.build

@@ -30,12 +30,8 @@ c_args = [
 vala_args = [
     '--vapidir', meson.current_source_dir() / 'vapi',
 ]
-if dep_webrtc_audio_processing.found()
-    dependencies += [dep_webrtc_audio_processing]
-    sources += files(
-        'src/voice_processor.vala',
-        'src/voice_processor_native.cpp',
-    )
+if dep_webrtcdsp.found()
+    dependencies += [dep_webrtcdsp]
     vala_args += ['-D', 'WITH_VOICE_PROCESSOR']
 endif
 if dep_gstreamer_rtp.version() == 'unknown' or dep_gstreamer_rtp.version().version_compare('>=1.16')

plugins/rtp/src/device.vala

@@ -438,13 +438,12 @@ public class Dino.Plugins.Rtp.Device : MediaDevice, Object {
         Gst.Bin bin = new Gst.Bin("voiceprocessorbin");
         Gst.Element converter = Gst.ElementFactory.make("audioconvert", @"dsp_convert_$id");
         Gst.Element resampler = Gst.ElementFactory.make("audioresample", @"dsp_resmaple_$id");
-
-        Gst.Element voiceproc = new VoiceProcessor(plugin.echoprobe as EchoProbe, element as Gst.Audio.StreamVolume);
-        voiceproc.name = @"dsp_$id";
+        Gst.Element voiceproc = Gst.ElementFactory.make("webrtcdsp", @"dsp_$id");
 
         bin.add_many(converter, resampler, voiceproc);
         converter.link(resampler);
         resampler.link(voiceproc);
+        voiceproc.@set("probe", "webrtcechoprobe0");
 
         Gst.Pad sink_pad = bin.find_unlinked_pad(Gst.PadDirection.SINK);
         Gst.Pad src_pad = bin.find_unlinked_pad(Gst.PadDirection.SRC);

plugins/rtp/src/plugin.vala

@@ -59,13 +59,14 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
     }
 
 #if WITH_VOICE_PROCESSOR
-    private Gst.Element create_echo_bin(Gst.Element element) {
+    private Gst.Element create_echo_probe() {
         Gst.Bin bin = new Gst.Bin("echoprobebin");
         Gst.Element converter = Gst.ElementFactory.make("audioconvert", "echo_convert_");
         Gst.Element resampler = Gst.ElementFactory.make("audioresample", "echo_resample_");
+        Gst.Element webrtcechoprobe = Gst.ElementFactory.make("webrtcechoprobe", "webrtcechoprobe0");
 
-        bin.add_many(element, converter, resampler);
-        element.link(converter);
+        bin.add_many(webrtcechoprobe, converter, resampler);
+        webrtcechoprobe.link(converter);
         converter.link(resampler);
 
         Gst.Pad sink_pad = bin.find_unlinked_pad(Gst.PadDirection.SINK);
@@ -100,11 +101,8 @@ public class Dino.Plugins.Rtp.Plugin : RootInterface, VideoCallPlugin, Object {
 
 #if WITH_VOICE_PROCESSOR
         // Audio echo probe
-        echoprobe = new EchoProbe();
-        if (echoprobe != null) {
-            echoprobe = create_echo_bin(echoprobe);
-            pipe.add(echoprobe);
-        }
+        echoprobe = create_echo_probe();
+        pipe.add(echoprobe);
 #endif
 
         // Pipeline