webrtc: Replace dsp and echoprobe by gst webrtcdsp and webrtcechoprobe

.github/workflows/build.yml

@@ -9,8 +9,8 @@ jobs:
           fetch-depth: 0
       - run: sudo apt-get update
       - run: sudo apt-get remove libunwind-14-dev
-      - run: sudo apt-get install -y build-essential gettext cmake valac libgee-0.8-dev libsqlite3-dev libgtk-4-dev libnotify-dev libgpgme-dev libsoup2.4-dev libgcrypt20-dev libqrencode-dev libnice-dev libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev libsrtp2-dev libwebrtc-audio-processing-dev libadwaita-1-dev libsignal-protocol-c-dev libcanberra-dev
+      - run: sudo apt-get install -y build-essential gettext cmake valac libgee-0.8-dev libsqlite3-dev libgtk-4-dev libnotify-dev libgpgme-dev libsoup2.4-dev libgcrypt20-dev libqrencode-dev libnice-dev libgstreamer1.0-dev libgstreamer-plugins-base1.0-dev libsrtp2-dev libadwaita-1-dev libsignal-protocol-c-dev libcanberra-dev
-      - run: ./configure --release --no-debug --with-tests --enable-plugin=notification-sound --prefix=/usr
+      - run: ./configure --release --no-debug --with-tests --enable-plugin=notification-sound --prefix=/usr --without-webrtc
       - run: cmake --build build
       - run: cmake --build build --target=test
       - name: Build DEB-package
@@ -38,8 +38,8 @@ jobs:
           fetch-depth: 0
       - 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: 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 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)

README.md

@@ -74,7 +74,7 @@ Make sure to install all [dependencies](https://github.com/dino/dino/wiki/Build#
 
 If you want to use `meson` build system, follow the next instructions:
 
-    meson setup build -Dcrypto-backend=gnutls -Dplugin-ice=enabled
+    meson setup build
     meson configure --prefix $PWD/build/install --libdir lib build
     meson compile -C build
     meson install -C build
@@ -85,6 +85,12 @@ If your `nice` library depends on `libsoup-2.4` (consider `ldd` output for the `
 Skip `meson configure` step, if you want to install the program globally.
 You can specify any convenient directory in the option `--prefix` where the program will be installed.
 
+If there is no `webrtcdsp` plugin in your system (check this by calling `gst-inspect-1.0 webrtcdsp`) you should pass extra argument:
+
+* `--without-webrtcdsp` for `./configure`;
+* `-Dplugin-rtp-webrtc-audio-processing=disabled` for `meson`;
+* `-DPLUGIN_RTP_WEBRTC_AUDIO_PROCESSING=OFF` for `cmake`.
+
 In addition, there is a git version of this package for **Arch Linux** on [AUR](https://aur.archlinux.org/packages/dino-plus-git)
 
 Build on Windows (x86_64)

build-win64.sh

@@ -74,7 +74,6 @@ prepare()
        mingw64/mingw-w64-x86_64-nsis \
        mingw64/mingw-w64-x86_64-libsignal-protocol-c \
        mingw64/mingw-w64-x86_64-icu \
-       mingw64/mingw-w64-x86_64-webrtc-audio-processing \
        mingw64/mingw-w64-x86_64-meson \
        git \
        make \

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)

cmake/FindWebRTCAudioProcessing.cmake

@@ -1,12 +0,0 @@
-include(PkgConfigWithFallback)
-find_pkg_config_with_fallback(WebRTCAudioProcessing
-    PKG_CONFIG_NAME webrtc-audio-processing
-    LIB_NAMES webrtc_audio_processing
-    INCLUDE_NAMES webrtc/modules/audio_processing/include/audio_processing.h
-    INCLUDE_DIR_SUFFIXES webrtc-audio-processing webrtc_audio_processing
-)
-
-include(FindPackageHandleStandardArgs)
-find_package_handle_standard_args(WebRTCAudioProcessing
-    REQUIRED_VARS WebRTCAudioProcessing_LIBRARY
-    VERSION_VAR WebRTCAudioProcessing_VERSION)

configure

@@ -1,7 +1,7 @@
 #!/bin/sh
 
 OPTS=`getopt -o "h" --long \
-help,fetch-only,no-debug,disable-fast-vapi,with-tests,release,with-libsoup3,\
+help,fetch-only,no-debug,disable-fast-vapi,with-tests,release,with-libsoup3,without-webrtcdsp,\
 enable-plugin:,disable-plugin:,\
 prefix:,program-prefix:,exec-prefix:,lib-suffix:,\
 bindir:,libdir:,includedir:,datadir:,\
@@ -21,6 +21,7 @@ DISABLE_FAST_VAPI=
 LIB_SUFFIX=
 NO_DEBUG=
 USE_SOUP3=
+PLUGIN_RTP_WEBRTC_AUDIO_PROCESSING=yes
 
 EXEC_PREFIX=
 BINDIR=
@@ -54,6 +55,7 @@ Configuration:
   --no-debug                 Build without debug symbols
   --release                  Configure to build an optimized release version
   --with-libsoup3            Build with libsoup-3.0
+  --without-webrtcdsp        Build without WebRTC audio processing
   --with-tests               Also build tests.
 
 Plugin configuration:
@@ -109,6 +111,7 @@ while true; do
         --valac-flags ) VALACFLAGS="$2"; shift; shift ;;
         --lib-suffix ) LIB_SUFFIX="$2"; shift; shift ;;
         --with-libsoup3 ) USE_SOUP3=yes; shift ;;
+        --without-webrtcdsp ) PLUGIN_RTP_WEBRTC_AUDIO_PROCESSING=no; shift ;;
         --disable-fast-vapi ) DISABLE_FAST_VAPI=yes; shift ;;
         --no-debug ) NO_DEBUG=yes; shift ;;
         --release ) BUILD_TYPE=RelWithDebInfo; shift ;;
@@ -219,6 +222,7 @@ cmake -G "$cmake_type" \
     -DBIN_INSTALL_DIR="$BINDIR" \
     -DINCLUDE_INSTALL_DIR="$INCLUDEDIR" \
     -DLIB_INSTALL_DIR="$LIBDIR" \
+    -DPLUGIN_RTP_WEBRTC_AUDIO_PROCESSING="$PLUGIN_RTP_WEBRTC_AUDIO_PROCESSING" \
     -Wno-dev \
     .. || exit 9
 

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()
+if dep_webrtcdsp.found()
-    dependencies += [dep_webrtc_audio_processing]
+    dependencies += [dep_webrtcdsp]
-    sources += files(
-        'src/voice_processor.vala',
-        'src/voice_processor_native.cpp',
-    )
     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

@@ -475,13 +475,12 @@ 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 = Gst.ElementFactory.make("webrtcdsp", @"dsp_$id");
-        Gst.Element voiceproc = new VoiceProcessor(plugin.echoprobe as EchoProbe, element as Gst.Audio.StreamVolume);
-        voiceproc.name = @"dsp_$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);
+        bin.add_many(webrtcechoprobe, converter, resampler);
-        element.link(converter);
+        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();
+        echoprobe = create_echo_probe();
-        if (echoprobe != null) {
+        pipe.add(echoprobe);
-            echoprobe = create_echo_bin(echoprobe);
-            pipe.add(echoprobe);
-        }
 #endif
 
         // Pipeline

plugins/rtp/src/voice_processor.vala

@@ -1,176 +0,0 @@
-using Gst;
-
-namespace Dino.Plugins.Rtp {
-public static extern Buffer adjust_to_running_time(Base.Transform transform, Buffer buf);
-}
-
-public class Dino.Plugins.Rtp.EchoProbe : Audio.Filter {
-    private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
-    private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
-    public Audio.Info audio_info { get; private set; }
-    public signal void on_new_buffer(Buffer buffer);
-    private uint period_samples;
-    private uint period_size;
-    private Base.Adapter adapter = new Base.Adapter();
-
-    static construct {
-        add_static_pad_template(sink_template);
-        add_static_pad_template(src_template);
-        set_static_metadata("Acoustic Echo Canceller probe", "Generic/Audio", "Gathers playback buffers for echo cancellation", "Dino Team <contact@dino.im>");
-    }
-
-    construct {
-        set_passthrough(true);
-    }
-
-    public override bool setup(Audio.Info info) {
-        audio_info = info;
-        period_samples = info.rate / 100; // 10ms buffers
-        period_size = period_samples * info.bpf;
-        return true;
-    }
-
-
-    public override FlowReturn transform_ip(Buffer buf) {
-        lock (adapter) {
-            adapter.push(adjust_to_running_time(this, buf));
-            while (adapter.available() > period_size) {
-                on_new_buffer(adapter.take_buffer(period_size));
-            }
-        }
-        return FlowReturn.OK;
-    }
-
-    public override bool stop() {
-        adapter.clear();
-        return true;
-    }
-}
-
-public class Dino.Plugins.Rtp.VoiceProcessor : Audio.Filter {
-    private static StaticPadTemplate sink_template = {"sink", PadDirection.SINK, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
-    private static StaticPadTemplate src_template = {"src", PadDirection.SRC, PadPresence.ALWAYS, {null, "audio/x-raw,rate=48000,channels=1,layout=interleaved,format=S16LE"}};
-    public Audio.Info audio_info { get; private set; }
-    private ulong process_outgoing_buffer_handler_id;
-    private uint adjust_delay_timeout_id;
-    private uint period_samples;
-    private uint period_size;
-    private Base.Adapter adapter = new Base.Adapter();
-    private EchoProbe? echo_probe;
-    private Audio.StreamVolume? stream_volume;
-    private ClockTime last_reverse;
-    private void* native;
-
-    static construct {
-        add_static_pad_template(sink_template);
-        add_static_pad_template(src_template);
-        set_static_metadata("Voice Processor (AGC, AEC, filters, etc.)", "Generic/Audio", "Pre-processes voice with WebRTC Audio Processing Library", "Dino Team <contact@dino.im>");
-    }
-
-    construct {
-        set_passthrough(false);
-    }
-
-    public VoiceProcessor(EchoProbe? echo_probe = null, Audio.StreamVolume? stream_volume = null) {
-        this.echo_probe = echo_probe;
-        this.stream_volume = stream_volume;
-    }
-
-    private static extern void* init_native(int stream_delay);
-    private static extern void setup_native(void* native);
-    private static extern void destroy_native(void* native);
-    private static extern void analyze_reverse_stream(void* native, Audio.Info info, Buffer buffer);
-    private static extern void process_stream(void* native, Audio.Info info, Buffer buffer);
-    private static extern void adjust_stream_delay(void* native);
-    private static extern void notify_gain_level(void* native, int gain_level);
-    private static extern int get_suggested_gain_level(void* native);
-    private static extern bool get_stream_has_voice(void* native);
-
-    public override bool setup(Audio.Info info) {
-        debug("VoiceProcessor.setup(%s)", info.to_caps().to_string());
-        audio_info = info;
-        period_samples = info.rate / 100; // 10ms buffers
-        period_size = period_samples * info.bpf;
-        adapter.clear();
-        setup_native(native);
-        return true;
-    }
-
-    public override bool start() {
-        native = init_native(150);
-        if (process_outgoing_buffer_handler_id == 0 && echo_probe != null) {
-            process_outgoing_buffer_handler_id = echo_probe.on_new_buffer.connect(process_outgoing_buffer);
-        }
-        if (stream_volume == null && sinkpad.get_peer() != null && sinkpad.get_peer().get_parent_element() is Audio.StreamVolume) {
-            stream_volume = sinkpad.get_peer().get_parent_element() as Audio.StreamVolume;
-        }
-        return true;
-    }
-
-    private bool adjust_delay() {
-        if (native != null) {
-            adjust_stream_delay(native);
-            return Source.CONTINUE;
-        } else {
-            adjust_delay_timeout_id = 0;
-            return Source.REMOVE;
-        }
-    }
-
-    private void process_outgoing_buffer(Buffer buffer) {
-        if (buffer.pts != uint64.MAX) {
-            last_reverse = buffer.pts;
-        }
-        analyze_reverse_stream(native, echo_probe.audio_info, buffer);
-        if (adjust_delay_timeout_id == 0 && echo_probe != null) {
-            adjust_delay_timeout_id = Timeout.add(1000, adjust_delay);
-        }
-    }
-
-    public override FlowReturn submit_input_buffer(bool is_discont, Buffer input) {
-        lock (adapter) {
-            if (is_discont) {
-                adapter.clear();
-            }
-            adapter.push(adjust_to_running_time(this, input));
-        }
-        return FlowReturn.OK;
-    }
-
-    public override FlowReturn generate_output(out Buffer output_buffer) {
-        lock (adapter) {
-            if (adapter.available() >= period_size) {
-                output_buffer = (Gst.Buffer) adapter.take_buffer(period_size).make_writable();
-                int old_gain_level = 0;
-                if (stream_volume != null) {
-                    old_gain_level = (int) (stream_volume.get_volume(Audio.StreamVolumeFormat.LINEAR) * 255.0);
-                    notify_gain_level(native, old_gain_level);
-                }
-                process_stream(native, audio_info, output_buffer);
-                if (stream_volume != null) {
-                    int new_gain_level = get_suggested_gain_level(native);
-                    if (old_gain_level != new_gain_level) {
-                        debug("Gain: %i -> %i", old_gain_level, new_gain_level);
-                        stream_volume.set_volume(Audio.StreamVolumeFormat.LINEAR, ((double)new_gain_level) / 255.0);
-                    }
-                }
-            }
-        }
-        return FlowReturn.OK;
-    }
-
-    public override bool stop() {
-        if (process_outgoing_buffer_handler_id != 0) {
-            echo_probe.disconnect(process_outgoing_buffer_handler_id);
-            process_outgoing_buffer_handler_id = 0;
-        }
-        if (adjust_delay_timeout_id != 0) {
-            Source.remove(adjust_delay_timeout_id);
-            adjust_delay_timeout_id = 0;
-        }
-        adapter.clear();
-        destroy_native(native);
-        native = null;
-        return true;
-    }
-}

plugins/rtp/src/voice_processor_native.cpp

@@ -1,148 +0,0 @@
-#include <algorithm>
-#include <gst/gst.h>
-#include <gst/audio/audio.h>
-#include <webrtc/modules/audio_processing/include/audio_processing.h>
-#include <webrtc/modules/interface/module_common_types.h>
-#include <webrtc/system_wrappers/include/trace.h>
-
-#define SAMPLE_RATE 48000
-#define SAMPLE_CHANNELS 1
-
-struct _DinoPluginsRtpVoiceProcessorNative {
-    webrtc::AudioProcessing *apm;
-    gint stream_delay;
-    gint last_median;
-    gint last_poor_delays;
-};
-
-extern "C" void *dino_plugins_rtp_adjust_to_running_time(GstBaseTransform *transform, GstBuffer *buffer) {
-    GstBuffer *copy = gst_buffer_copy(buffer);
-    GST_BUFFER_PTS(copy) = gst_segment_to_running_time(&transform->segment, GST_FORMAT_TIME, GST_BUFFER_PTS(buffer));
-    return copy;
-}
-
-extern "C" void *dino_plugins_rtp_voice_processor_init_native(gint stream_delay) {
-    _DinoPluginsRtpVoiceProcessorNative *native = new _DinoPluginsRtpVoiceProcessorNative();
-    webrtc::Config config;
-    config.Set<webrtc::ExtendedFilter>(new webrtc::ExtendedFilter(true));
-    config.Set<webrtc::ExperimentalAgc>(new webrtc::ExperimentalAgc(true, 85));
-    native->apm = webrtc::AudioProcessing::Create(config);
-    native->stream_delay = stream_delay;
-    native->last_median = 0;
-    native->last_poor_delays = 0;
-    return native;
-}
-
-extern "C" void dino_plugins_rtp_voice_processor_setup_native(void *native_ptr) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::AudioProcessing *apm = native->apm;
-    webrtc::ProcessingConfig pconfig;
-    pconfig.streams[webrtc::ProcessingConfig::kInputStream] =
-            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
-    pconfig.streams[webrtc::ProcessingConfig::kOutputStream] =
-            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
-    pconfig.streams[webrtc::ProcessingConfig::kReverseInputStream] =
-            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
-    pconfig.streams[webrtc::ProcessingConfig::kReverseOutputStream] =
-            webrtc::StreamConfig(SAMPLE_RATE, SAMPLE_CHANNELS, false);
-    apm->Initialize(pconfig);
-    apm->high_pass_filter()->Enable(true);
-    apm->echo_cancellation()->enable_drift_compensation(false);
-    apm->echo_cancellation()->set_suppression_level(webrtc::EchoCancellation::kModerateSuppression);
-    apm->echo_cancellation()->enable_delay_logging(true);
-    apm->echo_cancellation()->Enable(true);
-    apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kModerate);
-    apm->noise_suppression()->Enable(true);
-    apm->gain_control()->set_analog_level_limits(0, 255);
-    apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
-    apm->gain_control()->set_target_level_dbfs(3);
-    apm->gain_control()->set_compression_gain_db(9);
-    apm->gain_control()->enable_limiter(true);
-    apm->gain_control()->Enable(true);
-    apm->voice_detection()->set_likelihood(webrtc::VoiceDetection::Likelihood::kLowLikelihood);
-    apm->voice_detection()->Enable(true);
-}
-
-extern "C" void
-dino_plugins_rtp_voice_processor_analyze_reverse_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
-    webrtc::AudioProcessing *apm = native->apm;
-
-    GstMapInfo map;
-    gst_buffer_map(buffer, &map, GST_MAP_READ);
-
-    webrtc::AudioFrame frame;
-    frame.num_channels_ = info->channels;
-    frame.sample_rate_hz_ = info->rate;
-    frame.samples_per_channel_ = gst_buffer_get_size(buffer) / info->bpf;
-    memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
-
-    int err = apm->AnalyzeReverseStream(&frame);
-    if (err < 0) g_warning("voice_processor_native.cpp: ProcessReverseStream %i", err);
-
-    gst_buffer_unmap(buffer, &map);
-}
-
-extern "C" void dino_plugins_rtp_voice_processor_notify_gain_level(void *native_ptr, gint gain_level) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::AudioProcessing *apm = native->apm;
-    apm->gain_control()->set_stream_analog_level(gain_level);
-}
-
-extern "C" gint dino_plugins_rtp_voice_processor_get_suggested_gain_level(void *native_ptr) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::AudioProcessing *apm = native->apm;
-    return apm->gain_control()->stream_analog_level();
-}
-
-extern "C" bool dino_plugins_rtp_voice_processor_get_stream_has_voice(void *native_ptr) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::AudioProcessing *apm = native->apm;
-    return apm->voice_detection()->stream_has_voice();
-}
-
-extern "C" void dino_plugins_rtp_voice_processor_adjust_stream_delay(void *native_ptr) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::AudioProcessing *apm = native->apm;
-    int median, std, poor_delays;
-    float fraction_poor_delays;
-    apm->echo_cancellation()->GetDelayMetrics(&median, &std, &fraction_poor_delays);
-    poor_delays = (int)(fraction_poor_delays * 100.0);
-    if (fraction_poor_delays < 0 || (native->last_median == median && native->last_poor_delays == poor_delays)) return;
-    g_debug("voice_processor_native.cpp: Stream delay metrics: median=%i std=%i poor_delays=%i%%", median, std, poor_delays);
-    native->last_median = median;
-    native->last_poor_delays = poor_delays;
-    if (poor_delays > 90) {
-        native->stream_delay = std::min(std::max(0, native->stream_delay + std::min(48, std::max(median, -48))), 384);
-        g_debug("voice_processor_native.cpp: set stream_delay=%i", native->stream_delay);
-    }
-}
-
-extern "C" void
-dino_plugins_rtp_voice_processor_process_stream(void *native_ptr, GstAudioInfo *info, GstBuffer *buffer) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    webrtc::StreamConfig config(SAMPLE_RATE, SAMPLE_CHANNELS, false);
-    webrtc::AudioProcessing *apm = native->apm;
-
-    GstMapInfo map;
-    gst_buffer_map(buffer, &map, GST_MAP_READWRITE);
-
-    webrtc::AudioFrame frame;
-    frame.num_channels_ = info->channels;
-    frame.sample_rate_hz_ = info->rate;
-    frame.samples_per_channel_ = info->rate / 100;
-    memcpy(frame.data_, map.data, frame.samples_per_channel_ * info->bpf);
-
-    apm->set_stream_delay_ms(native->stream_delay);
-    int err = apm->ProcessStream(&frame);
-    if (err >= 0) memcpy(map.data, frame.data_, frame.samples_per_channel_ * info->bpf);
-    if (err < 0) g_warning("voice_processor_native.cpp: ProcessStream %i", err);
-
-    gst_buffer_unmap(buffer, &map);
-}
-
-extern "C" void dino_plugins_rtp_voice_processor_destroy_native(void *native_ptr) {
-    _DinoPluginsRtpVoiceProcessorNative *native = (_DinoPluginsRtpVoiceProcessorNative *) native_ptr;
-    delete native;
-}