Add native self-hosted instance connection to fluxer_desktop

Trimmed monorepo checkout (fluxer_desktop + packages/voice_engine_v2 + tools/ci) with a "Connect to a Different Server" menu item and popout that lets the desktop app switch to any self-hosted Fluxer instance, plus fixes for well-known discovery on single-domain self-hosted deployments and a false-positive ERR_ABORTED on same-origin client redirects during the switch. Defaults to chat.fluxr.chat and uses an isolated userData directory from the official build.
2026-07-01 18:22:43 -04:00
commit 682afacd30
1763 changed files with 613720 additions and 0 deletions
@@ -0,0 +1,36 @@
+diff --git a/BUILD.gn b/BUILD.gn
+index ca8d8faa61..13e07a2f28 100644
+--- a/BUILD.gn
+++ b/BUILD.gn
+@@ -24,6 +24,9 @@
+ import("//build/config/linux/pkg_config.gni")
+ import("//build/config/sanitizers/sanitizers.gni")
+ import("webrtc.gni")
+import("//third_party/libaom/options.gni")
+
+
+ if (rtc_enable_protobuf) {
+   import("//third_party/protobuf/proto_library.gni")
+ }
+@@ -331,6 +334,10 @@ config("common_config") {
+     defines += [ "WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE" ]
+   }
+ 
+  if (enable_libaom) {
+    defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
+  }
+
+   if (rtc_libvpx_build_vp9) {
+     defines += [ "RTC_ENABLE_VP9" ]
+   }
+@@ -565,6 +572,10 @@ if (!build_with_chromium) {
+       "pc:rtc_pc",
+       "sdk",
+       "video",
+      "//third_party/zlib",
+      "rtc_base:log_sinks",
+      "media:rtc_simulcast_encoder_adapter",
+      "api/crypto:frame_crypto_transformer",
+     ]
+ 
+     if (rtc_include_builtin_audio_codecs) {
@@ -0,0 +1,23 @@
+diff --git a/tools_webrtc/libs/generate_licenses.py b/tools_webrtc/libs/generate_licenses.py
+index d945b43..5f3e457 100755
+--- a/tools_webrtc/libs/generate_licenses.py
+++ b/tools_webrtc/libs/generate_licenses.py
+@@ -86,6 +86,18 @@ LIB_TO_LICENSES_DICT = {
+     # Compile time dependencies, no license needed:
+     'ow2_asm': [],
+     'jdk': [],
+
+    'ffmpeg':[
+        'third_party/ffmpeg/COPYING.GPLv2',
+        'third_party/ffmpeg/COPYING.GPLv3',
+        'third_party/ffmpeg/COPYING.LGPLv2.1',
+        'third_party/ffmpeg/COPYING.LGPLv3'
+    ],
+    'openh264': ['third_party/openh264/src/LICENSE'],
+    'catapult': [],
+    'google_benchmark': [],
+    'googletest': [],
+    'vinn': [],
+ }
+ 
+ # Third_party library _regex_ to licences mapping. Keys are regular expression
@@ -0,0 +1,22 @@
+--- src/buildtools/third_party/libunwind/BUILD.gn	2023-07-10 10:19:16
+++ src/buildtools/third_party/libunwind/BUILD.gn	2023-07-10 10:19:23
+@@ -21,7 +21,7 @@ config("libunwind_config") {
+ 
+ # TODO(crbug.com/40273848): Move this build file to third_party/libc++/BUILD.gn once submodule migration is done
+ source_set("libunwind") {
+-  visibility = [ "//buildtools/third_party/libc++abi" ]
+  visibility = [ "//buildtools/third_party/libc++abi", "//build/config:common_deps" ]
+   if (is_android) {
+     visibility += [ "//services/tracing/public/cpp" ]
+   }
+--- src/build/config/BUILD.gn	2023-07-10 10:23:49
+++ src/build/config/BUILD.gn	2023-07-10 10:23:54
+@@ -296,6 +296,8 @@ group("common_deps") {
+ 
+   if (use_custom_libcxx) {
+     public_deps += [ "//buildtools/third_party/libc++" ]
+  } else {
+    public_deps += [ "//buildtools/third_party/libunwind" ]
+   }
+ 
+   if (use_llvm_libatomic) {
@@ -0,0 +1,13 @@
+diff --git a/dav1d/BUILD.gn b/dav1d/BUILD.gn
+index 9348e15c3c..f34db7ff98 100644
+--- a/dav1d/BUILD.gn
+++ b/dav1d/BUILD.gn
+@@ -101,7 +101,7 @@ if (is_win) {
+     dav1d_copts += [ "-D_DARWIN_C_SOURCE" ]
+   }
+   if (is_linux || is_chromeos || is_android || current_os == "aix") {
+-    if (!is_clang) {
+    if (!is_clang && (current_cpu == "x86" || current_cpu == "x64")) {
+       dav1d_copts += [ "-D_GNU_SOURCE" ]
+     }
+   }
@@ -0,0 +1,19 @@
+diff --git a/config/compiler/BUILD.gn b/config/compiler/BUILD.gn
+index c09c9677b..3cfce80a1 100644
+--- a/config/compiler/BUILD.gn
+++ b/config/compiler/BUILD.gn
+@@ -721,14 +721,6 @@ config("compiler") {
+     } else {
+       cflags += [ "-ffp-contract=off" ]
+     }
+-
+-    # Enable ELF CREL (see crbug.com/357878242) for all platforms that use ELF.
+-    # TODO(crbug.com/376278218): This causes segfault on Linux ARM builds.
+-    # It also causes segfault on Linux s390x:
+-    # https://github.com/llvm/llvm-project/issues/149511
+-    if (is_linux && use_lld && current_cpu != "arm" && current_cpu != "s390x") {
+-      cflags += [ "-Wa,--crel,--allow-experimental-crel" ]
+-    }
+   }
+ 
+   # C11/C++11 compiler flags setup.
@@ -0,0 +1,13 @@
+diff --git a/libyuv.gni b/libyuv.gni
+index 3334df70..9859a121 100644
+--- a/libyuv.gni
+++ b/libyuv.gni
+@@ -24,7 +24,7 @@ declare_args() {
+   # errors on Fuchsia, macOS, and compilation errors on Windows.
+   # TODO: bug 359006069 - Remove the restriction after the linker and
+   # compilation errors are fixed.
+-  libyuv_use_sme = current_cpu == "arm64" && (is_linux || is_android)
+  libyuv_use_sme = false
+   libyuv_use_msa =
+       (current_cpu == "mips64el" || current_cpu == "mipsel") && mips_use_msa
+   libyuv_use_mmi =
@@ -0,0 +1,142 @@
+diff --git a/api/media_stream_interface.h b/api/media_stream_interface.h
+index fb1cc4e58e..85062ba60e 100644
+--- a/api/media_stream_interface.h
+++ b/api/media_stream_interface.h
+@@ -267,6 +267,11 @@ class RTC_EXPORT AudioSourceInterface : public MediaSourceInterface {
+   // (for some of the settings this approach is broken, e.g. setting
+   // audio network adaptation on the source is the wrong layer of abstraction).
+   virtual const AudioOptions options() const;
+
+  // Returns true if this source delivers audio externally (via AddSink),
+  // bypassing the ADM/AudioState audio distribution path.
+  // When true, AudioSendStream should not register with AudioState.
+  virtual bool is_external_source() const { return false; }
+ };
+ 
+ // Interface of the audio processor used by the audio track to collect
+diff --git a/audio/audio_send_stream.cc b/audio/audio_send_stream.cc
+index 76156ce830..10b59d3ff6 100644
+--- a/audio/audio_send_stream.cc
+++ b/audio/audio_send_stream.cc
+@@ -373,8 +373,13 @@ void AudioSendStream::Start() {
+   }
+   channel_send_->StartSend();
+   sending_ = true;
+-  audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
+-                                  encoder_num_channels_);
+  // Only register with AudioState if not using external source.
+  // External sources (like NativeAudioSource) deliver audio directly via AddSink,
+  // so we don't want AudioState to also send device audio to this stream.
+  if (!config_.external_source) {
+    audio_state()->AddSendingStream(this, encoder_sample_rate_hz_,
+                                    encoder_num_channels_);
+  }
+ }
+ 
+ void AudioSendStream::Stop() {
+@@ -386,7 +391,10 @@ void AudioSendStream::Stop() {
+   RemoveBitrateObserver();
+   channel_send_->StopSend();
+   sending_ = false;
+-  audio_state()->RemoveSendingStream(this);
+  // Only unregister if we registered (when not using external source).
+  if (!config_.external_source) {
+    audio_state()->RemoveSendingStream(this);
+  }
+ }
+ 
+ void AudioSendStream::SendAudioData(std::unique_ptr<AudioFrame> audio_frame) {
+diff --git a/call/audio_send_stream.h b/call/audio_send_stream.h
+index 84341b5cb1..9359777bc9 100644
+--- a/call/audio_send_stream.h
+++ b/call/audio_send_stream.h
+@@ -178,6 +178,12 @@ class AudioSendStream : public AudioSender {
+     // An optional frame transformer used by insertable streams to transform
+     // encoded frames.
+     scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer;
+
+    // When true, this stream uses an external audio source (not ADM).
+    // AudioState will NOT send device-captured audio to this stream.
+    // Audio is delivered directly via the source's AddSink mechanism.
+    // This prevents mixing of device audio with externally-sourced audio.
+    bool external_source = false;
+   };
+ 
+   virtual ~AudioSendStream() = default;
+diff --git a/media/base/audio_source.h b/media/base/audio_source.h
+index 04a7d19dfa..9f513f3c75 100644
+--- a/media/base/audio_source.h
+++ b/media/base/audio_source.h
+@@ -49,6 +49,10 @@ class AudioSource {
+   // to the source at a time.
+   virtual void SetSink(Sink* sink) = 0;
+ 
+  // Returns true if this source delivers audio externally (bypassing ADM).
+  // When true, AudioSendStream should not register with AudioState.
+  virtual bool is_external_source() const { return false; }
+
+  protected:
+   virtual ~AudioSource() {}
+ };
+diff --git a/media/engine/webrtc_voice_engine.cc b/media/engine/webrtc_voice_engine.cc
+index 762f9d584c..4ce07ddc9d 100644
+--- a/media/engine/webrtc_voice_engine.cc
+++ b/media/engine/webrtc_voice_engine.cc
+@@ -1017,6 +1017,14 @@ class WebRtcVoiceSendChannel::WebRtcAudioSendStream : public AudioSource::Sink {
+       RTC_DCHECK(source_ == source);
+       return;
+     }
+
+    // Check if this is an external audio source (delivers audio via AddSink).
+    // If so, mark the config so AudioState doesn't send device audio to this
+    // stream. This must be done before UpdateSendState() calls Start().
+    if (source->is_external_source() && !config_.external_source) {
+      config_.external_source = true;
+      stream_->Reconfigure(config_, nullptr);
+    }
+     source->SetSink(this);
+     source_ = source;
+     UpdateSendState();
+diff --git a/pc/rtp_sender.cc b/pc/rtp_sender.cc
+index d5edbbf0ed..c14ddfe868 100644
+--- a/pc/rtp_sender.cc
+++ b/pc/rtp_sender.cc
+@@ -786,6 +786,13 @@ void AudioRtpSender::SetSend() {
+   RTC_DCHECK_RUN_ON(signaling_thread_);
+   RTC_DCHECK(!stopped_);
+   RTC_DCHECK(can_send_track());
+
+  // Propagate is_external_source from AudioSourceInterface to the sink adapter.
+  // This ensures the voice engine knows not to mix ADM audio with external sources.
+  if (audio_track()->GetSource() && audio_track()->GetSource()->is_external_source()) {
+    sink_adapter_->set_is_external_source(true);
+  }
+
+   if (!media_channel_) {
+     RTC_LOG(LS_ERROR) << "SetAudioSend: No audio channel exists.";
+     return;
+diff --git a/pc/rtp_sender.h b/pc/rtp_sender.h
+index eaffd4ef0f..d0489df9e7 100644
+--- a/pc/rtp_sender.h
+++ b/pc/rtp_sender.h
+@@ -307,6 +307,12 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
+   LocalAudioSinkAdapter();
+   virtual ~LocalAudioSinkAdapter();
+ 
+  // Set whether the original AudioSourceInterface is an external source.
+  // This propagates the is_external_source state from AudioSourceInterface
+  // to this AudioSource adapter.
+  void set_is_external_source(bool value) { is_external_source_ = value; }
+  bool is_external_source() const override { return is_external_source_; }
+
+  private:
+   // AudioSinkInterface implementation.
+   void OnData(const void* audio_data,
+@@ -337,6 +343,7 @@ class LocalAudioSinkAdapter : public AudioTrackSinkInterface,
+   // Critical section protecting `sink_`.
+   Mutex lock_;
+   int num_preferred_channels_ = -1;
+  bool is_external_source_ = false;
+ };
+ 
+ class AudioRtpSender : public DtmfProviderInterface, public RtpSenderBase {
@@ -0,0 +1,17 @@
+diff --git a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+index 070257f072..61fd0c8f3b 100644
+--- a/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+++ b/modules/desktop_capture/linux/wayland/shared_screencast_stream.cc
+@@ -71,10 +71,10 @@ constexpr int CursorMetaSize(int w, int h) {
+           w * h * kCursorBpp);
+ }
+ 
+-constexpr PipeWireVersion kDmaBufModifierMinVersion = {.major = 0,
+const PipeWireVersion kDmaBufModifierMinVersion = {.major = 0,
+                                                        .minor = 3,
+                                                        .micro = 33};
+-constexpr PipeWireVersion kDropSingleModifierMinVersion = {.major = 0,
+const PipeWireVersion kDropSingleModifierMinVersion = {.major = 0,
+                                                            .minor = 3,
+                                                            .micro = 40};
+ 
@@ -0,0 +1,16 @@
+diff --git a/tools_webrtc/libs/generate_licenses.py b/tools_webrtc/libs/generate_licenses.py
+--- a/tools_webrtc/libs/generate_licenses.py
+++ b/tools_webrtc/libs/generate_licenses.py
+@@ -211,6 +211,12 @@ class LicenseBuilder:
+
+     def _get_third_party_libraries(self, buildfile_dir, target):
+         license_json = LicenseBuilder._run_gn(buildfile_dir, target)
+        # Strip any non-JSON content (e.g., GN warnings) before the actual JSON.
+        # GN may output warnings before the JSON when certain build args trigger
+        # deprecation notices.
+        json_start = license_json.find('{')
+        if json_start > 0:
+            license_json = license_json[json_start:]
+         try:
+             output = json.loads(license_json)
+         except:
@@ -0,0 +1,317 @@
+diff --git a/modules/video_coding/codecs/test/android_codec_factory_helper.cc b/modules/video_coding/codecs/test/android_codec_factory_helper.cc
+index f6a5046210..4ce5a79d25 100644
+--- a/modules/video_coding/codecs/test/android_codec_factory_helper.cc
+++ b/modules/video_coding/codecs/test/android_codec_factory_helper.cc
+@@ -54,9 +54,9 @@ void InitializeAndroidObjects() {
+ std::unique_ptr<VideoEncoderFactory> CreateAndroidEncoderFactory() {
+   JNIEnv* env = AttachCurrentThreadIfNeeded();
+   ScopedJavaLocalRef<jclass> factory_class =
+-      GetClass(env, "org/webrtc/HardwareVideoEncoderFactory");
+      GetClass(env, "livekit/org/webrtc/HardwareVideoEncoderFactory");
+   jmethodID factory_constructor = env->GetMethodID(
+-      factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;ZZ)V");
+      factory_class.obj(), "<init>", "(Llivekit/org/webrtc/EglBase$Context;ZZ)V");
+   ScopedJavaLocalRef<jobject> factory_object =
+       ScopedJavaLocalRef<jobject>::Adopt(
+           env, env->NewObject(factory_class.obj(), factory_constructor,
+@@ -69,9 +69,9 @@ std::unique_ptr<VideoEncoderFactory> CreateAndroidEncoderFactory() {
+ std::unique_ptr<VideoDecoderFactory> CreateAndroidDecoderFactory() {
+   JNIEnv* env = AttachCurrentThreadIfNeeded();
+   ScopedJavaLocalRef<jclass> factory_class =
+-      GetClass(env, "org/webrtc/HardwareVideoDecoderFactory");
+      GetClass(env, "livekit/org/webrtc/HardwareVideoDecoderFactory");
+   jmethodID factory_constructor = env->GetMethodID(
+-      factory_class.obj(), "<init>", "(Lorg/webrtc/EglBase$Context;)V");
+      factory_class.obj(), "<init>", "(Llivekit/org/webrtc/EglBase$Context;)V");
+   ScopedJavaLocalRef<jobject> factory_object =
+       ScopedJavaLocalRef<jobject>::Adopt(
+           env, env->NewObject(factory_class.obj(), factory_constructor,
+diff --git a/sdk/android/BUILD.gn b/sdk/android/BUILD.gn
+index 8ce74e2c8c..2e88760d0a 100644
+--- a/sdk/android/BUILD.gn
+++ b/sdk/android/BUILD.gn
+@@ -1446,11 +1446,13 @@ if (current_os == "linux" || is_android) {
+   generate_jni("generated_environment_jni") {
+     sources = [ "api/org/webrtc/Environment.java" ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_metrics_jni") {
+     sources = [ "api/org/webrtc/Metrics.java" ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   # Generated JNI for public JNI targets, matching order of targets
+@@ -1464,6 +1466,7 @@ if (current_os == "linux" || is_android) {
+       "src/java/org/webrtc/JniCommon.java",
+     ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_video_jni") {
+@@ -1493,11 +1496,13 @@ if (current_os == "linux" || is_android) {
+       "src/java/org/webrtc/WrappedNativeI420Buffer.java",
+     ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_video_egl_jni") {
+     sources = [ "src/java/org/webrtc/EglBase10Impl.java" ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_libvpx_vp8_jni") {
+@@ -1507,6 +1512,7 @@ if (current_os == "linux" || is_android) {
+     ]
+ 
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_libvpx_vp9_jni") {
+@@ -1516,18 +1522,21 @@ if (current_os == "linux" || is_android) {
+     ]
+ 
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_libaom_av1_encoder_jni") {
+     sources = [ "api/org/webrtc/LibaomAv1Encoder.java" ]
+ 
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_dav1d_jni") {
+     sources = [ "api/org/webrtc/Dav1dDecoder.java" ]
+ 
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_swcodecs_jni") {
+@@ -1537,11 +1546,13 @@ if (current_os == "linux" || is_android) {
+     ]
+ 
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_rtcerror_jni") {
+     sources = [ "src/java/org/webrtc/RtcError.java" ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_peerconnection_jni") {
+@@ -1585,11 +1596,13 @@ if (current_os == "linux" || is_android) {
+       "api/org/webrtc/TurnCustomizer.java",
+     ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_java_audio_jni") {
+     sources = [ "api/org/webrtc/audio/JavaAudioDeviceModule.java" ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_builtin_audio_codecs_jni") {
+@@ -1598,6 +1611,7 @@ if (current_os == "linux" || is_android) {
+       "api/org/webrtc/BuiltinAudioEncoderFactoryFactory.java",
+     ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   # Generated JNI for native API targets
+@@ -1609,17 +1623,20 @@ if (current_os == "linux" || is_android) {
+       "src/java/org/webrtc/WebRtcClassLoader.java",
+     ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   # Generated JNI for internal targets.
+ 
+   generate_jni("generated_logging_jni") {
+     sources = [ "src/java/org/webrtc/JNILogging.java" ]
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_audio_device_module_base_jni") {
+     sources = [ "src/java/org/webrtc/audio/WebRtcAudioManager.java" ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ 
+   generate_jni("generated_java_audio_device_module_native_jni") {
+@@ -1628,6 +1645,7 @@ if (current_os == "linux" || is_android) {
+       "src/java/org/webrtc/audio/WebRtcAudioTrack.java",
+     ]
+     namespace = "webrtc::jni"
+    package_prefix = android_package_prefix
+   }
+ }
+ 
+diff --git a/sdk/android/api/org/webrtc/PeerConnectionFactory.java b/sdk/android/api/org/webrtc/PeerConnectionFactory.java
+index 01679ad12b..7b8545902d 100644
+--- a/sdk/android/api/org/webrtc/PeerConnectionFactory.java
+++ b/sdk/android/api/org/webrtc/PeerConnectionFactory.java
+@@ -88,7 +88,7 @@ public class PeerConnectionFactory {
+       private String fieldTrials = "";
+       private boolean enableInternalTracer;
+       private NativeLibraryLoader nativeLibraryLoader = new NativeLibrary.DefaultLoader();
+-      private String nativeLibraryName = "jingle_peerconnection_so";
+      private String nativeLibraryName = "lkjingle_peerconnection_so";
+       @Nullable private Loggable loggable;
+       @Nullable private Severity loggableSeverity;
+ 
+diff --git a/sdk/android/src/jni/jni_helpers.h b/sdk/android/src/jni/jni_helpers.h
+index d86c8fa4ad..30f307a2ca 100644
+--- a/sdk/android/src/jni/jni_helpers.h
+++ b/sdk/android/src/jni/jni_helpers.h
+@@ -29,10 +29,10 @@
+ // boundary. crbug.com/655248
+ #define JNI_FUNCTION_DECLARATION(rettype, name, ...)                    \
+   __attribute__((force_align_arg_pointer)) extern "C" JNIEXPORT rettype \
+-      JNICALL Java_org_webrtc_##name(__VA_ARGS__)
+      JNICALL Java_livekit_org_webrtc_##name(__VA_ARGS__)
+ #else
+ #define JNI_FUNCTION_DECLARATION(rettype, name, ...) \
+-  extern "C" JNIEXPORT rettype JNICALL Java_org_webrtc_##name(__VA_ARGS__)
+  extern "C" JNIEXPORT rettype JNICALL Java_livekit_org_webrtc_##name(__VA_ARGS__)
+ #endif
+ 
+ namespace webrtc {
+diff --git a/sdk/android/src/jni/simulcast_video_encoder.cc b/sdk/android/src/jni/simulcast_video_encoder.cc
+index 6d4ba5b3b6..48d3cb3541 100644
+--- a/sdk/android/src/jni/simulcast_video_encoder.cc
+++ b/sdk/android/src/jni/simulcast_video_encoder.cc
+@@ -15,7 +15,14 @@ extern "C" {
+ #endif
+ 
+ // (VideoEncoderFactory primary, VideoEncoderFactory fallback, VideoCodecInfo info)
+-JNIEXPORT jlong JNICALL Java_org_webrtc_SimulcastVideoEncoder_nativeCreateEncoder(JNIEnv *env, jclass klass, jlong webrtcEnvRef, jobject primary, jobject fallback, jobject info) {
+JNI_FUNCTION_DECLARATION(jlong,
+                         SimulcastVideoEncoder_nativeCreateEncoder,
+                         JNIEnv *env,
+                         jclass klass,
+                         jlong webrtcEnvRef,
+                         jobject primary,
+                         jobject fallback,
+                         jobject info) {
+     RTC_LOG(LS_INFO) << "Create simulcast video encoder";
+     auto info_ref = JavaParamRef<jobject>::CreateLeaky(env, info);
+     SdpVideoFormat format = VideoCodecInfoToSdpVideoFormat(env, info_ref);
+diff --git a/tools_webrtc/android/build_aar.py b/tools_webrtc/android/build_aar.py
+index 2fbce97c60..130b3d5fa7 100755
+--- a/tools_webrtc/android/build_aar.py
+++ b/tools_webrtc/android/build_aar.py
+@@ -254,7 +254,7 @@ def BuildAar(build_dir,
+             Collect(aar_file, build_dir, arch, unstripped)
+ 
+     license_dir = os.path.dirname(os.path.realpath(output_file))
+-    GenerateLicenses(license_dir, build_dir, archs)
+    #GenerateLicenses(license_dir, build_dir, archs)
+ 
+ 
+ def main():
+diff --git a/webrtc.gni b/webrtc.gni
+index 468ba33b01..ca81ce298c 100644
+--- a/webrtc.gni
+++ b/webrtc.gni
+@@ -241,6 +241,10 @@ declare_args() {
+   #          hasn't been registered.
+   rtc_strict_field_trials = ""
+ 
+  if (is_android) {
+    android_package_prefix = "livekit"
+  }
+
+   # If different from "", symbols exported with RTC_OBJC_EXPORT will be prefixed
+   # with this string.
+   # See the definition of RTC_OBJC_TYPE_PREFIX in the code.
+diff --git a/third_party/jni_zero/BUILD.gn b/third_party/jni_zero/BUILD.gn
+index 7056bdd033c1..2b97de476de7 100644
+--- a/third_party/jni_zero/BUILD.gn
+++ b/third_party/jni_zero/BUILD.gn
+@@ -17,6 +17,7 @@ generate_jni("generate_jni") {
+     ":*",
+     "//components/cronet/android/*",
+   ]
+  package_prefix = "livekit"
+ }
+ 
+ generate_jar_jni("system_jni") {
+diff --git a/third_party/jni_zero/codegen/header_common.py b/third_party/jni_zero/codegen/header_common.py
+index 338cd748079f..be2577281a91 100644
+--- a/third_party/jni_zero/codegen/header_common.py
+++ b/third_party/jni_zero/codegen/header_common.py
+@@ -12,7 +12,8 @@ def class_accessors(sb, java_classes, module_name):
+   for java_class in java_classes:
+     if java_class in (java_types.OBJECT_CLASS, java_types.STRING_CLASS):
+       continue
+-    escaped_name = java_class.to_cpp()
+    escaped_name = java_class.class_without_prefix.to_cpp()
+    print("header accessor: ", escaped_name)
+     # #ifdef needed when multple .h files are #included that common classes.
+     sb(f"""\
+ #ifndef {escaped_name}_clazz_defined
+@@ -44,7 +45,7 @@ def class_accessor_expression(java_class):
+   if java_class == java_types.STRING_CLASS:
+     return 'jni_zero::g_string_class'
+ 
+-  return f'{java_class.to_cpp()}_clazz(env)'
+  return f'{java_class.class_without_prefix.to_cpp()}_clazz(env)'
+ 
+ 
+ def header_preamble(script_name,
+@@ -53,7 +54,7 @@ def header_preamble(script_name,
+                     user_includes=None,
+                     header_guard=None):
+   if header_guard is None:
+-    header_guard = f'{java_class.to_cpp()}_JNI'
+    header_guard = f'{java_class.class_without_prefix.to_cpp()}_JNI'
+   sb = []
+   sb.append(f"""\
+ // This file was generated by
+diff --git a/third_party/jni_zero/jni_zero.gni b/third_party/jni_zero/jni_zero.gni
+index 71826b906d10..22d8eaf6e232 100644
+--- a/third_party/jni_zero/jni_zero.gni
+++ b/third_party/jni_zero/jni_zero.gni
+@@ -123,7 +123,7 @@ template("_invoke_jni_zero") {
+ #   namespace: Registration functions will be wrapped into this. (optional)
+ #   priority_java_targets: List of java targets that, if using multiplexing,
+ #     will always be placed first in the sequential switch numbers. (optional)
+-#
+#   package_prefix: package prefix to add to jni
+ # Example
+ #   generate_jni_registration("chrome_jni_registration") {
+ #     java_targets = [ ":chrome_public_apk" ]
+@@ -327,6 +327,7 @@ template("generate_jni_impl") {
+                                "deps",
+                                "metadata",
+                                "public_deps",
+                               "package_prefix",
+                              ])
+       if (!defined(public_deps)) {
+         public_deps = []
+@@ -421,7 +422,9 @@ template("generate_jni_impl") {
+       if (defined(invoker.split_name)) {
+         args += [ "--split-name=${invoker.split_name}" ]
+       }
+-
+      if (defined(package_prefix)) {
+        args += [ "--package-prefix=${package_prefix}" ]
+      }
+       foreach(_name, _input_names) {
+         _name =
+             string_replace(get_path_info(_name, "name"), "\$", "__") + "_jni.h"
@@ -0,0 +1,62 @@
+diff --git a/rtc_base/boringssl_certificate.cc b/rtc_base/boringssl_certificate.cc
+index a4f919fbbe..fe3beeaed9 100644
+--- a/rtc_base/boringssl_certificate.cc
+++ b/rtc_base/boringssl_certificate.cc
+@@ -252,6 +252,12 @@ BoringSSLCertificate::BoringSSLCertificate(
+   RTC_DCHECK(cert_buffer_ != nullptr);
+ }
+ 
+BoringSSLCertificate::BoringSSLCertificate(
+    bssl::UniquePtr<CRYPTO_BUFFER> cert_buffer, SSL* ssl)
+    : cert_buffer_(std::move(cert_buffer)), ssl_(ssl) {
+  RTC_DCHECK(cert_buffer_ != nullptr);
+}
+
+ std::unique_ptr<BoringSSLCertificate> BoringSSLCertificate::Generate(
+     OpenSSLKeyPair* key_pair,
+     const SSLIdentityParams& params) {
+diff --git a/rtc_base/boringssl_certificate.h b/rtc_base/boringssl_certificate.h
+index b5a18d0843..53759fd064 100644
+--- a/rtc_base/boringssl_certificate.h
+++ b/rtc_base/boringssl_certificate.h
+@@ -34,6 +34,7 @@ namespace webrtc {
+ class BoringSSLCertificate final : public SSLCertificate {
+  public:
+   explicit BoringSSLCertificate(bssl::UniquePtr<CRYPTO_BUFFER> cert_buffer);
+  BoringSSLCertificate(bssl::UniquePtr<CRYPTO_BUFFER> cert_buffer, SSL* ssl);
+ 
+   static std::unique_ptr<BoringSSLCertificate> Generate(
+       OpenSSLKeyPair* key_pair,
+@@ -66,6 +67,10 @@ class BoringSSLCertificate final : public SSLCertificate {
+  private:
+   // A handle to the DER encoded certificate data.
+   bssl::UniquePtr<CRYPTO_BUFFER> cert_buffer_;
+  SSL* ssl_ = nullptr;
+
+ public:
+  SSL* ssl() const { return ssl_; }
+ };
+ 
+ }  //  namespace webrtc
+diff --git a/rtc_base/openssl_adapter.cc b/rtc_base/openssl_adapter.cc
+index b80e665d78..1f3dd28795 100644
+--- a/rtc_base/openssl_adapter.cc
+++ b/rtc_base/openssl_adapter.cc
+@@ -846,7 +846,7 @@ enum ssl_verify_result_t OpenSSLAdapter::SSLVerifyInternal(SSL* ssl,
+   std::vector<std::unique_ptr<SSLCertificate>> certs;
+   for (size_t i = 0; i < sk_CRYPTO_BUFFER_num(chain); ++i) {
+     certs.emplace_back(new BoringSSLCertificate(
+-        bssl::UpRef(sk_CRYPTO_BUFFER_value(chain, i))));
+        bssl::UpRef(sk_CRYPTO_BUFFER_value(chain, i)), ssl));
+   }
+ 
+   SSLCertChain cert_chain(std::move(certs));
+@@ -932,7 +932,7 @@ int OpenSSLAdapter::SSLVerifyInternal(int previous_status,
+       RTC_LOG(LS_ERROR) << "Failed to allocate CRYPTO_BUFFER.";
+       return previous_status;
+     }
+-    certs.emplace_back(new BoringSSLCertificate(std::move(crypto_buffer)));
+    certs.emplace_back(new BoringSSLCertificate(std::move(crypto_buffer), ssl));
+ #else
+     certs.emplace_back(new OpenSSLCertificate(x509_cert));
+ #endif
@@ -0,0 +1,14 @@
+diff --git a/build/config/compiler/BUILD.gn b/build/config/compiler/BUILD.gn
+index 8048ec5e8..4e00f7e53 100644
+--- a/build/config/compiler/BUILD.gn
+++ b/build/config/compiler/BUILD.gn
+@@ -1424,6 +1424,9 @@ config("default_warnings") {
+       # gethostbyname. Fires mostly in non-Chromium code. We probably
+       # want to remove this define eventually.
+       "_WINSOCK_DEPRECATED_NO_WARNINGS",
+
+      "_SILENCE_ALL_CXX17_DEPRECATION_WARNINGS",
+      "_SILENCE_ALL_CXX20_DEPRECATION_WARNINGS",
+     ]
+     if (!is_clang) {
+       # TODO(thakis): Remove this once