diff --git a/Build/Makefile b/Build/Makefile
index 83baecf721..7b210ca33d 100644
--- a/Build/Makefile
+++ b/Build/Makefile
@@ -35,7 +35,7 @@ android-build:
         CMAKE_EXTRA+="-DCMAKE_TOOLCHAIN_FILE=$(ANDROID_NDK_HOME)/build/cmake/android.toolchain.cmake \
         -DCMAKE_POSITION_INDEPENDENT_CODE=ON \
         -DANDROID_ABI=$(ANDROID_ABI) \
-        -DANDROID_PLATFORM=android-21 \
+        -DANDROID_PLATFORM=android-23 \
         -DBUILD_TEAMTALK_LIBRARY_LIB=ON \
         -DBUILD_TEAMTALK_LIBRARY_LIBPRO=ON \
         -DBUILD_TEAMTALK_CLIENTS=OFF \
diff --git a/ChangeLog.txt b/ChangeLog.txt
index 516fadac77..3e5bf6bc31 100644
--- a/ChangeLog.txt
+++ b/ChangeLog.txt
@@ -24,6 +24,7 @@ Default Qt Client
 - Fixed bug where all servers with same name was deleted on server deletion
 - Updated Qt to 6.11 on Windows and macOS
 Android Client
+- Minimum supported Android version is now 6.0 (Marshmallow)
 - Easy log on to server using Join Code
 - Fixed bug where TeamTalk would crash when entering TTS settings
 - Fixed bug where the initial volume of media files did not match the settings screen
diff --git a/Client/TeamTalkAndroid/build.gradle b/Client/TeamTalkAndroid/build.gradle
index 40d5c4f29f..18915f6cae 100644
--- a/Client/TeamTalkAndroid/build.gradle
+++ b/Client/TeamTalkAndroid/build.gradle
@@ -13,7 +13,7 @@ android {
 
     defaultConfig {
         applicationId = 'dk.bearware.gui'
-        minSdk = 21
+        minSdk = 23
         targetSdk = 36
         versionCode = 150
         versionName = '5.22'
diff --git a/Library/TeamTalkLib/avstream/MediaPlayback.cpp b/Library/TeamTalkLib/avstream/MediaPlayback.cpp
index 09b7f78696..1ffab4807a 100644
--- a/Library/TeamTalkLib/avstream/MediaPlayback.cpp
+++ b/Library/TeamTalkLib/avstream/MediaPlayback.cpp
@@ -27,6 +27,8 @@
 
 #if defined(ENABLE_WEBRTC)
 #include "avstream/WebRTCPreprocess.h"
+#include <api/audio/builtin_audio_processing_builder.h>
+#include <api/environment/environment_factory.h>
 #endif
 
 #include <algorithm>
@@ -208,7 +210,7 @@ bool MediaPlayback::SetupSpeexPreprocess(bool enableagc, const SpeexAGC& agc,
 bool MediaPlayback::SetupWebRTCPreprocess(const webrtc::AudioProcessing::Config& webrtc)
 {
     if (!m_apm)
-        m_apm = webrtc::AudioProcessingBuilder().Create();
+        m_apm = webrtc::BuiltinAudioProcessingBuilder().Build(webrtc::CreateEnvironment());
 
     m_apm->ApplyConfig(webrtc);
     if (m_apm->Initialize() != webrtc::AudioProcessing::kNoError)
diff --git a/Library/TeamTalkLib/avstream/MediaPlayback.h b/Library/TeamTalkLib/avstream/MediaPlayback.h
index f1e724013f..604922a6ca 100644
--- a/Library/TeamTalkLib/avstream/MediaPlayback.h
+++ b/Library/TeamTalkLib/avstream/MediaPlayback.h
@@ -142,7 +142,7 @@ class MediaPlayback : public soundsystem::StreamPlayer
     std::shared_ptr<SpeexPreprocess> m_preprocess_left, m_preprocess_right;
 #endif
 #if defined(ENABLE_WEBRTC)
-    rtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
+    webrtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
 #endif
     bool m_last_callback = false;
     ACE_Future<bool> m_drained;
diff --git a/Library/TeamTalkLib/avstream/SoundLoopback.cpp b/Library/TeamTalkLib/avstream/SoundLoopback.cpp
index 753ff34c9d..d2454c7ffb 100644
--- a/Library/TeamTalkLib/avstream/SoundLoopback.cpp
+++ b/Library/TeamTalkLib/avstream/SoundLoopback.cpp
@@ -27,6 +27,8 @@
 
 #if defined(ENABLE_WEBRTC)
 #include "avstream/WebRTCPreprocess.h"
+#include <api/audio/builtin_audio_processing_builder.h>
+#include <api/environment/environment_factory.h>
 #endif
 
 #include <cassert>
@@ -127,7 +129,7 @@ bool SoundLoopback::StartTest(int inputdevid, int outputdevid,
 #if defined(ENABLE_WEBRTC)
     if (IsEnabled(apm_cfg))
     {
-        m_apm = webrtc::AudioProcessingBuilder().Create();
+        m_apm = webrtc::BuiltinAudioProcessingBuilder().Build(webrtc::CreateEnvironment());
         if (!m_apm)
         {
             StopTest();
@@ -222,7 +224,7 @@ bool SoundLoopback::StartDuplexTest(int inputdevid, int outputdevid,
 #if defined(ENABLE_WEBRTC)
     if (IsEnabled(apm_cfg))
     {
-        m_apm = webrtc::AudioProcessingBuilder().Create();
+        m_apm = webrtc::BuiltinAudioProcessingBuilder().Build(webrtc::CreateEnvironment());
         if (!m_apm)
         {
             StopTest();
diff --git a/Library/TeamTalkLib/avstream/SoundLoopback.h b/Library/TeamTalkLib/avstream/SoundLoopback.h
index 8e6597ffd8..033fd46a8b 100644
--- a/Library/TeamTalkLib/avstream/SoundLoopback.h
+++ b/Library/TeamTalkLib/avstream/SoundLoopback.h
@@ -108,7 +108,7 @@ class SoundLoopback
     SpeexPreprocess m_preprocess_left, m_preprocess_right;
 #endif
 #if defined(ENABLE_WEBRTC)
-    rtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
+    webrtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
     std::vector<short> m_prev_buffer;
 #endif
     std::vector<short> m_preprocess_buffer_left, m_preprocess_buffer_right;
diff --git a/Library/TeamTalkLib/avstream/WebRTCPreprocess.cpp b/Library/TeamTalkLib/avstream/WebRTCPreprocess.cpp
index 95d518a430..fb28f2b028 100644
--- a/Library/TeamTalkLib/avstream/WebRTCPreprocess.cpp
+++ b/Library/TeamTalkLib/avstream/WebRTCPreprocess.cpp
@@ -26,7 +26,6 @@
 #include "codec/MediaUtil.h"
 #include "myace/MyACE.h"
 
-#include <absl/strings/string_view.h>
 #include <api/audio/audio_processing.h>
 #include <api/audio/audio_processing_statistics.h>
 #include <ace/SString.h>
@@ -38,24 +37,6 @@
 
 constexpr auto DEBUG_WEBRTC = 0;
 
-// webrtc::GainControlImpl queries this feature. Field trials is
-// excluded by passing rtc_exclude_field_trial_default=true to GN.
-namespace webrtc::field_trial {
-std::string FindFullName(absl::string_view trial_)
-{
-    std::string const trial(trial_);;
-#if defined(UNICODE)
-    ACE_TString str = LocalToUnicode(trial.c_str());
-#else
-    ACE_TString const str = trial.c_str();
-#endif
-    MYTRACE(ACE_TEXT("Querying feature: %s\n"), str.c_str());
-    return "Disabled";
-    //return "Enabled";
-}
-} // namespace field_trial
-  // namespace webrtc
-
 bool IsEnabled(const webrtc::AudioProcessing::Config& cfg)
 {
     return cfg.gain_controller1.enabled ||
@@ -136,7 +117,10 @@ int WebRTCPreprocess(webrtc::AudioProcessing& apm, const media::AudioFrame& infr
         if (stats != nullptr)
         {
             auto wstats = apm.GetStatistics();
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
             voice_detected |= wstats.voice_detected.value_or(false);
+#pragma clang diagnostic pop
         }
 
         in_index += in_cfg.num_frames();
@@ -146,7 +130,10 @@ int WebRTCPreprocess(webrtc::AudioProcessing& apm, const media::AudioFrame& infr
 
     if ((stats != nullptr) && n > 0)
     {
+#pragma clang diagnostic push
+#pragma clang diagnostic ignored "-Wdeprecated-declarations"
         stats->voice_detected = voice_detected;
+#pragma clang diagnostic pop
     }
 
     return infrm.input_samples;
diff --git a/Library/TeamTalkLib/build/webrtc/0001-Add-support-for-Visual-Studio-2026.patch b/Library/TeamTalkLib/build/webrtc/0001-Add-support-for-Visual-Studio-2026.patch
deleted file mode 100644
index 9ff995efde..0000000000
--- a/Library/TeamTalkLib/build/webrtc/0001-Add-support-for-Visual-Studio-2026.patch
+++ /dev/null
@@ -1,83 +0,0 @@
-From b451a63117d8bdf6a1c57518b1e45798e7bd0963 Mon Sep 17 00:00:00 2001
-From: hwangsihu <129564966+hwangsihu@users.noreply.github.com>
-Date: Thu, 4 Dec 2025 20:02:11 +0900
-Subject: [PATCH] Add support for Visual Studio 2026
-
----
- build/vs_toolchain.py | 22 +++++++++++++++-------
- 1 file changed, 15 insertions(+), 7 deletions(-)
-
-diff --git a/build/vs_toolchain.py b/build/vs_toolchain.py
-index f3a6fc7de..ecb7102b6 100755
---- a/build/vs_toolchain.py
-+++ b/build/vs_toolchain.py
-@@ -49,7 +49,8 @@ json_data_file = os.path.join(script_dir, 'win_toolchain.json')
- # The first version is assumed by this script to be the one that is packaged,
- # which makes a difference for the arm64 runtime.
- MSVS_VERSIONS = collections.OrderedDict([
--    ('2022', '17.0'),  # Default and packaged version of Visual Studio.
-+    ('2026', '18.0'),  # Default and packaged version of Visual Studio.
-+    ('2022', '17.0'),
-     ('2019', '16.0'),
-     ('2017', '15.0'),
- ])
-@@ -57,11 +58,16 @@ MSVS_VERSIONS = collections.OrderedDict([
- # List of preferred VC toolset version based on MSVS
- # Order is not relevant for this dictionary.
- MSVC_TOOLSET_VERSION = {
-+    '2026': 'VC145',
-     '2022': 'VC143',
-     '2019': 'VC142',
-     '2017': 'VC141',
- }
- 
-+MSVS_YEAR_TO_PATH = {
-+    '2026': '18',
-+}
-+
- def _HostIsWindows():
-   """Returns True if running on a Windows host (including under cygwin)."""
-   return sys.platform in ('win32', 'cygwin')
-@@ -189,8 +195,9 @@ def GetVisualStudioVersion():
-       program_files_path_variable = '%ProgramFiles%'
-     else:
-       program_files_path_variable = '%ProgramFiles(x86)%'
-+    path_version = MSVS_YEAR_TO_PATH.get(version, version)
-     path = os.path.expandvars(program_files_path_variable +
--                              '/Microsoft Visual Studio/%s' % version)
-+                              '/Microsoft Visual Studio/%s' % path_version)
-     if path and any(
-         os.path.exists(os.path.join(path, edition))
-         for edition in ('Enterprise', 'Professional', 'Community', 'Preview',
-@@ -220,22 +227,23 @@ def DetectVisualStudioPath():
-     program_files_path_variable = '%ProgramFiles%'
-   else:
-     program_files_path_variable = '%ProgramFiles(x86)%'
-+  path_version = MSVS_YEAR_TO_PATH.get(version_as_year, version_as_year)
-   for path in (os.environ.get('vs%s_install' % version_as_year),
-                os.path.expandvars(program_files_path_variable +
-                                   '/Microsoft Visual Studio/%s/Enterprise' %
--                                  version_as_year),
-+                                  path_version),
-                os.path.expandvars(program_files_path_variable +
-                                   '/Microsoft Visual Studio/%s/Professional' %
--                                  version_as_year),
-+                                  path_version),
-                os.path.expandvars(program_files_path_variable +
-                                   '/Microsoft Visual Studio/%s/Community' %
--                                  version_as_year),
-+                                  path_version),
-                os.path.expandvars(program_files_path_variable +
-                                   '/Microsoft Visual Studio/%s/Preview' %
--                                  version_as_year),
-+                                  path_version),
-                os.path.expandvars(program_files_path_variable +
-                                   '/Microsoft Visual Studio/%s/BuildTools' %
--                                  version_as_year)):
-+                                  path_version)):
-     if path and os.path.exists(path):
-       return path
- 
--- 
-2.51.2.windows.1
-
diff --git a/Library/TeamTalkLib/build/webrtc/0001-Add-teamtalk-static-lib.patch b/Library/TeamTalkLib/build/webrtc/0001-Add-teamtalk-static-lib.patch
new file mode 100644
index 0000000000..033368706f
--- /dev/null
+++ b/Library/TeamTalkLib/build/webrtc/0001-Add-teamtalk-static-lib.patch
@@ -0,0 +1,32 @@
+From 5fe3c2ad5ed6e8942774ca2aaddadc4cbac7591a Mon Sep 17 00:00:00 2001
+From: Sihu Hwang <129564966+hwangsihu@users.noreply.github.com>
+Date: Fri, 20 Feb 2026 11:11:48 +0900
+Subject: [PATCH] Add teamtalk static lib
+
+---
+ modules/audio_processing/BUILD.gn | 9 +++++++++
+ 1 file changed, 9 insertions(+)
+
+diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
+index 2997c457a3..24b08ec06b 100644
+--- a/modules/audio_processing/BUILD.gn
++++ b/modules/audio_processing/BUILD.gn
+@@ -228,6 +228,15 @@ rtc_library("audio_processing") {
+   }
+ }
+ 
++static_library("teamtalk") {
++  complete_static_lib = true
++  deps = [
++    ":audio_processing",
++    "../../api/audio:builtin_audio_processing_builder",
++    "../../api/environment:environment_factory",
++  ]
++}
++
+ rtc_library("residual_echo_detector") {
+   poisonous = [ "default_echo_detector" ]
+   configs += [ ":apm_debug_dump" ]
+-- 
+2.53.0.windows.1
+
diff --git a/Library/TeamTalkLib/build/webrtc/CMakeLists.txt b/Library/TeamTalkLib/build/webrtc/CMakeLists.txt
index 230a6a35ac..a4f3d79585 100644
--- a/Library/TeamTalkLib/build/webrtc/CMakeLists.txt
+++ b/Library/TeamTalkLib/build/webrtc/CMakeLists.txt
@@ -9,7 +9,7 @@ include(ExternalProject)
 set (TOOLCHAIN_INSTALL_PREFIX_WEBRTC ${TOOLCHAIN_INSTALL_PREFIX}/webrtc)
 
 if (MSVC)
-  # WebRTC for Windows depends on "Windows 11 SDK (10.0.22621.0)". This
+  # WebRTC for Windows depends on "Windows 11 SDK (10.0.26100.0)". This
   # can be installed from Visual Studio Installer.
   #
   # WebRTC for Windows depends on "C++ Clang Compiler for Windows (18.1.8)".
@@ -69,14 +69,12 @@ if (MSVC)
       file(MAKE_DIRECTORY ${WEBRTC_FETCH_PATH})
       set (WEBRTC_DOWNLOAD_CMD ${CMAKE_CURRENT_BINARY_DIR}/depottools-win.bat)
       set (WEBRTC_PATCH_CMD cd ${WEBRTC_SOURCE_ROOT})
-      set (WEBRTC_PATCH_CMD2 CALL git apply ${CMAKE_CURRENT_LIST_DIR}\\libteamtalk-r6818.patch)
-      set (WEBRTC_PATCH_CMD3 CALL git apply ${CMAKE_CURRENT_LIST_DIR}\\0001-Add-support-for-Visual-Studio-2026.patch)
+      set (WEBRTC_PATCH_CMD2 CALL git apply ${CMAKE_CURRENT_LIST_DIR}\\0001-Add-teamtalk-static-lib.patch)
     else()
       message (WARNING "WebRTC is already downloaded to ${WEBRTC_SOURCE_ROOT}. Skipping new fetch and patching")
       set (WEBRTC_DOWNLOAD_CMD "")
       set (WEBRTC_PATCH_CMD "")
       set (WEBRTC_PATCH_CMD2 "")
-      set (WEBRTC_PATCH_CMD3 "")
     endif()
 
     configure_file(${CMAKE_CURRENT_LIST_DIR}/webrtc_cfg_win.bat_sample ${CMAKE_CURRENT_BINARY_DIR}/webrtc_cfg_win.bat)
@@ -87,7 +85,6 @@ if (MSVC)
       DOWNLOAD_COMMAND  ${WEBRTC_DOWNLOAD_CMD}
       PATCH_COMMAND     ${WEBRTC_PATCH_CMD}
       COMMAND           ${WEBRTC_PATCH_CMD2}
-      COMMAND           ${WEBRTC_PATCH_CMD3}
       INSTALL_DIR       ${WEBRTC_INSTALL_ROOT}
       CONFIGURE_COMMAND ${CMAKE_CURRENT_BINARY_DIR}/webrtc_cfg_win.bat
       COMMAND           cd ${WEBRTC_SOURCE_ROOT}
@@ -184,7 +181,7 @@ else()
 
   elseif (${CMAKE_SYSTEM_NAME} MATCHES "Linux")
     set (WEBRTC_FETCH_PATH $ENV{HOME}/webrtc/${CMAKE_SYSTEM_NAME} CACHE FILEPATH "Root folder for WebRTC repository")
-    configure_file(${CMAKE_CURRENT_LIST_DIR}/args.gn_deb ${WEBRTC_INSTALL_ROOT}/args.gn @ONLY)
+    configure_file(${CMAKE_CURRENT_LIST_DIR}/args.gn_linux ${WEBRTC_INSTALL_ROOT}/args.gn @ONLY)
 
   else()
       message(WARNING "Unknown platform for WebRTC")
@@ -229,9 +226,9 @@ else()
     if (NOT EXISTS ${WEBRTC_SOURCE_ROOT})
       message (WARNING "WebRTC is not present in ${WEBRTC_SOURCE_ROOT}. Downloading... This takes a loooong time")
       set (WEBRTC_FETCH_CMD1 ${DEPOTTOOLS_EXPORT} && cd ${WEBRTC_FETCH_PATH} && fetch --nohooks ${WEBRTC_REPO_NAME} && gclient sync --with_branch_heads --with_tags)
-      set (WEBRTC_FETCH_CMD2 ${DEPOTTOOLS_EXPORT} && cd ${WEBRTC_SOURCE_ROOT} && git checkout branch-heads/6818)
+      set (WEBRTC_FETCH_CMD2 ${DEPOTTOOLS_EXPORT} && cd ${WEBRTC_SOURCE_ROOT} && git checkout branch-heads/7727)
       set (WEBRTC_FETCH_CMD3 ${DEPOTTOOLS_EXPORT} && cd ${WEBRTC_FETCH_PATH} && gclient sync -D)
-      set (WEBRTC_PATCH_CMD1 cd ${WEBRTC_SOURCE_ROOT} && git apply ${CMAKE_CURRENT_LIST_DIR}/libteamtalk-r6818.patch)
+      set (WEBRTC_PATCH_CMD1 cd ${WEBRTC_SOURCE_ROOT} && git apply ${CMAKE_CURRENT_LIST_DIR}/0001-Add-teamtalk-static-lib.patch)
     else()
       set (WEBRTC_FETCH_CMD1 true)
       set (WEBRTC_FETCH_CMD2 true)
diff --git a/Library/TeamTalkLib/build/webrtc/args.gn_android b/Library/TeamTalkLib/build/webrtc/args.gn_android
index b04a24a709..0977bbd822 100644
--- a/Library/TeamTalkLib/build/webrtc/args.gn_android
+++ b/Library/TeamTalkLib/build/webrtc/args.gn_android
@@ -1,6 +1,4 @@
 gtest_enable_absl_printers=false
-is_clang=false
-is_component_build=false
 is_debug=false
 libsrtp_build_boringssl=false
 libsrtp_ssl_root="/dev/null"
@@ -12,7 +10,6 @@ rtc_builtin_ssl_root_certificates=false
 rtc_disable_logging=true
 rtc_enable_protobuf=false
 rtc_enable_sctp=false
-rtc_exclude_field_trial_default=true
 rtc_include_opus=false
 rtc_include_tests=false
 rtc_libvpx_build_vp9=false
diff --git a/Library/TeamTalkLib/build/webrtc/args.gn_ios b/Library/TeamTalkLib/build/webrtc/args.gn_ios
index e0b05fe94d..9e280b5aba 100644
--- a/Library/TeamTalkLib/build/webrtc/args.gn_ios
+++ b/Library/TeamTalkLib/build/webrtc/args.gn_ios
@@ -3,7 +3,6 @@ enable_dsyms=true
 gtest_enable_absl_printers=false
 ios_deployment_target="@CMAKE_OSX_DEPLOYMENT_TARGET@"
 ios_enable_code_signing=false
-is_component_build=false
 is_debug=false
 libcxx_is_shared=false
 libsrtp_build_boringssl = false
@@ -16,7 +15,6 @@ rtc_builtin_ssl_root_certificates=false
 rtc_disable_logging=true
 rtc_enable_protobuf=false
 rtc_enable_sctp=false
-rtc_exclude_field_trial_default=true
 rtc_include_opus=false
 rtc_include_tests=false
 rtc_libvpx_build_vp9=false
diff --git a/Library/TeamTalkLib/build/webrtc/args.gn_deb b/Library/TeamTalkLib/build/webrtc/args.gn_linux
similarity index 86%
rename from Library/TeamTalkLib/build/webrtc/args.gn_deb
rename to Library/TeamTalkLib/build/webrtc/args.gn_linux
index 3fdec2ad27..aaf459c7bc 100644
--- a/Library/TeamTalkLib/build/webrtc/args.gn_deb
+++ b/Library/TeamTalkLib/build/webrtc/args.gn_linux
@@ -1,6 +1,5 @@
 gtest_enable_absl_printers=false
 is_clang=false
-is_component_build=false
 is_debug=false
 libsrtp_build_boringssl=false
 libsrtp_ssl_root="/dev/null"
@@ -12,7 +11,6 @@ rtc_builtin_ssl_root_certificates=false
 rtc_disable_logging=true
 rtc_enable_protobuf=false
 rtc_enable_sctp=false
-rtc_exclude_field_trial_default=true
 rtc_include_opus=false
 rtc_include_tests=false
 rtc_libvpx_build_vp9=false
@@ -21,8 +19,9 @@ rtc_use_x11=false
 rtc_use_x11_extensions=false
 treat_warnings_as_errors=false
 use_aura=false
-use_bundled_fontconfig=true
 use_custom_libcxx=false
+use_llvm_libatomic=false
 use_ozone=false
 use_rtti=true
+use_sysroot=false
 use_udev=false
diff --git a/Library/TeamTalkLib/build/webrtc/args.gn_mac b/Library/TeamTalkLib/build/webrtc/args.gn_mac
index 78c7e17f82..390606b612 100644
--- a/Library/TeamTalkLib/build/webrtc/args.gn_mac
+++ b/Library/TeamTalkLib/build/webrtc/args.gn_mac
@@ -1,5 +1,4 @@
 gtest_enable_absl_printers = false
-is_clang = false
 is_debug = false
 libsrtp_build_boringssl = false
 libsrtp_ssl_root = "/dev/null"
@@ -12,7 +11,6 @@ rtc_builtin_ssl_root_certificates = false
 rtc_disable_logging = true
 rtc_enable_protobuf = false
 rtc_enable_sctp = false
-rtc_exclude_field_trial_default = true
 rtc_include_opus = false
 rtc_include_tests = false
 rtc_libvpx_build_vp9 = false
diff --git a/Library/TeamTalkLib/build/webrtc/args.gn_win_debug b/Library/TeamTalkLib/build/webrtc/args.gn_win_debug
index 0f0019d694..90cbbcb1c2 100644
--- a/Library/TeamTalkLib/build/webrtc/args.gn_win_debug
+++ b/Library/TeamTalkLib/build/webrtc/args.gn_win_debug
@@ -1,5 +1,4 @@
 enable_iterator_debugging=true
-is_clang=true
 is_debug=true
 libsrtp_build_boringssl=false
 libsrtp_ssl_root=""
@@ -11,7 +10,6 @@ rtc_builtin_ssl_root_certificates=false
 rtc_disable_logging=true
 rtc_enable_protobuf=false
 rtc_enable_sctp=false
-rtc_exclude_field_trial_default=true
 rtc_include_opus=false
 rtc_include_tests=false
 rtc_libvpx_build_vp9=false
diff --git a/Library/TeamTalkLib/build/webrtc/args.gn_win_release b/Library/TeamTalkLib/build/webrtc/args.gn_win_release
index 699bd0fde5..be076854d3 100644
--- a/Library/TeamTalkLib/build/webrtc/args.gn_win_release
+++ b/Library/TeamTalkLib/build/webrtc/args.gn_win_release
@@ -1,4 +1,3 @@
-is_clang=true
 is_debug=false
 libsrtp_build_boringssl=false
 libsrtp_ssl_root=""
@@ -10,7 +9,6 @@ rtc_builtin_ssl_root_certificates=false
 rtc_disable_logging=true
 rtc_enable_protobuf=false
 rtc_enable_sctp=false
-rtc_exclude_field_trial_default=true
 rtc_include_opus=false
 rtc_include_tests=false
 rtc_libvpx_build_vp9=false
diff --git a/Library/TeamTalkLib/build/webrtc/depottools-win.bat_sample b/Library/TeamTalkLib/build/webrtc/depottools-win.bat_sample
index cd9e7d34ba..41793ec23d 100644
--- a/Library/TeamTalkLib/build/webrtc/depottools-win.bat_sample
+++ b/Library/TeamTalkLib/build/webrtc/depottools-win.bat_sample
@@ -6,7 +6,7 @@ CALL fetch --nohooks webrtc
 IF NOT "%ERRORLEVEL%" == "0" GOTO fail
 CD @WEBRTC_SOURCE_ROOT_NATIVE@
 IF NOT "%ERRORLEVEL%" == "0" GOTO fail
-CALL git checkout branch-heads/6818
+CALL git checkout branch-heads/7727
 IF NOT "%ERRORLEVEL%" == "0" GOTO fail
 CD @WEBRTC_FETCH_PATH_NATIVE@
 IF NOT "%ERRORLEVEL%" == "0" GOTO fail
diff --git a/Library/TeamTalkLib/build/webrtc/libteamtalk-r6818.patch b/Library/TeamTalkLib/build/webrtc/libteamtalk-r6818.patch
deleted file mode 100644
index 8e1372ca3b..0000000000
--- a/Library/TeamTalkLib/build/webrtc/libteamtalk-r6818.patch
+++ /dev/null
@@ -1,18 +0,0 @@
-diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
-index 03a63bb6c0..636d21f7c7 100644
---- a/modules/audio_processing/BUILD.gn
-+++ b/modules/audio_processing/BUILD.gn
-@@ -209,6 +209,13 @@ rtc_library("audio_processing") {
-   }
- }
- 
-+static_library("teamtalk") {
-+  complete_static_lib = true
-+  deps = [
-+    ":audio_processing",
-+  ]
-+}
-+
- rtc_library("residual_echo_detector") {
-   poisonous = [ "default_echo_detector" ]
-   configs += [ ":apm_debug_dump" ]
diff --git a/Library/TeamTalkLib/teamtalk/client/AudioThread.cpp b/Library/TeamTalkLib/teamtalk/client/AudioThread.cpp
index e05f6296a2..706b2e778d 100644
--- a/Library/TeamTalkLib/teamtalk/client/AudioThread.cpp
+++ b/Library/TeamTalkLib/teamtalk/client/AudioThread.cpp
@@ -30,6 +30,8 @@
 
 #if defined(ENABLE_WEBRTC)
 #include "avstream/WebRTCPreprocess.h"
+#include <api/audio/builtin_audio_processing_builder.h>
+#include <api/environment/environment_factory.h>
 #endif
 
 #include <cassert>
@@ -277,7 +279,7 @@ bool AudioThread::UpdatePreprocessor(const teamtalk::AudioPreprocessor& preproce
         }
 
         if (!m_apm)
-            m_apm = webrtc::AudioProcessingBuilder().Create();
+            m_apm = webrtc::BuiltinAudioProcessingBuilder().Build(webrtc::CreateEnvironment());
         m_apm->ApplyConfig(preprocess.webrtc);
         if (m_apm->Initialize() != webrtc::AudioProcessing::kNoError)
         {
diff --git a/Library/TeamTalkLib/teamtalk/client/AudioThread.h b/Library/TeamTalkLib/teamtalk/client/AudioThread.h
index 3474f4298e..e2e6608cb4 100644
--- a/Library/TeamTalkLib/teamtalk/client/AudioThread.h
+++ b/Library/TeamTalkLib/teamtalk/client/AudioThread.h
@@ -116,7 +116,7 @@ class AudioThread : protected ACE_Task<ACE_MT_SYNCH>
     std::unique_ptr<SpeexPreprocess> m_preprocess_left, m_preprocess_right;
 #endif
 #if defined(ENABLE_WEBRTC)
-    rtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
+    webrtc::scoped_refptr<webrtc::AudioProcessing> m_apm;
     std::unique_ptr<webrtc::AudioProcessingStats> m_aps;
 #endif
 #if defined(ENABLE_SPEEX)
diff --git a/Library/TeamTalkLib/test/CatchWebRTC.cpp b/Library/TeamTalkLib/test/CatchWebRTC.cpp
index ab78e07d3f..011e1dde68 100644
--- a/Library/TeamTalkLib/test/CatchWebRTC.cpp
+++ b/Library/TeamTalkLib/test/CatchWebRTC.cpp
@@ -27,6 +27,8 @@
 #include "codec/WaveFile.h"
 
 #include <api/audio/audio_processing.h>
+#include <api/audio/builtin_audio_processing_builder.h>
+#include <api/environment/environment_factory.h>
 #include <modules/audio_processing/agc/gain_control.h>
 #include <modules/audio_processing/audio_buffer.h>
 #include <modules/audio_processing/ns/noise_suppressor.h>
@@ -169,7 +171,7 @@ TEST_CASE("webrtc-apm")
     std::vector<int16_t> in_buff(af.channels * in_cfg.num_frames());
     std::vector<int16_t> apm_buff(af.channels * out_cfg.num_frames());
 
-    auto apm = webrtc::AudioProcessingBuilder().Create();
+    auto apm = webrtc::BuiltinAudioProcessingBuilder().Build(webrtc::CreateEnvironment());
 
     // first try gain_controller1
 
@@ -228,7 +230,7 @@ TEST_CASE("webrtc-double-gain")
     std::vector<int16_t> in_buff(af.channels * in_cfg.num_frames());
     std::vector<int16_t> apm_buff(af.channels * out_cfg.num_frames());
 
-    auto apm = webrtc::AudioProcessingBuilder().Create();
+    auto apm = webrtc::BuiltinAudioProcessingBuilder().Build(webrtc::CreateEnvironment());
 
     webrtc::AudioProcessing::Config apm_cfg;
     rawfile.Close();