Update WebRTC to R7727

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 \

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

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'

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)

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;

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();

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;

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;

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
+

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)

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

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

Library/TeamTalkLib/build/webrtc/args.gn_deb → 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

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

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