path: root/third_party/libwebrtc/moz-patch-stack/0035.patch

From: stransky <stransky@redhat.com>
Date: Thu, 5 Nov 2020 07:47:00 +0000
Subject: Bug 1654112 - Tweak upstream gn files for Firefox build. r=ng

Differential Revision: https://phabricator.services.mozilla.com/D130075
Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/127ace4d8887f11abb201d300a849772a2b519f8
---
 .gn                                           |  2 ++
 BUILD.gn                                      | 26 +++++++++++++-
 api/BUILD.gn                                  |  8 ++++-
 api/task_queue/BUILD.gn                       |  2 ++
 api/transport/BUILD.gn                        |  2 ++
 common_audio/BUILD.gn                         |  4 ---
 common_audio/fir_filter_avx2.cc               |  2 ++
 common_audio/intrin.h                         |  8 +++++
 media/BUILD.gn                                | 30 ++++++++++++++++
 modules/audio_coding/BUILD.gn                 |  2 +-
 modules/audio_device/BUILD.gn                 |  8 ++---
 modules/audio_processing/aec3/BUILD.gn        | 13 +++----
 .../aec3/adaptive_fir_filter_avx2.cc          |  3 +-
 .../audio_processing/agc2/rnn_vad/BUILD.gn    |  2 +-
 modules/desktop_capture/BUILD.gn              | 29 +---------------
 modules/portal/BUILD.gn                       | 24 +++++++++++++
 modules/video_capture/BUILD.gn                | 11 +-----
 modules/video_coding/BUILD.gn                 |  6 ++++
 rtc_base/BUILD.gn                             |  4 ++-
 rtc_base/system/BUILD.gn                      |  2 +-
 test/BUILD.gn                                 | 14 ++++++++
 webrtc.gni                                    | 34 ++++++++++++-------
 22 files changed, 162 insertions(+), 74 deletions(-)
 create mode 100644 common_audio/intrin.h

diff --git a/.gn b/.gn
index c9824916ad..d35ea79ced 100644
--- a/.gn
+++ b/.gn
@@ -69,6 +69,8 @@ default_args = {
   # Prevent jsoncpp to pass -Wno-deprecated-declarations to users
   jsoncpp_no_deprecated_declarations = false
 
+  use_custom_libcxx = false
+
   # Fixes the abi-revision issue.
   # TODO(https://bugs.webrtc.org/14437):  Remove this section if general
   # Chromium fix resolves the problem.
diff --git a/BUILD.gn b/BUILD.gn
index 5817d22227..8b2648a306 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -33,7 +33,7 @@ if (is_android) {
   import("//build/config/android/rules.gni")
 }
 
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
   # This target should (transitively) cause everything to be built; if you run
   # 'ninja default' and then 'ninja all', the second build should do no work.
   group("default") {
@@ -152,6 +152,10 @@ config("common_inherited_config") {
     defines += [ "WEBRTC_ENABLE_OBJC_SYMBOL_EXPORT" ]
   }
 
+  if (build_with_mozilla) {
+    defines += [ "WEBRTC_MOZILLA_BUILD" ]
+  }
+
   if (!rtc_builtin_ssl_root_certificates) {
     defines += [ "WEBRTC_EXCLUDE_BUILT_IN_SSL_ROOT_CERTS" ]
   }
@@ -460,9 +464,11 @@ config("common_config") {
   }
 }
 
+if (is_mac) {
 config("common_objc") {
   frameworks = [ "Foundation.framework" ]
 }
+}
 
 if (!build_with_chromium) {
   # Target to build all the WebRTC production code.
@@ -502,6 +508,23 @@ if (!build_with_chromium) {
       "sdk",
       "video",
     ]
+    if (build_with_mozilla) {
+      deps -= [
+        "api:create_peerconnection_factory",
+        "api:rtc_error",
+        "api:transport_api",
+        "api/crypto",
+        "api/rtc_event_log:rtc_event_log_factory",
+        "api/task_queue",
+        "api/task_queue:default_task_queue_factory",
+        "api/test/metrics",
+        "logging:rtc_event_log_api",
+        "p2p:rtc_p2p",
+        "pc:libjingle_peerconnection",
+        "pc:rtc_pc",
+        "sdk",
+      ]
+    }
 
     if (rtc_include_builtin_audio_codecs) {
       deps += [
@@ -521,6 +544,7 @@ if (!build_with_chromium) {
       deps += [
         "api/video:video_frame",
         "api/video:video_rtp_headers",
+        "test:rtp_test_utils",
       ]
     } else {
       deps += [
diff --git a/api/BUILD.gn b/api/BUILD.gn
index 33a6b0aaa6..ab5d6c91ce 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -35,7 +35,7 @@ rtc_source_set("callfactory_api") {
   ]
 }
 
-if (!build_with_chromium) {
+if (!build_with_chromium && !build_with_mozilla) {
   rtc_library("create_peerconnection_factory") {
     visibility = [ "*" ]
     allow_poison = [ "default_task_queue" ]
@@ -227,6 +227,7 @@ rtc_library("rtp_sender_interface") {
 }
 
 rtc_library("libjingle_peerconnection_api") {
+if (!build_with_mozilla) {
   visibility = [ "*" ]
   cflags = []
   sources = [
@@ -343,6 +344,7 @@ rtc_library("libjingle_peerconnection_api") {
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
+}
 
 rtc_source_set("frame_transformer_interface") {
   visibility = [ "*" ]
@@ -550,6 +552,7 @@ rtc_source_set("peer_network_dependencies") {
 }
 
 rtc_source_set("peer_connection_quality_test_fixture_api") {
+if (!build_with_mozilla) {
   visibility = [ "*" ]
   testonly = true
   sources = [ "test/peerconnection_quality_test_fixture.h" ]
@@ -600,6 +603,7 @@ rtc_source_set("peer_connection_quality_test_fixture_api") {
     "//third_party/abseil-cpp/absl/types:optional",
   ]
 }
+}
 
 rtc_source_set("frame_generator_api") {
   visibility = [ "*" ]
@@ -885,6 +889,7 @@ rtc_source_set("refcountedbase") {
   ]
 }
 
+if (!build_with_mozilla) {
 rtc_library("ice_transport_factory") {
   visibility = [ "*" ]
   sources = [
@@ -903,6 +908,7 @@ rtc_library("ice_transport_factory") {
     "rtc_event_log:rtc_event_log",
   ]
 }
+}
 
 rtc_library("neteq_simulator_api") {
   visibility = [ "*" ]
diff --git a/api/task_queue/BUILD.gn b/api/task_queue/BUILD.gn
index 69393b80ff..c9b4a5d0ec 100644
--- a/api/task_queue/BUILD.gn
+++ b/api/task_queue/BUILD.gn
@@ -30,6 +30,7 @@ rtc_library("task_queue") {
   ]
 }
 
+if (rtc_include_tests) {
 rtc_library("task_queue_test") {
   visibility = [ "*" ]
   testonly = true
@@ -78,6 +79,7 @@ rtc_library("task_queue_test") {
     ]
   }
 }
+}
 
 rtc_library("default_task_queue_factory") {
   visibility = [ "*" ]
diff --git a/api/transport/BUILD.gn b/api/transport/BUILD.gn
index 08f3d6d1d0..e0b31122b2 100644
--- a/api/transport/BUILD.gn
+++ b/api/transport/BUILD.gn
@@ -90,6 +90,7 @@ rtc_source_set("sctp_transport_factory_interface") {
 }
 
 rtc_source_set("stun_types") {
+if (!build_with_mozilla) {
   visibility = [ "*" ]
   sources = [
     "stun.cc",
@@ -110,6 +111,7 @@ rtc_source_set("stun_types") {
   ]
   absl_deps = [ "//third_party/abseil-cpp/absl/strings" ]
 }
+}
 
 if (rtc_include_tests) {
   rtc_source_set("test_feedback_generator_interface") {
diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
index 2ae6d32710..a45214f754 100644
--- a/common_audio/BUILD.gn
+++ b/common_audio/BUILD.gn
@@ -267,14 +267,10 @@ if (current_cpu == "x86" || current_cpu == "x64") {
       "resampler/sinc_resampler_avx2.cc",
     ]
 
-    if (is_win) {
-      cflags = [ "/arch:AVX2" ]
-    } else {
       cflags = [
         "-mavx2",
         "-mfma",
       ]
-    }
 
     deps = [
       ":fir_filter",
diff --git a/common_audio/fir_filter_avx2.cc b/common_audio/fir_filter_avx2.cc
index 9cb0f770ca..0031392f8a 100644
--- a/common_audio/fir_filter_avx2.cc
+++ b/common_audio/fir_filter_avx2.cc
@@ -15,6 +15,8 @@
 #include <string.h>
 #include <xmmintrin.h>
 
+#include "common_audio/intrin.h"
+
 #include "rtc_base/checks.h"
 #include "rtc_base/memory/aligned_malloc.h"
 
diff --git a/common_audio/intrin.h b/common_audio/intrin.h
new file mode 100644
index 0000000000..f6ff7f218f
--- /dev/null
+++ b/common_audio/intrin.h
@@ -0,0 +1,8 @@
+#if defined (__SSE__)
+  #include <immintrin.h>
+  #if defined (__clang__)
+    #include <avxintrin.h>
+    #include <avx2intrin.h>
+    #include <fmaintrin.h>
+  #endif
+#endif
diff --git a/media/BUILD.gn b/media/BUILD.gn
index 2e5c0bb872..4ddc8349a8 100644
--- a/media/BUILD.gn
+++ b/media/BUILD.gn
@@ -147,6 +147,27 @@ rtc_library("rtc_media_base") {
     "base/video_source_base.cc",
     "base/video_source_base.h",
   ]
+  if (build_with_mozilla) {
+    sources -= [
+      "base/adapted_video_track_source.cc",
+      "base/adapted_video_track_source.h",
+      "base/audio_source.h",
+      "base/delayable.h",
+      "base/media_channel.h",
+      "base/media_channel_impl.cc",
+      "base/media_channel_impl.h",
+      "base/media_engine.cc",
+      "base/media_engine.h",
+      "base/rid_description.cc",
+      "base/rid_description.h",
+      "base/rtp_utils.cc",
+      "base/rtp_utils.h",
+      "base/stream_params.cc",
+      "base/stream_params.h",
+      "base/turn_utils.cc",
+      "base/turn_utils.h",
+    ]
+  }
 }
 
 rtc_library("media_channel_impl") {
@@ -420,6 +441,9 @@ rtc_library("rtc_internal_video_codecs") {
     "../system_wrappers:field_trial",
     "../test:fake_video_codecs",
   ]
+  if (build_with_mozilla) {
+    deps -= [ "../test:fake_video_codecs" ]
+  }
 
   if (enable_libaom) {
     defines += [ "RTC_USE_LIBAOM_AV1_ENCODER" ]
@@ -449,6 +473,12 @@ rtc_library("rtc_internal_video_codecs") {
     # targets depend on :rtc_encoder_simulcast_proxy directly.
     "engine/encoder_simulcast_proxy.h",
   ]
+  if (build_with_mozilla) {
+    sources -= [
+      "engine/fake_video_codec_factory.cc",
+      "engine/fake_video_codec_factory.h",
+    ]
+  }
 }
 
 rtc_library("rtc_audio_video") {
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index dddc3edd83..eac0650a26 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -553,7 +553,7 @@ rtc_library("webrtc_opus_wrapper") {
     deps += [ rtc_opus_dir ]
     public_configs = [ "//third_party/opus:opus_config" ]
   } else if (build_with_mozilla) {
-    include_dirs = [ getenv("DIST") + "/include/opus" ]
+    public_configs = [ "//third_party/opus:opus_config" ]
   }
 }
 
diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
index 4a6a0ab41c..e35a442025 100644
--- a/modules/audio_device/BUILD.gn
+++ b/modules/audio_device/BUILD.gn
@@ -233,9 +233,9 @@ rtc_library("audio_device_impl") {
   ]
 
   if (build_with_mozilla) {
-    sources += [
-      "opensl/single_rw_fifo.cc",
-      "opensl/single_rw_fifo.h",
+    sources -= [
+      "include/test_audio_device.cc",
+      "include/test_audio_device.h",
     ]
   }
 
@@ -477,7 +477,7 @@ if (rtc_include_tests && !build_with_chromium) {
   }
 }
 
-if (!build_with_chromium && is_android) {
+if ((!build_with_chromium && !build_with_mozilla) && is_android) {
   rtc_android_library("audio_device_java") {
     sources = [
       "android/java/src/org/webrtc/voiceengine/BuildInfo.java",
diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
index f5eb5d5951..3e11a245a1 100644
--- a/modules/audio_processing/aec3/BUILD.gn
+++ b/modules/audio_processing/aec3/BUILD.gn
@@ -264,14 +264,11 @@ if (current_cpu == "x86" || current_cpu == "x64") {
       "vector_math_avx2.cc",
     ]
 
-    if (is_win) {
-      cflags = [ "/arch:AVX2" ]
-    } else {
-      cflags = [
-        "-mavx2",
-        "-mfma",
-      ]
-    }
+    cflags = [
+      "-mavx",
+      "-mavx2",
+      "-mfma",
+    ]
 
     deps = [
       ":adaptive_fir_filter",
diff --git a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
index 6c8c948026..44d4514275 100644
--- a/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
+++ b/modules/audio_processing/aec3/adaptive_fir_filter_avx2.cc
@@ -10,8 +10,7 @@
 
 #include "modules/audio_processing/aec3/adaptive_fir_filter.h"
 
-#include <immintrin.h>
-
+#include "common_audio/intrin.h"
 #include "rtc_base/checks.h"
 
 namespace webrtc {
diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
index 9093a68cf3..3003a585bd 100644
--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
@@ -122,7 +122,7 @@ rtc_source_set("vector_math") {
 if (current_cpu == "x86" || current_cpu == "x64") {
   rtc_library("vector_math_avx2") {
     sources = [ "vector_math_avx2.cc" ]
-    if (is_win) {
+    if (is_win && !build_with_mozilla) {
       cflags = [ "/arch:AVX2" ]
     } else {
       cflags = [
diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
index c9993dd47b..4aa9186527 100644
--- a/modules/desktop_capture/BUILD.gn
+++ b/modules/desktop_capture/BUILD.gn
@@ -342,37 +342,12 @@ rtc_library("desktop_capture") {
     ]
     deps += [ ":desktop_capture_objc" ]
   }
-
-  if (build_with_mozilla) {
-    sources += [
-      "desktop_device_info.cc",
-      "desktop_device_info.h",
-    ]
-    if (is_win) {
-      sources += [
-        "app_capturer_win.cc",
-        "win/desktop_device_info_win.cc",
-        "win/win_shared.cc",
-      ]
-    }
-  }
   if (rtc_use_x11_extensions || rtc_use_pipewire) {
     sources += [
       "mouse_cursor_monitor_linux.cc",
       "screen_capturer_linux.cc",
       "window_capturer_linux.cc",
     ]
-
-    if (build_with_mozilla && (is_linux || is_chromeos)) {
-      sources += [
-        "app_capturer_linux.cc",
-        "linux/x11/app_capturer_x11.cc",
-        "linux/x11/desktop_device_info_linux.cc",
-        "linux/x11/desktop_device_info_linux.h",
-        "linux/x11/shared_x_util.cc",
-        "linux/x11/shared_x_util.h",
-      ]
-    }
   }
 
   if (rtc_use_x11_extensions) {
@@ -536,9 +511,7 @@ rtc_library("desktop_capture") {
     deps += [ "../../rtc_base:sanitizer" ]
   }
 
-  if (!build_with_mozilla) {
-    deps += [ "//third_party/libyuv" ]
-  }
+  deps += [ "//third_party/libyuv" ]
 
   if (use_desktop_capture_differ_sse2) {
     deps += [ ":desktop_capture_differ_sse2" ]
diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
index d0756f269b..d7768b2323 100644
--- a/modules/portal/BUILD.gn
+++ b/modules/portal/BUILD.gn
@@ -11,6 +11,7 @@ import("//tools/generate_stubs/rules.gni")
 import("../../webrtc.gni")
 
 if ((is_linux || is_chromeos) && rtc_use_pipewire) {
+if (!build_with_mozilla) {
   pkg_config("gio") {
     packages = [
       "gio-2.0",
@@ -88,6 +89,12 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
       defines += [ "WEBRTC_USE_GIO" ]
     }
   }
+} else {
+  config("pipewire_all") {
+  }
+  config("pipewire_config") {
+  }
+}
 
   rtc_library("portal") {
     sources = [
@@ -120,5 +127,22 @@ if ((is_linux || is_chromeos) && rtc_use_pipewire) {
 
       deps += [ ":pipewire_stubs" ]
     }
+
+    if (build_with_mozilla) {
+      configs -= [
+        ":gio",
+        ":pipewire",
+        ":pipewire_config",
+      ]
+      deps -= [ ":pipewire_stubs" ]
+      defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ]
+      public_deps = [
+        "//third_party/pipewire",
+        "//third_party/drm",
+        "//third_party/gbm",
+        "//third_party/libepoxy"
+      ]
+    }
   }
 }
+
diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
index cfa5184d71..95548906c4 100644
--- a/modules/video_capture/BUILD.gn
+++ b/modules/video_capture/BUILD.gn
@@ -93,21 +93,12 @@ if (!build_with_chromium) {
         "strmiids.lib",
         "user32.lib",
       ]
-
-      if (build_with_mozilla) {
-        sources += [
-          "windows/BaseFilter.cpp",
-          "windows/BaseInputPin.cpp",
-          "windows/BasePin.cpp",
-          "windows/MediaType.cpp",
-        ]
-      }
     }
     if (is_fuchsia) {
       sources = [ "video_capture_factory_null.cc" ]
     }
 
-    if (build_with_mozilla && is_android) {
+    if (!build_with_mozilla && is_android) {
       include_dirs = [
         "/config/external/nspr",
         "/nsprpub/lib/ds",
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index fe63804b19..e1b5e4ba84 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -237,6 +237,12 @@ rtc_library("video_coding") {
     "video_receiver2.cc",
     "video_receiver2.h",
   ]
+  if (build_with_mozilla) {
+    sources += [
+      "event_wrapper.cc",
+      "event_wrapper.h",
+    ]
+  }
 
   deps = [
     ":codec_globals_headers",
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 0d8bd4f759..7e162cecbb 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -820,7 +820,9 @@ rtc_library("rtc_json") {
     "strings/json.h",
   ]
   deps = [ ":stringutils" ]
+if (!build_with_mozilla) {
   all_dependent_configs = [ "//third_party/jsoncpp:jsoncpp_config" ]
+}
   if (rtc_build_json) {
     deps += [ "//third_party/jsoncpp" ]
   } else {
@@ -2073,7 +2075,7 @@ if (rtc_include_tests) {
   }
 }
 
-if (is_android) {
+if (is_android && !build_with_mozilla) {
   rtc_android_library("base_java") {
     visibility = [ "*" ]
     sources = [
diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn
index 77f5139a2f..486b37590c 100644
--- a/rtc_base/system/BUILD.gn
+++ b/rtc_base/system/BUILD.gn
@@ -101,7 +101,7 @@ if (is_mac || is_ios) {
 rtc_source_set("warn_current_thread_is_deadlocked") {
   sources = [ "warn_current_thread_is_deadlocked.h" ]
   deps = []
-  if (is_android && !build_with_chromium) {
+  if (is_android && (!build_with_chromium && !build_with_mozilla)) {
     sources += [ "warn_current_thread_is_deadlocked.cc" ]
     deps += [
       "..:logging",
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 0c71115f0d..04a718c411 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -156,6 +156,7 @@ rtc_library("audio_test_common") {
   absl_deps = [ "//third_party/abseil-cpp/absl/memory" ]
 }
 
+if (!build_with_mozilla) {
 if (!build_with_chromium) {
   if (is_mac || is_ios) {
     rtc_library("video_test_mac") {
@@ -204,8 +205,12 @@ if (!build_with_chromium) {
     }
   }
 }
+}
 
 rtc_library("rtp_test_utils") {
+  if (build_with_mozilla) {
+  sources = []
+  } else {
   testonly = true
   sources = [
     "rtcp_packet_parser.cc",
@@ -215,6 +220,7 @@ rtc_library("rtp_test_utils") {
     "rtp_file_writer.cc",
     "rtp_file_writer.h",
   ]
+  }
 
   deps = [
     "../api:array_view",
@@ -467,7 +473,9 @@ rtc_library("video_test_support") {
   absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
 
   if (!is_ios) {
+    if (!build_with_mozilla) {
     deps += [ "//third_party:jpeg" ]
+    }
     sources += [ "testsupport/jpeg_frame_writer.cc" ]
   } else {
     sources += [ "testsupport/jpeg_frame_writer_ios.cc" ]
@@ -1036,6 +1044,10 @@ rtc_library("test_common") {
   if (!is_android && !build_with_chromium) {
     deps += [ "../modules/video_capture:video_capture_internal_impl" ]
   }
+  # This, or some form of it should be upstreamed.
+  if (!rtc_include_tests) {
+    deps -= [ "../rtc_base:task_queue_for_test" ]
+  }
 }
 
 rtc_library("mock_transport") {
@@ -1203,6 +1215,7 @@ rtc_library("copy_to_file_audio_capturer_unittest") {
   ]
 }
 
+if (!build_with_mozilla) {
 if (!build_with_chromium && is_android) {
   rtc_android_library("native_test_java") {
     testonly = true
@@ -1216,6 +1229,7 @@ if (!build_with_chromium && is_android) {
     ]
   }
 }
+}
 
 rtc_library("call_config_utils") {
   # TODO(bugs.webrtc.org/10814): Remove rtc_json_suppressions as soon as it
diff --git a/webrtc.gni b/webrtc.gni
index 8dfcc9d244..6ae1b2329c 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -35,6 +35,11 @@ if (is_mac) {
   import("//build/config/mac/rules.gni")
 }
 
+if (is_android) {
+  import("//build/config/android/config.gni")
+  import("//build/config/android/rules.gni")
+}
+
 if (is_fuchsia) {
   import("//build/config/fuchsia/config.gni")
 }
@@ -42,6 +47,11 @@ if (is_fuchsia) {
 # This declare_args is separated from the next one because args declared
 # in this one, can be read from the next one (args defined in the same
 # declare_args cannot be referenced in that scope).
+declare_args() {
+  # Enable to use the Mozilla internal settings.
+  build_with_mozilla = true
+}
+
 declare_args() {
   # Setting this to true will make RTC_EXPORT (see rtc_base/system/rtc_export.h)
   # expand to code that will manage symbols visibility.
@@ -84,7 +94,7 @@ declare_args() {
   # will tell the pre-processor to remove the default definition of the
   # SystemTimeNanos() which is defined in rtc_base/system_time.cc. In
   # that case a new implementation needs to be provided.
-  rtc_exclude_system_time = build_with_chromium
+  rtc_exclude_system_time = build_with_chromium || build_with_mozilla
 
   # Setting this to false will require the API user to pass in their own
   # SSLCertificateVerifier to verify the certificates presented from a
@@ -110,7 +120,7 @@ declare_args() {
 
   # Used to specify an external OpenSSL include path when not compiling the
   # library that comes with WebRTC (i.e. rtc_build_ssl == 0).
-  rtc_ssl_root = ""
+  rtc_ssl_root = "unused"
 
   # Enable when an external authentication mechanism is used for performing
   # packet authentication for RTP packets instead of libsrtp.
@@ -127,13 +137,13 @@ declare_args() {
   rtc_enable_bwe_test_logging = false
 
   # Set this to false to skip building examples.
-  rtc_build_examples = true
+  rtc_build_examples = false
 
   # Set this to false to skip building tools.
-  rtc_build_tools = true
+  rtc_build_tools = false
 
   # Set this to false to skip building code that requires X11.
-  rtc_use_x11 = ozone_platform_x11
+  rtc_use_x11 = use_x11
 
   # Set this to use PipeWire on the Wayland display server.
   # By default it's only enabled on desktop Linux (excludes ChromeOS) and
@@ -144,9 +154,6 @@ declare_args() {
   # Set this to link PipeWire and required libraries directly instead of using the dlopen.
   rtc_link_pipewire = false
 
-  # Enable to use the Mozilla internal settings.
-  build_with_mozilla = false
-
   # Experimental: enable use of Android AAudio which requires Android SDK 26 or above
   # and NDK r16 or above.
   rtc_enable_android_aaudio = false
@@ -227,7 +234,7 @@ declare_args() {
   # When set to true, a capturer implementation that uses the
   # Windows.Graphics.Capture APIs will be available for use. This introduces a
   # dependency on the Win 10 SDK v10.0.17763.0.
-  rtc_enable_win_wgc = is_win
+  rtc_enable_win_wgc = false
 
   # Includes the dav1d decoder in the internal decoder factory when set to true.
   rtc_include_dav1d_in_internal_decoder_factory = true
@@ -256,7 +263,7 @@ declare_args() {
   rtc_build_json = !build_with_mozilla
   rtc_build_libsrtp = !build_with_mozilla
   rtc_build_libvpx = !build_with_mozilla
-  rtc_libvpx_build_vp9 = !build_with_mozilla
+  rtc_libvpx_build_vp9 = true
   rtc_build_opus = !build_with_mozilla
   rtc_build_ssl = !build_with_mozilla
 
@@ -275,7 +282,7 @@ declare_args() {
 
   # Chromium uses its own IO handling, so the internal ADM is only built for
   # standalone WebRTC.
-  rtc_include_internal_audio_device = !build_with_chromium
+  rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla
 
   # Set this to true to enable the avx2 support in webrtc.
   # TODO: Make sure that AVX2 works also for non-clang compilers.
@@ -319,6 +326,9 @@ declare_args() {
   rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac)
 }
 
+# Enable liboam only on non-mozilla builds.
+enable_libaom = !build_with_mozilla
+
 # Make it possible to provide custom locations for some libraries (move these
 # up into declare_args should we need to actually use them for the GN build).
 rtc_libvpx_dir = "//third_party/libvpx"
@@ -1114,7 +1124,7 @@ if (is_mac || is_ios) {
   }
 }
 
-if (is_android) {
+if (is_android && !build_with_mozilla) {
   template("rtc_android_library") {
     android_library(target_name) {
       forward_variables_from(invoker,
-- 
2.34.1