patches/webrtc_src.diff

diff --git a/BUILD.gn b/BUILD.gn
index 5fe6f849f3..719626d516 100644
--- a/BUILD.gn
+++ b/BUILD.gn
@@ -335,7 +335,7 @@ config("common_config") {
         "-Wnon-virtual-dtor",
 
         # This is enabled for clang; enable for gcc as well.
-        "-Woverloaded-virtual",
+        # "-Woverloaded-virtual", # a lot of such errors when compiling with g++.
       ]
     }
 
@@ -445,9 +445,9 @@ if (!build_with_chromium) {
       ":webrtc_lib_link_test",
     ]
 
-    if (build_with_owt) {
-      visibility += [ "//talk/owt" ]
-    }
+    # if (build_with_owt) {
+    #   visibility += [ "//talk/owt" ]
+    # }
     sources = []
     complete_static_lib = true
     suppressed_configs += [ "//build/config/compiler:thin_archive" ]
diff --git a/api/test/video/BUILD.gn b/api/test/video/BUILD.gn
index 4ebb0c9167..a4a0d7b887 100644
--- a/api/test/video/BUILD.gn
+++ b/api/test/video/BUILD.gn
@@ -8,7 +8,7 @@
 
 import("../../../webrtc.gni")
 
-rtc_library("function_video_factory") {
+rtc_source_set("function_video_factory") {
   visibility = [ "*" ]
   testonly = true
   public = [
diff --git a/audio/utility/channel_mixer.cc b/audio/utility/channel_mixer.cc
index 8867a3eed4..4bd36622b3 100644
--- a/audio/utility/channel_mixer.cc
+++ b/audio/utility/channel_mixer.cc
@@ -10,6 +10,8 @@
 
 #include "audio/utility/channel_mixer.h"
 
+#include <cstring> // fix build with -std=gnu++17
+
 #include "audio/utility/channel_mixing_matrix.h"
 #include "rtc_base/checks.h"
 #include "rtc_base/logging.h"
diff --git a/build_overrides/build.gni b/build_overrides/build.gni
index 6025cddd04..2565ba0ec4 100644
--- a/build_overrides/build.gni
+++ b/build_overrides/build.gni
@@ -16,6 +16,8 @@ linux_use_bundled_binutils_override = true
 # only needed to support both WebRTC standalone and Chromium builds.
 build_with_chromium = false
 
+build_with_owt = false
+
 # Use our own suppressions files.
 asan_suppressions_file = "//build/sanitizers/asan_suppressions.cc"
 lsan_suppressions_file = "//tools_webrtc/sanitizers/lsan_suppressions_webrtc.cc"
@@ -35,7 +37,7 @@ lint_suppressions_file = "//tools_webrtc/android/suppressions.xml"
 # so we just ignore that assert. See https://crbug.com/648948 for more info.
 ignore_elf32_limitations = true
 
-if (is_win || is_ios || is_android) {
+if (is_win || is_ios || is_android || is_mac) {
   rtc_use_h265 = true
 } else {
   rtc_use_h265 = false
diff --git a/examples/BUILD.gn b/examples/BUILD.gn
index 4fd2d702aa..217f52056c 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -232,7 +232,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       # iOS must use WebRTC.framework which is dynamically linked.
       deps = [ "../sdk:framework_objc+link" ]
     } else {
-      deps = [ "../sdk:mac_framework_objc+link" ]
+      deps = [ "../sdk:mac_framework_objc_static" ]
     }
   }
 
@@ -301,7 +301,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
         "../sdk:ios_framework_bundle",
       ]
     } else {
-      deps += [ "../sdk:mac_framework_objc+link" ]
+      deps += [ "../sdk:mac_framework_objc_static" ]
     }
     libs = [
       "CoreMedia.framework",
@@ -528,7 +528,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       deps = [
         ":apprtc_common",
         ":apprtc_signaling",
-        "../sdk:mac_framework_objc+link",
+        "../sdk:mac_framework_objc_static",
       ]
     }
 
@@ -551,8 +551,8 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
 
       deps = [
         ":AppRTCMobile_lib",
-        "../sdk:mac_framework_bundle",
-        "../sdk:mac_framework_objc+link",
+        # "../sdk:mac_framework_bundle",
+        "../sdk:mac_framework_objc_static",
       ]
     }
   }
diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn
index ceee0c0f07..8c4f00dbb5 100644
--- a/modules/audio_coding/BUILD.gn
+++ b/modules/audio_coding/BUILD.gn
@@ -763,6 +763,8 @@ rtc_library("webrtc_opus") {
     public_deps += [ rtc_opus_dir ]  # no-presubmit-check TODO(webrtc:8603)
   } else if (build_with_mozilla) {
     include_dirs = [ "/media/libopus/include" ]
+  } else {
+    public_configs = [ "../../rtc_base:external_opus_library" ]
   }
 }
 
@@ -801,6 +803,8 @@ rtc_library("webrtc_multiopus") {
     public_deps += [ rtc_opus_dir ]
   } else if (build_with_mozilla) {
     include_dirs = [ "/media/libopus/include" ]
+  } else {
+    public_configs = [ "../../rtc_base:external_opus_library" ]
   }
 }
 
@@ -818,6 +822,8 @@ rtc_library("webrtc_opus_wrapper") {
     public_deps = [ rtc_opus_dir ]  # no-presubmit-check TODO(webrtc:8603)
   } else if (build_with_mozilla) {
     include_dirs = [ getenv("DIST") + "/include/opus" ]
+  } else {
+    public_configs = [ "../../rtc_base:external_opus_library" ]
   }
 
   deps = [
diff --git a/modules/audio_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 336846ed11..f0c388b453 100644
--- a/modules/audio_device/audio_device_buffer.cc
+++ b/modules/audio_device/audio_device_buffer.cc
@@ -456,7 +456,7 @@ void AudioDeviceBuffer::LogStats(LogState state) {
   }
   last_stats_ = stats;
 
-  int64_t time_to_wait_ms = next_callback_time - rtc::TimeMillis();
+  int64_t time_to_wait_ms = std::max(next_callback_time - rtc::TimeMillis(), int64_t(1));
   RTC_DCHECK_GT(time_to_wait_ms, 0) << "Invalid timer interval";
 
   // Keep posting new (delayed) tasks until state is changed to kLogStop.
diff --git a/modules/audio_processing/aec3/reverb_model_estimator.h b/modules/audio_processing/aec3/reverb_model_estimator.h
index 3b9971abae..efc291222e 100644
--- a/modules/audio_processing/aec3/reverb_model_estimator.h
+++ b/modules/audio_processing/aec3/reverb_model_estimator.h
@@ -13,6 +13,7 @@
 
 #include <array>
 #include <vector>
+#include <memory> // fix build with -std=gnu++17
 
 #include "absl/types/optional.h"
 #include "api/array_view.h"
diff --git a/modules/utility/BUILD.gn b/modules/utility/BUILD.gn
index 8710ed4729..2a8fa4e476 100644
--- a/modules/utility/BUILD.gn
+++ b/modules/utility/BUILD.gn
@@ -40,7 +40,7 @@ rtc_library("utility") {
   ]
 }
 
-rtc_library("mock_process_thread") {
+rtc_source_set("mock_process_thread") {
   testonly = true
   visibility = [ "*" ]
   sources = [ "include/mock/mock_process_thread.h" ]
diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
index 2f9d257e4a..65130a87ad 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -370,10 +370,13 @@ rtc_library("webrtc_h264") {
   ]
 
   if (rtc_use_h264) {
-    deps += [
-      "//third_party/ffmpeg",
-      "//third_party/openh264:encoder",
-    ]
+    deps += [ "//third_party/openh264:encoder" ]
+    if (rtc_build_ffmpeg) {
+      deps += [ "//third_party/ffmpeg" ]
+    } else {
+      configs += [ "../../rtc_base:external_ffmpeg_library" ]
+      all_dependent_configs = [ "../../rtc_base:external_ffmpeg_linking" ]
+    }
     if (!build_with_mozilla) {
       deps += [ "../../media:rtc_media_base" ]
     }
diff --git a/modules/video_coding/codecs/h264/h264_color_space.h b/modules/video_coding/codecs/h264/h264_color_space.h
index aec76efbac..918ea26f8b 100644
--- a/modules/video_coding/codecs/h264/h264_color_space.h
+++ b/modules/video_coding/codecs/h264/h264_color_space.h
@@ -19,7 +19,7 @@
 #include "api/video/color_space.h"
 
 extern "C" {
-#include "third_party/ffmpeg/libavcodec/avcodec.h"
+#include <libavcodec/avcodec.h>
 }  // extern "C"
 
 namespace webrtc {
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.cc b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
index 6725a3b7c7..a7e837ca01 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.cc
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.cc
@@ -21,9 +21,9 @@
 #include <memory>
 
 extern "C" {
-#include "third_party/ffmpeg/libavcodec/avcodec.h"
-#include "third_party/ffmpeg/libavformat/avformat.h"
-#include "third_party/ffmpeg/libavutil/imgutils.h"
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libavutil/imgutils.h>
 }  // extern "C"
 
 #include "api/video/color_space.h"
diff --git a/modules/video_coding/codecs/h264/h264_decoder_impl.h b/modules/video_coding/codecs/h264/h264_decoder_impl.h
index 80892de63b..70f34fb039 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.h
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.h
@@ -36,7 +36,7 @@
 // passed to ffmpeg.
 
 extern "C" {
-#include "third_party/ffmpeg/libavcodec/avcodec.h"
+#include <libavcodec/avcodec.h>
 }  // extern "C"
 
 #include "common_video/h264/h264_bitstream_parser.h"
diff --git a/modules/video_coding/utility/ivf_file_reader.cc b/modules/video_coding/utility/ivf_file_reader.cc
index 9667bb7cec..6fe4e88165 100644
--- a/modules/video_coding/utility/ivf_file_reader.cc
+++ b/modules/video_coding/utility/ivf_file_reader.cc
@@ -12,6 +12,7 @@
 
 #include <string>
 #include <vector>
+#include <cstring> // fix build with -std=gnu++17
 
 #include "api/video_codecs/video_codec.h"
 #include "modules/rtp_rtcp/source/byte_io.h"
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index ae49deb264..af192415f7 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -118,7 +118,7 @@ rtc_library("rtc_p2p") {
 }
 
 if (rtc_include_tests) {
-  rtc_library("fake_ice_transport") {
+  rtc_source_set("fake_ice_transport") {
     testonly = true
     visibility = [ "*" ]
     sources = [ "base/fake_ice_transport.h" ]
@@ -132,7 +132,7 @@ if (rtc_include_tests) {
     ]
   }
 
-  rtc_library("fake_port_allocator") {
+  rtc_source_set("fake_port_allocator") {
     testonly = true
     visibility = [ "*" ]
     sources = [ "base/fake_port_allocator.h" ]
diff --git a/p2p/base/basic_ice_controller.cc b/p2p/base/basic_ice_controller.cc
index aa20025b2c..506bf5cc9b 100644
--- a/p2p/base/basic_ice_controller.cc
+++ b/p2p/base/basic_ice_controller.cc
@@ -195,7 +195,7 @@ const Connection* BasicIceController::FindNextPingableConnection() {
         if (conn1 == conn2) {
           return false;
         }
-        return MorePingable(conn1, conn2) == conn2;
+        return MorePingable(std::min(conn1, conn2), std::max(conn1, conn2)) == conn2;
       });
   if (iter != pingable_connections.end()) {
     return *iter;
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index d078482d96..a1bb711199 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -24,6 +24,35 @@ if (!rtc_build_ssl) {
     assert(rtc_ssl_root != "",
            "You must specify rtc_ssl_root when rtc_build_ssl==0.")
     include_dirs = [ rtc_ssl_root ]
+    libs = rtc_ssl_libs
+  }
+}
+
+if (!rtc_build_ffmpeg) {
+  config("external_ffmpeg_linking") {
+    if (is_win && !is_clang) {
+      ldflags = [
+        "/NODEFAULTLIB:LIBCMT",
+
+        # libavcodec.a(me_cmp.o) : warning LNK4078: multiple '.rodata' sections found with different attributes (60500020)
+        "/ignore:4078",
+      ]
+    }
+  }
+  config("external_ffmpeg_library") {
+    assert(rtc_ffmpeg_root != "",
+           "You must specify rtc_ffmpeg_root when rtc_build_ffmpeg==0.")
+    include_dirs = [ rtc_ffmpeg_root ]
+    libs = rtc_ffmpeg_libs
+  }
+}
+
+if (!rtc_build_opus) {
+  config("external_opus_library") {
+    assert(rtc_opus_root != "",
+           "You must specify rtc_opus_root when rtc_build_opus==0.")
+    include_dirs = [ rtc_opus_root ]
+    libs = rtc_opus_libs
   }
 }
 
@@ -1326,7 +1355,7 @@ if (rtc_include_tests) {
         "win32_window_unittest.cc",
       ]
     }
-    if (is_posix || is_fuchsia) {
+    if ((is_posix || is_fuchsia) && rtc_build_ssl) {
       sources += [
         "openssl_adapter_unittest.cc",
         "openssl_session_cache_unittest.cc",
diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
index 7588baa4ea..81f5ccfe5e 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -23,6 +23,8 @@ group("sdk") {
     }
     if (is_ios) {
       public_deps += [ ":framework_objc" ]
+    } else if (is_mac) {
+      public_deps += [ ":mac_framework_objc_static" ]
     }
   }
 }
@@ -1399,6 +1401,107 @@ if (is_ios || is_mac) {
     }
 
     if (is_mac) {
+      rtc_source_set("mac_framework_objc_static") {
+        visibility = [ "*" ]
+        allow_poison = [ "software_video_codecs", "audio_codecs", "default_task_queue" ]
+        sources = [
+          "objc/api/peerconnection/RTCAudioSource.h",
+          "objc/api/peerconnection/RTCAudioTrack.h",
+          "objc/api/peerconnection/RTCCertificate.h",
+          "objc/api/peerconnection/RTCConfiguration.h",
+          "objc/api/peerconnection/RTCCryptoOptions.h",
+          "objc/api/peerconnection/RTCDataChannel.h",
+          "objc/api/peerconnection/RTCDataChannelConfiguration.h",
+          "objc/api/peerconnection/RTCDtmfSender.h",
+          "objc/api/peerconnection/RTCFieldTrials.h",
+          "objc/api/peerconnection/RTCIceCandidate.h",
+          "objc/api/peerconnection/RTCIceServer.h",
+          "objc/api/peerconnection/RTCLegacyStatsReport.h",
+          "objc/api/peerconnection/RTCMediaConstraints.h",
+          "objc/api/peerconnection/RTCMediaSource.h",
+          "objc/api/peerconnection/RTCMediaStream.h",
+          "objc/api/peerconnection/RTCMediaStreamTrack.h",
+          "objc/api/peerconnection/RTCMetrics.h",
+          "objc/api/peerconnection/RTCMetricsSampleInfo.h",
+          "objc/api/peerconnection/RTCPeerConnection.h",
+          "objc/api/peerconnection/RTCPeerConnectionFactory.h",
+          "objc/api/peerconnection/RTCPeerConnectionFactoryOptions.h",
+          "objc/api/peerconnection/RTCRtcpParameters.h",
+          "objc/api/peerconnection/RTCRtpCodecParameters.h",
+          "objc/api/peerconnection/RTCRtpEncodingParameters.h",
+          "objc/api/peerconnection/RTCRtpHeaderExtension.h",
+          "objc/api/peerconnection/RTCRtpParameters.h",
+          "objc/api/peerconnection/RTCRtpReceiver.h",
+          "objc/api/peerconnection/RTCRtpSender.h",
+          "objc/api/peerconnection/RTCRtpTransceiver.h",
+          "objc/api/peerconnection/RTCSSLAdapter.h",
+          "objc/api/peerconnection/RTCSessionDescription.h",
+          "objc/api/peerconnection/RTCTracing.h",
+          "objc/api/peerconnection/RTCVideoSource.h",
+          "objc/api/peerconnection/RTCVideoTrack.h",
+          "objc/api/video_codec/RTCVideoDecoderVP8.h",
+          "objc/api/video_codec/RTCVideoDecoderVP9.h",
+          "objc/api/video_codec/RTCVideoEncoderVP8.h",
+          "objc/api/video_codec/RTCVideoEncoderVP9.h",
+          "objc/api/video_frame_buffer/RTCNativeI420Buffer.h",
+          "objc/api/video_frame_buffer/RTCNativeMutableI420Buffer.h",
+          "objc/base/RTCCodecSpecificInfo.h",
+          "objc/base/RTCEncodedImage.h",
+          "objc/base/RTCI420Buffer.h",
+          "objc/base/RTCLogging.h",
+          "objc/base/RTCMacros.h",
+          "objc/base/RTCMutableI420Buffer.h",
+          "objc/base/RTCMutableYUVPlanarBuffer.h",
+          "objc/base/RTCRtpFragmentationHeader.h",
+          "objc/base/RTCVideoCapturer.h",
+          "objc/base/RTCVideoCodecInfo.h",
+          "objc/base/RTCVideoDecoder.h",
+          "objc/base/RTCVideoDecoderFactory.h",
+          "objc/base/RTCVideoEncoder.h",
+          "objc/base/RTCVideoEncoderFactory.h",
+          "objc/base/RTCVideoEncoderQpThresholds.h",
+          "objc/base/RTCVideoEncoderSettings.h",
+          "objc/base/RTCVideoFrame.h",
+          "objc/base/RTCVideoFrameBuffer.h",
+          "objc/base/RTCVideoRenderer.h",
+          "objc/base/RTCYUVPlanarBuffer.h",
+          "objc/components/capturer/RTCCameraVideoCapturer.h",
+          "objc/components/capturer/RTCFileVideoCapturer.h",
+          "objc/components/renderer/metal/RTCMTLNSVideoView.h",
+          "objc/components/renderer/opengl/RTCNSGLVideoView.h",
+          "objc/components/renderer/opengl/RTCVideoViewShading.h",
+          "objc/components/video_codec/RTCCodecSpecificInfoH264.h",
+          "objc/components/video_codec/RTCDefaultVideoDecoderFactory.h",
+          "objc/components/video_codec/RTCDefaultVideoEncoderFactory.h",
+          "objc/components/video_codec/RTCH264ProfileLevelId.h",
+          "objc/components/video_codec/RTCVideoDecoderFactoryH264.h",
+          "objc/components/video_codec/RTCVideoDecoderH264.h",
+          "objc/components/video_codec/RTCVideoEncoderFactoryH264.h",
+          "objc/components/video_codec/RTCVideoEncoderH264.h",
+          "objc/components/video_frame_buffer/RTCCVPixelBuffer.h",
+          "objc/helpers/RTCDispatcher.h",
+        ]
+        if (!build_with_chromium) {
+          sources += [
+            "objc/api/logging/RTCCallbackLogger.h",
+            "objc/api/peerconnection/RTCFileLogger.h",
+          ]
+        }
+
+        deps = [
+          ":base_objc",
+          ":default_codec_factory_objc",
+          ":native_api",
+          ":native_video",
+          ":videocapture_objc",
+          ":videocodec_objc",
+          ":videotoolbox_objc",
+        ]
+
+        configs += [ "..:common_objc" ]
+
+        public_configs = [ ":common_config_objc" ]
+      }
       mac_framework_bundle_with_umbrella_header("mac_framework_objc") {
         info_plist = "objc/Info.plist"
         output_name = "WebRTC"
diff --git a/sdk/objc/components/video_codec/nalu_rewriter.cc b/sdk/objc/components/video_codec/nalu_rewriter.cc
index ddf7346051..05d15f5d71 100644
--- a/sdk/objc/components/video_codec/nalu_rewriter.cc
+++ b/sdk/objc/components/video_codec/nalu_rewriter.cc
@@ -269,8 +269,14 @@ bool H265CMSampleBufferToAnnexBBuffer(
   // Get parameter set information.
   int nalu_header_size = 0;
   size_t param_set_count = 0;
-  OSStatus status = CMVideoFormatDescriptionGetHEVCParameterSetAtIndex(
+  OSStatus status = noErr;
+  if (__builtin_available(macOS 10.13, *)) {
+    status = CMVideoFormatDescriptionGetHEVCParameterSetAtIndex(
       description, 0, nullptr, nullptr, &param_set_count, &nalu_header_size);
+  } else {
+    RTC_LOG(LS_ERROR) << "H265 is unavailable on this system.";
+    return false;
+  }
   if (status != noErr) {
     RTC_LOG(LS_ERROR) << "Failed to get parameter set.";
     return false;
@@ -290,8 +296,13 @@ bool H265CMSampleBufferToAnnexBBuffer(
     size_t param_set_size = 0;
     const uint8_t* param_set = nullptr;
     for (size_t i = 0; i < param_set_count; ++i) {
-      status = CMVideoFormatDescriptionGetHEVCParameterSetAtIndex(
+      if (__builtin_available(macOS 10.13, *)) {
+        status = CMVideoFormatDescriptionGetHEVCParameterSetAtIndex(
           description, i, &param_set, &param_set_size, nullptr, nullptr);
+      } else {
+        RTC_LOG(LS_ERROR) << "H265 is unavailable on this system.";
+        return false;
+      }
       if (status != noErr) {
         RTC_LOG(LS_ERROR) << "Failed to get parameter set.";
         return false;
@@ -527,9 +538,15 @@ CMVideoFormatDescriptionRef CreateH265VideoFormatDescription(
 
   // Parse the SPS and PPS into a CMVideoFormatDescription.
   CMVideoFormatDescriptionRef description = nullptr;
-  OSStatus status = CMVideoFormatDescriptionCreateFromHEVCParameterSets(
+  OSStatus status = noErr;
+  if (__builtin_available(macOS 10.13, *)) {
+    status = CMVideoFormatDescriptionCreateFromHEVCParameterSets(
       kCFAllocatorDefault, 3, param_set_ptrs, param_set_sizes, 4, nullptr,
       &description);
+  } else {
+    RTC_LOG(LS_ERROR) << "H265 is unavailable on this system.";
+    return nullptr;
+  }
   if (status != noErr) {
     RTC_LOG(LS_ERROR) << "Failed to create video format description.";
     return nullptr;
diff --git a/test/BUILD.gn b/test/BUILD.gn
index 2e22476c80..7d649577b9 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -768,7 +768,7 @@ rtc_library("encoder_settings") {
   ]
 }
 
-rtc_library("rtc_expect_death") {
+rtc_source_set("rtc_expect_death") {
   testonly = true
   sources = [ "testsupport/rtc_expect_death.h" ]
   deps = [ ":test_support" ]
diff --git a/test/network/BUILD.gn b/test/network/BUILD.gn
index fc56001479..d3dd0e80fd 100644
--- a/test/network/BUILD.gn
+++ b/test/network/BUILD.gn
@@ -143,7 +143,7 @@ rtc_library("feedback_generator_unittest") {
   ]
 }
 
-rtc_library("network_emulation_unittests") {
+rtc_source_set("network_emulation_unittests") {
   testonly = true
   deps = [
     ":cross_traffic_unittest",
diff --git a/test/pc/e2e/BUILD.gn b/test/pc/e2e/BUILD.gn
index 8a5607a1e1..8b28e64dc8 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -42,7 +42,7 @@ if (rtc_include_tests) {
   }
 }
 
-rtc_library("peer_connection_quality_test_params") {
+rtc_source_set("peer_connection_quality_test_params") {
   visibility = [ "*" ]
   testonly = true
   sources = [ "peer_connection_quality_test_params.h" ]
@@ -62,7 +62,7 @@ rtc_library("peer_connection_quality_test_params") {
   ]
 }
 
-rtc_library("encoded_image_data_injector_api") {
+rtc_source_set("encoded_image_data_injector_api") {
   visibility = [ "*" ]
   testonly = true
   sources = [ "analyzer/video/encoded_image_data_injector.h" ]
diff --git a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
index b8f1740e46..4de32d7559 100644
--- a/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
+++ b/test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
@@ -10,6 +10,8 @@
 
 #include "test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.h"
 
+#include <cstring> // fix build with -std=gnu++17
+
 #include "api/stats_types.h"
 #include "rtc_base/logging.h"
 
diff --git a/test/pc/e2e/analyzer/video/default_encoded_image_data_injector.cc b/test/pc/e2e/analyzer/video/default_encoded_image_data_injector.cc
index 2634e6eea4..11d7f4b00e 100644
--- a/test/pc/e2e/analyzer/video/default_encoded_image_data_injector.cc
+++ b/test/pc/e2e/analyzer/video/default_encoded_image_data_injector.cc
@@ -12,6 +12,7 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <cstring> // fix build with -std=gnu++17
 
 #include "absl/memory/memory.h"
 #include "api/video/encoded_image.h"
diff --git a/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc b/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
index ec0d26b780..f51d57cfc1 100644
--- a/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
+++ b/test/pc/e2e/analyzer/video/single_process_encoded_image_data_injector.cc
@@ -12,6 +12,7 @@
 
 #include <algorithm>
 #include <cstddef>
+#include <cstring> // fix build with -std=gnu++17
 
 #include "absl/memory/memory.h"
 #include "api/video/encoded_image.h"
diff --git a/test/testsupport/jpeg_frame_writer.cc b/test/testsupport/jpeg_frame_writer.cc
index 8bf1ee4630..6817ee59c2 100644
--- a/test/testsupport/jpeg_frame_writer.cc
+++ b/test/testsupport/jpeg_frame_writer.cc
@@ -65,7 +65,7 @@ bool JpegFrameWriter::WriteFrame(const VideoFrame& input_frame, int quality) {
   cinfo.image_width = input_frame.width();
   cinfo.image_height = input_frame.height();
   cinfo.input_components = kColorPlanes;
-  cinfo.in_color_space = JCS_EXT_BGR;
+  cinfo.in_color_space = JCS_RGB; // JCS_EXT_BGR; <- this only in _turbo
   jpeg_set_defaults(&cinfo);
   jpeg_set_quality(&cinfo, quality, TRUE);
 
diff --git a/video/BUILD.gn b/video/BUILD.gn
index 14109c3494..21d768bb01 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -242,7 +242,7 @@ rtc_library("video_stream_encoder_impl") {
 }
 
 if (rtc_include_tests) {
-  rtc_library("video_mocks") {
+  rtc_source_set("video_mocks") {
     testonly = true
     sources = [ "test/mock_video_stream_encoder.h" ]
     deps = [
diff --git a/webrtc.gni b/webrtc.gni
index 96b654a28c..636d1ce00a 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -12,7 +12,7 @@ import("//build/config/sanitizers/sanitizers.gni")
 import("//build/config/sysroot.gni")
 import("//build/config/ui.gni")
 import("//build_overrides/build.gni")
-import("//build_overrides/ssl/ssl.gni")
+#import("//build_overrides/ssl/ssl.gni")
 
 if (!build_with_chromium && is_component_build) {
   print("The Gn argument `is_component_build` is currently " +
@@ -36,6 +36,25 @@ if (is_mac) {
   import("//build/config/mac/rules.gni")
 }
 
+declare_args() {
+  # Allow providing external ffmpeg
+  rtc_build_ffmpeg = true
+
+  # Used to specify an external ffmpeg include path when not compiling the
+  # library that comes with WebRTC (i.e. rtc_build_ffmpeg == 0).
+  rtc_ffmpeg_root = ""
+
+  # Used to specify an external ffmpeg libs.
+  rtc_ffmpeg_libs = []
+
+  # Used to specify and external opus include path when not compiling the
+  # library that comes with WebRTC (i.e. rtc_build_opus == 0).
+  rtc_opus_root = ""
+
+  # Used to specify an external opus libs.
+  rtc_opus_libs = []
+}
+
 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 +103,10 @@ 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 = owt_openssl_header_root
+  rtc_ssl_root = "" # owt_openssl_header_root
+
+  # Used to specify an external OpenSSL libs.
+  rtc_ssl_libs = []
 
   # Selects fixed-point code where possible.
   rtc_prefer_fixed_point = false
@@ -151,7 +173,7 @@ declare_args() {
   # Enabling H264 when building with MSVC is currently not supported, see
   # bugs.webrtc.org/9213#c13 for more info.
   rtc_use_h264 =
-      proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
+      proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang && rtc_build_ffmpeg)
 
   # By default, use normal platform audio support or dummy audio, but don't
   # use file-based audio playout and record.
@@ -213,7 +235,7 @@ declare_args() {
   rtc_build_libvpx = !build_with_mozilla
   rtc_libvpx_build_vp9 = !build_with_mozilla
   rtc_build_opus = !build_with_mozilla
-  rtc_build_ssl = !owt_use_openssl
+  rtc_build_ssl = !build_with_mozilla # owt_use_openssl
   rtc_build_usrsctp = !build_with_mozilla
 
   # Enable libevent task queues on platforms that support it.
@@ -619,7 +641,8 @@ template("rtc_static_library") {
 # For the same reason, testonly targets will always be expanded to
 # source_set in order to be sure that tests are present in the test binary.
 template("rtc_library") {
-  if (is_component_build || (defined(invoker.testonly) && invoker.testonly)) {
+  # if (is_component_build || (defined(invoker.testonly) && invoker.testonly)) {
+  if (is_component_build) {
     target_type = "source_set"
   } else {
     target_type = "static_library"