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/examples/BUILD.gn b/examples/BUILD.gn
index 4f76cad185..bb61efc235 100644
--- a/examples/BUILD.gn
+++ b/examples/BUILD.gn
@@ -197,7 +197,7 @@ 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" ]
     }
   }
 
@@ -266,7 +266,7 @@ 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",
@@ -493,7 +493,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
       deps = [
         ":apprtc_common",
         ":apprtc_signaling",
-        "../sdk:mac_framework_objc+link",
+        "../sdk:mac_framework_objc_static",
       ]
     }
 
@@ -516,8 +516,8 @@ 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_device/audio_device_buffer.cc b/modules/audio_device/audio_device_buffer.cc
index 336846ed11..5d0b280b36 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(), 1LL);
   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/utility/BUILD.gn b/modules/utility/BUILD.gn
index 4e85129e43..4bad1f728b 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 bca50d5f92..495d6c8143 100644
--- a/modules/video_coding/BUILD.gn
+++ b/modules/video_coding/BUILD.gn
@@ -345,10 +345,12 @@ 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" ]
+    }
     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 392ccaf563..526e394d16 100644
--- a/modules/video_coding/codecs/h264/h264_color_space.h
+++ b/modules/video_coding/codecs/h264/h264_color_space.h
@@ -17,13 +17,13 @@
 #ifdef WEBRTC_USE_H264
 
 #if defined(WEBRTC_WIN) && !defined(__clang__)
-#error "See: bugs.webrtc.org/9213#c13."
+//#error "See: bugs.webrtc.org/9213#c13."
 #endif
 
 #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 3c038e6425..187e6e007f 100644
--- a/modules/video_coding/codecs/h264/h264_decoder_impl.h
+++ b/modules/video_coding/codecs/h264/h264_decoder_impl.h
@@ -18,7 +18,7 @@
 #ifdef WEBRTC_USE_H264
 
 #if defined(WEBRTC_WIN) && !defined(__clang__)
-#error "See: bugs.webrtc.org/9213#c13."
+//#error "See: bugs.webrtc.org/9213#c13."
 #endif
 
 #include <memory>
@@ -40,7 +40,7 @@
 // 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/codecs/h264/h264_encoder_impl.h b/modules/video_coding/codecs/h264/h264_encoder_impl.h
index ba996366a3..1b9129f54c 100644
--- a/modules/video_coding/codecs/h264/h264_encoder_impl.h
+++ b/modules/video_coding/codecs/h264/h264_encoder_impl.h
@@ -18,7 +18,7 @@
 #ifdef WEBRTC_USE_H264
 
 #if defined(WEBRTC_WIN) && !defined(__clang__)
-#error "See: bugs.webrtc.org/9213#c13."
+//#error "See: bugs.webrtc.org/9213#c13."
 #endif
 
 #include <memory>
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/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 0805a5c549..c75732f766 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -24,6 +24,19 @@ 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_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 (is_win && !is_clang) {
+      ldflags = [ "/NODEFAULTLIB:LIBCMT" ]
+    }
   }
 }
 
@@ -1305,7 +1318,7 @@ 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 13793fb06b..40ee438e50 100644
--- a/sdk/BUILD.gn
+++ b/sdk/BUILD.gn
@@ -1386,6 +1386,126 @@ 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",
+          ":peerconnectionfactory_base_objc",
+          ":ui_objc",
+          ":videocapture_objc",
+          ":videocodec_objc",
+          ":videotoolbox_objc",
+          "../rtc_base:rtc_base_approved",
+        ]
+        if (rtc_use_metal_rendering) {
+          deps += [ ":metal_objc" ]
+        }
+        if (!build_with_chromium) {
+          deps += [
+            ":callback_logger_objc",
+            ":file_logger_objc",
+          ]
+        }
+
+        libs = [
+          "AVFoundation.framework",
+          "CoreGraphics.framework",
+          "CoreMedia.framework",
+          "OpenGL.framework",
+        ]
+
+        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/test/BUILD.gn b/test/BUILD.gn
index 464f44f60a..ca539baf5a 100644
--- a/test/BUILD.gn
+++ b/test/BUILD.gn
@@ -749,7 +749,7 @@ 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 4b01479c9b..410e450b74 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 9aef78a35d..258782e3f0 100644
--- a/test/pc/e2e/BUILD.gn
+++ b/test/pc/e2e/BUILD.gn
@@ -41,7 +41,7 @@ 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" ]
@@ -61,7 +61,7 @@ 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/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 1878f3216c..9c50cddfe1 100644
--- a/video/BUILD.gn
+++ b/video/BUILD.gn
@@ -240,7 +240,7 @@ 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 64a170c23f..1acd203a9b 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -35,6 +35,18 @@ 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 = []
+}
+
 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.
@@ -85,6 +97,9 @@ declare_args() {
   # library that comes with WebRTC (i.e. rtc_build_ssl == 0).
   rtc_ssl_root = ""
 
+  # Used to specify an external OpenSSL libs.
+  rtc_ssl_libs = []
+
   # Selects fixed-point code where possible.
   rtc_prefer_fixed_point = false
 
@@ -150,7 +165,7 @@ 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.
@@ -614,7 +629,8 @@ 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"