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patches/webrtc_src.diff

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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_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/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index d078482d96..6f0dfb1bc5 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -24,6 +24,26 @@ 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
}
}
@@ -1326,7 +1346,7 @@ 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..ef311d244c 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 +36,18 @@ 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.
@@ -84,7 +96,10 @@ 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 +166,7 @@ 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 +228,7 @@ 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 +634,8 @@ 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"
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