commit 2eb400c53144947874748fee1b1dc4588678440a parent 22b3401880c7d0a7f91dee5e3177fa7a1f8d4eea Author: Dan Baker <dbaker@mozilla.com> Date: Thu, 23 Oct 2025 15:54:38 -0600 Bug 1995393 - Vendor libwebrtc from ffd474e5fa Upstream commit: https://webrtc.googlesource.com/src/+/ffd474e5faee0954ea0ab9d90c81c02fb89ad10a Delete deprecated build flag for link-time injection of field trials. Such feature was removed, flag has no effect. Bug: webrtc:42220378 Change-Id: I0978df57dec9f511b63534030265e8826b29a278 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/404620 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#45351} Diffstat:
13 files changed, 36 insertions(+), 40 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor @@ -1,4 +1,4 @@ # ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc -libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-23T21:52:03.674407+00:00. +libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-23T21:54:26.795938+00:00. # base of lastest vendoring -645334dd76 +ffd474e5fa diff --git a/third_party/libwebrtc/moz-patch-stack/s0001.patch b/third_party/libwebrtc/moz-patch-stack/s0001.patch @@ -1501,10 +1501,10 @@ index b7933130bd..428fc9615d 100644 vcm_ = nullptr; } diff --git a/webrtc.gni b/webrtc.gni -index 3cf3cc5c06..1ba954a5fb 100644 +index fc856001de..daf74c1d45 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -114,7 +114,7 @@ declare_args() { +@@ -110,7 +110,7 @@ declare_args() { # Selects whether debug dumps for the audio processing module # should be generated. diff --git a/third_party/libwebrtc/moz-patch-stack/s0027.patch b/third_party/libwebrtc/moz-patch-stack/s0027.patch @@ -1280,7 +1280,7 @@ index e3122cbc18..b58bef6ad6 100644 "../api:sequence_checker", "../api/adaptation:resource_adaptation_api", diff --git a/webrtc.gni b/webrtc.gni -index 1ba954a5fb..2032395308 100644 +index daf74c1d45..ba5854bd23 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -35,6 +35,11 @@ if (is_mac) { @@ -1307,7 +1307,7 @@ index 1ba954a5fb..2032395308 100644 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. -@@ -83,7 +93,7 @@ declare_args() { +@@ -79,7 +89,7 @@ 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. @@ -1316,7 +1316,7 @@ index 1ba954a5fb..2032395308 100644 # Setting this to false will require the API user to pass in their own # SSLCertificateVerifier to verify the certificates presented from a -@@ -106,7 +116,7 @@ declare_args() { +@@ -102,7 +112,7 @@ 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). @@ -1325,7 +1325,7 @@ index 1ba954a5fb..2032395308 100644 # Enable when an external authentication mechanism is used for performing # packet authentication for RTP packets instead of libsrtp. -@@ -120,13 +130,13 @@ declare_args() { +@@ -116,13 +126,13 @@ declare_args() { rtc_exclude_audio_processing_module = false # Set this to false to skip building examples. @@ -1342,7 +1342,7 @@ index 1ba954a5fb..2032395308 100644 # Set this to use PipeWire on the Wayland display server. # By default it's only enabled on desktop Linux (excludes ChromeOS) and -@@ -137,9 +147,6 @@ declare_args() { +@@ -133,9 +143,6 @@ declare_args() { # Set this to link PipeWire and required libraries directly instead of using the dlopen. rtc_link_pipewire = false @@ -1352,7 +1352,7 @@ index 1ba954a5fb..2032395308 100644 # Experimental: enable use of Android AAudio which requires Android SDK 26 or above # and NDK r16 or above. rtc_enable_android_aaudio = false -@@ -281,7 +288,7 @@ declare_args() { +@@ -277,7 +284,7 @@ declare_args() { rtc_build_json = !build_with_mozilla rtc_build_libsrtp = !build_with_mozilla rtc_build_libvpx = !build_with_mozilla @@ -1361,7 +1361,7 @@ index 1ba954a5fb..2032395308 100644 rtc_build_opus = !build_with_mozilla rtc_build_ssl = !build_with_mozilla -@@ -290,7 +297,7 @@ declare_args() { +@@ -286,7 +293,7 @@ declare_args() { # Chromium uses its own IO handling, so the internal ADM is only built for # standalone WebRTC. @@ -1370,7 +1370,7 @@ index 1ba954a5fb..2032395308 100644 # Set this to true to enable the avx2 support in webrtc. # TODO: Make sure that AVX2 works also for non-clang compilers. -@@ -336,6 +343,9 @@ declare_args() { +@@ -332,6 +339,9 @@ declare_args() { rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac) } @@ -1380,7 +1380,7 @@ index 1ba954a5fb..2032395308 100644 # 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" -@@ -1208,7 +1218,7 @@ if (is_mac || is_ios) { +@@ -1204,7 +1214,7 @@ if (is_mac || is_ios) { } } diff --git a/third_party/libwebrtc/moz-patch-stack/s0034.patch b/third_party/libwebrtc/moz-patch-stack/s0034.patch @@ -276,10 +276,10 @@ index f45c664600..4c4366dad2 100644 config("x11_config") { if (rtc_use_x11_extensions) { diff --git a/webrtc.gni b/webrtc.gni -index 2032395308..c6f12efec0 100644 +index ba5854bd23..c35f08b426 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -160,13 +160,13 @@ declare_args() { +@@ -156,13 +156,13 @@ declare_args() { # Selects fixed-point code where possible. rtc_prefer_fixed_point = false diff --git a/third_party/libwebrtc/moz-patch-stack/s0045.patch b/third_party/libwebrtc/moz-patch-stack/s0045.patch @@ -74,10 +74,10 @@ index 20bf4afc44..697bedadb9 100644 #endif // defined(WEBRTC_POSIX) } diff --git a/webrtc.gni b/webrtc.gni -index c6f12efec0..840b752953 100644 +index c35f08b426..ad4460c554 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -353,7 +353,7 @@ rtc_opus_dir = "//third_party/opus" +@@ -349,7 +349,7 @@ rtc_opus_dir = "//third_party/opus" # Desktop capturer is supported only on Windows, OSX and Linux. rtc_desktop_capture_supported = diff --git a/third_party/libwebrtc/moz-patch-stack/s0069.patch b/third_party/libwebrtc/moz-patch-stack/s0069.patch @@ -10,10 +10,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/46fb51c90709be64c 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc.gni b/webrtc.gni -index 840b752953..93d4a8e55f 100644 +index ad4460c554..280b75076b 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -301,7 +301,7 @@ declare_args() { +@@ -297,7 +297,7 @@ declare_args() { # Set this to true to enable the avx2 support in webrtc. # TODO: Make sure that AVX2 works also for non-clang compilers. diff --git a/third_party/libwebrtc/moz-patch-stack/s0100.patch b/third_party/libwebrtc/moz-patch-stack/s0100.patch @@ -52,10 +52,10 @@ index ef3d60ee34..0d68cb4b29 100644 rtc_library("encoded_frame") { diff --git a/webrtc.gni b/webrtc.gni -index 93d4a8e55f..362ff142b5 100644 +index 280b75076b..420b8a1f89 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -490,8 +490,8 @@ all_poison_types = [ +@@ -486,8 +486,8 @@ all_poison_types = [ "software_video_codecs", ] diff --git a/third_party/libwebrtc/moz-patch-stack/s0101.patch b/third_party/libwebrtc/moz-patch-stack/s0101.patch @@ -10,10 +10,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/45b99d1ba95b46896 1 file changed, 71 insertions(+) diff --git a/webrtc.gni b/webrtc.gni -index 362ff142b5..ebb66f048c 100644 +index 420b8a1f89..c489e43642 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -694,6 +694,36 @@ template("rtc_source_set") { +@@ -690,6 +690,36 @@ template("rtc_source_set") { deps += [ "//third_party/abseil-cpp:absl" ] } } @@ -50,7 +50,7 @@ index 362ff142b5..ebb66f048c 100644 } } -@@ -928,6 +958,47 @@ template("rtc_library") { +@@ -924,6 +954,47 @@ template("rtc_library") { deps += [ "//third_party/abseil-cpp:absl" ] } } diff --git a/third_party/libwebrtc/moz-patch-stack/s0102.patch b/third_party/libwebrtc/moz-patch-stack/s0102.patch @@ -804,7 +804,7 @@ index ae891d22bd..4f2e3cc266 100644 } } diff --git a/webrtc.gni b/webrtc.gni -index ebb66f048c..4e6c39fdbb 100644 +index c489e43642..50566afb74 100644 --- a/webrtc.gni +++ b/webrtc.gni @@ -5,12 +5,12 @@ @@ -852,7 +852,7 @@ index ebb66f048c..4e6c39fdbb 100644 } if (build_with_chromium) { -@@ -376,8 +376,8 @@ rtc_common_configs = [ webrtc_root + ":common_config" ] +@@ -372,8 +372,8 @@ rtc_common_configs = [ webrtc_root + ":common_config" ] if (is_mac || is_ios) { if (filter_include(default_compiler_configs, @@ -863,7 +863,7 @@ index ebb66f048c..4e6c39fdbb 100644 } } -@@ -395,7 +395,7 @@ rtc_common_inherited_config = webrtc_root + ":common_inherited_config" +@@ -391,7 +391,7 @@ rtc_common_inherited_config = webrtc_root + ":common_inherited_config" # Common configs to remove or add in all rtc targets. rtc_remove_configs = [] if (!build_with_chromium && is_clang) { diff --git a/third_party/libwebrtc/moz-patch-stack/s0103.patch b/third_party/libwebrtc/moz-patch-stack/s0103.patch @@ -10,10 +10,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f18c05287ce831369 1 file changed, 12 insertions(+), 2 deletions(-) diff --git a/webrtc.gni b/webrtc.gni -index 4e6c39fdbb..bbb68a82d1 100644 +index 50566afb74..b199596ed1 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -539,7 +539,7 @@ template("rtc_test") { +@@ -535,7 +535,7 @@ template("rtc_test") { min_sdk_version = 23 target_sdk_version = 24 deps += [ @@ -22,7 +22,7 @@ index 4e6c39fdbb..bbb68a82d1 100644 webrtc_root + "sdk/android:native_test_jni_onload", webrtc_root + "sdk/android:base_java", webrtc_root + "test:native_test_java", -@@ -994,11 +994,21 @@ template("rtc_library") { +@@ -990,11 +990,21 @@ template("rtc_library") { modified_deps = [] foreach (dep, deps) { newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/") @@ -44,7 +44,7 @@ index 4e6c39fdbb..bbb68a82d1 100644 } } -@@ -1036,7 +1046,7 @@ template("rtc_executable") { +@@ -1032,7 +1042,7 @@ template("rtc_executable") { if (is_win) { deps += [ # Give executables the default manifest on Windows (a no-op elsewhere). diff --git a/third_party/libwebrtc/moz-patch-stack/s0107.patch b/third_party/libwebrtc/moz-patch-stack/s0107.patch @@ -147,10 +147,10 @@ index 9653fd651f..82b050a559 100644 rtc_library("ssl") { if (!build_with_mozilla) { diff --git a/webrtc.gni b/webrtc.gni -index bbb68a82d1..fddcdabae7 100644 +index b199596ed1..d36ef790d3 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -116,7 +116,7 @@ declare_args() { +@@ -112,7 +112,7 @@ 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). @@ -159,7 +159,7 @@ index bbb68a82d1..fddcdabae7 100644 # Enable when an external authentication mechanism is used for performing # packet authentication for RTP packets instead of libsrtp. -@@ -344,7 +344,7 @@ declare_args() { +@@ -340,7 +340,7 @@ declare_args() { } # Enable liboam only on non-mozilla builds. diff --git a/third_party/libwebrtc/moz-patch-stack/s0116.patch b/third_party/libwebrtc/moz-patch-stack/s0116.patch @@ -18,10 +18,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c8cf67108972d5cae 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/webrtc.gni b/webrtc.gni -index fddcdabae7..09e3f58634 100644 +index d36ef790d3..ff17358d2a 100644 --- a/webrtc.gni +++ b/webrtc.gni -@@ -136,7 +136,7 @@ declare_args() { +@@ -132,7 +132,7 @@ declare_args() { rtc_build_tools = false # Set this to false to skip building code that requires X11. diff --git a/third_party/libwebrtc/webrtc.gni b/third_party/libwebrtc/webrtc.gni @@ -79,10 +79,6 @@ declare_args() { # annotated symbols. rtc_enable_objc_symbol_export = rtc_enable_symbol_export - # Deprecated, this flag has no effect. - # TODO: bugs.webrtc.org/42220378 - Delete after August 15, 2025. - rtc_exclude_field_trial_default = false - # Setting this to true will define WEBRTC_EXCLUDE_METRICS_DEFAULT which # will tell the pre-processor to remove the default definition of symbols # needed to use metrics. In that case a new implementation needs to be