commit daa0543385fb69af5c3830b9f2e7414a724ccb82
parent b72bee702ed0f8dd225251404e870c0a4791a8ee
Author: Dan Baker <dbaker@mozilla.com>
Date: Wed, 22 Oct 2025 15:51:29 -0600
Bug 1995393 - Vendor libwebrtc from f6aade0ada
Upstream commit: https://webrtc.googlesource.com/src/+/f6aade0adad9ae1bc880c9246e2854a5276045c0
Separate assignment statements in echo_canceller3
Two variable assignments had been combined into one statement as they
were separated by commas, not semicolons. Based on the formatting of
surrounding code this looks unintentional.
(While not a logical bug, fixing as this caused a compilation issue in
one particular internal build configuration)
Bug: None
Change-Id: I076209010e012bc59617439523edba637d54bb10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/403186
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Joachim Reiersen <joachimr@meta.com>
Cr-Commit-Position: refs/heads/main@{#45318}
Diffstat:
33 files changed, 2062 insertions(+), 2086 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 2025-10-22T21:48:46.465357+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-10-22T21:51:10.236824+00:00.
# base of lastest vendoring
-85642b2e08
+f6aade0ada
diff --git a/third_party/libwebrtc/moz-patch-stack/s0100.patch b/third_party/libwebrtc/moz-patch-stack/s0100.patch
@@ -1,24 +1,68 @@
-From: Randell Jesup <rjesup@mozilla.com>
-Date: Tue, 1 Oct 2024 21:28:00 +0000
-Subject: Bug 1922048: fix comma instead of semicolon in libwebrtc AEC code
- r=mjf,webrtc-reviewers
+From: Michael Froman <mfroman@mozilla.com>
+Date: Thu, 10 Oct 2024 13:42:00 +0000
+Subject: Bug 1921707 - libwebrtc preprocessor, .json, .gni, and *.gn changes
+ r=ng,webrtc-reviewers
-Differential Revision: https://phabricator.services.mozilla.com/D224242
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fd7a2f8d10cf9d6e862b8f8bb8e675f0d9ddb4bf
+Differential Revision: https://phabricator.services.mozilla.com/D224076
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7017edc7cfc880bb19f94115d4f48a8c58920a9b
---
- modules/audio_processing/aec3/echo_canceller3.cc | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ .gn | 2 +-
+ media/BUILD.gn | 2 +-
+ modules/video_coding/BUILD.gn | 2 +-
+ webrtc.gni | 4 ++--
+ 4 files changed, 5 insertions(+), 5 deletions(-)
-diff --git a/modules/audio_processing/aec3/echo_canceller3.cc b/modules/audio_processing/aec3/echo_canceller3.cc
-index b6752efe73..068a0fd7f4 100644
---- a/modules/audio_processing/aec3/echo_canceller3.cc
-+++ b/modules/audio_processing/aec3/echo_canceller3.cc
-@@ -811,7 +811,7 @@ EchoCanceller3::EchoCanceller3(
- linear_output_framer_.reset(
- new BlockFramer(/*num_bands=*/1, num_capture_channels_));
- linear_output_block_ =
-- std::make_unique<Block>(/*num_bands=*/1, num_capture_channels_),
-+ std::make_unique<Block>(/*num_bands=*/1, num_capture_channels_);
- linear_output_sub_frame_view_ = std::vector<std::vector<ArrayView<float>>>(
- 1, std::vector<ArrayView<float>>(num_capture_channels_));
- }
+diff --git a/.gn b/.gn
+index 851c745732..cdbd705ec2 100644
+--- a/.gn
++++ b/.gn
+@@ -18,7 +18,7 @@ script_executable = "python3"
+ # The secondary source root is a parallel directory tree where
+ # GN build files are placed when they can not be placed directly
+ # in the source tree, e.g. for third party source trees.
+-secondary_source = "//build/secondary/"
++secondary_source = "//libwebrtc/"
+
+ # These are the targets to skip header checking by default. The files in targets
+ # matching these patterns (see "gn help label_pattern" for format) will not have
+diff --git a/media/BUILD.gn b/media/BUILD.gn
+index 0f2341a462..51d055bcf0 100644
+--- a/media/BUILD.gn
++++ b/media/BUILD.gn
+@@ -7,7 +7,7 @@
+ # be found in the AUTHORS file in the root of the source tree.
+
+ import("//build/config/linux/pkg_config.gni")
+-import("//third_party/libaom/options.gni")
++import("//libwebrtc/third_party/libaom/options.gni")
+ import("../webrtc.gni")
+
+ group("media") {
+diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
+index 2ef666d59a..2eddb81fd7 100644
+--- a/modules/video_coding/BUILD.gn
++++ b/modules/video_coding/BUILD.gn
+@@ -7,7 +7,7 @@
+ # be found in the AUTHORS file in the root of the source tree.
+
+ import("//build/config/linux/pkg_config.gni")
+-import("//third_party/libaom/options.gni")
++import("//libwebrtc/third_party/libaom/options.gni")
+ import("../../webrtc.gni")
+
+ rtc_library("encoded_frame") {
+diff --git a/webrtc.gni b/webrtc.gni
+index 10cc7468f6..afacae1027 100644
+--- a/webrtc.gni
++++ b/webrtc.gni
+@@ -487,8 +487,8 @@ all_poison_types = [
+ "software_video_codecs",
+ ]
+
+-absl_include_config = "//third_party/abseil-cpp:absl_include_config"
+-absl_define_config = "//third_party/abseil-cpp:absl_define_config"
++absl_include_config = "//libwebrtc/third_party/abseil-cpp:absl_include_config"
++absl_define_config = "//libwebrtc/third_party/abseil-cpp:absl_define_config"
+
+ # Abseil Flags are testonly, so this config will only be applied to WebRTC targets
+ # that are testonly.
diff --git a/third_party/libwebrtc/moz-patch-stack/s0101.patch b/third_party/libwebrtc/moz-patch-stack/s0101.patch
@@ -1,68 +1,100 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 10 Oct 2024 13:42:00 +0000
-Subject: Bug 1921707 - libwebrtc preprocessor, .json, .gni, and *.gn changes
- r=ng,webrtc-reviewers
+Subject: Bug 1921707 - webrtc.gni - filter out '//third_party/abseil-cpp/'
+ deps r=ng,webrtc-reviewers
-Differential Revision: https://phabricator.services.mozilla.com/D224076
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/7017edc7cfc880bb19f94115d4f48a8c58920a9b
+Differential Revision: https://phabricator.services.mozilla.com/D224079
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/45b99d1ba95b46896506de1c7dd92f19e4dc9207
---
- .gn | 2 +-
- media/BUILD.gn | 2 +-
- modules/video_coding/BUILD.gn | 2 +-
- webrtc.gni | 4 ++--
- 4 files changed, 5 insertions(+), 5 deletions(-)
+ webrtc.gni | 71 ++++++++++++++++++++++++++++++++++++++++++++++++++++++
+ 1 file changed, 71 insertions(+)
-diff --git a/.gn b/.gn
-index 851c745732..cdbd705ec2 100644
---- a/.gn
-+++ b/.gn
-@@ -18,7 +18,7 @@ script_executable = "python3"
- # The secondary source root is a parallel directory tree where
- # GN build files are placed when they can not be placed directly
- # in the source tree, e.g. for third party source trees.
--secondary_source = "//build/secondary/"
-+secondary_source = "//libwebrtc/"
-
- # These are the targets to skip header checking by default. The files in targets
- # matching these patterns (see "gn help label_pattern" for format) will not have
-diff --git a/media/BUILD.gn b/media/BUILD.gn
-index 0f2341a462..51d055bcf0 100644
---- a/media/BUILD.gn
-+++ b/media/BUILD.gn
-@@ -7,7 +7,7 @@
- # be found in the AUTHORS file in the root of the source tree.
-
- import("//build/config/linux/pkg_config.gni")
--import("//third_party/libaom/options.gni")
-+import("//libwebrtc/third_party/libaom/options.gni")
- import("../webrtc.gni")
-
- group("media") {
-diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
-index 2ef666d59a..2eddb81fd7 100644
---- a/modules/video_coding/BUILD.gn
-+++ b/modules/video_coding/BUILD.gn
-@@ -7,7 +7,7 @@
- # be found in the AUTHORS file in the root of the source tree.
-
- import("//build/config/linux/pkg_config.gni")
--import("//third_party/libaom/options.gni")
-+import("//libwebrtc/third_party/libaom/options.gni")
- import("../../webrtc.gni")
-
- rtc_library("encoded_frame") {
diff --git a/webrtc.gni b/webrtc.gni
-index 10cc7468f6..afacae1027 100644
+index afacae1027..bf4535e14c 100644
--- a/webrtc.gni
+++ b/webrtc.gni
-@@ -487,8 +487,8 @@ all_poison_types = [
- "software_video_codecs",
- ]
+@@ -691,6 +691,36 @@ template("rtc_source_set") {
+ deps += [ "//third_party/abseil-cpp:absl" ]
+ }
+ }
++
++ # Rather than widespread changes to most, if not all, BUILD.gn files
++ # we'll do some filtering on dependencies here.
++ if (build_with_mozilla && defined(deps)) {
++ # Now that abseil-cpp is moved to Mozilla's third_party directory
++ # we remove all the abseil-cpp dependency listings so moz.build
++ # generation succeed.
++ filtered_deps = []
++ foreach (dep, deps) {
++ newdep = string_replace(dep, "//third_party/abseil-cpp/", "")
++ if (newdep == dep) {
++ filtered_deps += [ dep ]
++ }
++ }
++ deps = []
++ deps = filtered_deps
++
++ # We don't build any of the test code, but many of the test
++ # targets list gtest/gmock as a dependency. We can simply
++ # filter them out here.
++ filtered_deps = []
++ foreach (dep, deps) {
++ newdep = string_replace(dep, "//testing/g", "")
++ if (newdep == dep) {
++ filtered_deps += [ dep ]
++ }
++ }
++ deps = []
++ deps = filtered_deps
++ }
+ }
+ }
--absl_include_config = "//third_party/abseil-cpp:absl_include_config"
--absl_define_config = "//third_party/abseil-cpp:absl_define_config"
-+absl_include_config = "//libwebrtc/third_party/abseil-cpp:absl_include_config"
-+absl_define_config = "//libwebrtc/third_party/abseil-cpp:absl_define_config"
+@@ -925,6 +955,47 @@ template("rtc_library") {
+ deps += [ "//third_party/abseil-cpp:absl" ]
+ }
+ }
++
++ # Rather than widespread changes to most, if not all, BUILD.gn files
++ # we'll do some filtering on dependencies here.
++ if (build_with_mozilla && defined(deps)) {
++ # Now that abseil-cpp is moved to Mozilla's third_party directory
++ # we remove all the abseil-cpp dependency listings so moz.build
++ # generation succeed.
++ filtered_deps = []
++ foreach (dep, deps) {
++ newdep = string_replace(dep, "//third_party/abseil-cpp/", "")
++ if (newdep == dep) {
++ filtered_deps += [ dep ]
++ }
++ }
++ deps = []
++ deps = filtered_deps
++
++ # We don't build any of the test code, but many of the test
++ # targets list gtest/gmock as a dependency. We can simply
++ # filter them out here.
++ filtered_deps = []
++ foreach (dep, deps) {
++ newdep = string_replace(dep, "//testing/g", "")
++ if (newdep == dep) {
++ filtered_deps += [ dep ]
++ }
++ }
++ deps = []
++ deps = filtered_deps
++
++ # Moving the google build directory requires looking for libwebrtc's
++ # third_party dependencies under libwebrtc/third_party instead of
++ # simply third_party.
++ modified_deps = []
++ foreach (dep, deps) {
++ newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/")
++ modified_deps += [ newdep ]
++ }
++ deps = []
++ deps = modified_deps
++ }
+ }
+ }
- # Abseil Flags are testonly, so this config will only be applied to WebRTC targets
- # that are testonly.
diff --git a/third_party/libwebrtc/moz-patch-stack/s0102.patch b/third_party/libwebrtc/moz-patch-stack/s0102.patch
@@ -1,100 +1,874 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 10 Oct 2024 13:42:00 +0000
-Subject: Bug 1921707 - webrtc.gni - filter out '//third_party/abseil-cpp/'
- deps r=ng,webrtc-reviewers
+Subject: Bug 1921707 - point to new build directory location in
+ third_party/libwebrtc r=ng,webrtc-reviewers
-Differential Revision: https://phabricator.services.mozilla.com/D224079
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/45b99d1ba95b46896506de1c7dd92f19e4dc9207
+Ran:
+ack -l "\"//build/config" third_party/libwebrtc | xargs sed -i.bak -E 's/\"\/\/build\/config/\"\/\/chromium\/build\/config/'
+ack -l "\"//build/dot" third_party/libwebrtc | xargs sed -i.bak -E 's/\"\/\/build\/dot/\"\/\/chromium\/build\/dot/'
+ack -l "\"//build/mac" third_party/libwebrtc | xargs sed -i.bak -E 's/\"\/\/build\/mac/\"\/\/chromium\/build\/mac/'
+find third_party -name "*.bak" | xargs rm
+hg revert third_party/libwebrtc/moz-patch-stack
+
+Differential Revision: https://phabricator.services.mozilla.com/D224543
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a342f5eca8622c8b918d6119f74aff96cc8cf582
---
- webrtc.gni | 71 ++++++++++++++++++++++++++++++++++++++++++++++++++++++
- 1 file changed, 71 insertions(+)
+ .gn | 4 +--
+ BUILD.gn | 14 +++++-----
+ api/BUILD.gn | 4 +--
+ api/audio/test/BUILD.gn | 4 +--
+ api/audio_codecs/BUILD.gn | 4 +--
+ api/audio_codecs/L16/BUILD.gn | 4 +--
+ api/audio_codecs/g711/BUILD.gn | 4 +--
+ api/audio_codecs/g722/BUILD.gn | 4 +--
+ api/audio_codecs/opus/BUILD.gn | 4 +--
+ api/audio_codecs/test/BUILD.gn | 4 +--
+ api/video/BUILD.gn | 4 +--
+ api/video_codecs/BUILD.gn | 4 +--
+ audio/BUILD.gn | 4 +--
+ common_audio/BUILD.gn | 4 +--
+ common_audio/third_party/ooura/BUILD.gn | 2 +-
+ examples/BUILD.gn | 14 +++++-----
+ .../androidapp/third_party/autobanh/BUILD.gn | 2 +-
+ examples/androidnativeapi/BUILD.gn | 4 +--
+ examples/androidvoip/BUILD.gn | 4 +--
+ logging/BUILD.gn | 4 +--
+ media/BUILD.gn | 2 +-
+ modules/audio_device/BUILD.gn | 4 +--
+ modules/audio_processing/aec3/BUILD.gn | 2 +-
+ modules/audio_processing/aecm/BUILD.gn | 2 +-
+ modules/audio_processing/agc/BUILD.gn | 2 +-
+ modules/audio_processing/agc2/BUILD.gn | 2 +-
+ .../audio_processing/agc2/rnn_vad/BUILD.gn | 8 +++---
+ modules/audio_processing/ns/BUILD.gn | 2 +-
+ modules/desktop_capture/BUILD.gn | 4 +--
+ modules/portal/BUILD.gn | 2 +-
+ modules/video_coding/BUILD.gn | 2 +-
+ pc/BUILD.gn | 4 +--
+ rtc_base/BUILD.gn | 8 +++---
+ rtc_base/memory/BUILD.gn | 4 +--
+ rtc_base/synchronization/BUILD.gn | 4 +--
+ rtc_base/system/BUILD.gn | 4 +--
+ rtc_tools/network_tester/BUILD.gn | 4 +--
+ sdk/BUILD.gn | 12 ++++----
+ system_wrappers/BUILD.gn | 4 +--
+ test/BUILD.gn | 6 ++--
+ test/fuzzers/BUILD.gn | 4 +--
+ webrtc.gni | 28 +++++++++----------
+ 42 files changed, 105 insertions(+), 105 deletions(-)
-diff --git a/webrtc.gni b/webrtc.gni
-index afacae1027..bf4535e14c 100644
---- a/webrtc.gni
-+++ b/webrtc.gni
-@@ -691,6 +691,36 @@ template("rtc_source_set") {
- deps += [ "//third_party/abseil-cpp:absl" ]
- }
+diff --git a/.gn b/.gn
+index cdbd705ec2..4ee826ec58 100644
+--- a/.gn
++++ b/.gn
+@@ -6,10 +6,10 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/dotfile_settings.gni")
++import("//chromium/build/dotfile_settings.gni")
+
+ # The location of the build configuration file.
+-buildconfig = "//build/config/BUILDCONFIG.gn"
++buildconfig = "//chromium/build/config/BUILDCONFIG.gn"
+
+ # The python interpreter to use by default. On Windows, this will look
+ # for vpython3.exe and vpython3.bat.
+diff --git a/BUILD.gn b/BUILD.gn
+index 550c75bafb..28fb57a248 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -21,15 +21,15 @@
+ # - visibility = [ ":*" ] - directory private.
+ # As a general guideline, only targets in api/ should have public visibility.
+
+-import("//build/config/linux/pkg_config.gni")
+-import("//build/config/sanitizers/sanitizers.gni")
++import("//chromium/build/config/linux/pkg_config.gni")
++import("//chromium/build/config/sanitizers/sanitizers.gni")
+ import("webrtc.gni")
+ if (rtc_enable_protobuf) {
+ import("//third_party/protobuf/proto_library.gni")
+ }
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ import("//third_party/jni_zero/jni_zero.gni")
+ }
+
+@@ -536,7 +536,7 @@ if (!build_with_chromium) {
+ sources = []
+
+ complete_static_lib = true
+- suppressed_configs += [ "//build/config/compiler:thin_archive" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:thin_archive" ]
+ defines = []
+
+ deps = [
+@@ -852,7 +852,7 @@ rtc_static_library("dcsctp") {
+ visibility = [ "//:default" ]
+ sources = []
+ complete_static_lib = true
+- suppressed_configs += [ "//build/config/compiler:thin_archive" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:thin_archive" ]
+ defines = []
+ deps = [
+ "net/dcsctp/public:factory",
+@@ -887,7 +887,7 @@ group("poison_software_video_codecs") {
+ if (!build_with_chromium) {
+ # Write debug logs to gn_logs.txt.
+ # This is also required for Siso builds.
+- import("//build/gn_logs.gni")
++ import("//chromium/build/gn_logs.gni")
+ lines = [
+ "Generated during 'gn gen' by //BUILD.gn.",
+ "",
+diff --git a/api/BUILD.gn b/api/BUILD.gn
+index ac2234a065..2aee479f4c 100644
+--- a/api/BUILD.gn
++++ b/api/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ group("api") {
+diff --git a/api/audio/test/BUILD.gn b/api/audio/test/BUILD.gn
+index b1060faae3..c567b6b0f3 100644
+--- a/api/audio/test/BUILD.gn
++++ b/api/audio/test/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ if (rtc_include_tests) {
+diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
+index 25ea12557b..aca296a06c 100644
+--- a/api/audio_codecs/BUILD.gn
++++ b/api/audio_codecs/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("audio_codecs_api") {
+diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
+index 8ce7122cec..b7ec211eaf 100644
+--- a/api/audio_codecs/L16/BUILD.gn
++++ b/api/audio_codecs/L16/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("audio_encoder_L16") {
+diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
+index 6c00e040fe..380e167dc6 100644
+--- a/api/audio_codecs/g711/BUILD.gn
++++ b/api/audio_codecs/g711/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("audio_encoder_g711") {
+diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
+index a3ee0d4e0c..322c52ac64 100644
+--- a/api/audio_codecs/g722/BUILD.gn
++++ b/api/audio_codecs/g722/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_source_set("audio_encoder_g722_config") {
+diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
+index 6df8361dbb..a86f34a973 100644
+--- a/api/audio_codecs/opus/BUILD.gn
++++ b/api/audio_codecs/opus/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("audio_encoder_opus_config") {
+diff --git a/api/audio_codecs/test/BUILD.gn b/api/audio_codecs/test/BUILD.gn
+index 40714b196e..2c6b39f59e 100644
+--- a/api/audio_codecs/test/BUILD.gn
++++ b/api/audio_codecs/test/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ if (rtc_include_tests) {
+diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
+index cd74281a22..16af840261 100644
+--- a/api/video/BUILD.gn
++++ b/api/video/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("video_rtp_headers") {
+diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
+index eddc2b7782..c767531ecd 100644
+--- a/api/video_codecs/BUILD.gn
++++ b/api/video_codecs/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("scalability_mode") {
+diff --git a/audio/BUILD.gn b/audio/BUILD.gn
+index eda3161da8..612028625a 100644
+--- a/audio/BUILD.gn
++++ b/audio/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("audio") {
+diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
+index 1e17368bf8..2dad7c98df 100644
+--- a/common_audio/BUILD.gn
++++ b/common_audio/BUILD.gn
+@@ -289,7 +289,7 @@ if (rtc_build_with_neon) {
+
+ if (target_cpu != "arm64") {
+ # Enable compilation for the NEON instruction set.
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+@@ -312,7 +312,7 @@ if (rtc_build_with_neon) {
+
+ if (target_cpu != "arm64") {
+ # Enable compilation for the NEON instruction set.
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+diff --git a/common_audio/third_party/ooura/BUILD.gn b/common_audio/third_party/ooura/BUILD.gn
+index 589142b014..2ee80ded55 100644
+--- a/common_audio/third_party/ooura/BUILD.gn
++++ b/common_audio/third_party/ooura/BUILD.gn
+@@ -40,7 +40,7 @@ rtc_library("fft_size_128") {
+
+ if (target_cpu != "arm64") {
+ # Enable compilation for the NEON instruction set.
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags += [ "-mfpu=neon" ]
}
-+
-+ # Rather than widespread changes to most, if not all, BUILD.gn files
-+ # we'll do some filtering on dependencies here.
-+ if (build_with_mozilla && defined(deps)) {
-+ # Now that abseil-cpp is moved to Mozilla's third_party directory
-+ # we remove all the abseil-cpp dependency listings so moz.build
-+ # generation succeed.
-+ filtered_deps = []
-+ foreach (dep, deps) {
-+ newdep = string_replace(dep, "//third_party/abseil-cpp/", "")
-+ if (newdep == dep) {
-+ filtered_deps += [ dep ]
-+ }
-+ }
-+ deps = []
-+ deps = filtered_deps
-+
-+ # We don't build any of the test code, but many of the test
-+ # targets list gtest/gmock as a dependency. We can simply
-+ # filter them out here.
-+ filtered_deps = []
-+ foreach (dep, deps) {
-+ newdep = string_replace(dep, "//testing/g", "")
-+ if (newdep == dep) {
-+ filtered_deps += [ dep ]
-+ }
-+ }
-+ deps = []
-+ deps = filtered_deps
-+ }
}
+diff --git a/examples/BUILD.gn b/examples/BUILD.gn
+index 6ec5e65609..73ef4bf072 100644
+--- a/examples/BUILD.gn
++++ b/examples/BUILD.gn
+@@ -9,15 +9,15 @@
+ import("../webrtc.gni")
+
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ import("//third_party/jni_zero/jni_zero.gni")
+ } else if (is_mac) {
+- import("//build/config/mac/rules.gni")
++ import("//chromium/build/config/mac/rules.gni")
+ } else if (is_ios) {
+- import("//build/config/ios/rules.gni")
++ import("//chromium/build/config/ios/rules.gni")
+ } else if (is_linux || is_chromeos) {
+- import("//build/config/linux/pkg_config.gni")
++ import("//chromium/build/config/linux/pkg_config.gni")
}
-@@ -925,6 +955,47 @@ template("rtc_library") {
- deps += [ "//third_party/abseil-cpp:absl" ]
+ group("examples") {
+@@ -632,7 +632,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
+ "../sdk:mediaconstraints_objc",
+ "../sdk:peerconnectionfactory_base_objc",
+ "../sdk:videocapture_objc",
+- "//build/config/ios:xctest",
++ "//chromium/build/config/ios:xctest",
+ "//third_party/ocmock",
+ ]
}
+@@ -737,7 +737,7 @@ if (is_linux || is_chromeos || is_win) {
+ "peerconnection/client/main_wnd.cc",
+ "peerconnection/client/main_wnd.h",
+ ]
+- configs += [ "//build/config/win:windowed" ]
++ configs += [ "//chromium/build/config/win:windowed" ]
+ deps += [
+ "../rtc_base:win32",
+ "../rtc_base:win32_socket_init",
+diff --git a/examples/androidapp/third_party/autobanh/BUILD.gn b/examples/androidapp/third_party/autobanh/BUILD.gn
+index b671239bae..e545b2cde3 100644
+--- a/examples/androidapp/third_party/autobanh/BUILD.gn
++++ b/examples/androidapp/third_party/autobanh/BUILD.gn
+@@ -7,7 +7,7 @@
+ # be found in the AUTHORS file in the root of the source tree.
+
+ if (is_android) {
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/rules.gni")
+
+ android_java_prebuilt("autobanh_java") {
+ jar_path = "lib/autobanh.jar"
+diff --git a/examples/androidnativeapi/BUILD.gn b/examples/androidnativeapi/BUILD.gn
+index 64cd43e923..6d36129869 100644
+--- a/examples/androidnativeapi/BUILD.gn
++++ b/examples/androidnativeapi/BUILD.gn
+@@ -41,8 +41,8 @@ if (is_android) {
+ "jni/onload.cc",
+ ]
+
+- suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+- configs += [ "//build/config/android:hide_all_but_jni" ]
++ suppressed_configs += [ "//chromium/build/config/android:hide_all_but_jni_onload" ]
++ configs += [ "//chromium/build/config/android:hide_all_but_jni" ]
+
+ deps = [
+ ":generated_jni",
+diff --git a/examples/androidvoip/BUILD.gn b/examples/androidvoip/BUILD.gn
+index 657e7a71f9..881dd69b03 100644
+--- a/examples/androidvoip/BUILD.gn
++++ b/examples/androidvoip/BUILD.gn
+@@ -50,8 +50,8 @@ if (is_android) {
+ "jni/onload.cc",
+ ]
+
+- suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+- configs += [ "//build/config/android:hide_all_but_jni" ]
++ suppressed_configs += [ "//chromium/build/config/android:hide_all_but_jni_onload" ]
++ configs += [ "//chromium/build/config/android:hide_all_but_jni" ]
+
+ deps = [
+ ":generated_jni",
+diff --git a/logging/BUILD.gn b/logging/BUILD.gn
+index 187019693e..e1edf6befb 100644
+--- a/logging/BUILD.gn
++++ b/logging/BUILD.gn
+@@ -11,8 +11,8 @@ if (rtc_enable_protobuf) {
+ import("//third_party/protobuf/proto_library.gni")
+ }
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ group("logging") {
+diff --git a/media/BUILD.gn b/media/BUILD.gn
+index 51d055bcf0..e50b924806 100644
+--- a/media/BUILD.gn
++++ b/media/BUILD.gn
+@@ -6,7 +6,7 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/config/linux/pkg_config.gni")
++import("//chromium/build/config/linux/pkg_config.gni")
+ import("//libwebrtc/third_party/libaom/options.gni")
+ import("../webrtc.gni")
+
+diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
+index a90fac2f4c..ae05c9d228 100644
+--- a/modules/audio_device/BUILD.gn
++++ b/modules/audio_device/BUILD.gn
+@@ -9,8 +9,8 @@
+ import("../../webrtc.gni")
+
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ config("audio_device_warnings_config") {
+diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
+index 847e5ccbea..2f6be208d2 100644
+--- a/modules/audio_processing/aec3/BUILD.gn
++++ b/modules/audio_processing/aec3/BUILD.gn
+@@ -124,7 +124,7 @@ rtc_library("aec3") {
+
+ defines = []
+ if (rtc_build_with_neon && target_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+diff --git a/modules/audio_processing/aecm/BUILD.gn b/modules/audio_processing/aecm/BUILD.gn
+index f01637f2da..d5bdc070be 100644
+--- a/modules/audio_processing/aecm/BUILD.gn
++++ b/modules/audio_processing/aecm/BUILD.gn
+@@ -30,7 +30,7 @@ rtc_library("aecm_core") {
+
+ if (target_cpu != "arm64") {
+ # Enable compilation for the NEON instruction set.
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags += [ "-mfpu=neon" ]
+ }
+ }
+diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
+index 45d08ad066..2d6f3ca0b4 100644
+--- a/modules/audio_processing/agc/BUILD.gn
++++ b/modules/audio_processing/agc/BUILD.gn
+@@ -86,7 +86,7 @@ rtc_library("legacy_agc") {
+ if (rtc_build_with_neon) {
+ if (target_cpu != "arm64") {
+ # Enable compilation for the NEON instruction set.
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+ }
+diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
+index 83f8889fff..54fcde3a18 100644
+--- a/modules/audio_processing/agc2/BUILD.gn
++++ b/modules/audio_processing/agc2/BUILD.gn
+@@ -258,7 +258,7 @@ rtc_library("vad_wrapper") {
+
+ defines = []
+ if (rtc_build_with_neon && current_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+index 0fb06b2a86..6ce62e7818 100644
+--- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
++++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
+@@ -19,7 +19,7 @@ rtc_library("rnn_vad") {
+
+ defines = []
+ if (rtc_build_with_neon && target_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+@@ -87,7 +87,7 @@ rtc_library("rnn_vad_layers") {
+
+ defines = []
+ if (rtc_build_with_neon && current_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+@@ -148,7 +148,7 @@ rtc_library("rnn_vad_pitch") {
+
+ defines = []
+ if (rtc_build_with_neon && current_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+@@ -271,7 +271,7 @@ if (rtc_include_tests) {
+
+ defines = []
+ if (rtc_build_with_neon && current_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
}
-+
-+ # Rather than widespread changes to most, if not all, BUILD.gn files
-+ # we'll do some filtering on dependencies here.
-+ if (build_with_mozilla && defined(deps)) {
-+ # Now that abseil-cpp is moved to Mozilla's third_party directory
-+ # we remove all the abseil-cpp dependency listings so moz.build
-+ # generation succeed.
-+ filtered_deps = []
-+ foreach (dep, deps) {
-+ newdep = string_replace(dep, "//third_party/abseil-cpp/", "")
-+ if (newdep == dep) {
-+ filtered_deps += [ dep ]
-+ }
-+ }
-+ deps = []
-+ deps = filtered_deps
-+
-+ # We don't build any of the test code, but many of the test
-+ # targets list gtest/gmock as a dependency. We can simply
-+ # filter them out here.
-+ filtered_deps = []
-+ foreach (dep, deps) {
-+ newdep = string_replace(dep, "//testing/g", "")
-+ if (newdep == dep) {
-+ filtered_deps += [ dep ]
-+ }
-+ }
-+ deps = []
-+ deps = filtered_deps
-+
-+ # Moving the google build directory requires looking for libwebrtc's
-+ # third_party dependencies under libwebrtc/third_party instead of
-+ # simply third_party.
-+ modified_deps = []
-+ foreach (dep, deps) {
-+ newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/")
-+ modified_deps += [ newdep ]
-+ }
-+ deps = []
-+ deps = modified_deps
-+ }
+
+diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
+index 0e96125f40..c72b3714c4 100644
+--- a/modules/audio_processing/ns/BUILD.gn
++++ b/modules/audio_processing/ns/BUILD.gn
+@@ -44,7 +44,7 @@ rtc_static_library("ns") {
+
+ defines = []
+ if (rtc_build_with_neon && target_cpu != "arm64") {
+- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
++ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
+ cflags = [ "-mfpu=neon" ]
+ }
+
+diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
+index 6f18642622..4b5acc5bdb 100644
+--- a/modules/desktop_capture/BUILD.gn
++++ b/modules/desktop_capture/BUILD.gn
+@@ -6,8 +6,8 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/config/linux/gtk/gtk.gni")
+-import("//build/config/ui.gni")
++import("//chromium/build/config/linux/gtk/gtk.gni")
++import("//chromium/build/config/ui.gni")
+ import("../../webrtc.gni")
+
+ use_desktop_capture_differ_sse2 = target_cpu == "x86" || target_cpu == "x64"
+diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
+index a693604389..72972d18e1 100644
+--- a/modules/portal/BUILD.gn
++++ b/modules/portal/BUILD.gn
+@@ -6,7 +6,7 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/config/linux/pkg_config.gni")
++import("//chromium/build/config/linux/pkg_config.gni")
+ import("//tools/generate_stubs/rules.gni")
+ import("../../webrtc.gni")
+
+diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
+index 2eddb81fd7..11da028bd3 100644
+--- a/modules/video_coding/BUILD.gn
++++ b/modules/video_coding/BUILD.gn
+@@ -6,7 +6,7 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/config/linux/pkg_config.gni")
++import("//chromium/build/config/linux/pkg_config.gni")
+ import("//libwebrtc/third_party/libaom/options.gni")
+ import("../../webrtc.gni")
+
+diff --git a/pc/BUILD.gn b/pc/BUILD.gn
+index 0c5df79fb4..70075055cf 100644
+--- a/pc/BUILD.gn
++++ b/pc/BUILD.gn
+@@ -30,8 +30,8 @@
+
+ import("../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ if (rtc_build_libsrtp) {
+diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
+index 9d319d5326..9653fd651f 100644
+--- a/rtc_base/BUILD.gn
++++ b/rtc_base/BUILD.gn
+@@ -9,8 +9,8 @@
+ import("../webrtc.gni")
+
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_source_set("protobuf_utils") {
+@@ -1894,8 +1894,8 @@ if (!rtc_rusty_base64) {
+ ]
}
+ } else {
+- import("//build/config/rust.gni")
+- import("//build/rust/rust_static_library.gni")
++ import("//chromium/build/config/rust.gni")
++ import("//chromium/build/rust/rust_static_library.gni")
+
+ rust_static_library("base64_rust_bridge") {
+ allow_unsafe = true # Needed for FFI with cxx crate.
+diff --git a/rtc_base/memory/BUILD.gn b/rtc_base/memory/BUILD.gn
+index 9a55af3ea9..3d272ba975 100644
+--- a/rtc_base/memory/BUILD.gn
++++ b/rtc_base/memory/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("aligned_malloc") {
+diff --git a/rtc_base/synchronization/BUILD.gn b/rtc_base/synchronization/BUILD.gn
+index 2a197152b9..4d5008874e 100644
+--- a/rtc_base/synchronization/BUILD.gn
++++ b/rtc_base/synchronization/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ rtc_library("yield") {
+diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn
+index 4a772795ed..de27f01bb0 100644
+--- a/rtc_base/system/BUILD.gn
++++ b/rtc_base/system/BUILD.gn
+@@ -8,8 +8,8 @@
+
+ import("../../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
}
+ rtc_source_set("arch") {
+diff --git a/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
+index 4fa98d73a2..72c01d50ab 100644
+--- a/rtc_tools/network_tester/BUILD.gn
++++ b/rtc_tools/network_tester/BUILD.gn
+@@ -180,8 +180,8 @@ if (is_android) {
+ "../../rtc_base:threading",
+ ]
+
+- suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
+- configs += [ "//build/config/android:hide_all_but_jni" ]
++ suppressed_configs += [ "//chromium/build/config/android:hide_all_but_jni_onload" ]
++ configs += [ "//chromium/build/config/android:hide_all_but_jni" ]
+
+ output_extension = "so"
+ }
+diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
+index c78cdec7cc..cc5d90b42f 100644
+--- a/sdk/BUILD.gn
++++ b/sdk/BUILD.gn
+@@ -9,12 +9,12 @@
+ import("//third_party/libaom/options.gni")
+ import("../webrtc.gni")
+ if (is_ios) {
+- import("//build/config/apple/mobile_config.gni")
+- import("//build/config/ios/ios_sdk.gni")
+- import("//build/config/ios/rules.gni")
++ import("//chromium/build/config/apple/mobile_config.gni")
++ import("//chromium/build/config/ios/ios_sdk.gni")
++ import("//chromium/build/config/ios/rules.gni")
+ }
+ if (is_mac) {
+- import("//build/config/mac/rules.gni")
++ import("//chromium/build/config/mac/rules.gni")
+ }
+
+ group("sdk") {
+@@ -67,7 +67,7 @@ if (is_ios || is_mac) {
+ ]
+
+ if (use_clang_coverage) {
+- configs = [ "//build/config/coverage:default_coverage" ]
++ configs = [ "//chromium/build/config/coverage:default_coverage" ]
+ }
+ }
+
+@@ -1280,7 +1280,7 @@ if (is_ios || is_mac) {
+ }
+
+ public_deps = [
+- "//build/config/ios:xctest",
++ "//chromium/build/config/ios:xctest",
+ "//third_party/ocmock",
+ ]
+ }
+diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn
+index 822b565610..f2ea548824 100644
+--- a/system_wrappers/BUILD.gn
++++ b/system_wrappers/BUILD.gn
+@@ -7,8 +7,8 @@
+ # be found in the AUTHORS file in the root of the source tree.
+
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+ import("../webrtc.gni")
+
+diff --git a/test/BUILD.gn b/test/BUILD.gn
+index f5039000f6..27fbece69e 100644
+--- a/test/BUILD.gn
++++ b/test/BUILD.gn
+@@ -6,10 +6,10 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/config/ui.gni")
++import("//chromium/build/config/ui.gni")
+ import("../webrtc.gni")
+ if (is_android) {
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/rules.gni")
+ import("//third_party/jni_zero/jni_zero.gni")
+ }
+
+@@ -415,7 +415,7 @@ if (is_ios && !build_with_chromium) {
+ testonly = true
+ visibility = [ "*" ]
+ sources = [ "ios/google_test_runner.mm" ]
+- configs += [ "//build/config/ios:xctest_config" ]
++ configs += [ "//chromium/build/config/ios:xctest_config" ]
+ frameworks = [ "UIKit.framework" ]
+ deps = [ ":google_test_runner_delegate" ]
+ }
+diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
+index 9d61ce653a..8d61065eab 100644
+--- a/test/fuzzers/BUILD.gn
++++ b/test/fuzzers/BUILD.gn
+@@ -6,7 +6,7 @@
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+
+-import("//build/config/features.gni")
++import("//chromium/build/config/features.gni")
+ import("//testing/libfuzzer/fuzzer_test.gni")
+ import("../../webrtc.gni")
+
+@@ -73,7 +73,7 @@ template("webrtc_fuzzer_test") {
+ }
+
+ if (!build_with_chromium && is_clang) {
+- suppressed_configs = [ "//build/config/clang:find_bad_constructs" ]
++ suppressed_configs = [ "//chromium/build/config/clang:find_bad_constructs" ]
+ }
+ }
+ }
+diff --git a/webrtc.gni b/webrtc.gni
+index bf4535e14c..b9c2178996 100644
+--- a/webrtc.gni
++++ b/webrtc.gni
+@@ -5,12 +5,12 @@
+ # tree. An additional intellectual property rights grant can be found
+ # in the file PATENTS. All contributing project authors may
+ # be found in the AUTHORS file in the root of the source tree.
+-import("//build/config/arm.gni")
+-import("//build/config/features.gni")
+-import("//build/config/mips.gni")
+-import("//build/config/ozone.gni")
+-import("//build/config/sanitizers/sanitizers.gni")
+-import("//build/config/sysroot.gni")
++import("//chromium/build/config/arm.gni")
++import("//chromium/build/config/features.gni")
++import("//chromium/build/config/mips.gni")
++import("//chromium/build/config/ozone.gni")
++import("//chromium/build/config/sanitizers/sanitizers.gni")
++import("//chromium/build/config/sysroot.gni")
+ import("//build_overrides/build.gni")
+
+ if (!build_with_chromium && is_component_build) {
+@@ -28,20 +28,20 @@ if (!build_with_chromium && is_component_build) {
+ }
+
+ if (is_ios) {
+- import("//build/config/ios/rules.gni")
++ import("//chromium/build/config/ios/rules.gni")
+ }
+
+ if (is_mac) {
+- import("//build/config/mac/rules.gni")
++ import("//chromium/build/config/mac/rules.gni")
+ }
+
+ if (is_android) {
+- import("//build/config/android/config.gni")
+- import("//build/config/android/rules.gni")
++ import("//chromium/build/config/android/config.gni")
++ import("//chromium/build/config/android/rules.gni")
+ }
+
+ if (is_fuchsia) {
+- import("//build/config/fuchsia/config.gni")
++ import("//chromium/build/config/fuchsia/config.gni")
+ }
+
+ if (build_with_chromium) {
+@@ -373,8 +373,8 @@ rtc_common_configs = [ webrtc_root + ":common_config" ]
+
+ if (is_mac || is_ios) {
+ if (filter_include(default_compiler_configs,
+- [ "//build/config/compiler:enable_arc" ]) == []) {
+- rtc_common_configs += [ "//build/config/compiler:enable_arc" ]
++ [ "//chromium/build/config/compiler:enable_arc" ]) == []) {
++ rtc_common_configs += [ "//chromium/build/config/compiler:enable_arc" ]
+ }
+ }
+
+@@ -392,7 +392,7 @@ 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) {
+- rtc_remove_configs += [ "//build/config/clang:find_bad_constructs" ]
++ rtc_remove_configs += [ "//chromium/build/config/clang:find_bad_constructs" ]
+ }
+ rtc_add_configs = rtc_common_configs
+ rtc_prod_configs = [ webrtc_root + ":rtc_prod_config" ]
diff --git a/third_party/libwebrtc/moz-patch-stack/s0103.patch b/third_party/libwebrtc/moz-patch-stack/s0103.patch
@@ -1,874 +1,55 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Thu, 10 Oct 2024 13:42:00 +0000
-Subject: Bug 1921707 - point to new build directory location in
- third_party/libwebrtc r=ng,webrtc-reviewers
+Subject: Bug 1921707 - add filtering for new build directory location in
+ third_party/libwebrtc/webrtc.gni r=ng,webrtc-reviewers
-Ran:
-ack -l "\"//build/config" third_party/libwebrtc | xargs sed -i.bak -E 's/\"\/\/build\/config/\"\/\/chromium\/build\/config/'
-ack -l "\"//build/dot" third_party/libwebrtc | xargs sed -i.bak -E 's/\"\/\/build\/dot/\"\/\/chromium\/build\/dot/'
-ack -l "\"//build/mac" third_party/libwebrtc | xargs sed -i.bak -E 's/\"\/\/build\/mac/\"\/\/chromium\/build\/mac/'
-find third_party -name "*.bak" | xargs rm
-hg revert third_party/libwebrtc/moz-patch-stack
-
-Differential Revision: https://phabricator.services.mozilla.com/D224543
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/a342f5eca8622c8b918d6119f74aff96cc8cf582
+Differential Revision: https://phabricator.services.mozilla.com/D224545
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f18c05287ce831369f44715a4174facff4f12652
---
- .gn | 4 +--
- BUILD.gn | 14 +++++-----
- api/BUILD.gn | 4 +--
- api/audio/test/BUILD.gn | 4 +--
- api/audio_codecs/BUILD.gn | 4 +--
- api/audio_codecs/L16/BUILD.gn | 4 +--
- api/audio_codecs/g711/BUILD.gn | 4 +--
- api/audio_codecs/g722/BUILD.gn | 4 +--
- api/audio_codecs/opus/BUILD.gn | 4 +--
- api/audio_codecs/test/BUILD.gn | 4 +--
- api/video/BUILD.gn | 4 +--
- api/video_codecs/BUILD.gn | 4 +--
- audio/BUILD.gn | 4 +--
- common_audio/BUILD.gn | 4 +--
- common_audio/third_party/ooura/BUILD.gn | 2 +-
- examples/BUILD.gn | 14 +++++-----
- .../androidapp/third_party/autobanh/BUILD.gn | 2 +-
- examples/androidnativeapi/BUILD.gn | 4 +--
- examples/androidvoip/BUILD.gn | 4 +--
- logging/BUILD.gn | 4 +--
- media/BUILD.gn | 2 +-
- modules/audio_device/BUILD.gn | 4 +--
- modules/audio_processing/aec3/BUILD.gn | 2 +-
- modules/audio_processing/aecm/BUILD.gn | 2 +-
- modules/audio_processing/agc/BUILD.gn | 2 +-
- modules/audio_processing/agc2/BUILD.gn | 2 +-
- .../audio_processing/agc2/rnn_vad/BUILD.gn | 8 +++---
- modules/audio_processing/ns/BUILD.gn | 2 +-
- modules/desktop_capture/BUILD.gn | 4 +--
- modules/portal/BUILD.gn | 2 +-
- modules/video_coding/BUILD.gn | 2 +-
- pc/BUILD.gn | 4 +--
- rtc_base/BUILD.gn | 8 +++---
- rtc_base/memory/BUILD.gn | 4 +--
- rtc_base/synchronization/BUILD.gn | 4 +--
- rtc_base/system/BUILD.gn | 4 +--
- rtc_tools/network_tester/BUILD.gn | 4 +--
- sdk/BUILD.gn | 12 ++++----
- system_wrappers/BUILD.gn | 4 +--
- test/BUILD.gn | 6 ++--
- test/fuzzers/BUILD.gn | 4 +--
- webrtc.gni | 28 +++++++++----------
- 42 files changed, 105 insertions(+), 105 deletions(-)
+ webrtc.gni | 14 ++++++++++++--
+ 1 file changed, 12 insertions(+), 2 deletions(-)
-diff --git a/.gn b/.gn
-index cdbd705ec2..4ee826ec58 100644
---- a/.gn
-+++ b/.gn
-@@ -6,10 +6,10 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/dotfile_settings.gni")
-+import("//chromium/build/dotfile_settings.gni")
-
- # The location of the build configuration file.
--buildconfig = "//build/config/BUILDCONFIG.gn"
-+buildconfig = "//chromium/build/config/BUILDCONFIG.gn"
-
- # The python interpreter to use by default. On Windows, this will look
- # for vpython3.exe and vpython3.bat.
-diff --git a/BUILD.gn b/BUILD.gn
-index 550c75bafb..28fb57a248 100644
---- a/BUILD.gn
-+++ b/BUILD.gn
-@@ -21,15 +21,15 @@
- # - visibility = [ ":*" ] - directory private.
- # As a general guideline, only targets in api/ should have public visibility.
-
--import("//build/config/linux/pkg_config.gni")
--import("//build/config/sanitizers/sanitizers.gni")
-+import("//chromium/build/config/linux/pkg_config.gni")
-+import("//chromium/build/config/sanitizers/sanitizers.gni")
- import("webrtc.gni")
- if (rtc_enable_protobuf) {
- import("//third_party/protobuf/proto_library.gni")
- }
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- import("//third_party/jni_zero/jni_zero.gni")
- }
-
-@@ -536,7 +536,7 @@ if (!build_with_chromium) {
- sources = []
-
- complete_static_lib = true
-- suppressed_configs += [ "//build/config/compiler:thin_archive" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:thin_archive" ]
- defines = []
-
- deps = [
-@@ -852,7 +852,7 @@ rtc_static_library("dcsctp") {
- visibility = [ "//:default" ]
- sources = []
- complete_static_lib = true
-- suppressed_configs += [ "//build/config/compiler:thin_archive" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:thin_archive" ]
- defines = []
- deps = [
- "net/dcsctp/public:factory",
-@@ -887,7 +887,7 @@ group("poison_software_video_codecs") {
- if (!build_with_chromium) {
- # Write debug logs to gn_logs.txt.
- # This is also required for Siso builds.
-- import("//build/gn_logs.gni")
-+ import("//chromium/build/gn_logs.gni")
- lines = [
- "Generated during 'gn gen' by //BUILD.gn.",
- "",
-diff --git a/api/BUILD.gn b/api/BUILD.gn
-index ac2234a065..2aee479f4c 100644
---- a/api/BUILD.gn
-+++ b/api/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- group("api") {
-diff --git a/api/audio/test/BUILD.gn b/api/audio/test/BUILD.gn
-index b1060faae3..c567b6b0f3 100644
---- a/api/audio/test/BUILD.gn
-+++ b/api/audio/test/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- if (rtc_include_tests) {
-diff --git a/api/audio_codecs/BUILD.gn b/api/audio_codecs/BUILD.gn
-index 25ea12557b..aca296a06c 100644
---- a/api/audio_codecs/BUILD.gn
-+++ b/api/audio_codecs/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("audio_codecs_api") {
-diff --git a/api/audio_codecs/L16/BUILD.gn b/api/audio_codecs/L16/BUILD.gn
-index 8ce7122cec..b7ec211eaf 100644
---- a/api/audio_codecs/L16/BUILD.gn
-+++ b/api/audio_codecs/L16/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("audio_encoder_L16") {
-diff --git a/api/audio_codecs/g711/BUILD.gn b/api/audio_codecs/g711/BUILD.gn
-index 6c00e040fe..380e167dc6 100644
---- a/api/audio_codecs/g711/BUILD.gn
-+++ b/api/audio_codecs/g711/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("audio_encoder_g711") {
-diff --git a/api/audio_codecs/g722/BUILD.gn b/api/audio_codecs/g722/BUILD.gn
-index a3ee0d4e0c..322c52ac64 100644
---- a/api/audio_codecs/g722/BUILD.gn
-+++ b/api/audio_codecs/g722/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_source_set("audio_encoder_g722_config") {
-diff --git a/api/audio_codecs/opus/BUILD.gn b/api/audio_codecs/opus/BUILD.gn
-index 6df8361dbb..a86f34a973 100644
---- a/api/audio_codecs/opus/BUILD.gn
-+++ b/api/audio_codecs/opus/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("audio_encoder_opus_config") {
-diff --git a/api/audio_codecs/test/BUILD.gn b/api/audio_codecs/test/BUILD.gn
-index 40714b196e..2c6b39f59e 100644
---- a/api/audio_codecs/test/BUILD.gn
-+++ b/api/audio_codecs/test/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- if (rtc_include_tests) {
-diff --git a/api/video/BUILD.gn b/api/video/BUILD.gn
-index cd74281a22..16af840261 100644
---- a/api/video/BUILD.gn
-+++ b/api/video/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("video_rtp_headers") {
-diff --git a/api/video_codecs/BUILD.gn b/api/video_codecs/BUILD.gn
-index eddc2b7782..c767531ecd 100644
---- a/api/video_codecs/BUILD.gn
-+++ b/api/video_codecs/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("scalability_mode") {
-diff --git a/audio/BUILD.gn b/audio/BUILD.gn
-index eda3161da8..612028625a 100644
---- a/audio/BUILD.gn
-+++ b/audio/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("audio") {
-diff --git a/common_audio/BUILD.gn b/common_audio/BUILD.gn
-index 1e17368bf8..2dad7c98df 100644
---- a/common_audio/BUILD.gn
-+++ b/common_audio/BUILD.gn
-@@ -289,7 +289,7 @@ if (rtc_build_with_neon) {
-
- if (target_cpu != "arm64") {
- # Enable compilation for the NEON instruction set.
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-@@ -312,7 +312,7 @@ if (rtc_build_with_neon) {
-
- if (target_cpu != "arm64") {
- # Enable compilation for the NEON instruction set.
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-diff --git a/common_audio/third_party/ooura/BUILD.gn b/common_audio/third_party/ooura/BUILD.gn
-index 589142b014..2ee80ded55 100644
---- a/common_audio/third_party/ooura/BUILD.gn
-+++ b/common_audio/third_party/ooura/BUILD.gn
-@@ -40,7 +40,7 @@ rtc_library("fft_size_128") {
-
- if (target_cpu != "arm64") {
- # Enable compilation for the NEON instruction set.
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags += [ "-mfpu=neon" ]
- }
- }
-diff --git a/examples/BUILD.gn b/examples/BUILD.gn
-index 6ec5e65609..73ef4bf072 100644
---- a/examples/BUILD.gn
-+++ b/examples/BUILD.gn
-@@ -9,15 +9,15 @@
- import("../webrtc.gni")
-
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- import("//third_party/jni_zero/jni_zero.gni")
- } else if (is_mac) {
-- import("//build/config/mac/rules.gni")
-+ import("//chromium/build/config/mac/rules.gni")
- } else if (is_ios) {
-- import("//build/config/ios/rules.gni")
-+ import("//chromium/build/config/ios/rules.gni")
- } else if (is_linux || is_chromeos) {
-- import("//build/config/linux/pkg_config.gni")
-+ import("//chromium/build/config/linux/pkg_config.gni")
- }
-
- group("examples") {
-@@ -632,7 +632,7 @@ if (is_ios || (is_mac && target_cpu != "x86")) {
- "../sdk:mediaconstraints_objc",
- "../sdk:peerconnectionfactory_base_objc",
- "../sdk:videocapture_objc",
-- "//build/config/ios:xctest",
-+ "//chromium/build/config/ios:xctest",
- "//third_party/ocmock",
- ]
- }
-@@ -737,7 +737,7 @@ if (is_linux || is_chromeos || is_win) {
- "peerconnection/client/main_wnd.cc",
- "peerconnection/client/main_wnd.h",
- ]
-- configs += [ "//build/config/win:windowed" ]
-+ configs += [ "//chromium/build/config/win:windowed" ]
+diff --git a/webrtc.gni b/webrtc.gni
+index b9c2178996..34a8ee592d 100644
+--- a/webrtc.gni
++++ b/webrtc.gni
+@@ -536,7 +536,7 @@ template("rtc_test") {
+ min_sdk_version = 21
+ target_sdk_version = 24
deps += [
- "../rtc_base:win32",
- "../rtc_base:win32_socket_init",
-diff --git a/examples/androidapp/third_party/autobanh/BUILD.gn b/examples/androidapp/third_party/autobanh/BUILD.gn
-index b671239bae..e545b2cde3 100644
---- a/examples/androidapp/third_party/autobanh/BUILD.gn
-+++ b/examples/androidapp/third_party/autobanh/BUILD.gn
-@@ -7,7 +7,7 @@
- # be found in the AUTHORS file in the root of the source tree.
-
- if (is_android) {
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/rules.gni")
-
- android_java_prebuilt("autobanh_java") {
- jar_path = "lib/autobanh.jar"
-diff --git a/examples/androidnativeapi/BUILD.gn b/examples/androidnativeapi/BUILD.gn
-index 64cd43e923..6d36129869 100644
---- a/examples/androidnativeapi/BUILD.gn
-+++ b/examples/androidnativeapi/BUILD.gn
-@@ -41,8 +41,8 @@ if (is_android) {
- "jni/onload.cc",
- ]
-
-- suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
-- configs += [ "//build/config/android:hide_all_but_jni" ]
-+ suppressed_configs += [ "//chromium/build/config/android:hide_all_but_jni_onload" ]
-+ configs += [ "//chromium/build/config/android:hide_all_but_jni" ]
-
- deps = [
- ":generated_jni",
-diff --git a/examples/androidvoip/BUILD.gn b/examples/androidvoip/BUILD.gn
-index 657e7a71f9..881dd69b03 100644
---- a/examples/androidvoip/BUILD.gn
-+++ b/examples/androidvoip/BUILD.gn
-@@ -50,8 +50,8 @@ if (is_android) {
- "jni/onload.cc",
- ]
-
-- suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
-- configs += [ "//build/config/android:hide_all_but_jni" ]
-+ suppressed_configs += [ "//chromium/build/config/android:hide_all_but_jni_onload" ]
-+ configs += [ "//chromium/build/config/android:hide_all_but_jni" ]
-
- deps = [
- ":generated_jni",
-diff --git a/logging/BUILD.gn b/logging/BUILD.gn
-index 187019693e..e1edf6befb 100644
---- a/logging/BUILD.gn
-+++ b/logging/BUILD.gn
-@@ -11,8 +11,8 @@ if (rtc_enable_protobuf) {
- import("//third_party/protobuf/proto_library.gni")
- }
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- group("logging") {
-diff --git a/media/BUILD.gn b/media/BUILD.gn
-index 51d055bcf0..e50b924806 100644
---- a/media/BUILD.gn
-+++ b/media/BUILD.gn
-@@ -6,7 +6,7 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/config/linux/pkg_config.gni")
-+import("//chromium/build/config/linux/pkg_config.gni")
- import("//libwebrtc/third_party/libaom/options.gni")
- import("../webrtc.gni")
-
-diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
-index a90fac2f4c..ae05c9d228 100644
---- a/modules/audio_device/BUILD.gn
-+++ b/modules/audio_device/BUILD.gn
-@@ -9,8 +9,8 @@
- import("../../webrtc.gni")
-
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- config("audio_device_warnings_config") {
-diff --git a/modules/audio_processing/aec3/BUILD.gn b/modules/audio_processing/aec3/BUILD.gn
-index 847e5ccbea..2f6be208d2 100644
---- a/modules/audio_processing/aec3/BUILD.gn
-+++ b/modules/audio_processing/aec3/BUILD.gn
-@@ -124,7 +124,7 @@ rtc_library("aec3") {
-
- defines = []
- if (rtc_build_with_neon && target_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-diff --git a/modules/audio_processing/aecm/BUILD.gn b/modules/audio_processing/aecm/BUILD.gn
-index f01637f2da..d5bdc070be 100644
---- a/modules/audio_processing/aecm/BUILD.gn
-+++ b/modules/audio_processing/aecm/BUILD.gn
-@@ -30,7 +30,7 @@ rtc_library("aecm_core") {
-
- if (target_cpu != "arm64") {
- # Enable compilation for the NEON instruction set.
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags += [ "-mfpu=neon" ]
- }
- }
-diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
-index 45d08ad066..2d6f3ca0b4 100644
---- a/modules/audio_processing/agc/BUILD.gn
-+++ b/modules/audio_processing/agc/BUILD.gn
-@@ -86,7 +86,7 @@ rtc_library("legacy_agc") {
- if (rtc_build_with_neon) {
- if (target_cpu != "arm64") {
- # Enable compilation for the NEON instruction set.
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
- }
-diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
-index 83f8889fff..54fcde3a18 100644
---- a/modules/audio_processing/agc2/BUILD.gn
-+++ b/modules/audio_processing/agc2/BUILD.gn
-@@ -258,7 +258,7 @@ rtc_library("vad_wrapper") {
-
- defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-diff --git a/modules/audio_processing/agc2/rnn_vad/BUILD.gn b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
-index 0fb06b2a86..6ce62e7818 100644
---- a/modules/audio_processing/agc2/rnn_vad/BUILD.gn
-+++ b/modules/audio_processing/agc2/rnn_vad/BUILD.gn
-@@ -19,7 +19,7 @@ rtc_library("rnn_vad") {
-
- defines = []
- if (rtc_build_with_neon && target_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-@@ -87,7 +87,7 @@ rtc_library("rnn_vad_layers") {
-
- defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-@@ -148,7 +148,7 @@ rtc_library("rnn_vad_pitch") {
-
- defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-@@ -271,7 +271,7 @@ if (rtc_include_tests) {
-
- defines = []
- if (rtc_build_with_neon && current_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-diff --git a/modules/audio_processing/ns/BUILD.gn b/modules/audio_processing/ns/BUILD.gn
-index 0e96125f40..c72b3714c4 100644
---- a/modules/audio_processing/ns/BUILD.gn
-+++ b/modules/audio_processing/ns/BUILD.gn
-@@ -44,7 +44,7 @@ rtc_static_library("ns") {
-
- defines = []
- if (rtc_build_with_neon && target_cpu != "arm64") {
-- suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
-+ suppressed_configs += [ "//chromium/build/config/compiler:compiler_arm_fpu" ]
- cflags = [ "-mfpu=neon" ]
- }
-
-diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
-index 6f18642622..4b5acc5bdb 100644
---- a/modules/desktop_capture/BUILD.gn
-+++ b/modules/desktop_capture/BUILD.gn
-@@ -6,8 +6,8 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/config/linux/gtk/gtk.gni")
--import("//build/config/ui.gni")
-+import("//chromium/build/config/linux/gtk/gtk.gni")
-+import("//chromium/build/config/ui.gni")
- import("../../webrtc.gni")
-
- use_desktop_capture_differ_sse2 = target_cpu == "x86" || target_cpu == "x64"
-diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
-index a693604389..72972d18e1 100644
---- a/modules/portal/BUILD.gn
-+++ b/modules/portal/BUILD.gn
-@@ -6,7 +6,7 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/config/linux/pkg_config.gni")
-+import("//chromium/build/config/linux/pkg_config.gni")
- import("//tools/generate_stubs/rules.gni")
- import("../../webrtc.gni")
-
-diff --git a/modules/video_coding/BUILD.gn b/modules/video_coding/BUILD.gn
-index 2eddb81fd7..11da028bd3 100644
---- a/modules/video_coding/BUILD.gn
-+++ b/modules/video_coding/BUILD.gn
-@@ -6,7 +6,7 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/config/linux/pkg_config.gni")
-+import("//chromium/build/config/linux/pkg_config.gni")
- import("//libwebrtc/third_party/libaom/options.gni")
- import("../../webrtc.gni")
-
-diff --git a/pc/BUILD.gn b/pc/BUILD.gn
-index 0c5df79fb4..70075055cf 100644
---- a/pc/BUILD.gn
-+++ b/pc/BUILD.gn
-@@ -30,8 +30,8 @@
-
- import("../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- if (rtc_build_libsrtp) {
-diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
-index 9d319d5326..9653fd651f 100644
---- a/rtc_base/BUILD.gn
-+++ b/rtc_base/BUILD.gn
-@@ -9,8 +9,8 @@
- import("../webrtc.gni")
-
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_source_set("protobuf_utils") {
-@@ -1894,8 +1894,8 @@ if (!rtc_rusty_base64) {
- ]
- }
- } else {
-- import("//build/config/rust.gni")
-- import("//build/rust/rust_static_library.gni")
-+ import("//chromium/build/config/rust.gni")
-+ import("//chromium/build/rust/rust_static_library.gni")
-
- rust_static_library("base64_rust_bridge") {
- allow_unsafe = true # Needed for FFI with cxx crate.
-diff --git a/rtc_base/memory/BUILD.gn b/rtc_base/memory/BUILD.gn
-index 9a55af3ea9..3d272ba975 100644
---- a/rtc_base/memory/BUILD.gn
-+++ b/rtc_base/memory/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("aligned_malloc") {
-diff --git a/rtc_base/synchronization/BUILD.gn b/rtc_base/synchronization/BUILD.gn
-index 2a197152b9..4d5008874e 100644
---- a/rtc_base/synchronization/BUILD.gn
-+++ b/rtc_base/synchronization/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_library("yield") {
-diff --git a/rtc_base/system/BUILD.gn b/rtc_base/system/BUILD.gn
-index 4a772795ed..de27f01bb0 100644
---- a/rtc_base/system/BUILD.gn
-+++ b/rtc_base/system/BUILD.gn
-@@ -8,8 +8,8 @@
-
- import("../../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- rtc_source_set("arch") {
-diff --git a/rtc_tools/network_tester/BUILD.gn b/rtc_tools/network_tester/BUILD.gn
-index 4fa98d73a2..72c01d50ab 100644
---- a/rtc_tools/network_tester/BUILD.gn
-+++ b/rtc_tools/network_tester/BUILD.gn
-@@ -180,8 +180,8 @@ if (is_android) {
- "../../rtc_base:threading",
- ]
-
-- suppressed_configs += [ "//build/config/android:hide_all_but_jni_onload" ]
-- configs += [ "//build/config/android:hide_all_but_jni" ]
-+ suppressed_configs += [ "//chromium/build/config/android:hide_all_but_jni_onload" ]
-+ configs += [ "//chromium/build/config/android:hide_all_but_jni" ]
-
- output_extension = "so"
- }
-diff --git a/sdk/BUILD.gn b/sdk/BUILD.gn
-index c78cdec7cc..cc5d90b42f 100644
---- a/sdk/BUILD.gn
-+++ b/sdk/BUILD.gn
-@@ -9,12 +9,12 @@
- import("//third_party/libaom/options.gni")
- import("../webrtc.gni")
- if (is_ios) {
-- import("//build/config/apple/mobile_config.gni")
-- import("//build/config/ios/ios_sdk.gni")
-- import("//build/config/ios/rules.gni")
-+ import("//chromium/build/config/apple/mobile_config.gni")
-+ import("//chromium/build/config/ios/ios_sdk.gni")
-+ import("//chromium/build/config/ios/rules.gni")
- }
- if (is_mac) {
-- import("//build/config/mac/rules.gni")
-+ import("//chromium/build/config/mac/rules.gni")
- }
-
- group("sdk") {
-@@ -67,7 +67,7 @@ if (is_ios || is_mac) {
- ]
-
- if (use_clang_coverage) {
-- configs = [ "//build/config/coverage:default_coverage" ]
-+ configs = [ "//chromium/build/config/coverage:default_coverage" ]
+- "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
++ "//chromium/build/android/gtest_apk:native_test_instrumentation_test_runner_java",
+ webrtc_root + "sdk/android:native_test_jni_onload",
+ webrtc_root + "sdk/android:base_java",
+ webrtc_root + "test:native_test_java",
+@@ -991,11 +991,21 @@ template("rtc_library") {
+ modified_deps = []
+ foreach (dep, deps) {
+ newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/")
++ newdep = string_replace(newdep, "//build", "//chromium/build")
+ modified_deps += [ newdep ]
+ }
+ deps = []
+ deps = modified_deps
}
++ if (build_with_mozilla && defined(data_deps)) {
++ modified_deps = []
++ foreach (dep, data_deps) {
++ newdep = string_replace(dep, "//build/", "//chromium/build/")
++ modified_deps += [ newdep ]
++ }
++ data_deps = []
++ data_deps = modified_deps
++ }
}
-
-@@ -1280,7 +1280,7 @@ if (is_ios || is_mac) {
- }
-
- public_deps = [
-- "//build/config/ios:xctest",
-+ "//chromium/build/config/ios:xctest",
- "//third_party/ocmock",
- ]
- }
-diff --git a/system_wrappers/BUILD.gn b/system_wrappers/BUILD.gn
-index 822b565610..f2ea548824 100644
---- a/system_wrappers/BUILD.gn
-+++ b/system_wrappers/BUILD.gn
-@@ -7,8 +7,8 @@
- # be found in the AUTHORS file in the root of the source tree.
-
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
}
- import("../webrtc.gni")
-diff --git a/test/BUILD.gn b/test/BUILD.gn
-index f5039000f6..27fbece69e 100644
---- a/test/BUILD.gn
-+++ b/test/BUILD.gn
-@@ -6,10 +6,10 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/config/ui.gni")
-+import("//chromium/build/config/ui.gni")
- import("../webrtc.gni")
- if (is_android) {
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/rules.gni")
- import("//third_party/jni_zero/jni_zero.gni")
- }
-
-@@ -415,7 +415,7 @@ if (is_ios && !build_with_chromium) {
- testonly = true
- visibility = [ "*" ]
- sources = [ "ios/google_test_runner.mm" ]
-- configs += [ "//build/config/ios:xctest_config" ]
-+ configs += [ "//chromium/build/config/ios:xctest_config" ]
- frameworks = [ "UIKit.framework" ]
- deps = [ ":google_test_runner_delegate" ]
- }
-diff --git a/test/fuzzers/BUILD.gn b/test/fuzzers/BUILD.gn
-index 9d61ce653a..8d61065eab 100644
---- a/test/fuzzers/BUILD.gn
-+++ b/test/fuzzers/BUILD.gn
-@@ -6,7 +6,7 @@
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
-
--import("//build/config/features.gni")
-+import("//chromium/build/config/features.gni")
- import("//testing/libfuzzer/fuzzer_test.gni")
- import("../../webrtc.gni")
-
-@@ -73,7 +73,7 @@ template("webrtc_fuzzer_test") {
+@@ -1033,7 +1043,7 @@ template("rtc_executable") {
+ if (is_win) {
+ deps += [
+ # Give executables the default manifest on Windows (a no-op elsewhere).
+- "//build/win:default_exe_manifest",
++ "//chromium/build/win:default_exe_manifest",
+ ]
}
- if (!build_with_chromium && is_clang) {
-- suppressed_configs = [ "//build/config/clang:find_bad_constructs" ]
-+ suppressed_configs = [ "//chromium/build/config/clang:find_bad_constructs" ]
- }
- }
- }
-diff --git a/webrtc.gni b/webrtc.gni
-index bf4535e14c..b9c2178996 100644
---- a/webrtc.gni
-+++ b/webrtc.gni
-@@ -5,12 +5,12 @@
- # tree. An additional intellectual property rights grant can be found
- # in the file PATENTS. All contributing project authors may
- # be found in the AUTHORS file in the root of the source tree.
--import("//build/config/arm.gni")
--import("//build/config/features.gni")
--import("//build/config/mips.gni")
--import("//build/config/ozone.gni")
--import("//build/config/sanitizers/sanitizers.gni")
--import("//build/config/sysroot.gni")
-+import("//chromium/build/config/arm.gni")
-+import("//chromium/build/config/features.gni")
-+import("//chromium/build/config/mips.gni")
-+import("//chromium/build/config/ozone.gni")
-+import("//chromium/build/config/sanitizers/sanitizers.gni")
-+import("//chromium/build/config/sysroot.gni")
- import("//build_overrides/build.gni")
-
- if (!build_with_chromium && is_component_build) {
-@@ -28,20 +28,20 @@ if (!build_with_chromium && is_component_build) {
- }
-
- if (is_ios) {
-- import("//build/config/ios/rules.gni")
-+ import("//chromium/build/config/ios/rules.gni")
- }
-
- if (is_mac) {
-- import("//build/config/mac/rules.gni")
-+ import("//chromium/build/config/mac/rules.gni")
- }
-
- if (is_android) {
-- import("//build/config/android/config.gni")
-- import("//build/config/android/rules.gni")
-+ import("//chromium/build/config/android/config.gni")
-+ import("//chromium/build/config/android/rules.gni")
- }
-
- if (is_fuchsia) {
-- import("//build/config/fuchsia/config.gni")
-+ import("//chromium/build/config/fuchsia/config.gni")
- }
-
- if (build_with_chromium) {
-@@ -373,8 +373,8 @@ rtc_common_configs = [ webrtc_root + ":common_config" ]
-
- if (is_mac || is_ios) {
- if (filter_include(default_compiler_configs,
-- [ "//build/config/compiler:enable_arc" ]) == []) {
-- rtc_common_configs += [ "//build/config/compiler:enable_arc" ]
-+ [ "//chromium/build/config/compiler:enable_arc" ]) == []) {
-+ rtc_common_configs += [ "//chromium/build/config/compiler:enable_arc" ]
- }
- }
-
-@@ -392,7 +392,7 @@ 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) {
-- rtc_remove_configs += [ "//build/config/clang:find_bad_constructs" ]
-+ rtc_remove_configs += [ "//chromium/build/config/clang:find_bad_constructs" ]
- }
- rtc_add_configs = rtc_common_configs
- rtc_prod_configs = [ webrtc_root + ":rtc_prod_config" ]
diff --git a/third_party/libwebrtc/moz-patch-stack/s0104.patch b/third_party/libwebrtc/moz-patch-stack/s0104.patch
@@ -1,55 +1,23 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Thu, 10 Oct 2024 13:42:00 +0000
-Subject: Bug 1921707 - add filtering for new build directory location in
- third_party/libwebrtc/webrtc.gni r=ng,webrtc-reviewers
+Date: Tue, 15 Oct 2024 16:46:00 -0500
+Subject: Bug 1924098 - (fix-e432503389) upstream changed signature of
+ LibvpxVp9Encoder::InitAndSetControlSettings
-Differential Revision: https://phabricator.services.mozilla.com/D224545
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f18c05287ce831369f44715a4174facff4f12652
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fadad7695efe0d9cc6b999a613b9e1f536e660bd
---
- webrtc.gni | 14 ++++++++++++--
- 1 file changed, 12 insertions(+), 2 deletions(-)
+ modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/webrtc.gni b/webrtc.gni
-index b9c2178996..34a8ee592d 100644
---- a/webrtc.gni
-+++ b/webrtc.gni
-@@ -536,7 +536,7 @@ template("rtc_test") {
- min_sdk_version = 21
- target_sdk_version = 24
- deps += [
-- "//build/android/gtest_apk:native_test_instrumentation_test_runner_java",
-+ "//chromium/build/android/gtest_apk:native_test_instrumentation_test_runner_java",
- webrtc_root + "sdk/android:native_test_jni_onload",
- webrtc_root + "sdk/android:base_java",
- webrtc_root + "test:native_test_java",
-@@ -991,11 +991,21 @@ template("rtc_library") {
- modified_deps = []
- foreach (dep, deps) {
- newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/")
-+ newdep = string_replace(newdep, "//build", "//chromium/build")
- modified_deps += [ newdep ]
- }
- deps = []
- deps = modified_deps
- }
-+ if (build_with_mozilla && defined(data_deps)) {
-+ modified_deps = []
-+ foreach (dep, data_deps) {
-+ newdep = string_replace(dep, "//build/", "//chromium/build/")
-+ modified_deps += [ newdep ]
-+ }
-+ data_deps = []
-+ data_deps = modified_deps
-+ }
- }
- }
-
-@@ -1033,7 +1043,7 @@ template("rtc_executable") {
- if (is_win) {
- deps += [
- # Give executables the default manifest on Windows (a no-op elsewhere).
-- "//build/win:default_exe_manifest",
-+ "//chromium/build/win:default_exe_manifest",
- ]
- }
-
+diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+index f0f2e3b853..34f50ee61a 100644
+--- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
++++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
+@@ -1338,7 +1338,7 @@ int LibvpxVp9Encoder::UpdateCodecFrameSize(
+ // bugs in trying to do it the "right" way, we basically re-do
+ // the initialization.
+ vpx_codec_destroy(encoder_); // clean up old state
+- int result = InitAndSetControlSettings(&codec_);
++ int result = InitAndSetControlSettings();
+ if (result == WEBRTC_VIDEO_CODEC_OK) {
+ // TODO: Mozilla rates have become much more complicated, we need to store
+ // more state or find another way of doing this.
diff --git a/third_party/libwebrtc/moz-patch-stack/s0105.patch b/third_party/libwebrtc/moz-patch-stack/s0105.patch
@@ -1,23 +1,31 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Tue, 15 Oct 2024 16:46:00 -0500
-Subject: Bug 1924098 - (fix-e432503389) upstream changed signature of
- LibvpxVp9Encoder::InitAndSetControlSettings
+Date: Tue, 29 Oct 2024 18:20:00 +0000
+Subject: Bug 1925119 - add dcsctp to libwebrtc build. r=bwc,webrtc-reviewers
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/fadad7695efe0d9cc6b999a613b9e1f536e660bd
+Differential Revision: https://phabricator.services.mozilla.com/D225892
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/90a33b06e51017d25542a95c2a460bfed41c553b
---
- modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ BUILD.gn | 3 ++-
+ 1 file changed, 2 insertions(+), 1 deletion(-)
-diff --git a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
-index f0f2e3b853..34f50ee61a 100644
---- a/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
-+++ b/modules/video_coding/codecs/vp9/libvpx_vp9_encoder.cc
-@@ -1338,7 +1338,7 @@ int LibvpxVp9Encoder::UpdateCodecFrameSize(
- // bugs in trying to do it the "right" way, we basically re-do
- // the initialization.
- vpx_codec_destroy(encoder_); // clean up old state
-- int result = InitAndSetControlSettings(&codec_);
-+ int result = InitAndSetControlSettings();
- if (result == WEBRTC_VIDEO_CODEC_OK) {
- // TODO: Mozilla rates have become much more complicated, we need to store
- // more state or find another way of doing this.
+diff --git a/BUILD.gn b/BUILD.gn
+index 28fb57a248..2e315c79c4 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -565,6 +565,7 @@ if (!build_with_chromium) {
+ "call",
+ "common_audio",
+ "common_video",
++ ":dcsctp",
+ "logging:rtc_event_log_api",
+ "media",
+ "modules",
+@@ -849,7 +850,7 @@ if (rtc_include_tests && !build_with_chromium) {
+ # Build target for standalone dcsctp
+ rtc_static_library("dcsctp") {
+ # Only the root target should depend on this.
+- visibility = [ "//:default" ]
++ visibility = [ "//:default", "//libwebrtc:webrtc" ]
+ sources = []
+ complete_static_lib = true
+ suppressed_configs += [ "//chromium/build/config/compiler:thin_archive" ]
diff --git a/third_party/libwebrtc/moz-patch-stack/s0106.patch b/third_party/libwebrtc/moz-patch-stack/s0106.patch
@@ -1,31 +1,25 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Tue, 29 Oct 2024 18:20:00 +0000
-Subject: Bug 1925119 - add dcsctp to libwebrtc build. r=bwc,webrtc-reviewers
+Date: Tue, 19 Nov 2024 16:05:00 -0600
+Subject: Bug 1928110 - (fix-b9c4c242d4) handle upstream last_arrival_timestamp
+ to last_arrival_ntp_timestamp
-Differential Revision: https://phabricator.services.mozilla.com/D225892
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/90a33b06e51017d25542a95c2a460bfed41c553b
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/6918cf2d9126fa3e3676386349f6e383c298d8ab
---
- BUILD.gn | 3 ++-
- 1 file changed, 2 insertions(+), 1 deletion(-)
+ modules/rtp_rtcp/source/rtcp_receiver.cc | 4 ++--
+ 1 file changed, 2 insertions(+), 2 deletions(-)
-diff --git a/BUILD.gn b/BUILD.gn
-index 28fb57a248..2e315c79c4 100644
---- a/BUILD.gn
-+++ b/BUILD.gn
-@@ -565,6 +565,7 @@ if (!build_with_chromium) {
- "call",
- "common_audio",
- "common_video",
-+ ":dcsctp",
- "logging:rtc_event_log_api",
- "media",
- "modules",
-@@ -849,7 +850,7 @@ if (rtc_include_tests && !build_with_chromium) {
- # Build target for standalone dcsctp
- rtc_static_library("dcsctp") {
- # Only the root target should depend on this.
-- visibility = [ "//:default" ]
-+ visibility = [ "//:default", "//libwebrtc:webrtc" ]
- sources = []
- complete_static_lib = true
- suppressed_configs += [ "//chromium/build/config/compiler:thin_archive" ]
+diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
+index 9061eec513..cb3111127a 100644
+--- a/modules/rtp_rtcp/source/rtcp_receiver.cc
++++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
+@@ -393,8 +393,8 @@ void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
+ MutexLock lock(&rtcp_receiver_lock_);
+ *packet_count = remote_sender_.packets_sent;
+ *octet_count = remote_sender_.bytes_sent;
+- *ntp_timestamp_ms = remote_sender_.last_arrival_timestamp.ToMs();
+- *remote_ntp_timestamp_ms = remote_sender_.last_remote_timestamp.ToMs();
++ *ntp_timestamp_ms = remote_sender_.last_arrival_ntp_timestamp.ToMs();
++ *remote_ntp_timestamp_ms = remote_sender_.last_remote_ntp_timestamp.ToMs();
+ }
+
+ std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
diff --git a/third_party/libwebrtc/moz-patch-stack/s0107.patch b/third_party/libwebrtc/moz-patch-stack/s0107.patch
@@ -1,25 +1,170 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Tue, 19 Nov 2024 16:05:00 -0600
-Subject: Bug 1928110 - (fix-b9c4c242d4) handle upstream last_arrival_timestamp
- to last_arrival_ntp_timestamp
+From: Nico Grunbaum <na-g@nostrum.com>
+Date: Wed, 27 Nov 2024 03:41:00 +0000
+Subject: Bug 1921154 - libwebrtc build related changes for
+ AV1;r=mjf,webrtc-reviewers
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/6918cf2d9126fa3e3676386349f6e383c298d8ab
+These are most of the changes needed to get a build with AV1.
+The crypto/openssl changes are later altered in the stack adding an NSPR based implementation of the functionality that was missing.
+
+Differential Revision: https://phabricator.services.mozilla.com/D228540
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/4527c41ef71d7683c60d9c435991a0d93bc5ce4d
---
- modules/rtp_rtcp/source/rtcp_receiver.cc | 4 ++--
- 1 file changed, 2 insertions(+), 2 deletions(-)
+ BUILD.gn | 4 ++++
+ media/BUILD.gn | 24 ++++++++++++++----------
+ rtc_base/BUILD.gn | 4 ++++
+ webrtc.gni | 4 ++--
+ 4 files changed, 24 insertions(+), 12 deletions(-)
-diff --git a/modules/rtp_rtcp/source/rtcp_receiver.cc b/modules/rtp_rtcp/source/rtcp_receiver.cc
-index 9061eec513..cb3111127a 100644
---- a/modules/rtp_rtcp/source/rtcp_receiver.cc
-+++ b/modules/rtp_rtcp/source/rtcp_receiver.cc
-@@ -393,8 +393,8 @@ void RTCPReceiver::RemoteRTCPSenderInfo(uint32_t* packet_count,
- MutexLock lock(&rtcp_receiver_lock_);
- *packet_count = remote_sender_.packets_sent;
- *octet_count = remote_sender_.bytes_sent;
-- *ntp_timestamp_ms = remote_sender_.last_arrival_timestamp.ToMs();
-- *remote_ntp_timestamp_ms = remote_sender_.last_remote_timestamp.ToMs();
-+ *ntp_timestamp_ms = remote_sender_.last_arrival_ntp_timestamp.ToMs();
-+ *remote_ntp_timestamp_ms = remote_sender_.last_remote_ntp_timestamp.ToMs();
- }
-
- std::vector<ReportBlockData> RTCPReceiver::GetLatestReportBlockData() const {
+diff --git a/BUILD.gn b/BUILD.gn
+index 2e315c79c4..174ce0d2f4 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -521,6 +521,9 @@ if (!rtc_build_ssl) {
+ "crypto",
+ "ssl",
+ ]
++ if (build_with_mozilla) {
++ libs = []
++ }
+ }
+ }
+
+@@ -615,6 +618,7 @@ if (!build_with_chromium) {
+ "api/environment:environment_factory",
+ "api/video:video_frame",
+ "api/video:video_rtp_headers",
++ "api/video_codecs:builtin_video_decoder_factory",
+ "test:rtp_test_utils",
+ ]
+ # Added when we removed deps in other places to avoid building
+diff --git a/media/BUILD.gn b/media/BUILD.gn
+index e50b924806..ac722ddda3 100644
+--- a/media/BUILD.gn
++++ b/media/BUILD.gn
+@@ -12,12 +12,10 @@ import("../webrtc.gni")
+
+ group("media") {
+ deps = []
+- if (!build_with_mozilla) {
+- deps += [
+- ":rtc_media",
+- ":rtc_media_base",
+- ]
+- }
++ deps += [
++ ":rtc_media",
++ ":rtc_media_base",
++ ]
+ }
+
+ config("rtc_media_defines_config") {
+@@ -281,6 +279,9 @@ rtc_library("media_engine") {
+ "base/media_engine.cc",
+ "base/media_engine.h",
+ ]
++ deps -= [
++ ":media_channel_impl",
++ ]
+ }
+ }
+
+@@ -432,6 +433,7 @@ rtc_library("codec_list") {
+ ]
+ }
+
++# MOZILLA Bug 1931241 - how much of this do we need?
+ rtc_library("rtp_utils") {
+ if (!build_with_mozilla) {
+ sources = [
+@@ -478,7 +480,6 @@ rtc_library("media_constants") {
+ }
+
+ rtc_library("turn_utils") {
+-if (!build_with_mozilla) {
+ sources = [
+ "base/turn_utils.cc",
+ "base/turn_utils.h",
+@@ -489,17 +490,14 @@ if (!build_with_mozilla) {
+ "../rtc_base/system:rtc_export",
+ ]
+ }
+-}
+
+ rtc_library("rid_description") {
+-if (!build_with_mozilla) {
+ sources = [
+ "base/rid_description.cc",
+ "base/rid_description.h",
+ ]
+ deps = [ ":codec" ]
+ }
+-}
+
+ rtc_library("rtc_simulcast_encoder_adapter") {
+ visibility = [ "*" ]
+@@ -717,6 +715,12 @@ rtc_library("rtc_audio_video") {
+ "engine/webrtc_voice_engine.cc",
+ "engine/webrtc_voice_engine.h",
+ ]
++ if (build_with_mozilla) {
++ sources -= [
++ "engine/webrtc_video_engine.cc",
++ "engine/webrtc_voice_engine.cc",
++ ]
++ }
+
+ public_configs = []
+ if (!build_with_chromium) {
+diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
+index 9653fd651f..82b050a559 100644
+--- a/rtc_base/BUILD.gn
++++ b/rtc_base/BUILD.gn
+@@ -1533,6 +1533,7 @@ rtc_source_set("ssl_header") {
+ }
+
+ rtc_library("digest") {
++if (!build_with_mozilla) {
+ visibility = [ "*" ]
+ sources = [
+ "message_digest.cc",
+@@ -1554,8 +1555,10 @@ rtc_library("digest") {
+ configs += [ "..:external_ssl_library" ]
+ }
+ }
++}
+
+ rtc_library("crypto_random") {
++if (!build_with_mozilla) {
+ visibility = [ "*" ]
+ sources = [
+ "crypto_random.cc",
+@@ -1576,6 +1579,7 @@ rtc_library("crypto_random") {
+ configs += [ "..:external_ssl_library" ]
+ }
+ }
++}
+
+ rtc_library("ssl") {
+ if (!build_with_mozilla) {
+diff --git a/webrtc.gni b/webrtc.gni
+index 34a8ee592d..d29a86112b 100644
+--- a/webrtc.gni
++++ b/webrtc.gni
+@@ -116,7 +116,7 @@ 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 = "unused"
++ rtc_ssl_root = ""
+
+ # Enable when an external authentication mechanism is used for performing
+ # packet authentication for RTP packets instead of libsrtp.
+@@ -341,7 +341,7 @@ declare_args() {
+ }
+
+ # Enable liboam only on non-mozilla builds.
+-enable_libaom = !build_with_mozilla
++enable_libaom = true
+
+ # 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).
diff --git a/third_party/libwebrtc/moz-patch-stack/s0108.patch b/third_party/libwebrtc/moz-patch-stack/s0108.patch
@@ -1,170 +1,23 @@
-From: Nico Grunbaum <na-g@nostrum.com>
-Date: Wed, 27 Nov 2024 03:41:00 +0000
-Subject: Bug 1921154 - libwebrtc build related changes for
- AV1;r=mjf,webrtc-reviewers
+From: Michael Froman <mfroman@mozilla.com>
+Date: Mon, 13 Jan 2025 08:14:00 -0600
+Subject: Bug 1942428 - (fix-69cc695699) uncomment an arg that upstream thinks
+ is unused
-These are most of the changes needed to get a build with AV1.
-The crypto/openssl changes are later altered in the stack adding an NSPR based implementation of the functionality that was missing.
-
-Differential Revision: https://phabricator.services.mozilla.com/D228540
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/4527c41ef71d7683c60d9c435991a0d93bc5ce4d
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/73f9e0e64eb65e9f2579b0f3eb34b88e2b5e5d74
---
- BUILD.gn | 4 ++++
- media/BUILD.gn | 24 ++++++++++++++----------
- rtc_base/BUILD.gn | 4 ++++
- webrtc.gni | 4 ++--
- 4 files changed, 24 insertions(+), 12 deletions(-)
+ modules/rtp_rtcp/source/rtcp_sender.cc | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/BUILD.gn b/BUILD.gn
-index 2e315c79c4..174ce0d2f4 100644
---- a/BUILD.gn
-+++ b/BUILD.gn
-@@ -521,6 +521,9 @@ if (!rtc_build_ssl) {
- "crypto",
- "ssl",
- ]
-+ if (build_with_mozilla) {
-+ libs = []
-+ }
- }
- }
-
-@@ -615,6 +618,7 @@ if (!build_with_chromium) {
- "api/environment:environment_factory",
- "api/video:video_frame",
- "api/video:video_rtp_headers",
-+ "api/video_codecs:builtin_video_decoder_factory",
- "test:rtp_test_utils",
- ]
- # Added when we removed deps in other places to avoid building
-diff --git a/media/BUILD.gn b/media/BUILD.gn
-index e50b924806..ac722ddda3 100644
---- a/media/BUILD.gn
-+++ b/media/BUILD.gn
-@@ -12,12 +12,10 @@ import("../webrtc.gni")
-
- group("media") {
- deps = []
-- if (!build_with_mozilla) {
-- deps += [
-- ":rtc_media",
-- ":rtc_media_base",
-- ]
-- }
-+ deps += [
-+ ":rtc_media",
-+ ":rtc_media_base",
-+ ]
- }
-
- config("rtc_media_defines_config") {
-@@ -281,6 +279,9 @@ rtc_library("media_engine") {
- "base/media_engine.cc",
- "base/media_engine.h",
- ]
-+ deps -= [
-+ ":media_channel_impl",
-+ ]
- }
- }
-
-@@ -432,6 +433,7 @@ rtc_library("codec_list") {
- ]
- }
-
-+# MOZILLA Bug 1931241 - how much of this do we need?
- rtc_library("rtp_utils") {
- if (!build_with_mozilla) {
- sources = [
-@@ -478,7 +480,6 @@ rtc_library("media_constants") {
- }
-
- rtc_library("turn_utils") {
--if (!build_with_mozilla) {
- sources = [
- "base/turn_utils.cc",
- "base/turn_utils.h",
-@@ -489,17 +490,14 @@ if (!build_with_mozilla) {
- "../rtc_base/system:rtc_export",
- ]
- }
--}
-
- rtc_library("rid_description") {
--if (!build_with_mozilla) {
- sources = [
- "base/rid_description.cc",
- "base/rid_description.h",
- ]
- deps = [ ":codec" ]
- }
--}
-
- rtc_library("rtc_simulcast_encoder_adapter") {
- visibility = [ "*" ]
-@@ -717,6 +715,12 @@ rtc_library("rtc_audio_video") {
- "engine/webrtc_voice_engine.cc",
- "engine/webrtc_voice_engine.h",
- ]
-+ if (build_with_mozilla) {
-+ sources -= [
-+ "engine/webrtc_video_engine.cc",
-+ "engine/webrtc_voice_engine.cc",
-+ ]
-+ }
-
- public_configs = []
- if (!build_with_chromium) {
-diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
-index 9653fd651f..82b050a559 100644
---- a/rtc_base/BUILD.gn
-+++ b/rtc_base/BUILD.gn
-@@ -1533,6 +1533,7 @@ rtc_source_set("ssl_header") {
- }
-
- rtc_library("digest") {
-+if (!build_with_mozilla) {
- visibility = [ "*" ]
- sources = [
- "message_digest.cc",
-@@ -1554,8 +1555,10 @@ rtc_library("digest") {
- configs += [ "..:external_ssl_library" ]
- }
- }
-+}
-
- rtc_library("crypto_random") {
-+if (!build_with_mozilla) {
- visibility = [ "*" ]
- sources = [
- "crypto_random.cc",
-@@ -1576,6 +1579,7 @@ rtc_library("crypto_random") {
- configs += [ "..:external_ssl_library" ]
- }
- }
-+}
-
- rtc_library("ssl") {
- if (!build_with_mozilla) {
-diff --git a/webrtc.gni b/webrtc.gni
-index 34a8ee592d..d29a86112b 100644
---- a/webrtc.gni
-+++ b/webrtc.gni
-@@ -116,7 +116,7 @@ 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 = "unused"
-+ rtc_ssl_root = ""
-
- # Enable when an external authentication mechanism is used for performing
- # packet authentication for RTP packets instead of libsrtp.
-@@ -341,7 +341,7 @@ declare_args() {
- }
-
- # Enable liboam only on non-mozilla builds.
--enable_libaom = !build_with_mozilla
-+enable_libaom = true
-
- # 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).
+diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
+index 4f7ec4f751..14dcdd9cfa 100644
+--- a/modules/rtp_rtcp/source/rtcp_sender.cc
++++ b/modules/rtp_rtcp/source/rtcp_sender.cc
+@@ -215,7 +215,7 @@ bool RTCPSender::Sending() const {
+ return sending_;
+ }
+
+-void RTCPSender::SetSendingStatus(const FeedbackState& /* feedback_state */,
++void RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
+ bool sending) {
+ bool sendRTCPBye = false;
+ {
diff --git a/third_party/libwebrtc/moz-patch-stack/s0109.patch b/third_party/libwebrtc/moz-patch-stack/s0109.patch
@@ -1,23 +1,40 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Mon, 13 Jan 2025 08:14:00 -0600
-Subject: Bug 1942428 - (fix-69cc695699) uncomment an arg that upstream thinks
- is unused
+From: Andreas Pehrson <apehrson@mozilla.com>
+Date: Tue, 28 Jan 2025 21:45:00 +0000
+Subject: Bug 1938047 - Link adapted_video_track_source.cc and generate webrtc
+ moz.build files. r=ng,webrtc-reviewers
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/73f9e0e64eb65e9f2579b0f3eb34b88e2b5e5d74
+Differential Revision: https://phabricator.services.mozilla.com/D234863
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b6dd815fc9d2df718a1b48b96f58f99c0cfb7e19
---
- modules/rtp_rtcp/source/rtcp_sender.cc | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ media/BUILD.gn | 11 -----------
+ 1 file changed, 11 deletions(-)
-diff --git a/modules/rtp_rtcp/source/rtcp_sender.cc b/modules/rtp_rtcp/source/rtcp_sender.cc
-index 4f7ec4f751..14dcdd9cfa 100644
---- a/modules/rtp_rtcp/source/rtcp_sender.cc
-+++ b/modules/rtp_rtcp/source/rtcp_sender.cc
-@@ -215,7 +215,7 @@ bool RTCPSender::Sending() const {
- return sending_;
- }
-
--void RTCPSender::SetSendingStatus(const FeedbackState& /* feedback_state */,
-+void RTCPSender::SetSendingStatus(const FeedbackState& feedback_state,
- bool sending) {
- bool sendRTCPBye = false;
- {
+diff --git a/media/BUILD.gn b/media/BUILD.gn
+index ac722ddda3..31df4610c7 100644
+--- a/media/BUILD.gn
++++ b/media/BUILD.gn
+@@ -54,11 +54,6 @@ rtc_library("rtc_media_base") {
+ "base/video_broadcaster.h", # Used downstream
+ "base/video_common.h", # Used downstream
+ ]
+- if (build_with_mozilla) {
+- sources -= [
+- "base/adapted_video_track_source.h",
+- ]
+- }
+ deps = [
+ ":adapted_video_track_source",
+ ":codec",
+@@ -139,12 +134,6 @@ rtc_library("adapted_video_track_source") {
+ "base/adapted_video_track_source.cc",
+ "base/adapted_video_track_source.h",
+ ]
+- if (build_with_mozilla) {
+- sources -= [
+- "base/adapted_video_track_source.cc",
+- "base/adapted_video_track_source.h",
+- ]
+- }
+ deps = [
+ ":video_adapter",
+ ":video_broadcaster",
diff --git a/third_party/libwebrtc/moz-patch-stack/s0110.patch b/third_party/libwebrtc/moz-patch-stack/s0110.patch
@@ -1,40 +1,49 @@
From: Andreas Pehrson <apehrson@mozilla.com>
Date: Tue, 28 Jan 2025 21:45:00 +0000
-Subject: Bug 1938047 - Link adapted_video_track_source.cc and generate webrtc
- moz.build files. r=ng,webrtc-reviewers
+Subject: Bug 1938047 - Hook up VideoStreamFactory with EncoderInfo like the
+ libwebrtc factory. r=webrtc-reviewers,dbaker
-Differential Revision: https://phabricator.services.mozilla.com/D234863
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b6dd815fc9d2df718a1b48b96f58f99c0cfb7e19
+This is necessary for the factory to factor in the encoder's resolution
+alignment requirements when calculating stream resolutions.
+
+Differential Revision: https://phabricator.services.mozilla.com/D235702
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8b5653392e88ccd0c2b0675c761bb22733364225
---
- media/BUILD.gn | 11 -----------
- 1 file changed, 11 deletions(-)
+ video/config/video_encoder_config.h | 4 ++++
+ video/video_stream_encoder.cc | 1 +
+ 2 files changed, 5 insertions(+)
-diff --git a/media/BUILD.gn b/media/BUILD.gn
-index ac722ddda3..31df4610c7 100644
---- a/media/BUILD.gn
-+++ b/media/BUILD.gn
-@@ -54,11 +54,6 @@ rtc_library("rtc_media_base") {
- "base/video_broadcaster.h", # Used downstream
- "base/video_common.h", # Used downstream
- ]
-- if (build_with_mozilla) {
-- sources -= [
-- "base/adapted_video_track_source.h",
-- ]
-- }
- deps = [
- ":adapted_video_track_source",
- ":codec",
-@@ -139,12 +134,6 @@ rtc_library("adapted_video_track_source") {
- "base/adapted_video_track_source.cc",
- "base/adapted_video_track_source.h",
- ]
-- if (build_with_mozilla) {
-- sources -= [
-- "base/adapted_video_track_source.cc",
-- "base/adapted_video_track_source.h",
-- ]
-- }
- deps = [
- ":video_adapter",
- ":video_broadcaster",
+diff --git a/video/config/video_encoder_config.h b/video/config/video_encoder_config.h
+index 7fda03ffa8..8ca8462dd0 100644
+--- a/video/config/video_encoder_config.h
++++ b/video/config/video_encoder_config.h
+@@ -26,6 +26,7 @@
+ #include "api/video_codecs/sdp_video_format.h"
+ #include "api/video_codecs/spatial_layer.h"
+ #include "api/video_codecs/video_codec.h"
++#include "api/video_codecs/video_encoder.h"
+
+ namespace webrtc {
+
+@@ -153,6 +154,9 @@ class VideoEncoderConfig {
+ int frame_height,
+ const VideoEncoderConfig& encoder_config) = 0;
+
++ // Mozilla: Patching in necessary info our custom VideoStreamFactory will need to know.
++ virtual void SetEncoderInfo(const VideoEncoder::EncoderInfo& info) {}
++
+ protected:
+ ~VideoStreamFactoryInterface() override {}
+ };
+diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
+index fa0308f0f7..22d03dba57 100644
+--- a/video/video_stream_encoder.cc
++++ b/video/video_stream_encoder.cc
+@@ -1055,6 +1055,7 @@ void VideoStreamEncoder::ReconfigureEncoder() {
+ std::vector<VideoStream> streams;
+ if (encoder_config_.video_stream_factory) {
+ // Note: only tests set their own EncoderStreamFactory...
++ encoder_config_.video_stream_factory->SetEncoderInfo(encoder_->GetEncoderInfo());
+ streams = encoder_config_.video_stream_factory->CreateEncoderStreams(
+ env_.field_trials(), last_frame_info_->width, last_frame_info_->height,
+ encoder_config_);
diff --git a/third_party/libwebrtc/moz-patch-stack/s0111.patch b/third_party/libwebrtc/moz-patch-stack/s0111.patch
@@ -1,49 +1,23 @@
-From: Andreas Pehrson <apehrson@mozilla.com>
-Date: Tue, 28 Jan 2025 21:45:00 +0000
-Subject: Bug 1938047 - Hook up VideoStreamFactory with EncoderInfo like the
- libwebrtc factory. r=webrtc-reviewers,dbaker
+From: Michael Froman <mfroman@mozilla.com>
+Date: Wed, 5 Feb 2025 18:03:00 +0000
+Subject: Bug 1942171 - add builtin_audio_processing_builder to the libwebrtc
+ build. r=pehrsons,webrtc-reviewers
-This is necessary for the factory to factor in the encoder's resolution
-alignment requirements when calculating stream resolutions.
-
-Differential Revision: https://phabricator.services.mozilla.com/D235702
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8b5653392e88ccd0c2b0675c761bb22733364225
+Differential Revision: https://phabricator.services.mozilla.com/D234594
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/27c29807e6f46562d2b6bae4b7b06a5966290afe
---
- video/config/video_encoder_config.h | 4 ++++
- video/video_stream_encoder.cc | 1 +
- 2 files changed, 5 insertions(+)
+ BUILD.gn | 1 +
+ 1 file changed, 1 insertion(+)
-diff --git a/video/config/video_encoder_config.h b/video/config/video_encoder_config.h
-index 7fda03ffa8..8ca8462dd0 100644
---- a/video/config/video_encoder_config.h
-+++ b/video/config/video_encoder_config.h
-@@ -26,6 +26,7 @@
- #include "api/video_codecs/sdp_video_format.h"
- #include "api/video_codecs/spatial_layer.h"
- #include "api/video_codecs/video_codec.h"
-+#include "api/video_codecs/video_encoder.h"
-
- namespace webrtc {
-
-@@ -153,6 +154,9 @@ class VideoEncoderConfig {
- int frame_height,
- const VideoEncoderConfig& encoder_config) = 0;
+diff --git a/BUILD.gn b/BUILD.gn
+index 174ce0d2f4..600993950b 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -615,6 +615,7 @@ if (!build_with_chromium) {
-+ // Mozilla: Patching in necessary info our custom VideoStreamFactory will need to know.
-+ virtual void SetEncoderInfo(const VideoEncoder::EncoderInfo& info) {}
-+
- protected:
- ~VideoStreamFactoryInterface() override {}
- };
-diff --git a/video/video_stream_encoder.cc b/video/video_stream_encoder.cc
-index fa0308f0f7..22d03dba57 100644
---- a/video/video_stream_encoder.cc
-+++ b/video/video_stream_encoder.cc
-@@ -1055,6 +1055,7 @@ void VideoStreamEncoder::ReconfigureEncoder() {
- std::vector<VideoStream> streams;
- if (encoder_config_.video_stream_factory) {
- // Note: only tests set their own EncoderStreamFactory...
-+ encoder_config_.video_stream_factory->SetEncoderInfo(encoder_->GetEncoderInfo());
- streams = encoder_config_.video_stream_factory->CreateEncoderStreams(
- env_.field_trials(), last_frame_info_->width, last_frame_info_->height,
- encoder_config_);
+ if (build_with_mozilla) {
+ deps += [
++ "api/audio:builtin_audio_processing_builder",
+ "api/environment:environment_factory",
+ "api/video:video_frame",
+ "api/video:video_rtp_headers",
diff --git a/third_party/libwebrtc/moz-patch-stack/s0112.patch b/third_party/libwebrtc/moz-patch-stack/s0112.patch
@@ -1,23 +1,26 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Wed, 5 Feb 2025 18:03:00 +0000
-Subject: Bug 1942171 - add builtin_audio_processing_builder to the libwebrtc
- build. r=pehrsons,webrtc-reviewers
+Date: Fri, 7 Feb 2025 15:53:00 -0600
+Subject: Bug 1942428 - (fix-882b32d00f) fix
+ -Wtautological-unsigned-zero-compare in payload_type.h
-Differential Revision: https://phabricator.services.mozilla.com/D234594
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/27c29807e6f46562d2b6bae4b7b06a5966290afe
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f3ed04250e9ec1b22731ac690eb66bc96ff894d6
---
- BUILD.gn | 1 +
- 1 file changed, 1 insertion(+)
+ call/payload_type.h | 4 ++++
+ 1 file changed, 4 insertions(+)
-diff --git a/BUILD.gn b/BUILD.gn
-index 174ce0d2f4..600993950b 100644
---- a/BUILD.gn
-+++ b/BUILD.gn
-@@ -615,6 +615,7 @@ if (!build_with_chromium) {
-
- if (build_with_mozilla) {
- deps += [
-+ "api/audio:builtin_audio_processing_builder",
- "api/environment:environment_factory",
- "api/video:video_frame",
- "api/video:video_rtp_headers",
+diff --git a/call/payload_type.h b/call/payload_type.h
+index d91da50fcc..a5e120417e 100644
+--- a/call/payload_type.h
++++ b/call/payload_type.h
+@@ -33,7 +33,11 @@ class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
+ if (rtcp_mux && (id > 63 && id < 96)) {
+ return false;
+ }
++#if defined(WEBRTC_MOZILLA_BUILD)
++ return id <= 127; // id is unsigned, therefor always >= 0
++#else
+ return id >= 0 && id <= 127;
++#endif
+ }
+ template <typename Sink>
+ friend void AbslStringify(Sink& sink, const PayloadType pt) {
diff --git a/third_party/libwebrtc/moz-patch-stack/s0113.patch b/third_party/libwebrtc/moz-patch-stack/s0113.patch
@@ -1,26 +1,23 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Fri, 7 Feb 2025 15:53:00 -0600
-Subject: Bug 1942428 - (fix-882b32d00f) fix
- -Wtautological-unsigned-zero-compare in payload_type.h
+Date: Fri, 7 Mar 2025 17:43:00 -0600
+Subject: Bug 1952339 - (fix-68e7b00928) add default param to emplace() to fix
+ compile issue
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f3ed04250e9ec1b22731ac690eb66bc96ff894d6
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/26c916591791b13a3d8888066c22c5ba49c86c47
---
- call/payload_type.h | 4 ++++
- 1 file changed, 4 insertions(+)
+ call/rtp_config.cc | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/call/payload_type.h b/call/payload_type.h
-index d91da50fcc..a5e120417e 100644
---- a/call/payload_type.h
-+++ b/call/payload_type.h
-@@ -33,7 +33,11 @@ class PayloadType : public StrongAlias<class PayloadTypeTag, uint8_t> {
- if (rtcp_mux && (id > 63 && id < 96)) {
- return false;
- }
-+#if defined(WEBRTC_MOZILLA_BUILD)
-+ return id <= 127; // id is unsigned, therefor always >= 0
-+#else
- return id >= 0 && id <= 127;
-+#endif
+diff --git a/call/rtp_config.cc b/call/rtp_config.cc
+index 115912cd46..5e08467d1f 100644
+--- a/call/rtp_config.cc
++++ b/call/rtp_config.cc
+@@ -258,7 +258,7 @@ RtpStreamConfig RtpConfig::GetStreamConfig(size_t index) const {
+ stream_config.raw_payload = raw_payload;
+ if (!rtx.ssrcs.empty()) {
+ RTC_DCHECK_EQ(ssrcs.size(), rtx.ssrcs.size());
+- auto& stream_config_rtx = stream_config.rtx.emplace();
++ auto& stream_config_rtx = stream_config.rtx.emplace(RtpStreamConfig::Rtx());
+ stream_config_rtx.ssrc = rtx.ssrcs[index];
+ stream_config_rtx.payload_type = rtx.payload_type;
}
- template <typename Sink>
- friend void AbslStringify(Sink& sink, const PayloadType pt) {
diff --git a/third_party/libwebrtc/moz-patch-stack/s0114.patch b/third_party/libwebrtc/moz-patch-stack/s0114.patch
@@ -1,23 +1,63 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Fri, 7 Mar 2025 17:43:00 -0600
-Subject: Bug 1952339 - (fix-68e7b00928) add default param to emplace() to fix
- compile issue
+From: Jan Grulich <jgrulich@redhat.com>
+Date: Mon, 14 Apr 2025 16:37:00 +0000
+Subject: Bug 1954903 - Update libwebrtc GN configs r=webrtc-reviewers,mjf
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/26c916591791b13a3d8888066c22c5ba49c86c47
+Update GN config files for the special handling in gn_processor.py for
+gbm, drm and PipeWire libraries.
+
+Also remove libwebrtc/third_party/[gbm,drm,pipewire] libs for which we
+now have a special handling in gn_processor.py.
+
+Differential Revision: https://phabricator.services.mozilla.com/D242101
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/506d4efaf2dc29472623beff64cc2bd0d5555afd
---
- call/rtp_config.cc | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ modules/desktop_capture/BUILD.gn | 4 ++++
+ modules/portal/BUILD.gn | 4 +---
+ modules/video_capture/BUILD.gn | 2 +-
+ 3 files changed, 6 insertions(+), 4 deletions(-)
-diff --git a/call/rtp_config.cc b/call/rtp_config.cc
-index 115912cd46..5e08467d1f 100644
---- a/call/rtp_config.cc
-+++ b/call/rtp_config.cc
-@@ -258,7 +258,7 @@ RtpStreamConfig RtpConfig::GetStreamConfig(size_t index) const {
- stream_config.raw_payload = raw_payload;
- if (!rtx.ssrcs.empty()) {
- RTC_DCHECK_EQ(ssrcs.size(), rtx.ssrcs.size());
-- auto& stream_config_rtx = stream_config.rtx.emplace();
-+ auto& stream_config_rtx = stream_config.rtx.emplace(RtpStreamConfig::Rtx());
- stream_config_rtx.ssrc = rtx.ssrcs[index];
- stream_config_rtx.payload_type = rtx.payload_type;
+diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
+index 4b5acc5bdb..484a47894b 100644
+--- a/modules/desktop_capture/BUILD.gn
++++ b/modules/desktop_capture/BUILD.gn
+@@ -553,6 +553,10 @@ rtc_library("desktop_capture") {
+ "../../rtc_base:sanitizer",
+ "../portal",
+ ]
++
++ if (build_with_mozilla) {
++ defines += [ "WEBRTC_USE_PIPEWIRE" ]
++ }
+ }
+
+ if (rtc_enable_win_wgc) {
+diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
+index 72972d18e1..70b4739ad0 100644
+--- a/modules/portal/BUILD.gn
++++ b/modules/portal/BUILD.gn
+@@ -139,11 +139,9 @@ if (!build_with_mozilla) {
+ deps -= [ ":pipewire_stubs" ]
+ defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ]
+ public_deps = [
+- "../../third_party/pipewire",
+- "../../third_party/drm",
+- "../../third_party/gbm",
+ "../../third_party/libepoxy"
+ ]
++ defines += [ "WEBRTC_USE_PIPEWIRE" ]
+ }
}
+ }
+diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
+index 83c1b21260..e32e3ab37e 100644
+--- a/modules/video_capture/BUILD.gn
++++ b/modules/video_capture/BUILD.gn
+@@ -118,7 +118,7 @@ if (!build_with_chromium || is_linux || is_chromeos) {
+ ]
+ if (build_with_mozilla) {
+ configs -= [ "../portal:pipewire_base" ]
+- public_deps = [ "../../third_party/pipewire" ]
++ defines = [ "WEBRTC_USE_PIPEWIRE" ]
+ }
+ }
+ }
diff --git a/third_party/libwebrtc/moz-patch-stack/s0115.patch b/third_party/libwebrtc/moz-patch-stack/s0115.patch
@@ -1,63 +1,24 @@
-From: Jan Grulich <jgrulich@redhat.com>
-Date: Mon, 14 Apr 2025 16:37:00 +0000
-Subject: Bug 1954903 - Update libwebrtc GN configs r=webrtc-reviewers,mjf
+From: Michael Froman <mfroman@mozilla.com>
+Date: Tue, 13 May 2025 22:39:00 +0000
+Subject: Bug 1966238 - (fix-bf85c38fb3) fix build for our added 'pid' field in
+ DesktopCapturer::Source r?pehrsons!
-Update GN config files for the special handling in gn_processor.py for
-gbm, drm and PipeWire libraries.
-
-Also remove libwebrtc/third_party/[gbm,drm,pipewire] libs for which we
-now have a special handling in gn_processor.py.
-
-Differential Revision: https://phabricator.services.mozilla.com/D242101
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/506d4efaf2dc29472623beff64cc2bd0d5555afd
+Differential Revision: https://phabricator.services.mozilla.com/D249164
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b6b37905a574e7706d98eb683e6800c9525815e7
---
- modules/desktop_capture/BUILD.gn | 4 ++++
- modules/portal/BUILD.gn | 4 +---
- modules/video_capture/BUILD.gn | 2 +-
- 3 files changed, 6 insertions(+), 4 deletions(-)
+ modules/desktop_capture/mac/screen_capturer_sck.mm | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
-index 4b5acc5bdb..484a47894b 100644
---- a/modules/desktop_capture/BUILD.gn
-+++ b/modules/desktop_capture/BUILD.gn
-@@ -553,6 +553,10 @@ rtc_library("desktop_capture") {
- "../../rtc_base:sanitizer",
- "../portal",
- ]
-+
-+ if (build_with_mozilla) {
-+ defines += [ "WEBRTC_USE_PIPEWIRE" ]
-+ }
- }
-
- if (rtc_enable_win_wgc) {
-diff --git a/modules/portal/BUILD.gn b/modules/portal/BUILD.gn
-index 72972d18e1..70b4739ad0 100644
---- a/modules/portal/BUILD.gn
-+++ b/modules/portal/BUILD.gn
-@@ -139,11 +139,9 @@ if (!build_with_mozilla) {
- deps -= [ ":pipewire_stubs" ]
- defines -= [ "WEBRTC_DLOPEN_PIPEWIRE" ]
- public_deps = [
-- "../../third_party/pipewire",
-- "../../third_party/drm",
-- "../../third_party/gbm",
- "../../third_party/libepoxy"
- ]
-+ defines += [ "WEBRTC_USE_PIPEWIRE" ]
- }
+diff --git a/modules/desktop_capture/mac/screen_capturer_sck.mm b/modules/desktop_capture/mac/screen_capturer_sck.mm
+index 49c923e56b..0d0750f63e 100644
+--- a/modules/desktop_capture/mac/screen_capturer_sck.mm
++++ b/modules/desktop_capture/mac/screen_capturer_sck.mm
+@@ -444,7 +444,7 @@ bool ScreenCapturerSck::GetSourceList(SourceList* sources) {
+ sources->clear();
+ EnsurePickerHandle();
+ if (picker_handle_) {
+- sources->push_back({picker_handle_->Source(), std::string()});
++ sources->push_back({picker_handle_->Source(), 0, std::string()});
}
+ return true;
}
-diff --git a/modules/video_capture/BUILD.gn b/modules/video_capture/BUILD.gn
-index 83c1b21260..e32e3ab37e 100644
---- a/modules/video_capture/BUILD.gn
-+++ b/modules/video_capture/BUILD.gn
-@@ -118,7 +118,7 @@ if (!build_with_chromium || is_linux || is_chromeos) {
- ]
- if (build_with_mozilla) {
- configs -= [ "../portal:pipewire_base" ]
-- public_deps = [ "../../third_party/pipewire" ]
-+ defines = [ "WEBRTC_USE_PIPEWIRE" ]
- }
- }
- }
diff --git a/third_party/libwebrtc/moz-patch-stack/s0116.patch b/third_party/libwebrtc/moz-patch-stack/s0116.patch
@@ -1,24 +1,32 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Tue, 13 May 2025 22:39:00 +0000
-Subject: Bug 1966238 - (fix-bf85c38fb3) fix build for our added 'pid' field in
- DesktopCapturer::Source r?pehrsons!
+Date: Thu, 10 Jul 2025 20:03:00 +0000
+Subject: Bug 1975060 - improve build file generation regarding x11 usage.
+ r=dbaker
-Differential Revision: https://phabricator.services.mozilla.com/D249164
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b6b37905a574e7706d98eb683e6800c9525815e7
+Upstream will be removing use_x11. We override the use_x11 settings
+from within gn_processor.py anyway, so we can remove our hack in
+third_party/libwebrtc/webrtc.gni, and then make gn_processor.py modify
+ozone_platform_x11 rather than use_x11.
+
+Note: these changes produce identical moz.build files for libwebrtc
+and abseil-cpp.
+
+Differential Revision: https://phabricator.services.mozilla.com/D255713
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c8cf67108972d5caefbfbb66146e26e5358493c2
---
- modules/desktop_capture/mac/screen_capturer_sck.mm | 2 +-
+ webrtc.gni | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/modules/desktop_capture/mac/screen_capturer_sck.mm b/modules/desktop_capture/mac/screen_capturer_sck.mm
-index 49c923e56b..0d0750f63e 100644
---- a/modules/desktop_capture/mac/screen_capturer_sck.mm
-+++ b/modules/desktop_capture/mac/screen_capturer_sck.mm
-@@ -444,7 +444,7 @@ bool ScreenCapturerSck::GetSourceList(SourceList* sources) {
- sources->clear();
- EnsurePickerHandle();
- if (picker_handle_) {
-- sources->push_back({picker_handle_->Source(), std::string()});
-+ sources->push_back({picker_handle_->Source(), 0, std::string()});
- }
- return true;
- }
+diff --git a/webrtc.gni b/webrtc.gni
+index d29a86112b..ee377f06d0 100644
+--- a/webrtc.gni
++++ b/webrtc.gni
+@@ -136,7 +136,7 @@ declare_args() {
+ rtc_build_tools = false
+
+ # Set this to false to skip building code that requires X11.
+- rtc_use_x11 = use_x11
++ rtc_use_x11 = ozone_platform_x11
+
+ # Set this to use PipeWire on the Wayland display server.
+ # By default it's only enabled on desktop Linux (excludes ChromeOS) and
diff --git a/third_party/libwebrtc/moz-patch-stack/s0117.patch b/third_party/libwebrtc/moz-patch-stack/s0117.patch
@@ -1,32 +1,26 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Thu, 10 Jul 2025 20:03:00 +0000
-Subject: Bug 1975060 - improve build file generation regarding x11 usage.
- r=dbaker
+Date: Wed, 16 Jul 2025 21:08:00 +0300
+Subject: Bug 1974813 - turn off rust in libwebrtc since we don't use any
+ libwebrtc rust code. r=ng?
-Upstream will be removing use_x11. We override the use_x11 settings
-from within gn_processor.py anyway, so we can remove our hack in
-third_party/libwebrtc/webrtc.gni, and then make gn_processor.py modify
-ozone_platform_x11 rather than use_x11.
+Note: no moz.build changes are caused by this change.
-Note: these changes produce identical moz.build files for libwebrtc
-and abseil-cpp.
-
-Differential Revision: https://phabricator.services.mozilla.com/D255713
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/c8cf67108972d5caefbfbb66146e26e5358493c2
+Differential Revision: https://phabricator.services.mozilla.com/D257316
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/14ee98c4832a55bf32116c925540ab0f53e46cd3
---
- webrtc.gni | 2 +-
+ .gn | 2 +-
1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/webrtc.gni b/webrtc.gni
-index d29a86112b..ee377f06d0 100644
---- a/webrtc.gni
-+++ b/webrtc.gni
-@@ -136,7 +136,7 @@ declare_args() {
- rtc_build_tools = false
+diff --git a/.gn b/.gn
+index 4ee826ec58..0d50e80115 100644
+--- a/.gn
++++ b/.gn
+@@ -89,7 +89,7 @@ default_args = {
+ use_fuzztest_wrapper = false
- # Set this to false to skip building code that requires X11.
-- rtc_use_x11 = use_x11
-+ rtc_use_x11 = ozone_platform_x11
+ # Enable Rust in WebRTC
+- enable_rust = true
++ enable_rust = false # Mozilla
+ enable_rust_cxx = true
+ enable_chromium_prelude = true
- # Set this to use PipeWire on the Wayland display server.
- # By default it's only enabled on desktop Linux (excludes ChromeOS) and
diff --git a/third_party/libwebrtc/moz-patch-stack/s0118.patch b/third_party/libwebrtc/moz-patch-stack/s0118.patch
@@ -1,26 +1,69 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Wed, 16 Jul 2025 21:08:00 +0300
-Subject: Bug 1974813 - turn off rust in libwebrtc since we don't use any
- libwebrtc rust code. r=ng?
+Date: Fri, 18 Jul 2025 16:07:00 -0500
+Subject: Bug 1971201 - (fix-70fb834e8c) most all of libwebrtc has converted
+ from rtc:: to webrtc::
-Note: no moz.build changes are caused by this change.
-
-Differential Revision: https://phabricator.services.mozilla.com/D257316
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/14ee98c4832a55bf32116c925540ab0f53e46cd3
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d51a82971b9e755d5514572fcd8ddc557153c8ef
---
- .gn | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ modules/rtp_rtcp/source/rtp_header_extensions.h | 4 ++--
+ modules/video_capture/video_capture.h | 2 +-
+ modules/video_capture/video_capture_impl.cc | 2 +-
+ modules/video_capture/video_capture_impl.h | 2 +-
+ 4 files changed, 5 insertions(+), 5 deletions(-)
-diff --git a/.gn b/.gn
-index 4ee826ec58..0d50e80115 100644
---- a/.gn
-+++ b/.gn
-@@ -89,7 +89,7 @@ default_args = {
- use_fuzztest_wrapper = false
+diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
+index cc811cae3a..46d80e424a 100644
+--- a/modules/rtp_rtcp/source/rtp_header_extensions.h
++++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
+@@ -321,10 +321,10 @@ class CsrcAudioLevel {
+ static constexpr const char* kUri =
+ "urn:ietf:params:rtp-hdrext:csrc-audio-level";
+
+- static bool Parse(rtc::ArrayView<const uint8_t> data,
++ static bool Parse(webrtc::ArrayView<const uint8_t> data,
+ CsrcAudioLevelList* csrcAudioLevels);
+ static size_t ValueSize(const CsrcAudioLevelList& csrcAudioLevels);
+- static bool Write(rtc::ArrayView<uint8_t> data,
++ static bool Write(webrtc::ArrayView<uint8_t> data,
+ const CsrcAudioLevelList& csrcAudioLevels);
+ };
+ #endif
+diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
+index a7c1764c51..8a91ca2f83 100644
+--- a/modules/video_capture/video_capture.h
++++ b/modules/video_capture/video_capture.h
+@@ -123,7 +123,7 @@ class VideoCaptureModule : public RefCountInterface {
+
+ // Remove capture data callback
+ virtual void DeRegisterCaptureDataCallback(
+- rtc::VideoSinkInterface<VideoFrame> *dataCallback) = 0;
++ webrtc::VideoSinkInterface<VideoFrame> *dataCallback) = 0;
+
+ // Start capture device
+ virtual int32_t StartCapture(const VideoCaptureCapability& capability) = 0;
+diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
+index 62a4b21a5f..98d7081b1d 100644
+--- a/modules/video_capture/video_capture_impl.cc
++++ b/modules/video_capture/video_capture_impl.cc
+@@ -120,7 +120,7 @@ void VideoCaptureImpl::RegisterCaptureDataCallback(
+ }
- # Enable Rust in WebRTC
-- enable_rust = true
-+ enable_rust = false # Mozilla
- enable_rust_cxx = true
- enable_chromium_prelude = true
+ void VideoCaptureImpl::DeRegisterCaptureDataCallback(
+- rtc::VideoSinkInterface<VideoFrame>* dataCallBack) {
++ webrtc::VideoSinkInterface<VideoFrame>* dataCallBack) {
+ MutexLock lock(&api_lock_);
+ auto it = _dataCallBacks.find(dataCallBack);
+ if (it != _dataCallBacks.end()) {
+diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h
+index 4d5b83547f..220b8cff27 100644
+--- a/modules/video_capture/video_capture_impl.h
++++ b/modules/video_capture/video_capture_impl.h
+@@ -63,7 +63,7 @@ class RTC_EXPORT VideoCaptureImpl : public VideoCaptureModule {
+ virtual void RegisterCaptureDataCallback(
+ RawVideoSinkInterface* dataCallback) override;
+ void DeRegisterCaptureDataCallback(
+- rtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
++ webrtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
+ int32_t StopCaptureIfAllClientsClose() override;
+ int32_t SetCaptureRotation(VideoRotation rotation) override;
diff --git a/third_party/libwebrtc/moz-patch-stack/s0119.patch b/third_party/libwebrtc/moz-patch-stack/s0119.patch
@@ -1,69 +1,82 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Fri, 18 Jul 2025 16:07:00 -0500
-Subject: Bug 1971201 - (fix-70fb834e8c) most all of libwebrtc has converted
- from rtc:: to webrtc::
+From: Michael Froman <mjfroman@mac.com>
+Date: Mon, 15 Sep 2025 14:53:58 -0500
+Subject: Bug 1973648 - move applicable libwebrtc changes to our new
+ libwebrtc_overrides location. r=ng
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/d51a82971b9e755d5514572fcd8ddc557153c8ef
+Differential Revision: https://phabricator.services.mozilla.com/D258499
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/19164e58323d59c1f66f7fcf671b8d0b6366f708
---
- modules/rtp_rtcp/source/rtp_header_extensions.h | 4 ++--
- modules/video_capture/video_capture.h | 2 +-
- modules/video_capture/video_capture_impl.cc | 2 +-
- modules/video_capture/video_capture_impl.h | 2 +-
- 4 files changed, 5 insertions(+), 5 deletions(-)
+ call/call.cc | 13 -------------
+ call/call.h | 11 ++---------
+ modules/desktop_capture/BUILD.gn | 5 +++++
+ 3 files changed, 7 insertions(+), 22 deletions(-)
-diff --git a/modules/rtp_rtcp/source/rtp_header_extensions.h b/modules/rtp_rtcp/source/rtp_header_extensions.h
-index cc811cae3a..46d80e424a 100644
---- a/modules/rtp_rtcp/source/rtp_header_extensions.h
-+++ b/modules/rtp_rtcp/source/rtp_header_extensions.h
-@@ -321,10 +321,10 @@ class CsrcAudioLevel {
- static constexpr const char* kUri =
- "urn:ietf:params:rtp-hdrext:csrc-audio-level";
-
-- static bool Parse(rtc::ArrayView<const uint8_t> data,
-+ static bool Parse(webrtc::ArrayView<const uint8_t> data,
- CsrcAudioLevelList* csrcAudioLevels);
- static size_t ValueSize(const CsrcAudioLevelList& csrcAudioLevels);
-- static bool Write(rtc::ArrayView<uint8_t> data,
-+ static bool Write(webrtc::ArrayView<uint8_t> data,
- const CsrcAudioLevelList& csrcAudioLevels);
+diff --git a/call/call.cc b/call/call.cc
+index dc1e44995c..827991c4d1 100644
+--- a/call/call.cc
++++ b/call/call.cc
+@@ -514,19 +514,6 @@ class Call final : public webrtc::Call,
};
- #endif
-diff --git a/modules/video_capture/video_capture.h b/modules/video_capture/video_capture.h
-index a7c1764c51..8a91ca2f83 100644
---- a/modules/video_capture/video_capture.h
-+++ b/modules/video_capture/video_capture.h
-@@ -123,7 +123,7 @@ class VideoCaptureModule : public RefCountInterface {
+ } // namespace internal
+
+-std::string Call::Stats::ToString(int64_t time_ms) const {
+- char buf[1024];
+- SimpleStringBuilder ss(buf);
+- ss << "Call stats: " << time_ms << ", {";
+- ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
+- ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
+- ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
+- ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
+- ss << "rtt_ms: " << rtt_ms;
+- ss << '}';
+- return ss.str();
+-}
+-
+ std::unique_ptr<Call> Call::Create(CallConfig config) {
+ std::unique_ptr<RtpTransportControllerSendInterface> transport_send;
+ if (config.rtp_transport_controller_send_factory != nullptr) {
+diff --git a/call/call.h b/call/call.h
+index 87e1884583..420a754edd 100644
+--- a/call/call.h
++++ b/call/call.h
+@@ -25,6 +25,7 @@
+ #include "api/transport/bitrate_settings.h"
+ #include "call/audio_receive_stream.h"
+ #include "call/audio_send_stream.h"
++#include "call/call_basic_stats.h"
+ #include "call/call_config.h"
+ #include "call/flexfec_receive_stream.h"
+ #include "call/packet_receiver.h"
+@@ -50,15 +51,7 @@ namespace webrtc {
- // Remove capture data callback
- virtual void DeRegisterCaptureDataCallback(
-- rtc::VideoSinkInterface<VideoFrame> *dataCallback) = 0;
-+ webrtc::VideoSinkInterface<VideoFrame> *dataCallback) = 0;
+ class Call {
+ public:
+- struct Stats {
+- std::string ToString(int64_t time_ms) const;
+-
+- int send_bandwidth_bps = 0; // Estimated available send bandwidth.
+- int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
+- int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
+- int64_t pacer_delay_ms = 0;
+- int64_t rtt_ms = -1;
+- };
++ using Stats = CallBasicStats;
- // Start capture device
- virtual int32_t StartCapture(const VideoCaptureCapability& capability) = 0;
-diff --git a/modules/video_capture/video_capture_impl.cc b/modules/video_capture/video_capture_impl.cc
-index 62a4b21a5f..98d7081b1d 100644
---- a/modules/video_capture/video_capture_impl.cc
-+++ b/modules/video_capture/video_capture_impl.cc
-@@ -120,7 +120,7 @@ void VideoCaptureImpl::RegisterCaptureDataCallback(
- }
+ static std::unique_ptr<Call> Create(CallConfig config);
- void VideoCaptureImpl::DeRegisterCaptureDataCallback(
-- rtc::VideoSinkInterface<VideoFrame>* dataCallBack) {
-+ webrtc::VideoSinkInterface<VideoFrame>* dataCallBack) {
- MutexLock lock(&api_lock_);
- auto it = _dataCallBacks.find(dataCallBack);
- if (it != _dataCallBacks.end()) {
-diff --git a/modules/video_capture/video_capture_impl.h b/modules/video_capture/video_capture_impl.h
-index 4d5b83547f..220b8cff27 100644
---- a/modules/video_capture/video_capture_impl.h
-+++ b/modules/video_capture/video_capture_impl.h
-@@ -63,7 +63,7 @@ class RTC_EXPORT VideoCaptureImpl : public VideoCaptureModule {
- virtual void RegisterCaptureDataCallback(
- RawVideoSinkInterface* dataCallback) override;
- void DeRegisterCaptureDataCallback(
-- rtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
-+ webrtc::VideoSinkInterface<VideoFrame>* dataCallback) override;
+diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
+index 484a47894b..5c843cfc2b 100644
+--- a/modules/desktop_capture/BUILD.gn
++++ b/modules/desktop_capture/BUILD.gn
+@@ -390,6 +390,11 @@ rtc_library("desktop_capture") {
+ ]
+ if (build_with_mozilla) {
+ libs -= [ "Xtst" ]
++ # Mozilla provides custom impl of x_error_trap.*
++ sources -= [
++ "linux/x11/x_error_trap.cc",
++ "linux/x11/x_error_trap.h",
++ ]
+ }
+ }
- int32_t StopCaptureIfAllClientsClose() override;
- int32_t SetCaptureRotation(VideoRotation rotation) override;
diff --git a/third_party/libwebrtc/moz-patch-stack/s0120.patch b/third_party/libwebrtc/moz-patch-stack/s0120.patch
@@ -1,82 +1,29 @@
-From: Michael Froman <mjfroman@mac.com>
-Date: Mon, 15 Sep 2025 14:53:58 -0500
-Subject: Bug 1973648 - move applicable libwebrtc changes to our new
- libwebrtc_overrides location. r=ng
+From: Michael Froman <mfroman@mozilla.com>
+Date: Thu, 7 Aug 2025 11:37:00 -0500
+Subject: Bug 1980479 - (fix-50544d8dc4) add missing stub implementations for
+ SSLStreamAdapter static methods.
-Differential Revision: https://phabricator.services.mozilla.com/D258499
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/19164e58323d59c1f66f7fcf671b8d0b6366f708
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8fee7e6d9736f8d1dfe78e9bc2c4676ca86d0405
---
- call/call.cc | 13 -------------
- call/call.h | 11 ++---------
- modules/desktop_capture/BUILD.gn | 5 +++++
- 3 files changed, 7 insertions(+), 22 deletions(-)
+ rtc_base/ssl_stream_adapter.h | 5 +++--
+ 1 file changed, 3 insertions(+), 2 deletions(-)
-diff --git a/call/call.cc b/call/call.cc
-index dc1e44995c..827991c4d1 100644
---- a/call/call.cc
-+++ b/call/call.cc
-@@ -514,19 +514,6 @@ class Call final : public webrtc::Call,
- };
- } // namespace internal
+diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h
+index ca2f6c48c4..24fc299b82 100644
+--- a/rtc_base/ssl_stream_adapter.h
++++ b/rtc_base/ssl_stream_adapter.h
+@@ -244,11 +244,12 @@ class SSLStreamAdapter : public StreamInterface {
+ static bool IsAcceptableCipher(int cipher, KeyType key_type);
+ static bool IsAcceptableCipher(absl::string_view cipher, KeyType key_type);
--std::string Call::Stats::ToString(int64_t time_ms) const {
-- char buf[1024];
-- SimpleStringBuilder ss(buf);
-- ss << "Call stats: " << time_ms << ", {";
-- ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
-- ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
-- ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
-- ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
-- ss << "rtt_ms: " << rtt_ms;
-- ss << '}';
-- return ss.str();
--}
--
- std::unique_ptr<Call> Call::Create(CallConfig config) {
- std::unique_ptr<RtpTransportControllerSendInterface> transport_send;
- if (config.rtp_transport_controller_send_factory != nullptr) {
-diff --git a/call/call.h b/call/call.h
-index 87e1884583..420a754edd 100644
---- a/call/call.h
-+++ b/call/call.h
-@@ -25,6 +25,7 @@
- #include "api/transport/bitrate_settings.h"
- #include "call/audio_receive_stream.h"
- #include "call/audio_send_stream.h"
-+#include "call/call_basic_stats.h"
- #include "call/call_config.h"
- #include "call/flexfec_receive_stream.h"
- #include "call/packet_receiver.h"
-@@ -50,15 +51,7 @@ namespace webrtc {
-
- class Call {
- public:
-- struct Stats {
-- std::string ToString(int64_t time_ms) const;
--
-- int send_bandwidth_bps = 0; // Estimated available send bandwidth.
-- int max_padding_bitrate_bps = 0; // Cumulative configured max padding.
-- int recv_bandwidth_bps = 0; // Estimated available receive bandwidth.
-- int64_t pacer_delay_ms = 0;
-- int64_t rtt_ms = -1;
-- };
-+ using Stats = CallBasicStats;
-
- static std::unique_ptr<Call> Create(CallConfig config);
-
-diff --git a/modules/desktop_capture/BUILD.gn b/modules/desktop_capture/BUILD.gn
-index 484a47894b..5c843cfc2b 100644
---- a/modules/desktop_capture/BUILD.gn
-+++ b/modules/desktop_capture/BUILD.gn
-@@ -390,6 +390,11 @@ rtc_library("desktop_capture") {
- ]
- if (build_with_mozilla) {
- libs -= [ "Xtst" ]
-+ # Mozilla provides custom impl of x_error_trap.*
-+ sources -= [
-+ "linux/x11/x_error_trap.cc",
-+ "linux/x11/x_error_trap.h",
-+ ]
- }
- }
+- static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups();
++ static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups()
++ { return {}; }
+ static std::optional<std::string> GetEphemeralKeyExchangeCipherGroupName(
+ uint16_t group_id);
+ static std::vector<uint16_t> GetDefaultEphemeralKeyExchangeCipherGroups(
+- const FieldTrialsView* field_trials);
++ const FieldTrialsView* field_trials) { return {}; }
+ ////////////////////////////////////////////////////////////////////////////
+ // Testing only member functions
diff --git a/third_party/libwebrtc/moz-patch-stack/s0121.patch b/third_party/libwebrtc/moz-patch-stack/s0121.patch
@@ -1,29 +1,41 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Thu, 7 Aug 2025 11:37:00 -0500
-Subject: Bug 1980479 - (fix-50544d8dc4) add missing stub implementations for
- SSLStreamAdapter static methods.
+Date: Thu, 11 Sep 2025 14:17:00 -0500
+Subject: Bug 1988042 - add BSD support to new version of libwebrtc's
+ cpu_info.cc r?ng!
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8fee7e6d9736f8d1dfe78e9bc2c4676ca86d0405
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e3b37acb8816aefb9134a68cafad8c7d5e145179
---
- rtc_base/ssl_stream_adapter.h | 5 +++--
- 1 file changed, 3 insertions(+), 2 deletions(-)
+ rtc_base/cpu_info.cc | 7 +++++--
+ 1 file changed, 5 insertions(+), 2 deletions(-)
-diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h
-index ca2f6c48c4..24fc299b82 100644
---- a/rtc_base/ssl_stream_adapter.h
-+++ b/rtc_base/ssl_stream_adapter.h
-@@ -244,11 +244,12 @@ class SSLStreamAdapter : public StreamInterface {
- static bool IsAcceptableCipher(int cipher, KeyType key_type);
- static bool IsAcceptableCipher(absl::string_view cipher, KeyType key_type);
+diff --git a/rtc_base/cpu_info.cc b/rtc_base/cpu_info.cc
+index 0ce48a02f1..f0bff56911 100644
+--- a/rtc_base/cpu_info.cc
++++ b/rtc_base/cpu_info.cc
+@@ -29,14 +29,14 @@
+ #include <unistd.h>
+ #elif defined(WEBRTC_FUCHSIA)
+ #include <zircon/syscalls.h>
+-#elif defined(WEBRTC_LINUX)
++#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
+ #include <unistd.h>
+ #endif // WEBRTC_LINUX
-- static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups();
-+ static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups()
-+ { return {}; }
- static std::optional<std::string> GetEphemeralKeyExchangeCipherGroupName(
- uint16_t group_id);
- static std::vector<uint16_t> GetDefaultEphemeralKeyExchangeCipherGroups(
-- const FieldTrialsView* field_trials);
-+ const FieldTrialsView* field_trials) { return {}; }
-
- ////////////////////////////////////////////////////////////////////////////
- // Testing only member functions
+ #if defined(WEBRTC_ARCH_X86_FAMILY) && defined(_MSC_VER)
+ #include <intrin.h>
+ #endif
+-#if defined(WEBRTC_ARCH_ARM_FAMILY) && defined(WEBRTC_LINUX)
++#if defined(WEBRTC_ARCH_ARM_FAMILY) && (defined(WEBRTC_LINUX) || defined(WEBRTC_BSD))
+ #include <asm/hwcap.h>
+ #include <sys/auxv.h>
+ #endif
+@@ -176,6 +176,9 @@ bool Supports(ISA instruction_set_architecture) {
+ if (instruction_set_architecture == ISA::kNeon) {
+ #if defined(WEBRTC_ANDROID)
+ return 0 != (android_getCpuFeatures() & ANDROID_CPU_ARM_FEATURE_NEON);
++#elif defined(WEBRTC_BSD)
++ // OpenBSD does not have getauxval() and does not have /proc reading
++ return false;
+ #elif defined(WEBRTC_LINUX)
+ uint64_t hwcap = 0;
+ hwcap = getauxval(AT_HWCAP);
diff --git a/third_party/libwebrtc/moz-patch-stack/s0122.patch b/third_party/libwebrtc/moz-patch-stack/s0122.patch
@@ -1,41 +1,42 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Thu, 11 Sep 2025 14:17:00 -0500
-Subject: Bug 1988042 - add BSD support to new version of libwebrtc's
- cpu_info.cc r?ng!
+From: Michael Froman <mjfroman@mac.com>
+Date: Tue, 16 Sep 2025 09:26:22 -0500
+Subject: Bug 1985396 - Cherry-pick upstream libwebrtc commit 73898ac18b
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/e3b37acb8816aefb9134a68cafad8c7d5e145179
+Upstream commit: https://webrtc.googlesource.com/src/+/73898ac18b498c3322f71b9543a2b77f465a47af
+ APM data dumpers: add missing include
+
+ Currently, audioproc_f fails to build with apm_debug_dump enabled.
+
+ Bug: webrtc:442745059
+ Change-Id: I349468891d798341abdaea32042cbe23c4722f75
+ Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407641
+ Commit-Queue: Sam Zackrisson <saza@webrtc.org>
+ Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
+ Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
+ Cr-Commit-Position: refs/heads/main@{#45547}
---
- rtc_base/cpu_info.cc | 7 +++++--
- 1 file changed, 5 insertions(+), 2 deletions(-)
+ modules/audio_processing/logging/apm_data_dumper.cc | 5 ++++-
+ 1 file changed, 4 insertions(+), 1 deletion(-)
-diff --git a/rtc_base/cpu_info.cc b/rtc_base/cpu_info.cc
-index 0ce48a02f1..f0bff56911 100644
---- a/rtc_base/cpu_info.cc
-+++ b/rtc_base/cpu_info.cc
-@@ -29,14 +29,14 @@
- #include <unistd.h>
- #elif defined(WEBRTC_FUCHSIA)
- #include <zircon/syscalls.h>
--#elif defined(WEBRTC_LINUX)
-+#elif defined(WEBRTC_LINUX) || defined(WEBRTC_BSD)
- #include <unistd.h>
- #endif // WEBRTC_LINUX
+diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
+index b26d90fbc8..d86d8d8d3b 100644
+--- a/modules/audio_processing/logging/apm_data_dumper.cc
++++ b/modules/audio_processing/logging/apm_data_dumper.cc
+@@ -10,13 +10,16 @@
- #if defined(WEBRTC_ARCH_X86_FAMILY) && defined(_MSC_VER)
- #include <intrin.h>
- #endif
--#if defined(WEBRTC_ARCH_ARM_FAMILY) && defined(WEBRTC_LINUX)
-+#if defined(WEBRTC_ARCH_ARM_FAMILY) && (defined(WEBRTC_LINUX) || defined(WEBRTC_BSD))
- #include <asm/hwcap.h>
- #include <sys/auxv.h>
+ #include "modules/audio_processing/logging/apm_data_dumper.h"
+
+-
+ // Check to verify that the define is properly set.
+ #if !defined(WEBRTC_APM_DEBUG_DUMP) || \
+ (WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
+ #error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
#endif
-@@ -176,6 +176,9 @@ bool Supports(ISA instruction_set_architecture) {
- if (instruction_set_architecture == ISA::kNeon) {
- #if defined(WEBRTC_ANDROID)
- return 0 != (android_getCpuFeatures() & ANDROID_CPU_ARM_FEATURE_NEON);
-+#elif defined(WEBRTC_BSD)
-+ // OpenBSD does not have getauxval() and does not have /proc reading
-+ return false;
- #elif defined(WEBRTC_LINUX)
- uint64_t hwcap = 0;
- hwcap = getauxval(AT_HWCAP);
+
++#if WEBRTC_APM_DEBUG_DUMP == 1
++#include "rtc_base/strings/string_builder.h"
++#endif
++
+ namespace webrtc {
+ namespace {
+
diff --git a/third_party/libwebrtc/moz-patch-stack/s0123.patch b/third_party/libwebrtc/moz-patch-stack/s0123.patch
@@ -1,42 +1,68 @@
-From: Michael Froman <mjfroman@mac.com>
-Date: Tue, 16 Sep 2025 09:26:22 -0500
-Subject: Bug 1985396 - Cherry-pick upstream libwebrtc commit 73898ac18b
+From: Michael Froman <mfroman@mozilla.com>
+Date: Thu, 18 Sep 2025 12:09:00 -0500
+Subject: Bug 1985396 - Cherry-pick upstream libwebrtc commit 63a59de808
-Upstream commit: https://webrtc.googlesource.com/src/+/73898ac18b498c3322f71b9543a2b77f465a47af
- APM data dumpers: add missing include
+Upstream commit: https://webrtc.googlesource.com/src/+/63a59de808ba7669e60bded703e71f360e5337e0
+ dcsctp: Avoid iterator invalidation
- Currently, audioproc_f fails to build with apm_debug_dump enabled.
+ The ExpireOutstandingChunks method iterated through the
+ `outstanding_data_` container while conditionally calling
+ `AbandonAllFor`. The `AbandonAllFor` method can, in some cases, add new
+ elements to `outstanding_data_`, which invalidates the iterators used by
+ the loop in `ExpireOutstandingChunks`. This could lead to undefined
+ behavior.
- Bug: webrtc:442745059
- Change-Id: I349468891d798341abdaea32042cbe23c4722f75
- Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/407641
- Commit-Queue: Sam Zackrisson <saza@webrtc.org>
- Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
- Reviewed-by: Lionel Koenig <lionelk@webrtc.org>
- Cr-Commit-Position: refs/heads/main@{#45547}
+ This aligns the C++ implementation to the Rust dcsctp port, which
+ handles this correctly by using a two-pass approach.
+
+ Bug: chromium:443213199
+ Change-Id: Ib5ac73b743c5ed900a69806088780c213536a4a6
+ Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408002
+ Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
+ Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
+ Auto-Submit: Victor Boivie <boivie@webrtc.org>
+ Cr-Commit-Position: refs/heads/main@{#45566}
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b85c57c2d6f0604075bf639a9a7342ece543cdf1
---
- modules/audio_processing/logging/apm_data_dumper.cc | 5 ++++-
- 1 file changed, 4 insertions(+), 1 deletion(-)
+ net/dcsctp/tx/outstanding_data.cc | 16 ++++++++++++----
+ 1 file changed, 12 insertions(+), 4 deletions(-)
-diff --git a/modules/audio_processing/logging/apm_data_dumper.cc b/modules/audio_processing/logging/apm_data_dumper.cc
-index b26d90fbc8..d86d8d8d3b 100644
---- a/modules/audio_processing/logging/apm_data_dumper.cc
-+++ b/modules/audio_processing/logging/apm_data_dumper.cc
-@@ -10,13 +10,16 @@
-
- #include "modules/audio_processing/logging/apm_data_dumper.h"
-
--
- // Check to verify that the define is properly set.
- #if !defined(WEBRTC_APM_DEBUG_DUMP) || \
- (WEBRTC_APM_DEBUG_DUMP != 0 && WEBRTC_APM_DEBUG_DUMP != 1)
- #error "Set WEBRTC_APM_DEBUG_DUMP to either 0 or 1"
- #endif
+diff --git a/net/dcsctp/tx/outstanding_data.cc b/net/dcsctp/tx/outstanding_data.cc
+index 074e3c0461..63e750b5fc 100644
+--- a/net/dcsctp/tx/outstanding_data.cc
++++ b/net/dcsctp/tx/outstanding_data.cc
+@@ -402,6 +402,7 @@ std::vector<std::pair<TSN, Data>> OutstandingData::GetChunksToBeRetransmitted(
+ }
-+#if WEBRTC_APM_DEBUG_DUMP == 1
-+#include "rtc_base/strings/string_builder.h"
-+#endif
+ void OutstandingData::ExpireOutstandingChunks(Timestamp now) {
++ std::vector<UnwrappedTSN> tsns_to_expire;
+ UnwrappedTSN tsn = last_cumulative_tsn_ack_;
+ for (const Item& item : outstanding_data_) {
+ tsn.Increment();
+@@ -411,15 +412,22 @@ void OutstandingData::ExpireOutstandingChunks(Timestamp now) {
+ if (item.is_abandoned()) {
+ // Already abandoned.
+ } else if (item.is_nacked() && item.has_expired(now)) {
+- RTC_DLOG(LS_VERBOSE) << "Marking nacked chunk " << *tsn.Wrap()
+- << " and message " << *item.data().mid
+- << " as expired";
+- AbandonAllFor(item);
++ tsns_to_expire.push_back(tsn);
+ } else {
+ // A non-expired chunk. No need to iterate any further.
+ break;
+ }
+ }
+
- namespace webrtc {
- namespace {
++ for (UnwrappedTSN tsn_to_expire : tsns_to_expire) {
++ // The item is retrieved by TSN, as AbandonAllFor may have modified
++ // `outstanding_data_` and invalidated iterators from the first loop.
++ Item& item = GetItem(tsn_to_expire);
++ RTC_DLOG(LS_WARNING) << "Marking nacked chunk " << *tsn_to_expire.Wrap()
++ << " and message " << *item.data().mid
++ << " as expired";
++ AbandonAllFor(item);
++ }
+ RTC_DCHECK(IsConsistent());
+ }
diff --git a/third_party/libwebrtc/moz-patch-stack/s0124.patch b/third_party/libwebrtc/moz-patch-stack/s0124.patch
@@ -1,68 +1,44 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Thu, 18 Sep 2025 12:09:00 -0500
-Subject: Bug 1985396 - Cherry-pick upstream libwebrtc commit 63a59de808
+Date: Wed, 17 Sep 2025 15:41:00 +0000
+Subject: Bug 1988042 - android build fix in
+ third_party/libwebrtc/rtc_base/cpu_info.cc r?ng!
-Upstream commit: https://webrtc.googlesource.com/src/+/63a59de808ba7669e60bded703e71f360e5337e0
- dcsctp: Avoid iterator invalidation
+Supports(...) is really used in unittests and outside of unittests
+it is only used when WEBRTC_ARCH_X86_FAMILY is defined. For now,
+we'll return false when checking for arm/neon suport.
- The ExpireOutstandingChunks method iterated through the
- `outstanding_data_` container while conditionally calling
- `AbandonAllFor`. The `AbandonAllFor` method can, in some cases, add new
- elements to `outstanding_data_`, which invalidates the iterators used by
- the loop in `ExpireOutstandingChunks`. This could lead to undefined
- behavior.
-
- This aligns the C++ implementation to the Rust dcsctp port, which
- handles this correctly by using a two-pass approach.
-
- Bug: chromium:443213199
- Change-Id: Ib5ac73b743c5ed900a69806088780c213536a4a6
- Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408002
- Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
- Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
- Auto-Submit: Victor Boivie <boivie@webrtc.org>
- Cr-Commit-Position: refs/heads/main@{#45566}
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/b85c57c2d6f0604075bf639a9a7342ece543cdf1
+Differential Revision: https://phabricator.services.mozilla.com/D265018
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/eae7b6ec4f7ca191e8eaa8e36787142528f2c132
---
- net/dcsctp/tx/outstanding_data.cc | 16 ++++++++++++----
- 1 file changed, 12 insertions(+), 4 deletions(-)
+ rtc_base/cpu_info.cc | 9 +++++++++
+ 1 file changed, 9 insertions(+)
-diff --git a/net/dcsctp/tx/outstanding_data.cc b/net/dcsctp/tx/outstanding_data.cc
-index 074e3c0461..63e750b5fc 100644
---- a/net/dcsctp/tx/outstanding_data.cc
-+++ b/net/dcsctp/tx/outstanding_data.cc
-@@ -402,6 +402,7 @@ std::vector<std::pair<TSN, Data>> OutstandingData::GetChunksToBeRetransmitted(
- }
-
- void OutstandingData::ExpireOutstandingChunks(Timestamp now) {
-+ std::vector<UnwrappedTSN> tsns_to_expire;
- UnwrappedTSN tsn = last_cumulative_tsn_ack_;
- for (const Item& item : outstanding_data_) {
- tsn.Increment();
-@@ -411,15 +412,22 @@ void OutstandingData::ExpireOutstandingChunks(Timestamp now) {
- if (item.is_abandoned()) {
- // Already abandoned.
- } else if (item.is_nacked() && item.has_expired(now)) {
-- RTC_DLOG(LS_VERBOSE) << "Marking nacked chunk " << *tsn.Wrap()
-- << " and message " << *item.data().mid
-- << " as expired";
-- AbandonAllFor(item);
-+ tsns_to_expire.push_back(tsn);
- } else {
- // A non-expired chunk. No need to iterate any further.
- break;
- }
- }
-+
-+ for (UnwrappedTSN tsn_to_expire : tsns_to_expire) {
-+ // The item is retrieved by TSN, as AbandonAllFor may have modified
-+ // `outstanding_data_` and invalidated iterators from the first loop.
-+ Item& item = GetItem(tsn_to_expire);
-+ RTC_DLOG(LS_WARNING) << "Marking nacked chunk " << *tsn_to_expire.Wrap()
-+ << " and message " << *item.data().mid
-+ << " as expired";
-+ AbandonAllFor(item);
-+ }
- RTC_DCHECK(IsConsistent());
- }
-
+diff --git a/rtc_base/cpu_info.cc b/rtc_base/cpu_info.cc
+index f0bff56911..b51c9e950e 100644
+--- a/rtc_base/cpu_info.cc
++++ b/rtc_base/cpu_info.cc
+@@ -25,7 +25,9 @@
+ #elif defined(WEBRTC_MAC)
+ #include <sys/sysctl.h>
+ #elif defined(WEBRTC_ANDROID)
++#if !defined(WEBRTC_MOZILLA_BUILD)
+ #include <cpu-features.h>
++#endif
+ #include <unistd.h>
+ #elif defined(WEBRTC_FUCHSIA)
+ #include <zircon/syscalls.h>
+@@ -175,7 +177,14 @@ bool Supports(ISA instruction_set_architecture) {
+ #elif defined(WEBRTC_ARCH_ARM_FAMILY)
+ if (instruction_set_architecture == ISA::kNeon) {
+ #if defined(WEBRTC_ANDROID)
++#if defined(WEBRTC_MOZILLA_BUILD)
++ // We should revisit this if Supports(...) is ever more widely used.
++ // Currently, it is used in unittests and when WEBRTC_ARCH_X86_FAMILY
++ // is defined in non-unittest code.
++ return false;
++#else
+ return 0 != (android_getCpuFeatures() & ANDROID_CPU_ARM_FEATURE_NEON);
++#endif
+ #elif defined(WEBRTC_BSD)
+ // OpenBSD does not have getauxval() and does not have /proc reading
+ return false;
diff --git a/third_party/libwebrtc/moz-patch-stack/s0125.patch b/third_party/libwebrtc/moz-patch-stack/s0125.patch
@@ -1,44 +1,39 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Wed, 17 Sep 2025 15:41:00 +0000
-Subject: Bug 1988042 - android build fix in
- third_party/libwebrtc/rtc_base/cpu_info.cc r?ng!
+Date: Thu, 18 Sep 2025 22:33:00 -0500
+Subject: Bug 1985396 - (fix-38c5a66082) fix windows build failure caused by a
+ macro expansion
-Supports(...) is really used in unittests and outside of unittests
-it is only used when WEBRTC_ARCH_X86_FAMILY is defined. For now,
-we'll return false when checking for arm/neon suport.
+Commit 38c5a66082 adds the include rtc_base/strings/string_builder.h to
+third_party/libwebrtc/modules/desktop_capture/desktop_geometry.h, but we
+pick this up through a chain of includes in CamerasParent.cpp. On Windows
+builds we then get a failure because Shlwapi.h has a macro that expands
+StrCat to StrCatW, which means we're changing absl::StrCat to
+absl::StrCatW which doesn't exist.
-Differential Revision: https://phabricator.services.mozilla.com/D265018
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/eae7b6ec4f7ca191e8eaa8e36787142528f2c132
+We can avoid the desktop_geometry.h include and forward declare
+"class DesktopVector" to work around this issue.
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9541460d001e1a62ae567a18d810e35e50ac791c
---
- rtc_base/cpu_info.cc | 9 +++++++++
- 1 file changed, 9 insertions(+)
+ modules/desktop_capture/desktop_capturer.h | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/rtc_base/cpu_info.cc b/rtc_base/cpu_info.cc
-index f0bff56911..b51c9e950e 100644
---- a/rtc_base/cpu_info.cc
-+++ b/rtc_base/cpu_info.cc
-@@ -25,7 +25,9 @@
- #elif defined(WEBRTC_MAC)
- #include <sys/sysctl.h>
- #elif defined(WEBRTC_ANDROID)
-+#if !defined(WEBRTC_MOZILLA_BUILD)
- #include <cpu-features.h>
-+#endif
- #include <unistd.h>
- #elif defined(WEBRTC_FUCHSIA)
- #include <zircon/syscalls.h>
-@@ -175,7 +177,14 @@ bool Supports(ISA instruction_set_architecture) {
- #elif defined(WEBRTC_ARCH_ARM_FAMILY)
- if (instruction_set_architecture == ISA::kNeon) {
- #if defined(WEBRTC_ANDROID)
-+#if defined(WEBRTC_MOZILLA_BUILD)
-+ // We should revisit this if Supports(...) is ever more widely used.
-+ // Currently, it is used in unittests and when WEBRTC_ARCH_X86_FAMILY
-+ // is defined in non-unittest code.
-+ return false;
-+#else
- return 0 != (android_getCpuFeatures() & ANDROID_CPU_ARM_FEATURE_NEON);
-+#endif
- #elif defined(WEBRTC_BSD)
- // OpenBSD does not have getauxval() and does not have /proc reading
- return false;
+diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
+index c85d0eba7a..8219358e1e 100644
+--- a/modules/desktop_capture/desktop_capturer.h
++++ b/modules/desktop_capture/desktop_capturer.h
+@@ -23,7 +23,6 @@
+ #include "modules/desktop_capture/delegated_source_list_controller.h"
+ #include "modules/desktop_capture/desktop_capture_metadata.h"
+ #include "modules/desktop_capture/desktop_capture_types.h"
+-#include "modules/desktop_capture/desktop_frame.h"
+ #include "modules/desktop_capture/desktop_geometry.h"
+ #include "modules/desktop_capture/shared_memory.h"
+ #include "rtc_base/system/rtc_export.h"
+@@ -34,6 +33,7 @@ void RTC_EXPORT LogDesktopCapturerFullscreenDetectorUsage();
+
+ class DesktopCaptureOptions;
+ class DesktopFrame;
++class DesktopVector;
+
+ // Abstract interface for screen and window capturers.
+ class RTC_EXPORT DesktopCapturer {
diff --git a/third_party/libwebrtc/moz-patch-stack/s0126.patch b/third_party/libwebrtc/moz-patch-stack/s0126.patch
@@ -1,39 +1,22 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Thu, 18 Sep 2025 22:33:00 -0500
-Subject: Bug 1985396 - (fix-38c5a66082) fix windows build failure caused by a
+Date: Fri, 19 Sep 2025 10:46:00 -0500
+Subject: Bug 1985396 - (fix-948f6e7a4e) fix windows build failure caused by a
macro expansion
-Commit 38c5a66082 adds the include rtc_base/strings/string_builder.h to
-third_party/libwebrtc/modules/desktop_capture/desktop_geometry.h, but we
-pick this up through a chain of includes in CamerasParent.cpp. On Windows
-builds we then get a failure because Shlwapi.h has a macro that expands
-StrCat to StrCatW, which means we're changing absl::StrCat to
-absl::StrCatW which doesn't exist.
-
-We can avoid the desktop_geometry.h include and forward declare
-"class DesktopVector" to work around this issue.
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9541460d001e1a62ae567a18d810e35e50ac791c
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/5368b134e503299a9cef8590ec91fbc3cf4dcc8a
---
- modules/desktop_capture/desktop_capturer.h | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ modules/desktop_capture/desktop_capturer.h | 1 -
+ 1 file changed, 1 deletion(-)
diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
-index c85d0eba7a..8219358e1e 100644
+index 8219358e1e..9b1efa0bb5 100644
--- a/modules/desktop_capture/desktop_capturer.h
+++ b/modules/desktop_capture/desktop_capturer.h
@@ -23,7 +23,6 @@
#include "modules/desktop_capture/delegated_source_list_controller.h"
#include "modules/desktop_capture/desktop_capture_metadata.h"
#include "modules/desktop_capture/desktop_capture_types.h"
--#include "modules/desktop_capture/desktop_frame.h"
- #include "modules/desktop_capture/desktop_geometry.h"
+-#include "modules/desktop_capture/desktop_geometry.h"
#include "modules/desktop_capture/shared_memory.h"
#include "rtc_base/system/rtc_export.h"
-@@ -34,6 +33,7 @@ void RTC_EXPORT LogDesktopCapturerFullscreenDetectorUsage();
-
- class DesktopCaptureOptions;
- class DesktopFrame;
-+class DesktopVector;
- // Abstract interface for screen and window capturers.
- class RTC_EXPORT DesktopCapturer {
diff --git a/third_party/libwebrtc/moz-patch-stack/s0127.patch b/third_party/libwebrtc/moz-patch-stack/s0127.patch
@@ -1,22 +1,47 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Fri, 19 Sep 2025 10:46:00 -0500
-Subject: Bug 1985396 - (fix-948f6e7a4e) fix windows build failure caused by a
- macro expansion
+Date: Wed, 8 Oct 2025 17:49:00 +0000
+Subject: Bug 1980501 - use forward class decl rather than include
+ audio_state.h r?ng!
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/5368b134e503299a9cef8590ec91fbc3cf4dcc8a
+This avoids code outside of libwebrtc inadvertently including
+rtc_base/containers/flat_set.h which uses c++20 features. Once
+the rest of the Mozilla tree is compiling with c++20, we can
+remove this modification (see Bug 1768116).
+
+Differential Revision: https://phabricator.services.mozilla.com/D267993
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9c487a574d218cc1c261b646c501db4d159754f0
---
- modules/desktop_capture/desktop_capturer.h | 1 -
- 1 file changed, 1 deletion(-)
+ audio/audio_receive_stream.h | 12 ++++++++++++
+ 1 file changed, 12 insertions(+)
-diff --git a/modules/desktop_capture/desktop_capturer.h b/modules/desktop_capture/desktop_capturer.h
-index 8219358e1e..9b1efa0bb5 100644
---- a/modules/desktop_capture/desktop_capturer.h
-+++ b/modules/desktop_capture/desktop_capturer.h
-@@ -23,7 +23,6 @@
- #include "modules/desktop_capture/delegated_source_list_controller.h"
- #include "modules/desktop_capture/desktop_capture_metadata.h"
- #include "modules/desktop_capture/desktop_capture_types.h"
--#include "modules/desktop_capture/desktop_geometry.h"
- #include "modules/desktop_capture/shared_memory.h"
- #include "rtc_base/system/rtc_export.h"
+diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
+index 4b6c57b871..3f054776d2 100644
+--- a/audio/audio_receive_stream.h
++++ b/audio/audio_receive_stream.h
+@@ -34,7 +34,11 @@
+ #include "api/transport/rtp/rtp_source.h"
+ #include "api/units/time_delta.h"
+ #include "api/units/timestamp.h"
++// This can be removed after Bug 1768116 enables
++// c++20 builds across the entire Mozilla tree.
++#if !defined(WEBRTC_MOZILLA_BUILD)
+ #include "audio/audio_state.h"
++#endif
+ #include "call/audio_receive_stream.h"
+ #include "call/audio_state.h"
+ #include "call/syncable.h"
+@@ -47,6 +51,14 @@ class PacketRouter;
+ class RtpStreamReceiverControllerInterface;
+ class RtpStreamReceiverInterface;
++// This can be removed after Bug 1768116 enables
++// c++20 builds across the entire Mozilla tree.
++#if defined(WEBRTC_MOZILLA_BUILD)
++namespace internal {
++class AudioState;
++}
++#endif
++
+ namespace voe {
+ class ChannelReceiveInterface;
+ } // namespace voe
diff --git a/third_party/libwebrtc/moz-patch-stack/s0128.patch b/third_party/libwebrtc/moz-patch-stack/s0128.patch
@@ -1,47 +1,40 @@
From: Michael Froman <mfroman@mozilla.com>
Date: Wed, 8 Oct 2025 17:49:00 +0000
-Subject: Bug 1980501 - use forward class decl rather than include
- audio_state.h r?ng!
+Subject: Bug 1980501 - enable c++20 builds in libwebrtc. r?ng!
-This avoids code outside of libwebrtc inadvertently including
-rtc_base/containers/flat_set.h which uses c++20 features. Once
-the rest of the Mozilla tree is compiling with c++20, we can
-remove this modification (see Bug 1768116).
-
-Differential Revision: https://phabricator.services.mozilla.com/D267993
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/9c487a574d218cc1c261b646c501db4d159754f0
+Differential Revision: https://phabricator.services.mozilla.com/D267994
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/43de54a6e52daf0a06956ae01951ef1e29d46e82
---
- audio/audio_receive_stream.h | 12 ++++++++++++
- 1 file changed, 12 insertions(+)
+ BUILD.gn | 12 +++++++++++-
+ 1 file changed, 11 insertions(+), 1 deletion(-)
-diff --git a/audio/audio_receive_stream.h b/audio/audio_receive_stream.h
-index 4b6c57b871..3f054776d2 100644
---- a/audio/audio_receive_stream.h
-+++ b/audio/audio_receive_stream.h
-@@ -34,7 +34,11 @@
- #include "api/transport/rtp/rtp_source.h"
- #include "api/units/time_delta.h"
- #include "api/units/timestamp.h"
-+// This can be removed after Bug 1768116 enables
-+// c++20 builds across the entire Mozilla tree.
-+#if !defined(WEBRTC_MOZILLA_BUILD)
- #include "audio/audio_state.h"
-+#endif
- #include "call/audio_receive_stream.h"
- #include "call/audio_state.h"
- #include "call/syncable.h"
-@@ -47,6 +51,14 @@ class PacketRouter;
- class RtpStreamReceiverControllerInterface;
- class RtpStreamReceiverInterface;
+diff --git a/BUILD.gn b/BUILD.gn
+index 600993950b..3ba008076e 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -315,6 +315,16 @@ config("common_config") {
+ cflags_objc = []
+ defines = []
-+// This can be removed after Bug 1768116 enables
-+// c++20 builds across the entire Mozilla tree.
-+#if defined(WEBRTC_MOZILLA_BUILD)
-+namespace internal {
-+class AudioState;
-+}
-+#endif
++ # We should be able to remove this after c++20 is enabled
++ # across the Mozilla tree.
++ if (build_with_mozilla) {
++ if (is_win) {
++ cflags_cc += [ "-std:c++20" ]
++ } else {
++ cflags_cc += [ "-std=gnu++20" ]
++ }
++ }
+
- namespace voe {
- class ChannelReceiveInterface;
- } // namespace voe
+ if (rtc_enable_protobuf) {
+ defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
+ } else {
+@@ -416,7 +426,7 @@ config("common_config") {
+ # "-Wnested-externs", (C/Obj-C only)
+ ]
+ cflags_objc += [ "-Wstrict-prototypes" ]
+- cflags_cc = [
++ cflags_cc += [
+ "-Wnon-virtual-dtor",
+
+ # This is enabled for clang; enable for gcc as well.
diff --git a/third_party/libwebrtc/moz-patch-stack/s0129.patch b/third_party/libwebrtc/moz-patch-stack/s0129.patch
@@ -1,40 +1,23 @@
From: Michael Froman <mfroman@mozilla.com>
-Date: Wed, 8 Oct 2025 17:49:00 +0000
-Subject: Bug 1980501 - enable c++20 builds in libwebrtc. r?ng!
+Date: Wed, 8 Oct 2025 16:39:00 -0500
+Subject: Bug 1993083 - (fix-032f8c223e) add missing stub implementation for
+ SSLStreamAdapter static method.
-Differential Revision: https://phabricator.services.mozilla.com/D267994
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/43de54a6e52daf0a06956ae01951ef1e29d46e82
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/293f1877fead30723696b8f5cb354bdf819af5d3
---
- BUILD.gn | 12 +++++++++++-
- 1 file changed, 11 insertions(+), 1 deletion(-)
+ rtc_base/ssl_stream_adapter.h | 2 +-
+ 1 file changed, 1 insertion(+), 1 deletion(-)
-diff --git a/BUILD.gn b/BUILD.gn
-index 600993950b..3ba008076e 100644
---- a/BUILD.gn
-+++ b/BUILD.gn
-@@ -315,6 +315,16 @@ config("common_config") {
- cflags_objc = []
- defines = []
+diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h
+index 24fc299b82..ec16887eac 100644
+--- a/rtc_base/ssl_stream_adapter.h
++++ b/rtc_base/ssl_stream_adapter.h
+@@ -247,7 +247,7 @@ class SSLStreamAdapter : public StreamInterface {
+ static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups()
+ { return {}; }
+ static std::optional<std::string> GetEphemeralKeyExchangeCipherGroupName(
+- uint16_t group_id);
++ uint16_t group_id) { return std::nullopt; }
+ static std::vector<uint16_t> GetDefaultEphemeralKeyExchangeCipherGroups(
+ const FieldTrialsView* field_trials) { return {}; }
-+ # We should be able to remove this after c++20 is enabled
-+ # across the Mozilla tree.
-+ if (build_with_mozilla) {
-+ if (is_win) {
-+ cflags_cc += [ "-std:c++20" ]
-+ } else {
-+ cflags_cc += [ "-std=gnu++20" ]
-+ }
-+ }
-+
- if (rtc_enable_protobuf) {
- defines += [ "WEBRTC_ENABLE_PROTOBUF=1" ]
- } else {
-@@ -416,7 +426,7 @@ config("common_config") {
- # "-Wnested-externs", (C/Obj-C only)
- ]
- cflags_objc += [ "-Wstrict-prototypes" ]
-- cflags_cc = [
-+ cflags_cc += [
- "-Wnon-virtual-dtor",
-
- # This is enabled for clang; enable for gcc as well.
diff --git a/third_party/libwebrtc/moz-patch-stack/s0130.patch b/third_party/libwebrtc/moz-patch-stack/s0130.patch
@@ -1,23 +1,22 @@
-From: Michael Froman <mfroman@mozilla.com>
-Date: Wed, 8 Oct 2025 16:39:00 -0500
-Subject: Bug 1993083 - (fix-032f8c223e) add missing stub implementation for
- SSLStreamAdapter static method.
+From: Jan-Ivar Bruaroey <jib@mozilla.com>
+Date: Wed, 8 Oct 2025 17:05:00 +0000
+Subject: Bug 1868223 - Add api:frame_transformer_factory to BUILD.gn r=mjf
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/293f1877fead30723696b8f5cb354bdf819af5d3
+Differential Revision: https://phabricator.services.mozilla.com/D267541
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8e7778c59894c284b8bddd39b731e999fe2e807c
---
- rtc_base/ssl_stream_adapter.h | 2 +-
- 1 file changed, 1 insertion(+), 1 deletion(-)
+ BUILD.gn | 1 +
+ 1 file changed, 1 insertion(+)
-diff --git a/rtc_base/ssl_stream_adapter.h b/rtc_base/ssl_stream_adapter.h
-index 24fc299b82..ec16887eac 100644
---- a/rtc_base/ssl_stream_adapter.h
-+++ b/rtc_base/ssl_stream_adapter.h
-@@ -247,7 +247,7 @@ class SSLStreamAdapter : public StreamInterface {
- static std::set<uint16_t> GetSupportedEphemeralKeyExchangeCipherGroups()
- { return {}; }
- static std::optional<std::string> GetEphemeralKeyExchangeCipherGroupName(
-- uint16_t group_id);
-+ uint16_t group_id) { return std::nullopt; }
- static std::vector<uint16_t> GetDefaultEphemeralKeyExchangeCipherGroups(
- const FieldTrialsView* field_trials) { return {}; }
+diff --git a/BUILD.gn b/BUILD.gn
+index 3ba008076e..da75caaed6 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -625,6 +625,7 @@ if (!build_with_chromium) {
+ if (build_with_mozilla) {
+ deps += [
++ "api:frame_transformer_factory",
+ "api/audio:builtin_audio_processing_builder",
+ "api/environment:environment_factory",
+ "api/video:video_frame",
diff --git a/third_party/libwebrtc/moz-patch-stack/s0131.patch b/third_party/libwebrtc/moz-patch-stack/s0131.patch
@@ -1,22 +0,0 @@
-From: Jan-Ivar Bruaroey <jib@mozilla.com>
-Date: Wed, 8 Oct 2025 17:05:00 +0000
-Subject: Bug 1868223 - Add api:frame_transformer_factory to BUILD.gn r=mjf
-
-Differential Revision: https://phabricator.services.mozilla.com/D267541
-Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/8e7778c59894c284b8bddd39b731e999fe2e807c
----
- BUILD.gn | 1 +
- 1 file changed, 1 insertion(+)
-
-diff --git a/BUILD.gn b/BUILD.gn
-index 3ba008076e..da75caaed6 100644
---- a/BUILD.gn
-+++ b/BUILD.gn
-@@ -625,6 +625,7 @@ if (!build_with_chromium) {
-
- if (build_with_mozilla) {
- deps += [
-+ "api:frame_transformer_factory",
- "api/audio:builtin_audio_processing_builder",
- "api/environment:environment_factory",
- "api/video:video_frame",
