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Bug 1952339 - Vendor libwebrtc from b300a9cc05

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Upstream commit: https://webrtc.googlesource.com/src/+/b300a9cc05bf9735d735edfeade09216e9068888
    OpenH264 library enabled in Android Chromium build

    H264 codec components are included in Chromium build on Android when
    OpenH264 library is enabled in Chromium media using media_use_openh264
    build flag.

    Bug: chromium:40519162
    Change-Id: I4fcc341a7d208b399a294b1f86774860f70f83b6
    Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/372680
    Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
    Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
    Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
    Cr-Commit-Position: refs/heads/main@{#43940}

Differential Revision: https://phabricator.services.mozilla.com/D243941
This commit is contained in:
Michael Froman
2025-03-07 15:39:51 -06:00
parent be6c08786b
commit eefba4d644
12 changed files with 42 additions and 36 deletions
Download Patch File Download Diff File Expand all files Collapse all files

4
third_party/libwebrtc/README.mozilla.last-vendor vendored
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@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-03-07T21:38:53.492911+00:00.
libwebrtc updated from /home/mfroman/mozilla/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-03-07T21:39:41.205208+00:00.
# base of lastest vendoring
3a3ebb512e
b300a9cc05

2
third_party/libwebrtc/moz-patch-stack/0001.patch vendored
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@@ -1649,7 +1649,7 @@ index 0a9226ef6f..620c1c02f3 100644
vcm_ = nullptr;
}
diff --git a/webrtc.gni b/webrtc.gni
index e33d52902a..d6628dc52a 100644
index cd861f91a0..d77a4310f2 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -123,7 +123,7 @@ declare_args() {

10
third_party/libwebrtc/moz-patch-stack/0030.patch vendored
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@@ -1310,7 +1310,7 @@ index 7cca46263c..ee2b94ed45 100644
"../api/adaptation:resource_adaptation_api",
"../api/environment",
diff --git a/webrtc.gni b/webrtc.gni
index d6628dc52a..e635ff0866 100644
index d77a4310f2..29286c5b95 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -35,6 +35,11 @@ if (is_mac) {
@@ -1382,7 +1382,7 @@ index d6628dc52a..e635ff0866 100644
# Experimental: enable use of Android AAudio which requires Android SDK 26 or above
# and NDK r16 or above.
rtc_enable_android_aaudio = false
@@ -284,7 +291,7 @@ declare_args() {
@@ -290,7 +297,7 @@ declare_args() {
rtc_build_json = !build_with_mozilla
rtc_build_libsrtp = !build_with_mozilla
rtc_build_libvpx = !build_with_mozilla
@@ -1391,7 +1391,7 @@ index d6628dc52a..e635ff0866 100644
rtc_build_opus = !build_with_mozilla
rtc_build_ssl = !build_with_mozilla
@@ -293,7 +300,7 @@ declare_args() {
@@ -299,7 +306,7 @@ declare_args() {
# Chromium uses its own IO handling, so the internal ADM is only built for
# standalone WebRTC.
@@ -1400,7 +1400,7 @@ index d6628dc52a..e635ff0866 100644
# Set this to true to enable the avx2 support in webrtc.
# TODO: Make sure that AVX2 works also for non-clang compilers.
@@ -333,6 +340,9 @@ declare_args() {
@@ -339,6 +346,9 @@ declare_args() {
rtc_enable_grpc = rtc_enable_protobuf && (is_linux || is_mac)
}
@@ -1410,7 +1410,7 @@ index d6628dc52a..e635ff0866 100644
# Make it possible to provide custom locations for some libraries (move these
# up into declare_args should we need to actually use them for the GN build).
rtc_libvpx_dir = "//third_party/libvpx"
@@ -1199,7 +1209,7 @@ if (is_mac || is_ios) {
@@ -1205,7 +1215,7 @@ if (is_mac || is_ios) {
}
}

6
third_party/libwebrtc/moz-patch-stack/0039.patch vendored
View File

@@ -276,7 +276,7 @@ index 059a4dadc1..9e687b8787 100644
config("x11_config") {
if (rtc_use_x11_extensions) {
diff --git a/webrtc.gni b/webrtc.gni
index e635ff0866..16a1e59e9a 100644
index 29286c5b95..dc0553d964 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -169,13 +169,13 @@ declare_args() {
@@ -293,5 +293,5 @@ index e635ff0866..16a1e59e9a 100644
- (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"
+ (target_cpu == "arm" && arm_use_neon) || target_cpu == "arm64"
# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
# all platforms except Android and iOS. Because FFmpeg can be built
# Enable this to build OpenH264 encoder/FFmpeg decoder. When building WebRTC
# as part of Chromium, this is delegated to `media_use_openh264`. When

4
third_party/libwebrtc/moz-patch-stack/0051.patch vendored
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@@ -74,10 +74,10 @@ index d64ea689bb..c3c6955a7b 100644
#endif // defined(WEBRTC_POSIX)
}
diff --git a/webrtc.gni b/webrtc.gni
index 16a1e59e9a..933080a544 100644
index dc0553d964..c67c8d5ea5 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -350,7 +350,7 @@ rtc_opus_dir = "//third_party/opus"
@@ -356,7 +356,7 @@ rtc_opus_dir = "//third_party/opus"
# Desktop capturer is supported only on Windows, OSX and Linux.
rtc_desktop_capture_supported =

4
third_party/libwebrtc/moz-patch-stack/0078.patch vendored
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@@ -10,10 +10,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/46fb51c90709be64c
1 file changed, 1 insertion(+), 1 deletion(-)
diff --git a/webrtc.gni b/webrtc.gni
index 933080a544..08736c53ef 100644
index c67c8d5ea5..93fd8fc8ed 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -304,7 +304,7 @@ declare_args() {
@@ -310,7 +310,7 @@ declare_args() {
# Set this to true to enable the avx2 support in webrtc.
# TODO: Make sure that AVX2 works also for non-clang compilers.

4
third_party/libwebrtc/moz-patch-stack/0126.patch vendored
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@@ -61,7 +61,7 @@ index 7e044d26c3..3053b829f2 100644
rtc_library("encoded_frame") {
diff --git a/webrtc.gni b/webrtc.gni
index 08736c53ef..50e7bdf856 100644
index 93fd8fc8ed..bd0af3a4c5 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -11,7 +11,7 @@ import("//build/config/mips.gni")
@@ -73,7 +73,7 @@ index 08736c53ef..50e7bdf856 100644
if (!build_with_chromium && is_component_build) {
print("The Gn argument `is_component_build` is currently " +
@@ -487,8 +487,8 @@ all_poison_types = [
@@ -493,8 +493,8 @@ all_poison_types = [
"software_video_codecs",
]

6
third_party/libwebrtc/moz-patch-stack/0127.patch vendored
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@@ -10,10 +10,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/45b99d1ba95b46896
1 file changed, 71 insertions(+)
diff --git a/webrtc.gni b/webrtc.gni
index 50e7bdf856..fe50741224 100644
index bd0af3a4c5..33d81920f4 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -685,6 +685,36 @@ template("rtc_source_set") {
@@ -691,6 +691,36 @@ template("rtc_source_set") {
deps += [ "//third_party/abseil-cpp:absl" ]
}
}
@@ -50,7 +50,7 @@ index 50e7bdf856..fe50741224 100644
}
}
@@ -919,6 +949,47 @@ template("rtc_library") {
@@ -925,6 +955,47 @@ template("rtc_library") {
deps += [ "//third_party/abseil-cpp:absl" ]
}
}

6
third_party/libwebrtc/moz-patch-stack/0128.patch vendored
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@@ -874,7 +874,7 @@ index fb107ef1a3..b718fa583d 100644
declare_args() {
diff --git a/webrtc.gni b/webrtc.gni
index fe50741224..4466757c0f 100644
index 33d81920f4..7e6602dfd7 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -5,12 +5,12 @@
@@ -922,7 +922,7 @@ index fe50741224..4466757c0f 100644
}
if (build_with_chromium) {
@@ -373,8 +373,8 @@ rtc_common_configs = [ webrtc_root + ":common_config" ]
@@ -379,8 +379,8 @@ rtc_common_configs = [ webrtc_root + ":common_config" ]
if (is_mac || is_ios) {
if (filter_include(default_compiler_configs,
@@ -933,7 +933,7 @@ index fe50741224..4466757c0f 100644
}
}
@@ -392,7 +392,7 @@ rtc_common_inherited_config = webrtc_root + ":common_inherited_config"
@@ -398,7 +398,7 @@ 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) {

8
third_party/libwebrtc/moz-patch-stack/0129.patch vendored
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@@ -10,10 +10,10 @@ Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/f18c05287ce831369
1 file changed, 12 insertions(+), 2 deletions(-)
diff --git a/webrtc.gni b/webrtc.gni
index 4466757c0f..254896c313 100644
index 7e6602dfd7..fe68d3317d 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -536,7 +536,7 @@ template("rtc_test") {
@@ -542,7 +542,7 @@ template("rtc_test") {
min_sdk_version = 21
target_sdk_version = 24
deps += [
@@ -22,7 +22,7 @@ index 4466757c0f..254896c313 100644
webrtc_root + "sdk/android:native_test_jni_onload",
webrtc_root + "sdk/android:base_java",
webrtc_root + "test:native_test_java",
@@ -985,11 +985,21 @@ template("rtc_library") {
@@ -991,11 +991,21 @@ template("rtc_library") {
modified_deps = []
foreach (dep, deps) {
newdep = string_replace(dep, "//third_party/", "//libwebrtc/third_party/")
@@ -44,7 +44,7 @@ index 4466757c0f..254896c313 100644
}
}
@@ -1027,7 +1037,7 @@ template("rtc_executable") {
@@ -1033,7 +1043,7 @@ template("rtc_executable") {
if (is_win) {
deps += [
# Give executables the default manifest on Windows (a no-op elsewhere).

4
third_party/libwebrtc/moz-patch-stack/0136.patch vendored
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@@ -181,7 +181,7 @@ index 2c83d6614d..87c62f0e99 100644
rtc_library("ssl") {
if (!build_with_mozilla) {
diff --git a/webrtc.gni b/webrtc.gni
index 254896c313..8fac542d4f 100644
index fe68d3317d..17306b77b7 100644
--- a/webrtc.gni
+++ b/webrtc.gni
@@ -125,7 +125,7 @@ declare_args() {
@@ -193,7 +193,7 @@ index 254896c313..8fac542d4f 100644
# Enable when an external authentication mechanism is used for performing
# packet authentication for RTP packets instead of libsrtp.
@@ -341,7 +341,7 @@ declare_args() {
@@ -347,7 +347,7 @@ declare_args() {
}
# Enable liboam only on non-mozilla builds.

20
third_party/libwebrtc/webrtc.gni vendored
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@@ -177,18 +177,24 @@ declare_args() {
rtc_build_with_neon =
(target_cpu == "arm" && arm_use_neon) || target_cpu == "arm64"
# Enable this to build OpenH264 encoder/FFmpeg decoder. This is supported on
# all platforms except Android and iOS. Because FFmpeg can be built
# with/without H.264 support, `ffmpeg_branding` has to separately be set to a
# value that includes H.264, for example "Chrome". If FFmpeg is built without
# H.264, compilation succeeds but `H264DecoderImpl` fails to initialize.
# Enable this to build OpenH264 encoder/FFmpeg decoder. When building WebRTC
# as part of Chromium, this is delegated to `media_use_openh264`. When
# building WebRTC as a standalone library, this is supported on all platforms
# except Android and iOS. Because FFmpeg can be built with/without H.264
# support, `ffmpeg_branding` has to separately be set to a value that
# includes H.264, for example "Chrome". If FFmpeg is built without H.264,
# compilation succeeds but `H264DecoderImpl` fails to initialize.
# CHECK THE OPENH264, FFMPEG AND H.264 LICENSES/PATENTS BEFORE BUILDING.
# http://www.openh264.org, https://www.ffmpeg.org/
#
# Enabling H264 when building with MSVC is currently not supported, see
# bugs.webrtc.org/9213#c13 for more info.
rtc_use_h264 =
proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
if (build_with_chromium) {
rtc_use_h264 = media_use_openh264
} else {
rtc_use_h264 =
proprietary_codecs && !is_android && !is_ios && !(is_win && !is_clang)
}
# Use system OpenH264
rtc_system_openh264 = false