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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/moz-patch-stack/0082.patch
parentInitial commit. (diff)
downloadfirefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz
firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/moz-patch-stack/0082.patch')
-rw-r--r--third_party/libwebrtc/moz-patch-stack/0082.patch189
1 files changed, 189 insertions, 0 deletions
diff --git a/third_party/libwebrtc/moz-patch-stack/0082.patch b/third_party/libwebrtc/moz-patch-stack/0082.patch
new file mode 100644
index 0000000000..ad98ccfed2
--- /dev/null
+++ b/third_party/libwebrtc/moz-patch-stack/0082.patch
@@ -0,0 +1,189 @@
+From: Michael Froman <mfroman@mozilla.com>
+Date: Wed, 8 Mar 2023 00:26:00 +0000
+Subject: Bug 1820869 - avoid building unreachable files. r=ng,webrtc-reviewers
+
+Differential Revision: https://phabricator.services.mozilla.com/D171922
+Mercurial Revision: https://hg.mozilla.org/mozilla-central/rev/88b3cc6bbece7c53d00e124713330f3d34d2789d
+---
+ BUILD.gn | 9 +++++++++
+ call/BUILD.gn | 10 ++++++++++
+ media/BUILD.gn | 7 ++++++-
+ modules/audio_device/BUILD.gn | 11 ++++++++++-
+ rtc_base/BUILD.gn | 2 ++
+ webrtc.gni | 2 +-
+ 6 files changed, 38 insertions(+), 3 deletions(-)
+
+diff --git a/BUILD.gn b/BUILD.gn
+index 6515866c2d..465c4d9bfd 100644
+--- a/BUILD.gn
++++ b/BUILD.gn
+@@ -549,6 +549,15 @@ if (!build_with_chromium) {
+ "api/video:video_rtp_headers",
+ "test:rtp_test_utils",
+ ]
++ # Added when we removed deps in other places to avoid building
++ # unreachable sources. See Bug 1820869.
++ deps += [
++ "api/video_codecs:video_codecs_api",
++ "api/video_codecs:rtc_software_fallback_wrappers",
++ "media:rtc_encoder_simulcast_proxy",
++ "modules/video_coding:webrtc_vp8",
++ "modules/video_coding:webrtc_vp9",
++ ]
+ } else {
+ deps += [
+ "api",
+diff --git a/call/BUILD.gn b/call/BUILD.gn
+index 26618aee80..fb23b7ef39 100644
+--- a/call/BUILD.gn
++++ b/call/BUILD.gn
+@@ -352,6 +352,16 @@ rtc_library("call") {
+ "//third_party/abseil-cpp/absl/strings",
+ "//third_party/abseil-cpp/absl/types:optional",
+ ]
++ if (build_with_mozilla) { # See Bug 1820869.
++ sources -= [
++ "call_factory.cc",
++ "degraded_call.cc",
++ ]
++ deps -= [
++ ":fake_network",
++ ":simulated_network",
++ ]
++ }
+ }
+
+ rtc_source_set("receive_stream_interface") {
+diff --git a/media/BUILD.gn b/media/BUILD.gn
+index 4ddc8349a8..daca67e033 100644
+--- a/media/BUILD.gn
++++ b/media/BUILD.gn
+@@ -442,7 +442,10 @@ rtc_library("rtc_internal_video_codecs") {
+ "../test:fake_video_codecs",
+ ]
+ if (build_with_mozilla) {
+- deps -= [ "../test:fake_video_codecs" ]
++ deps -= [
++ "../modules/video_coding:webrtc_multiplex", # See Bug 1820869.
++ "../test:fake_video_codecs",
++ ]
+ }
+
+ if (enable_libaom) {
+@@ -477,6 +480,8 @@ rtc_library("rtc_internal_video_codecs") {
+ sources -= [
+ "engine/fake_video_codec_factory.cc",
+ "engine/fake_video_codec_factory.h",
++ "engine/internal_encoder_factory.cc", # See Bug 1820869.
++ "engine/multiplex_codec_factory.cc", # See Bug 1820869.
+ ]
+ }
+ }
+diff --git a/modules/audio_device/BUILD.gn b/modules/audio_device/BUILD.gn
+index e35a442025..61cd531edd 100644
+--- a/modules/audio_device/BUILD.gn
++++ b/modules/audio_device/BUILD.gn
+@@ -30,6 +30,7 @@ rtc_source_set("audio_device_default") {
+ }
+
+ rtc_source_set("audio_device") {
++if (!build_with_mozilla) { # See Bug 1820869.
+ visibility = [ "*" ]
+ public_deps = [
+ ":audio_device_api",
+@@ -40,6 +41,7 @@ rtc_source_set("audio_device") {
+ ":audio_device_impl",
+ ]
+ }
++}
+
+ rtc_source_set("audio_device_api") {
+ visibility = [ "*" ]
+@@ -58,6 +60,7 @@ rtc_source_set("audio_device_api") {
+ }
+
+ rtc_library("audio_device_buffer") {
++if (!build_with_mozilla) { # See Bug 1820869.
+ sources = [
+ "audio_device_buffer.cc",
+ "audio_device_buffer.h",
+@@ -85,6 +88,7 @@ rtc_library("audio_device_buffer") {
+ "../../system_wrappers:metrics",
+ ]
+ }
++}
+
+ rtc_library("audio_device_generic") {
+ sources = [
+@@ -180,6 +184,7 @@ rtc_source_set("audio_device_module_from_input_and_output") {
+ # Contains default implementations of webrtc::AudioDeviceModule for Windows,
+ # Linux, Mac, iOS and Android.
+ rtc_library("audio_device_impl") {
++if (!build_with_mozilla) { # See Bug 1820869.
+ visibility = [ "*" ]
+ deps = [
+ ":audio_device_api",
+@@ -373,6 +378,7 @@ rtc_library("audio_device_impl") {
+ ]
+ }
+ }
++}
+
+ if (is_mac) {
+ rtc_source_set("audio_device_impl_frameworks") {
+@@ -390,6 +396,7 @@ if (is_mac) {
+ }
+ }
+
++if (!build_with_mozilla) { # See Bug 1820869.
+ rtc_source_set("mock_audio_device") {
+ visibility = [ "*" ]
+ testonly = true
+@@ -406,8 +413,10 @@ rtc_source_set("mock_audio_device") {
+ "../../test:test_support",
+ ]
+ }
++}
+
+-if (rtc_include_tests && !build_with_chromium) {
++# See Bug 1820869 for !build_with_mozilla.
++if (rtc_include_tests && !build_with_chromium && !build_with_mozilla) {
+ rtc_library("audio_device_unittests") {
+ testonly = true
+
+diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
+index 3cd0bfff06..0b1e2a6208 100644
+--- a/rtc_base/BUILD.gn
++++ b/rtc_base/BUILD.gn
+@@ -283,6 +283,7 @@ rtc_library("sample_counter") {
+ absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
+ }
+
++if (!build_with_mozilla) { # See Bug 1820869.
+ rtc_library("timestamp_aligner") {
+ visibility = [ "*" ]
+ sources = [
+@@ -296,6 +297,7 @@ rtc_library("timestamp_aligner") {
+ "system:rtc_export",
+ ]
+ }
++}
+
+ rtc_library("zero_memory") {
+ visibility = [ "*" ]
+diff --git a/webrtc.gni b/webrtc.gni
+index 1b21d329b2..46a9433141 100644
+--- a/webrtc.gni
++++ b/webrtc.gni
+@@ -221,7 +221,7 @@ declare_args() {
+ # video codecs they depends on will not be included in libwebrtc.{a|lib}
+ # (they will still be included in libjingle_peerconnection_so.so and
+ # WebRTC.framework)
+- rtc_include_builtin_video_codecs = true
++ rtc_include_builtin_video_codecs = !build_with_mozilla # See Bug 1820869.
+
+ # When set to true and in a standalone build, it will undefine UNICODE and
+ # _UNICODE (which are always defined globally by the Chromium Windows
+--
+2.34.1
+