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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/moz-patch-stack/0082.patch | |
parent | Initial commit. (diff) | |
download | firefox-esr-upstream.tar.xz firefox-esr-upstream.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.patch | 189 |
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 + |