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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
|
