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