Bug 1437345 - Don't build audio_device after bug 1437345. r?dminor draft
authorJan Beich <jbeich@FreeBSD.org>
Wed, 14 Mar 2018 17:09:06 +0000
changeset 767549 0a8bf54891bd8ccea8466dc486a7ae01c832f889
parent 767347 80b4777a6421d8df4bb27ac23fb607c318a3006c
push id102631
push userbmo:jbeich@FreeBSD.org
push dateWed, 14 Mar 2018 20:02:24 +0000
reviewersdminor
bugs1437345
milestone61.0a1
Bug 1437345 - Don't build audio_device after bug 1437345. r?dminor MozReview-Commit-ID: JjLucnqna0A
media/webrtc/trunk/webrtc/build/webrtc.gni
media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn
--- a/media/webrtc/trunk/webrtc/build/webrtc.gni
+++ b/media/webrtc/trunk/webrtc/build/webrtc.gni
@@ -154,17 +154,17 @@ declare_args() {
 
   rtc_restrict_logging = build_with_chromium || build_with_mozilla
 
   # Excluded in Chromium since its prerequisites don't require Pulse Audio.
   rtc_include_pulse_audio = !build_with_chromium
 
   # Chromium uses its own IO handling, so the internal ADM is only built for
   # standalone WebRTC.
-  rtc_include_internal_audio_device = !build_with_chromium
+  rtc_include_internal_audio_device = !build_with_chromium && !build_with_mozilla
 
   # Include tests in standalone checkout.
   rtc_include_tests = !build_with_chromium && !build_with_mozilla
 }
 
 # 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).
 if (build_with_mozilla) {
--- a/media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn
@@ -128,46 +128,36 @@ rtc_static_library("audio_device") {
         "android/opensles_player.h",
         "android/opensles_recorder.cc",
         "android/opensles_recorder.h",
       ]
       libs = [
         "log",
         "OpenSLES",
       ]
-
-      if (build_with_mozilla) {
-        include_dirs += [
-          "/config/external/nspr",
-          "/nsprpub/lib/ds",
-          "/nsprpub/pr/include",
-        ]
-      }
     }
     if (rtc_use_dummy_audio_file_devices) {
       defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]
     } else {
       if (is_linux) {
         sources += [
+          "linux/alsasymboltable_linux.cc",
+          "linux/alsasymboltable_linux.h",
+          "linux/audio_device_alsa_linux.cc",
+          "linux/audio_device_alsa_linux.h",
+          "linux/audio_mixer_manager_alsa_linux.cc",
+          "linux/audio_mixer_manager_alsa_linux.h",
           "linux/latebindingsymboltable_linux.cc",
           "linux/latebindingsymboltable_linux.h",
         ]
-        libs = [ "dl" ]
-        if (!build_with_mozilla) {
-          sources += [
-            "linux/alsasymboltable_linux.cc",
-            "linux/alsasymboltable_linux.h",
-            "linux/audio_device_alsa_linux.cc",
-            "linux/audio_device_alsa_linux.h",
-            "linux/audio_mixer_manager_alsa_linux.cc",
-            "linux/audio_mixer_manager_alsa_linux.h",
-          ]
-          defines += [ "LINUX_ALSA" ]
-          libs += [ "X11" ]
-        }
+        defines += [ "LINUX_ALSA" ]
+        libs = [
+          "dl",
+          "X11",
+        ]
         if (rtc_include_pulse_audio) {
           sources += [
             "linux/audio_device_pulse_linux.cc",
             "linux/audio_device_pulse_linux.h",
             "linux/audio_mixer_manager_pulse_linux.cc",
             "linux/audio_mixer_manager_pulse_linux.h",
             "linux/pulseaudiosymboltable_linux.cc",
             "linux/pulseaudiosymboltable_linux.h",