Bug 987979: Patch 3 - Fix various build issues in webrtc.org/Mozilla integration. r=rjesup

config/system-headers

@@ -347,7 +347,6 @@ commdlg.h
 compat.h
 condapi.h
 ConditionalMacros.h
-config.h
 conio.h
 console.h
 ControlDefinitions.h

content/media/webrtc/MediaEngineWebRTC.cpp

@@ -131,12 +131,10 @@ MediaEngineWebRTC::EnumerateVideoDevices(nsTArray<nsRefPtr<MediaEngineVideoSourc
   MutexAutoLock lock(mMutex);
 
 #ifdef MOZ_WIDGET_ANDROID
-  jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
-
   // get the JVM
   JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
 
-  if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
+  if (webrtc::VideoEngine::SetAndroidObjects(jvm) != 0) {
     LOG(("VieCapture:SetAndroidObjects Failed"));
     return;
   }

ipc/chromium/src/base/basictypes.h

@@ -169,13 +169,15 @@ inline To implicit_cast(From const &f) {
 // the expression is false, most compilers will issue a warning/error
 // containing the name of the variable.
 
+// Avoid multiple definitions for webrtc
+#if !defined(COMPILE_ASSERT)
 template <bool>
 struct CompileAssert {
 };
 
-#undef COMPILE_ASSERT
 #define COMPILE_ASSERT(expr, msg) \
   typedef CompileAssert<(bool(expr))> msg[bool(expr) ? 1 : -1]
+#endif
 
 // Implementation details of COMPILE_ASSERT:
 //

media/mtransport/third_party/nICEr/src/stun/addrs.c

@@ -53,8 +53,8 @@ static char *RCSSTRING __UNUSED__="$Id: addrs.c,v 1.2 2008/04/28 18:21:30 ekr Ex
 #undef __unused
 #include <linux/sysctl.h>
 #endif
-#include <net/if.h>
 #ifndef LINUX
+#include <net/if.h>
 #if !defined(__OpenBSD__) && !defined(__NetBSD__)
 #include <net/if_var.h>
 #endif

media/mtransport/third_party/nICEr/src/stun/stun.h

@@ -39,13 +39,15 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 #else
 #include <sys/param.h>
 #include <sys/socket.h>
-#include <net/if.h>
 #ifndef LINUX
+#include <net/if.h>
 #if !defined(__OpenBSD__) && !defined(__NetBSD__)
 #include <net/if_var.h>
 #endif
 #include <net/if_dl.h>
 #include <net/if_types.h>
+#else
+#include <linux/if.h>
 #endif
 #ifndef BSD
 #include <net/route.h>

media/webrtc/signaling/signaling.gyp

@@ -50,7 +50,6 @@
         './include',
         './src/sipcc/include',
         './src/sipcc/cpr/include',
-        '../../../ipc/chromium/src',
         '../../../ipc/chromium/src/base/third_party/nspr',
         '../../../xpcom/base',
         '../../../dom/base',
@@ -63,9 +62,6 @@
         '../trunk/webrtc/modules/interface',
         '../trunk/webrtc/peerconnection',
         '../../libyuv/include',
-        '../../../netwerk/srtp/src/include',
-        '../../../netwerk/srtp/src/crypto/include',
-        '../../../ipc/chromium/src',
         '../../mtransport/third_party/nrappkit/src/util/libekr',
       ],
 
@@ -160,8 +156,6 @@
         './src/mediapipeline/MediaPipeline.cpp',
         './src/mediapipeline/MediaPipelineFilter.h',
         './src/mediapipeline/MediaPipelineFilter.cpp',
-        './src/mediapipeline/SrtpFlow.h',
-        './src/mediapipeline/SrtpFlow.cpp',
       ],
 
       #
@@ -194,12 +188,28 @@
       # Conditionals
       #
       'conditions': [
+        # hack so I can change the include flow for SrtpFlow
+        ['build_with_mozilla==1', {
+          'sources': [
+            './src/mediapipeline/SrtpFlow.h',
+            './src/mediapipeline/SrtpFlow.cpp',
+          ],
+          'include_dirs!': [
+            '../trunk/webrtc',
+          ],
+          'include_dirs': [
+            '../../../netwerk/srtp/src/include',
+            '../../../netwerk/srtp/src/crypto/include',
+          ],
+        }],
         ['moz_webrtc_omx==1', {
           'sources': [
             './src/media-conduit/WebrtcOMXH264VideoCodec.cpp',
             './src/media-conduit/OMXVideoCodec.cpp',
           ],
           'include_dirs': [
+            # hack on hack to re-add it after SrtpFlow removes it
+            '../../webrtc/trunk/webrtc',
             '../../../content/media/omx',
             '../../../gfx/layers/client',
           ],
@@ -238,6 +248,7 @@
           ],
 
           'defines': [
+            'OS_LINUX',
             'SIP_OS_LINUX',
             '_GNU_SOURCE',
             'LINUX',
@@ -252,6 +263,7 @@
           'include_dirs': [
           ],
           'defines': [
+            'OS_WIN',
             'SIP_OS_WINDOWS',
             'WIN32',
             'GIPS_VER=3480',
@@ -279,6 +291,7 @@
           'include_dirs': [
           ],
           'defines': [
+            'OS_MACOSX',
             'SIP_OS_OSX',
             'OSX',
             '_FORTIFY_SOURCE=2',

media/webrtc/signaling/src/callcontrol/CallControlManagerImpl.cpp

@@ -18,6 +18,8 @@
 #include "CallControlManagerImpl.h"
 #include "csf_common.h"
 
+#include "base/platform_thread.h"
+
 extern "C"
 {
 #include "config_api.h"

media/webrtc/signaling/src/media-conduit/VideoConduit.cpp

@@ -245,12 +245,10 @@ MediaConduitErrorCode WebrtcVideoConduit::Init(WebrtcVideoConduit *other)
   } else {
 
 #ifdef MOZ_WIDGET_ANDROID
-    jobject context = jsjni_GetGlobalContextRef();
-
     // get the JVM
     JavaVM *jvm = jsjni_GetVM();
 
-    if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
+    if (webrtc::VideoEngine::SetAndroidObjects(jvm) != 0) {
       CSFLogError(logTag,  "%s: could not set Android objects", __FUNCTION__);
       return kMediaConduitSessionNotInited;
     }

media/webrtc/trunk/build/all.gyp

@@ -104,7 +104,7 @@
             }],
           ],
         }],
-        ['toolkit_uses_gtk==1', {
+        ['(toolkit_uses_gtk==1) and (build_with_mozilla==0)', {
           'dependencies': [
             '../tools/gtk_clipboard_dump/gtk_clipboard_dump.gyp:*',
             '../tools/xdisplaycheck/xdisplaycheck.gyp:*',

media/webrtc/trunk/build/filename_rules.gypi

@@ -66,7 +66,7 @@
         ['exclude', '(^|/)x11_[^/]*\\.(h|cc)$'],
       ],
     }],
-    ['<(toolkit_uses_gtk)!=1 or >(nacl_untrusted_build)==1', {
+    ['(<(toolkit_uses_gtk)!=1 or >(nacl_untrusted_build)==1) and (build_with_mozilla==0)', {
       'sources/': [
         ['exclude', '_gtk(_browsertest|_unittest)?\\.(h|cc)$'],
         ['exclude', '(^|/)gtk/'],

media/webrtc/trunk/webrtc/modules/video_capture/include/video_capture.h

@@ -14,13 +14,13 @@
 #include "webrtc/modules/interface/module.h"
 #include "webrtc/modules/video_capture/include/video_capture_defines.h"
 
-#ifdef ANDROID
+#if defined(ANDROID) && !defined(WEBRTC_GONK)
 #include <jni.h>
 #endif
 
 namespace webrtc {
 
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_GONK)
 int32_t SetCaptureAndroidVM(JavaVM* javaVM);
 #endif
 

media/webrtc/trunk/webrtc/video_engine/include/vie_base.h

@@ -21,7 +21,7 @@
 
 #include "webrtc/common_types.h"
 
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(MOZ_WIDGET_GONK)
 #include <jni.h>
 #endif
 
@@ -64,7 +64,7 @@ class WEBRTC_DLLEXPORT VideoEngine {
   // user receives callbacks for generated trace messages.
   static int SetTraceCallback(TraceCallback* callback);
 
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(MOZ_WIDGET_GONK)
   // Android specific.
   static int SetAndroidObjects(JavaVM* java_vm);
 #endif

media/webrtc/trunk/webrtc/video_engine/vie_impl.cc

@@ -155,7 +155,7 @@ int VideoEngine::SetTraceCallback(TraceCallback* callback) {
   return Trace::SetTraceCallback(callback);
 }
 
-#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
+#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_GONK)
 int VideoEngine::SetAndroidObjects(JavaVM* javaVM) {
   WEBRTC_TRACE(kTraceApiCall, kTraceVideo, kModuleId,
                "SetAndroidObjects()");

mobile/android/base/moz.build

@@ -83,12 +83,15 @@ stjar.javac_flags = ['-Xlint:none']
 
 if CONFIG['MOZ_WEBRTC']:
     video_root = TOPSRCDIR + '/media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/'
+    video_render_root = TOPSRCDIR + '/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/'
     audio_root = TOPSRCDIR + '/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/'
     wrjar = add_java_jar('webrtc')
     wrjar.sources += [
         video_root + 'CaptureCapabilityAndroid.java',
         video_root + 'VideoCaptureAndroid.java',
         video_root + 'VideoCaptureDeviceInfoAndroid.java',
+        video_render_root + 'ViEAndroidGLES20.java',
+        video_render_root + 'ViERenderer.java',
     ]
     wrjar.sources += [
         audio_root + 'AudioManagerAndroid.java',