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Bug 987979: Patch 3 - Fix various build issues in webrtc.org/Mozilla integration. r=rjesup
This commit is contained in:
parent
21318d2311
commit
964601c191
config
content/media/webrtc
ipc/chromium/src/base
media
mtransport/third_party/nICEr/src/stun
webrtc
signaling
trunk
build
webrtc
mobile/android/base
@ -347,7 +347,6 @@ commdlg.h
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compat.h
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condapi.h
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ConditionalMacros.h
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config.h
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conio.h
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console.h
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ControlDefinitions.h
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@ -131,12 +131,10 @@ MediaEngineWebRTC::EnumerateVideoDevices(nsTArray<nsRefPtr<MediaEngineVideoSourc
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MutexAutoLock lock(mMutex);
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#ifdef MOZ_WIDGET_ANDROID
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jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
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// get the JVM
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JavaVM *jvm = mozilla::AndroidBridge::Bridge()->GetVM();
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if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
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if (webrtc::VideoEngine::SetAndroidObjects(jvm) != 0) {
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LOG(("VieCapture:SetAndroidObjects Failed"));
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return;
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}
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@ -169,13 +169,15 @@ inline To implicit_cast(From const &f) {
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// the expression is false, most compilers will issue a warning/error
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// containing the name of the variable.
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// Avoid multiple definitions for webrtc
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#if !defined(COMPILE_ASSERT)
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template <bool>
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struct CompileAssert {
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};
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#undef COMPILE_ASSERT
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#define COMPILE_ASSERT(expr, msg) \
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typedef CompileAssert<(bool(expr))> msg[bool(expr) ? 1 : -1]
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#endif
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// Implementation details of COMPILE_ASSERT:
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//
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@ -53,8 +53,8 @@ static char *RCSSTRING __UNUSED__="$Id: addrs.c,v 1.2 2008/04/28 18:21:30 ekr Ex
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#undef __unused
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#include <linux/sysctl.h>
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#endif
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#include <net/if.h>
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#ifndef LINUX
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#include <net/if.h>
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#if !defined(__OpenBSD__) && !defined(__NetBSD__)
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#include <net/if_var.h>
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#endif
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@ -39,13 +39,15 @@ OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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#else
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#include <sys/param.h>
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#include <sys/socket.h>
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#include <net/if.h>
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#ifndef LINUX
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#include <net/if.h>
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#if !defined(__OpenBSD__) && !defined(__NetBSD__)
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#include <net/if_var.h>
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#endif
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#include <net/if_dl.h>
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#include <net/if_types.h>
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#else
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#include <linux/if.h>
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#endif
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#ifndef BSD
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#include <net/route.h>
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@ -50,7 +50,6 @@
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'./include',
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'./src/sipcc/include',
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'./src/sipcc/cpr/include',
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'../../../ipc/chromium/src',
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'../../../ipc/chromium/src/base/third_party/nspr',
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'../../../xpcom/base',
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'../../../dom/base',
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@ -63,9 +62,6 @@
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'../trunk/webrtc/modules/interface',
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'../trunk/webrtc/peerconnection',
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'../../libyuv/include',
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'../../../netwerk/srtp/src/include',
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'../../../netwerk/srtp/src/crypto/include',
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'../../../ipc/chromium/src',
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'../../mtransport/third_party/nrappkit/src/util/libekr',
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],
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@ -160,8 +156,6 @@
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'./src/mediapipeline/MediaPipeline.cpp',
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'./src/mediapipeline/MediaPipelineFilter.h',
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'./src/mediapipeline/MediaPipelineFilter.cpp',
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'./src/mediapipeline/SrtpFlow.h',
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'./src/mediapipeline/SrtpFlow.cpp',
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],
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#
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@ -194,12 +188,28 @@
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# Conditionals
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#
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'conditions': [
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# hack so I can change the include flow for SrtpFlow
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['build_with_mozilla==1', {
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'sources': [
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'./src/mediapipeline/SrtpFlow.h',
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'./src/mediapipeline/SrtpFlow.cpp',
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],
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'include_dirs!': [
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'../trunk/webrtc',
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],
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'include_dirs': [
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'../../../netwerk/srtp/src/include',
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'../../../netwerk/srtp/src/crypto/include',
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],
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}],
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['moz_webrtc_omx==1', {
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'sources': [
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'./src/media-conduit/WebrtcOMXH264VideoCodec.cpp',
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'./src/media-conduit/OMXVideoCodec.cpp',
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],
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'include_dirs': [
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# hack on hack to re-add it after SrtpFlow removes it
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'../../webrtc/trunk/webrtc',
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'../../../content/media/omx',
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'../../../gfx/layers/client',
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],
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@ -238,6 +248,7 @@
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],
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'defines': [
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'OS_LINUX',
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'SIP_OS_LINUX',
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'_GNU_SOURCE',
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'LINUX',
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@ -252,6 +263,7 @@
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'include_dirs': [
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],
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'defines': [
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'OS_WIN',
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'SIP_OS_WINDOWS',
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'WIN32',
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'GIPS_VER=3480',
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@ -279,6 +291,7 @@
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'include_dirs': [
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],
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'defines': [
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'OS_MACOSX',
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'SIP_OS_OSX',
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'OSX',
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'_FORTIFY_SOURCE=2',
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@ -18,6 +18,8 @@
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#include "CallControlManagerImpl.h"
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#include "csf_common.h"
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#include "base/platform_thread.h"
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extern "C"
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{
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#include "config_api.h"
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@ -245,12 +245,10 @@ MediaConduitErrorCode WebrtcVideoConduit::Init(WebrtcVideoConduit *other)
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} else {
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#ifdef MOZ_WIDGET_ANDROID
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jobject context = jsjni_GetGlobalContextRef();
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// get the JVM
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JavaVM *jvm = jsjni_GetVM();
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if (webrtc::VideoEngine::SetAndroidObjects(jvm, (void*)context) != 0) {
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if (webrtc::VideoEngine::SetAndroidObjects(jvm) != 0) {
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CSFLogError(logTag, "%s: could not set Android objects", __FUNCTION__);
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return kMediaConduitSessionNotInited;
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}
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@ -104,7 +104,7 @@
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}],
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],
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}],
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['toolkit_uses_gtk==1', {
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['(toolkit_uses_gtk==1) and (build_with_mozilla==0)', {
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'dependencies': [
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'../tools/gtk_clipboard_dump/gtk_clipboard_dump.gyp:*',
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'../tools/xdisplaycheck/xdisplaycheck.gyp:*',
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@ -66,7 +66,7 @@
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['exclude', '(^|/)x11_[^/]*\\.(h|cc)$'],
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],
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}],
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['<(toolkit_uses_gtk)!=1 or >(nacl_untrusted_build)==1', {
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['(<(toolkit_uses_gtk)!=1 or >(nacl_untrusted_build)==1) and (build_with_mozilla==0)', {
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'sources/': [
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['exclude', '_gtk(_browsertest|_unittest)?\\.(h|cc)$'],
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['exclude', '(^|/)gtk/'],
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@ -14,13 +14,13 @@
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#include "webrtc/modules/interface/module.h"
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#include "webrtc/modules/video_capture/include/video_capture_defines.h"
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#ifdef ANDROID
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#if defined(ANDROID) && !defined(WEBRTC_GONK)
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#include <jni.h>
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#endif
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namespace webrtc {
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_GONK)
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int32_t SetCaptureAndroidVM(JavaVM* javaVM);
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#endif
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@ -21,7 +21,7 @@
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#include "webrtc/common_types.h"
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(MOZ_WIDGET_GONK)
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#include <jni.h>
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#endif
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@ -64,7 +64,7 @@ class WEBRTC_DLLEXPORT VideoEngine {
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// user receives callbacks for generated trace messages.
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static int SetTraceCallback(TraceCallback* callback);
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(MOZ_WIDGET_GONK)
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// Android specific.
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static int SetAndroidObjects(JavaVM* java_vm);
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#endif
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@ -155,7 +155,7 @@ int VideoEngine::SetTraceCallback(TraceCallback* callback) {
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return Trace::SetTraceCallback(callback);
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}
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD)
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#if defined(ANDROID) && !defined(WEBRTC_CHROMIUM_BUILD) && !defined(WEBRTC_GONK)
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int VideoEngine::SetAndroidObjects(JavaVM* javaVM) {
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WEBRTC_TRACE(kTraceApiCall, kTraceVideo, kModuleId,
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"SetAndroidObjects()");
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@ -83,12 +83,15 @@ stjar.javac_flags = ['-Xlint:none']
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if CONFIG['MOZ_WEBRTC']:
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video_root = TOPSRCDIR + '/media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/webrtc/videoengine/'
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video_render_root = TOPSRCDIR + '/media/webrtc/trunk/webrtc/modules/video_render/android/java/src/org/webrtc/videoengine/'
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audio_root = TOPSRCDIR + '/media/webrtc/trunk/webrtc/modules/audio_device/android/java/src/org/webrtc/voiceengine/'
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wrjar = add_java_jar('webrtc')
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wrjar.sources += [
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video_root + 'CaptureCapabilityAndroid.java',
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video_root + 'VideoCaptureAndroid.java',
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video_root + 'VideoCaptureDeviceInfoAndroid.java',
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video_render_root + 'ViEAndroidGLES20.java',
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video_render_root + 'ViERenderer.java',
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]
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wrjar.sources += [
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audio_root + 'AudioManagerAndroid.java',
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