Move talk/app/webrtc to webrtc/api
The previously disabled warnings that were inherited from
talk/build/common.gypi are now replaced by target-specific disabling
of only the failing warnings. Additional disabling was needed since the stricter
compilation warnings that applies to code in webrtc/.
License headers will be updated in a follow-up CL.
Other modifications:
* Updated the header guards.
* Sorted the includes using chromium/src/tools/sort-headers.py
except for these files:
talk/app/webrtc/peerconnectionendtoend_unittest.cc
talk/app/webrtc/java/jni/androidmediadecoder_jni.cc
talk/app/webrtc/java/jni/androidmediaencoder_jni.cc
webrtc/media/devices/win32devicemanager.cc
The HAVE_SCTP define was added for the peerconnection_unittests target
in api_tests.gyp.
I also checked that none of
SRTP_RELATIVE_PATH
HAVE_SRTP
HAVE_WEBRTC_VIDEO
HAVE_WEBRTC_VOICE
were used by the talk/app/webrtc code.
For Chromium, the following changes will need to be applied to the roll CL that updates the
DEPS for WebRTC and libjingle:
https://codereview.chromium.org/1615433002
BUG=webrtc:5418
NOPRESUBMIT=True
R=deadbeef@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1610243002 .
Cr-Commit-Position: refs/heads/master@{#11545}
diff --git a/webrtc/api/test/androidtestinitializer.cc b/webrtc/api/test/androidtestinitializer.cc
new file mode 100644
index 0000000..17118c0
--- /dev/null
+++ b/webrtc/api/test/androidtestinitializer.cc
@@ -0,0 +1,74 @@
+/*
+ * libjingle
+ * Copyright 2015 Google Inc.
+ *
+ * Redistribution and use in source and binary forms, with or without
+ * modification, are permitted provided that the following conditions are met:
+ *
+ * 1. Redistributions of source code must retain the above copyright notice,
+ * this list of conditions and the following disclaimer.
+ * 2. Redistributions in binary form must reproduce the above copyright notice,
+ * this list of conditions and the following disclaimer in the documentation
+ * and/or other materials provided with the distribution.
+ * 3. The name of the author may not be used to endorse or promote products
+ * derived from this software without specific prior written permission.
+ *
+ * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
+ * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
+ * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
+ * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
+ * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
+ * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
+ * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
+ * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
+ * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
+ * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
+ */
+
+#include "webrtc/api/test/androidtestinitializer.h"
+
+#include <pthread.h>
+
+// Note: this dependency is dangerous since it reaches into Chromium's base.
+// There's a risk of e.g. macro clashes. This file may only be used in tests.
+// Since we use Chromes build system for creating the gtest binary, this should
+// be fine.
+#include "base/android/context_utils.h"
+#include "base/android/jni_android.h"
+
+#include "webrtc/api/java/jni/classreferenceholder.h"
+#include "webrtc/api/java/jni/jni_helpers.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/ssladapter.h"
+#include "webrtc/voice_engine/include/voe_base.h"
+
+namespace webrtc {
+
+namespace {
+
+static pthread_once_t g_initialize_once = PTHREAD_ONCE_INIT;
+
+// There can only be one JNI_OnLoad in each binary. So since this is a GTEST
+// C++ runner binary, we want to initialize the same global objects we normally
+// do if this had been a Java binary.
+void EnsureInitializedOnce() {
+ RTC_CHECK(::base::android::IsVMInitialized());
+ JNIEnv* jni = ::base::android::AttachCurrentThread();
+ JavaVM* jvm = NULL;
+ RTC_CHECK_EQ(0, jni->GetJavaVM(&jvm));
+ jobject context = ::base::android::GetApplicationContext();
+
+ RTC_CHECK_GE(webrtc_jni::InitGlobalJniVariables(jvm), 0);
+ RTC_CHECK(rtc::InitializeSSL()) << "Failed to InitializeSSL()";
+ webrtc_jni::LoadGlobalClassReferenceHolder();
+
+ webrtc::VoiceEngine::SetAndroidObjects(jvm, context);
+}
+
+} // anonymous namespace
+
+void InitializeAndroidObjects() {
+ RTC_CHECK_EQ(0, pthread_once(&g_initialize_once, &EnsureInitializedOnce));
+}
+
+} // namespace webrtc