diff options
author | Matthew Waters <matthew@centricular.com> | 2020-06-26 16:19:03 +1000 |
---|---|---|
committer | Matthew Waters <matthew@centricular.com> | 2020-08-19 20:42:16 +1000 |
commit | 8b4d1567124a5ad6999cf00b90721bfdc659c261 (patch) | |
tree | 547b05319dd74586bd233cfdc46625174ea480ec | |
parent | 101d9965e54d380aa76f1c88372fc3374369ed9e (diff) |
webrtc/android: initialize the debug category
Fixes possible critical/crash on startup
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/25>
-rw-r--r-- | webrtc/android/app/src/main/jni/Android.mk | 2 | ||||
-rw-r--r-- | webrtc/android/app/src/main/jni/webrtc.c | 4 |
2 files changed, 4 insertions, 2 deletions
diff --git a/webrtc/android/app/src/main/jni/Android.mk b/webrtc/android/app/src/main/jni/Android.mk index c0ab0f8..326781d 100644 --- a/webrtc/android/app/src/main/jni/Android.mk +++ b/webrtc/android/app/src/main/jni/Android.mk @@ -31,7 +31,7 @@ GSTREAMER_NDK_BUILD_PATH := $(GSTREAMER_ROOT)/share/gst-android/ndk-build/ include $(GSTREAMER_NDK_BUILD_PATH)/plugins.mk -GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc volume +GSTREAMER_PLUGINS_CORE_CUSTOM := coreelements app audioconvert audiorate audioresample videoconvert videorate videoscale videotestsrc volume GSTREAMER_PLUGINS_CODECS_CUSTOM := videoparsersbad vpx opus audioparsers opusparse androidmedia GSTREAMER_PLUGINS_NET_CUSTOM := tcp rtsp rtp rtpmanager udp srtp webrtc dtls nice GSTREAMER_PLUGINS := $(GSTREAMER_PLUGINS_CORE_CUSTOM) $(GSTREAMER_PLUGINS_CODECS_CUSTOM) $(GSTREAMER_PLUGINS_NET_CUSTOM) \ diff --git a/webrtc/android/app/src/main/jni/webrtc.c b/webrtc/android/app/src/main/jni/webrtc.c index e5b47fc..7a30dd9 100644 --- a/webrtc/android/app/src/main/jni/webrtc.c +++ b/webrtc/android/app/src/main/jni/webrtc.c @@ -822,6 +822,8 @@ native_class_init (JNIEnv * env, jclass klass) __android_log_print (ANDROID_LOG_ERROR, "GstPlayer", "%s", message); (*env)->ThrowNew (env, exception_class, message); } + GST_DEBUG_CATEGORY_INIT (debug_category, "webrtc", 0, + "GStreamer Android WebRTC"); //gst_debug_set_threshold_from_string ("gl*:7", FALSE); } @@ -905,7 +907,7 @@ JNI_OnLoad (JavaVM * vm, void *reserved) java_vm = vm; if ((*vm)->GetEnv (vm, (void **) &env, JNI_VERSION_1_4) != JNI_OK) { - __android_log_print (ANDROID_LOG_ERROR, "GstPlayer", + __android_log_print (ANDROID_LOG_ERROR, "GstWebRTC", "Could not retrieve JNIEnv"); return 0; } |