Guys. I need some help with WebRTC. I am developing an Android app that streams from a Service to a AngularJS client using Node.Js and Express as a signaling server. And I am using libjingle packaged by pristine.
Every time I stop the peer from the client it gives me the following error:
09-10 01:43:06.161 5935-5935/org.myapp E/rtc﹕ #
# Fatal error in ../../talk/app/webrtc/java/jni/peerconnection_jni.cc, line 956
# Check failed: 0 == (reinterpret_cast<MediaStreamTrackInterface*>(j_p))->Release() (0 vs. 1)
# Unexpected refcount.
#
I have absolutely no idea of what's going. I have spoken with one developer from pristine and he said that this error is not really seen frequently.
Did anybody ever run into it?
Tracing back: The call site invokes CHECK_RELEASE, which verifies that the last reference is being released. So, it looks like you've got an extra reference to the MediaStreamTrackInterface.
Related
I am developing app of video chat in Web RTC.
I am going through documentation of Web RTC.
I have followed all steps mentioned for Getting the Code
export GYP_DEFINES="OS=android"
fetch --nohooks webrtc_android
gclient sync
I have all src code of web rtc in my root directory. As per documentation say we have to compile source to use in our application. Here i got error while compiling source using following command.
$ gn gen out/Default --args='target_os="android" target_cpu="arm"'
ERROR at //build/config/android/internal_rules.gni:636:3: Can't load input file.
import("//third_party/ijar/ijar.gni")
^-----------------------------------
Unable to load:
/Users/c86/webrtc-checkout/src/third_party/ijar/ijar.gni
I also checked in the secondary tree for:
/Users/c86/webrtc-checkout/src/build/secondary/third_party/ijar/ijar.gni
See //build/config/android/rules.gni:8:1: whence it was imported.
import("//build/config/android/internal_rules.gni")
^-------------------------------------------------
See //webrtc/examples/BUILD.gn:12:3: whence it was imported.
import("//build/config/android/rules.gni")
^----------------------------------------
See //BUILD.gn:13:5: which caused the file to be included.
"//webrtc/examples",
^------------------
How can I solve this? Am I going in right direction to develop app of video chat on Web RTC?
Building WebRTC for Android is only supported on Ubuntu(14 / 16).
Currently not supported in macOS.
I installed the QPython app on Android and executed a script name helloworld.py. It used androidhelper's method maketoast to show a message with the Android api.
I tried writing the exact same script on another file but it is giving me the following error:
com.googlecode.android_scripting.rpc.RpcError: Unknown RPC.
Then I tried to modify helloworld.py and it's now giving me the above error.
I searched all over the internet for this error but no solutions.
By the way, I use a rooted Samsung Galaxy Tab 3 LITE T110N.
I already tried using the android module, but it doesn't exist. So does sl4a doesn't exist.
I use python 2.7.
import androidhelper
droid=androidhelper.Android()
droid.makeToast("Hello, sl4a!")
#not maketoast
#it's makeToast!
#read this:
#http://www.mithril.com.au/android/doc/index.html
#when you call sl4a,you have to tell sl4a what to do, if sl4a can't recognize it,
#sl4a throws an Unknown RPC error
#Thanks
I am trying to provision the Nymi band from my android app. I am using the sample code from the Nymi Android SDK 2.0 (Can be downloaded from here:). I removed all the code related to the Nymulator(Nymi band emulator). I want to test the provisioning with the real Nymi band. However,when I click the provision button, the app get crashed. I tried to debug it with the exception breakpoints and I got the "getsockopt errno 88" error in my debugger:
I am attaching the screenshot too.
I did some research on google , and found that the errno 88 is
#define ENOTSOCK 88 /* Socket operation on non-socket */
During the app launch, I see the following message in the Debug Console:
Connected to the target VM, address: 'localhost:8604', transport: 'socket'
which I think,the connection to socket is fine. I am not sure,what makes the socket connection go wrong later.
This is what I see in my LOgCat after I click the provision button:(I am posting only some part of the error message,because the error message in Logcat is way too long)
Edit: I added some breakpoints,and I found where the app is crashing. I found this code:
try {
System.loadLibrary("_nymi_sdk_net");
_loaded = true;
} catch (UnsatisfiedLinkError var6) {
return false;
}
At system.load library, I am getting the error.
Edit 2: I changed the libraries from net to native ,as net is for emulators and native is for device. This time new error occurs at the same line:System.loadLibrary("_nymi_sdk");
Screenshot:
(Disclaimer: I know nothing of this SDK or the band, and this answer is just a speculation. And I've deleted my earlier answer as it was a missed shot in the dark.)
There is a minor hint in the SDK's README.md:
Prerequisites:
Android Development:
[...]
To use the BasicExample app with a Nymi Band, the NCL library in Examples\BasicExample\libs must be replaces with the native library in
NCL\native\libs
You must remove all files and folders from BasicExample\libs and copy all files from NCL\native\libs there instead.
(I wonder why they didn't emphasize this much harder...)
So the libs folder should contain the following folders and files:
When in a call using the Sinch SDK on Android, the following error is thrown:
09-14 17:06:38.223 28454-28484/com.callcasual.callcasual E/sinch-android-rtc﹕ ERROR ; (17: 6:38:237 | 1) 28493; (voe_audio_processing_impl.cc:857): GetEcMetricsStatus: not supported
When searching for that error it appears to be an issue with WebRTC, however I cannot find a way to solve the issue.
This is only indicating that echo metrics are not available, the voice will go through fine during the call anyway. Why WebRTC logs this as an error I'm not sure but we are looking at disabling this in a future release of the Sinch SDK.
I'm using this FFMPEG port for Android.
My Goal is to read RTSP stream eventually.
I can open a stream from a local file (located at "/storage/emulated/0/bunny.mp4") ,
using *avformat_open_input* function.
However when I try to open the same file from remote location like *"http_://192.168.1.183/bunny.mp4*" , it returns error -2 .
The whole cpp code is here.
Any ideas ?
Check if you have INTERNET permission in your manifest.
Also check that you are building ffmpeg with tcp and rtsp protocol enabled. And codecs enabled, too. I believe FFMPEG's configure prints this info when building and also it's in the build log.
Also, there's a FFMPEG function to print error message by code, don't remember the name though. So you can call it and see what -2 error means.
This here project has FFMPEG configured correctly https://github.com/dannoy/dolphin-player look in the p folder