(libjingle signaling + webrtc voice engine)
Discussion: webrtc-jingle
- Working example android and ios apps of libjingle and webrtc voice backend.
- Based on libjingle trunk and webrtc trunk updated on regular intervals.
- Added improvements for stability and missing pieces for mobile implementation.
- Can make calls between two phones, or between gmail and a phone.
- Happy for any help, please see tickets, and send a pull request.
- Download and install depot_tools
# mkdir webrtcjingleproject
# cd webrtcjingleproject
# gclient config https://github.com/lukeweber/webrtc-jingle-client.git --name trunk
# gclient sync
or for an older stable build, take the head of the stable branch revision.
# gclient sync --revision PUT_STABLE_HEAD_REV_HERE
- android NDK r8e
- Android SDK
- eclipse
- Maven v3.0.3+
- Add the following to your environment, i.e. .bashrc or .bash_profile
#Android
export ANDROID_SDK_ROOT=/path/to/sdk/
export ANDROID_NDK_ROOT=/path/to/ndk/
export PATH=$PATH:$ANDROID_NDK_ROOT:$ANDROID_SDK_ROOT:$ANDROID_SDK_ROOT/platform-tools
#mvn variables
export ANDROID_HOME=$ANDROID_SDK_ROOT
export ANDROID_NDK_HOME=$ANDROID_NDK_ROOT
- Build, deploy to phone, and start debugger in one script: tools/badit_android.py
- Build the apks: cd trunk/android && mvn install
- To run a debugger: build/android/gdb_apk -p com.tuenti.voice.example -s VoiceClientService -l android/voice-client-native/obj/${build_profile}/local/${app_abi}
- For debugging the ndk compile outside of maven, I sometimes prefer and use a light shell wrapper instead of maven for building only the c++ project. trunk/android/voice-client-native && ./build.sh
- Build debug code jni in debug mode: cd trunk/android/voice-client-core && ./build.sh
- Generate unittest apk: tools/gen_tests_apk.sh
- Install unittest : adb install -r adb install -r voice_testing/${app_abi}/${lib}/${lib}-debug.apk
- Prepare
- Run unittest:
# adb shell mkdir /sdcard/talk
# adb shell am start -n org.chromium.native_test/org.chromium.native_test.ChromeNativeTestActivity
# See adb logcat adb | grep libjingle
- Fetch unittest logs: adb pull /sdcard/talk talk-logs
- NO UI YET, need more changes in libjingle core to make this work.
- DEPS build on linux and mac.
- Need to wire the java code in example app, using third_party/webrtc/video_engine/test/android/src/org/webrtc/videoengineapp/WebRTCDemo.java as a template.
- webrtcvideoengine.cc will certainly need changes, as will VideoRenderer to enable passing a java ref down to webrtc, contact me if you want to give this a shot.
# cd android/voice-client-core/
# ln -s [insert_full_path_here]/trunk/third_party/libvpx/source/libvpx jni/libvpx
# cd third_party/libvpx/source
# git pull https://gerrit.chromium.org/gerrit/webm/libvpx refs/changes/99/41299/1
# libvpx/configure --target=armv7-android-gcc --disable-examples --sdk-path=$ANDROID_NDK_ROOT --enable-error-concealment --enable-realtime-only --disable-vp9 --enable-pic
# Open jni/libvpx/build/make/Android.mk => change BUILD_SHARED_LIBRARY to BUILD_STATIC_LIBRARY
#
- OSX machine
- Download the latest xcode and command line tools.
- Apply the following patch to third_party/expat
Index: expat.gyp
===================================================================
--- expat.gyp (revision 169394)
+++ expat.gyp (working copy)
@@ -7,7 +7,7 @@
'conditions': [
# On Linux, we implicitly already depend on expat via fontconfig;
# let's not pull it in twice.
- ['os_posix == 1 and OS != "mac" and OS != "android"', {
+ ['os_posix == 1 and OS != "mac" and OS != "android" and OS != "ios"', {
'use_system_expat%': 1,
}, {
'use_system_expat%': 0,- Change your .gclient file in trunk/../.gclient
--- .gclient
+++ .gclient
@@ -8,3 +8,4 @@
"safesync_url": "",
},
]
+target_os = ['ios']- Run gclient sync again to fetch xmppframework.
- Autogenerate an xcode project with gyp with the following command:
./build/gyp_chromium --depth=. -DOS=ios -Dinclude_tests=0 -Denable_protobuf=0 -Denable_video=0 webrtcjingle.gyp
- open trunk/webrtcjingle.xcodeproj
- Modify myJID, and myPassword in AppDelegate.mm.
- Modify user you wish to call in ios/VoiceClientExample/ViewController.mm => [appDelegate call:@"user@gmail.com"];
- If using an xmpp server make sure to change the flag isGtalk in login in VoiceClientDelegate.mm.
- In xcode, build and deploy
- May experience issues about sse from audio_processing.gypi. If you push to an IOS device add -Dtarget_arch=arm. If emulator, the other command will probably work.
- Opus is currently alpha in implementation, is hard set to 48kHz in webrtc. Android mic is set to 16kHz, meaning you'll probably upsample/downsample all audio by 3x.
- Add "WEBRTC_BUILD_WITH_OPUS := true" to android/voice-client-core/jni/Android.mk
- Modify offer kCodecPrefs in third_party/libjingle/talk/media/webrtc/webrtcvoiceengine.cc to include OPUS and exclude ISAC.
- Native code won't build on a windows machine.
- VM with Ubuntu 64bit Linux
- Recommended disk of at least 3GB. Current build cache is approx, 1.5GB.
- 64bit Android Linux NDK required.
- Video from jreyes https://www.youtube.com/watch?v=f0NU-E8l_qQ