I'm failing using the build scripts. After cloning the repo I ran through the steps form the README.

> git clone https://github.com/pristineio/webrtc-build-scripts.git
> cd webrtc-build-scripts
> source ios/build.sh
> export WEBRTC_DEBUG=true
> dance
Get the current working directory so we can change directories back when done
If no directory where depot tools should be...
Make directory for gclient called Depot Tools
Pull the depot tools project from chromium source into the depot tools directory
Cloning into '/Users/tobias/Projekte/webrtc-build-scripts/depot_tools'...
remote: Sending approximately 13.43 MiB ...
remote: Total 11922 (delta 7651), reused 11922 (delta 7651)
Receiving objects: 100% (11922/11922), 13.43 MiB | 395.00 KiB/s, done.
Resolving deltas: 100% (7652/7652), done.
Checking connectivity... done.
Go back to working directory
Get the current working directory so we can change directories back when done
If no directory where depot tools should be...
Change directory into the depot tools
Pull the depot tools down to the latest
Already up-to-date.
Go back to working directory
Get the current working directory so we can change directories back when done
If no directory where depot tools should be...
Change directory into the depot tools
Pull the depot tools down to the latest
Already up-to-date.
Go back to working directory
choose_code_signing:1: = not found

After this the depot_tools are updated/downloaded correctly but the webrtc sources have not been checked out (the directory "webrtc" is empty).

I am grateful for every tip.

build_apprtc for Android causes "die: error: must run as root" for ninja

Here what I see in logs:

 Build AppRTCDemo in  (arch: x86_64) mode
 log: ninja version 0.1.3 initializing
 die: error: must run as root

Looks like something wrong with some paths in
https://github.com/pristineio/webrtc-build-scripts/blob/master/android/build.sh

I receive a lot of errors when call build_apprtc:

  /vagrant/webrtc/src/build/android/envsetup.sh: No such file or directory

etc

I've pulled down the build scripts from the master branch today (so that would be commit 7570628). I'm trying to build for iOS. Both get_webrtc and build_webrtc are generating the message below (illegal primary in regular expression). Hopefully it's not harmful.

.....
Get the current working directory so we can change directories back when done
If no directory where depot tools should be...
Change directory into the depot tools
Pull the depot tools down to the latest
Already up-to-date.
Go back to working directory
awk: illegal primary in regular expression ) |" at |"
input record number 1, file
source line number 1
Using code signing identity
....

Is there a workaround for building the android libs on my OS? This is as far as I get (with timer incremeting forever...) when I execute get_webrtc:

________ running '/usr/bin/python -c import os,sys;script = os.path.join("trunk","check_root_dir.py");_ = os.system("%s %s" % (sys.executable,script)) if os.path.exists(script) else 0' in '/home/mondain/workspace/github/webrtc-build-scripts/android/webrtc'

________ running '/usr/bin/python -u src/sync_chromium.py --target-revision 601e6f32704fed600914fd5198c24395c4dc36f5' in '/home/mondain/workspace/github/webrtc-build-scripts/android/webrtc'
Running "gclient sync --force --revision src@601e6f32704fed600914fd5198c24395c4dc36f5 --gclientfile .gclient.tmp --delete_unversioned_trees --reset --upstream --deps=unix,android" in /home/mondain/workspace/github/webrtc-build-scripts/android/webrtc/src/chromium
[0:01:00] Still working on:
[0:01:00]   src
[0:01:10] Still working on:
[0:01:10]   src
[0:01:20] Still working on:
[0:01:20]   src
[0:01:30] Still working on:
[0:01:30]   src
[0:01:40] Still working on:
[0:01:40]   src
[0:01:50] Still working on:
[0:01:50]   src

I downloaded the pod revision 7730.2.0 but after the pod install command ended I got a libWebRTC.a that is only 352.5mb and it's only built for i386, not armv7. Anything that I am missing?

Firstly, not all of the dependencies seem to be automatically installed. After running get_webrtc I got the following

________ running '/usr/bin/python src/webrtc/build/gyp_webrtc -Dextra_gyp_flag=0' in '/home/vagrant/webrtc'
Updating projects from gyp files...
Package glib-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `glib-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'glib-2.0' found
Package gobject-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gobject-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gobject-2.0' found
Package gthread-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gthread-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gthread-2.0' found
Package gtk+-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gtk+-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gtk+-2.0' found
gyp: Call to 'pkg-config --libs-only-l glib-2.0 gobject-2.0 gthread-2.0 gtk+-2.0' returned exit status 1.
Package gobject-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gobject-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gobject-2.0' found
Package gthread-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gthread-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gthread-2.0' found
Package gtk+-2.0 was not found in the pkg-config search path.
Perhaps you should add the directory containing `gtk+-2.0.pc'
to the PKG_CONFIG_PATH environment variable
No package 'gtk+-2.0' found
gyp: Call to 'pkg-config --libs-only-l gobject-2.0 gthread-2.0 gtk+-2.0' returned exit status 1.
Error: Command /usr/bin/python src/webrtc/build/gyp_webrtc -Dextra_gyp_flag=0 returned non-zero exit status 1 in /home/vagrant/webrtc

To resolve I ran /webrtc/src/build/install-build-deps-android.sh
After doing this I was able to run get_webrtc without any errors. I then tried build_apprtc. The following is the output

vagrant@precise64:~$ export WEBRTC_DEBUG=true
vagrant@precise64:~$ build_apprtc 
Change directory into the depot tools
Pull the depot tools down to the latest
Already up-to-date.
Setting up build environment for Android
Export the base settings of GYP_DEFINES so we can define how we want to build
User has not specified any gyp defines so we proceed with default
GYP_DEFINES=OS=android host_os=linux libjingle_java=1 build_with_libjingle=1 build_with_chromium=0 enable_tracing=1 enable_android_opensl=1 target_arch=arm arm_neon=1 armv7=1
Run gclient hooks

________ running '/usr/bin/python -c import os,sys;script = os.path.join('trunk','check_root_dir.py');_ = os.system('%s %s' % (sys.executable,script)) if os.path.exists(script) else 0' in '/home/vagrant/webrtc'

________ running '/usr/bin/python -u src/sync_chromium.py --target-revision de13cf44ff5aa5815dd742c2cedcaece02601f4b' in '/home/vagrant/webrtc'
Chromium already up to date: de13cf44ff5aa5815dd742c2cedcaece02601f4b

________ running '/usr/bin/python src/setup_links.py' in '/home/vagrant/webrtc'

________ running 'download_from_google_storage --directory --recursive --num_threads=10 --no_auth --bucket chromium-webrtc-resources src/resources' in '/home/vagrant/webrtc'

________ running '/usr/bin/python src/webrtc/build/gyp_webrtc -Dextra_gyp_flag=0' in '/home/vagrant/webrtc'
Updating projects from gyp files...
Hook '/usr/bin/python src/webrtc/build/gyp_webrtc -Dextra_gyp_flag=0' took 12.54 secs
Build AppRTCDemo in Debug (arch: arm) mode
ninja: Entering directory `out/Debug/'
[629/2043] CC obj/chromium/src/third_party/openmax_dl/dl/sp/src/arm/openmax_dl_armv7.detect.o
FAILED: /home/vagrant/webrtc/src/third_party/android_tools/ndk//toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/bin/arm-linux-androideabi-gcc -MMD -MF obj/chromium/src/third_party/openmax_dl/dl/sp/src/arm/openmax_dl_armv7.detect.o.d -DV8_DEPRECATION_WARNINGS -D_FILE_OFFSET_BITS=64 -DNO_TCMALLOC -DDISABLE_NACL -DDL_ARM_NEON -DCHROMIUM_BUILD -DUSE_LIBJPEG_TURBO=1 -DENABLE_WEBRTC=1 -DUSE_PROPRIETARY_CODECS -DENABLE_BROWSER_CDMS -DENABLE_CONFIGURATION_POLICY -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DENABLE_EGLIMAGE=1 -DENABLE_AUTOFILL_DIALOG=1 -DCLD_VERSION=1 -DENABLE_PRINTING=1 -DENABLE_MANAGED_USERS=1 -DVIDEO_HOLE=1 -DENABLE_LOAD_COMPLETION_HACKS=1 -DUSE_OPENSSL=1 -DUSE_OPENSSL_CERTS=1 -DANDROID -D__GNU_SOURCE=1 -DUSE_STLPORT=1 -D_STLP_USE_PTR_SPECIALIZATIONS=1 '-DCHROME_BUILD_ID=""' -DHAVE_SYS_UIO_H -DDYNAMIC_ANNOTATIONS_ENABLED=1 -DWTF_USE_DYNAMIC_ANNOTATIONS=1 -D_DEBUG -I../../chromium/src/third_party/openmax_dl -Igen -I../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures -fstack-protector --param=ssp-buffer-size=4 -fno-strict-aliasing -Wno-unused-parameter -Wno-missing-field-initializers -fvisibility=hidden -pipe -fPIC -Wno-unused-local-typedefs -Wno-format -march=armv7-a -mtune=generic-armv7-a -mfloat-abi=softfp -mthumb -fno-tree-sra -fno-caller-saves -Wno-psabi -mthumb-interwork -ffunction-sections -funwind-tables -g -fstack-protector -fno-short-enums -finline-limit=64 -Wa,--noexecstack --sysroot=/home/vagrant/webrtc/src/third_party/android_tools/ndk//platforms/android-14/arch-arm -isystem/home/vagrant/webrtc/src/third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport -Os -g -gdwarf-4 -fdata-sections -ffunction-sections -fomit-frame-pointer -funwind-tables   -c ../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c -o obj/chromium/src/third_party/openmax_dl/dl/sp/src/arm/openmax_dl_armv7.detect.o
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c: In function 'SetFFTRoutines':
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:29:29: error: lvalue required as left operand of assignment
     omxSP_FFTFwd_RToCCS_F32 = omxSP_FFTFwd_RToCCS_F32_Sfs;
                             ^
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:30:29: error: lvalue required as left operand of assignment
     omxSP_FFTInv_CCSToR_F32 = omxSP_FFTInv_CCSToR_F32_Sfs;
                             ^
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:34:31: error: 'omxSP_FFTFwd_RToCCS_F32_Sfs_vfp' undeclared (first use in this function)
     omxSP_FFTFwd_RToCCS_F32 = omxSP_FFTFwd_RToCCS_F32_Sfs_vfp;
                               ^
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:34:31: note: each undeclared identifier is reported only once for each function it appears in
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:35:31: error: 'omxSP_FFTInv_CCSToR_F32_Sfs_vfp' undeclared (first use in this function)
     omxSP_FFTInv_CCSToR_F32 = omxSP_FFTInv_CCSToR_F32_Sfs_vfp;
                               ^
In file included from ../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:15:0:
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c: At top level:
../../chromium/src/third_party/openmax_dl/dl/sp/api/omxSP.h:2553:33: error: 'omxSP_FFTFwd_RToCCS_F32_Sfs' redeclared as different kind of symbol
 #define omxSP_FFTFwd_RToCCS_F32 omxSP_FFTFwd_RToCCS_F32_Sfs
                                 ^
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:77:13: note: in expansion of macro 'omxSP_FFTFwd_RToCCS_F32'
 OMXResult (*omxSP_FFTFwd_RToCCS_F32)(
             ^
../../chromium/src/third_party/openmax_dl/dl/sp/api/omxSP.h:2293:11: note: previous declaration of 'omxSP_FFTFwd_RToCCS_F32_Sfs' was here
 OMXResult omxSP_FFTFwd_RToCCS_F32_Sfs(
           ^
../../chromium/src/third_party/openmax_dl/dl/sp/api/omxSP.h:2554:33: error: 'omxSP_FFTInv_CCSToR_F32_Sfs' redeclared as different kind of symbol
 #define omxSP_FFTInv_CCSToR_F32 omxSP_FFTInv_CCSToR_F32_Sfs
                                 ^
../../chromium/src/third_party/openmax_dl/dl/sp/src/arm/detect.c:82:13: note: in expansion of macro 'omxSP_FFTInv_CCSToR_F32'
 OMXResult (*omxSP_FFTInv_CCSToR_F32)(
             ^
../../chromium/src/third_party/openmax_dl/dl/sp/api/omxSP.h:2526:11: note: previous declaration of 'omxSP_FFTInv_CCSToR_F32_Sfs' was here
 OMXResult omxSP_FFTInv_CCSToR_F32_Sfs(
           ^
[629/2043] CC obj/chromium/src/third_party/usrsctp/usrsctplib/netinet/usrsctplib.sctp_usrreq.o
ninja: build stopped: subcommand failed.
Debug build for apprtc failed for revision 7456

Am I doing something wrong or is this something that needs to be fixed?

I'm using the provided build scrips (thanks by the way!) to build a WebRTC library and I'm adding into my own project. When I try to include arm64 in the valid architectures for the project, I receive linker warnings regarding missing symbols for x86_64. I think that is because the simulator uses x86_64 (I'm pretty new to iOS...) the library built by default seems to only have arm7, arm64, and i386. Is it possible to get it built and working for arm64/x86_64 in an iOS project?

As background, I think Apple is going to start requiring projects use arm64 and want to make sure my project conforms to their standard.

The current most recent build in cocoapods--7298--hits a crash in libvpx. Is there a way to get a more recent build, or are more recent builds also hitting that same issue?

The current pod exposes all symbols in the webrtc project such as nss/sha512.cc containing symbols such as _SHA256_Update. Those symbols are susceptible to conflict with other libraries such as openssl.

Is it possible to just expose ObjectC related symbols in the pod distribution?

Thanks.

After cloning the repo, I ran some commands according to the README,
'dance' script told some error messages.
What was wrong?

$ git clone https://github.com/pristineio/webrtc-build-scripts.git
$ export WEBRTC_DEBUG=true
$ cd webrtc-build-scripts/
$ source ios/build.sh
$ dance
...
________ running 'download_from_google_storage --no_resume --platform=darwin --no_auth --bucket chromium-gn -s src/buildtools/mac/gn.sha1' in '/Users/nishimura/webrtc-build-scripts/ios/webrtc/src/chromium'
Error: Command download_from_google_storage --no_resume --platform=darwin --no_auth --bucket chromium-gn -s src/buildtools/mac/gn.sha1 returned non-zero exit status 1 in /Users/nishimura/webrtc-build-scripts/ios/webrtc/src/chromium
File gs://chromium-gn/f62a6b6e14c39aa4e4bc2a19a2184877c9488e1e for src/buildtools/mac/gn does not exist.
0> File gs://chromium-gn/f62a6b6e14c39aa4e4bc2a19a2184877c9488e1e for src/buildtools/mac/gn does not exist, skipping.
Error: Command /usr/local/Cellar/python/2.7.9/Frameworks/Python.framework/Versions/2.7/Resources/Python.app/Contents/MacOS/Python -u src/sync_chromium.py --target-revision 601e6f32704fed600914fd5198c24395c4dc36f5 returned non-zero exit status 2 in /Users/nishimura/webrtc-build-scripts/ios/webrtc
Hook '/usr/local/Cellar/python/2.7.9/Frameworks/Python.framework/Versions/2.7/Resources/Python.app/Contents/MacOS/Python -u src/sync_chromium.py --target-revision 601e6f32704fed600914fd5198c24395c4dc36f5' took 27.44 secs
mkdir: created directory '/Users/nishimura/webrtc-build-scripts/ios/webrtc/libjingle_peerconnection_builds'
Running ninja
ninja: Entering directory `out_ios_armeabi_v7a/Debug-iphoneos/'
ninja: fatal: chdir to 'out_ios_armeabi_v7a/Debug-iphoneos/' - No such file or directory
Running libtool
error: /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool: can't open file: /Users/nishimura/webrtc-build-scripts/ios/webrtc/src/out_ios_armeabi_v7a/Debug-iphoneos/*.a (No such file or directory)
.gclient file in parent directory /Users/nishimura/webrtc-build-scripts/ios/webrtc might not be the file you want to use

I don't know how to configure for 64 bit in iOS

Is this correct?

pod "libjingle_peerconnection"

post_install do |installer_representation|
installer_representation.project.targets.each do |target|
target.build_configurations.each do |config|
config.build_settings['ONLY_ACTIVE_ARCH'] = 'NO'
config.build_settings['VALID_ARCHS'] = ['armv7', 'i386','arm64']
end
end
end

Hi,

I tried today to build iOS version of libs. It worked about a week ago.

Today I got:

@@@@@@@@@@@@@@@@@@@
A C T I O N R E Q I R E D
@@@@@@@@@@@@@@@@@@@
It looks like you have a legacy checkout where the solution's top-level
directory is named 'trunk'. From now on, it must be named 'src'.

What you need to do is to:

1. Edit your .gclient file and change the solution name from 'trunk' to 'src'
2. Rename your 'trunk' directory to 'src'
3. Re-run gclient sync (or gclient runhooks)

I tried to edit .gclient as error message suggest, but it looks this file is generated by script.

Any idea what to change?

Thanks,

Jindrich

Hi,
This problem of downloading all of chromium seems to have become worse.
We can't get the Android to ever complete while running in Vagrant image.
iOS tends to fail most of the time too and sometimes completes after 4 hours or so.

Do you have any suggestions on improving this?
Perhaps not using --force in the chromium bit so it can be re-run if it stalls?

Can you please try the Android Vagrant build on a Mac and see if you get the same problems.

Hi,

I tried cocoapods version of lib and it works fine with demo project. However, when linking in my project, I get:

ld: b/bl/blx thumb2 branch out of range (35827308 max is +/-16MB): from _WebRtcIlbcfix_CbSearch (0x00123138) to _WebRtcSpl_CrossCorrelation (0x0234E990) in '_WebRtcIlbcfix_CbSearch'

Any idea, how to fix this issue?

Thanks,

Jindrich

I've tried to follow the README.md guide that you guys have on Mac Yosemite and Ubuntu 14.04. After executing build_apprtc I see such messages:

[3:20:19]   src/third_party/android_tools
[3:20:29] Still working on:
[3:20:29]   src/third_party/android_tools
[3:20:39] Still working on:
[3:20:39]   src/third_party/android_tools
[3:20:49] Still working on:
[3:20:49]   src/third_party/android_tools
[3:20:59] Still working on:
[3:20:59]   src/third_party/android_tools
[3:21:09] Still working on:
[3:21:09]   src/third_party/android_tools
[3:21:19] Still working on:
[3:21:19]   src/third_party/android_tools
[3:21:29] Still working on:
[3:21:29]   src/third_party/android_tools
[3:21:39] Still working on:
[3:21:39]   src/third_party/android_tools
[3:21:49] Still working on:

That goes on forever and is never done. Am I missing something?

Hi. I was using webrtc-build-scripts for updating webrtc library for a while.
About 15h ago AppRTCDemo(IOS) updated to WebSocket based signaling(r7852).
With this change AppRTCDemo has some merge issues.
Do I need to download entire repository with using webrtc-build-scripts?
And another question.
Will webrtc-build-scripts upgrade for r7852 (if it needs to)?

I'm using libWebRTC in a project, where I need to make some small changes to the Objective-C wrapper. I would like to do the change locally in the build step as a patch (and submit it later to the official repo). What is the best way to include this automatically in the build process?

Hi,

Compile with XCode 6.1.1, an idea?

________ running '/usr/bin/python src/webrtc/build/gyp_webrtc -Dextra_gyp_flag=0' in '/Users/robert/Dev/Web/webrtc-build-scripts-master/ios/webrtc'
Updating projects from gyp files...
Traceback (most recent call last):
File "src/webrtc/build/gyp_webrtc", line 100, in
gyp_rc = gyp.main(args)
File "/Users/robert/Dev/Web/webrtc-build-scripts-master/ios/webrtc/src/chromium/src/tools/gyp/pylib/gyp/init.py", line 525, in main
return gyp_main(args)
File "/Users/robert/Dev/Web/webrtc-build-scripts-master/ios/webrtc/src/chromium/src/tools/gyp/pylib/gyp/init.py", line 510, in gyp_main
generator.GenerateOutput(flat_list, targets, data, params)
File "/Users/robert/Dev/Web/webrtc-build-scripts-master/ios/webrtc/src/chromium/src/tools/gyp/pylib/gyp/generator/ninja.py", line 2376, in GenerateOutput
pool.map(CallGenerateOutputForConfig, arglists)
File "/System/Library/Frameworks/Python.framework/Versions/2.7/lib/python2.7/multiprocessing/pool.py", line 250, in map
return self.map_async(func, iterable, chunksize).get()
File "/System/Library/Frameworks/Python.framework/Versions/2.7/lib/python2.7/multiprocessing/pool.py", line 554, in get
raise self._value
AssertionError: Multiple codesigning fingerprints for identity: iPhone Developer
Error: Command /usr/bin/python src/webrtc/build/gyp_webrtc -Dextra_gyp_flag=0 returned non-zero exit status 1 in /Users/robert/Dev/Web/webrtc-build-scripts-master/ios/webrtc
ninja: Entering directory `out_ios_armeabi_v7a/Debug-iphoneos/'
ninja: error: build.ninja:98: unexpected EOF
description = MERGE INFOPL
^ near here
error: /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool: can't open file: /Users/robert/Dev/Web/webrtc-build-scripts-master/ios/webrtc/src/out_ios_armeabi_v7a/Debug-iphoneos/*.a (No such file or directory)
Running depot tools as root is sad

What's the right way to switch the camera from the front-camera to the rear-camera while the call is going.

I tried this, it works a few times, then it crashes. Any idea why?

[self.localVideoTrack removeRenderer:(RTCEAGLVideoView*)[self.viewController localVideoTrackViewForCallController:self]];

        RTCMediaStream* lms = self.peerConnection.localStreams[1];
        [lms removeVideoTrack:self.localVideoTrack];
        [self.peerConnection removeStream:lms];

        NSString *newVideoTrackId = nil;
        NSString *oldVideoTrackId = nil;
        if (_rearCamera) {
            oldVideoTrackId = @"ARDAMSv1";
            newVideoTrackId = @"ARDAMSv0";
        }
        else {
            oldVideoTrackId = @"ARDAMSv0";
            newVideoTrackId = @"ARDAMSv1";
        }

        RTCVideoCapturer* capturer = [RTCVideoCapturer capturerWithDeviceName:newCameraID];
        self.videoSource = [_peerConnectionFactory videoSourceWithCapturer:capturer constraints:self.videoConstraints];
        self.localVideoTrack = [_peerConnectionFactory videoTrackWithID:newVideoTrackId source:self.videoSource];

        [lms addVideoTrack:self.localVideoTrack];
        [self.peerConnection addStream:lms];

        RTCSessionDescription* remoteDesc = self.peerConnection.remoteDescription;
        RTCSessionDescription *localDesc = [[RTCSessionDescription alloc] initWithType:self.peerConnection.localDescription.type sdp:[self.peerConnection.localDescription.description stringByReplacingOccurrencesOfString:oldVideoTrackId withString:newVideoTrackId]];

            [self.peerConnection setLocalDescriptionWithDelegate:self sessionDescription:localDesc];
            [self.peerConnection setRemoteDescriptionWithDelegate:self sessionDescription:remoteDesc];

Firefox 38 will require Perfect Forward Secrecy for DTLS. The current iOS library is built using the standard SSL library (NSS) which doesn't support PFS. The guys at &yet have verified that building with the OpenSSL fork BoringSSL (which Chromium has moved to, iirc) solves the problem with Firefox 38 interopability.

This should be a small matter of adding use_openssl=1 to the GYP_DEFINES in build.sh.

________ running '/usr/bin/python -u src/sync_chromium.py --target-revision b0c3ed39916e25bed2900b653974672a39fcb254' in '/Users/username/Documents/github-repo/blumr-ios64/webrtc-build-scripts/ios/webrtc'
Running "gclient sync --force --revision src@b0c3ed39916e25bed2900b653974672a39fcb254 --gclientfile .gclient.tmp --delete_unversioned_trees --reset --upstream --deps=ios,mac" in /Users/username/Documents/github-repo/blumr-ios64/webrtc-build-scripts/ios/webrtc/src/chromium
[0:01:00] Still working on:
[0:01:00] src
[0:01:10] Still working on:
[0:01:10] src

We use the libjingle_peerconnection pod. The AppRTC example app recently switched to using the third party library SocketRocket, which is therefore now included in the library created by this build script.

We already use the SocketRocket library in our app, which causes duplicate symbols in our project. The issue is described here.

I'm a bit confused as to why the example app libraries are included in the binary at all. Could they either be removed, or at least prefixed in order to avoid duplicate symbol errors?

I am running on a device (ios7)
When I connect, I manage to establish a connection

SDP onSuccess() - possibly drain candidates
GAE onMessage type - candidate
GAE onMessage type - candidate
GAE onMessage type - candidate

but I see no video but a black box and on the website https://apprtc.appspot.com/?r=def I see myself twice from the computer camera (the other peer)

Whoo hoo! After a bunch of days of downloading and cursing I finally got it to compile. I wanted to document my issues and my solutions to you.

1. I updated the vagrant vm to ubuntu/trusty64. I kept seeing things about deprecated support and to move to trusty. Don't know in the end if it was necessary but I don't feel it was harmful.

2. I had to change the ANDROID_TOOLCHAINS variable to "$WEBRTC_ROOT/src/chromium/src/third_party/android_tools/ndk/toolchains"

3. I had to change the location of envsetup.sh to $WEBRTC_ROOT/src/chromium/src/build/android/envsetup.sh

4. I added --force to gclient sync. Don't know if this made a difference.

5. I have my partition set to Mac OS Extended (Journaled). I needed to create a Mac OS Extended (Case-sensitive, Journaled) partition. I had a bunch of issues fetching webrtc. I would constantly get you have unstaged changes from gclient. It just couldn't complete. Whatever I tried, I continued to get unstaged changes. I couldn't do a git checkout . git diff | git apply --reverse git stash git reset --hard origin/master git clean -df git commit -am "remove" && git reset --hard head^ . Tried an interactive checkout. Tried checking out files one by one from master. Files still had differences. Moving the entire directory to a case sensitive partition fixed this. Debugging this one sucked the most.

6. I could not get hard links to work on the shared /vagrant folder. I would get this error:

   Downloading /vagrant/webrtc/src/chromium/src/third_party/binutils/Linux_x64/binutils.tar.bz2
   tar: ./bin/ld.gold: Cannot hard link to `./bin/ld': Operation not permitted
   

   I tried https://github.com/gael-ian/vagrant-bindfs , http://serverfault.com/questions/501599/vagrant-synced-folders-and-vboxinternal2-sharedfoldersenablesymlinkscreate , hashicorp/vagrant#713 (comment) , plus others I cannot remember. I eventually gave up on this one and moved it to a different folder in my vagrant box that was not the shared project directory. The issue went away. This means I changed the PROJECT_ROOT environment variable.

7. Then, there were a lot of unmet dependencies. I dunno I just kinda winged it one by one until eventually I just took this list: https://groups.google.com/a/chromium.org/forum/#!topic/chromium-dev/Rm1nX_LFYTo and hoped it worked. I removed all the ttf stuff.

8. I added

   create_directory_if_not_found "$ARCH_OUT"
   create_directory_if_not_found "$ARCH_OUT/$BUILD_TYPE"
   

   before exec_ninja: https://github.com/pristineio/webrtc-build-scripts/blob/master/android/build.sh#L222 . Don't know if it was necessary but I got issues about "Directory not found". This happened way before I solved these other issues so it might not be necessary.

vagrant version 1.7.2 (installed with homebrew-cask)

Sorry this isn't a pull request. I didn't want to submit all my code changes in a PR. I felt it was important to at least document the issues I had. That way if other people run into problems then at least they'll see they're not the only one.

Here's an adorable baby koala.

The readme states:

When the scripts are done you can find the .jar and .so file in
$WEBRTC_HOME under "libjingle_peerconnection_builds".

That variable does not exist. I also can't find a file named libjingle_peerconnection.jar anywhere. However, there are two files named libjingle_peerconnection_java.jar which are identical:

webrtc-build-scripts/android/webrtc/src/out_android_armeabi-v7a/Release/gen/libjingle_peerconnection_java
webrtc-build-scripts/android/webrtc/src/out_android_armeabi-v7a/Release/lib.java/libjingle_peerconnection_java.jar

1. First of all, thanks for the building scripts.
   After I've done setup all the environment (source build.sh, install dependencies, use jdk 1.8), I tried to build apprtc (arch=x64, debug=true). But I got errors after all:

   [28/2637] CC obj/chromium/src/third_party/boringssl/src/crypto/asn1/boringssl.a_d2i_fp.o
   FAILED: /home/sysu/webrtc-build-
   scripts/android/webrtc/src/third_party/android_tools/ndk//toolchains/x86_64-4.9/prebuilt/linux-x86_64/bin/x86_64-linux-android-gcc -MMD -MF obj/chromium/src/third_party/boringssl/src/crypto/asn1/boringssl.a_d2i_fp.o.d -DV8_DEPRECATION_WARNINGS -D_FILE_OFFSET_BITS=64 -DNO_TCMALLOC -DDISABLE_NACL -DCHROMIUM_BUILD -DCR_CLANG_REVISION=231690-1 -DUSE_LIBJPEG_TURBO=1 -DENABLE_WEBRTC=1 -DUSE_PROPRIETARY_CODECS -DENABLE_BROWSER_CDMS -DENABLE_CONFIGURATION_POLICY -DENABLE_NOTIFICATIONS -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DDONT_EMBED_BUILD_METADATA -DENABLE_AUTOFILL_DIALOG=1 -DCLD_VERSION=1 -DENABLE_PRINTING=1 -DENABLE_BASIC_PRINTING=1 -DENABLE_SUPERVISED_USERS=1 -DVIDEO_HOLE=1 -DV8_USE_EXTERNAL_STARTUP_DATA -DBORINGSSL_IMPLEMENTATION -DBORINGSSL_NO_STATIC_INITIALIZER -DUSE_LIBPCI=1 -DUSE_OPENSSL=1 -DUSE_OPENSSL_CERTS=1 -DANDROID -D__GNU_SOURCE=1 -DUSE_STLPORT=1 -D_STLP_USE_PTR_SPECIALIZATIONS=1 '-DCHROME_BUILD_ID=""' -DHAVE_SYS_UIO_H -DDYNAMIC_ANNOTATIONS_ENABLED=1 -DWTF_USE_DYNAMIC_ANNOTATIONS=1 -D_DEBUG -Igen -I../../chromium/src/third_party/boringssl/src/include -I../../chromium/src/third_party/boringssl/src/crypto -fstack-protector --param=ssp-buffer-size=4 -fno-strict-aliasing -Wno-unused-parameter -Wno-missing-field-initializers -fvisibility=hidden -pipe -fPIC -Wno-unused-local-typedefs -Wno-format -m64 -march=x86-64 -ffunction-sections -funwind-tables -g -fstack-protector -fno-short-enums -finline-limit=64 -Wa,--noexecstack --sysroot=../../third_party/android_tools/ndk//platforms/android-21/arch-x86_64 -isystem../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport -Os -g -gdwarf-4 -fdata-sections -ffunction-sections -fomit-frame-pointer -funwind-tables   -c ../../chromium/src/third_party/boringssl/src/crypto/asn1/a_d2i_fp.c -o obj/chromium/src/third_party/boringssl/src/crypto/asn1/boringssl.a_d2i_fp.o
   In file included from ../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport/limits.h:30:0,
                    from /usr/include/../include/limits.h:123,
                    *************** hundreds of same line **************
                    from ../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport/limits.h:30,
                    from /usr/include/limits.h:123,
                    from ../../chromium/src/third_party/boringssl/src/crypto/asn1/a_d2i_fp.c:59:
   /usr/include/../include/limits.h:123:26: error: #include nested too deeply
   # include_next <limits.h>
                            ^
   usr/include/../include/limits.h:143:30: error: #include nested too deeply
   # include <bits/posix1_lim.h>
                                ^
   /usr/include/../include/limits.h:147:30: error: #include nested too deeply
   # include <bits/posix2_lim.h>
                                ^
   In file included from /usr/include/bits/posix1_lim.h:160:0,
                    from /usr/include/../include/limits.h:143,
                    from ../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport/limits.h:30,
                    from /usr/include/../include/limits.h:123,
                    *************** hundreds of same line **************
                    from ../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport/limits.h:30,
                    from /usr/include/../include/limits.h:123,
                    from ../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport/limits.h:30,
                    from /usr/include/limits.h:123,
                    from ../../chromium/src/third_party/boringssl/src/crypto/asn1/a_d2i_fp.c:59:
   /usr/include/bits/local_lim.h:38:26: error: #include nested too deeply
   #include <linux/limits.h>
                            ^
   ../../chromium/src/third_party/boringssl/src/crypto/asn1/a_d2i_fp.c: In function 'asn1_d2i_read_bio':
   ../../chromium/src/third_party/boringssl/src/crypto/asn1/a_d2i_fp.c:235:16: error: 'INT_MAX' undeclared (first use in this function)
    if (want > INT_MAX /* BIO_read takes an int length */ ||
               ^
   ../../chromium/src/third_party/boringssl/src/crypto/asn1/a_d2i_fp.c:235:16: note: each undeclared identifier is reported only once for each function it appears in
   [28/2637] CC obj/chromium/src/third_party/libvpx/source/libvpx/vp9/encoder/x86/libvpx_intrinsics_avx2.vp9_dct_avx2.o
   ninja: build stopped: subcommand failed.
   Debug build for apprtc failed for revision 8726

2. Seems like the -Werror turns on, so I remove it, and tried again, then I got:

   Running ninja
   ninja: Entering directory `out_android_x86_64/Debug'
   [208/2656] CC obj/chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu_features.cpu-features.o
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c: In function 'extract_cpuinfo_field':
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c:195:21: warning: incompatible implicit declaration of built-in function 'strlen'
    int  fieldlen = strlen(field);
                    ^
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c:204:11: warning: assignment makes pointer from integer without a cast
        p = memmem(p, bufend-p, field, fieldlen);
          ^
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c:216:10: warning: incompatible implicit declaration of built-in function 'memchr'
    p  = memchr(p, ':', bufend-p);
         ^
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c:232:5: warning: incompatible implicit declaration of built-in function 'memcpy'
    memcpy(result, p, len);
    ^
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c: In function 'has_list_item':
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c:246:19: warning: incompatible implicit declaration of built-in function 'strlen'
    int itemlen = strlen(item);
                  ^
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c: In function 'cpulist_parse':
   ../../chromium/src/third_party/android_tools/ndk/sources/android/cpufeatures/cpu-features.c:378:13: warning: incompatible implicit declaration of built-in function 'memchr'
        q = memchr(p, ',', end-p);
            ^
   [383/2656] CC obj/chromium/src/third_party/libevent/libevent.select.o
   FAILED: /home/sysu/webrtc-build-scripts/android/webrtc/src/third_party/android_tools/ndk//toolchains/x86_64-4.9/prebuilt/linux-x86_64/bin/x86_64-linux-android-gcc -D__u32=int -MMD -MF obj/chromium/src/third_party/libevent/libevent.select.o.d -DV8_DEPRECATION_WARNINGS -D_FILE_OFFSET_BITS=64 -DNO_TCMALLOC -DDISABLE_NACL -DCHROMIUM_BUILD -DCR_CLANG_REVISION=231690-1 -DUSE_LIBJPEG_TURBO=1 -DENABLE_WEBRTC=1 -DUSE_PROPRIETARY_CODECS -DENABLE_BROWSER_CDMS -DENABLE_CONFIGURATION_POLICY -DENABLE_NOTIFICATIONS -DDISCARDABLE_MEMORY_ALWAYS_SUPPORTED_NATIVELY -DSYSTEM_NATIVELY_SIGNALS_MEMORY_PRESSURE -DDONT_EMBED_BUILD_METADATA -DENABLE_AUTOFILL_DIALOG=1 -DCLD_VERSION=1 -DENABLE_PRINTING=1 -DENABLE_BASIC_PRINTING=1 -DENABLE_SUPERVISED_USERS=1 -DVIDEO_HOLE=1 -DV8_USE_EXTERNAL_STARTUP_DATA -DHAVE_CONFIG_H -DUSE_LIBPCI=1 -DUSE_OPENSSL=1 -DUSE_OPENSSL_CERTS=1 -DANDROID -D__GNU_SOURCE=1 -DUSE_STLPORT=1 -D_STLP_USE_PTR_SPECIALIZATIONS=1 '-DCHROME_BUILD_ID=""' -DHAVE_SYS_UIO_H -DDYNAMIC_ANNOTATIONS_ENABLED=1 -DWTF_USE_DYNAMIC_ >ANNOTATIONS=1 -D_DEBUG -Igen -I../../chromium/src/third_party/libevent/android -fstack-protector --param=ssp-buffer-size=4 -fno-strict-aliasing -Wno-unused-parameter -Wno-missing-field-initializers -fvisibility=hidden -pipe -fPIC -Wno-unused-local-typedefs -Wno-format -m64 -march=x86-64 -ffunction-sections -funwind-tables -g -fstack-protector -fno-short-enums -finline-limit=64 -Wa,--noexecstack --sysroot=../../third_party/android_tools/ndk//platforms/android-21/arch-x86_64 -I../../third_party/android_tools/ndk//sources/cxx-stl/stlport/stlport -Os -g -gdwarf-4 -fdata-sections -ffunction-sections -fomit-frame-pointer -funwind-tables   -c ../../chromium/src/third_party/libevent/select.c -o obj/chromium/src/third_party/libevent/libevent.select.o
   ../../chromium/src/third_party/libevent/select.c:67:23: error: conflicting types for 'fd_mask'
   typedef unsigned long fd_mask;
                         ^
   In file included from /usr/include/sys/types.h:219:0,
                    from ../../chromium/src/third_party/libevent/select.c:33:
   /usr/include/sys/select.h:82:19: note: previous declaration of 'fd_mask' was here
   typedef __fd_mask fd_mask;
                     ^
   [383/2656] CC obj/chromium/src/third_party/libvpx/source/libvpx/vp9/encoder/x86/libvpx_intrinsics_sse2.vp9_dct_sse2.o
   ninja: build stopped: subcommand failed.

Can anyone tells me what's the problem of my build?
P.S.
My compiler's gcc 4.8, OS is Ubuntu 14.10

Thanks.

Hi! I'm using mac for building android library, btw I had a successful building for iOS version with r8509.
Here's what I did the following:

• go to my previous folder /android
• run vagrant up and ssh
• git clone https://github.com/pristineio/webrtc-build-scripts.git
• go to webrtc-build-scripts
• run source android/build.sh
• run get_webrtc 8509
• run build_apprtc

But, I'm stucked with this line:
[5:34:49] Still working on:
[5:34:49] src/third_party/android_tools

Hi, I was trying to run the AppRTCDemo on Mac and got the error:

ld: warning: directory not found for option '-L/Users/rbehera/webrtc/libjingle_peerconnection_builds/Debug-iphoneos'
ld: warning: directory not found for option '-L/repo/tomfisher/google/webrtc-build-scripts/ios/webrtc/libjingle_peerconnection_builds/Profile-iphoneos'
Undefined symbols for architecture i386:
  "_OBJC_CLASS_$_ARDSignalingParams", referenced from:
      objc-class-ref in APPRTCAppClient.o
ld: symbol(s) not found for architecture i386
clang: error: linker command failed with exit code 1 (use -v to see invocation)

using either r7810 or r7827, as well as from command line or xcode.

I'm on OSX and using the vagrant vm started form the android directory.

When I run webrtc_get it finishes with this

Updated to revision 7413.
Syncing projects: 100% (3/3), done.                      

________ running '/usr/bin/python -c import os, sys;script = os.path.join('trunk', 'check_root_dir.py');_ = os.system('%s %s' % (sys.executable, script)) if os.path.exists(script) else 0' in '/home/vagrant/webrtc'
@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@
                      A C T I O N     R E Q I R E D
@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@@

It looks like you have a legacy checkout where the solution's top-level
directory is named 'trunk'. From now on, it must be named 'src'.

What you need to do is to:

1. Edit your .gclient file and change the solution name from 'trunk' to 'src'
2. Rename your 'trunk' directory to 'src'
3. Re-run gclient sync (or gclient runhooks)

________ running '/usr/bin/python -u src/sync_chromium.py --target-revision 2d714fae183152299b3cbf0056eab5fe8bb75e87' in '/home/vagrant/webrtc'
/usr/bin/python: can't open file 'src/sync_chromium.py': [Errno 2] No such file or directory
Error: Command /usr/bin/python -u src/sync_chromium.py --target-revision 2d714fae183152299b3cbf0056eab5fe8bb75e87 returned non-zero exit status 2 in /home/vagrant/webrtc

What am I supposed to do here? There's already a directory named 'src' at the same level as 'trunk'.

I just did a pull today and had problems building. I think it's because we probably need to be getting DL_ARM_NEON_OPTIONAL defined instead of DL_ARM_NEON. Here is the writeup I put on the discuss-webrtc (figuring that you probably don't set the DL_ARM_NEON). https://groups.google.com/forum/#!topic/discuss-webrtc/UtA_12H3fMI

My guess is that it's not your code setting the DL_ARM_NEON, but I thought I'd bring it to your attention just in case.

Thank you for providing these scripts. Trying to compile the webrtc library with Google's out of date instructions is a horrific experience :-).

This is for revision 6958.

build_apprtc eventually fails to setup the build environment with the following:

Setting up build environment for Android
-bash: <~>/webrtc/trunk/build/install-build-deps-android.sh: No such file or directory
-bash: <~>/webrtc/trunk/build/android/envsetup.sh: No such file or directory

Then it becomes stuck:

[0:01:00] Still working on:
[0:01:00]   src
[0:01:10] Still working on:
[0:01:10]   src
[0:01:20] Still working on:
[0:01:20]   src
....

Android arm64 build is commented out and there is a note that the application can not yet build for arm64 successfully. Is this fix near in the roadmap?

https://github.com/pristineio/webrtc-build-scripts/blob/master/android/build.sh#L293

    # Uncomment once the application can successfully build for arm64
    #export WEBRTC_ARCH=armv8
    #prepare_gyp_defines &&
    #execute_build

I'm trying to compile following the steps on documentation and also on http://tech.pristine.io/build-android-apprtc/ but when I execute "build_apprtc" it keeps there and showing still working each some time..

mike@ultramike:~/workspace/webrtc/webrtc-build-scripts$ build_apprtc
Setting up build environment for Android
bash: /home/mike/workspace/webrtc/webrtc-build-scripts/android/webrtc/src/build/android/envsetup.sh: No existe el archivo o el directorio
Export the base settings of GYP_DEFINES so we can define how we want to build
User has not specified any gyp defines so we proceed with default
ARMv7 with Neon Build
GYP_DEFINES=OS=android host_os=linux libjingle_java=1 build_with_libjingle=1 build_with_chromium=0 enable_tracing=1 enable_android_opensl=1 OS=android
Run gclient hooks

________ running '/usr/bin/python -c import os,sys;script = os.path.join("trunk","check_root_dir.py");_ = os.system("%s %s" % (sys.executable,script)) if os.path.exists(script) else 0' in '/home/mike/workspace/webrtc/webrtc-build-scripts/android/webrtc'

________ running '/usr/bin/python -u src/sync_chromium.py --target-revision a6eafec7a5adaa933252b6601822d0ab312d660a' in '/home/mike/workspace/webrtc/webrtc-build-scripts/android/webrtc'
Running "gclient sync --force --revision src@a6eafec7a5adaa933252b6601822d0ab312d660a --gclientfile .gclient.tmp --delete_unversioned_trees --reset --upstream --deps=unix,android" in /home/mike/workspace/webrtc/webrtc-build-scripts/android/webrtc/src/chromium
[0:01:00] Still working on:
[0:01:00]   src
[0:01:10] Still working on:
[0:01:10]   src
[0:01:20] Still working on:
[0:01:20]   src
[0:01:30] Still working on:

And keeps there..

[0:10:50] Still working on:
[0:10:50]   src
[0:11:00] Still working on:
[0:11:00]   src
[0:11:10] Still working on:
[0:11:10]   src
[0:11:20] Still working on:
[0:11:20]   src
[0:11:30] Still working on:
[0:11:30]   src
[0:11:40] Still working on:
[0:11:40]   src
[0:11:50] Still working on:
[0:11:50]   src
...

Log

WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:490): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:490): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:490): Unexpected codec: G722/16000/2 (119)
opus/48000/2 (111)
CN/8000/1 (13)
CN/16000/1 (105)
CN/32000/1 (106)
telephone-event/8000/1 (126)
red/8000/1 (127)
WebRtcVideoEngine::WebRtcVideoEngine
webrtc: (vie_impl.cc:133): SetTraceFilter: filter: 8206
webrtc: (vie_impl.cc:138): SetTraceCallback:
WebRtcVoiceEngine::Init
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
Build: Aug 13 2014 10:37:23 ?
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, opus_fec: false, }
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 0
Error(webrtcvideoengine.cc:1469): webrtc: (voe_audio_processing_impl.cc:1001): SetTypingDetectionStatus: not supported
Warning(webrtcvoiceengine.cc:867): SetTypingDetectionStatus(0) failed, err=8003
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
Opus FEC is enabled? 0
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
WebRtcVoiceEngine::Init Done!
WebRtcVideoEngine::Init
WebRtcVideoEngine::InitVideoEngine
WebRtc VideoEngine Version:
VideoEngine 3.55.0
Build: Aug 13 2014 10:37:27 ?
webrtc: (vie_base_impl.cc:68): SetVoiceEngine: SetVoiceEngine
VideoEngine Init done
webrtc: (vie_render_impl.cc:64): RegisterVideoRenderModule:
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, opus_fec: false, }
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 0
Warning(webrtcvoiceengine.cc:867): SetTypingDetectionStatus(0) failed, err=8003
Error(webrtcvideoengine.cc:1469): webrtc: (voe_audio_processing_impl.cc:1001): SetTypingDetectionStatus: not supported
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
Opus FEC is enabled? 0
2014-08-13 10:55:21.853 AppRTCDemo[511:60b] ICE servers:
(
"RTCICEServer: [stun:stun.l.google.com:19302::]",
"RTCICEServer: [turn:192.158.30.23:3478?transport=udp:1407984921:47912914:0ggPO+/q3+oxXoFXTXdOKQHaI8k=]"
)
Allowing SCTP data engine.
Generating identity.
Created VideoCapturer for Front Camera
Failed to find best capture format, fall back to the requested format I420 640x480x30
VAdapt input interval changed from 0 to 33333333
Camera 'com.apple.avfoundation.avcapturedevice.built-in_video:1' started with format I420 640x480x30, elapsed time 0 ms
2014-08-13 10:55:26.155 AppRTCDemo[511:3107] WARNING: -[<AVCaptureConnection: 0x18182dc0> setVideoMinFrameDuration:] is deprecated. Please use AVCaptureDevice setActiveVideoMinFrameDuration
2014-08-13 10:55:26.157 AppRTCDemo[511:3107] WARNING: -[<AVCaptureConnection: 0x18182dc0> setVideoMaxFrameDuration:] is deprecated. Please use AVCaptureDevice setActiveVideoMaxFrameDuration
2014-08-13 10:55:26.166 AppRTCDemo[511:60b] PCO onRenegotiationNeeded - ignoring because AppRTC has a predefined negotiation strategy
Captured frame size 480x640. Expected format I420 640x480x30
VAdapt Input Resolution Change: Previous input resolution: 640x480 New input resolution: 480x640 New output resolution: 480x640
VAdapt Frame: scaled 0 / out 90 / in 90 Changes: 0 Input: 480x640 i33333333 Scale: 1 Output: 480x640 i33333333 Changed: false
VAdapt Frame: scaled 0 / out 180 / in 180 Changes: 0 Input: 480x640 i33333333 Scale: 1 Output: 480x640 i33333333 Changed: false
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Created channel for audio
Setting voice channel options: AudioOptions {}
Set voice channel options. Current options: AudioOptions {}
webrtc: (remote_bitrate_estimator_single_stream.cc:258): RemoteBitrateEstimatorFactory: Instantiating.
webrtc: (vie_base_impl.cc:167): Video channel created: 0
webrtc: (vie_network_impl.cc:74): RegisterSendTransport: channel: 0
webrtc: (vie_network_impl.cc:137): SetMTU: channel: 0 mtu: 1200
webrtc: (vie_rtp_rtcp_impl.cc:285): SetRTCPStatus: channel: 0 mode: 1
webrtc: (vie_rtp_rtcp_impl.cc:521): SetKeyFrameRequestMethod: channel: 0 method: 1
webrtc: (vie_rtp_rtcp_impl.cc:388): SetNACKStatus: channel: 0 on
Warning(webrtcvideoengine.cc:1469): webrtc: (rtp_packet_history.cc:48): Purging packet history in order to re-set status.
NACK enabled for channel 0
webrtc: (vie_base_impl.cc:211): ConnectAudioChannel: ConnectAudioChannel, video channel 0, audio channel 0
webrtc: (vie_rtp_rtcp_impl.cc:558): SetRembStatus: channel: 0 sender: off receiver: off
webrtc: (vie_rtp_rtcp_impl.cc:590): SetReceiveTimestampOffsetStatus: channel: 0enable: off id: 0
webrtc: (vie_rtp_rtcp_impl.cc:627): SetReceiveAbsoluteSendTimeStatus: channel: 0enable: off id: 0
webrtc: (vie_image_process_impl.cc:187): EnableColorEnhancement: video_channel: 0 enable: off
webrtc: (vie_codec_impl.cc:502): RegisterDecoderObserver for channel 0
webrtc: (vie_capture_impl.cc:106): External capture device allocated: 4097
webrtc: (vie_capture_impl.cc:139): Connect capture id 4097 to channel 0
webrtc: (vie_codec_impl.cc:469): RegisterEncoderObserver for channel 0
webrtc: (vie_rtp_rtcp_impl.cc:571): SetSendTimestampOffsetStatus: channel: 0enable: off id: 0
webrtc: (vie_rtp_rtcp_impl.cc:608): SetSendAbsoluteSendTimeStatus: channel: 0enable: off id: 0
webrtc: (vie_rtp_rtcp_impl.cc:658): SetTransmissionSmoothingStatus: channel: 0 enable: on
webrtc: (vie_rtp_rtcp_impl.cc:558): SetRembStatus: channel: 0 sender: off receiver: off
webrtc: (vie_rtp_rtcp_impl.cc:388): SetNACKStatus: channel: 0 on
Warning(webrtcvideoengine.cc:1469): webrtc: (rtp_packet_history.cc:48): Purging packet history in order to re-set status.
NACK enabled for channel 0
webrtc: (vie_base_impl.cc:301): StartReceive: StartReceive 0
Created channel for video
Improved WIFI BWE called.
webrtc: (vie_network_impl.cc:166): SetBandwidthEstimationConfig: channel: 0
webrtc: (remote_bitrate_estimator_single_stream.cc:258): RemoteBitrateEstimatorFactory: Instantiating.
Session:1009845214941252125 Old state:STATE_INIT New state:STATE_SENTINITIATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Setting local voice description
Add send ssrc: 1389120337
Setting receive voice codecs:
ISAC/16000/1 (103)
opus/48000/2 (111)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
Changing voice state, recv=0 send=0
Setting local video description
webrtc: (vie_rtp_rtcp_impl.cc:590): SetReceiveTimestampOffsetStatus: channel: 0enable: on id: 2
webrtc: (vie_rtp_rtcp_impl.cc:627): SetReceiveAbsoluteSendTimeStatus: channel: 0enable: on id: 3
AddSendStream {id:ARDAMSv0;ssrcs:[1971932771,4176019807];ssrc_groups:{semantics:FID;ssrcs:[1971932771,4176019807]};cname:EE8iRHRBSF3zd0xv;sync_label:ARDAMS}
webrtc: (vie_rtp_rtcp_impl.cc:114): SetLocalSSRC: channel: 0 ssrc: 1971932771
webrtc: (vie_rtp_rtcp_impl.cc:114): SetLocalSSRC: channel: 0 ssrc: 4176019807
webrtc: (vie_rtp_rtcp_impl.cc:321): SetRTCPCName: channel: 0 rtcp_cname: EE8iRHRBSF3zd0xv
webrtc: (vie_rtp_rtcp_impl.cc:114): SetLocalSSRC: channel: 0 ssrc: 1971932771
Add send ssrc: 1971932771
webrtc: (vie_codec_impl.cc:278): SetReceiveCodec for channel 0
webrtc: (vie_codec_impl.cc:279): Codec type 0, payload type d
webrtc: (vie_codec_impl.cc:278): SetReceiveCodec for channel 0
webrtc: (vie_codec_impl.cc:279): Codec type 3, payload type t
webrtc: (vie_codec_impl.cc:278): SetReceiveCodec for channel 0
webrtc: (vie_codec_impl.cc:279): Codec type 4, payload type u
webrtc: (vie_rtp_rtcp_impl.cc:227): SetRtxReceivePayloadType: channel: 0 payload_type: 96
Buffer latency is 0
webrtc: (vie_rtp_rtcp_impl.cc:474): SetSenderBufferingMode: channel: 0 target_delay_ms: 0
Warning(webrtcvideoengine.cc:1469): webrtc: (rtp_packet_history.cc:48): Purging packet history in order to re-set status.
webrtc: (vie_rtp_rtcp_impl.cc:501): SetReceiverBufferingMode: channel: 0 target_delay_ms: 0
Changing video state, recv=0 send=0
Setting voice channel options: AudioOptions {}
Set voice channel options. Current options: AudioOptions {}
webrtc: (vie_base_impl.cc:78): RegisterCpuOveruseObserver: RegisterCpuOveruseObserver on channel 0
webrtc: (vie_base_impl.cc:78): RegisterCpuOveruseObserver: RegisterCpuOveruseObserver on channel 0
WebRtcOveruseObserver enable: 0
Local and Remote descriptions must be applied to get SSL Role of the session.
Transport: audio, allocating candidates
Transport: audio, allocating candidates
Transport: video, allocating candidates
Transport: video, allocating candidates
2014-08-13 10:55:35.927 AppRTCDemo[511:60b] PCO onSignalingStateChange: 1
2014-08-13 10:55:35.930 AppRTCDemo[511:60b] PCO onIceGatheringChange. 1
2014-08-13 10:55:35.932 AppRTCDemo[511:60b] PCO onIceGatheringChange. 1
2014-08-13 10:55:35.954 AppRTCDemo[511:60b] PCO onIceGatheringChange. 1
2014-08-13 10:55:35.956 AppRTCDemo[511:60b] PCO onIceGatheringChange. 1
Jingle:Net[ppp0:10.9.0.0/24:Unknown]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[ppp0:10.9.0.0/24:Unknown]]: Port created
AllocationSequence: UDPPort will be handling the STUN candidate generation.
Adding allocated port for audio
Jingle:Port[audio:1:0::Net[ppp0:10.9.0.0/24:Unknown]]: Added port to allocator
Jingle:Net[en0:10.1.30.0/24:Unknown]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[en0:10.1.30.0/24:Unknown]]: Port created
2014-08-13 10:55:36.020 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[audio] Index[0] Sdp[candidate:2364124319 1 udp 2122129151 10.9.0.63 54292 typ host generation 0]
2014-08-13 10:55:36.023 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[audio] Index[0] Sdp[candidate:2364124319 2 udp 2122129151 10.9.0.63 54292 typ host generation 0]
2014-08-13 10:55:36.024 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[video] Index[1] Sdp[candidate:2364124319 1 udp 2122129151 10.9.0.63 54292 typ host generation 0]
2014-08-13 10:55:36.026 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[video] Index[1] Sdp[candidate:2364124319 2 udp 2122129151 10.9.0.63 54292 typ host generation 0]
AllocationSequence: UDPPort will be handling the STUN candidate generation.
Adding allocated port for audio
Jingle:Port[audio:1:0::Net[en0:10.1.30.0/24:Unknown]]: Added port to allocator
2014-08-13 10:55:36.033 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[audio] Index[0] Sdp[candidate:2595736997 1 udp 2122063615 10.1.30.194 61272 typ host generation 0]
2014-08-13 10:55:36.035 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[audio] Index[0] Sdp[candidate:2595736997 2 udp 2122063615 10.1.30.194 61272 typ host generation 0]
2014-08-13 10:55:36.037 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[video] Index[1] Sdp[candidate:2595736997 1 udp 2122063615 10.1.30.194 61272 typ host generation 0]
2014-08-13 10:55:36.038 AppRTCDemo[511:60b] PCO onICECandidate.
Mid[video] Index[1] Sdp[candidate:2595736997 2 udp 2122063615 10.1.30.194 61272 typ host generation 0]
Jingle:Net[ppp0:10.9.0.0/24:Unknown]: Allocation Phase=Relay
Jingle:Port[:1:0:relay:Net[ppp0:10.9.0.0/24:Unknown]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:relay:Net[ppp0:10.9.0.0/24:Unknown]]: Added port to allocator
Jingle:Port[audio:1:0:relay:Net[ppp0:10.9.0.0/24:Unknown]]: Trying to connect to TURN server via udp @ 192.158.30.23:3478
Jingle:Net[en0:10.1.30.0/24:Unknown]: Allocation Phase=Relay
Jingle:Port[:1:0:relay:Net[en0:10.1.30.0/24:Unknown]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:relay:Net[en0:10.1.30.0/24:Unknown]]: Added port to allocator
Jingle:Port[audio:1:0:relay:Net[en0:10.1.30.0/24:Unknown]]: Trying to connect to TURN server via udp @ 192.158.30.23:3478
Jingle:Net[ppp0:10.9.0.0/24:Unknown]: Allocation Phase=Tcp
Jingle:Port[:1:0:local:Net[ppp0:10.9.0.0/24:Unknown]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:local:Net[ppp0:10.9.0.0/24:Unknown]]: Added port to allocator
Error(common.cc:59): ../../talk/app/webrtc/webrtcsdp.cc(1726): ASSERT FAILED: it->address().port() == 0 || it->address().port() == cricket::DISCARD_PORT @ BuildCandidate
Error(common.cc:59): ../../talk/app/webrtc/webrtcsdp.cc(1727): ASSERT FAILED: it->tcptype() == cricket::TCPTYPE_ACTIVE_STR @ BuildCandidate
(lldb)

debug with touch5 :)

Hi,

I was able to follow the instructions from README.md and then I generated some so and jar files. But I've seen this - http://andrii.sergiienko.me/?go=all/building-webrtc-demo-for-android/ - and I think it shows us how to build an AppRTCDemo that works with http://apprtc.appspot.com/ I think.

So how can I build the same demo ? In that article it uses some ninja command. Any help would be useful!

Regards

Hi,

Using latest webrtc-build-scripts.
Success link using debug build library.
Failed link using release build library.
Please help to build app with using release library.

[Environment]
Xcode : 6.1
libjingle : 8355
Rename libWebRTC-xxxx-arm-intel-Release.a to libWebRTC-Universal-Release.a

[Library info]
$ lipo libWebRTC-Universal-Release.a -info
Architectures in the fat file: libWebRTC-Universal-Release.a are: armv7 i386 x86_64 arm64

[Build log]
/Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool: for architecture x86_64 object: /Users/______/libWebRTC-Universal-Release.a(x86_abi_support.o) malformed object (string table at offset 0 with a size of 8, overlaps Mach-O headers at offset 0 with a size of 144)
Command /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/libtool failed with exit code 1

Like Cocoapods, an Android Maven artifact would be extremely convenient since most of users just need the built jars.

Pardon me if it is too Naive. See the steps. steps does not work.

On Mac or Windows

If you don't have a Ubuntu machine available, or you are too lazy to setup a virtual machine manually, you can build WebRTC for Android on your Mac or Windows PC through our Vagrant script.

First of all, you need to download and install Vagrant. After that, from the /android directory, you need to execute the following in you shell:
1 [installed]
2 [cloned this repos]
3 [changed directory to android]

If you need to use private SSH keys from your host computer

Execute this line of code to ensure your private key is added to your identity

ssh-add -L
4[ This command does not work]

If there are no identities, add them by:

ssh-add ~/.ssh/id_rsa

Boot up and provision the Vagrant box

vagrant up

SSH into the Vagrant box

vagrant ssh

I dont see any webrtc logs on console when using the latest version, Have you turned of the logs ? I am using libjingle_peerconnection v 7298.2.0 from cocoapods. Version 7243 has logs.

Hi,

Recently I've used this build script for creating iOS webRTC framework and then I decided to use your library through Cocoapods. Problem is that app crashes each time after I enable RTCVideoTrack (maybe a second later):

[videoTrack setEnabled:!mute]; //mute is some BOOL

I have this call stack, hope it can help.

Hello gentlemen,

I'm trying to do another build with latest webrtc code but when I go to libjingle_peerconnection_build directory I can see the very same jar file that I had created at 5 of feb.

I've tried also removing this directory, build scripts recreated it and add into it the same jar with modification date from 5 of feb.

There's any way to clean all and create everything ?

I'm doing..

• get_webrtc
• build_apprtc

Regards.

Latest available build 8005.2.0 is missing RTC* headers. Decided to rollback to the build 7972.2.0 which has headers.

I'm running the build scripts on Mac Yosemite (with vagrant) and this is how the get_webrtc command crashed after 2hours (with errors), anyone know what can be done to fix this:

[2:00:21] Still working on:
[2:00:21]   src/third_party/android_tools
[2:00:22] Still working on:
[2:00:22]   src/third_party/android_tools

________ running '/usr/bin/python src/build/landmines.py' in '/vagrant/webrtc/src/chromium'

________ running '/usr/bin/python src/build/download_nacl_toolchains.py --mode nacl_core_sdk sync' in '/vagrant/webrtc/src/chromium'

________ running '/usr/bin/python src/build/download_sdk_extras.py' in '/vagrant/webrtc/src/chromium'

________ running '/usr/bin/python src/chrome/installer/linux/sysroot_scripts/install-debian.wheezy.sysroot.py --linux-only' in '/vagrant/webrtc/src/chromium'

________ running '/usr/bin/python src/build/vs_toolchain.py update' in '/vagrant/webrtc/src/chromium'

________ running '/usr/bin/python src/tools/clang/scripts/update.py --if-needed' in '/vagrant/webrtc/src/chromium'
  % Total    % Received % Xferd  Average Speed   Time    Time     Time  Current
                                 Dload  Upload   Total   Spent    Left  Speed
100 23.9M  100 23.9M    0     0  1450k      0  0:00:16  0:00:16 --:--:-- 3112k
Trying to download prebuilt clang
clang 223108 unpacked
Hook '/usr/bin/python src/tools/clang/scripts/update.py --if-needed' took 18.60 secs

________ running '/usr/bin/python src/build/util/lastchange.py -o src/build/util/LASTCHANGE' in '/vagrant/webrtc/src/chromium'

________ running '/usr/bin/python src/build/util/lastchange.py -s src/third_party/WebKit -o src/build/util/LASTCHANGE.blink' in '/vagrant/webrtc/src/chromium'

________ running 'download_from_google_storage --no_resume --platform=win32 --no_auth --bucket chromium-gn -s src/buildtools/win/gn.exe.sha1' in '/vagrant/webrtc/src/chromium'

________ running 'download_from_google_storage --no_resume --platform=darwin --no_auth --bucket chromium-gn -s src/buildtools/mac/gn.sha1' in '/vagrant/webrtc/src/chromium'

________ running 'download_from_google_storage --no_resume --platform=linux* --no_auth --bucket chromium-gn -s src/buildtools/linux32/gn.sha1' in '/vagrant/webrtc/src/chromium'
0> Downloading src/buildtools/linux32/gn...
Hook 'download_from_google_storage --no_resume '--platform=linux*' --no_auth --bucket chromium-gn -s src/buildtools/linux32/gn.sha1' took 55.64 secs

________ running 'download_from_google_storage --no_resume --platform=linux* --no_auth --bucket chromium-gn -s src/buildtools/linux64/gn.sha1' in '/vagrant/webrtc/src/chromium'
0> Downloading src/buildtools/linux64/gn...
Hook 'download_from_google_storage --no_resume '--platform=linux*' --no_auth --bucket chromium-gn -s src/buildtools/linux64/gn.sha1' took 45.03 secs

________ running 'download_from_google_storage --no_resume --platform=win32 --no_auth --bucket chromium-clang-format -s src/buildtools/win/clang-format.exe.sha1' in '/vagrant/webrtc/src/chromium'

________ running 'download_from_google_storage --no_resume --platform=darwin --no_auth --bucket chromium-clang-format -s src/buildtools/mac/clang-format.sha1' in '/vagrant/webrtc/src/chromium'

________ running 'download_from_google_storage --no_resume --platform=linux* --no_auth --bucket chromium-clang-format -s src/buildtools/linux64/clang-format.sha1' in '/vagrant/webrtc/src/chromium'
0> Downloading src/buildtools/linux64/clang-format...
Hook 'download_from_google_storage --no_resume '--platform=linux*' --no_auth --bucket chromium-clang-format -s src/buildtools/linux64/clang-format.sha1' took 47.21 secs

________ running '/usr/bin/python src/third_party/binutils/download.py' in '/vagrant/webrtc/src/chromium'
Error: Command /usr/bin/python src/third_party/binutils/download.py returned non-zero exit status 1 in /vagrant/webrtc/src/chromium
0> Downloading /vagrant/webrtc/src/chromium/src/third_party/binutils/Linux_x64/binutils.tar.bz2...
tar: ./bin/ld.gold: Cannot hard link to `./bin/ld': Operation not permitted
tar: Exiting with failure status due to previous errors
Downloading /vagrant/webrtc/src/chromium/src/third_party/binutils/Linux_x64/binutils.tar.bz2
Extracting /vagrant/webrtc/src/chromium/src/third_party/binutils/Linux_x64/binutils.tar.bz2
Traceback (most recent call last):
  File "src/third_party/binutils/download.py", line 118, in <module>
    sys.exit(main(sys.argv))
  File "src/third_party/binutils/download.py", line 106, in main
    return FetchAndExtract(arch)
  File "src/third_party/binutils/download.py", line 91, in FetchAndExtract
    subprocess.check_call(['tar', 'axf', tarball], cwd=outdir)
  File "/usr/lib/python2.7/subprocess.py", line 511, in check_call
    raise CalledProcessError(retcode, cmd)
subprocess.CalledProcessError: Command '['tar', 'axf', '/vagrant/webrtc/src/chromium/src/third_party/binutils/Linux_x64/binutils.tar.bz2']' returned non-zero exit status 2
Hook '/usr/bin/python src/third_party/binutils/download.py' took 53.78 secs
Error: Command /usr/bin/python -u src/sync_chromium.py --target-revision 4664fe0d123f948bafa7b942717fc5847e61971c returned non-zero exit status 2 in /vagrant/webrtc
Hook '/usr/bin/python -u src/sync_chromium.py --target-revision 4664fe0d123f948bafa7b942717fc5847e61971c' took 7448.52 secs

Hi,

I'm trying to build webrtc for android using these commands:
get_webrtc
build_apprtc
export WEBRTC_ARCH=armv7 #or armv8, x86, or x86_64
prepare_gyp_defines && execute_build

** The zip file that is generated is less than 918 bytes and has a few empty folders in it.

I'm getting this message which includes an error:

In file included from ../../webrtc/modules/audio_device/audio_device_impl.cc:28:0:
../../webrtc/modules/audio_device/android/audio_device_template.h: In instantiation of 'webrtc::AudioDeviceTemplate<InputType, OutputType>::AudioDeviceTemplate(int32_t) [with InputType = webrtc::OpenSlesInput; OutputType = webrtc::OpenSlesOutput; int32_t = int]':
../../webrtc/modules/audio_device/audio_device_impl.cc:280:85: required from here
../../webrtc/modules/audio_device/android/audio_device_template.h:40:16: error: no matching function for call to 'webrtc::OpenSlesInput::OpenSlesInput()'
input_() {
^
../../webrtc/modules/audio_device/android/audio_device_template.h:40:16: note: candidate is:
In file included from ../../webrtc/modules/audio_device/audio_device_impl.cc:31:0:
../../webrtc/modules/audio_device/android/opensles_input.h:38:3: note: webrtc::OpenSlesInput::OpenSlesInput(int32_t, webrtc::PlayoutDelayProvider_)
OpenSlesInput(const int32_t id, PlayoutDelayProvider_ delay_provider);
^
../../webrtc/modules/audio_device/android/opensles_input.h:38:3: note: candidate expects 2 arguments, 0 provided
[8/157] CXX obj/webrtc/modules/audio_device/android/audio_device.opensles_input.o
ninja: build stopped: subcommand failed.
Copy JAR File
cp: cannot stat ‘/Repos/webrtc-build-scripts/android/webrtc/src/out_android_armeabi_v7a/Release/libjingle_peerconnection.jar’: No such file or directory
/Repos/webrtc-build-scripts/android/webrtc/src/third_party/android_tools/ndk/toolchains/arm-linux-androideabi-4.9/prebuilt/linux-x86_64/bin/arm-linux-androideabi-strip: '/Repos/webrtc-build-scripts/android/webrtc/src/out_android_armeabi_v7a/Release/libjingle_peerconnection_so.so': No such file
adding: libs/ (stored 0%)
adding: res/ (stored 0%)
adding: jniLibs/ (stored 0%)
adding: jniLibs/x86/ (stored 0%)
adding: jniLibs/armeabi_v7a/ (stored 0%)
adding: jniLibs/x86_64/ (stored 0%)
Release build for apprtc complete for revision 8353

It'd be really useful to have these if possible, but I'm not sure if libvpx and libyuv support these architectures yet? Any ideas?

At the time of writing it's 7348

When running build_webrtc or build_webrtc_mac, I am getting an awk error obtaining a signing identity. This looks to be most likely whats making my builds fail.

awk: illegal primary in regular expression ) |" at  |"
 input record number 1, file
 source line number 1
Using code signing identity

Been trying to play around with the syntax to get it to work but haven't been getting anywhere. I figured it would be a simple escaping issue but escaping those characters didn't work. Of note: I am using zsh but was able to reproduce on bash as well.

awk: illegal primary in regular expression ) |" at |"
input record number 1, file
source line number 1
Using code signing identity
sed: /Users/huangzan/Downloads/webrtc-build-scripts/webrtc-build-scripts/ios/webrtc/src/build/common.gypi: No such file or directory
.gclient file in parent directory /Users/huangzan/Downloads/webrtc-build-scripts/webrtc-build-scripts/ios/webrtc might not be the file you want to use

________ running '/usr/bin/python -c import os,sys;script = os.path.join("trunk","check_root_dir.py");_ = os.system("%s %s" % (sys.executable,script)) if os.path.exists(script) else 0' in '/Users/huangzan/Downloads/webrtc-build-scripts/webrtc-build-scripts/ios/webrtc'

________ running '/usr/bin/python -u src/sync_chromium.py --target-revision 271c6cca48a6cef32c0f3add3b17b700707deec5' in '/Users/huangzan/Downloads/webrtc-build-scripts/webrtc-build-scripts/ios/webrtc'
Running "gclient sync --force --revision src@271c6cca48a6cef32c0f3add3b17b700707deec5 --gclientfile .gclient.tmp --delete_unversioned_trees --reset --upstream --deps=ios,mac" in /Users/huangzan/Downloads/webrtc-build-scripts/webrtc-build-scripts/ios/webrtc/src/chromium
[0:01:00] Still working on:
[0:01:00] src
[0:01:10] Still working on:
[0:01:10] src
[0:01:20] Still working on:
[0:01:20] src
[0:01:30] Still working on:

XCODE 6.1.1

Hey pristine.io,

Any chance you can push an updated version of your libjingle maven artifact? I've been downloading the webrtc source for the past 16 hours and it's still not done.

Thanks.

Here's a picture of two adorable baby platypuses to encourage you to update the artifact: