Peer Connection example does not work on Linux VM

[GoogleCodeExporter]

What steps will reproduce the problem?

$ gclient config http://webrtc.googlecode.com/svn/trunk
$ gclient sync --force
$ cd trunk
$ ninja -C out/Debug peerconnection_server
$ ninja -C out/Debug peerconnection_client
$ cd trunk/out/Debug 
$ ./peerconnection_server
Server listening on port 8888

New Terminal window

$ cd 'pwd'/trunk/out/Debug 
$ ./peerconnection_client

window appears click connect

open html file in peer connection folder

hit connect in .html 

click connect button in peerconnection_client window

highlight html clien and hit enter

output:

SwitchToConnectUI
Headers received
OnClose
OnPeerConnected
OnSignedIn
SwitchToPeerList
SEND_MESSAGE_TO_PEER
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
ISAC/32000/1 (104)
Warning(webrtcvoiceengine.cc:466): Unexpected codec: ISAC/48000/1 (105)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:466): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:466): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:466): 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
WebRtcVoiceEngine::Init
webrtc: Thread with name:Trace started 
webrtc: Thread with name:ProcessThread started 
webrtc: CheckPlatform
webrtc: current platform is LINUX
webrtc: CreatePlatformSpecificObjects
webrtc: output: kPlatformDefaultAudio
webrtc: attempting to use the Linux PulseAudio APIs...
webrtc: PulseAudioIsSupported
webrtc: SetPulseAudioObjects
webrtc:   the PulseAudio objects for the mixer has been set
webrtc:   checking PulseAudio version: 1.1
webrtc: Close
webrtc: CloseSpeaker
webrtc: CloseMicrophone
webrtc: *** Linux Pulse Audio is supported ***
webrtc: Linux PulseAudio APIs will be utilized
webrtc: AttachAudioBuffer
webrtc: number of registered modules has increased to 2
webrtc:   OS info: Linux
webrtc: SetPulseAudioObjects
webrtc:   the PulseAudio objects for the mixer has been set
webrtc:   checking PulseAudio version: 1.1
webrtc: Thread with name:webrtc_audio_module_rec_thread started 
webrtc:   number of availiable output devices is 2
webrtc: AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex=0)
webrtc:   the output mixer device is now open
webrtc: CloseSpeaker
webrtc: output: available=-1262240475
webrtc: AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex=0)
webrtc:   the output mixer device is now open
webrtc:   number of availiable input devices is 2
webrtc: AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex=1)
webrtc:   the input mixer device is now open
webrtc: CloseMicrophone
webrtc: output: available=1
webrtc: AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex=1)
webrtc:   the input mixer device is now open
webrtc: output: available=1
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
Build: svn:Unavailable(issue687) Nov 17 2013 23:19:22 d
External recording and playout build
Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, 
swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: 
false, experimental_aec: false, aec_dump: false, }
Adjusting AGC level from default -3dB to -3dB
webrtc:      AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable() => 
available=256, available
webrtc: output: available=1
webrtc: 
TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0xb34ec768)
webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0xb34ec768)
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
ISAC/32000/1 (104)
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 3.46.0
Build: svn:Unavailable(issue687) Nov 17 2013 23:19:30 d
VideoEngine Init done
Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, 
swap: false, typing: true, conference: false, agc_delta: 0, experimental_agc: 
false, experimental_aec: false, aec_dump: false, }
Adjusting AGC level from default -3dB to -3dB
Setting microphone to (id=0, name=Default device) and speaker to (id=0, 
name=Default device)
Set microphone to (id=0 name=Default device) and speaker to (id=0 name=Default 
device)
Enumerating V4L2 devices
V4L2 device metadata found at /sys/class/video4linux/
Found V4L2 capture device /dev/video0
Trying /sys/class/video4linux/video0/name
Name for video0 is VirtualBox Webcam - Integrated 
Total V4L2 devices found : 1
Warning(webrtcvideocapturer.cc:170): Failed to find capturer for id: /dev/video0
Error(common.cc:76): ../../talk/app/webrtc/videosource.cc(374): ASSERT FAILED: 
capturer != NULL @ Create
Trace/breakpoint trap (core dumped)

What version of the product are you using? On what operating system?

Using Ubuntu 12.04 LTS

Please provide any additional information below.

Have V4L2 installed. Webcam works with RTC in Chrome web browser.

Virtual machine is oracle virtualbox 4.3.2

Original issue reported on code.google.com by mdelvall...@gmail.com on 17 Nov 2013 at 11:25

[GoogleCodeExporter]

Did you resolve this issue?
Now I am facing this issue, not sure how to proceed.
Any input in this regard, is highly appreciated.
kamal

Original comment by palei.ka...@gmail.com on 11 Apr 2014 at 9:11

[GoogleCodeExporter]

I have the same problem with the v4l2loopback device

Original comment by office.b...@gmail.com on 30 Apr 2014 at 6:04