How to add Audio only call's

[VijaiAaditya]

Is it possible to make audio only call's. Can you please help me how to add voice only call support.

I tried by adding

AudioManager audioManager =
((AudioManager) getSystemService(AUDIO_SERVICE));
// TODO(fischman): figure out how to do this Right(tm) and remove the
// suppression.
@SuppressWarnings("deprecation")
boolean isWiredHeadsetOn = audioManager.isWiredHeadsetOn();
audioManager.setMode(isWiredHeadsetOn ?
AudioManager.MODE_IN_CALL : AudioManager.MODE_IN_COMMUNICATION);
audioManager.setSpeakerphoneOn(!isWiredHeadsetOn);

lMS.addTrack(factory.createAudioTrack("ARDAMSa0"));
pc.addStream(lMS, new MediaConstraints());

pcConstraints.mandatory.add(new MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"));
pcConstraints.mandatory.add(new MediaConstraints.KeyValuePair("OfferToReceiveVideo", "false"));

But nothing helped me to make a audio only call.

[pchab]

Hi,

I've currently no way of testing this, but I will still try to help you.
Here is what I would try :

• In WebRtcClient :
  create a new method that does the same thing as setCamera but without video
  :

  public void setAudio(){
  lMS = factory.createLocalMediaStream("ARDAMS");
  lMS.addTrack(factory.createAudioTrack("ARDAMSa0"));

  mListener.onLocalStream(lMS);
  }

• In RTCactivity :
  in startCam() call setAudio() instead of setCamera(...)

At this point, you should still have audio working and a green (or black)
screen.
You can then :

• Remove the videoStreamsView and videoRenderer from RTCactivity
• Change the PeerConnection Constraints "OfferToReceiveVideo" to false

Hope this will work and good luck !
I'll try to find some androids and time to test this.

Pierre

On Thu, Jun 26, 2014 at 8:37 AM, Vijay notifications@github.com wrote:

Is it possible to make audio only call's. Can you please help me how to
add voice only call support.

I tried by adding

AudioManager audioManager =
((AudioManager) getSystemService(AUDIO_SERVICE));
// TODO(fischman): figure out how to do this Right(tm) and remove the
// suppression.
@SuppressWarnings https://github.com/SuppressWarnings("deprecation")
boolean isWiredHeadsetOn = audioManager.isWiredHeadsetOn();
audioManager.setMode(isWiredHeadsetOn ?
AudioManager.MODE_IN_CALL : AudioManager.MODE_IN_COMMUNICATION);
audioManager.setSpeakerphoneOn(!isWiredHeadsetOn);

lMS.addTrack(factory.createAudioTrack("ARDAMSa0"));
pc.addStream(lMS, new MediaConstraints());

pcConstraints.mandatory.add(new
MediaConstraints.KeyValuePair("OfferToReceiveAudio", "true"));
pcConstraints.mandatory.add(new
MediaConstraints.KeyValuePair("OfferToReceiveVideo", "false"));

But nothing helped me to make a audio only call.

—
Reply to this email directly or view it on GitHub
#8.

[VijaiAaditya]

no Pierre i already tried it . Its not working.

[VijaiAaditya]

I commented out the

@OverRide
public void onLocalStream(MediaStream localStream) {
//localStream.videoTracks.get(0).addRenderer(new VideoRenderer(new VideoCallbacks(vsv, 0)));
//this will not render video component and I can pass only audio
}

from RTCActivity.java and also added

public void setAudio(){
lMS = factory.createLocalMediaStream("ARDAMS");
lMS.addTrack(factory.createAudioTrack("ARDAMSa0"));

mListener.onLocalStream(lMS);

}

now clients are getting connected but app forcefully closed after establishing connection. My logcat log is as bellow. I am unable to understand why its crashing the app.

---

06-27 15:56:16.385: I/libjingle(8296): WebRtcVideoEngine::WebRtcVideoEngine
06-27 15:56:16.385: I/libjingle(8296): WebRtcVoiceEngine::Init
06-27 15:56:16.395: I/libjingle(8296): webrtc: Thread with name:Trace started
06-27 15:56:16.395: I/libjingle(8296): webrtc: Thread with name:ProcessThread started
06-27 15:56:16.395: I/libjingle(8296): webrtc: CheckPlatform
06-27 15:56:16.395: I/libjingle(8296): webrtc: current platform is ANDROID
06-27 15:56:16.395: I/libjingle(8296): webrtc: CreatePlatformSpecificObjects
06-27 15:56:16.395: I/libjingle(8296): webrtc: output: kPlatformDefaultAudio
06-27 15:56:16.395: I/libjingle(8296): webrtc: Android JNI Audio APIs will be utilized
06-27 15:56:16.395: I/libjingle(8296): webrtc: AttachAudioBuffer
06-27 15:56:16.395: I/libjingle(8296): webrtc: number of registered modules has increased to 2
06-27 15:56:16.395: I/libjingle(8296): webrtc: OS info: Android
06-27 15:56:16.395: I/libjingle(8296): webrtc: Trying configured recording sampling rate 16000
06-27 15:56:16.395: I/libjingle(8296): webrtc: Recording sample rate set to (16)
06-27 15:56:16.395: I/libjingle(8296): webrtc: Trying configured playback sampling rate 16000
06-27 15:56:16.405: I/libjingle(8296): webrtc: Playback sample rate set to (16)
06-27 15:56:16.405: I/libjingle(8296): webrtc: Thread with name:jni_audio_capture_thread started
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=2006877095
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=1
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=0
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=0
06-27 15:56:16.405: I/libjingle(8296): webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0x77dc2a28)
06-27 15:56:16.405: I/libjingle(8296): webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0x77dc2a28)
06-27 15:56:16.405: I/libjingle(8296): WebRtc VoiceEngine Version:
06-27 15:56:16.405: I/libjingle(8296): VoiceEngine 4.1.0
06-27 15:56:16.405: I/libjingle(8296): Build: svn:Unavailable(issue687) Aug 14 2013 13:52:46 d
06-27 15:56:16.405: I/libjingle(8296): Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
06-27 15:56:16.405: E/libjingle(8296): Error(webrtcvideoengine.cc:1265): webrtc: (voe_audio_processing_impl.cc:990): virtual int webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not supported
06-27 15:56:16.405: W/libjingle(8296): Warning(webrtcvoiceengine.cc:747): SetTypingDetectionStatus(0) failed, err=8003
06-27 15:56:16.405: I/libjingle(8296): Adjusting AGC level from default -3dB to -3dB
06-27 15:56:16.405: I/libjingle(8296): WebRtc VoiceEngine codecs:
06-27 15:56:16.405: I/libjingle(8296): opus/48000/2 (111)
06-27 15:56:16.405: I/libjingle(8296): ISAC/16000/1 (103)
06-27 15:56:16.405: I/libjingle(8296): G722/16000/1 (9)
06-27 15:56:16.405: I/libjingle(8296): ILBC/8000/1 (102)
06-27 15:56:16.405: I/libjingle(8296): PCMU/8000/1 (0)
06-27 15:56:16.405: I/libjingle(8296): PCMA/8000/1 (8)
06-27 15:56:16.405: I/libjingle(8296): CN/48000/1 (107)
06-27 15:56:16.405: I/libjingle(8296): CN/32000/1 (106)
06-27 15:56:16.405: I/libjingle(8296): CN/16000/1 (105)
06-27 15:56:16.405: I/libjingle(8296): CN/8000/1 (13)
06-27 15:56:16.405: I/libjingle(8296): red/8000/1 (127)
06-27 15:56:16.405: I/libjingle(8296): telephone-event/8000/1 (126)
06-27 15:56:16.415: I/libjingle(8296): WebRtcVoiceEngine::Init Done!
06-27 15:56:16.415: I/libjingle(8296): WebRtcVideoEngine::Init
06-27 15:56:16.415: I/libjingle(8296): WebRtcVideoEngine::InitVideoEngine
06-27 15:56:16.415: I/libjingle(8296): WebRtc VideoEngine Version:
06-27 15:56:16.415: I/libjingle(8296): VideoEngine 3.39.0
06-27 15:56:16.415: I/libjingle(8296): Build: svn:Unavailable(issue687) Aug 14 2013 13:52:43 d
06-27 15:56:16.415: I/libjingle(8296): VideoEngine Init done
06-27 15:56:16.425: I/libjingle(8296): Applying audio options: AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, aec_dump: false, }
06-27 15:56:16.425: W/libjingle(8296): Warning(webrtcvoiceengine.cc:747): SetTypingDetectionStatus(0) failed, err=8003
06-27 15:56:16.425: E/libjingle(8296): Error(webrtcvideoengine.cc:1265): webrtc: (voe_audio_processing_impl.cc:990): virtual int webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not supported
06-27 15:56:16.425: I/libjingle(8296): Adjusting AGC level from default -3dB to -3dB
06-27 15:56:16.446: D/dalvikvm(8296): GC_FOR_ALLOC freed 222K, 3% free 9312K/9564K, paused 19ms, total 19ms
06-27 15:56:16.526: D/fr.pchab.AndroidRTC.WebRtcClient(8296): WebRtcClient connected.
06-27 15:56:16.556: D/fr.pchab.AndroidRTC.WebRtcClient(8296): MessageHandler.onEvent() id
06-27 15:56:16.556: D/HybiParser(8296): Creating frame for: 5:::{"args":[{"type":"init","to":"nxHXPjq2Con4OAkazBRn"}],"name":"message"} op: 1 err: -1
06-27 15:56:16.556: D/HybiParser(8296): Creating frame for: 5:::{"args":[{"privacy":true,"name":"android_test"}],"name":"readyToStream"} op: 1 err: -1
06-27 15:56:16.586: D/fr.pchab.AndroidRTC.WebRtcClient(8296): MessageHandler.onEvent() message
06-27 15:56:16.586: D/fr.pchab.AndroidRTC.WebRtcClient(8296): new Peer: nxHXPjq2Con4OAkazBRn 0
06-27 15:56:16.586: D/dalvikvm(8296): Trying to load lib /data/app-lib/fr.pchab.AndroidRTC-1/libjingle_peerconnection_so.so 0x41e95eb0
06-27 15:56:16.586: D/dalvikvm(8296): Shared lib '/data/app-lib/fr.pchab.AndroidRTC-1/libjingle_peerconnection_so.so' already loaded in same CL 0x41e95eb0
06-27 15:56:16.596: D/fr.pchab.AndroidRTC.WebRtcClient(8296): CreateAnswerCommand
06-27 15:56:16.596: I/libjingle(8296): Ignored line: c=IN IP4 0.0.0.0
06-27 15:56:16.596: I/libjingle(8296): Created channel for audio
06-27 15:56:16.596: I/libjingle(8296): Session:1813408327742074693 Old state:STATE_INIT New state:STATE_RECEIVEDINITIATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
06-27 15:56:16.596: I/libjingle(8296): Setting remote voice description
06-27 15:56:16.596: I/libjingle(8296): Selected voice codec opus/48000/1 (111), bitrate=32000
06-27 15:56:16.596: W/libjingle(8296): Warning(webrtcvoiceengine.cc:1660): Unknown codec ISAC/16000/1 (103)
06-27 15:56:16.596: I/libjingle(8296): Recv stream 3111079734 reuse default channel
06-27 15:56:16.596: I/libjingle(8296): Stopping playout for channel #0
06-27 15:56:16.596: I/libjingle(8296): Add remote ssrc: 3111079734
06-27 15:56:16.596: I/libjingle(8296): Enabling audio level header extension with ID 1
06-27 15:56:16.596: I/libjingle(8296): Setting voice channel options: AudioOptions {}
06-27 15:56:16.596: I/libjingle(8296): Set voice channel options. Current options: AudioOptions {}
06-27 15:56:16.596: I/libjingle(8296): Changing voice state, recv=0 send=0
06-27 15:56:16.596: E/libjingle(8296): Error(webrtcvideoengine.cc:1265): webrtc: (voe_volume_control_impl.cc:545): virtual int webrtc::VoEVolumeControlImpl::SetOutputVolumePan(int, float, float): not supported
06-27 15:56:16.606: W/libjingle(8296): Warning(webrtcvoiceengine.cc:2446): SetOutputVolumePan(0, 1, 1) failed, err=8003
06-27 15:56:16.606: I/libjingle(8296): SetOutputScaling to left=1 right=1 for channel 0 and ssrc 3111079734
06-27 15:56:16.606: D/fr.pchab.AndroidRTC.WebRtcClient(8296): onAddStream VLUlH6RHekGhLlz4nlynWohOdFRZceNGPWJj
06-27 15:56:16.606: W/System.err(8296): java.lang.IndexOutOfBoundsException
06-27 15:56:16.606: W/System.err(8296): at java.util.LinkedList.get(LinkedList.java:519)
06-27 15:56:16.606: W/System.err(8296): at fr.pchab.AndroidRTC.RTCActivity.onAddRemoteStream(RTCActivity.java:145)
06-27 15:56:16.606: W/System.err(8296): at fr.pchab.AndroidRTC.WebRtcClient$Peer.onAddStream(WebRtcClient.java:286)
06-27 15:56:16.606: W/System.err(8296): at dalvik.system.NativeStart.run(Native Method)
06-27 15:56:16.606: E/libjingle(8296): Error(peerconnection_jni.cc:591): ../../talk/app/webrtc/java/jni/peerconnection_jni.cc:591: error during CallVoidMethod
06-27 15:56:16.606: A/libc(8296): Fatal signal 6 (SIGABRT) at 0x00002068 (code=-6), thread 8366 (chab.AndroidRTC)

[deepak-rathi]

Issue 2614: Android libjingle recent compile crashes Application that Java exception can not catch (Reported by stephen....@gmail.com, Nov 11, 2013)

https://code.google.com/p/webrtc/issues/detail?id=2614

[pchab]

I think you're on the right track.
This is a simple error, your remoteStreams doesn't have videoTracks so
get(0) is out of bound in the onAddRemoteStream callback.

You could make it so that the listener callbacks for streams
(onAddRemoteStream, onRemoveRemoteStream, onLocalStream in WebRtcClient)
are not called for a audio-only call.

On Fri, Jun 27, 2014 at 12:41 PM, Vijay notifications@github.com wrote:

I commented out the

@OverRide https://github.com/Override
public void onLocalStream(MediaStream localStream) {
//localStream.videoTracks.get(0).addRenderer(new VideoRenderer(new
VideoCallbacks(vsv, 0)));
//this will not render video component and I can pass only audio
}

from RTCActivity.java and also added

public void setAudio(){
lMS = factory.createLocalMediaStream("ARDAMS");
lMS.addTrack(factory.createAudioTrack("ARDAMSa0"));

mListener.onLocalStream(lMS);

}

now clients are getting connected but app forcefully closed after
establishing connection. My logcat log is as bellow. I am unable to

understand why its crashing the app.

06-27 15:56:16.385: I/libjingle(8296): WebRtcVideoEngine::WebRtcVideoEngine
06-27 15:56:16.385: I/libjingle(8296): WebRtcVoiceEngine::Init
06-27 15:56:16.395: I/libjingle(8296): webrtc: Thread with name:Trace
started
06-27 15:56:16.395: I/libjingle(8296): webrtc: Thread with
name:ProcessThread started
06-27 15:56:16.395: I/libjingle(8296): webrtc: CheckPlatform
06-27 15:56:16.395: I/libjingle(8296): webrtc: current platform is ANDROID
06-27 15:56:16.395: I/libjingle(8296): webrtc:
CreatePlatformSpecificObjects
06-27 15:56:16.395: I/libjingle(8296): webrtc: output:
kPlatformDefaultAudio
06-27 15:56:16.395: I/libjingle(8296): webrtc: Android JNI Audio APIs will
be utilized
06-27 15:56:16.395: I/libjingle(8296): webrtc: AttachAudioBuffer
06-27 15:56:16.395: I/libjingle(8296): webrtc: number of registered
modules has increased to 2
06-27 15:56:16.395: I/libjingle(8296): webrtc: OS info: Android
06-27 15:56:16.395: I/libjingle(8296): webrtc: Trying configured recording
sampling rate 16000
06-27 15:56:16.395: I/libjingle(8296): webrtc: Recording sample rate set
to (16)
06-27 15:56:16.395: I/libjingle(8296): webrtc: Trying configured playback
sampling rate 16000
06-27 15:56:16.405: I/libjingle(8296): webrtc: Playback sample rate set to
(16)
06-27 15:56:16.405: I/libjingle(8296): webrtc: Thread with
name:jni_audio_capture_thread started
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=2006877095
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=1
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=0
06-27 15:56:16.405: I/libjingle(8296): webrtc: output: available=0
06-27 15:56:16.405: I/libjingle(8296): webrtc:
TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0x77dc2a28)
06-27 15:56:16.405: I/libjingle(8296): webrtc:
OutputMixer::SetAudioProcessingModule(audioProcessingModule=0x77dc2a28)
06-27 15:56:16.405: I/libjingle(8296): WebRtc VoiceEngine Version:
06-27 15:56:16.405: I/libjingle(8296): VoiceEngine 4.1.0
06-27 15:56:16.405: I/libjingle(8296): Build: svn:Unavailable(issue687)
Aug 14 2013 13:52:46 d
06-27 15:56:16.405: I/libjingle(8296): Applying audio options:
AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false,
typing: false, conference: false, agc_delta: 0, experimental_agc: false,
experimental_aec: false, aec_dump: false, }
06-27 15:56:16.405: E/libjingle(8296): Error(webrtcvideoengine.cc:1265):
webrtc: (voe_audio_processing_impl.cc:990): virtual int
webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not
supported
06-27 15:56:16.405: W/libjingle(8296): Warning(webrtcvoiceengine.cc:747):
SetTypingDetectionStatus(0) failed, err=8003
06-27 15:56:16.405: I/libjingle(8296): Adjusting AGC level from default
-3dB to -3dB
06-27 15:56:16.405: I/libjingle(8296): WebRtc VoiceEngine codecs:
06-27 15:56:16.405: I/libjingle(8296): opus/48000/2 (111)
06-27 15:56:16.405: I/libjingle(8296): ISAC/16000/1 (103)
06-27 15:56:16.405: I/libjingle(8296): G722/16000/1 (9)
06-27 15:56:16.405: I/libjingle(8296): ILBC/8000/1 (102)
06-27 15:56:16.405: I/libjingle(8296): PCMU/8000/1 (0)
06-27 15:56:16.405: I/libjingle(8296): PCMA/8000/1 (8)
06-27 15:56:16.405: I/libjingle(8296): CN/48000/1 (107)
06-27 15:56:16.405: I/libjingle(8296): CN/32000/1 (106)
06-27 15:56:16.405: I/libjingle(8296): CN/16000/1 (105)
06-27 15:56:16.405: I/libjingle(8296): CN/8000/1 (13)
06-27 15:56:16.405: I/libjingle(8296): red/8000/1 (127)
06-27 15:56:16.405: I/libjingle(8296): telephone-event/8000/1 (126)
06-27 15:56:16.415: I/libjingle(8296): WebRtcVoiceEngine::Init Done!
06-27 15:56:16.415: I/libjingle(8296): WebRtcVideoEngine::Init
06-27 15:56:16.415: I/libjingle(8296): WebRtcVideoEngine::InitVideoEngine
06-27 15:56:16.415: I/libjingle(8296): WebRtc VideoEngine Version:
06-27 15:56:16.415: I/libjingle(8296): VideoEngine 3.39.0
06-27 15:56:16.415: I/libjingle(8296): Build: svn:Unavailable(issue687)
Aug 14 2013 13:52:43 d
06-27 15:56:16.415: I/libjingle(8296): VideoEngine Init done
06-27 15:56:16.425: I/libjingle(8296): Applying audio options:
AudioOptions {aec: true, agc: true, ns: true, hf: true, swap: false,
typing: false, conference: false, agc_delta: 0, experimental_agc: false,
experimental_aec: false, aec_dump: false, }
06-27 15:56:16.425: W/libjingle(8296): Warning(webrtcvoiceengine.cc:747):
SetTypingDetectionStatus(0) failed, err=8003
06-27 15:56:16.425: E/libjingle(8296): Error(webrtcvideoengine.cc:1265):
webrtc: (voe_audio_processing_impl.cc:990): virtual int
webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not
supported
06-27 15:56:16.425: I/libjingle(8296): Adjusting AGC level from default
-3dB to -3dB
06-27 15:56:16.446: D/dalvikvm(8296): GC_FOR_ALLOC freed 222K, 3% free
9312K/9564K, paused 19ms, total 19ms
06-27 15:56:16.526: D/fr.pchab.AndroidRTC.WebRtcClient(8296): WebRtcClient
connected.
06-27 15:56:16.556: D/fr.pchab.AndroidRTC.WebRtcClient(8296):
MessageHandler.onEvent() id
06-27 15:56:16.556: D/HybiParser(8296): Creating frame for:
5:::{"args":[{"type":"init","to":"nxHXPjq2Con4OAkazBRn"}],"name":"message"}
op: 1 err: -1
06-27 15:56:16.556: D/HybiParser(8296): Creating frame for:
5:::{"args":[{"privacy":true,"name":"android_test"}],"name":"readyToStream"}
op: 1 err: -1
06-27 15:56:16.586: D/fr.pchab.AndroidRTC.WebRtcClient(8296):
MessageHandler.onEvent() message
06-27 15:56:16.586: D/fr.pchab.AndroidRTC.WebRtcClient(8296): new Peer:
nxHXPjq2Con4OAkazBRn 0
06-27 15:56:16.586: D/dalvikvm(8296): Trying to load lib
/data/app-lib/fr.pchab.AndroidRTC-1/libjingle_peerconnection_so.so
0x41e95eb0
06-27 15:56:16.586: D/dalvikvm(8296): Shared lib
'/data/app-lib/fr.pchab.AndroidRTC-1/libjingle_peerconnection_so.so'
already loaded in same CL 0x41e95eb0
06-27 15:56:16.596: D/fr.pchab.AndroidRTC.WebRtcClient(8296):
CreateAnswerCommand
06-27 15:56:16.596: I/libjingle(8296): Ignored line: c=IN IP4 0.0.0.0
06-27 15:56:16.596: I/libjingle(8296): Created channel for audio
06-27 15:56:16.596: I/libjingle(8296): Session:1813408327742074693 Old
state:STATE_INIT New state:STATE_RECEIVEDINITIATE
Type:urn:xmpp:jingle:apps:rtp:1 Transport:
http://www.google.com/transport/p2p
06-27 15:56:16.596: I/libjingle(8296): Setting remote voice description
06-27 15:56:16.596: I/libjingle(8296): Selected voice codec opus/48000/1
(111), bitrate=32000
06-27 15:56:16.596: W/libjingle(8296): Warning(webrtcvoiceengine.cc:1660):
Unknown codec ISAC/16000/1 (103)
06-27 15:56:16.596: I/libjingle(8296): Recv stream 3111079734 reuse
default channel
06-27 15:56:16.596: I/libjingle(8296): Stopping playout for channel #0
06-27 15:56:16.596: I/libjingle(8296): Add remote ssrc: 3111079734
06-27 15:56:16.596: I/libjingle(8296): Enabling audio level header
extension with ID 1
06-27 15:56:16.596: I/libjingle(8296): Setting voice channel options:
AudioOptions {}
06-27 15:56:16.596: I/libjingle(8296): Set voice channel options. Current
options: AudioOptions {}
06-27 15:56:16.596: I/libjingle(8296): Changing voice state, recv=0 send=0
06-27 15:56:16.596: E/libjingle(8296): Error(webrtcvideoengine.cc:1265):
webrtc: (voe_volume_control_impl.cc:545): virtual int
webrtc::VoEVolumeControlImpl::SetOutputVolumePan(int, float, float): not
supported
06-27 15:56:16.606: W/libjingle(8296): Warning(webrtcvoiceengine.cc:2446):
SetOutputVolumePan(0, 1, 1) failed, err=8003
06-27 15:56:16.606: I/libjingle(8296): SetOutputScaling to left=1 right=1
for channel 0 and ssrc 3111079734
06-27 15:56:16.606: D/fr.pchab.AndroidRTC.WebRtcClient(8296): onAddStream
VLUlH6RHekGhLlz4nlynWohOdFRZceNGPWJj
06-27 15:56:16.606: W/System.err(8296): java.lang.IndexOutOfBoundsException
06-27 15:56:16.606: W/System.err(8296): at
java.util.LinkedList.get(LinkedList.java:519)
06-27 15:56:16.606: W/System.err(8296): at
fr.pchab.AndroidRTC.RTCActivity.onAddRemoteStream(RTCActivity.java:145)
06-27 15:56:16.606: W/System.err(8296): at
fr.pchab.AndroidRTC.WebRtcClient$Peer.onAddStream(WebRtcClient.java:286)
06-27 15:56:16.606: W/System.err(8296): at
dalvik.system.NativeStart.run(Native Method)
06-27 15:56:16.606: E/libjingle(8296): Error(peerconnection_jni.cc:591):
../../talk/app/webrtc/java/jni/peerconnection_jni.cc:591: error during
CallVoidMethod
06-27 15:56:16.606: A/libc(8296): Fatal signal 6 (SIGABRT) at 0x00002068
(code=-6), thread 8366 (chab.AndroidRTC)

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#8 (comment).

[VijaiAaditya]

ok thank u very much Pierre Chabardes its worked .

[EvaniRam]

Is it possible to improve voice call clarity by noise and echo cancellation ?? . Can you please help me in how to remove noise and echo in android application.

[borhanreo]

How can i get all user list? please help me