search

gshORTON / webm

Automatically exported from code.google.com/p/webm
0 stars 0 forks source link

webrtc and mobilevlckit #1000

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago 
Hi,
I'm using webrtc and mobilevlckit linked to my ios-project as a static 
libraries. I faced with an error described below. One webrtc's dev referenced 
my here. Any idea what problem is?

2015-04-30 18:45:55.230 asd[1893:451287] MYRTC peerConnectionWithICEServers
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:566): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:566): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/8000/1 (9)
Warning(webrtcvoiceengine.cc:566): Unexpected codec: G722/8000/2 (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)
WebRtcVideoEngine2::WebRtcVideoEngine2()
WebRtcVoiceEngine::Init
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
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, delay_agnostic_aec: false, 
experimental_ns: false, aec_dump: false, }
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 0
Error(webrtcvoiceengine.cc:1443): webrtc: (voe_audio_processing_impl.cc:1005): 
virtual int webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not 
supported
Warning(webrtcvoiceengine.cc:959): SetTypingDetectionStatus(0) failed, err=8003
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Delay agnostic aec is enabled? 0
Experimental aec is enabled? 0
Experimental ns is enabled? 0
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
G722/8000/1 (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!
WebRtcVideoEngine2::Init
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, delay_agnostic_aec: false, 
experimental_ns: false, aec_dump: false, }
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 0
Error(webrtcvoiceengine.cc:1443): webrtc: (voe_audio_processing_impl.cc:1005): 
virtual int webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not 
supported
Warning(webrtcvoiceengine.cc:959): SetTypingDetectionStatus(0) failed, err=8003
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Delay agnostic aec is enabled? 0
Experimental aec is enabled? 0
Experimental ns is enabled? 0
Making key pair
Allowing SCTP data engine.
Returning key pair
Making certificate for WebRTC
Returning certificate
Making key pair
2015-04-30 18:45:56.192 asd[1893:451287] ws > message
{"event":"message","data":{"payload":{"type":"offer","sdp":"v=0\r\no=- 
6648610871812824200 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio 
video\r\na=msid-semantic: WMS ARDAMS\r\nm=audio 9 RTP/SAVPF 111 103 9 102 0 8 
106 105 13 127 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 
0.0.0.0\r\na=ice-ufrag:PIH+Zs0Hf+gZKpP3\r\na=ice-pwd:wqB4fckBDuTXGdA2oTLJ3rW9\r\
na=fingerprint:sha-1 
B0:B1:7F:90:45:CC:A6:A0:43:E5:8E:78:1F:06:15:8A:76:0A:25:98\r\na=setup:actpass\r
\na=mid:audio\r\na=extmap:1 
urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:3 
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=sendrecv\r\na=rt
cp-mux\r\na=rtpmap:111 opus/48000/2\r\na=fmtp:111 minptime=10; 
useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:9 
G722/8000\r\na=rtpmap:102 ILBC/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 
PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 
CN/8000\r\na=rtpmap:127 red/8000\r\na=rtpmap:126 
telephone-event/8000\r\na=maxptime:60\r\na=ssrc:3097431544 
cname:WxA3eSfZkgRrE+S7\r\na=ssrc:3097431544 msid:ARDAMS 
ARDAMSa0\r\na=ssrc:3097431544 mslabel:ARDAMS\r\na=ssrc:3097431544 
label:ARDAMSa0\r\nm=video 9 RTP/SAVPF 100 116 117 120 96\r\nc=IN IP4 
0.0.0.0\r\na=rtcp:9 IN IP4 
0.0.0.0\r\na=ice-ufrag:PIH+Zs0Hf+gZKpP3\r\na=ice-pwd:wqB4fckBDuTXGdA2oTLJ3rW9\r\
na=fingerprint:sha-1 
B0:B1:7F:90:45:CC:A6:A0:43:E5:8E:78:1F:06:15:8A:76:0A:25:98\r\na=setup:actpass\r
\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 
http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=sendrecv\r\na=rt
cp-mux\r\na=rtpmap:100 VP8/90000\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 
nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:100 goog-remb\r\na=rtpmap:116 
red/90000\r\na=rtpmap:117 ulpfec/90000\r\na=rtpmap:120 
H264/90000\r\na=rtcp-fb:120 ccm fir\r\na=rtcp-fb:120 nack\r\na=rtcp-fb:120 nack 
pli\r\na=rtcp-fb:120 goog-remb\r\na=rtpmap:96 rtx/90000\r\na=fmtp:96 
apt=100\r\na=ssrc-group:FID 6743164 559108273\r\na=ssrc:6743164 
cname:WxA3eSfZkgRrE+S7\r\na=ssrc:6743164 msid:ARDAMS ARDAMSv0\r\na=ssrc:6743164 
mslabel:ARDAMS\r\na=ssrc:6743164 label:ARDAMSv0\r\na=ssrc:559108273 
cname:WxA3eSfZkgRrE+S7\r\na=ssrc:559108273 msid:ARDAMS 
ARDAMSv0\r\na=ssrc:559108273 mslabel:ARDAMS\r\na=ssrc:559108273 
label:ARDAMSv0\r\n"},"from":"C53E5F70-8CD6-4CFE-95C4-08AD5274CE40","type":"offer
"}}
2015-04-30 18:45:56.714 asd[1893:451287] MYRTC setRemoteDescriptionWithDelegate
2015-04-30 18:45:56.715 asd[1893:451287] MYRTC peerConnection 
setRemoteDescriptionWithDelegate
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Returning key pair
Making certificate for WebRTC
Returning certificate
Created channel for audio
Setting voice channel options: AudioOptions {}
Set voice channel options.  Current options: AudioOptions {}
CreateChannel: With voice channel. Options: VideoOptions {process: 0.1, low: 
0.65, high: 0.85, num channels for early receive: 0, }
webrtc: (vie_render_impl.cc:64): virtual int 
webrtc::ViERenderImpl::RegisterVideoRenderModule(webrtc::VideoRender &): 
webrtc: (remote_bitrate_estimator_single_stream.cc:285): 
RemoteBitrateEstimatorFactory: Instantiating.
Error(webrtcvoiceengine.cc:1443): webrtc: (generic_encoder.cc:94): Failed to 
initialize the encoder associated with payload name: VP8
Error(webrtcvoiceengine.cc:1443): webrtc: (codec_database.cc:307): Failed to 
initialize video encoder.
Error(webrtcvoiceengine.cc:1443): webrtc: (video_sender.cc:144): Failed to 
initialize set encoder with payload name 'VP8'.

#
# Fatal error in ../../webrtc/video/call.cc, line 224
# Check failed: base_channel_id_ != -1
#

Original issue reported on code.google.com by unregister03 on 7 May 2015 at 2:10

GoogleCodeExporter commented 9 years ago 
marco, 

Can you take a look to see if this should go to webrtc's issue tracker?

Thanks

Original comment by ya...@google.com on 7 May 2015 at 10:13

GoogleCodeExporter commented 9 years ago

Original comment by ya...@google.com on 7 May 2015 at 10:14

GoogleCodeExporter commented 9 years ago 
yes should be filed in webrtc issue tracker: 
https://code.google.com/p/webrtc/issues/list

Original comment by marpan@google.com on 7 May 2015 at 10:20

GoogleCodeExporter commented 9 years ago 
[deleted comment]
GoogleCodeExporter commented 9 years ago 
https://code.google.com/p/webrtc/issues/detail?id=4622 
They suggested to post here

Original comment by unregister03 on 8 May 2015 at 7:54