新问题,看应该是视频编解码那里出问题了.找找看.
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
ISAC/32000/1 (104)
Unexpected codec: ISAC/48000/1 (105)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Unexpected codec: PCMU/8000/2 (110)
Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/8000/1 (9)
Unexpected codec: G722/8000/2 (119)
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: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -18774
webrtc: CheckPlatform
webrtc: current platform is IOS
webrtc: CreatePlatformSpecificObjects
webrtc: iPhone Audio APIs will be utilized
webrtc: AttachAudioBuffer
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -134
webrtc: output: available=0
webrtc: output: available=0
webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0x2e630510)
webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0x2e630510)
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, extended_filter_aec: false, delay_agnostic_aec: false, experimental_ns: false, aec_dump: false, }
webrtc: Built-in AEC not supported on this platform
High pass filter enabled? 1
Stereo swapping enabled? 0
NetEq capacity is 50
NetEq fast mode? 0
Typing detection is enabled? 0
webrtc: (voe_audio_processing_impl.cc:1007): virtual int webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not supported
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
Extended filter aec is enabled? 0
Experimental ns is enabled? 0
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
ISAC/32000/1 (104)
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, audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, extended_filter_aec: false, delay_agnostic_aec: false, experimental_ns: false, aec_dump: false, }
webrtc: Built-in AEC not supported on this platform
High pass filter enabled? 1
Stereo swapping enabled? 0
NetEq capacity is 50
NetEq fast mode? 0
Typing detection is enabled? 0
webrtc: (voe_audio_processing_impl.cc:1007): virtual int webrtc::VoEAudioProcessingImpl::SetTypingDetectionStatus(bool): not supported
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
Extended filter aec is enabled? 0
Experimental ns is enabled? 0
Making key pair
2015-07-28 17:35:44.626 rtcdemo[215:4150] Client connecting.
2015-07-28 17:35:44.631 rtcdemo[215:4150] Joining room:365865178 on room server.
Returning key pair
Making certificate for WebRTC
Returning certificate
2015-07-28 17:35:50.233 rtcdemo[215:4150] Joined room:365865178 on room server.
2015-07-28 17:35:50.236 rtcdemo[215:4150] SocketRocket: In debug mode. Allowing connection to any root cert
2015-07-28 17:35:50.237 rtcdemo[215:4150] Opening WebSocket.
2015-07-28 17:35:50.239 rtcdemo[215:4150] Client connected.
Allowing SCTP data engine.
Making key pair
Returning key pair
Making certificate for WebRTC
Returning certificate
Capture Requested 640x480x30
Supported NV12 640x480x30 distance 0
Best NV12 640x480x30 Interval 33333333 distance 0
VAdapt input interval changed from 0 to 33333333
2015-07-28 17:35:50.931 rtcdemo[215:4536] WARNING: Renegotiation needed but unimplemented.
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
webrtc: (neteq_impl.cc:98): NetEq config: sample_rate_hz=16000, enable_audio_classifier=false, max_packets_in_buffer=50, background_noise_mode=2, playout_mode=0, enable_fast_accelerate=0
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -1008
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -248
webrtc: Channel::SendRTCPPacket() failed to send RTCP packet due to invalid transport object
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -323
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -2
webrtc: Channel::SendRTCPPacket() failed to send RTCP packet due to invalid transport object
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -3003
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -1
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -1
Setting voice channel options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -1
Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_max_packets: 50, audio_jitter_buffer_fast_accelerate: false, }
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -64
Created channel for audio
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -2
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -190
CreateChannel: With voice channel. Options: VideoOptions {process: 0.1, low: 0.65, high: 0.85, num channels for early receive: 0, }
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -1
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -1
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -65
webrtc: (remote_bitrate_estimator_single_stream.cc:285): RemoteBitrateEstimatorFactory: Instantiating.
webrtc: (process_thread_impl.cc:31): TimeUntilNextProcess returned an invalid value -2165
webrtc: (send_side_bandwidth_estimation.cc:290): Estimated available bandwidth 0 kbps is below configured min bitrate 10 kbps.
webrtc: (generic_encoder.cc:94): Failed to initialize the encoder associated with payload name: VP8
webrtc: (codec_database.cc:307): Failed to initialize video encoder.
webrtc: (video_sender.cc:107): Failed to initialize set encoder with payload name 'VP8'.
#
# Fatal error in ../../../mysrc/webrtc/video//call.cc, line 179
# Check failed: channel_group_->CreateSendChannel( base_channel_id_, 0, &transport_adapter_, num_cpu_cores_, true)
#
#
(lldb)
