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MarshalX / tgcalls

Voice chats, private incoming and outgoing calls in Telegram for Developers
https://t.me/tgcallslib
GNU Lesser General Public License v3.0
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[2.0.0dev3] munmap_chunk(): invalid pointer when using asyncio.sleep #129

Closed nitanmarcel closed 3 years ago

nitanmarcel commented 3 years ago

In the dev version there's a new error munmap_chunk(): invalid pointer when asyncio.sleep or something similar is used.

TgCalls version:

pytgcalls==2.0.0.dev3
tgcalls==2.0.0.dev2

Example to reproduce:

import asyncio
import os
from pytgcalls import GroupCallFactory
import pyrogram

API_HASH = None
API_ID = None

# chat or channel where you want to play audiofile_example_MP3_700KB.mp3
CHAT_PEER = "@tgcallschat"
CLIENT_TYPE = GroupCallFactory.MTPROTO_CLIENT_TYPE.PYROGRAM
# for Telethon uncomment line below
# CLIENT_TYPE = pytgcalls.GroupCallFactory.MTPROTO_CLIENT_TYPE.TELETHON

def on_played_data(gc, length):
    pass

async def main(client):
    await client.start()
    while not client.is_connected:
        await asyncio.sleep(1)

    group_call_factory = GroupCallFactory(client, CLIENT_TYPE)
    group_call_raw = group_call_factory.get_raw_group_call(
        on_played_data=on_played_data)
    await group_call_raw.start(CHAT_PEER)

    await asyncio.sleep(1) # Removing this exists the script succesfully.

if __name__ == "__main__":
    pyro_client = pyrogram.Client(
        os.environ.get('SESSION_NAME', 'pytgcalls'),
        api_hash=os.environ.get('API_HASH', API_HASH),
        api_id=os.environ.get('API_ID', API_ID)
    )

    loop = asyncio.get_event_loop()
    loop.run_until_complete(main(pyro_client))

Logs:

Pyrogram v1.2.9, Copyright (C) 2017-2021 Dan <https://github.com/delivrance>
Licensed under the terms of the GNU Lesser General Public License v3 or later (LGPLv3+)

DEBUG:pytgcalls.dispatcher.dispatcher:Build storage of handlers for dispatcher.
DEBUG:pytgcalls.implementation.group_call_native:Create a new native instance...
tgcalls v2.0.0.2 DEV, Copyright (C) 2020-2021 Il`ya (Marshal) <https://github.com/MarshalX>
Licensed under the terms of the GNU Lesser General Public License v3 (LGPLv3)

DEBUG:pytgcalls.implementation.group_call_native:Native instance created.
DEBUG:pytgcalls.implementation.group_call_native:Start native group call...
DEBUG:pytgcalls.implementation.group_call_native:Emit join payload...
DEBUG:pytgcalls.implementation.group_call_native:Successfully connected to VC with ssrc=98920714 as InputPeerUser.
DEBUG:pytgcalls.implementation.group_call_native:Group call update...
DEBUG:pytgcalls.implementation.group_call_native:<UpdateGroupCallWrapper>({'chat_id': 1412793637, 'call': <GroupCallWrapper>({'id': -6067902465567988211, 'params': pyrogram.raw.types.DataJSON(data='{"transport":{"candidates":[{"generation":"0","component":"1","protocol":"udp","port":"32000","ip":"2001:67c:4e8:f102:4:0:285:4","foundation":"1","id":"6b591e6557bd78d102580cfe6","priority":"2130706431","type":"host","network":"0"},{"generation":"0","component":"1","protocol":"udp","port":"32000","ip":"91.108.9.68","foundation":"2","id":"53e534b157bd78d105b7ae1a8","priority":"2130706431","type":"host","network":"0"}],"xmlns":"urn:xmpp:jingle:transports:ice-udp:1","ufrag":"7uno61fcr8bjtb","rtcp-mux":true,"pwd":"27eo6b36jirs47e35rchb0h16m","fingerprints":[{"fingerprint":"79:6B:52:8C:E4:1C:0F:8A:BC:C7:7D:43:6F:41:3B:6A:91:21:F6:42:77:01:4E:8B:D5:10:E7:CE:9D:5B:44:60","setup":"active","hash":"sha-256"}]},"audio":{"payload-types":[{"id":111,"name":"opus","clockrate":48000,"channels":2,"parameters":{"minptime":10,"useinbandfec":1},"rtcp-fbs":[{"type":"transport-cc"}]},{"id":126,"name":"telephone-event","clockrate":8000,"channels":1}],"rtp-hdrexts":[{"id":1,"uri":"urn:ietf:params:rtp-hdrext:ssrc-audio-level"},{"id":2,"uri":"http:\\/\\/www.webrtc.org\\/experiments\\/rtp-hdrext\\/abs-send-time"},{"id":3,"uri":"http:\\/\\/www.ietf.org\\/id\\/draft-holmer-rmcat-transport-wide-cc-extensions-01"}]}}')})})
DEBUG:pytgcalls.implementation.group_call_native:Set join response payload...
DEBUG:pytgcalls.implementation.group_call_native:Set connection mode GroupConnectionMode.GroupConnectionModeRtc.
DEBUG:pytgcalls.implementation.group_call_native:Join response payload was set.
DEBUG:pytgcalls.implementation.group_call_native:Group call participants update...
DEBUG:pytgcalls.implementation.group_call_native:<UpdateGroupCallParticipantsWrapper>({'participants': [<GroupCallParticipantWrapper>({'source': 98920714, 'left': False, 'peer': pyrogram.raw.types.PeerUser(user_id=942356877), 'muted': True, 'can_self_unmute': False, 'is_self': True})]})
DEBUG:pytgcalls.implementation.group_call_native:Group call participants update...
DEBUG:pytgcalls.implementation.group_call_native:<UpdateGroupCallParticipantsWrapper>({'participants': [<GroupCallParticipantWrapper>({'source': 98920714, 'left': False, 'peer': pyrogram.raw.types.PeerUser(user_id=942356877), 'muted': True, 'can_self_unmute': False, 'is_self': True})]})
DEBUG:pytgcalls.implementation.group_call_native:Network state updated...
DEBUG:pytgcalls.dispatcher.dispatcher:Trigger NETWORK_STATUS_CHANGED handlers...
DEBUG:pytgcalls.implementation.group_call_native:Set is muted on native instance side. New value: False.
DEBUG:pytgcalls.dispatcher.dispatcher:Get NETWORK_STATUS_CHANGED handlers...
DEBUG:pytgcalls.implementation.group_call_native:Set is muted on server side. New value: False.
DEBUG:pytgcalls.implementation.group_call_native:New network state is True.
DEBUG:pytgcalls.implementation.group_call_native:Start status (call action) worker...
munmap_chunk(): invalid pointer
Aborted

WebRTC Logs:

INFO:pyrogram.crypto.aes:Using TgCrypto
(field_trial.cc:140): Setting field trial string:WebRTC-Audio-Allocation/min:32kbps,max:2048kbps/WebRTC-Audio-OpusMinPacketLossRate/Enabled-1/WebRTC-TaskQueuePacer/Enabled/WebRTC-VP8ConferenceTemporalLayers/1/
(openssl_key_pair.cc:38): Making key pair
(audio_processing_impl.cc:277): Injected APM submodules:
Echo control factory: 0
Echo detector: 0
Capture analyzer: 1
Capture post processor: 0
Render pre processor: 0
(openssl_key_pair.cc:91): Returning key pair
(openssl_certificate.cc:59): Making certificate for WebRTC
(WrappedAudioDeviceModuleImpl.cpp:16): Create
(WrappedAudioDeviceModuleImpl.cpp:52): CreateForTest
(audio_device_buffer.cc:64): AudioDeviceBuffer::ctor
(audio_device_impl.cc:137): current platform is Linux
(openssl_certificate.cc:109): Returning certificate
(audio_device_impl.cc:313): AttachAudioBuffer
(audio_device_buffer.cc:180): SetRecordingSampleRate(0)
(audio_device_buffer.cc:186): SetPlayoutSampleRate(0)
(audio_device_buffer.cc:200): SetRecordingChannels(0)
(audio_device_buffer.cc:206): SetPlayoutChannels(0)
(audio_device_impl.cc:333): Init
(dtls_srtp_transport.cc:62): Setting RTCP Transport on null transport 0
(webrtc_voice_engine.cc:269): WebRtcVoiceEngine::WebRtcVoiceEngine
(dtls_srtp_transport.cc:67): Setting RTP Transport on null transport 0
(webrtc_voice_engine.cc:291): WebRtcVoiceEngine::Init
(basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
(basic_port_allocator.cc:375): Start getting ports with turn_port_prune_policy 0
(p2p_transport_channel.cc:532): Set backup connection ping interval to 25000 milliseconds.
(p2p_transport_channel.cc:541): Set ICE receiving timeout to 2500 milliseconds
(p2p_transport_channel.cc:548): Set ping most likely connection to 1
(p2p_transport_channel.cc:555): Set stable_writable_connection_ping_interval to 2500
(p2p_transport_channel.cc:568): Set presume writable when fully relayed to 0
(p2p_transport_channel.cc:586): Set regather_on_failed_networks_interval to 8000
(p2p_transport_channel.cc:593): Set receiving_switching_delay to 1000
(p2p_transport_channel.cc:466): Set ICE ufrag: aykE pwd: PXQETX8zVxkinkbbyg/Cv8ro on transport transport
(dtls_srtp_transport.cc:62): Setting RTCP Transport on transport transport 0
(dtls_srtp_transport.cc:67): Setting RTP Transport on transport transport 50007580
(audio_device_impl.cc:333): Init
(audio_device_impl.cc:677): SetPlayoutDevice(0)
(audio_device_impl.cc:366): InitSpeaker
(adm_helpers.cc:48): Unable to access speaker.
(audio_device_impl.cc:581): StereoPlayoutIsAvailable
(audio_device_impl.cc:588): output: 1
(audio_device_impl.cc:593): SetStereoPlayout(1)
(audio_device_buffer.cc:206): SetPlayoutChannels(2)
(audio_device_impl.cc:739): SetRecordingDevice(0)
(audio_device_impl.cc:372): InitMicrophone
(audio_device_impl.cc:535): StereoRecordingIsAvailable
(audio_device_impl.cc:542): output: 1
(audio_device_impl.cc:547): SetStereoRecording(1)
(audio_device_buffer.cc:200): SetRecordingChannels(2)
(audio_device_impl.cc:852): RegisterAudioCallback
(webrtc_voice_engine.cc:387): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 1, agc: 1, ns: 1, hf: 1, swap: 0, audio_jitter_buffer_max_packets: 200, audio_jitter_buffer_fast_accelerate: 0, audio_jitter_buffer_min_delay_ms: 0, audio_jitter_buffer_enable_rtx_handling: 0, typing: 1, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 1, }
(audio_device_impl.cc:868): BuiltInAECIsAvailable
(basic_port_allocator.cc:111): Filtered out ignored networks:
(audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform
(audio_device_impl.cc:871): output: 0
(audio_device_impl.cc:884): BuiltInAGCIsAvailable
(audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform
(audio_device_impl.cc:887): output: 0
(audio_device_impl.cc:900): BuiltInNSIsAvailable
(audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform
(audio_device_impl.cc:903): output: 0
(webrtc_voice_engine.cc:496): Stereo swapping enabled? 0
(webrtc_voice_engine.cc:501): NetEq capacity is 200
(webrtc_voice_engine.cc:507): NetEq fast mode? 0
(webrtc_voice_engine.cc:513): NetEq minimum delay is 0
(webrtc_voice_engine.cc:519): NetEq handle reordered packets? 0
(webrtc_voice_engine.cc:539): Experimental ns is enabled? 0
(basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
(webrtc_voice_engine.cc:590): NS set to 1
(basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
(basic_port_allocator.cc:861): Network manager has started
(basic_port_allocator.cc:111): Filtered out ignored networks:
(basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
(basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
(basic_port_allocator.cc:861): Network manager has started
(webrtc_voice_engine.cc:594): Typing detection is enabled? 1
(basic_port_allocator.cc:111): Filtered out ignored networks:
(basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
(basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
(basic_port_allocator.cc:111): Filtered out ignored networks:
(basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
(basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
(audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255 }, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
(basic_port_allocator.cc:111): Filtered out ignored networks:
(basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
(basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
(basic_port_allocator.cc:776): Allocate ports on 1 networks
(basic_port_allocator.cc:111): Filtered out ignored networks:
(basic_port_allocator.cc:113): Net[lo:0:0:0:x:x:x:x:x/128:Loopback:id=3]
(basic_port_allocator.cc:113): Net[lo:127.0.0.x/8:Loopback:id=2]
(basic_port_allocator.cc:776): Allocate ports on 1 networks
(basic_port_allocator.cc:1361): Net[eth0:172.26.208.x/20:Ethernet:id=1]: Allocation Phase=Udp
(port.cc:186): Port[50008b70::1:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port created with network cost 0
(basic_port_allocator.cc:885): Adding allocated port for
(basic_port_allocator.cc:907): Port[50008b70::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Added port to allocator
(basic_port_allocator.cc:925): Port[50008b70::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Gathered candidate: Cand[:1775382669:0:udp:2122260224:172.26.223.x:49257:local::0:gtub:en+GY6Ywf7nVYrSzRX6hShHr:1:0:0]
(basic_port_allocator.cc:958): Port[50008b70::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port ready.
(basic_port_allocator.cc:1069): Port[50008b70::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port completed gathering candidates.
(basic_port_allocator.cc:1474): AllocationSequence: No STUN server configured, skipping.
(basic_port_allocator.cc:1361): Net[eth0:172.26.208.x/20:Ethernet:id=1]: Allocation Phase=Udp
(port.cc:186): Port[50005410::1:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port created with network cost 0
(agc_manager_direct.cc:68): [agc] GetMinMicLevel
(basic_port_allocator.cc:885): Adding allocated port for
(basic_port_allocator.cc:907): Port[50005410::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Added port to allocator
(agc_manager_direct.cc:72): [agc] Using default min mic level: 12
(basic_port_allocator.cc:925): Port[50005410::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Gathered candidate: Cand[:1775382669:0:udp:2122260224:172.26.223.x:34481:local::0:YRAv:BvvijVGhCHBCEv11oe73gPQT:1:0:0]
(basic_port_allocator.cc:958): Port[50005410::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port ready.
(basic_port_allocator.cc:1069): Port[50005410::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port completed gathering candidates.
(basic_port_allocator.cc:1474): AllocationSequence: No STUN server configured, skipping.
(audio_device_impl.cc:845): Recording
(audio_device_impl.cc:745): SetRecordingDevice
(AudioDeviceHelper.cpp:33): setAudioInputDevice(): SetRecordingDevice(kDefaultCommunicationDevice) failed: -1.
(audio_device_impl.cc:814): Playing
(audio_device_impl.cc:683): SetPlayoutDevice
(AudioDeviceHelper.cpp:81): setAudioOutputDevice(): SetPlayoutDevice(kDefaultCommunicationDevice) failed: -1.
(audio_device_impl.cc:751): InitPlayout
(audio_device_impl.cc:777): PlayoutIsInitialized
(audio_device_buffer.cc:186): SetPlayoutSampleRate(48000)
(audio_device_buffer.cc:206): SetPlayoutChannels(2)
(audio_device_impl.cc:757): output: 0
(audio_device_impl.cc:789): StartPlayout
(audio_device_impl.cc:814): Playing
(RawAudioDevice.cpp:199): Started playout capture Python callback
(audio_device_impl.cc:796): output: 0
(audio_device_buffer.cc:287): Size of playout buffer: 960
(bitrate_prober.cc:72): Bandwidth probing enabled, set to inactive
(cpu_info.cc:53): Available number of cores: 8
(aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
(remote_bitrate_estimator_single_stream.cc:72): RemoteBitrateEstimatorSingleStream: Instantiating.
(remote_estimator_proxy.cc:50): Maximum interval between transport feedback RTCP messages (ms): 250
(webrtc_voice_engine.cc:1619): Setting voice channel options: AudioOptions {audio_jitter_buffer_fast_accelerate: 1, audio_jitter_buffer_min_delay_ms: 50, }
(webrtc_voice_engine.cc:387): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_fast_accelerate: 1, audio_jitter_buffer_min_delay_ms: 50, }
(webrtc_voice_engine.cc:507): NetEq fast mode? 1
(webrtc_voice_engine.cc:513): NetEq minimum delay is 50
(webrtc_voice_engine.cc:539): Experimental ns is enabled? 0
(audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255 }, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
(webrtc_voice_engine.cc:1637): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_fast_accelerate: 1, audio_jitter_buffer_min_delay_ms: 50, }
(channel.cc:150): Created channel: {mid: audio1, media_type: audio}
(rtp_demuxer.cc:154): Added sink = 48044c08 for criteria {mid: audio1, rsid: <empty>, ssrcs: [], payload_types = []}
(channel.cc:911): Setting local voice description for {mid: audio1, media_type: audio}
(webrtc_voice_engine.cc:1479): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[112:l16:48000:0:1]], extensions: []}
(webrtc_voice_engine.cc:1647): Setting receive voice codecs.
(rtp_demuxer.cc:250): Removed sink = 48044c08 bindings
(rtp_demuxer.cc:154): Added sink = 48044c08 for criteria {mid: audio1, rsid: <empty>, ssrcs: [], payload_types = [111, 112, ]}
(channel.cc:902): Changing voice state, recv=0 send=0 for {mid: audio1, media_type: audio}
(channel.cc:978): Setting remote voice description for {mid: audio1, media_type: audio}
(webrtc_voice_engine.cc:1433): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:0:2], AudioCodec[112:l16:48000:0:1]], extensions: [], extmap-allow-mixed: true, max_bandwidth_bps: 1300000, mid: audio1, options: AudioOptions {}}
(webrtc_voice_engine.cc:2341): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
(webrtc_voice_engine.cc:1619): Setting voice channel options: AudioOptions {}
(webrtc_voice_engine.cc:387): WebRtcVoiceEngine::ApplyOptions: AudioOptions {audio_jitter_buffer_fast_accelerate: 1, audio_jitter_buffer_min_delay_ms: 50, }
(webrtc_voice_engine.cc:507): NetEq fast mode? 1
(webrtc_voice_engine.cc:513): NetEq minimum delay is 50
(webrtc_voice_engine.cc:539): Experimental ns is enabled? 0
(audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 1 }, echo_canceller: { enabled: 1, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 1, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 1 }, gain_controller1: { enabled: 1, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255 }, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 1 }, level_estimation: { enabled: 0 }}
(webrtc_voice_engine.cc:1637): Set voice channel options. Current options: AudioOptions {audio_jitter_buffer_fast_accelerate: 1, audio_jitter_buffer_min_delay_ms: 50, }
(webrtc_voice_engine.cc:1999): AddRecvStream: {ssrcs:[1];ssrc_groups:;stream_ids:stream1;}
(delay_manager.cc:85): Delay manager config: quantile=0.97 forget_factor=0.9993 start_forget_weight=2 resample_interval_ms=0 max_history_ms=2000
(decision_logic.cc:64): NetEq decision logic settings: estimate_dtx_delay=1 time_stretch_cn=1 target_level_window_ms=100
(neteq_impl.cc:166): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=true, enable_muted_state=true, enable_rtx_handling=false, extra_output_delay_ms=0
(audio_receive_stream.cc:124): AudioReceiveStream: 1
(rtp_demuxer.cc:154): Added sink = 48093e90 for criteria {mid: <empty>, rsid: <empty>, ssrcs: [1, ], payload_types = []}
(audio_receive_stream.cc:391): AudioReceiveStream::ConfigureStream: {rtp: {remote_ssrc: 1, local_ssrc: 4195875351, transport_cc: off, nack: {rtp_history_ms: 0}, extensions: []}, rtcp_send_transport: (Transport), sync_group: stream1}
(channel.cc:757): Add remote ssrc: 1 to {mid: audio1, media_type: audio}
(rtp_demuxer.cc:250): Removed sink = 48044c08 bindings
(rtp_demuxer.cc:154): Added sink = 48044c08 for criteria {mid: audio1, rsid: <empty>, ssrcs: [1, ], payload_types = [111, 112, ]}
(channel.cc:902): Changing voice state, recv=0 send=0 for {mid: audio1, media_type: audio}
(webrtc_voice_engine.cc:2064): ResetUnsignaledRecvStream.
(rtp_demuxer.cc:250): Removed sink = 48044c08 bindings
(rtp_demuxer.cc:154): Added sink = 48044c08 for criteria {mid: audio1, rsid: <empty>, ssrcs: [1, ], payload_types = []}
(channel.cc:547): Channel enabled: {mid: audio1, media_type: audio}
(audio_device_impl.cc:814): Playing
(channel.cc:902): Changing voice state, recv=1 send=0 for {mid: audio1, media_type: audio}
(basic_port_allocator.cc:1361): Net[eth0:172.26.208.x/20:Ethernet:id=1]: Allocation Phase=Relay
(basic_port_allocator.cc:1361): Net[eth0:172.26.208.x/20:Ethernet:id=1]: Allocation Phase=Relay
(basic_port_allocator.cc:1361): Net[eth0:172.26.208.x/20:Ethernet:id=1]: Allocation Phase=Tcp
(port.cc:186): Port[50029ff0::1:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port created with network cost 0
(basic_port_allocator.cc:885): Adding allocated port for
(basic_port_allocator.cc:907): Port[50029ff0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Added port to allocator
(basic_port_allocator.cc:925): Port[50029ff0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Gathered candidate: Cand[:659672189:0:tcp:1518280448:172.26.223.x:58193:local::0:gtub:en+GY6Ywf7nVYrSzRX6hShHr:1:0:0]
(basic_port_allocator.cc:958): Port[50029ff0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port ready.
(basic_port_allocator.cc:1069): Port[50029ff0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port completed gathering candidates.
(basic_port_allocator.cc:1142): All candidates gathered for pooled session.
(basic_port_allocator.cc:1361): Net[eth0:172.26.208.x/20:Ethernet:id=1]: Allocation Phase=Tcp
(port.cc:186): Port[5002a7b0::1:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port created with network cost 0
(basic_port_allocator.cc:885): Adding allocated port for
(basic_port_allocator.cc:907): Port[5002a7b0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Added port to allocator
(basic_port_allocator.cc:925): Port[5002a7b0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Gathered candidate: Cand[:659672189:0:tcp:1518280448:172.26.223.x:48809:local::0:YRAv:BvvijVGhCHBCEv11oe73gPQT:1:0:0]
(basic_port_allocator.cc:958): Port[5002a7b0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port ready.
(basic_port_allocator.cc:1069): Port[5002a7b0::0:0:local:Net[eth0:172.26.208.x/20:Ethernet:id=1]]: Port completed gathering candidates.
(basic_port_allocator.cc:1142): All candidates gathered for pooled session.
(GroupInstanceCustomImpl.cpp:2909): 7720.000: setJoinResponsePayload
(p2p_transport_channel.cc:953): P2PTransportChannel: transport, component 0 gathering complete
(p2p_transport_channel.cc:477): Received remote ICE parameters: ufrag=476u1fcsms10j, renomination disabled
(connection.cc:312): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--W|-|0|0|9115038259943047167|-]: Connection created
(p2p_transport_channel.cc:1406): Channel[transport|0|__]: Created connection with origin: 2, total: 1
(p2p_transport_channel.cc:1860): Channel[transport|0|__]: Transport channel state changed from 0 to 2
(p2p_transport_channel.cc:1640): Channel[transport|0|__]: Have a pingable connection for the first time; starting to ping.
(dtls_transport.cc:367): DtlsTransport[transport|0|__]: DTLS setup complete.
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--W|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=424c46765077395435354a6d, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=577372447368695159327535, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=674c4a48682b595367766d6b, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=384250624d2f46504c754f41, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=3339454b527837526b507063, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=727156542f624f676b784a58, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=76686f634f5349586d484469, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=39724a454872586b57424c55, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=4f4644743254556e68666a37, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=4e773462536d48416b514f50, use_candidate=0, nomination=0
(connection.cc:1167): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Sent STUN BINDING request, id=6d644d7a6d5a5572674f654e, use_candidate=0, nomination=0
(connection.cc:1082): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|C--I|-|0|0|9115038259943047167|-]: Received STUN BINDING response, id=424c46765077395435354a6d, code=0, rtt=502, pings_since_last_response=424c46765077395435354a6d 577372447368695159327535 674c4a48682b595367766d6b 384250624d2f46504c754f41 3339454b527837526b507063 ... 6 more
(connection.cc:1305): Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:60PVSd87:0:0:local:udp:172.26.223.x:49257->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|CRWS|-|0|0|9115038259943047167|502]: Updating local candidate type to prflx.
(p2p_transport_channel.cc:277): Switching selected connection due to: candidate pair state changed
(p2p_transport_channel.cc:1754): Channel[transport|0|__]: New selected connection: Conn[5002ce60:transport:Net[eth0:172.26.208.x/20:Ethernet:id=1]:DDQETqLm:0:0:prflx:udp:86.124.125.x:50855->jJM/wLP7:1:2130706431:host:udp:91.108.9.x:32002|CRWS|S|0|0|7962116755336200191|502]
(channel.cc:373): Network route changed for {mid: audio1, media_type: audio}
(rtp_transport_controller_send.cc:301): Network route changed on transport transport: new_route = [ connected: 1 local: [ 1/1 Ethernet turn: 0 ] remote: [ 1/0 Ethernet turn: 0 ] packet_overhead_bytes: 28 ]
(dtls_transport.cc:818): DtlsTransport[transport|0|__]: configuring DTLS handshake timeout 1004 based on ICE RTT 502
(dtls_transport.cc:723): DtlsTransport[transport|0|__]: DtlsTransport: Started DTLS handshake
(srtp_transport.cc:365): The params in SRTP transport are reset.
(dtls_transport.cc:651): DtlsTransport[transport|0|__]: DTLS handshake complete.
(dtls_srtp_transport.cc:218): Extracting keys from transport: transport
(call.cc:1228): UpdateAggregateNetworkState: aggregate_state change to up
(rtp_transport_controller_send.cc:603): Creating fallback congestion controller
(alr_experiment.cc:79): Using ALR experiment settings: pacing factor: 1, max pacer queue length: 2875, ALR bandwidth usage percent: 80, ALR start budget level percent: 40, ALR end budget level percent: -60, ALR experiment group ID: 3
(trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network state predictor
(trendline_estimator.cc:185): Using Trendline filter for delay change estimation with settings sort:false,cap:false,beginning_packets:7,end_packets:7,cap_uncertainty:0,window_size:20 and no network state predictor
(aimd_rate_control.cc:113): Using aimd rate control with back off factor 0.85
(delay_based_bwe.cc:104): Initialized DelayBasedBwe with separate audio overuse detectionenabled:false,packet_threshold:10,time_threshold:1 s and alr limited backoff disabled
(delay_based_bwe.cc:357): BWE Setting start bitrate to: 32 kbps
(bitrate_allocator.cc:394): Current BWE 32000
(bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (32000:60:5)
(bitrate_prober.cc:114): Probe cluster (bitrate:min bytes:min packets): (32000:60:5)
(srtp_transport.cc:310): SRTP activated with negotiated parameters: send cipher_suite 7 recv cipher_suite 7
(channel.cc:575): Channel writable ({mid: audio1, media_type: audio}) for the first time
(channel.cc:902): Changing voice state, recv=1 send=0 for {mid: audio1, media_type: audio}
(sctp_transport.cc:308): InitializeUsrSctp
(GroupInstanceCustomImpl.cpp:2384): 7721.517: setIsRtcConnected: 1
(SctpDataChannelProviderInterfaceImpl.cpp:58): Outgoing DataChannel message: {"colibriClass": "ReceiverVideoConstraints", "constraints": {}, "defaultConstraints": {"maxHeight": 0}, "onStageEndpoints": []}
(webrtc_voice_engine.cc:1619): Setting voice channel options: AudioOptions {aec: 0, agc: 0, ns: 0, hf: 0, typing: 0, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 0, }
(webrtc_voice_engine.cc:387): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 0, agc: 0, ns: 0, hf: 0, typing: 0, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 0, }
(audio_device_impl.cc:868): BuiltInAECIsAvailable
(audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform
(audio_device_impl.cc:871): output: 0
(audio_device_impl.cc:884): BuiltInAGCIsAvailable
(audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform
(audio_device_impl.cc:887): output: 0
(audio_device_impl.cc:900): BuiltInNSIsAvailable
(audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform
(audio_device_impl.cc:903): output: 0
(webrtc_voice_engine.cc:539): Experimental ns is enabled? 0
(webrtc_voice_engine.cc:590): NS set to 0
(webrtc_voice_engine.cc:594): Typing detection is enabled? 0
(audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 0 }, echo_canceller: { enabled: 0, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 0, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 0 }, gain_controller1: { enabled: 0, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255 }, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 0 }, level_estimation: { enabled: 0 }}
(webrtc_voice_engine.cc:1637): Set voice channel options. Current options: AudioOptions {aec: 0, agc: 0, ns: 0, hf: 0, typing: 0, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 0, }
(channel.cc:150): Created channel: {mid: 0, media_type: audio}
(rtp_demuxer.cc:154): Added sink = 48092878 for criteria {mid: 0, rsid: <empty>, ssrcs: [], payload_types = []}
(channel.cc:575): Channel writable ({mid: 0, media_type: audio}) for the first time
(channel.cc:902): Changing voice state, recv=0 send=0 for {mid: 0, media_type: audio}
(channel.cc:911): Setting local voice description for {mid: 0, media_type: audio}
(webrtc_voice_engine.cc:1479): WebRtcVoiceMediaChannel::SetRecvParameters: {codecs: [AudioCodec[111:opus:48000:128:2]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}]}
(webrtc_voice_engine.cc:1647): Setting receive voice codecs.
(webrtc_voice_engine.cc:1934): AddSendStream: {ssrcs:[1012635232];ssrc_groups:;stream_ids:;}
(delay_manager.cc:85): Delay manager config: quantile=0.97 forget_factor=0.9993 start_forget_weight=2 resample_interval_ms=0 max_history_ms=2000
(decision_logic.cc:64): NetEq decision logic settings: estimate_dtx_delay=1 time_stretch_cn=1 target_level_window_ms=100
(neteq_impl.cc:166): NetEq config: sample_rate_hz=16000, enable_post_decode_vad=true, max_packets_in_buffer=200, min_delay_ms=0, enable_fast_accelerate=false, enable_muted_state=false, enable_rtx_handling=false, extra_output_delay_ms=0
(audio_coding_module.cc:224): Created
(audio_send_stream.cc:162): AudioSendStream: 1012635232
(audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1012635232, extmap-allow-mixed: true, extensions: [], c_name: }, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: -1, max_bitrate_bps: -1, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: <unset>}
(audio_send_stream.cc:886): Config is invalid: min_bitrate_bps=-1; max_bitrate_bps=-1; both expected greater or equal to 0
(channel.cc:706): Add send stream ssrc: 1012635232 into {mid: 0, media_type: audio}
(channel.cc:902): Changing voice state, recv=0 send=0 for {mid: 0, media_type: audio}
(channel.cc:978): Setting remote voice description for {mid: 0, media_type: audio}
(webrtc_voice_engine.cc:1433): WebRtcVoiceMediaChannel::SetSendParameters: {codecs: [AudioCodec[111:opus:48000:128:2]], extensions: [{uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}], extmap-allow-mixed: true, max_bandwidth_bps: 1300000, mid: 0, options: AudioOptions {}}
(audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1012635232, extmap-allow-mixed: true, extensions: [], c_name: }, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 6000, max_bitrate_bps: 6000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 510000, minptime: 10, ptime: 120, usedtx: 0, useinbandfec: 0, x-google-max-bitrate: 128, x-google-min-bitrate: 128, x-google-start-bitrate: 128}}}}
(audio_send_stream.cc:886): Config is invalid: min_bitrate_bps=-1; max_bitrate_bps=-1; both expected greater or equal to 0
(webrtc_voice_engine.cc:1857): Recreate all the receive streams because the send codec has changed.
(delay_based_bwe.cc:357): BWE Setting start bitrate to: 32 kbps
(audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1012635232, extmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: }, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 6000, max_bitrate_bps: 6000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 510000, minptime: 10, ptime: 120, usedtx: 0, useinbandfec: 0, x-google-max-bitrate: 128, x-google-min-bitrate: 128, x-google-start-bitrate: 128}}}}
(audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1012635232, mid: 0, extmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: }, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 6000, max_bitrate_bps: 6000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 510000, minptime: 10, ptime: 120, usedtx: 0, useinbandfec: 0, x-google-max-bitrate: 128, x-google-min-bitrate: 128, x-google-start-bitrate: 128}}}}
(webrtc_voice_engine.cc:2341): WebRtcVoiceMediaChannel::SetMaxSendBitrate.
(audio_send_stream.cc:230): AudioSendStream::ConfigureStream: {rtp: {ssrc: 1012635232, mid: 0, extmap-allow-mixed: true, extensions: [{uri: http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01, id: 3}, {uri: http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}, {uri: urn:ietf:params:rtp-hdrext:ssrc-audio-level, id: 1}], c_name: }, rtcp_report_interval_ms: 5000, send_transport: (Transport), min_bitrate_bps: 6000, max_bitrate_bps: 6000, has audio_network_adaptor_config: false, has_dscp: false, send_codec_spec: {nack_enabled: false, transport_cc_enabled: true, cng_payload_type: <unset>, red_payload_type: <unset>, payload_type: 111, format: {name: opus, clockrate_hz: 48000, num_channels: 2, parameters: {maxaveragebitrate: 510000, minptime: 10, ptime: 120, usedtx: 0, useinbandfec: 0, x-google-max-bitrate: 128, x-google-min-bitrate: 128, x-google-start-bitrate: 128}}}}
(webrtc_voice_engine.cc:1619): Setting voice channel options: AudioOptions {}
(webrtc_voice_engine.cc:387): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 0, agc: 0, ns: 0, hf: 0, typing: 0, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 0, }
(audio_device_impl.cc:868): BuiltInAECIsAvailable
(audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform
(audio_device_impl.cc:871): output: 0
(audio_device_impl.cc:884): BuiltInAGCIsAvailable
(audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform
(audio_device_impl.cc:887): output: 0
(audio_device_impl.cc:900): BuiltInNSIsAvailable
(audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform
(audio_device_impl.cc:903): output: 0
(webrtc_voice_engine.cc:539): Experimental ns is enabled? 0
(webrtc_voice_engine.cc:590): NS set to 0
(webrtc_voice_engine.cc:594): Typing detection is enabled? 0
(audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 0 }, echo_canceller: { enabled: 0, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 0, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 0 }, gain_controller1: { enabled: 0, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255 }, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 0 }, level_estimation: { enabled: 0 }}
(webrtc_voice_engine.cc:1637): Set voice channel options. Current options: AudioOptions {aec: 0, agc: 0, ns: 0, hf: 0, typing: 0, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 0, }
(channel.cc:902): Changing voice state, recv=0 send=0 for {mid: 0, media_type: audio}
(webrtc_voice_engine.cc:2064): ResetUnsignaledRecvStream.
(channel.cc:547): Channel enabled: {mid: 0, media_type: audio}
(webrtc_voice_engine.cc:387): WebRtcVoiceEngine::ApplyOptions: AudioOptions {aec: 0, agc: 0, ns: 0, hf: 0, typing: 0, experimental_agc: 0, experimental_ns: 0, residual_echo_detector: 0, }
(audio_device_impl.cc:868): BuiltInAECIsAvailable
(audio_device_generic.cc:18): BuiltInAECIsAvailable: Not supported on this platform
(audio_device_impl.cc:871): output: 0
(audio_device_impl.cc:884): BuiltInAGCIsAvailable
(audio_device_generic.cc:28): BuiltInAGCIsAvailable: Not supported on this platform
(audio_device_impl.cc:887): output: 0
(audio_device_impl.cc:900): BuiltInNSIsAvailable
(audio_device_generic.cc:38): BuiltInNSIsAvailable: Not supported on this platform
(audio_device_impl.cc:903): output: 0
(webrtc_voice_engine.cc:539): Experimental ns is enabled? 0
(webrtc_voice_engine.cc:590): NS set to 0
(webrtc_voice_engine.cc:594): Typing detection is enabled? 0
(audio_processing_impl.cc:533): AudioProcessing::ApplyConfig: AudioProcessing::Config{ pipeline: { maximum_internal_processing_rate: 48000, multi_channel_render: 0, multi_channel_capture: 0 }, pre_amplifier: { enabled: 0, fixed_gain_factor: 1 },capture_level_adjustment: { enabled: 0, pre_gain_factor: 1, post_gain_factor: 1, analog_mic_gain_emulation: { enabled: 0, initial_level: 255 }}, high_pass_filter: { enabled: 0 }, echo_canceller: { enabled: 0, mobile_mode: 0, enforce_high_pass_filtering: 1 }, noise_suppression: { enabled: 0, level: High }, transient_suppression: { enabled: 0 }, voice_detection: { enabled: 0 }, gain_controller1: { enabled: 0, mode: AdaptiveAnalog, target_level_dbfs: 3, compression_gain_db: 9, enable_limiter: 1, analog_level_minimum: 0, analog_level_maximum: 255 }, gain_controller2: { enabled: 0, fixed_digital: { gain_db: 0 }, adaptive_digital: { enabled: 0, noise_estimator: NoiseFloor, vad_reset_period_ms: 1500, adjacent_speech_frames_threshold: 12, max_gain_change_db_per_second: 3, max_output_noise_level_dbfs: -50, sse2_allowed: 1, avx2_allowed: 1, neon_allowed: 1}}, residual_echo_detector: { enabled: 0 }, level_estimation: { enabled: 0 }}
(audio_device_impl.cc:783): RecordingIsInitialized
(audio_device_impl.cc:845): Recording
(audio_device_impl.cc:764): InitRecording
(audio_device_impl.cc:783): RecordingIsInitialized
(audio_device_buffer.cc:180): SetRecordingSampleRate(48000)
(audio_device_buffer.cc:200): SetRecordingChannels(2)
(audio_device_impl.cc:770): output: 0
(channel.cc:902): Changing voice state, recv=0 send=1 for {mid: 0, media_type: audio}
(bitrate_allocator.cc:523): UpdateAllocationLimits : total_requested_min_bitrate: 33866 bps, total_requested_padding_bitrate: 0 bps, total_requested_max_bitrate: 2049866 bps
(audio_device_impl.cc:845): Recording
(goog_cc_network_control.cc:346): max bitrate smaller than min bitrate
(goog_cc_network_control.cc:350): start bitrate smaller than min bitrate
(audio_device_impl.cc:764): InitRecording
(audio_device_impl.cc:783): RecordingIsInitialized
(audio_device_impl.cc:820): StartRecording
(audio_device_impl.cc:845): Recording
(RawAudioDevice.cpp:243): Started recording from Python
(audio_device_impl.cc:827): output: 0
(audio_device_buffer.cc:238): Size of recording buffer: 960
munmap_chunk(): invalid pointer
(goog_cc_network_control.cc:346): max bitrate smaller than min bitrate
(goog_cc_network_control.cc:350): start bitrate smaller than min bitrate
(delay_based_bwe.cc:357): BWE Setting start bitrate to: 33866 bps
(goog_cc_network_control.cc:346): max bitrate smaller than min bitrate
(goog_cc_network_control.cc:350): start bitrate smaller than min bitrate
Aborted