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__gmon_start___ITM_deregisterTMCloneTable_ITM_registerTMCloneTable__cxa_finalizeg_type_class_peek_parent_gst_debug_register_funcptrgst_element_class_add_static_pad_templategst_element_class_set_static_metadatag_type_class_adjust_private_offsetgst_object_unrefg_freeg_mutex_lockg_type_nameg_mutex_unlockg_value_get_booleang_value_dup_stringg_value_get_enumg_value_get_intg_value_set_booleang_value_set_enumg_value_set_intg_value_set_stringgst_adapter_cleargst_planar_audio_adapter_cleargst_mini_object_make_writablegst_segment_to_running_time_gst_debug_mingst_adapter_pushgst_planar_audio_adapter_pushgst_debug_loggst_planar_audio_adapter_configureg_param_spec_stringg_object_class_install_propertyg_param_spec_booleang_param_spec_enumg_param_spec_intgst_type_mark_as_plugin_apig_enum_register_staticg_intern_static_stringgst_audio_filter_get_typeg_type_register_static_simple_gst_debug_category_newgst_adapter_newgst_planar_audio_adapter_newgst_audio_info_initg_object_refgst_adapter_availablegst_planar_audio_adapter_flushgst_planar_audio_adapter_take_buffergst_audio_meta_api_get_typegst_buffer_get_metagst_buffer_newgst_buffer_copy_regiongst_buffer_appendgst_buffer_new_allocategst_buffer_memsetgst_buffer_add_audio_meta_gst_element_error_printfgst_stream_error_quarkgst_element_message_fullgst_mini_object_replacegst_audio_buffer_mapgst_audio_buffer_unmapgst_planar_audio_adapter_availablegst_adapter_prev_ptsgst_util_uint64_scale_intgst_adapter_flushgst_adapter_take_buffergst_planar_audio_adapter_prev_ptsgst_mini_object_unrefg_return_if_fail_warning__stack_chk_fail_Unwind_Resumegst_buffer_get_audio_level_metagst_segment_to_stream_timegst_structure_newgst_message_new_elementgst_element_post_message_ZN6webrtc20AudioProcessingStatsD1Evgst_buffer_add_audio_level_metagst_adapter_pts_at_discontgst_planar_audio_adapter_pts_at_discontg_strcmp0gst_resource_error_quarkgst_object_ref_ZN6webrtc15AudioProcessing6Config8PipelineC1Ev_ZN6webrtc22AudioProcessingBuilderC1Ev_ZN6webrtc22AudioProcessingBuilder6CreateEv_ZN6webrtc22AudioProcessingBuilderD1Evg_type_class_refg_type_class_unrefg_enum_get_valueg_list_removegst_event_parse_latencygst_query_new_latencygst_pad_querygst_query_parse_latencygst_mini_object_copyg_mutex_initg_list_prependgst_element_registerg_once_init_enter_pointerg_once_init_leave_pointergst_plugin_webrtcdsp_get_descgst_plugin_webrtcdsp_registergst_plugin_register_static__gxx_personality_v0libgstbadaudio-1.0.so.0libgstbase-1.0.so.0libgstreamer-1.0.so.0libgobject-2.0.so.0libglib-2.0.so.0libgstaudio-1.0.so.0libwebrtc-audio-processing-1.so.3libstdc++.so.6libgcc_s.so.1libc.so.6libgstwebrtcdsp.soGCC_3.0CXXABI_1.3GLIBC_ABI_DT_RELRGLIBC_2.4GLIBC_2.2.5$ORIGIN/../../gst-libs/gst/audio  P&y  y  ӯk  B ii  ui  ؟"^efXȜМ؜      (08@HPX`hpx !#$ȝ%Н&؝'()*+,-./ 0(10283@4H5P6X7`8h9p:x;<=>?@ABCDȞEОF؞GHIJKLMNO P(Q0R8S@THUPVXW`YhZp[x\]_`abcdghȟiПjpm6؆ unknown errornot enabledunspecified errorcreating failedunsupported componentunsupported functionnull pointerbad parameterbad sample ratebad data lengthbad number of channelsfile IO errorstream parameter not setgst_webrtc_echo_probe_stopgst_webrtc_echo_probe_setupGeneric/AudioAcoustic Echo Canceller probegst_webrtc_dsp_finalizegst_webrtc_dsp_set_propertygst_webrtc_dsp_get_propertygst_webrtc_dsp_startgst_webrtc_dsp_stopgst_webrtc_dsp_setupwebrtcechoprobe0Echo ProbeHigh Pass Filterhigh-pass-filterEcho Cancelecho-cancelGstWebrtcEchoSuppressionLevelecho-suppression-levelnoise-suppressionNoise Suppression Levelnoise-suppression-levelGain Controlgain-controlexperimental-agcExtended Filterextended-filterdelay-agnosticTarget Level dBFStarget-level-dbfsCompression Gain dBcompression-gain-dbStartup Minimum Volumestartup-min-volumeLimiterlimiterGstWebrtcGainControlModeGain Control Modegain-control-modeVoice Detectionvoice-detectionvoice-detection-frame-size-msvoice-detection-likelihoodGstWebrtcDspwebrtcdspGstWebrtcEchoProbebuf != NULLerror: %snow hasno longer hasgst_webrtc_dsp_process_streamstream-has-voicestream-timevoice-activitygst_webrtc_vad_post_activityNOT Enabling High Pass filterEnabling Echo CancellationEnabling Noise Suppressionwebrtcechoprobe1.26.11gst-plugins-badLGPL2026-03-10sinksrcAdaptive Digitaladaptive-digitalFixed Digitalfixed-digitalAdaptive Analogadaptive-analogLow SuppressionModerate SuppressionmoderateVery High Suppressionvery-highhigh SuppressionVery Low Likelihoodvery-lowModerate LikelihoodHigh Likelihoodșؙ(8HXLМLLL0H`xhLLԞ\t̝,Dgst_webrtc_echo_probe_src_eventgst_webrtc_echo_probe_transform_ipNicolas Dufresne Gathers playback buffers for webrtcdsp../ext/webrtcdsp/gstwebrtcdsp.cpp%s:%d: invalid %s id %u for "%s" of type '%s' in '%s'gst_webrtc_dsp_submit_input_bufferReceived discont, clearing adapter.../ext/webrtcdsp/gstwebrtcechoprobe.cppsetting format to %s with %i Hz and %i channelswebrtcdsp format produce too big period (maximum is %d samples and we have %u samples), reduce the number of channels or the rate.gst_webrtc_dsp_generate_outputNicolas Dufresne Pre-processes voice with WebRTC Audio Processing LibraryVoice Processor (AGC, AEC, filters, etc.)The name of the webrtcechoprobe element that record the audio being played through loud speakers. Must be set before PAUSED state.Enable or disable high pass filteringEnable or disable echo cancellerControls the aggressiveness of the suppressor. A higher level trades off double-talk performance for increased echo suppression.Echo Suppression Level (does nothing)Enable or disable noise suppressionGstWebrtcNoiseSuppressionLevelControls the aggressiveness of the suppression. Increasing the level will reduce the noise level at the expense of a higher speech distortion.Enable or disable automatic digital gain controlEnable or disable experimental automatic gain control.Experimental AGC (does nothing)Enable or disable the extended filter.Enable or disable the delay agnostic mode.Delay agnostic mode (does nothing)Sets the target peak |level| (or envelope) of the gain control in dBFS (decibels from digital full-scale).Sets the maximum |gain| the digital compression stage may apply, in dB.At startup the experimental AGC moves the microphone volume up to |startup_min_volume| if the current microphone volume is set too low. No effect if experimental-agc isn't enabled.When enabled, the compression stage will hard limit the signal to the target level. Otherwise, the signal will be compressed but not limited above the target level.Controls the mode of the compression stageEnable or disable the voice activity detectorSets the |size| of the frames in ms on which the VAD will operate. Larger frames will improve detection accuracy, but reduce the frequency of updatesVoice detection frame size in milliseconds (does nothing)GstWebrtcVoiceDetectionLikelihoodSpecifies the likelihood that a frame will be declared to contain voice.Voice detection likelihood (does nothing)libwebrtcdsp wrapping elementsGstFlowReturn gst_webrtc_dsp_analyze_reverse_stream(GstWebrtcDsp*, GstClockTime)Echo Probe has rate %i , while the DSP is running at rate %i, use a caps filter to ensure those are the same.gst_webrtc_dsp_analyze_reverse_streamReverse stream analyses failed: %s.Failed to filter the audio: %s.Posting voice activity message, stream %s voiceNo echo probe with name %s found.Enabling Voice Activity DetectionEnabling Digital Gain Control, target level dBFS %d, compression gain dB %d, limiter %senabled, mode: %sWe have a latency of %u:%02u:%02u.%09u and delay of %imsVoice pre-processing using WebRTC Audio Processing Libraryhttp://download.fedoraproject.orgFedora GStreamer-plugins-bad packageaudio/x-raw, format = (string) S16LE, layout = (string) interleaved, rate = (int) { 48000, 32000, 16000, 8000 }, channels = (int) [1, MAX];audio/x-raw, format = (string) F32LE, layout = (string) non-interleaved, rate = (int) { 48000, 32000, 16000, 8000 }, channels = (int) [1, MAX] FA@@@H;{X@͈X؊lX8H8H\Xh H,l@tخ8 Ht(zRx $z`FJ w?9*3$"DP\ pd$EC A ` pIEC A~ ,EC BEEID K ,PEC BEEEHJ E $0EC E  <HEC C| ,`EC BFx H [ E (ďEC BEH K (8jEC A N | L $|AJ EN B d MAJ E| 09EC Ak zPLRx H$ {AC DHK. 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