Current File : //lib64/gstreamer-1.0/libgstwebrtcdsp.so
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IIx LIHIt qIx Jt= x I|$ 1l[ Q HH success unspecified error creating failed unsupported component unsupported function null pointer bad parameter bad sample rate bad data length bad number of channels file IO error stream parameter not set not enabled unkown error gstwebrtcdsp.cpp error: %s now has no longer has stream-has-voice stream-time voice-activity gst_webrtc_dsp_finalize gst_webrtc_dsp_set_property gst_webrtc_dsp_get_property gst_webrtc_dsp_start gst_webrtc_dsp_stop gst_webrtc_dsp_setup Generic/Audio webrtcechoprobe0 Echo Probe High Pass Filter high-pass-filter Echo Cancel echo-cancel GstWebrtcEchoSuppressionLevel Echo Suppression Level echo-suppression-level noise-suppression Noise Suppression Level noise-suppression-level Gain Control gain-control Experimental AGC experimental-agc Extended Filter extended-filter Delay Agnostic delay-agnostic Target Level dBFS target-level-dbfs Compression Gain dB compression-gain-db Startup Minimum Volume startup-min-volume Limiter limiter GstWebrtcGainControlMode Gain Control Mode gain-control-mode Voice Detection voice-detection voice-detection-frame-size-ms Voice Detection Likelihood voice-detection-likelihood NOT Enabling High Pass filter Enabling Echo Cancellation Enabling Noise Suppression GstWebrtcDsp webrtcdsp webrtcechoprobe 1.16.1 http://www.redhat.com GStreamer-plugins-bad package gst-plugins-bad LGPL 2019-09-23 Very Low Likelihood very-low Moderate Likelihood moderate High Likelihood Adaptive Digital adaptive-digital Fixed Digital fixed-digital Low Suppression Moderate Suppression Very High Suppression very-high high Suppression src sink 0@P`p,D\t 4Ld| (@XpX gst_webrtc_dsp_start gst_webrtc_dsp_submit_input_buffer gst_webrtc_vad_post_message gst_webrtc_dsp_process_stream gst_webrtc_dsp_analyze_reverse_stream gst_webrtc_dsp_setup Echo Probe has rate %i , while the DSP is running at rate %i, use a caps filter to ensure those are the same. Reverse stream analyses failed: %s. Failed to filter the audio: %s. Posting voice activity message, stream %s voice Received discont, clearing adapter. %s:%d: invalid %s id %u for "%s" of type '%s' in '%s' gst_webrtc_dsp_submit_input_buffer gst_webrtc_dsp_generate_output Nicolas Dufresne <nicolas.dufresne@collabora.com> Pre-processes voice with WebRTC Audio Processing Library Voice 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 filtering Enable or disable echo canceller Controls the aggressiveness of the suppressor. A higher level trades off double-talk performance for increased echo suppression. Enable or disable noise suppression GstWebrtcNoiseSuppressionLevel Controls 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 control Enable or disable experimental automatic gain control. Enable or disable the extended filter. Enable or disable the delay agnostic mode. 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 stage Enable or disable the voice activity detector Sets the |size| of the frames in ms on which the VAD will operate. Larger frames will improve detection accuracy, but reduce the frequency of updates Voice Detection Frame Size Milliseconds GstWebrtcVoiceDetectionLikelihood Specifies the likelihood that a frame will be declared to contain voice. setting format to %s with %i Hz and %i channels Failed to initialize WebRTC Audio Processing library webrtc::AudioProcessing::Initialize() failed: %s Enabling Digital Gain Control, target level dBFS %d, compression gain dB %d, limiter %senabled, mode: %s Enabling Voice Activity Detection, frame size %d milliseconds, likelihood: %s webrtcdsp format produce too big period (maximum is %lu samples and we have %u samples), reduce the number of channels or the rate. libwebrtcdsp wrapping elements Voice pre-processing using WebRTC Audio Processing Library No echo probe with name %s found. audio/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] N6webrtc6Config10BaseOptionE N6webrtc10AudioFrameE N6webrtc6Config6OptionINS_14ExtendedFilterEEE N6webrtc6Config6OptionINS_15ExperimentalAgcEEE N6webrtc6Config6OptionINS_13DelayAgnosticEEE gst_webrtc_echo_probe_src_event gst_webrtc_echo_probe_transform_ip Nicolas Dufresne <nicolas.dufrsesne@collabora.com> Gathers playback buffers for webrtcdsp Accoustic Echo Canceller probe We have a latency of %u:%02u:%02u.%09u and delay of %ims gst_webrtc_echo_probe_stop gst_webrtc_echo_probe_setup gstwebrtcechoprobe.cpp GstWebrtcEchoProbe gst_webrtc_echo_probe_src_event gst_webrtc_echo_probe_setup ;` + | t TP D t tP T t 4 ( D $X Tt t X $ D 8 X d $
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