Assist Microphone does not allow user to check audio quality and samples (gain, microphone multiplier) to get optimal results from Wakeword or Whisper TTS #3830
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I second this, even if storing the requests is possible a privacy concern, it should be possible to get some stats like true peak dB, loudness and apparent noise floor. Also, it's tough to determine if the config is being written to or read at all, so knowing if any audio is making it's way through is a requirement to debugging. |
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Describe the issue you are experiencing
Whilst there are clear options for gain and microphone multiplier - there seems to be no easy way to capture some of the audio AFTER these parameters are applied to see how well the microphone is performing and whether it is distorting, too quiet or other problems may be occurring.
So getting good quality seem to be a case of juggling gain and microphone volume multiplier, turning on logging in Whisper to see what transcribed text comes through and guessing whether it's too soft, too muffled, or distorted etc.
What type of installation are you running?
Home Assistant OS
Which operating system are you running on?
Home Assistant Operating System
Which add-on are you reporting an issue with?
Whisper
What is the version of the add-on?
1.2.0
Steps to reproduce the issue
...
System Health information
System Information
Home Assistant Community Store
Home Assistant Cloud
Home Assistant Supervisor
Dashboards
Recorder
Anything in the Supervisor logs that might be useful for us?
Anything in the add-on logs that might be useful for us?
s6-rc: info: service s6rc-oneshot-runner: starting s6-rc: info: service s6rc-oneshot-runner successfully started s6-rc: info: service fix-attrs: starting s6-rc: info: service fix-attrs successfully started s6-rc: info: service legacy-cont-init: starting s6-rc: info: service legacy-cont-init successfully started s6-rc: info: service assist_microphone: starting s6-rc: info: service assist_microphone successfully started s6-rc: info: service discovery: starting DEBUG:root:Namespace(mic_uri=None, mic_command='arecord -r 16000 -c 1 -f S16_LE -t raw', mic_command_rate=16000, mic_command_width=2, mic_command_channels=1, mic_command_samples_per_chunk=1024, mic_volume_multiplier=32.0, mic_noise_suppression=0, mic_auto_gain=12, mic_seconds_to_mute_after_awake_wav=0.5, mic_no_mute_during_awake_wav=False, mic_channel_index=None, snd_uri=None, snd_command='aplay -r 16000 -c 1 -f S16_LE -t raw', snd_command_rate=16000, snd_command_width=2, snd_command_channels=1, snd_volume_multiplier=2.0, wake_uri=None, wake_word_name=[], wake_command=None, wake_command_rate=16000, wake_command_width=2, wake_command_channels=1, wake_refractory_seconds=5.0, vad=False, vad_threshold=0.5, vad_trigger_level=1, vad_buffer_seconds=2, vad_wake_word_timeout=5.0, event_uri=None, startup_command=None, detect_command=None, detection_command=None, transcript_command=None, stt_start_command=None, stt_stop_command=None, synthesize_command=None, tts_start_command=None, tts_stop_command=None, tts_played_command=None, streaming_start_command=None, streaming_stop_command=None, error_command=None, connected_command=None, disconnected_command=None, awake_wav='/usr/src/sounds/awake.wav', done_wav='/usr/src/sounds/done.wav', uri='tcp://0.0.0.0:10700', name='assist microphone', area=None, no_zeroconf=True, zeroconf_name=None, zeroconf_host=None, debug_recording_dir=None, debug=True, log_format='%(levelname)s:%(name)s:%(message)s') INFO:root:Ready DEBUG:root:Connecting to mic service: ['arecord', '-r', '16000', '-c', '1', '-f', 'S16_LE', '-t', 'raw'] DEBUG:root:Connecting to snd service: ['aplay', '-r', '16000', '-c', '1', '-f', 'S16_LE', '-t', 'raw'] INFO:root:Connected to services DEBUG:root:Using webrtc audio enhancements DEBUG:root:Connected to mic service Recording raw data 'stdin' : Signed 16 bit Little Endian, Rate 16000 Hz, Mono [15:45:26] INFO: Successfully sent discovery information to Home Assistant. s6-rc: info: service discovery successfully started s6-rc: info: service legacy-services: starting s6-rc: info: service legacy-services successfully started DEBUG:root:Server set: 12457378106121 INFO:root:Connected to server INFO:root:Streaming audio DEBUG:root:Event(type='run-pipeline', data={'start_stage': 'wake', 'end_stage': 'tts', 'restart_on_end': True, 'snd_format': {'rate': 16000, 'width': 2, 'channels': 1}}, payload=None) DEBUG:root:Ping enabledAdditional information
After many hours, I chose:
Noise suppression: 0
Audio gain: 12
Microphone volume multipler: 32 (yes this high!)
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