snowboy wake word detection is not working with respeaker 2 mic pi hat.

[Arrow66]

i've been trying to make respeaker detect my wakeword for 2 days .still cant make it out .i thing the problem is with 16bit mono recording all i can hear is a white noise .but stereo recording is working fine no issue at all.tried pyaudio ,arecord . all of them have same problem while mono recording .

can some one help me fix this problem. i can provide any info u guys want

[xiongyihui]

Have you tried to use audacity to find out if it's the case? Maybe one microphone is broken

[Arrow66]

Audacity is not working with mono recording .any way to fix ?

[xiongyihui]

Which ALSA device do you select? The default will support mono recording

[Arrow66]

no its not working . do i need to make any changes to .asounrc file

pcm.!default {
type hw
card 1
format S16_LE
}

ctl.!default {
type hw
card 1
}

``

[Arrow66]

these are my device info

==============================
Default recording device number: 9
Default playback device number: 9

Device ID: 0
Device name: bcm2835 ALSA: - (hw:0,0)
Host name: ALSA
Recording channels: 0
Playback channels: 2
Low Recording Latency: -1.000000
Low Playback Latency: 0.005805
High Recording Latency: -1.000000
High Playback Latency: 0.034830
Supported Rates:
8000
9600
11025
12000
15000
16000
22050
24000
32000
44100
48000

Device ID: 1
Device name: bcm2835 ALSA: IEC958/HDMI (hw:0,1)
Host name: ALSA
Recording channels: 0
Playback channels: 2
Low Recording Latency: -1.000000
Low Playback Latency: 0.005805
High Recording Latency: -1.000000
High Playback Latency: 0.034830
Supported Rates:
44100
48000

Device ID: 2
Device name: seeed-2mic-voicecard: - (hw:1,0)
Host name: ALSA
Recording channels: 2
Playback channels: 2
Low Recording Latency: 0.005805
Low Playback Latency: 0.008707
High Recording Latency: 0.034830
High Playback Latency: 0.034830
Supported Rates:
8000
11025
16000
22050
32000
44100
48000

Device ID: 3
Device name: sysdefault
Host name: ALSA
Recording channels: 0
Playback channels: 128
Low Recording Latency: -1.000000
Low Playback Latency: 0.005805
High Recording Latency: -1.000000
High Playback Latency: 0.034830
Supported Rates:
8000
9600
11025
12000
15000
16000
22050
24000
32000
44100
48000
88200
176400

Device ID: 4
Device name: playback
Host name: ALSA
Recording channels: 0
Playback channels: 128
Low Recording Latency: -1.000000
Low Playback Latency: 0.125000
High Recording Latency: -1.000000
High Playback Latency: 0.125000
Supported Rates:
8000
9600
12000
15000
16000
24000
32000
44100
48000
88200
96000
176400
192000

Device ID: 5
Device name: capture
Host name: ALSA
Recording channels: 128
Playback channels: 0
Low Recording Latency: 0.125000
Low Playback Latency: -1.000000
High Recording Latency: 0.125000
High Playback Latency: -1.000000
Supported Rates:

Device ID: 6
Device name: dmixed
Host name: ALSA
Recording channels: 0
Playback channels: 2
Low Recording Latency: -1.000000
Low Playback Latency: 0.125000
High Recording Latency: -1.000000
High Playback Latency: 0.125000
Supported Rates:
48000

Device ID: 7
Device name: array
Host name: ALSA
Recording channels: 2
Playback channels: 0
Low Recording Latency: 0.125000
Low Playback Latency: -1.000000
High Recording Latency: 0.125000
High Playback Latency: -1.000000
Supported Rates:

Device ID: 8
Device name: dmix
Host name: ALSA
Recording channels: 0
Playback channels: 2
Low Recording Latency: -1.000000
Low Playback Latency: 0.021333
High Recording Latency: -1.000000
High Playback Latency: 0.021333
Supported Rates:
48000

Device ID: 9
Device name: default
Host name: ALSA
Recording channels: 2
Playback channels: 2
Low Recording Latency: 0.005805
Low Playback Latency: 0.008707
High Recording Latency: 0.034830
High Playback Latency: 0.034830
Supported Rates:
8000
11025
16000
22050
32000
44100
48000

Selected recording device: 9 - default
Selected playback device: 9 - default
Supported Rates:
8000
11025
16000
22050
32000
44100
48000

Available mixers:

Available recording sources:

Available playback volumes:
0 - PCM:0

Recording volume is emulated
Playback volume is emulated

[Arrow66]

default recording device can record in stereo with no issue . but in the case of mono something is not right ..
this is the screen of audacity when i'm recording in stereo .take a look at audio meter

this one when recording mono

[Arrow66]

[Arrow66]

can some one please help me fix this issue .i did a fresh install of raspbian stretch today .still the problem exist

[xiongyihui]

Are the volumes of the two microphone the same?
You can use alsamixer -c 1 to set the volumes.

If you want to to restore the default volumes, try sudo alsactl --file=/etc/voicecard/wm8960_asound.state restore

[xiongyihui]

By the way, you can remove your alsa configuration file ~/.asound.rc and just use the default alsa configuration file /etc/asound.conf from the driver

[Arrow66]

@xiongyihui i Tried all of the steps u've mentioned . still the problem exist these are my alsa-infos
alsa - info : http://www.alsa-project.org/db/?f=bfd07e437056c0a9560290340fa9b4c0a5ade7e1

[Arrow66]

as you guys can see audio recorded from left input mic is just too noisy .but audio from right one working perfectly . i can findout this by playing tracks individually .but when i play them together there is no issue audio was perfect .

And i noticed that when i'm recording mono left input mic is activated . is there anyway i can use my right mic when recording mono ?.

like this one mentioned in this thread https://unix.stackexchange.com/questions/254219/alsa-capturing-mono-microphone-as-right-channel-of-stereo

[Arrow66]

@xiongyihui help me bro .

[Pillar1989]

@arjunan47 I think you need buy new one.

[Arrow66]

@Pillar1989 is that the only option I've left . any way to fix ?

[xiongyihui]

@arjunan47 When did you buy the mic hat?

@xyu6

[Arrow66]

my mic hat was just 2 moths old
i'm from india and i bought it from indian reseller https://www.fabtolab.com/respeaker-hat
my mic hat sometimes automatically turn on its led lights .i need to unplug and replug my rp3 to the power . sometime i need to plugin rp3 adapter to different sockets to make the lights turn off .
does this relates to my issue ?
is there any firmware updates available for me
@xiongyihui any help?

[Arrow66]

@xyu6 any help

[Arrow66]

 def get_callback(self):
    def callback(in_data, frame_count, time_info, status):
        sound = AudioSegment(in_data,sample_width=2, channels=1, frame_rate=16000)
        self.wavefile.writeframes(sound.raw_data)
        return in_data, pyaudio.paContinue
    return callback

by this pyaudio callback function i can convert stereo audio to mono in realtime using pydub .but the problem is only 32bit pcm is working . but when i change
wavefile.setsampwidth(4)
wavefile sampwidth to 2 ie 16bit pcm it becomes completely laggy .
any way to fix ?

[xiongyihui]

@arjunan47 Did you want to save audio data to .wav file? You can use wave module, See https://docs.python.org/2/library/wave.html

import wave
wav = wave.open('audio.wav', 'wb')
# ...

PS. As the left channel is broken, before it is fixed, you can recording stereo audio data and select the right channel to use. To get right channel from stereo:

import audioop

stereo = ''  # stereo audio data
width = 2
right = audioop.tomono(stereo, width, 0, 1)

[Arrow66]

@xiongyihui bro is it possible to record audio in realtime using buffers/chunks and extract mono from it?.may be I could use it for wake word detection using snowboy

[Arrow66]

Thanks you 😄guys finally snowboy works

[Arrow66]

i don't what i did wrong , one day it starts to work .nextday it stops .
``
import pyaudio
import wave
import time
import audioop
import snowboydetect
import threading
import Queue
detector = snowboydetect.SnowboyDetect(resource_filename = '/home

            /pi/xxxxxcommon.res'.encode(),model_str ='/home/xxxxx.umdl'.encode())

      class SnowboyRec(threading.Thread):
               def __init__(self,channels = 2,
              rate = 16000, frames_per_buffer = 1024):

            threading.Thread.__init__(self)
            self.channels = channels

            self.rate = rate
            self.frames_per_buffer = frames_per_buffer
            self._pa = pyaudio.PyAudio()

            self._stream = None
   def start_recording(self):
        detector = snowboydetect.SnowboyDetect(resource_filename = 'xxxx'.encode(),model_str ='    
          xxxxxxxx'.encode())
        detector.SetAudioGain(9)
        detector.ApplyFrontend(False)
        detector.SetSensitivity('0.9'.encode())
        print('entered')

        self._stream = self._pa.open(format=pyaudio.paInt16,
                                        channels=self.channels,
                                        rate=self.rate,
                                        input_device_index = 2,
                                        input=True,
                                        frames_per_buffer=self.frames_per_buffer,
                                        stream_callback=self.get_callback())

        print('start')
        self._stream.start_stream()
        return self

def get_callback(self):
        def callback(in_data, frame_count, time_info, status):
            d = audioop.tomono(in_data, 2, 0, 1)
            status = detector.RunDetection(d)
            print(status)
            if status == 1:
                print('keyword spotted')
            return in_data, pyaudio.paContinue
        return callback

if name == 'main':
snow = SnowboyRec()
snow.start_recording()
it works when i speak so loud
`