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Triggering code from sound input: how to?

Started by dicky96, January 07, 2013, 02:52:06 AM

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dedndave


Siekmanski

Hi Dave

I'm still alive  :t
Little free time for programming  :(
Creative coders use backward thinking techniques as a strategy.

Gunther

Hi Marinus,

good to see you again. I hope you're doing well.

Gunther
You have to know the facts before you can distort them.

Donkey

Been playing around with getting sound from the PC microphone and though I am getting data I have no idea at all how to interpret it, here's the code:

DATA SECTION
caps WAVEINCAPSA <>
pBuffer PTR ?

CODE SECTION

START:
invoke GetMicrophone,0
test eax,eax
js >
invoke GetMicInput,eax,0
:
invoke ExitProcess,0

GetMicrophone FRAME Shiss
uses ebx,edi,esi
invoke waveInGetNumDevs
mov ebx,eax
xor esi,esi
mov edi,-1
:
invoke waveInGetDevCaps,esi,offset caps,SIZEOF WAVEINCAPSA
mov eax,offset caps.szPname + 10
mov B[eax],0
invoke lstrcmpi,offset caps.szPname,"microphone"
test eax,eax
jnz >
mov edi,esi
jmp >>.EXIT
:
inc esi
cmp esi,ebx
jl <
.EXIT
mov eax,edi
ret
endf

GetMicInput FRAME uID,pSound
uses edi,esi,ebx
LOCAL hWAV:%HANDLE
LOCAL wfx:WAVEFORMATEX
LOCAL whdr:WAVEHDR

mov D[hWAV],-1

mov W[wfx.wFormatTag],WAVE_FORMAT_PCM
mov W[wfx.nChannels],2
mov D[wfx.nSamplesPerSec],8000
mov W[wfx.nBlockAlign],2
mov D[wfx.nAvgBytesPerSec],16000
mov W[wfx.wBitsPerSample],8
mov W[wfx.cbSize],0

invoke waveInOpen,offset hWAV,[uID],offset wfx,offset MonitorMicrophone,0,CALLBACK_FUNCTION

// Create a buffer (8MB)

invoke GlobalAlloc,GMEM_FIXED | GMEM_ZEROINIT,8*1024*1024
mov [pBuffer],eax

mov [whdr.lpData],eax
mov D[whdr.dwBufferLength],8*1024*1024
mov D[whdr.dwBytesRecorded],0
mov D[whdr.dwUser],0
mov D[whdr.dwFlags],0
mov D[whdr.dwLoops],0
mov D[whdr.lpNext],0
mov D[whdr.reserved],0

invoke waveInPrepareHeader,[hWAV],offset whdr,SIZEOF WAVEHDR
invoke waveInAddBuffer, [hWAV],offset whdr,SIZEOF WAVEHDR

invoke waveInStart,[hWAV]

invoke Sleep,1000

invoke waveInStop,[hWAV]

invoke waveInClose,[hWAV]

invoke GlobalFree,[pBuffer]

ret
endf

MonitorMicrophone FRAME hwi, msg, dwInstance, dwParam1, dwParam2

cmp D[msg],WIM_CLOSE
jne >
ret

:
cmp D[msg],WIM_DATA
jne >>
mov eax,[dwParam1]
mov ecx,[eax+WAVEHDR.dwBytesRecorded]
mov eax,[eax+WAVEHDR.lpData]

ret

:
cmp D[msg],WIM_OPEN
jne >
ret

:
ret
endf


Everything returns without error and I have 15830 bytes recorded, there is definitely data written to the buffer but I don't know enough about WAV data to know how to use or interpret the data.

Edgar
"Ahhh, what an awful dream. Ones and zeroes everywhere...[shudder] and I thought I saw a two." -- Bender
"It was just a dream, Bender. There's no such thing as two". -- Fry
-- Futurama

Donkey's Stable

dedndave

you could probably store the data as a wav file and play it back with wmp to test it

Donkey

Quote from: dedndave on January 08, 2013, 02:10:01 PM
you could probably store the data as a wav file and play it back with wmp to test it

There doesn't seem to be a RIFF header, only raw data. I may try to prepend a RIFF header to the data as well as the fmt and data chunk information and try to play it using PlaySound with SND_MEMORY | SND_NODEFAULT. But that's for another day.
"Ahhh, what an awful dream. Ones and zeroes everywhere...[shudder] and I thought I saw a two." -- Bender
"It was just a dream, Bender. There's no such thing as two". -- Fry
-- Futurama

Donkey's Stable

Donkey

Ok, this bothered me too much to leave it till tomorrow. It was the fact that there was no RIFF header. The following code builds the header and plays back what the microphone recorded.

The RIFF header takes the format:



DATA SECTION
caps WAVEINCAPSA <>
pBuffer PTR ?

CODE SECTION

START:
invoke GetMicrophone,0
test eax,eax
js >
invoke GetMicInput,eax,0
:
invoke ExitProcess,0

GetMicrophone FRAME Shiss
uses ebx,edi,esi
invoke waveInGetNumDevs
mov ebx,eax
xor esi,esi
mov edi,-1
:
invoke waveInGetDevCaps,esi,offset caps,SIZEOF WAVEINCAPSA
mov eax,offset caps.szPname + 10
mov B[eax],0
invoke lstrcmpi,offset caps.szPname,"microphone"
test eax,eax
jnz >
mov edi,esi
jmp >>.EXIT
:
inc esi
cmp esi,ebx
jl <
.EXIT
mov eax,edi
ret
endf

GetMicInput FRAME uID,pSound
uses edi,esi,ebx
LOCAL hWAV:%HANDLE
LOCAL wfx:WAVEFORMATEX
LOCAL whdr:WAVEHDR

mov D[hWAV],-1

mov W[wfx.wFormatTag],WAVE_FORMAT_PCM
mov W[wfx.nChannels],2
mov D[wfx.nSamplesPerSec],8000
mov W[wfx.nBlockAlign],2
mov D[wfx.nAvgBytesPerSec],16000
mov W[wfx.wBitsPerSample],8
mov W[wfx.cbSize],0

invoke waveInOpen,offset hWAV,[uID],offset wfx,offset MonitorMicrophone,0,CALLBACK_FUNCTION

// Create a buffer (8MB) + RIFF header

invoke GlobalAlloc,GMEM_FIXED | GMEM_ZEROINIT,8*1024*1024 + 44
mov [pBuffer],eax

add eax,44 // leave space for the WAV header
mov [whdr.lpData],eax

mov D[whdr.dwBufferLength],8*1024*1024
mov D[whdr.dwBytesRecorded],0
mov D[whdr.dwUser],0
mov D[whdr.dwFlags],0
mov D[whdr.dwLoops],0
mov D[whdr.lpNext],0
mov D[whdr.reserved],0

invoke waveInPrepareHeader,[hWAV],offset whdr,SIZEOF WAVEHDR
invoke waveInAddBuffer, [hWAV],offset whdr,SIZEOF WAVEHDR


invoke waveInStart,[hWAV]

invoke Sleep,1000

invoke waveInStop,[hWAV]

invoke waveInClose,[hWAV]

invoke GlobalFree,[pBuffer]

ret
endf

MonitorMicrophone FRAME hwi, msg, dwInstance, dwParam1, dwParam2

cmp D[msg],WIM_CLOSE
jne >
ret

:
cmp D[msg],WIM_DATA
jne >>
mov eax,[dwParam1]
mov ecx,[eax+WAVEHDR.dwBytesRecorded]
mov eax,[eax+WAVEHDR.lpData]

invoke BuildWAVFormat,[pBuffer],ecx

ret

:
cmp D[msg],WIM_OPEN
jne >
ret

:
ret
endf

BuildWAVFormat FRAME pData,dwDataSize
uses ebx,esi,edi

mov ebx,[pData]
mov D[ebx],"RIFF"
mov eax,[dwDataSize]
add eax,36
mov [ebx+4],eax
mov D[ebx+8],"WAVE"
mov D[ebx+12],"fmt "
mov W[ebx+16],16
mov W[ebx+20],1
mov W[ebx+22],2
mov D[ebx+24],8000
mov D[ebx+28],16000
mov W[ebx+32],2
mov W[ebx+34],8
mov D[ebx+36],"data"
mov eax,[dwDataSize]
mov [ebx+40],eax

invoke PlaySound,[pData],NULL,SND_MEMORY | SND_NODEFAULT

ret
endf


Just to let you know, I screwed around with this for 20 minutes until I figured out I had my mic muted :)
"Ahhh, what an awful dream. Ones and zeroes everywhere...[shudder] and I thought I saw a two." -- Bender
"It was just a dream, Bender. There's no such thing as two". -- Fry
-- Futurama

Donkey's Stable

dedndave

very cool, Edgar   :t

the microphone thing - not hard to do - lol

Donkey

Quote from: dedndave on January 08, 2013, 03:51:45 PM
very cool, Edgar   :t

the microphone thing - not hard to do - lol

Thanks Dave,

With a bit of work it can easily be turned into an application that will record data from a microphone and save it to a WAV file, a neat application if anyone has a use for it. Also I'm trying to figure out how to parse the raw WAV data to do an FFT on it so I can spot specific frequencies but I'm not really that good with that sort of stuff. I figure it would pretty much fit the bill for the thread subject if I could isolate a specific frequency and trigger an event based on it.
"Ahhh, what an awful dream. Ones and zeroes everywhere...[shudder] and I thought I saw a two." -- Bender
"It was just a dream, Bender. There's no such thing as two". -- Fry
-- Futurama

Donkey's Stable

dedndave

i don't remember much about it, other than everything in a RIFF file is stored in "chunks"
i think the data is some form of pulse-code modulation
the header seems to outline the parameters

Siekmanski

Hi Donkey

The Goertzel algorithm is perfect and fast to get a specific frequency. 

Goertzel algorithm:

samples      = range -1.0 to 1.0 (floating point)
N            = number of samples to process
Pi           = 3.141592653589793238
frequency    = frequency to look for
samplerate   = sample rate of the data


; Precalculate coeff(s)

       coeff = Cos((Pi * 2.0 / N) * (frequency / samplerate * N + 0.5) * 2.0)

; Processing loop
       q1,q2,i == 0

loop:  q0 = (coeff * q1) - q2 + samples[i]
       q2 = q1
       q1 = q0
       i   = i + 1

      goto loop until i == N

      Magnitude = Sqrt((q1 * q1) + (q2 * q2) - (q1 * q2 * coeff) / N * 2)


Creative coders use backward thinking techniques as a strategy.

Siekmanski

Hi Donkey

16 bit WAV samples are signed data
8 bit WAV samples are unsigned so you have to convert them to signed data

xor eax,eax
mov al,255
xor eax,10000000b

al is now 127
Creative coders use backward thinking techniques as a strategy.

Siekmanski

Some old test pieces with Goertzel in action.

Creative coders use backward thinking techniques as a strategy.

Donkey

Quote from: Siekmanski on January 09, 2013, 01:37:54 AM
Hi Donkey

16 bit WAV samples are signed data
8 bit WAV samples are unsigned so you have to convert them to signed data

xor eax,eax
mov al,255
xor eax,10000000b

al is now 127

Thanks , since I'm building the WAV files I can just build them as 16 bit, easier that way. I'll take a look at Goertzel, thanks for the code, I'm missing enough hair as it is and didn't look forward to pulling any more out  :biggrin:
"Ahhh, what an awful dream. Ones and zeroes everywhere...[shudder] and I thought I saw a two." -- Bender
"It was just a dream, Bender. There's no such thing as two". -- Fry
-- Futurama

Donkey's Stable

dicky96

Actually I have decided to try the external hardware method putting logic 1 onto input pins of the parallel port when there is a beat

This is not to discredit other suggestions, it is more about what  I understand best (hardware) and the time taken for me to develop a solution

If I use windows sleep function to activate my code around every 80mS I doubt that the delay in picking up a beat would be noticable as i am not stepping my display once per beat, i'm just changing the sequence direction/colour/etc on each beat

Also as was suggested, I may as well put win98 on this PC as it is gonna be dedicated to this task (as I mentioned I have plenty of old PC hardware around) - whether I can find a copy of win98 is another matter lol!  Why dont microsoft just give away old obselete OS for free as they have no more use for them?

In the case of win98 can I just read/write directly to the parallel port data/control registers at 378h and 37ah?

And does the sleep function work just the same?

cheers
Rich