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Author Topic: HOW do i sample (not play) audio?  (Read 9967 times)

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Offline Parkland

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Re: HOW do i sample (not play) audio?
« Reply #30 on: April 28, 2020, 07:26:28 pm »
It works in windows, I'll try the linux sfml library. Didn't even consider that being an issue. I'm not the sharpest board in the metal shop.

Is there a way to live sample mic ? Or do you have to record samples then analyze data from the recorded sample?
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Offline Parkland

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Re: HOW do i sample (not play) audio?
« Reply #31 on: April 28, 2020, 11:50:56 pm »
I tried to use this program to load the buffer and keep displaying it live, seems like the waveform does'nt match up though so I'm probably doing it wrong
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Offline Petr

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Re: HOW do i sample (not play) audio?
« Reply #32 on: April 29, 2020, 11:20:10 am »
Hi. Try this:

Code: QB64: [Select]
  1. DECLARE LIBRARY "SFML_Audio_Wrapper", "sfml\libsfml-audio"
  2.     SUB Mic_Start (BYVAL SampleRate&)
  3.     SUB Mic_End
  4.     SUB Mic_Play
  5.     SUB Mic_Pause
  6.     SUB Mic_Stop
  7.     SUB Mic_Save (Filename$)
  8.     FUNCTION Mic_Get_SampleCount&
  9.     FUNCTION Mic_Get_Sample% (BYVAL SampleNum&)
  10.     FUNCTION Sound_Load (FileName$)
  11.     FUNCTION Sound_Num_Samples
  12.     FUNCTION Sound_Get_Sample% (BYVAL SampleNum&)
  13.     SUB SNDRAW_Save (RawSnd%, BYVAL NumSamples&, BYVAL SampleRate&, BYVAL Channels&, FileName$)
  14.     FUNCTION Sound_Num_Channels&
  16.  
  17. SCREEN _NEWIMAGE(800, 600, 256)
  18.  
  19. '// Record 5 seconds of audio @ 36000 samples per second
  20. Mic_Start 44100
  22.  
  23. '// End the recording
  24. Mic_End
  25.  
  26. '// Play the mic sound directly
  27. Mic_Play
  29. 'after this delay is record size: 44100 (microphone sample rate) * 5 seconds
  30.  
  31.  
  32.  
  33.  
  34.  
  35. '// Load the samples into an array
  36. '// This is if you want to analyse the sound in anyway or distort it or something
  37. NumSamples& = Mic_Get_SampleCount& - 1
  38. DIM Sample(NumSamples&) AS INTEGER
  39. FOR I& = 0 TO NumSamples&
  40.     Sample(I&) = Mic_Get_Sample(I&)
  42.  
  43.  
  44. '// Play the samples stored in the array using sndraw - i dont now what you need to / or * the value by to get accurate
  45. '// playback from sndraw
  46.  
  47. TYPE SND
  48.     left AS SINGLE
  49.     right AS SINGLE
  51. size = NumSamples&
  52. DIM S(size) AS SND
  53.  
  54.  
  55.  
  56. FOR I& = 0 TO NumSamples& - 1 STEP 2
  57.     _SNDRAW Sample(I&) / 65535, Sample(I& + 1) / 65535
  58.     S(j).left = Sample(I&) / 65535
  59.     S(j).right = Sample(I& + 1) / 65535
  60.     j = j + 1
  61.  
  62.     X = X + 1: IF X > _WIDTH THEN CLS: X = 1
  63.     OYl = Yl
  64.     Yl = 200 + S(j&).left * 200
  65.     OYr = Yr
  66.     Yr = 400 + S(j&).right * 200
  67.     LINE (X - 1, OYl)-(X, Yl)
  68.     LINE (X - 1, OYr)-(X, Yr)
  69.  
  70.     DO WHILE _SNDRAWLEN > 0: LOOP
  71.     j& = j& + 1
  73.  
  74. SAVESOUND16S S(), "next_test.wav"
  75.  
  76.  
  77.  
  78. REM Mic_Save ("microphone.wav")
  79.  
  80. SUB SAVESOUND16S (arr() AS SND, file AS STRING)
  81. TYPE head16
  82.     chunk AS STRING * 4 '       4 bytes  (RIFF)
  83.     size AS LONG '              4 bytes  (file size)
  84.     fomat AS STRING * 4 '       4 bytes  (WAVE)
  85.     sub1 AS STRING * 4 '        4 bytes  (fmt )
  86.     subchunksize AS LONG '      4 bytes  (lo / hi), $00000010 for PCM audio
  87.     format AS INTEGER '         2 bytes  (0001 = standard PCM, 0101 = IBM mu-law, 0102 = IBM a-law, 0103 = IBM AVC ADPCM)
  88.     channels AS INTEGER '       2 bytes  (1 = mono, 2 = stereo)
  89.     rate AS LONG '              4 bytes  (sample rate, standard is 44100)
  90.     ByteRate AS LONG '          4 bytes  (= sample rate * number of channels * (bits per channel /8))
  91.     Block AS INTEGER '          2 bytes  (block align = number of channels * bits per sample /8)
  92.     Bits AS INTEGER '           2 bytes  (bits per sample. 8 = 8, 16 = 16)
  93.     subchunk2 AS STRING * 4 '   4 bytes  ("data")  contains begin audio samples
  94.     lenght AS LONG '            4 bytes  Data block size
  95. END TYPE '                     44 bytes  total
  96. DIM H16 AS head16
  98.  
  99. H16.chunk = "RIFF"
  100. H16.size = 44 + UBOUND(arr) * 4 'two channels, it create 16 bit, stereo wav file, one sample use 2 bytes to one channel
  101.  
  102. H16.fomat = "WAVE"
  103. H16.sub1 = "fmt "
  104. H16.subchunksize = 16
  105. H16.format = 1
  106. H16.channels = 2
  107. H16.rate = 44100
  108. H16.ByteRate = 44100 * 2 * 16 / 8
  109. H16.Block = 4
  110. H16.Bits = 16
  111. H16.subchunk2 = "data"
  112. H16.lenght = UBOUND(arr) * 4
  113. IF _FILEEXISTS(file$) THEN KILL file$
  114.  
  115. OPEN file$ FOR BINARY AS #ch
  116. PUT #ch, , H16
  117. DIM LeftChannel AS INTEGER, RightChannel AS INTEGER
  118.  
  119. FOR audiodata = 0 TO UBOUND(arr)
  120.     LeftChannel = arr(audiodata).left * 32768
  121.     RightChannel = arr(audiodata).right * 32768
  122.  
  123.     PUT #ch, , LeftChannel
  124.     PUT #ch, , RightChannel
  128.  

There are several ways to proceed: Do you want a program that allows you to move left - right to view the painted curve, or, a simpler program - to make a BMP output file that will have a resolution (for example) of 50000 * 600 and will contain the resulting curves?
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Offline MB

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Re: HOW do i sample (not play) audio?
« Reply #33 on: April 29, 2020, 02:20:51 pm »
Parkland,

When I do this code (running under 32-bit windows):

   Mic_Start 22000
   _DELAY 2
   Mic_End

I collect 33673 samples, my first 10515 samples are all zeros. samples 10516 through 44188 have data.

Every time I sample my hardware lags for around 10k samples when using the 22000 sample rate.

Do a simpler test of just running this codet:

   Mic_Start 22000
   _DELAY 5
   Mic_End
   Mic_Save ("test.raw") ' change .raw to a format you use on your platform.

just as a sanity check to see how much lag/(if any) you have before data begins to accumulate.

- MB

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Offline Unseen Machine

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Re: HOW do i sample (not play) audio?
« Reply #34 on: April 29, 2020, 02:54:42 pm »
Quote
Is there a way to live sample mic ? Or do you have to record samples then analyze data from the recorded sample?

There is (well almost) but this library doesnt support it. I know that in SFML 2.5 you can get samples back whilst a mic is still recording and i think i remember every 100mili seconds or something. I'll have a look into it and see if i can put something together in the next day or so.

As for the rest of the questions...i got no idea i wrote the wrapper the best part of a decade ago and have never used it since, i look forward to seeing what you guys make with it...

Unseen
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Offline MB

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Re: HOW do i sample (not play) audio?
« Reply #35 on: April 29, 2020, 03:01:36 pm »
Parkland,

This is the code I used to see the leading dead space, file "rec2sav.bas" attached.

Code just tells you amount of samples where about the empty buffer ends and the noise floor begins. Then the total samples.


Unseen, I love that you made this wrapper... as I only need to record samples and analyze data, this is a neat tool in my books!

- MB
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Offline Parkland

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Re: HOW do i sample (not play) audio?
« Reply #36 on: April 29, 2020, 08:41:20 pm »
I don't know how well this approach works, but it may do what I need. (see attached file)  [ You are not allowed to view this attachment ]  
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Offline Parkland

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Re: HOW do i sample (not play) audio?
« Reply #37 on: April 29, 2020, 08:50:49 pm »
By chance does anyone have any idea / experience with what method is used to figure out frequencies?
Possibly similar to FFT?

I tried measuring every time the wave crosses 0, but that only works for the loudest frequency.
I tried measuring every peak and valley, but that only works for the smallest frequency.

Wondering what process would normally be employed if you want to know strengths of certain frequencies in the sound.
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Offline MB

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Re: HOW do i sample (not play) audio?
« Reply #38 on: April 30, 2020, 08:07:29 am »
Quote from: Parkland on April 29, 2020, 08:50:49 pm
By chance does anyone have any idea / experience with what method is used to figure out frequencies?
Possibly similar to FFT?

I tried measuring every time the wave crosses 0, but that only works for the loudest frequency.
I tried measuring every peak and valley, but that only works for the smallest frequency.

Wondering what process would normally be employed if you want to know strengths of certain frequencies in the sound.

Crossing zero only works well with a single cycle style of waveform. 

I have been porting over some of my own code that I made in C for one of my "Teensy 4.0" microprocessor into QB64; however, it is far from being ready to share as is it implements the Laplace Transform.

QB64 user "_vince" made a real nice clean sample of a fast algorithm for computing the DFT for N=2^k in QB64. He also list a optimization for real only input FFT in the same thread, take a look at his post in this link:

https://www.qb64.org/forum/index.php?topic=1938.msg111661;topicseen#msg111661

-MB
« Last Edit: April 30, 2020, 08:26:39 am by MB »
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Offline _vince

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Re: HOW do i sample (not play) audio?
« Reply #39 on: April 30, 2020, 10:22:01 pm »
I've revised that FFT post to address some of this interest, particularly the last example

Code: QB64: [Select]
  1. 'defdbl a-z
  2.  
  3. const sw = 1024
  4. const sh = 600
  5.  
  7. 'pi = 2*asin(1)
  8. pi = 4*atn(1)
  9.  
  10. declare sub rfft(xx_r(), xx_i(), x_r(), n)
  11. declare sub fft(xx_r(), xx_i(), x_r(), x_i(), n)
  12. declare sub dft(xx_r(), xx_i(), x_r(), x_i(), n)
  13.  
  14.  
  15. dim x_r(sw-1), x_i(sw-1)
  16. dim xx_r(sw-1), xx_i(sw-1)
  18.  
  19. for i=0 to sw-1
  20.         x_r(i) = 100*sin(2*pi*(sw*2000/44000)*i/sw) + (100*rnd - 50)
  22.  
  23.  
  24. 'screenres sw, sh, 32
  25. screen _newimage(sw, sh, 32)
  26.  
  27. 'plot signal
  28. pset (0, sh/4 - x_r(0))
  29. for i=0 to sw-1
  30.         line -(i, sh/4 - x_r(i)), _rgb(255,0,0)
  32. line (0, sh/4)-step(sw,0), _rgb(255,0,0),,&h5555
  33.  
  34. _printstring (0, 0), "2000 kHz signal with RND noise sampled at 44 kHz in 1024 samples"
  35.  
  36.  
  37. rfft xx_r(), xx_i(), x_r(), sw
  38.  
  39. 'plot its fft
  40. pset (0, 50+3*sh/4 - 0.005*sqr(xx_r(0)*xx_r(0) + xx_i(0)*xx_i(0)) ), _rgb(255,255,0)
  41. for i=0 to sw/2
  42.         line -(i*2, 50+3*sh/4 - 0.005*sqr(xx_r(i)*xx_r(i) + xx_i(i)*xx_i(i)) ), _rgb(255,255,0)
  44. line (0, 50+3*sh/4)-step(sw,0), _rgb(255,255,0),,&h5555
  45.  
  46.  
  47. 'find peak
  49. max = 0
  50. m = 0
  51. for i=0 to sw/2
  52.         d = 0.01*sqr(xx_r(i)*xx_r(i) + xx_i(i)*xx_i(i))
  53.         if d > max then
  54.                 max = d
  55.                 m = i
  56.         end if
  58.  
  59. _printstring (0, sh/2), "m_peak ="+str$(m)
  60. _printstring (0, sh/2 + 16), "f_peak = m_peak * 44 kHz / 1024 samples = "+str$(m*44000/1024)+" Hz"
  61.  
  62. 'apply frequency correction, only works for some signals
  64. dim u_r as double, u_i as double
  65. dim v_r as double, v_i as double
  66.  
  67. u_r = xx_r(m - 1) - xx_r(m + 1)
  68. u_i = xx_i(m - 1) - xx_i(m + 1)
  69. v_r = 2*xx_r(m) - xx_r(m - 1) - xx_r(m + 1)
  70. v_i = 2*xx_i(m) - xx_i(m - 1) - xx_i(m + 1)
  71. c = (u_r*v_r + u_i*v_i)/(v_r*v_r + v_i*v_i)
  72.  
  73. _printstring (0, sh/2 + 2*16), "f_corrected = "+str$((m+c)*44000/1024)+" Hz"
  74.  
  77.  
  78.  
  79. sub rfft(xx_r(), xx_i(), x_r(), n)
  80.         dim w_r as double, w_i as double, wm_r as double, wm_i as double
  81.         dim u_r as double, u_i as double, v_r as double, v_i as double
  82.  
  83.         log2n = log(n/2)/log(2)
  84.  
  85.         for i=0 to n/2 - 1
  86.                 rev = 0
  87.                 for j=0 to log2n - 1
  88.                         if i and (2^j) then rev = rev + (2^(log2n - 1 - j))
  89.                 next
  90.  
  91.                 xx_r(i) = x_r(2*rev)
  92.                 xx_i(i) = x_r(2*rev + 1)
  93.         next
  94.  
  95.         for i=1 to log2n
  96.                 m = 2^i
  97.                 wm_r = cos(-2*pi/m)
  98.                 wm_i = sin(-2*pi/m)
  99.  
  100.                 for j=0 to n/2 - 1 step m
  101.                         w_r = 1
  102.                         w_i = 0
  103.  
  104.                         for k=0 to m/2 - 1
  105.                                 p = j + k
  106.                                 q = p + (m \ 2)
  107.  
  108.                                 u_r = w_r*xx_r(q) - w_i*xx_i(q)
  109.                                 u_i = w_r*xx_i(q) + w_i*xx_r(q)
  110.                                 v_r = xx_r(p)
  111.                                 v_i = xx_i(p)
  112.  
  113.                                 xx_r(p) = v_r + u_r
  114.                                 xx_i(p) = v_i + u_i
  115.                                 xx_r(q) = v_r - u_r
  116.                                 xx_i(q) = v_i - u_i
  117.  
  118.                                 u_r = w_r
  119.                                 u_i = w_i
  120.                                 w_r = u_r*wm_r - u_i*wm_i
  121.                                 w_i = u_r*wm_i + u_i*wm_r
  122.                         next
  123.                 next
  124.         next
  125.  
  126.         xx_r(n/2) = xx_r(0)
  127.         xx_i(n/2) = xx_i(0)
  128.  
  129.         for i=1 to n/2 - 1
  130.                 xx_r(n/2 + i) = xx_r(n/2 - i)
  131.                 xx_i(n/2 + i) = xx_i(n/2 - i)
  132.         next
  133.  
  134.         dim xpr as double, xpi as double
  135.         dim xmr as double, xmi as double
  136.  
  137.         for i=0 to n/2 - 1
  138.                 xpr = (xx_r(i) + xx_r(n/2 + i)) / 2
  139.                 xpi = (xx_i(i) + xx_i(n/2 + i)) / 2
  140.  
  141.                 xmr = (xx_r(i) - xx_r(n/2 + i)) / 2
  142.                 xmi = (xx_i(i) - xx_i(n/2 + i)) / 2
  143.  
  144.                 xx_r(i) = xpr + xpi*cos(2*pi*i/n) - xmr*sin(2*pi*i/n)
  145.                 xx_i(i) = xmi - xpi*sin(2*pi*i/n) - xmr*cos(2*pi*i/n)
  146.         next
  147.  
  148.         'symmetry, complex conj
  149.         for i=0 to n/2 - 1
  150.                 xx_r(n/2 + i) = xx_r(n/2 - 1 - i)
  151.                 xx_i(n/2 + i) =-xx_i(n/2 - 1 - i)
  152.         next
  154.  
« Last Edit: April 30, 2020, 10:32:53 pm by _vince »
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Offline Parkland

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Re: HOW do i sample (not play) audio?
« Reply #40 on: May 04, 2020, 03:30:33 pm »
Wow! very awesome.
Thanks everyone.

Also attached is the most recent version of what I glued together for live sampling
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