Sugestions or Comments about my Sound Tutorial

The tutorial (under construction):
http://www.vbforums.com/showthread.php?t=388562

Please post any sugestions or comments about my tutorial here.

What would you like to see in my tutorial ? (keep in mind that it is about sound using DirectXSound)

Any comments about what I already have ? something that needs more explaining, or not clear enough ?
Did I make any mistakes in that thread ? am I saying something that is not true/wrong ?

Keep in mind that it is under construction, and it will be for a long time. I'm doing this in my free time (at work, and home).

Re: Sugestions or Comments about my Sound Tutorial

You said,

Quote:

Originally Posted by CVMichael

Title: VB6.0 – Sound and DirectXSound Tutorial

I thought you would cover non-Dx sound too.

Will you cover MIDI ? That's what I'm interested in. :)

I wish for a good (preferebly non-Dx) MIDI tutorial. But the way you started, I guess DirectX woun't be too hard. :thumb:

I want to build a software synthesizer that can save compositions as MIDI files.

P.S. Why that dummy file is there ? :D

Re: Sugestions or Comments about my Sound Tutorial

Sorry, I don't know anything about MIDI, so I won't cover it.

About the dummy file: vbforums.com will display the attached picture in the attachment section if there is only one picture attached. If there is more than one picture, it won't display any of them. This way i'm tricking it into not displaying the picture, because I don't want to display it there, I want to display the picture at the beginning of the post.

The non-DX sound, I mean wave files, creating and modifying sounds directly in the Byte (or Integer) arrays, as opposed to modify the sound in the DX buffer, or DX functions.

Re: Sugestions or Comments about my Sound Tutorial

It looks good so far. :)

You can put the image attachment in a post in the Support Images forum, and just link to it.

The only comment I have is the "Sound format: What is ..." section seems to run-on, I'd prefer to see it either as a list, or with each item (eg: Block Align) underlined and/or bold to make them stand out.

Re: Sugestions or Comments about my Sound Tutorial

Thanks...

OK... I'll do that (underlineing, and bold)

I also want to make some animated GIFs for the next section, but I'm having trouble finding a good and free GIF maker program.
http://www.vbforums.com/showthread.php?t=388956

I tried one (IrfanView 3.98) from download.com and I found the interface so difficult, I could not figure out how to even make a simple GIF, so I just un-installed it.

Re: Sugestions or Comments about my Sound Tutorial

I saw a post last week linking to a good simple one - but have no idea what it was called or which thread it was in tho :(

Re: Sugestions or Comments about my Sound Tutorial

"What is sound ?":
What about frequencies, and their representation as samples?

"Sound format":
Samplerate - I think you should also explain the Nyquist frequency.

"How to read or write to a wave file ?":
I thought those headers were called chunks.
Also your WaveReadFormat function may not work for all wavs,
there could also be other chunks (for example comments) before
the data chunk, or the data could be compressed, so the format
chunk could be much bigger.

I'm excited about the more advaned topics like tone recognition
and resampling. Will you also cover the FFT and digital filters (IIR, FIR)?

For the ACM encoder, I could give you a class I've written,
if you're interested.

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by rm_03

"What is sound ?":
What about frequencies, and their representation as samples?

I can explain about frequencies... but I'm not sure what you mean by "and their representation as samples"

Quote:

Originally Posted by rm_03

"Sound format":
Samplerate - I think you should also explain the Nyquist frequency.

I don't know anything about Nyquist frequency, I heared before about Nyquist theorem. But as I was saying don't know anything about it.
Maybe you can exaplain it, and I'll put it in the tutorial (giving credit to you for that part of course)

Quote:

Originally Posted by rm_03

"How to read or write to a wave file ?":
I thought those headers were called chunks.

Yea... I do have a "ChunkHeader" structure there...

Quote:

Originally Posted by rm_03

Also your WaveReadFormat function may not work for all wavs,
there could also be other chunks (for example comments) before
the data chunk, or the data could be compressed, so the format
chunk could be much bigger.

That's what this code does (From the WaveReadFormat):
VB Code:
' Ignore chunks until we get to the "data" chunk.
    Get #InFileNum, , chunk
    Do While chunk.lType <> &H61746164
        For i = 1 To chunk.lLen
            Get #InFileNum, , by
        Next
        Get #InFileNum, , chunk
    Loop
Quote:

Originally Posted by rm_03

Will you also cover the FFT and digital filters (IIR, FIR)?

I was thinking to write something about FFT, but I have only little experience with it.
And I don't know about filters, but I'm excited to learn. And when I know something then i'll put it in the tutorial.
I was actually looking for a long time for code (or formulas) about digital filters, but I could not find anything usefull.

Quote:

Originally Posted by rm_03

For the ACM encoder, I could give you a class I've written,
if you're interested.

I saw you code already, not long before you posted it in the CodeBank, but one problem I found with it is that it does not convert in real time, you have to give it a wave file, and then it will convert to MP3.
I want to do this in real time, the code that I will explaing in the tutorial will be for both actually, for real time, and separate converting wave file to MP3 file.

Re: Sugestions or Comments about my Sound Tutorial

Quote:

I can explain about frequesncies... but I'm not sure what you mean by "and their representation as samples"

Basic stuff like how does a frequency at 10 Hz with the amplitude 16000 look like, and how to use sample points to create such a frequency.

Quote:

Nyquist frequency

Never heard of the "Nyquist theorem" ;), but the Nyquist frequency (the highest possible freq) would
be samplerate / 2, because you need at least 2 sample points for a frequency.

Quote:

That's what this code does (From the WaveReadFormat):

Oops, didn't see that, sorry :)

Quote:

I was actually looking for a long time for code (or formulas) about digital filters, but I could not find anything usefull.

Have you visited MusicDSP yet?
http://www.musicdsp.org/archive.php?classid=3
"RBJ Audio-EQ-Cookbook" is a good resource for various IIR filters.

I'd love to see that tutorial grow, there's nothing comparable to it in the VB world yet.

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by rm_03

Basic stuff like how does a frequency at 10 Hz with the amplitude 16000 look like, and how to use sample points to create such a frequency.

I was planing to show how to create tones of certain frequencyes, if you look at "My Objectives", You will see "How to dinamically create tones/beeps with custom frequencies."
I'm gonna put some images showing the wave data in that section of the tutorial.

Quote:

Originally Posted by rm_03

Never heard of the "Nyquist theorem" ;), but the Nyquist frequency (the highest possible freq) would
be samplerate / 2, because you need at least 2 sample points for a frequency.

Ow yea, that ring a bell :D
Yea, I think I'll explain it, but I have to read a little more about it, so that I can explain in words that actually make sence :)

Quote:

Originally Posted by rm_03

Have you visited MusicDSP yet?
http://www.musicdsp.org/archive.php?classid=3
"RBJ Audio-EQ-Cookbook" is a good resource for various IIR filters.

Nice !
I will definitelly spend some time checking the filters, and if I can get (at least one) of them to work, then I'll put that in the tutorial too.

Quote:

Originally Posted by rm_03

there's nothing comparable to it in the VB world yet.

What do you mean by that ?
You mean no one made a tutorial about sound in VB yet ? (I doubt that)

Re: Sugestions or Comments about my Sound Tutorial

Quote:

You mean no one made a tutorial about sound in VB yet ? (I doubt that)

At least I havn't seen anything for VB which deals with the lower level parts.
You just can't call "how to use MCI/WMP" a sound tutorial.

BTW, IIR filter for 16 bit:

in a module:

Code:

Option Explicit Private Const Pi As Single = 3.14159265358979 Private Const LN2 As Single = 0.693147180559945 Public Enum IIR_TYPE IIR_LOW_PASS IIR_HIGH_PASS IIR_BAND_PASS IIR_NOTCH IIR_PEAK_EQ IIR_LOW_SHELVE IIR_HIGH_SHELVE End Enum Public Function CreateBiquadIIR( _ ByVal iirtype As IIR_TYPE, _ ByVal dBGain As Single, _ ByVal freq As Single, _ ByVal srate As Single, _ ByVal bandwidth As Single _ ) As IIRFilter Dim A As Single, omega As Single Dim sn As Single, cs As Single Dim Alpha As Single, Beta As Single Dim a0 As Single, a1 As Single, a2 As Single Dim b0 As Single, b1 As Single, b2 As Single Dim filter As IIRFilter Set filter = New IIRFilter A = 10 ^ (dBGain / 40) omega = 2 * Pi * freq / srate sn = Sin(omega) cs = Cos(omega) Alpha = sn * sinh(LN2 / 2 * bandwidth * omega / sn) Beta = Sqr(A + A) Select Case iirtype Case IIR_LOW_PASS b0 = (1 - cs) / 2 b1 = 1 - cs b2 = (1 - cs) / 2 a0 = 1 + Alpha a1 = -2 * cs a2 = 1 - Alpha Case IIR_HIGH_PASS b0 = (1 + cs) / 2 b1 = -(1 + cs) b2 = (1 + cs) / 2 a0 = 1 + Alpha a1 = -2 * cs a2 = 1 - Alpha Case IIR_BAND_PASS b0 = Alpha b1 = 0 b2 = -Alpha a0 = 1 + Alpha a1 = -2 * cs a2 = 1 - Alpha Case IIR_NOTCH b0 = 1 b1 = -2 * cs b2 = 1 a0 = 1 + Alpha a1 = -2 * cs a2 = 1 - Alpha Case IIR_PEAK_EQ b0 = 1 + (Alpha * A) b1 = -2 * cs b2 = 1 - (Alpha * A) a0 = 1 + (Alpha / A) a1 = -2 * cs a2 = 1 - (Alpha / A) Case IIR_LOW_SHELVE b0 = A * ((A + 1) - (A - 1) * cs + Beta * sn) b1 = 2 * A * ((A - 1) - (A + 1) * cs) b2 = A * ((A + 1) - (A - 1) * cs - Beta * sn) a0 = (A + 1) + (A - 1) * cs + Beta * sn a1 = -2 * ((A - 1) + (A + 1) * cs) a2 = (A + 1) + (A - 1) * cs - Beta * sn Case IIR_HIGH_SHELVE b0 = A * ((A + 1) + (A - 1) * cs + Beta * sn) b1 = -2 * A * ((A - 1) + (A + 1) * cs) b2 = A * ((A + 1) + (A - 1) * cs - Beta * sn) a0 = (A + 1) - (A - 1) * cs + Beta * sn a1 = 2 * ((A - 1) - (A + 1) * cs) a2 = (A + 1) - (A - 1) * cs - Beta * sn End Select filter.a0 = b0 / a0 filter.a1 = b1 / a0 filter.a2 = b2 / a0 filter.a3 = a1 / a0 filter.a4 = a2 / a0 filter.x1 = 0: filter.x2 = 0 filter.y1 = 0: filter.y2 = 0 Set CreateBiquadIIR = filter End Function Private Function sinh(angle As Single) As Single sinh = (Exp(angle) - Exp(-angle)) / 2# End Function

class "IIRFilter":

Code:

Option Explicit ' http://www.dspguru.com/sw/lib/biquad.c Public a0 As Single Public a1 As Single Public a2 As Single Public a3 As Single Public a4 As Single Public x1 As Single Public x2 As Single Public y1 As Single Public y2 As Single Public Sub ProcessSamples( _ intSamples() As Integer, _ ByVal datalen As Long _ ) Dim result As Single Dim sample As Single Dim i As Long For i = 0 To datalen sample = intSamples(i) / 32767 result = a0 * sample + a1 * x1 + a2 * x2 - a3 * y1 - a4 * y2 x2 = x1 x1 = sample y2 = y1 y1 = result If result > 1# Then intSamples(i) = 32767 ElseIf result < -1# Then intSamples(i) = -32768 Else intSamples(i) = CInt(result * 32767) End If Next End Sub

Usage:

Code:

Dim clsIIR as IIRFilter Dim intSamples(44100 - 1) As Integer ' could be any size Set clsIIR = CreateBiquadIIR(IIR_PEAK_EQ, 6, 170, 44100, 1) clsIIR.ProcessSamples intSamples, UBound(intSamples)

Will boost your signal at 170 Hz by 6 dB with a bandwidth of 1 octave.
I love them :) Create some of them in a row and you got an equalizer.

Re: Sugestions or Comments about my Sound Tutorial

OMG !
I LOVE THAT...
I've been looking for an equalizer code for so long...

Where have you been till now ? :D

Though I have not tried it yet... owwww... I can't wait to have more free time ! to fool around with that code, and figure out how it works :)

I'm at work right now, and I'm busy at the moment. I hope I have time when I get home to look at it.

If you have more stuff like this, POST IT ! :D

Re: Sugestions or Comments about my Sound Tutorial

Not much left...

FFT with windowing:

Code:

Option Explicit Private Const Pi As Single = 3.14159265358979 Private Const AngleNum As Single = 2 * Pi Private lngPow2(31) As Long Public Enum FFT_WINDOW WINDOW_FUNC_NONE WINDOW_FUNC_HANNING WINDOW_FUNC_HAMMING WINDOW_FUNC_BLACKMAN End Enum ' http://www.fullspectrum.com/deeth/main.html Public Sub FastFourierTransform( _ NumSamples As Long, _ RealIn() As Integer, _ RealOut() As Single, _ Optional wnd As FFT_WINDOW = WINDOW_FUNC_NONE _ ) Dim NumBits As Long Dim Rev As Long Dim index As Long Dim i As Long, j As Long Dim k As Long, n As Long Dim BlockSize As Long, BlockEnd As Long Dim DeltaAngle As Single, DeltaAr As Single Dim Alpha As Single, Beta As Single Dim TR As Single, TI As Single Dim AR As Single, AI As Single Dim dblWnd() As Single Dim imagout() As Single ReDim imagout(NumSamples - 1) As Single dblWnd = CreateWindow(wnd, NumSamples) For i = 0 To 16 If (NumSamples And lngPow2(i)) <> 0 Then NumBits = i Exit For End If Next For i = 0 To (NumSamples - 1) index = i Rev = 0 For k = 0 To NumBits - 1 Rev = (Rev * 2) Or (index And 1) index = index \ 2 Next j = Rev RealOut(j) = RealIn(i) * dblWnd(i) Next BlockEnd = 1 BlockSize = 2 Do While BlockSize <= NumSamples DeltaAngle = AngleNum / BlockSize Alpha = Sin(0.5 * DeltaAngle) Alpha = 2# * Alpha * Alpha Beta = Sin(DeltaAngle) For i = 0 To NumSamples - 1 Step BlockSize AR = 1# AI = 0# j = i For n = 0 To BlockEnd - 1 k = j + BlockEnd TR = AR * RealOut(k) - AI * imagout(k) TI = AI * RealOut(k) + AR * imagout(k) RealOut(k) = RealOut(j) - TR imagout(k) = imagout(j) - TI RealOut(j) = RealOut(j) + TR imagout(j) = imagout(j) + TI DeltaAr = Alpha * AR + Beta * AI AI = AI - (Alpha * AI - Beta * AR) AR = AR - DeltaAr j = j + 1& Next Next BlockEnd = BlockSize BlockSize = BlockSize * 2& Loop End Sub Public Sub InitPower2() Dim i As Long For i = 0 To 30 lngPow2(i) = 2 ^ i Next lngPow2(31) = &H80000000 End Sub Public Function CreateWindow( _ wnd As FFT_WINDOW, _ ByVal Length As Long _ ) As Single() Dim dblOut() As Single Dim i As Long ReDim dblOut(Length - 1) As Single Select Case wnd Case WINDOW_FUNC_NONE For i = 0 To Length - 1 dblOut(i) = 1 Next Case WINDOW_FUNC_HANNING For i = 0 To Length - 1 dblOut(i) = 0.5 * (1 - Cos(i * 2 * Pi / (Length - 1))) Next Case WINDOW_FUNC_HAMMING For i = 0 To Length - 1 dblOut(i) = 0.54 - (0.46 * Cos((i) * 2 * Pi / (Length - 1))) Next Case WINDOW_FUNC_BLACKMAN For i = 0 To Length - 1 dblOut(i) = 0.42 - (0.5 * Cos((i) * 2 * Pi / (Length - 1))) + (0.08 * Cos((i) * 4 * Pi / (Length - 1))) Next End Select CreateWindow = dblOut End Function

Usage:

Code:

InitPower2 ' needs to be called only once FastFourierTransform 1024, intSamplesInput, sngSamplesOutput, WINDOW_FUNC_HANNING

Will apply an Hanning window on intSamplesInput and return the
frequency domain in sngSamplesOutput (NumSamples has to be 2^N),
where each element in the array represents samplerate / NumSamples bandwidth, if I remember correctly.

Changing volume:

Code:

Public Enum VOL_UNIT VOL_DECIBEL VOL_PERCENT VOL_FACTOR End Enum Public Sub ChangeVolume( _ intSamples() As Integer, _ ByVal datalen As Long, _ ByVal value As Single, _ ByVal unit As VOL_UNIT _ ) Dim sngFactor As Single Dim sngResult As Single Dim i As Long Select Case unit Case VOL_DECIBEL sngFactor = 10 ^ (value / 20) Case VOL_PERCENT sngFactor = value / 100 Case VOL_FACTOR sngFactor = value End Select For i = 0& To datalen sngResult = intSamples(i) * sngFactor If sngResult > 32767# Then intSamples(i) = 32767 ElseIf sngResult < -32768# Then intSamples(i) = -32768 Else intSamples(i) = CInt(sngResult) End If Next End Sub

Usage:

Code:

ChangeVolume intSamples, UBound(intSamples), 6, VOL_DECIBEL

Will boost the volume by 6 dB.

Echo:

Code:

Private intEcho() As Integer Private lngEchoPos As Long Private lngEchoLength As Single Public Sub DSPEcho( _ intSamples() As Integer, _ ByVal datalength As Long _ ) Dim i As Long For i = 0 To datalength intSamples(i) = norm(CLng(intSamples(i)) + intEcho(lngEchoPos)) intEcho(lngEchoPos) = intSamples(i) * lngEchoLength lngEchoPos = lngEchoPos + 1 If lngEchoPos > UBound(intEcho) Then lngEchoPos = 0 End If Next End Sub Public Sub DSPEchoSettings( _ ByVal samplerate As Long, _ ByVal echo_length_ms As Long, _ ByVal echo_length As Single _ ) Dim lngEchoPoints As Long lngEchoPoints = samplerate / 1000 * echo_length_ms ReDim intEcho(lngEchoPoints - 1) As Integer lngEchoLength = echo_length lngEchoPos = 0 End Sub

Usage:

Code:

DSPEchoSettings 44100, 500, 0.4 DSPEcho intSamples, UBound(intSamples)

Will add an echo to the signal wich is 500 ms long and will be multiplied by 0.4 after each reverb.

Decibel Full Scale:

Code:

Public Function dBFS(ByVal amplitude As Long) As Double If amplitude = 0 Then dBFS = -96 Else dBFS = 20 * ((Log(Abs(amplitude) / 32768)) / Log(10)) End If End Function

Will return a value from -96 to 0 dBFS.

I also have written encoders and decoders for wav/mp3/ape/ogg/wma/cda (for a streaming player), but with german comments, if you're interested.
I just don't have the time to translate it all, else I would've already posted it at PSC.

Re: Sugestions or Comments about my Sound Tutorial

I've seen a few "using sound in VB" type tutorials, but certainly nothing this complete - although I must admit I haven't looked particularly hard!

It looking to me as if your tutorial will become a must-bookmark page. :thumb:

Re: Sugestions or Comments about my Sound Tutorial

I also have some FFT code in C++ and VB that I used before, but I never had time to look at it, and understand how it works, and how to use it in practical cases.
I only used it before to show that "wave preview" thing while the music is recorded/played...
From what I know you can do neet stuff like sound recognition with FFT, but never figured it out how to implement it for that.

The echo and volume thing I already know how to do, it's easy... if you look at "My Objectives" I already have those listed there...

Quote:

Originally Posted by rm_03

I also have written encoders and decoders for wav/mp3/ape/ogg/wma/cda (for a streaming player), but with german comments, if you're interested.
I just don't have the time to translate it all, else I would've already posted it at PSC.

Yes, I am interested... I don't care if it's in another language, as long as the code is clearly written, I will understand what is going on (as long as I have enough time). I actually never understood someone else's comments.

What anoys me with comments is something like:

Open "c:\some file.txt" for binary as #1 ' Open the file

I'm like: "Really... I thought you close the file with that statement..."
So comments don't matter to me.

But I have to admit, I'm doing the same thing in my tutorial :D

1 Attachment(s)

Re: Sugestions or Comments about my Sound Tutorial

The basic idea behind the Fourier Transformation is to split a periodic function into some sine and cosine functions.
That way you can also obtain the frequencies of an audio signal.
The output array of the FFT function I posted above contains:

element 0: d/c offset (whatever that is...)
element 1 - n/2: frequencies
element n/2 - n: again frequencies, but in reversed order (maybe cosine values)

each element in 1 - n/2 is a frequency band with a bandwidth of samplerate/num_samples Hz.
So, the more samples you use, the more accurate it will get, but also the slower.
Also, the FFT is fastest with 2^n samples.
I use the FFT for a Winamp like visualisation of audio, so I only use 512 samples (very fast compiled).

I don't think you need to know more about the FFT (maybe a bit about the DCT).

Quote:

I don't care if it's in another language

Of course the DLLs wich do the actual job are not written in VB.
Anyway, I attached my project. The UI of the test projects is german,
but I think it's easy to guess.

Re: Sugestions or Comments about my Sound Tutorial

What do you guys think of the tutorial so far ?

(posting also to bump...)

Re: Sugestions or Comments about my Sound Tutorial

I really like the resampling function, I tried that before but failed on downsampling, although I havn't tested yours yet.
It looks like you're using linear interpolation, cubic or Hermite interpolation should increase the quality.
(http://astronomy.swin.edu.au/~pbourk...ion/index.html)
I've also heard you should do low pass filtering to remove frequencies above the new Nyquist frequency when downsampling, but never dealed with it.

Anyway, I've just ported a Goertzel algorithm (Fourier transformation, but only for 1 frequency)
for tone detection to VB,
maybe you can use it for your tone detection topic:

Code:

Function Goertzel( _ sngData() As Single, _ ByVal N As Long, _ ByVal freq As Single, _ ByVal sampr As Long _ ) As Single Dim Skn As Single Dim Skn1 As Single Dim Skn2 As Single Dim c As Single Dim c2 As Single Dim i As Long c = 2 * PI * freq / sampr c2 = Cos(c) For i = 0 To N - 1 Skn2 = Skn1 Skn1 = Skn Skn = 2 * c2 * Skn1 - Skn2 + sngData(i) Next Goertzel = Skn - Exp(-c) * Skn1 End Function Function power(ByVal value As Single) As Single power = 20 * Log(Abs(value)) / Log(10) End Function

Usage: (returns dB of 8000 Hz in the signal at 44100 samples/s)

Code:

dB = power(Goertzel(sngSignal, UBound(sngSignal) + 1, 8000, 44100))

I also did some testing with DTMF tone detection,
it can detect a generated tone in 0.01 seconds in the IDE
(compiled with optimizations 0.001 s), which is pretty fast on a P3 550 Mhz, I think.

Re: Sugestions or Comments about my Sound Tutorial

That is becoming one huge thread!! It may be better if it was split across a few threads (Basic/Medium/Advanced?), with an Index thread to link them together - it's up to you tho, it's just a thought!

Another thing, instead of having the images uploaded into the thread itself, you can create a thread in the Support Images forum which contains them (as I do for the FAQs). You could also do this with the other attachments if you like.

Regarding the audio tutorial under construction

It seems the audio tutorial here contains the following error in the first WaveReadFormat() routine:

VB Code:

If header.lFormatLength < 16 Then Exit Function

header.lFormatLength holds 16 for either 8, 16 or 32 bits WAVE files.

The correct procedure would be to ask for the WaveFormat wBitsPerSample value, but we cannot simply "replace" it here since "Dim WaveFmt As WaveFormat" is declared in another different function named Create_Wave_File_Example(). (or, this lFormatLength might have nothing to do with the actual WAV bits and be a totally different parameter not explained in the tutorial)

Anyway, I have joined both WaveReadFormat and Create_Wave_File in one, in a function that simply retrieves True if the file is useable and False if not, and shows a messagebox when not. The function only let pass 16bit files, and I also use the WavDataStart variable to store the position of the actual WAV audio data start.

VB Code:

Public Function CheckWaveFile(ByVal sFileName As String) as Boolean
'Returns True if file is PCM WAVE and and 16-bit
 
Dim wHeader As FileHeader
Dim wFormat As WaveFormat
Dim wChunk As ChunkHeader
Dim lDataLength As Long
Dim wB as Byte
Dim wL as Long
 
    WavNum = FreeFile
    Open sFileName For Binary Access Read As #WavNum
    
    Get #WavNum, 1, wHeader
    
    If wHeader.lRiff <> &H46464952 Then GoTo InvalidFile   ' Check for "RIFF" tag and exit if not found.
    If wHeader.lWave <> &H45564157 Then GoTo InvalidFile   ' Check for "WAVE" tag and exit if not found.
    If wHeader.lFormat <> &H20746D66 Then GoTo InvalidFile ' Check for "fmt " tag and exit if not found.
    
    Get #WavNum, , wFormat ' Retrieve format.
    
    If wFormat.wBitsPerSample <> 16 Then GoTo InvalidFile ' check if file is 16-bit
    
    ' Seek to next chunk by discarding any format bytes.
    For wL = 1 To wHeader.lFormatLength - 16
        Get #WavNum, , wB
    Next
    
    ' Ignore chunks until we get to the "data" chunk.
    Get #WavNum, , wChunk
    Do While wChunk.lType <> &H61746164
        For wL = 1 To wChunk.lLen
            Get #WavNum, , wB
        Next
        Get #WavNum, , wChunk
    Loop
    
    WavDataStart = LOF(WavNum) - wChunk.lLen + 1 'get position of audio data start
    WavBytesPerSec = wFormat.nAvgBytesPerSec
    WavBytesPerSample = (wFormat.wBitsPerSample / 8) * wFormat.nChannels
    
    'Debug.Print "wav data start = " & WavDataStart
    'Debug.Print "bytes per second = " & WavBytesPerSec
        
    Close #WavNum
    
    Exit Function
    
InvalidFile:
    MsgBox "Failed to load selected wave file. Only 16-bit PCM WAV (uncompressed) files are supported.", vbCritical, App.ProductName
    Close #WavNum
    
End Function

The tutorial so far is great and very clear, good work.

Re: Sugestions or Comments about my Sound Tutorial

Inuya5ha:

CVMichael has asked the comments/suggestions/etal be posted here so they are all in one place for him to look at. Therefore, I've merged the thread you posted in ClassicVb with this one.

Thanks for your feedback. :)

Re: Sugestions or Comments about my Sound Tutorial

Thanks for merging Hack.

Inuya5ha, the variable that returns how many bits the wave has is wFormat.wBitsPerSample.

So, to complete your function, put this line at the end of the function (Just before the "Exit Function" statement).
VB Code:
CheckWaveFile = wFormat.wBitsPerSample = 16

Re: Sugestions or Comments about my Sound Tutorial

Ow... I just noticed that you have the right code in the function:
VB Code:
If wFormat.wBitsPerSample <> 16 Then GoTo InvalidFile ' check if file is 16-bit
I don't understand what is the problem then ?

Why do you mention about header.lFormatLength, when you do have the right code ?

header.lFormatLength just returns the length of the structure, wich is always the same.

1 Attachment(s)

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by rm_03

It looks like you're using linear interpolation, cubic or Hermite interpolation should increase the quality.
(http://astronomy.swin.edu.au/~pbourk...ion/index.html)
I've also heard you should do low pass filtering to remove frequencies above the new Nyquist frequency when downsampling, but never dealed with it.

Thanks for the link.

I implemented the Cubic and Hermite interpolation from the link you gave me, but I don't see significant diferences between linear and the other 2.

Hermite looks best compared to the original.

I will make another test program that will convert wave files and save them into new ones, so that I can hear the diference.

I attached a test project, please take a look at it, and tell me what you think ?

I did not have time to take a look at the rest of the code you gave me. I'm very busy lately.

Re: Sugestions or Comments about my Sound Tutorial

Quote:

I implemented the Cubic and Hermite interpolation from the link you gave me, but I don't see significant diferences between linear and the other 2.

Take a look at this:
http://musicdsp.org/files/other001.gif

Quote:

I attached a test project, please take a look at it, and tell me what you think ?

Very nice stuff, hopefully this has some real time capabilities :)

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by rm_03

Take a look at this:
http://musicdsp.org/files/other001.gif

I have no clue what i'm seeing in that gif...

Quote:

Originally Posted by rm_03

Very nice stuff, hopefully this has some real time capabilities :)

Well, if in VB is not fast enough, it can always be converted to a C++ DLL, and used in VB like API... that will be fast for sure :)

Re: Sugestions or Comments about my Sound Tutorial

Quote:

I have no clue what i'm seeing in that gif...

X is frequency, Y is dB at that freq.
The blue line is perfect interpolation, how it should look like.
The green line is "no interpolation", and turns into a lot of audible distortion.
The pink line is linear interpolation, which already has a better quality.
The black line is cubic interpolation, and has the lowest distortion.

A perfect interpolation would be using sinc functions, I heard.

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by rm_03

Usage: (returns dB of 8000 Hz in the signal at 44100 samples/s)

Code:

dB = power(Goertzel(sngSignal, UBound(sngSignal) + 1, 8000, 44100))

I also did some testing with DTMF tone detection,
it can detect a generated tone in 0.01 seconds in the IDE
(compiled with optimizations 0.001 s), which is pretty fast on a P3 550 Mhz, I think.

I finally had time to try it.

I'm speechless, very nice !

It's soo precise !
I tried with a dual tone with frequencies (tones from the phone): 770 Hz and 1209 Hz (wich coresponds to #4 on the phone)

I made a short program that tests with all the frequencies from the phone, and I got this:
First number is the frequency and in brackets is the dB
Test1:
350(075.87) 440(060.86) 480(046.48) 620(076.66) 697(077.03) 770(171.13) 852(068.98) 941(042.27) 1209(171.14) 1336(067.06) 1477(062.78)
Test2:
350(075.87) 440(060.86) 480(046.48) 620(076.66) 697(077.03) 771(111.53) 852(068.98) 941(042.27) 1209(171.14) 1336(067.06) 1477(062.78)

If you notice In the first test for the frequency 770 the volume is 171, and for frequency 1209 is the same.
The second test, I changed it to test for 771 Hz instead of 770 Hz, and the result is 111 dB

The diference in frequency is only 1 Hz, but a very big diference in dB (wich is very good)

Pretty fast too...

Though I made the test with digitaly created wave files (wich have perfect quality), not actual tone from the phone. I want to test for that too, but I don't have time right now.

Re: Sugestions or Comments about my Sound Tutorial

I have a question about your attached oscilloscope program
I understand as much as I can from the code but I could not understant where is the call to this function "DirectSoundRecord_GotWaveData"
It seems that we are calling this function after specific time...
where are we calling this function and how to change this specific time???

Re: Sugestions or Comments about my Sound Tutorial

GotWaveData is an event that comes from the DirectSound form (frmDX_Record.frm).

If you want to change the interval that event is called, you have to change the buffer size, and that is done in the Initialize function.
Look at the line in red:
VB Code:
If HalfBufferLen <= 0 Then
            ' make half of the buffer to be 100 ms
            [COLOR=Red]HalfBuffLen = .lAvgBytesPerSec / 10[/COLOR]
        Else
            ' using a "custom" size buffer
            HalfBuffLen = HalfBufferLen
        End If
From that line the program chooses how big the buffer is, therefore how often the GotWaveData is called.

Re: Sugestions or Comments about my Sound Tutorial

Thanks ;)

Re: Sugestions or Comments about my Sound Tutorial

I don't have time to read this whole thread but I do have some comments and suggestions.

I know you are trying to make it easier to understand for us DS noobs, but your split buffer tutorial can't even read all the wave files! Just 16 bit stereo I think. So you are gonna need to add some code to support all wavs like one of your tutorials have. Not to mention not have it load a wav whenever it finishes like it currently is doing. And unfortunately, your "support all wavs code" doesn't have a sample project, so I don't know what to do with it really. :(

I would also like you to add these tutorials:

How to reverse the sounds so you can play it backwards.
Allow more splits in the buffer
Add special effects like chorus, flange, reverb, etc.
How to equalize the sound.
Play only the bass or treble portion of the sound.
Changing the volumn of the buffer as well as the master volumn.
How to crossfade.
And we end the tutorial with a much more improved scratching program. :bigyello:

I also would like to see a DirectSound module engine with all the tools you need to manipulate, record, play, pause, stop, etc., sounds. Otherwise I'm gonna be borrowing code from your scratching prog to make one on my own. Messed up on the last one cause the sound was skipping or jumping a little bit.

How to adjust mic input level?

:wave: Hi CVMichael,
First let me say a big thank you for the exellent vb6 tone record and analyse app. It was very interesting and just what I was looking for.
I have adapted it and now have it all working and it is brilliant! :thumb:

I have one small problem left however, I need to adjust the mic record level in the vb6 app. My plan is to adjust the level until I get a good signal (i.e. I can analyse a tone which is being played and get the correct number).

I have looked at technet, etc. but cannot find anywhere how to adjust the mic level. Can you help please?

Many thanks

Steve

Re: How to adjust mic input level?

Quote:

Originally Posted by steve6375

I have one small problem left however, I need to adjust the mic record level in the vb6 app. My plan is to adjust the level until I get a good signal (i.e. I can analyse a tone which is being played and get the correct number).

Unfortunatelly, I don't know how to do that either, but if you do find code on how to change the rec volume, then I'm pretty sure I can figure out how to auto change to optimal level...

Re: Sugestions or Comments about my Sound Tutorial

Thanks for the reply.
I found this -http://www.codeguru.com/csharp/csharp/cs_graphics/sound/article.php/c10931/ but am not at all sure about translating it to vb6...

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by steve6375

Thanks for the reply.
I found this -http://www.codeguru.com/csharp/csharp/cs_graphics/sound/article.php/c10931/ but am not at all sure about translating it to vb6...

Actually, that does not help, but the link within it does: http://www.borg.com/~jglatt/tech/mixer.htm

But I don't have time to learn how to use the API's and make an application for this. Maybe if you find code using the APIs listed there ?

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by rm_03

element 0: d/c offset (whatever that is...)

That tells you where the center of the waveform is, relative to a reference point (usually 0 volts, or ground). A pure AC waveform will have a 0 DC offset, but signals can "ride on" a fixed (or even varying, in which case the DC offset will vary with time) DC level.

@CVMichael:
Look in the AllAPI API Guide - they're in there, with VB6 (and VB.net) code.

Re: Sugestions or Comments about my Sound Tutorial

About Tone Recognition, see this code... Is simple and functional... Easy !

http://www.planet-source-code.com/vb...65667&lngWId=1

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by Robertovo

About Tone Recognition, see this code... Is simple and functional... Easy !

http://www.planet-source-code.com/vb...65667&lngWId=1

rm_03 already posted code for the Goertzel algorithm in this thread at post #18

And if you look at this thread http://www.vbforums.com/showthread.php?t=459138 at post #10, you will see I posted an application that will detect some frequencies

Re: Sugestions or Comments about my Sound Tutorial

It was a nice and good reference on recording sound mix source. but I have found a big bug in your recording source code. That is a runtime error if you record over 2 gbytes, it will show an error of a bad record number.
How can I solve this problem ? I saw that it is an vb automation error.

have a nice day .... and post your reply as soon as possible. plz...^^ :wave:

Re: Sugestions or Comments about my Sound Tutorial

Quote:

Originally Posted by windowbase

It was a nice and good reference on recording sound mix source. but I have found a big bug in your recording source code. That is a runtime error if you record over 2 gbytes, it will show an error of a bad record number.
How can I solve this problem ? I saw that it is an vb automation error.

have a nice day .... and post your reply as soon as possible. plz...^^ :wave:

Welcome to the forums :wave:

About your problem: I would be surprized if it didn't give you an error...

The program is made with VB functions. Internally VB keeps track of file position using a Long data type... A long data type has a maximum of 2^31=2,147,483,648 bytes... that's 2 GBytes.

If you want it to work with more than 2 GBytes, then change all the functions for file processing to API functions.

I don't want to change the code for file processing to API becuase the code is about sound processing, not file processing...

I never would have thought that anyone would record for that long anyways...

If you want to record for that long, then I think it is better to save it encoded to MP3 directly, you can record much longer like that.

For recording directly to MP3, take a look at this thread:
MP3 Encoder with Blade encoder
This project is not using API also, but because it's recording to MP3, you can record up to 37 hours (using encoding 128bps) for a 2 GByte file.

Re: Sugestions or Comments about my Sound Tutorial

Are you trying to create a file larger tha 2GB on A FAT32 volume? If so it won't work - you need an NTFS volume.

Re: Sugestions or Comments about my Sound Tutorial

Only 1,824 views and 41 ( now 42 ) replies since the 17th of this month. Why so unpopular?

Re: Sugestions or Comments about my Sound Tutorial

I guess not many people like to play with sound ?

Re: Sugestions or Comments about my Sound Tutorial

Thanks for this nice tutorial....
Before looking at your tutorial the things like "Wave File Format", Playing wave files and DirectX etc. were alien to me... Now I am very confident in using Directsound and playing streams of incoming PCM Audio data...

Re: Sugestions or Comments about my Sound Tutorial

Thank you, your welcome... too bad I never found time to finish it... (and when I had time, I forgot about it :p )

And welcome to the forums :wave:

Re: Sugestions or Comments about my Sound Tutorial

Hi Michael. It's been about a month since I downloaded Direct_Sound, so this is long overdue.

Great Job!

Yesterday, I read your tutorial. I won't pretend to understand all the code but it will become more apparent as I use and manipulate your code in my own projects. You deserve a ...
Big thumbs up!!!:thumb: :thumb: :thumb::thumb: :thumb: :thumb: :thumb: :thumb:

Re: Sugestions or Comments about my Sound Tutorial

Thank you CDRIVE, I'm glad it helped you.

I wish I had more time to write about all the things I know about sound, but my current job & life style does not give me enough free time for this.

Re: Sugestions or Comments about my Sound Tutorial

Hi CVMichael. Bless you for having gone to the trouble of sharing the intricacies of DirectSound programming with us all. I am well on the way to completing a project based on your explanations and example code, however am stuck on one little thing (well I hope it's little!). In your 'Simple DirectSound' example, you make this statement:

DsBuffer.SetCurrentPosition 0
'You can use the SetCurrentPosition to jump around a sound file as you like; which can be useful

I'd like a user to be able to select where to start sound play from. I'm hoping this can be accomplished via a slider on the form, the length of which always represents 100% of the sound length in the buffer, and wherever the slider knob is positioned, indicates to the user where in the sound duration the sound will start from. To do this, I would need to know how many bytes are in the buffer for the loaded sound, correct?

If so, can you please suggest what procedure, function to use to determine how many bytes are taken by the sound in the buffer, so that instead of Sound(Index).DSBuffer.SetCurrentPosition 0, I can specify Sound(Index).DSBuffer.SetCurrentPosition X, where X is the number of bytes equivalent to where the slider is positioned?

Thankyou so much

Re: Sugestions or Comments about my Sound Tutorial

Michael,

Love the tutorial. Thanks!

I am trying to make a "recording studio" on my PC for my son. I have recorded various sounds (.wav's) from my Yamaha key. What I want to do is create buttons that play them, and the ability to record what's been played. So...I will NOT be recording from an external device, but rather from the sounds that are already part of my app. Any quick tips on how to go about this? Does my question make sense?

Thanks!

Bryce

Re: Sugestions or Comments about my Sound Tutorial

Hi Bryce, welcome to the forums :wave:

Well, there are 3 ways to record that I can think of...

1) is to record the actions, for example, save in a file like this:
time difference from last action
what button was clicked
time difference from last action
what button was clicked
... and so on

When you play back,
wait for amount of time
simulate pressing the button
wait for amount of time
simulate pressing the button
... and so on

This method will create the smallest size of file possible. becuase you are saving only the actions done not the actual sounds.

2) Create the wave file as you go
Well, I don't know how you play the wave sounds right now, so my answer depends on that.
If you send me your project (you already know my e-mail), I can see what is the best way to do this.

3) The way I said before in the e-mail I sent you, simply record...
But I think you have the impression that recording can be done only from outside sources, and that is wrong.

All you have to do is tell Windows / sound card where to record FROM

You do this by going into your sound volume, and change to recording, then choose the source:
See this: http://www.vbforums.com/showpost.php...55&postcount=4

In your case you have to choose either "record master" or "wave"

Re: Sugestions or Comments about my Sound Tutorial

Michael,

Thanks for the reply.

You're right, I was a bit confused (your point #3). Thanks for clarifying that!

I really like your option #1, given the "light" nature of it. I'll give it a go.

Thanks again!

Bryce

Re: Sugestions or Comments about my Sound Tutorial

Hey CVMicheal! :wave:

I don't know if you are like this or not, but whenever I buy a new game or some kind of electronic device, I tend to read/skim the instruction manual (mainly look at the pictures) just to get a feel for what I'm getting involved with.

I have not read your tutorial... or even clicked the link yet. But I'm about to, and after reading this entire thread, I'm very excited to read the tutorial. :)

I'm an electrical engineering student currently starting my junior year and I can see alot of similarities in what I take in my classes (mainly math) and what I have seen in the code examples. I have a special project in mind where I need to detect sound frequencies from a microphone and I think I can actually do it... or at least get a better understanding with this tutorial.

So, just wanted to say hi and let you know that I am greatly looking forward to reading your tutorial. :)

Thanks in advance! Cheers!

- Zero

Re: Sugestions or Comments about my Sound Tutorial

Hi Zero, and welcome to the forums :wave:

Detecting frequencies is not covered in my tutorial, but there is code right here in this thread that can detect frequencies, thanks to rm_03.

You still need to know how to record sound to be able to do it, and that you can learn from my tutorial.

By the way guys, I forgot to say this before, I also posted my sound tutorial on CodeGuru, here:
http://www.codeguru.com/vb/gen/vb_mu...php/c15297__1/

Re: Sugestions or Comments about my Sound Tutorial

Michael,

I'm a bit rusty in VB (been "out of it" for a while now...). I'm having trouble recording the time an event (ie. button click takes place. If I use Now(), I don't seem to get enough precision if the clicks are close together (time doesn't change). Next I tried using a timer setting its interval to 1 and counting the milliseconds between clicks. When I wait two seconds between clicks, it only seems to have incremented to about 300 "counts" (would think it would be 2000?). I must be missing something obvious...

Also, would you happen to have any examples of code to do something like "create a .wav file with a one second long middle C...?"

Thanks for the assist!

Bryce

Re: Sugestions or Comments about my Sound Tutorial

You can use the Timer() to get a more precise value. It's precision is 1000 / 64 = 15.625 milliseconds

If you need even more precision than that, let me know

To create tones, see this:
http://www.vbforums.com/showthread.php?t=356482

Re: Sugestions or Comments about my Sound Tutorial

Although being new to sound engineering so to speak, I am not new to VB and I thought the tutorial was very kool, understandable and helped me quite a bit. I am working with a control from MultiMedia Softlabs Soind recorder and Sound Editor and so far it seems pretty easy to use, but your tutorial helped me to understand more of what it was doing under the covers. I would like to use this control and then your tutorial so that I could read the file in memory byte by byte to get the highest peak values (db level) so that I can normalize the file. If you know anything about this tool would be great.

Re: Sugestions or Comments about my Sound Tutorial

Michael,

I hesitate to post this, since I should be able to figure out the answer by looking through the code you've provided, but...

If I want to change the duration of a "note," what parm/variable in your example(s) would I use to do that? I get a bit confused when looking at "samples per second" and the other variables (ie. "Len")...

Also, do you have anything that shows what frequency each note on a "scale" would be?

Thanks again!

Bryce

Re: Sugestions or Comments about my Sound Tutorial

Actually I don't know what frequency belongs to what note, but searching on Google, i found this, and it looks pretty good:
http://www.lenardaudio.com/education/03_db.html

Yes, the Length parameter of GenerateTone function controls the length of the note/sound, but you need a buffer that can hold the length you give.

For example if your buffer is one second, and you pass the Length for 2 seconds, then of course it won't be able to generate a 2 second sound if the buffer is only one second.

The buffer should be as long as ALL the sounds you plan to generate.

Also, note that the Length parameter is in samples, so then you have to multiply your SamplesPerSec to the time you want to generate the note.

For example if your SamplesPerSec is 44100, and you want to generate a 2.5 second note, then the Length should be 44100 * 2.5 = 110250 (and the buffer should be equal or at least that)

Re: Sugestions or Comments about my Sound Tutorial

Michael,

Thanks once again!

Bryce

Also, the other thing I'm struggling a bit with is 1) generating the tone then 2) playing the tone back immediately (maybe not "immediately..." One button generates, another button plays...). I don't have any issues with generating, then saving, then finding the file I generated and playing it with Windows Media Player, for example, but when I try to play it from the same VB app in which I generate it, I'm not getting any playback.

That may be a little fuzzy, but any thoughts?