ILMAUserCallback

Implement this user callback interface in order to receive audio buffers. Register the callback by setting the either the ILMACallback::CallbackObj or the ILMACallback::CallbackObj2 property. Each interface has one method, which is called by the LEAD Audio Callback Filter whenever the filter receives an audio buffer.

DirectShow determines the duration of the audio buffer. However, if you use the LEADTOOLS Multimedia toolkit, you can change the size of the audio buffers using the get_AudioBufferSize property.

Use this interface with VB (6 or earlier) or other programming languages that cannot deal with pointers. This interface does not work well in .NET. For .NET programming, use the ILMAUserCallback2 (uses a pointer to the data, cast as a 32-bit long value) or ILMAUserCallback3 (uses a pointer to the data, cast as a 64-bit LONGLONG value.) interface instead.

Data types:

AudCalBkErrConstants:

typedef enum  
{ 
   AUDCALBK_ERR_OK, 
   AUDCALBK_ERR_DROP, 
   AUDCALBK_ERR_DELIVERLASTSAMPLE, 
} AudCalBkErrConstants; 

Lists the possible error code values that can be returned by the CallbackProc method.

Interface Methods:

HRESULT CallbackProc (VARIANT* pData, long lDataSize, long lBitsPerSample, long lChannels, long lSamplesPerSec, long lAvgBytesPerSec);

This function will be called each time an audio sample is received by the filter.

Parameters

Returns

Comments

Format of audio data:

The general format of the audio data (lChannels) is as follows:

Sample 0

In this case,

pData(0) is Sample 0, Channel 0   
pData(1) is Sample 0, Channel 1 
pData(lChannels - 1) is Sample 0, Channel lChannels -1   
pData(lChannels) is Sample 1, Channel 0   
pData(lChannels + 1) is Sample 1, Channel 1 

The format for mono data is simple, because there is only one channel. In this case, every value in the array is one sample:

pData(0) is Sample 0   
pData(1) is Sample 1 

The format for stereo data is still simple: there are two values per sample (one for left channel and the other for right channel):

pData(0) is Sample 0, Left channel   
pData(1) is Sample 0, Right channel   
pData(2) is Sample 1, Left channel   
pData(3) is Sample 1, Right channel 

The format of the value is different depending on whether lBitsPerChannel is 8 or 16.

Format of 8-bit audio data:

The values in the array are unsigned and between 0 and 255. The real audio value will be obtained by subtracting 128 from the array value and should have a value between -128 and 127. If you want to change the value, you must remember to add 128 before putting back the data.

Here is some Visual Basic code that doubles the intensity of the sound for pData(i):

Dim val As Long ' use a signed data type 
val = pData(i) - 128 ' convert pData(i) to long. val is now the audio value, in the 0..255 range 
val = val * 2 - double the sound intensity 
' convert the value to signed 8-bit 
val = pData(i) - 128 
' double the audio value 
val = val * 2 
' clip the output to -128..127 
If val > 127 Then 
   val = 127 
ElseIf val < -128 Then 
   val = -128 
End If 
' write the data out, adding back 128 
pData(i) = val + 128 

Note how the real audio value was clipped to the -128 ..127 range before adding 128. This is important, otherwise distortions are introduced.

Format of 16-bit data:

The values in the array are signed and between -32768 and 32767. They contain the real audio value and it is not necessary to do the conversions like for the 8-bit data.

Here is some Visual Basic code that doubles the intensity of the sound for pData(i):

' use a signed data type. Use long (32-bit in VB) instead of Integer (16-bit in VB) to avoid overflows when multiplying by 2 
Dim val As Long 
val = pData(i) 
val = val * 2 
' clip the output to --32768..32767 
If val > 32767 Then 
   val = 32767 
ElseIf (val < -32768) Then 
   val = -32768 
End If 
pData(i) = val