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Implementing DoProcessOutput

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[The feature associated with this page, Windows Media Player SDK, is a legacy feature. It has been superseded by MediaPlayer. MediaPlayer has been optimized for Windows 10 and Windows 11. Microsoft strongly recommends that new code use MediaPlayer instead of Windows Media Player SDK, when possible. Microsoft suggests that existing code that uses the legacy APIs be rewritten to use the new APIs if possible.]

To process audio data, you'll need to perform several steps in DoProcessOutput. The following steps use the plug-in wizard sample code as examples. If you want to create an audio DSP plug-in that processes media content of the same type that the plug-in wizard sample code does, you'll only need to change the actual processing code referred to in step 6. Following are all the steps implemented in DoProcessOutput:

  • If the plug-in is not currently enabled, simply copy the data unchanged into the output buffer. If your plug-in converts the data into a different format, you must also do the conversion processing here.

    // Test whether the plug-in is disabled by the user.
if (!m_bEnabled)
{
    // Just copy the data without changing it.
    memcpy(pbOutputData, m_pbInputData, *cbBytesProcessed);

    return S_OK;
}
    • Retrieve a pointer to the input format structure. You'll need to retrieve member data from this structure, so copy the pointer from m_mtInput.pbformat to a local pointer variable of a type that matches the format structure type. The following example stores a pointer to a WAVEFORMATEX input format structure:
    WAVEFORMATEX *pWave = (WAVEFORMATEX*) m_mtInput.pbFormat;
    • Calculate the number of samples to process. The sample code that the plug-in wizard generates performs this step by dividing the number of bytes to process by the nBlockAlign member of the WAVEFORMATEX input format structure, and then multiplying the result by the number of channels, which was stored in the nChannels member. The following example is from the plug-in wizard sample code:
    DWORD dwSamplesToProcess = (cbBytesProcessed / pWave->nBlockAlign) * pWave->nChannels;
    1. Determine the bit depth of the audio. The plug-in wizard sample code determines 8-bit or 16-bit audio by inspecting the wBitsPerSample member of the WAVEFORMATEX structure. It then uses that value in a switch statement to provide separate processing routines for each bit depth. You may need to use a different technique when dealing with other format types and bit depths.
    2. Create a loop to step through the audio samples in the input buffer.
    3. Retrieve a sample from the input buffer. You do this by dereferencing the input data pointer and storing the result in a variable of type int. For 16-bit audio, you must recast the BYTE pointer to a short pointer to handle the greater audio sample precision. Once you have the value, you can immediately increment the pbInputData pointer so that it points to the next sample. The following examples demonstrate this:
    // For 8-bit audio.
int i = *pbInputData++;

    -or-

    // For 16-bit audio.
// Recast the pointer.
short *pwInputData = (short *) pbInputData; 

// Enter the loop and then get the input sample.
int i = *pwInputData++;
    1. Perform the processing. This is where you apply the algorithms that change the sample somehow. What you do here is up to you.
    2. Write the processed data to the output buffer. Immediately increment the pointer to the output buffer, as in the following example:
    *pwOutputData++ = i;
    1. Repeat the loop until all the samples have been processed.
    2. Return an appropriate HRESULT.

Implementing an Audio DSP Plug-in