How to convert audio byte to samples

Question

This is my struct

/* wave data block header */
typedef struct wavehdr_tag {
    LPSTR       lpData;                 /* pointer to locked data buffer */
    DWORD       dwBufferLength;         /* length of data buffer */
    DWORD       dwBytesRecorded;        /* used for input only */
    DWORD_PTR   dwUser;                 /* for client's use */
    DWORD       dwFlags;                /* assorted flags (see defines) */
    DWORD       dwLoops;                /* loop control counter */
    struct wavehdr_tag FAR *lpNext;     /* reserved for driver */
    DWORD_PTR   reserved;               /* reserved for driver */
} WAVEHDR, *PWAVEHDR, NEAR *NPWAVEHDR, FAR *LPWAVEHDR;

I have this variable WAVEHDR waveHeader;

I record 10 secs from microphone and waveHeader->lpData has my raw recorded data, and waveHeader->dwBytesRecorded is the raw data's length Now I want to calculate the volume in each second to say which second has highest volume and which one has the lowest.

I know I should sum the absolute values and divide by the number of samples

I used sum += abs(waveHeader->lpData[i]); for i from 0 to length of one secs data, but it doesn't give me a good result

it always gives me the same result for each second, but I am silent in some seconds and speak in some...

I read I have to add samples, not bytes How should I convert waveHeader->lpData[i] to samples?

//len = length of one secs data (waveHeader->dwBytesRecorded/10)
for (int i=0; i<len; i++)
{
    sum += abs(waveHeader->lpData[i]);
}

Kirshbaum

The WAVEFORMATEX that you passed to waveInOpen gives you nChannels and wBitsPerSample. Multiple the two together and divide by 8 and that's the number of bytes per sample. – Anxious

@JonathanPotter nChannels = 2 and wBitsPerSample = 16 (2*16)/8 is 4 so each sample has 4 bytes, yes? Now what should I do to sum the absolute values and divide by the number of samples? I got confused :( – Kirshbaum

For 4-byte samples, you would cast lpData to a DWORD*. But you probably want to handle the two channels individually (i.e. they're not actually 4 byte samples, they're 2x2 byte samples) so you could cast it to a WORD* and then calculate the average for each channel. – Anxious

@JonathanPotter both channel are the same, so if I calculate the average for one channel it's enough, I think I misunderstood something, because in any way I calculate the average, the result is unusable / I add some code in the question. – Kirshbaum

You're not casting the pointer in that code, all you're doing is casting individual bytes to words. – Anxious

By the way, what you trying to implement? Are you trying to render a "peak meter" or some visual indicator of volume on the screen? Or are you trying to do some other sort of analysis of the signal? The reason why I ask, is if you are trying get a visual rendering of the signal, there's lots of easy cheats you can apply to make this easy without having to do sophisticated signal processing. – Seigniory

Perhaps you should attach a sample WAV file you record. Specifically, WAVEHDR is a ell known structure on its own but no one knows what format (values) you are using and whether the entire WAV you create is valid. – Neology

@RomanR. Thanks for the time you left, I succeeded miraculously – Kirshbaum

Answer 1

You have the WAVEFORMATEX used for capturing the audio, right? If so, you can modify the following routine to meet your needs:

void ProcessSamples(WAVEHDR* header, WAVEFORMATEX* format)
{
    BYTE* pData = (BYTE*)(header->data);
    DWORD dwNumSamples = header->dwBytesRecorded / format->nBlockAlign;

    // 16-bit stereo, the most common format
    if ((format->wBitsPerSample == 16) && (format->nChannels == 2))
    {
        for (DWORD index = 0; index < dwNumSamples; index++)
        {
            short left = *(short*)pData; pData+=2;
            short right = *(short*)pData; pData+=2;
        }
    }
    else if ((format->wBitsPerSample == 16) && (format->nChannels == 1))
    {
        for (DWORD index = 0; index < dwNumSamples; index++)
        {
            short monoSample = *(short*)pData; pData+=2;
        }
    }
    else if ((format->wBitsPerSample == 8) && (format->nChannels == 2))
    {
        // 8-bit samples are unsigned.
        // "128" is the median silent value
        // normalize to a "signed" value
        for (DWORD index = 0; index < dwNumSamples; index++)
        {
            signed char left = (*(signed char*)pData) - 128; pData += 1;
            signed char right = (*(signed char*)pData) - 128; pData += 1;
        }
    }
    else if ((format->wBitsPerSample == 8) && (format->nChannels == 1))
    {
        for (DWORD index = 0; index < dwNumSamples; index++)
        {
            signed char monosample = (*(signed char*)pData) - 128; pData += 1;
        }
    }
}

Seigniory

hi, I search and find your answer and try to use your code in my project but I have a problem, this is my question my problem is I don't know how to memcpy_s the data? and I don't sure is my problem because of samples and bytes or is something else..? I'm new in voice, please help me in my project – Colchicum