Using winmm.lib to decode real time LTC input?

[antithing]

Hi, and thank you very much for this code. I am streaming in LTC from a master clock, and am attempting to pick it up using winmm.lib, which streams the signal into a:

short int waveIn[44100 * 3];

Am i correct in thinking that libltc decoding needs an unsigned char* ?

Can I use something similar to the below code, or do i need a queue of some kind for real time streaming?

The signal is checked and is valid ltc.

Thanks!

void Start()
{
	const int NUMPTS = 44100 * 3;   // 3 seconds
	int sampleRate = 44100;

	HWAVEIN      hWaveIn;
	MMRESULT result;

	WAVEFORMATEX pFormat;
	pFormat.wFormatTag = WAVE_FORMAT_PCM;     // simple, uncompressed format
	pFormat.nChannels = 1;                    //  1=mono, 2=stereo
	pFormat.nSamplesPerSec = sampleRate;      // 44100
	pFormat.nAvgBytesPerSec = sampleRate * 2;   // = nSamplesPerSec * n.Channels *    wBitsPerSample/8
	pFormat.nBlockAlign = 2;                  // = n.Channels * wBitsPerSample/8
	pFormat.wBitsPerSample = 16;              //  16 for high quality, 8 for telephone-grade
	pFormat.cbSize = 0;

	// Specify recording parameters
	result = waveInOpen(&hWaveIn, WAVE_MAPPER, &pFormat,
		0L, 0L, WAVE_FORMAT_DIRECT);

	WAVEHDR      WaveInHdr;
	WaveInHdr.lpData = (LPSTR)waveIn;
	WaveInHdr.dwBufferLength = NUMPTS * 2;
	WaveInHdr.dwBytesRecorded = 0;
	WaveInHdr.dwUser = 0L;
	WaveInHdr.dwFlags = 0L;
	WaveInHdr.dwLoops = 0L;
	waveInPrepareHeader(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));

	// Insert a wave input buffer
	result = waveInAddBuffer(hWaveIn, &WaveInHdr, sizeof(WAVEHDR));

	// decode stuff
	int apv = 1920;
	ltcsnd_sample_t sound[BUFFER_SIZE];
	size_t n =0;
	long int total =0;
	LTCDecoder *decoder;
	LTCFrameExt frame;
	decoder = ltc_decoder_create(apv, 32);

	// Commence sampling input
	result = waveInStart(hWaveIn);

	int num = 0;
	while (num < 200)
	{
		cout << "recording..." << endl;

		sound[0] = waveIn[0]; // this is wrong. Do I need to convert to unsigned char?

		ltc_decoder_write(decoder, sound, n, total); // do i need to write before i read?
		
                ltc_decoder_read(decoder, &frame);
		SMPTETimecode stime;
		ltc_frame_to_time(&stime, &frame.ltc, 1);

		std::cout << stime.frame << std::endl;		  // prints a strange character	
		
		num += 1;
	}
	
	waveInClose(hWaveIn);
	ltc_decoder_free(decoder);
	PlayRecord();
}

[x42]

Perhaps try using 8bit directly (instead of 16)

wBitsPerSample = 16;              //  16 for high quality, 8 for telephone-grade

Note you'll also have to set nAvgBytesPerSec = nSamplesPerSec in that case.

Alternatively sound[0] = waveIn[0] / 256 but I don't know if winmm produces signed or unsigned data.
You'll also have to convert sound[0]... sound[total -1] , not just the first element of the data-buffer.

[antithing]

Thank you. I will try that.
What is the best way to copy the right amount of data from waveIn to sound?
Sound = waveIn gives an error, do I need to do a for loop for the max buffer size? (waveIn[44100 * 3];)
Thanks again!

[antithing]

Hi, sorry to bug you again, I am still stuck on this. i have adjusted my main loop above to:

while (num < 200)
	{
		cout << "recording..." << endl;

		for (unsigned int i = 0; i < BUFFER_SIZE; ++i) {
			
			sound[i] = waveIn[i] / 256;
			ltc_decoder_write(decoder, sound, n, total);
		}
				
	    ltc_decoder_read(decoder, &frame);
	    SMPTETimecode stime;


		ltc_frame_to_time(&stime, &frame.ltc, 1);

		std::cout << stime.frame << std::endl;			
		
		num += 1;
	}

But it just prints an empty line. Where am i going wrong? Thanks again.

[conducive2]

@antithing Hello, have you solved this problem? I also encountered similar problems.