How to parse audio data into LTC (linear timecode) string representation

Using Cinder
1

Hi!

Does anyone know how to use properly this library (GitHub - x42/libltc: Linear/Logitudinal Time Code (LTC) Library) for parsing audio data from audio input node and converting it to timecode?
For some reason timecode, in the form of “hours:minutes:seconds:frames”, is printed correct after each 2-3 seconds, while in between wrong values are printed.
I am using mobile app (https://play.google.com/store/apps/details?id=com.AaronBernstein.ltctimecodegeneratorpro&hl=sr__%23Latn&gl=US) for timecode generation and jack input for sending data to my cinder app.
I followed each step in libltc/tests/ltcdecode.c. The only difference is that in the example a binary file is used for reading bytes of audio data, while I am using cinder audio buffer (floats data) for the same purpose.

My implementation is the following:

				const float* data = mMonitorNode->getBuffer().getData();
				float* nonConstData = const_cast<float*>(data);
				ltc_decoder_write_float(ltcDecoder, nonConstData, mMonitorNode->getBuffer().getSize(), total);
				while (ltc_decoder_read(ltcDecoder, &frame)) {
					SMPTETimecode stime;
					ltc_frame_to_time(&stime, &frame.ltc, 1);
					timecode.hour = stime.hours;
					timecode.min = stime.mins;
					timecode.sec = stime.secs;
					timecode.frame = stime.frame;
				}
				gui::LabelText("LTC (H:M:S:F)", "%.2d:%.2d:%.2d:%.2d", timecode.hour, timecode.min, timecode.sec, timecode.frame);
				total += mMonitorNode->getBuffer().getSize();

Thanks in advance,
Dusan

2

Hi all,

I would like to ask more specific question regarding this problem:
Does anyone know how cinder audio buffer transforms raw bytes of audio data (e.g. from .wav file) to float representation?

3

Hi,

Not sure about your LTC needs (haven’t needed to use that myself), but concerning your question around audio buffer raw bytes → floats (and assuming the audio isn’t stored in your file/source as floats since that doesn’t take any transformation): if the platform you’re on has built-in conversion support (eg iOS and OSX), we use that. Otherwise a ci::audio::dsp::Converter is used, with the most common implementation being with r8brain (here). You can see how that’s done in the audio::Source api, or looking at one of the platform specific implementations of that.

Hope that helps,
Rich

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