Converting audio samples into an audible file format

Question

I've been trying to convert a huge amount of data (about 900 MB) to an audible file format for a few days now. I've been given a .dat file containing 900 millions floating-point samples (one per line) representing 90 seconds of music at 10 MHz.

I downsampled to 40 KHz but now I don't know how I could possibly listen to the audio hidden in those bytes. I'm writing a C++ program in a Linux environment but if any one knows how to accomplish this task using Matlab, Octave, Python, Audacity, MPlayer or any other tool, please come forth and speak :) Contributions in any amount are greatly appreciated.

    head -n 5 ~/input.dat
    -2.4167
    -7.5322e-016
    -0.2283
     0.13581
    -0.51926

Thank you all guys. Wish I could vote for your answers :( – mr_hyde Apr 02 '13 at 21:12 — mr_hyde, Apr 02 '13 at 21:12

score 1 · Answer 1 · answered Apr 02 '13 at 17:56

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The wav file format is a rather simple one.

You just need to write the 44 byte header block defined in that link, followed by your data converted to integers.

answered Apr 02 '13 at 17:56

Drew Dormann

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score 1 · Accepted Answer · edited May 23 '17 at 12:29

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Target your sample rate to 44100 hz (or 48000, 22050, 11025, or 8000 hz)

Convert your audio samples to 16-bit signed integers (-32768 to +32767).

Follow the instructions on WAV file synthesis here: WAV File Synthesis From Scratch - C

edited May 23 '17 at 12:29

Community

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answered Apr 02 '13 at 17:58

selbie

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score 1 · Answer 3 · answered Apr 02 '13 at 18:30

If you have a sequence of bytes and want to convert it to audio, all you need to do is write a header to it. Since you mentioned that you can use MatLAB, I would recommend wavwrite command. It is simple, tried and tested and excellent for prototyping. Here is the link to the documentation:
http://www.mathworks.in/help/matlab/ref/wavwrite.html

Here are some steps you may need to take in case you are using wavwrite.
- Since your input data is floating point, scale the data in your file to within a range of [-1, 1].
- Once data is scaled plug and chug into the function call.
- Play the wav file using wavplay command.

score 1 · Answer 4 · answered Dec 21 '14 at 00:26

WAV allows for floating-point sample data and a wide range of sample-rates (1Hz to 4.2GHz in 1Hz increments, if memory serves).

You don't need to bother with converting to integer values. Just set up the WAV file's header appropriately and write 32-bit floats as binary data in the data section.

From a storage perspective, the 10MHz sample-rate is no problem for a WAV file. Playback, however, will require conversion to something the hardware can handle. The upper limit these days is typically 96 or 192 kHz.

Converting audio samples into an audible file format

4 Answers4