These simple programs convert raw AC3 (Dolby Digital, A/52) and DTS streams into WAV files containing the S/PDIF fake PCM encoding thereof. This means that if such a .wav file is replayed correctly bit-by-bit via the S/PDIF output of a device and supplied to a standards compliant decoder, it will be recognized as a standards compliant AC3/DTS stream, indistinguishable from a stream generated e.g. by a DVD player.
This is useful to e.g. author CDs containing compressed multichannel data (such as DTS encoded CDs that commercially available). Take care: Playing back the files produced via speakers at high volume may damage these or your hearing.
These sources are very old (from around 2006/2007) and show my coding abilities back then when I started with C. I hope to clean this up and modernize it at some point in the future.
This tool takes a raw .ac3 file and converts it into a .wav file of the same sample rate containing one AC3 frame plus padding zeros for the remaining time of that frame's time slot.
In a similar fashion, this tool translates a .dts file to a .wav file. Currently it only supports 16 bit encapsulation, i.e. the data stream is included in the .wav file 1:1. The DTS specification also defines a 14 bit format for use with CDs where there is a possibility that the data will be played back via speakers in order to reduce the possibility of speaker/hearing damage when the medium is played back without a DTS decoder. In this case, only the lower 14 bits of a word would be used, with the high 2 bits set to zero. This encoding may be added to dts2spdif later on.
As usual, this code comes with no warranty whatsoever. I am especially not responsible if you blow up your speakers or ears using these tools.
Do whatever you like with this code, but if you use it for your own application I would appreciate an attribution note in the documentation. Thanks!
-- norly
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