Advanced Home Theater Audio Terminology
The purpose of this section is to provide the more advanced user a little more detail about the various audio terms you will encounter (Thanks Chris). This is a work in progress so please keep checking back.
DTS or Digital Theater Sound has been around since about 1972 when Terry Beard and Jim Ketcham developed a sound format in conjunction with TODD-AO. The format as used in theaters is a CD-ROM containing the audio tracks held in synch with the film by means of a code embedded in the film. The first practical use of the format was in 1992 with Jurassic Park. The theater specification are 1.235Mbit/sec of the CD's bandwidth of 1.411Mbits/sec. Compression is 2.9:1 fir 16 bits and 4.3:1 for 24 bits. For use in DVDs the specs are 1.509Mbits/sec for the 48 kHz sampling. There is also a 754 kbits/sec used where space is tight. Redundant information in the audio tracks is compressed using a codec similar to computer compression programs like PKZip. This greatly reduces the size required for the audio track. The compression with DTS is less than that of the similar Dolby digital format so DTS tracks require more of the DVD's bandwidth that Dolby.
In DTS's case, a technique called "forward-adaptive bit allocation" is used. Using this technique, the allocation of data to each sub-band is predetermined exclusively by the encoder. This information is explicitly conveyed to the decoder along with the actual bits to be used. Forward-adaptive bit allocation's primary advantage is that the psychoacoustic model used resides exclusively within the encoder. Because the model is encoder-based, extremely complex psychoacoustic coding algorithms can be used (as decoder processing ability isn't a limiting factor). Forward-adaptive bit allocation also allows psychoacoustic model modifications and improvements to be passed directly on to installed decoders, essentially "future-proofing" DTS decoders from premature obsolescence.
Pulse-code modulation: Modulation in which a signal is sampled, and the magnitude (with respect to a fixed reference) of each sample is quantized and digitized for transmission over a common transmission medium. In conventional PCM, before being digitized, the analog data may be processed (e.g., compressed), but once digitized, the PCM signal is not subjected to further processing (e.g., digital compaction) before being multiplexed into the aggregate data stream. PCM pulse trains may be interleaved with pulse trains from other channels.
Each pulse of a pulse train takes the standard value nearest its actual value when modulated. The modulating wave can be faithfully reproduced. The amplitude range has been divided into 5 standard values. Each pulse is given whatever standard value is nearest its actual instantaneous value. The same amplitude range has been divided into 10 standard levels. The curve is a much closer approximation of the modulating wave, than is the 5-level quantized curve. The greater the number of standard levels used, the more closely the quantized wave approximates the original. This is also made evident by the fact that an infinite number of standard levels exactly duplicates the conditions of nonquantization (the original analog waveform).