Some basic audio questions

[mito2005]

I couldn't find the answers to these simple questions using the search engine or the Audio FAQ.

1. Are DTS files encoded at 48KHz?
2. What does the word "discrete" mean in the surround context? (5.1 discrete, for instance)
3. My receiver displays DD when playing dtswav files. Is it converting them?


More questions to follow. Thank you for your time answering these newbie questions.


:search:

[daphy]

1. Are DTS files encoded at 48KHz?
sometimes, if a DVD-Video is your source
2. What does the word "discrete" mean in the surround context? (5.1 discrete, for instance)
there are 6 single channels encoded inside the tracks which could be seperated by a decoder (your amp for example) while replaying it - a f.e. surround Prologic encoded track is 2.0 track which includes other channels in matrixes (~> mixed in) not standalone.
3. My receiver displays DD when playing dtswav files. Is it converting them?
What´s the use of double postings :thanks:

[ursamtl]

1. Are DTS files encoded at 48KHz?
Not always. For example, DTSWAV files for creating surround CDs must be encoded at 44.1kHz to be recognized as audio CDs by DVD players. DTS files must be encoded at 48kHz or 96 kHz to be recognized as DTS soundtracks accompanying video on DVDs.

2. What does the word "discrete" mean in the surround context? (5.1 discrete, for instance)
Discrete means they are completely separate, autonomous audio channels that need no extraction or "de-matrixing" to be heard. Dolby Digital 5.1 provides up to 6 discrete channels. Dolby surround, Dolby Pro Logic, and Dolby Pro Logic II do not provide discrete channels, but rather combine the audio information together into 2 channels. The extra channels are then derived by manipulating the audio information in the 2 channels to extract something resembling the original channels that were matrixed.

As for your third question, sorry I have no idea why your receiver is displaying the DD. Check your manual. I doubt very much that it's actually "converting" the data from DTS to DD.

Regards,
Steve.

[mito2005]

Another batch of questions (perhaps these could be included in the Audio FAQ):

1. Do you notice quality degradation If DTS is converted from 48KHz to 44.1KHz?
2. What about quality difference between 96KHz and 48KHz?
3. Are the DTS files I downloaded in 44.1 or 48khz?


Thanks. :thanks:

[daphy]

3. Are the DTS files I downloaded in 44.1 or 48khz?
They are surely illegal ;) but if they have a WAV extension they are at 44.1KHz -> a DTS extension means 48-96 KHz (honestly I never saw a 96KHz in my life :rolleyes: )
2. What about quality difference between 96KHz and 48KHz?
some say it´s better because it has a higher sampling rate - if your home equipment ever proof this, it´s another question. More interesting might be the question: Is the quality of a 16bit ->20,24bit DTS better?
1. Do you notice quality degradation If DTS is converted from 48KHz to 44.1KHz?
In my opinon and a age over 30 -> it´s philosophy, importent are two things:
- you will never get a better sound if you decode the lossy format for upsampling and reencoding
- upsampling a uncompressed WAV also doesn´t improve the quality :p

[ursamtl]

As daphy pointed out, upsampling will not improve quality. In other words, it will not add any audio information that wasn't already there. A sampling rate's most obvious effect is on high-frequency response. The higher the sampling rate, the higher the maximum high frequency that can be recorded. A basic rule of thumb is that the maximum frequency is about half the sampling rate. So, if your source file has no audio information above, for example, 16kHz (which falls well below a maximum of 22kHz, or half a 44.1kHz sampling rate), then upsampling to 48kHz and raising the "frequency ceiling" to 24 kHz is not going to add any audio information above 16kHz. It just adds space that would have been used had the audio information been available (and a dog in the same room so he could hear it ;).

However, there are some points to consider. Some computer soundcards are optimized for operation at 48kHz (Creative Labs Soundblaster and Audigy cards are well known for this, but there are others). Their chips resample the audio to 48kHz, which can introduce intermodulation distortion. Software resampling with an accurate resampling program such as ssrc can do a much better job. Even on a friend's PC with a onboard sound (SoundMax), the difference between original 44.1kHz audio files and the same files resampled with ssrc_hp to 48kHz is clearly audible, especially through a decent set of headphones. High frequencies such as cymbals seem more open and detailed. Reverb tails on these sounds tend to sound more natural and open. This being said, the actual sound data was already in the 44.1kHz file; it was simply a matter of playing it back properly. I find playing back audio files with Foobar2000 loaded with the ssrc resampling DSP gives me the best sound on my computer. Other programs such as WinAmp have similar resampling plugins, etc.

The main reason for permanently altering an audio file would be if you intend to process the files in any way afterwards. Increasing the sampling rate and, more importantly, bit resolution gives you a lot more data to work with if you want to use digital signal processing such as reverb, conversion to surround, compression, equalization, etc. Of course it takes more processing power to do this, but the results will be better. The original sound data might not be improved, but any reverb or new sound data created by the process will be in a higher resolution.