Digital audio formats convertion (AC3/DTS/AAC/etc.)

[DnA]

Hi folks,
do you have any idea how to convert files among these digital audio formats without losing the additional information (LFE channel, DRC, voice normalization level, etc.)?

[daphy]

more details plz - I guess none knows what you mean!

[DnA]

What I mean is there are specific attributes for the digital audio multichannel formats like AC3 and DTS.
For example if you want to convert DTS file to Dolby Digital you should also keep audio stream information like Dynamic Rage Compression, Dialog normalization, etc. If you try to convert the file (DTS->AC3) via 6 .WAV files this information will be lost.
I hope this information gives more clarity to the problem.

[Rockaria]

Sure I hear the AC3 contains some parameters for the 'Dolby-certified decoding' such as Dialog Normalization, center level, surround level and the DRC as you mentioned.
But as for the DTS, I've heard of nothing like that( I could be wrong here).

Anyway, there are two process steps in the transcoding : decoding(to a wave buffer) and encoding(from the wave buffer).

If the decoder is not broken(i.e. Dolby certified...), it will apply all the parameters to the wave spectrum.
So the encoding from the buffer, is another (artificial) job to make it close to the original(or proper for the purposes).

[DnA]

I agree that some of the information will be preserved and properly decoded. But for example, the DRC goal is to be set in the audio decoder (amplifier, software decoder, etc.) by the user while watching a movie. Once the audio file is converted (via PCM stream), this information is lost - you will have multi channel audio with the DRC settings from the convertion. Therefore I think that it would be convenient, if after converting among AC3, AAC and DTS this information is kept. ;)

[Rockaria]

I can't agree with you more.:)
Yes, DRC and maybe some other parameters are play(decoding)time options, the flexibility of which will be lost by the transcoding.