Sony doesn't make DVD-A/V players, unless they just started making them. SACD was/is a competitor to DVD-A, so Sony was pushing SACD exclusively. If your player is a DVD-V/SACD/CD player, then it is likely to have a 1 bit/high sampling rate Digital to Analog Converter only, because SACD is a 1 bit system. A lot of conventional CD players also use a 1 bit sigma delta oversampling DA converter as well, even though CDs are 16 bit. If you are running analog outputs from your disc player to your receiver, then upsampling 16/44.1 > 16/48 will most likely improve the sound quality, because, no offesnse, your DA converter is less than ideal. I spent a lot of money on an SACD player and about 20 SACDs myself.
A multibit sigma-delta oversampling DA converter is much better and is found in higher end CD players and most DVD-A/V players (I think, at least the higher end ones). Companies use 1 bit converters because they are cheaper than multibit ones.
If you are sending a digital signal from your DVD player to your receiver or preamp/processor, then it depends what kind of DA converter is in that component.
It would be a good idea to burn a CD of a show and then burn an audio only 16/48 DVD-video with Audio DVD creator. Then do a listening test and decide what is best for you. Here is the thing, though: If you ever upgrade to a Blu-Ray/DVD-Audio/Video player with a true multibit converter, then those upsampled 16/48 audio only DVD-Vs will sound WORSE than CDs because you now have a great DA converter. Upsampled files will no longer sound better. In fact, upsampled files will sound worse unless they are a direct integer multiple (2x, 3x, 4x, etc.) of the original file:
"When changing the sampling rate, it is better to maintain an integer multiple of the original signal's sample rate, so the processing is kept simple. More importantly, the end result is more accurate, thus enabling a higher fidelity of sound reproduction. A two times (2x) oversampling system will double the sampling rate, by adding one easy to find numerical value in between each actual sample. For example, when a 44.1kHz digital signal is processed, a 88.2kHz digital signal is obtained. It is simple, effective and precise because it is a direct multiple of the original digital signal. For an upsampler to make a 96kHz digital signal from a 44.1kHz signal, it will have to perform awkward mathematical operations to obtain a 96kHz signal. (96kHz / 44.1KHz equals 2.1768707…). This results in a less accurate output from the digital filter, with everything else following (i.e. digital-to-analog conversion and analog filtering) also being less accurate. As well, exactly like oversampling, the artificially higher sampling frequency created by an upsampler doesn't increase the actual frequency response of the system, but simply increases the lower limit of the frequencies that need to be eliminated."
So it would be a waste of a lot of discs (not THAT much money, I know, but it adds up after a while) to convert 16/44.1 to 16/48 if you ever get a Blu-Ray/DVD-A/V player with a multibit converter, or a receiver or pre/pro with multibit converters. You can burn a DVD-A disc and keep the files at 16/44.1.
I read that there is a great free program to resample, it is supposed to be better than Audacity, I'll look around for it. Wavelab is better than Soundforge from what I've read. I think foobar2000 will perform upsamlping to 48kHz on the fly, which improves the sound on my crappy soundcard/DA converter when listening through headphones. I'll have to do some listening tests and see how foobar2000 compares to Wavelab's upsampling.