Mary, at the risk of repeating what others have probably said more
coherently, allow me.
All sound is analog by nature, it slides continually from soft to loud, from low to high frequency, like sliding down a slide. Now, imagine that slide becomes a set of stairs, that is what happens in the conversion of the analog sound to a digital signal.
Sample size is the actual number of bits that are in each sample of the audio signal. 16 bits is pretty standard, 24 is better. It is like touching corduroy fabric. If it is a wide wale, your finger will only touch a few rows at a time. If it is narrow, your finger will touch more rows. If it is really superfine wale, you will touch many rows with just a fingertip.
In the same way, the larger the sample and the faster the sample rate, the more of the sound will be captured exactly as it sounded originally.
Your cd collection is all done at 16 bits per sample and at a sample rate of 44,100 per second. There is a law, if you will, that says that the highest frequency you can record is one half the sample rate, it is called the Nyquist frequency. That means that at a rate of 44,100 samples a second, the highest frequency sound you can record is just over 22,000 hertz, which is adequate for most things.
The size of the sample determines how many different sound levels are possible. A 16 bit sample has 2 to the 16th power different sound levels. A 24 bit sample has many more. This means, in practical terms, that the larger the sample size, the lower the noise and more detail will be picked up. For most purposes, a 16 bit sample is adequate.
The bit rate is different. It has to do with the compressing that takes place when turning a pcm file, oops, I mean a wav file, into a compressed format like mp3. A bit rate is expressed in thousands of bits per second, so 128 kbps means 128,000 bits per second. While this may seem fast, it is about one tenth of what pure digitized audio at cd quality would consist of.
The higher the bit rate, the more lifelike the sound will be. Music at 128 kbps, a common mp3 compression, sounds fairly good until you hear drums, cymbals and other high frequency sounds. They end up being injured by the compression that takes place. MP3 compression is called a lossy compression, meaning that some data is lost forever when the audio is compressed. This is why turning it back into a wav file does not make it sound any better.
So, the ds-2 recording at a sample rate of 44,100 samples per second and a sample size of 16 bits gives you a nearly cd quality master file. When this is compressed to 128 kbps, you do lose some quality, but you also have a file that is only one tenth the space of the master file.
I do not know, but the ds-2 may do the mp3 encoding on the fly, or maybe not. In either case, given the specs, it is capable of producing a decent sounding audio file. I figure that a half decent external microphone will contribute to the quality as they probably did not put high end mikes in an inexpensive digital recorder.
Hope this helps. If I simply confused things too much, you may have my ration of chocolate for an hour...well, maybe for half an hour.