I’ve been recording audio streams from my music player of choice using the X-Fi’s Audio Creation Mode’s recorder as it works well for my purposes. The streams come in @ 192, 160, & 128kps MP3 format depending on the internet station. I record the streams in WAV format (1411kps) so that I can later molest them (trim, mix, reverb, echo, etc.) in whatever way suits me @ the time and re-save them w/out losing much, if any, quality. I assumed that if I saved them in any lossy compression format I’d lose some quality after I worked on the files and resaved them. Make sense? Finally, I compress the finished (or untouched) files to AAC for personal listening.

My questions are (1) whether or not it makes any sense to do that final compression @ higher than 192kps AAC since that’s highest bitrate the source comes in as. When I used lossless encoding (for highest possible quality) the bitrate is never less than 1000kps which surprises me since the bitrate of the source is only 192kps at its highest. Even if I don’t toy with the WAV file and compress straight from the originally recorded file this remains the case.

I’ve settled on using between 224kbps and 320kbps for that final compression…(2) is this just a waste of space? (3) Would I get pretty much the same quality as the source, in your opinion, if I just compressed the files @ 192kps AAC? (4) Why do you think the lossless files remain at such a high bitrate when the source is so comparatively low? I had the expectation that the lossless files would be smaller because the WAV file is only capturing up to 192kbps audio. I apologize if I’ve not written this clearly (…went to public schools. Shocked )

I know I may be delving into stupidity but I don’t have any audio engineering background and I obviously don’t have an understanding of how WAV and AAC compression really work.

Here are my answers to your questions:
1. I wouldn't go higher than 192kbps AAC since your original source is 192kbps mp3 (presumably). There really is no need to go above the source bitrate since you won't gain any quality. I think recording anything above 192kbps VBR AAC is just a waste of space. It will probably provide quality that is close to your source. You could always do a blind ABX test between the modified WAV and the 192kbps AAC. I wouldn't decide anything until doing a ABX test.
2. See number one.
3. See number one.
4. This is the nature of the WAV format. The WAV format is built in the standard that it will ALWAYS encode audio at 1411kbps no matter what source the WAV is recorded from. So even though your source is a 192kbps mp3, converting it to a WAV will produce a 1411kbps WAV (keep in mind that they are two completely different formats).

Thanks for the reply. Though I understand this about WAV files I thought the lossless compression would be smaller than ~1000K because the WAV file had only captured 192kbps compressed streams...not...say...900kpbs. I think you're saying that once its in WAV format it doesn't matter what the original bitrate was...it could have originally been 2kbps. The result would be the same because WAV format is 1411, so a ~1000kbps lossless encoded file. I guess lossless will only trim so much from the file its encoded from regardless of the characteristics of the content.

Why dont you use Streamripper to directly record the mp3 stream? I did it a lot in the past, and it worked fine (apart from the fact that the tracks were not cut correctly at the right place when listening to my favorite station radio paradise [now comes with 192 kbps]).

Out of curiosity I downloaded and installed it again right now, and it works fine with current winamp but with the same track cutting problem. Maybe there is a workaround I don't know. But you can put the interesting tracks together and recut again using mp3DirectCut. That's what I did when I was using Streamripper on a regular basis.

What I don't know is whether or not Streamripper is still legal to use but I guess it is.

Other than that in any case where you don't have access to the original compressed format and must use the soundcard output or a virtual or real CD as a temporary medium (like with downloaded DRM protected music) or in case you want to do some editing as you said which cannot be done with mp3Directcut I think it's best to re-encode with a very high quality procedure in order to keep the re-encoding issue down. My favorite is wavPack lossy high @ ~350 kbps (but usually I buy CDs even if I like just a few tracks). In case of a 128 kbps mp3 stream I'd lower this ambition a bit - AAC in the 200 kbps region will be fine.

But if you can restrict your editing to cutting (trimming) or fading in or out it's best to use Streamripper and mp3Directcut.

I might consider StreamStripper, but again, remember I want to be able to edit and re-save the files with as high quality as possible which I why I'm using WAV format. My WAV editor, unfortunately will not save files in AAC format, which is what I'd like to use instead of MP3 or WMA since I'm using iTunes as my player of choice (60GB library from my OS X days that I'm not anxious to move to another player).

Thanks again.

Your wav editor can certainly store the result as a wav file. You can then encode to anything you like using for instance foobar. For converting into AAC I guess you can use iTunes too (I don't know much about iTunes).

Though it doesn't match your needs perfectly you might want to consider using Streamripper for the mere fact that you get your tracks well separated with filenames carrying the artists' and tracks' names. Probably you can read mp3 files into your editor (otherwise you have to convert them into wavs first).

I just played around with the streamripper options and found a way to overcome the cutting problem though not perfectly. I changed the splitting options 'Search Window Pre:' to 12000, 'Search Window Post:' to 1000, and 'Padding Pre:' to 12000, and this makes the tracks overlapping so that only the beginning and the end of the tracks have to be trimmed but nothing is missing in each track any more (at least according to current experience). This will not be the optimal setting, but I've tried so many other settings without success so I don't touch it at the moment.

I wasn't listening to internet radio a lot in recent times and was astonished how many 320, 256 and 192 kbps mp3 stations are available on shoutcast.com. Motivated by the Streamripper FAQ page I found Icecast has some interesting stations too especially Virgin Radio Classic Rock which comes with Vorbis -q5. You can streamrip this all using Streamripper.

BTW can you imagine using Rockbox firmware on your iPod? You can use wavPack lossy in case you're interested.

Good info again. Thanks all. The kids @ Tom's Hardware Guide SC forum just had no idea.