Hi,

I need to archive some FM Radio broadcasts and I want to save the programs in the highest possible quality, but without being overkill for the source media (an FM broadcast, clearly lower quality than CD).

My goal:
- quality perceptively the same (or acceptablely the same, without ringing etc.) as the original FM Broadcast
- as small a file as possible
- preferrably an mp3, for compaibility with hardware audio players.
- I'd really like to shoot for 64-128kbps, closer to 64

I have read that a 22,050 kHz 16-bit stereo recording is higher quality than an FM brodcast, so I'm assuming that should be opitmal and that anything higher would be overkill. Can I even go lower? (FYI -- i can't record any higher, since these ar 4 hour broadcasts and that will result in a file larger than 2gb, exceeding the limits of the wav format (this is not the Fat32/NTFS issue).

Finally, space is a big concern, so I want to compress these. I'm really only a LAME mp3s user. I have read that 112kbps is comparable to FM, but cannot find any details if that's 112@44.1 or 112@22.050, stero/mono etc. -- so I don't know if i should trust this.

I have also read that some other codecs do a much better job at low bitrates and for lower quality source, so i'm open to those suggestions as well. I really did want to stick to mp3s though, since i can play those files on various hardware mp3 players i have.

So, does anyone out there have any technical expertise on this? What should be my optimal encoder and settings to balance space and quality?

Thanks,

Jon

on the same topic
http://www.hydrogenaudio.org/forums/index....012&hl=fm+radio

I think fm radio has more bandwidth than 22khz. I would suggest using 32khz if that is possible, and compatible with your soundcard. At 32khz, I would think that somewhere between 96-128 kpbs abr would work well at this frequency, with mp3. With ogg or mp3pro, you could probably get away with 64-80 kpbs.

A little correction: fm radio has more bandwidth than 11 kHz. It would be good to reproduce up to 16 kHz (thus 32kHz sampling rate) if that is possible.

There is a carrier or multiplexing frequency present on fm broadcasts somewhere around 18khz.

Better quality tapedecks used to offer an option to filter this out, so it may be above the ath and eating up bits for no reason. Using -Y with VBR might solve the problem.

A lowpass at 17.5 or so would work too. I don't think there is any real "content" above this level, but I could be very wrong in this because I have not looked into such things for ages.

IIRC FM radio audio bandwidth is 15KHz, so 32KHz samplerate would be sufficient. The quality will probably suck because almost all stations use heavy compression to make their material sound louder.

heh.

So then it's clear-cut, eh? --alt-preset 80 or --alt-preset 96 would be just right, I'd say.

Thanks for all your responses, you all rock. While i'm a veteran at mp3s with over 10,000 encodes under my belt, for this new FM-specific issue i'm a newbie

In doing the math, i need to record a 4.5 hour broadcast every weekday, and would like to archive an entire week on one 700mb CD. That roughly relegates me to 64kbps to squeeze 22.5 hours on a single CD.

That being said I have a few more questions/clarifications:

1. Regarding the sample rate. I don't truly understand the whole aliasing issue, but i've always been told to sample high and then downsample in software so you get averaging of samples. I have an SB Live!, so is 32kHz optimal and 44.1 overkill or is 44.1 optimal

Does anyone have a link to a good explanation (with diagrams, i'm a visual thinker) to this whole aliasing/sampling issue?

2. Assume I initally sample high, based on the quality of an FM broadcast, would it make sense for me to encode at a lower sample rate like 22.050 or 16? -- in other words, would a 44.1kHz @ 64kbps be similar in quality (or better or worse) to the same recording downsampled to 22 or 16 @ 64kbps

3. I checked the specs on my tuner and the frequency resonse is 30Hz - 15 kHz -- should i then set my hipass and lowpass filters to this?

4. The broadcast will be primarily (but not entirely) talk radio, so would that effect your codec choices? There will be musical portions i'd like to retain as best as possible.

5. Since i will definately be encoding at 64kbps, should I go with mp3Pro as opposed to mp3? Is mp3Pro compatible with regular mp3 players (i want to use these files with a hardware mp3 player).

I've never used mp3Pro, so what's the best mp3Pro encoder, equivalent to LAME for mp3? -- Also, does it support commandline encoding, as i need to run the encodes in batch

Thanks again,

Jon

I don't know how SBLive's input resampling compares to its output resampling, but I think Pio2001 recently stated that even if you record the input at 48kHz (SBLive's native samplerate), the card is still doing a 48kHz -> 48kHz resample, which might cause audible distortion in high frequencies. Basically, there's nothing you can do to avoid the SBLive's problematic hardware resampling when you record, so there doesn't seem to be a reason why you should capture your radio recording at a samplerate other than the one you finally want to encode with. Capturing at 44.1kHz and then converting to 32kHz will do more harm than good. Plus, you're going to want to chop off all the high freqs when you encode anyway (because FM doesn't transmit a musical signal over XXkHz... 15kHz, I guess?)

Anyway, should consider encoding the MP3 at 44.1kHz (or 48kHz), as opposed to 32kHz, to take advantage of MP3's improved time resolution at higher samplerates.

And I don't know about MP3Pro's compatibility with portable players, but it's not nearly as good as plain-old MP3 compatibility. I mean, MP3Pro will play on any MP3 player, but you'll lose all the high frequencies if the player isn't specifically MP3Pro compatible. So MP3Pro and Ogg Vorbis have essentially the same hardware compatibility currently (virtually nil).

And you also said that you were recording primarily voice, but you also want to record music. For this application, a VBR codec would be ideal. However, Lame VBR is unthinkably bad at bitrates this low, and I don't know how good MP3Pro VBR is. But we do know that Ogg Vorbis VBR is pretty decent at this bitrate. If you experiment a bit, you could find a good -q setting that averages 64kbps over a long recording of talk and music. So give Ogg Vorbis consideration for your project.

FhG 64 kbit/s Intensity Stereo with 44.1 kHz sample cutoff at 12000 Hz would be great

By any chance, are you recording Howard Stern?

Also, you might want to check out the Windows Media 9 encoder. I've been checking it out and it seems to outperform mp3 at low bitrates if you're looking for the least filesize. I recorded Stern last week just for kicks and found that 48 - 56 kbps with a 22000hz cutoff sounds phenomenal for archiving. Also, compatibility is of course, a given. Most mp3 players out there have WMA support, as well do the various burning programs.

Otherwise, I'd say Ogg -q0 would be great. However, due to lack of hardware and burning support, its not the right choice for you. At least not yet.

Actually I am recording "Ron and Fez" -- There's a high likelihood that they'll be cancelled in NYC since WNEW is like the worst mananged radio station in the world and they love to take shows away from us just when they get popular. I've had too many other shows I like disappear without a trace, so I wanna archive a ton of these for the future.

Once I get this worked out, I'm gonna post a how-to doc for others trying to do the same thing.

Truth be told, 64kbps mp3 actually sounds good enough, but i still wanna do some more tests to make sure i get the most out of these little files.

I'm still searching for a command line mp3pro encoder... anyone know of one?

Thanks