Strategy for time-indexed audio archive with lossy compression

For part of one of my projects, I am considering developing an audio archive for internet radio stations. This archive would be indexed and addressable by date/time.

For example, the server would connect to a stream (generally encoded in MP3), and save the st开发者_如何学Cream data. A client could connect to this server and request audio from 2011-07-05 15:58:30 to 2011-07-05 15:59:37. The server would return the audio data to the client for playback.

My initial thought was to save the data to 1-minute chunks of raw MP3 data to disk, and reference these files from a database. The server would be dumb to the stream/file format, and wouldn't understand mpeg frames. It would simply pass on data to the client, dividing the chunks up linearly to send. It would be up to the client to sync to the stream. This is not unlike how internet radio servers run in general. SHOUTcast servers simply output the data, byte for byte, that is sent to them from the encoder. When a client connects, data is sent, regardless of whether or not it even ends on an MP3 frame. It is up to the client to sync.

I am wondering if there might be a better approach, maximizing compatibility with clients and audio formats. Any thoughts on how to go about this?

The only other thing I can think of is decoding the MP3 to raw PCM audio and re-encoding as necessary when requested. I would prefer not to go this route due to the disk space required, and the loss of quality when re-encoding.

This question is language-agnostic, but if it is helpful, I will likely implement a solution in PHP with MySQL as the database.

You don't have to worry about this, since ALL mp3 that I accessed over shoutcast is Constant Bitrate. Do you don't have to index it. I have POC project that had archive in 5 minute chunks, then uses PHP to combine that files and pseudo-stream it to the winamp via shoutcast. It worked!

And since you are working with mp3, you can assume (and you'll assume correctly) that the density of the captured file is linear, so to access 30 second of the 60 second file you should seek in the middle. Since mp3 decoders are robust enough, you don't have to track the frames at all here.

AACplus, whole different story. It's inherent VBR.