[HOMBRESINIESTRO]

It interrupts playing every 10-20 seconds for a little while. Apart from that everything seems to work just fine. I have itergrated the express basestation it in my existing Airport network with an extreme basestation and "old" Airport Clients.
When setting up a network for the express station itself it works without stuttering, but I want to be able to surf the net _and_ listen to my music on the stereo.

Does anyone else have this problem and knows a possible solution. There are six meters between my express basestation and my iMac, btw.

[HOMBRESINIESTRO]

ah, and there are no mobile phones or microwaves in the way!

[Krypton]

I think I heard that using Airtunes at 802.11b speeds (due to having older Airport cards on the network) can cause stuttering, but at 802.11g speeds there is enough bandwidth to do all the things you want.

[qnxde]

Uncompressed audio is 150KB/sec, AirTunes uses Apple Lossless which on average halves that, to 75kb/sec. Surely even 802.11b can sustain that sort of a bit rate - my DSL is faster than that!

[f1000]

Originally posted by qnxde:
Uncompressed audio is 150KB/sec, AirTunes uses Apple Lossless which on average halves that, to 75kb/sec. Surely even 802.11b can sustain that sort of a bit rate - my DSL is faster than that!

Depends on the audio format: AIFF requires more than 150 kbps (more like 1.4 Mbps).

Also, does Apple Lossless halve MP3 data rates?

[qnxde]

yes, I said 150KB/sec (150 kilobytes a second) - that's still only 75 kilobytes a second for lossless - which even localtalk can almost sustain - nevermind 802.11b or any other network protocol

[f1000]

Originally posted by qnxde:
yes, I said 150KB/sec (150 kilobytes a second) - that's still only 75 kilobytes a second for lossless - which even localtalk can almost sustain - nevermind 802.11b or any other network protocol

My mistake, I didn't notice the 'KB'.

[CatOne]

Originally posted by f1000:
Depends on the audio format: AIFF requires more than 150 kbps (more like 1.4 Mbps).

Also, does Apple Lossless halve MP3 data rates?

No, Apple Lossless runs about 50% to 80% of AIFF (uncompressed) data rates. It's a considerably higher bit rate than MP3 or AAC, as it's "lossless."

And AirTunes uses Apple Lossless over the air (and it's encrypted). If your music is encoded in AAC or MP3 or whatever, it is uncompressed, and then recompressed with Apple Lossless before transmission.

[Millennium]

Originally posted by f1000:
Depends on the audio format: AIFF requires more than 150 kbps (more like 1.4 Mbps).

Also, does Apple Lossless halve MP3 data rates?

Nope.

Here's the thing: datarate is nothing more than a measure of file size. A 128kbps MP3 stream will be the same size as a 128kbps Apple Lossless stream of the same song, which itself will be the same size as a corresponding 128kbps AAC stream. The actual file sizes may differ slightly because of metadata, but the streams will be exactly the same size if they were recorded from the same source at the same bitrate.

What makes MP3 "good" is that it sounds decent at low datarates. AAC is better because it sounds better at those same datarates. Essentially it's not a matter of how good the codec sounds, but how much data can be thrown out while still sounding good. A 128kbps AAC sounds better than a 128kbps MP3 not because it throws away more or less data (it doesn't), but because it is better at picking what data to keep and what to throw away.

AAC doesn't throw out any data at all. This makes its datarate very high. It can compress losslessly, which crunches the datarate down a little, but lossless compression can only go so far because you can't throw any data out (if you do, then the codec is no longer lossless).

[f1000]

Originally posted by Millennium:
AAC doesn't throw out any data at all. This makes its datarate very high. It can compress losslessly, which crunches the datarate down a little, but lossless compression can only go so far because you can't throw any data out (if you do, then the codec is no longer lossless).

CatOne points out something that should've been obvious to me; that is, iTunes must first decompress music files into a raw data format that can then be recompressed using ALE and sent to the AirPort Express unit. The recompressed music files shouldn't be any smaller than their original lossy counterparts; thus, ALE's real benefit is to serve as a lingua franca to simplify the playback circuitry in the AP Express unit.

Now, you've gone and thrown a monkey-wrench into my understanding of the universe by saying that AAC doesn't throw out any data at all. I hope you meant ALE and not AAC.

[HOMBRESINIESTRO]

After reading all this, is it normal that my "old" Airport network can't handle the streaming or is it not. I.e. can I hope for an update from Apple to fix the problem or is it a real hardware limitation?

[gperks]

Mine swill pause for a moment or two when the signal strength changes. Using APGrapher I have about a 70% connection. If I walk between my router and Mac, it drops to the low 60s and the music has a good chance of pausing.

The router to AE distance is a similar one from router to Mac, just through a different wall.

AEv2 could use a larger anti-skip buffer :-)

[Millennium]

Originally posted by f1000:
Now, you've gone and thrown a monkey-wrench into my understanding of the universe by saying that AAC doesn't throw out any data at all. I hope you meant ALE and not AAC.

You're correct; that was a typo on my part. My bad.

[AKcrab]

Originally posted by HOMBRESINIESTRO:
After reading all this, is it normal that my "old" Airport network can't handle the streaming or is it not. I.e. can I hope for an update from Apple to fix the problem or is it a real hardware limitation?

I am using an Airport Express in a mixed b/g environment. I have and XP machine with a PCI card that is g compatible, but my G4 867 is only b compatible.

Both machines connect with a good signal, and *both* can stream music to the AE via iTunes.

There is a slight lag at the beginning of a stream, which is unfortunate, but i suppose that's to be expected. It's usually only at the beginning of a stream, not between tracks.

[bmedina]

iTunes must first decompress music files into a raw data format that can then be recompressed using ALE and sent to the AirPort Express unit. The recompressed music files shouldn't be any smaller than their original lossy counterparts; thus, ALE's real benefit is to serve as a lingua franca to simplify the playback circuitry in the AP Express unit.

Yes, iTunes has to convert everything to WAV/AIFF and then recompress to ALE on the fly. The only exception is if the file already is ALE, then of course it doesn't recompress. The broadcast ALE stream will always be 700-1000 kbps regardless of whether or not the source files are lossy or lossless.

The drawback to this method is a performance hit to the computer that is broacasting. The great benefit of this method is ensured forward compatibility with any format a future version of iTunes may support.

...lossless compression can only go so far because you can't throw any data out

Then how does the file size cut in half? Of course you're throwing data out; that's the whole point.

if you do, then the codec is no longer lossless.

No, a codec is lossless if the bits you put into an encoder are identical to the bits that come out of the decoder. It can throw out as much data as it wants as long as it meets that one criteria.

[ShotgunEd]

Originally posted by bmedina:
Then how does the file size cut in half? Of course you're throwing data out; that's the whole point.

As far as I understand, lossless encryption uses 'symbols' to represent recurring pieces of data. It doesn't throw data out, but rather finds a more efficient way of representing it.

With aac and mp3 the sound quality worsens as actual sound data is ditched, usually at the low end (thus low quality mp3s have crap bass response).

[bmedina]

Originally posted by ShotgunEd:
As far as I understand, lossless encryption uses 'symbols' to represent recurring pieces of data. It doesn't throw data out, but rather finds a more efficient way of representing it.

That would be one way of "throwing out data" without sacrificing quality. My point is, you can't go from a 100MB file to a 50MB file without "throwing out data."

[Gee4orce]

Depends on your definition of 'data'. Strictly speaking, you don't throw away data ('information'); if you did, it would be lossy (the loss of information, or data). All you are throwing away is redundant bytes that can be compressed by saying "here comes a hundred x0123s", instead of "x0123 x0123 x0123 x0123 x0123.... etc".

[chris v]

HOMBRE, Have you tried changing the streaming buffer settings in iTunes prefs?

CV

[HOMBRESINIESTRO]

Originally posted by chris v:
HOMBRE, Have you tried changing the streaming buffer settings in iTunes prefs?

Just tried it. Doesn't change anything. :-/ Are you sure this setting is for Airport express streaming?

[chris v]

Originally posted by HOMBRESINIESTRO:
Just tried it. Doesn't change anything. :-/ Are you sure this setting is for Airport express streaming?

No, it was a blind stab in the dark.

[Spheric Harlot]

Originally posted by ShotgunEd:
As far as I understand, lossless encryption uses 'symbols' to represent recurring pieces of data. It doesn't throw data out, but rather finds a more efficient way of representing it.

With aac and mp3 the sound quality worsens as actual sound data is ditched, usually at the low end (thus low quality mp3s have crap bass response).

The high end suffers dramatically, too.

[ShotgunEd]

I notice it significantly more in the low end. I'm sure thats just me though. I like to EQ with a boost in the low and high end anyways, what with the middle carrying most of the noise.

At least thats the way it works with guitars and the like. I've no idea whether this carries over to mp3s.

[bmedina]

Originally posted by Gee4orce:
Depends on your definition of 'data'. Strictly speaking, you don't throw away data ('information'); if you did, it would be lossy (the loss of information, or data). All you are throwing away is redundant bytes that can be compressed by saying "here comes a hundred x0123s", instead of "x0123 x0123 x0123 x0123 x0123.... etc".

Data = bits.

With lossless codecs, the encoder throws away bits, but the decoder can exactly reconstruct the thrown-away bits. Lossy codecs don't have this property.

The high end suffers dramatically, too.

Only at low bit rates. In that case, most encoders use a low-pass filter to get rid of the highest frequencies. Since those frequencies are generally the hardest to encode, it's a good tradeoff, as our hearing is more sensitive to lower frequencies.

[AKcrab]

Originally posted by AKcrab:
There is a slight lag at the beginning of a stream, which is unfortunate, but i suppose that's to be expected. It's usually only at the beginning of a stream, not between tracks.

I guess I wasn't alone with this.
Seems to me Apple could give me a better "fix".

http://docs.info.apple.com/article.html?artnum=108072

This happens only when connecting to certain digital receivers, and it usually only happens to the first song in a stream. These digital receivers are simply synchronizing their clocks with the music stream, which may take a few seconds. This is normal when using such a receiver.

Synchronizing their clocks? wtf? There's no clock on my receiver.

[Person Man]

Originally posted by AKcrab:
I guess I wasn't alone with this.
Seems to me Apple could give me a better "fix".

http://docs.info.apple.com/article.html?artnum=108072

Synchronizing their clocks? wtf? There's no clock on my receiver.

Synchronizing the sound processor clock. All processor chips have an internal "clock."

[AKcrab]

Originally posted by Person Man:
Synchronizing the sound processor clock. All processor chips have an internal "clock."

Ahhh.. Thanks.
I hope Apple will put an option in the prefs to somehow get around this.