Duet Receiver - High Res Music Files?

[Robo]

According to Benchmark's site, 16-bit word-length truncation is a Known Problem with iTunes v6 and QuickTime v7 for Mac. Benchmark shows the distortion graphs at http://www.benchmarkmedia.com/wiki/i...g_24-bit_files. Benchmark implies that this doesn't applies to the PC version.

If Snapper is showing that this truncation occurs during the file compression rather than the playback stage, then someone should check Apple's claims of "Lossless Compression". I'm now wondering whether my lossless compressed PC files (which look to be about the right size for 24 bits) might have been padded out with 0's!

[Eric Seaberg]

Again, iTunes said the files were 24-bit. I can check the files with ProTools as well, if you really care.

[Robo]

Thanks for zeroing in on the word length issue. I had been thinking that it was the sample rate that was hindering SqueezeCenter from streaming my high res files. As explained in earlier postings, these files had been compressed from wav format into AppleLossless format using iTunes 7.6 for Windows. All my Slim receivers (Duet - up to 48kHz, Transporter - up to 96 kHz, and SoftSqueeze 3.6 up to 96 kHz) just yield motorboating noises (on both the digital and analogue outputs) when I try to listen to them via SqueezeCenter.

I hadn't noticed that some of my high res files are actually 44.1kHz 24 bit files. These particular files may suffer from another issue, so I can't definitively blame their inability to stream on their having a 24 bit word length.

Does anyone NOT have trouble streaming to their Slim devices 24 bit music that has been compressed into Apple Lossless format using iTunes 7.6 for Windows?

On a Mac, the problem may be even more severe. As I interpret Eric's tests using Snapper, users of iTunes v6 for Mac may not even be getting a full quota of 24 bits in their 24 bit compressed files. Has anyone else observed this? Or is it just an artifact of the decompression routine used by Snapper? If true, it would have potential implications for audio archives.

[Robo]

Spurred by Eric's experiment I have plugged a mike and mike pre-amp into my PC's sound card, opened Nero Wave Editor and created a bunch of wav files, each with a different sample rate and word length.

I dragged the wav files into iTunes for Windows and got iTunes to convert them to Apple Lossless (ALAC files). I asked SqueezeCenter to look for new and changed music: it found and catalogued both the ALAC and the wav files.

The results of selecting these files in SqueezeCenter and attempting to stream them to my receivers are as follows -

ALAC files:
44.1kHz 16 bit: Plays on the Duet Receiver and the Transporter;
44.1 24 bit and 48kHz 24 bit: When I stream these files to either the Duet Receiver or Transporter I get continual "alac.exe has stopped working" messages on the PC until I stop SqueezeCenter;
96kHz 16 bit: Plays at half speed on the Transporter
96 kHz 24 bit: When I stream to the Transporter I get continual "alac.exe has stopped working" messages on the PC until I stop SqueezeCenter.
The "alac.exe has stopped working" message was occasionally accompanied by loud motorboating noises from my upstairs and downstairs stereos.

wav files:
48kHz 24 bit: Plays on the Duet Receiver and the Transporter;
96kHz 24 bit: Plays on the Transporter.
I haven't checked how the other wav files perform.

[Eric Seaberg]

Any 96k or higher files I have I actually encode to Dolby AC3 or DTS, even as a stereo mix, mainly to save streaming bandwidth. My decoder keeps the 96k/24bit word and the Transporter just thinks it's a 48k file.

As far as the Transporter playing your 96k file at the wrong speed, that does sound like ALAC is transcoding it thinking it's a 48k file, hence the half-speed playback. As far as 24-bit, I'll have to try playing some of my 88.2/24bit files on the Transporter to see what happens WITHOUT transcoding, since they're already AIFF.

[Eric Seaberg]

Just for fun I took one of my original 88.2kHz/24bit files (AIFF) and put it in my Music folder. My Transporter played it at 1/2 speed!! Just to make sure it wasn't the AIFF itself, I converted it to a WAV file... same results.

Odd!! Is this a 'bug' in 7.01 or firmware 37?

[fholo]

Was a solution (or explanation?) for this ever found?

[Phil Leigh]

Was a solution (or explanation?) for this ever found?

Sure - just convert to the FLAC format and everything always plays properly, regardless of sampling rate/bit-depth. Wav has no generally adopted standard way of representing > 44.1/16 in its header structures and some software can't handle the unnoffical extensions for this.