Apple's Lossless Audio Codec (ALAC) Now Open Source
Posted by: sector51 on 28 October 2011
Just caught this news on macosforgedotorg
The Apple Lossless Audio Codec (ALAC) is a lossless audio codec developed by Apple and deployed on all of it's platforms and devices for some years now. Apple is making the Apple Lossless Audio Codec (ALAC) available as an open source project. Full details can be found on the Apple Lossless Audio Codec project page.
Finally!
Puzzled by this thread. Had a good chat with Trevor at the TomTom bash and very interesting. But not as open as I would have hoped.
Uncompressed audio is uncompressed audio (IMHO). Shurely it's the process of decoding/transcoding which may influence power supplies - nothing to do with bit-perfect-ness. There's a good word. That takes processor cycles which ultimately affects ergs. No surprise that this is dependent on hardware.
Nothing new here - stuff affects power supplies. We listen to modified mains.
Design a good power supply. End of?
Still on Trespass......
Something else to get those of you so inclined hot and bothered. All streams sent to an apple express are packed up in ALAC before sending and decompressed by the express.
Lets consider the size of the processor in an airport express.
And you guys are arguing a say quad core i7 processor will struggle with this. Haha.
Simon
Next you'll be saying that Naim is optimised for girl singers with acoustic guitar accompaniments!
Have you never been in the Music Room? 
SinS, you prefer straight WAV over transcoded FLACs to WAV on Asset? Thanks!
S
Simon
Joe - i owe you a visit. We can test out various files - see you on FB. 
James
James,
That sounds like a good plan, see you over there. There is no hope for me I'm getting great pleasure from Spotify 320. LOL If only some of these old mates in the Music Room had better taste, I've been sampling some of their current favs, jeeze they listen to some odd stuff. 
Joe
S
Simon, Naim talking to Linn would be better imo .. here's a quote from one of their engineers:
"We have done extensive measurements on power supply disturbance recently, and have compared results for both FLAC and WAV streaming. Our findings are as follows :
1. If we measure the power rail that feeds the main processor in the DS we can clearly see identifiable disturbance patterns due to audio decoding and network activity. These patterns do look different for WAV and FLAC - WAV shows more clearly defined peaks due to regular network activity and processing, while FLAC shows more broadband disturbance due to increased (but more random) processor activity.
2. If we measure the power rails that feed the audio clock and the DAC we see no evidence of any processor related disturbances. There is no measurable difference (down to a noise floor measured in micro-volts) between FLAC and WAV in any of the audio power rails.
3. Highly accurate measurements of clock jitter and audio distortion/noise also show no difference between WAV and FLAC.
The extensive filtering, multi-layered regulation, and careful circuit layout in the DS ensure that there is in excess of 60dB of attenuation across the audio band between the main digital supply, and the supplies that feed the DAC and the audio clock. Further, the audio components themselves add an additional degree of attenuation between their power supply and their output. Direct and indirect measurements confirm that there is no detectable interaction between processor load and audio performance."
google "murrays power supply disturbances"
I will be interested to hear whether the mk2 boards improve this. There is also Naims design consideration of modularity, and disturbances on the Powerlines feeding the streaming board (which I believe does the decoding) could radiate. This could couple elsewhere. A mitigation would be to braid or twist the power supply wires and other connection wires to balance them but from pictures I have seen I don't think this is the case. I guess the Naim engineers are looking or have looked very closely at this. This area is often quite challanging in my expierience because the simulators I have seen don't accurately model these EMC interaction methods. Regular near field and far field antenna design modelling tools are hard enough.
Simon
So WAV and FLAC stream unpacking processes in NAIM are not identical then?
And why do we assume that full, bit-perfect unpacking occurs prior to DA-Conversion?
And what if DAC takes the raw WAV or FLAC stream for direct analogue conversion without prior unpacking???
>And why do we assume that full, bit-perfect unpacking occurs prior to DA-Conversion?
Because it does-
DACs convert streams of bits in to analogue - that is all. If you send them a word processing document then they'll try to convert it to an analogue sound - not tried, but assume you'll get something like Lou Reed's Metal Machine Music. Sort of liking playing your Sinclair ZX81 tapes in your Nakamichi. [Somebody correct me if this is wrong]
And what if DAC takes the raw WAV or FLAC stream for direct analogue conversion without prior unpacking?
It doesn't - whether the bits come from a CD transport, ALAC, WAV, AIFF, FLAC or some other format, you DAC needs something to unpack the bits from it for it to work.
A badly set-up computer can make a mess of things - for instance if you are playing space invaders while listening to some Robert Calvert then you can mix the sounds of innocent little aliens getting zapped with Bob's telling us that Einstein was not a handsome fellow.
But this has nothing to do with a DAC trying to play a FLAC file - if ever tries to then something is wrong with computer - ALAC, WAV, AIFF, FLAC all have the same music in them bit-for-bit and that is they sound the same on the same set-up if you disregard any noise caused by processing the different formats, which must be too small for me to hear in my set-up. As computers have background processes that run for housekeeping purposes, I'd suggest if you played the same file twice that the noise levels could vary. Perhaps this is a job for FPGAs.
All the best, Guy