You can use dBpoweramp or Foobar 2000. I'm certain there are other programs you can try as well.

There's already some discussion of this on HA. Just type in 24 to 16 into the search engine. Sorry I can't be a bit more helpful...I don't find myself converting 24-bit to 16 too often. I did do this with the Nine Inch Nails "The Slip" release for use on my iPod. Seems to work just fine in 16 bit ALAC. I retained the original for archival purposes.

I'm not at all certain how you plan to compress any more than 2:1 to 3:1 without using a lossy codec.

• It has been asked for codecs, not applications. Why do you recommend Foobar and dBpoweramp?
• It has not been asked for a conversion to 16 bit. Why do you apologize for not having any experience with that?
• How is your interest in Nine Inch Nails in any way related here?
• What has the OP's question got to do with your iPod?
• How is the question related to ALAC?
• Why should anyone care that you retained the original?
• Why are you "not at all certain" that the OP's goal is not achievable without lossy codecs? Hint: He posted in the "other lossy codecs" forum.

I care immensely about you & everything you stand for.

Quote from: Engelsstaub on 2010-11-25 09:37:17

You can use dBpoweramp or Foobar 2000. I'm certain there are other programs you can try as well.

There's already some discussion of this on HA. Just type in 24 to 16 into the search engine. Sorry I can't be a bit more helpful...I don't find myself converting 24-bit to 16 too often. I did do this with the Nine Inch Nails "The Slip" release for use on my iPod. Seems to work just fine in 16 bit ALAC. I retained the original for archival purposes.

I'm not at all certain how you plan to compress any more than 2:1 to 3:1 without using a lossy codec.

• It has been asked for codecs, not applications. Why do you recommend Foobar and dBpoweramp?
• It has not been asked for a conversion to 16 bit. Why do you apologize for not having any experience with that?
• How is your interest in Nine Inch Nails in any way related here?
• What has the OP's question got to do with your iPod?
• How is the question related to ALAC?
• Why should anyone care that you retained the original?
• Why are you "not at all certain" that the OP's goal is not achievable without lossy codecs? Hint: He posted in the "other lossy codecs" forum.

There is not one partial sentence with at least some sense to it.

Pepper that with your sig:

Quote

I care immensely about you & everything you stand for.

Epic post!

One of my favorites this year.

 


To the OP:

It would help if you could provide more details about your realtime requirements, especially maximum latency and available processing power.

It doesn't have anything to do with another post I made recently, does it?

^ Can you believe this guy?
Is that old D.R.I. logo as avatar?
The only things I noticed posted by him were numerous repeatings to common HA terms and his Apples of course

Show some horizon man

and don't forget TOS about writing language, we are not making Babylon

thanks everyone, my project is using an ARM7 cpu to compress and decompress HiFi audio/music(24bit 48K sample rate that is 24*48=1152Kbps) in real time, and Tx/RX the compressed data using 2.4G wireless, latency equal or less than 3ms, the max wireless speed is about 500Kbps, so we need compress rate 3:1 or 4:1,  that is lossy codec.  I tried adpcm souce, but the frequency spectrum analysis  showing mang high frequency distortion. I want to do this using subband adpcm or other algorithms. are there any open source project concerning about this 24bit compress, or some papers i can reference?

Quote from: yg2003618 on 2010-11-26 02:09:52

thanks everyone, my project is using an ARM7 cpu to compress and decompress HiFi audio/music(24bit 48K sample rate that is 24*48=1152Kbps) in real time, and Tx/RX the compressed data using 2.4G wireless, latency equal or less than 3ms, the max wireless speed is about 500Kbps, so we need compress rate 3:1 or 4:1,  that is lossy codec.  I tried adpcm souce, but the frequency spectrum analysis  showing mang high frequency distortion. I want to do this using subband adpcm or other algorithms. are there any open source project concerning about this 24bit compress, or some papers i can reference?

3ms latency rules out pretty much any mainstream format (including AAC and MP3).  Does it need to be free?  Apt-X has a very low latency subband codec, perhaps they'll sell you something if you're on a commercial product.  Otherwise some kind of ADPCM or custom subband codec might be your best bet.

Doesn't compressing a 24-bit file to a lossy codec (other than a hybrid like WavPack) theoretically defeat the supposed benefits of DVD Audio quality? I'm seriously wondering. I was a bit confused before, of course, but this has me  more befuddled. I've never heard of such a want or need and wonder why one would do this.

I guess what I mean is: wouldn't just converting this to a 16-bit lossy file be about the same? AFAIK the lossy codecs will discard most of the stuff that's supposed to be in the 24-bit range. (Sorry if my terminology is imprecise or if my question seems too novice. I come to HA because I'm not that knowledgeable about some of the things most of you are and am trying to learn rather than "teach.")

I've been looking online for a solution to your question but the only thing I can find is Wavpack as soiaf pointed out.

Edited for a grammatical error.

Quote from: saratoga on 2010-11-26 04:57:37

Quote from: yg2003618 on 2010-11-26 02:09:52

thanks everyone, my project is using an ARM7 cpu to compress and decompress HiFi audio/music(24bit 48K sample rate that is 24*48=1152Kbps) in real time, and Tx/RX the compressed data using 2.4G wireless, latency equal or less than 3ms, the max wireless speed is about 500Kbps, so we need compress rate 3:1 or 4:1,  that is lossy codec.  I tried adpcm souce, but the frequency spectrum analysis  showing mang high frequency distortion. I want to do this using subband adpcm or other algorithms. are there any open source project concerning about this 24bit compress, or some papers i can reference?

3ms latency rules out pretty much any mainstream format (including AAC and MP3).  Does it need to be free?  Apt-X has a very low latency subband codec, perhaps they'll sell you something if you're on a commercial product.  Otherwise some kind of ADPCM or custom subband codec might be your best bet.

Otherwise some kind of ADPCM or custom subband codec might be your best bet. ?

could you please tell more?

Doesn't compressing a 24-bit file to a lossy codec (other than a hybrid like WavPack) theoretically defeat the supposed benefits of DVD Audio quality? I'm seriously wondering

While more than 16 bits doesn't make much sense as a final delivery format, there are applications for higher bit depths. For example, when recording live, 8 bits of additional headroom free from the necessity to continually adjust the input gain to fit into a 16 bit window.

It depends, if high dynamic range is a top priority (at the cost of others), then yes.

And especially regarding ultra low delay codecs: 144 samples per frame is such a tight limit for the set of possible filtering operations,

maybe a higher "vertical" input resolution can help the encoder make better choices under these extreme circumstances. But that's only a guess.

I think thats unlikely.  Hes using ARM7, which means 32 bit fixed point inputs to the encoder, no matter what the actual PCM is (since arm7 lacks dsp extensions for things like 16 bit multiplies).

In practice, since he is using lossy anyway, the choice the input PCM format is likely to be of no consequence.

Could you please explain the inference? I don't understand it. Especially how it is supposed to refute to the paragraph you have cited.