audio editor for 64-bit .wav files

[tony tony chopper]

We (Sonic Foundry/Sony) defined a 64 bit RIFF format so that very large files could be created.

Do you happen to have other proprietary wav chunks to store tempo btw? I'm interested in reading tempo (& any other kind of info) from something more reliable or precise than the ACID chunk. Although I can assume that at Sony you keep using the ACID chunk.

(I know that Syntrilium also had a chunk for that in Cooledit, but could never get any info because Adobe is impossible to reach)


It would be easy to define yet another new chunk, but it would only add to the mess that the wav format is.

[formlabs]

As SonyPCH and UltraJv said, these files are for long recordings. It's got nothing to do with extreme dynamic range.

From the Cubase manual, Export Audio Mixdown section:

Wave 64 is a proprietary format developed by Sonic
Foundry Inc. In terms of audio quality, Wave 64 files are
identical to standard wave files, but in the file headers
Wave 64 files use 64-bit values for addressing where wave
files use 32-bit values. The consequence of this is that
Wave 64 files can be considerably larger than standard
Wave files. Wave 64 is therefore a good file format choice
for really long recordings (file sizes over 2GB), e.g. live surround
recordings. Wave 64 files have the extension ".w64".

Ok, that makes sense to me. To get around file the file size limits of WAV and AIF files, Apple came up with CAF, which can hold days worth of audio in one file.

[BertKoor]

You guys do bounce (instruments and similar) to 32-bit float right?

Well, actually... If you'd bounce to 16bit ints, chances are you won't notice any difference in the end product. Since individual tracks end up something like 10dB softer in the final mix, there's enough resolution there to hide any truncation artefacts. Provided ofcourse the mixing engine works in 32bits (but hey, they all do nowadays)

[tony tony chopper]

or orchestral pieces where you really want to catch the nuance of that Stradivarius.

Same as he wrote above, chances are that the recording's background noise is in the 16bit realm, waaaaaaaaaaaaay beyond 64bit or 32bit.
(that's for normal people, because the rest will just tell you that a stradivarius can't be recorded with bits)
I really can't imagine any good reason to want to save 64bit audio files (except marketing).

[LawrenceF]

You guys do bounce (instruments and similar) to 32-bit float right?

Well, actually... If you'd bounce to 16bit ints, chances are you won't notice any difference in the end product. Since individual tracks end up something like 10dB softer in the final mix, there's enough resolution there to hide any truncation artefacts. Provided ofcourse the mixing engine works in 32bits (but hey, they all do nowadays)

So you bounce your instrument and track stems and track consolidations (like in the middle of a production) to 16 bit files? I wouldn't do that but it probably doesn't matter much in the big picture.. especially in pop music. I stem and bounce 32-bit float, always have.

Call it my "comfort level".

[kmmcdonald]

The reason that I initially posed this question is as follows:

I purchased the Galbanum Architecture Waveforms library in 64-bit format. I have been using them for importing into Rapture (in which they, for one reason or another, sound better to me than the original Rapture library), and for importing into Alchemy resynthesis for further processing in the additive editor.

I have discovered that Alchemy cannot effectively resynthesize single-cycle waveforms. Therefore, I need to create .wav files with the waveform repeated a few times for import into Alchemy. Thus, I needed an audio editor that can work with these files. So far, Alchemy imports and resynthesizes the new, longer, files well.

From what was available to me (Audacity, Pyro Audio Creator, and Wavosaur), only Wavosaur could do the job.

Keith

[ishkabbible]

Nothing like resurrecting a dead thread....
I got here from asking exactly the same question today, because Zynewave Podium DOES bounce to 64 bit floating point .WAV files, and they claim, rightly or wrongly, that mixing in double precision results in better sound when you have many tracks. However, once I have the final mix, I need to convert it to 32-bit floats before any of the commonly available (i.e. free) MP3 encoders can handle it. So in reality what I need is something that can read these 64-bit float .WAV files, round them to 32 bit floats, and write them back out.
And I found that tool already in my toolbx - Audacity 2.0 does exactly what I need, and I expect it will also do what OP needs.

[orange]

for some reason i got a nervous twitch in my right eye reeding this thread....

[ishkabbible]

for some reason i got a nervous twitch in my right eye reeding this thread....