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Topic: FrAD: Fourier Analogue-in-Digital (Analogue Audio Archive Codec) (Read 963 times) previous topic - next topic
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FrAD: Fourier Analogue-in-Digital (Analogue Audio Archive Codec)

(@Admins: since has not been approved as news, this is the normal discussion version of the post)

Sounds more like a "wannabe" than a working project, but some (Archivist Licensed) source code seems available:
Fourier Analogue-in-Digital is an uncompressed codec like WAV or AIFF rather than FLAC or ALAC. Unlike WAV or AIFF, which store PCM, FrAD stores energy density per frequency.

FrAD is a codec that nobody owns, nobody patents, and is standardised in so much detail that anyone can easily implement it in any language, even if the original implementation is lost. It also supports tags, cover art, and sequential search, and the original implementation can be supported by any OS on any machine that supports Python.

The main difference between any other formats and FrAD is its resilience to corruption and robustness. For many audio files that are branched fron PCM, even a flip of a bit or two can result in popping noise, failure to play the corrupted frame, or even the loss of all subsequent audio information. FrAD, however, provides robust error recovery such that sporadic flipping of a few bytes is not even a concern.

Goals of FrAD (more)
  • Preserving analogue waveforms intact in digital file.
  • Ensuring the analogue waveform remains intact even if the file is corrupted
  • Making all data frames work independently

Format specs:

Dev's help request:
Hybrid Multimedia Production Suite will be a platform-indipendent open source suite for advanced audio/video contents production.
Official git:

Re: FrAD: Fourier Analogue-in-Digital (Analogue Audio Archive Codec)

Reply #1
I fail to see what is analog about this.

So instead of sampling in the time domain (as PCM does) sampling happens in the frequency domain. But those frequencies are still represented by numbers which makes it digital. There is nothing analog about it. Sampling in the time domain also 'preserves analog waveforms' just fine.

The FAQ has an interesting take on what is lossless:
The loss in 12 and 16 bits can be taken as a significant loss of quality that anyone can hear [...] and that the loss is not actually a serious quality loss as you might fear. [...] Anything above 24-bit is beyond the dynamic range that the human ear can perceive, 0-80 dB SPL, so it's hard for human ears to notice the difference. So even though the PCM itself may not be preserved intact, the human ear will still have the exact same auditory experience, and we can say that it meets the definition of ‘lossless’.
By that definition, many codecs usually regarded as lossy are suddenly lossless too.
While complete lossless is virtually impossible outside of using PCM as it is, FrAD guarantees that at bit depths of 24 bits and above, the sound quality is indistinguishable from the original PCM to human and animal hearing.
So, seriously, what is the point of this?
Music: sounds arranged such that they construct feelings.