Yes, adding a readily playable header may not be a good idea in the
wrong hands and could be an option, but this method would allow
compression of an audio file (or more generally, a signal file) with
perfect recovery regardless of format or possible corruption.
Archiving corrupted or experimental files is a valid, albeit infrequent,
usage case.
I'm aware that the raw format allows this but at the expense of the need
of manual parameter input. In some cases, such as in adaptive sample
rate (https://ieeexplore.ieee.org/document/1105415) the parameters may
be unknown or meaningless. My proposal is to automatically estimate the
needed information under the assumption that the file contains some sort
of signal and possibly some ad hoc metadata.
Just in case somebody in the future might find this idea promising :).
By the way, I think a better name for "foreign metadata" would be
"non-standard metadata" or "extended metadata"
Regards,
Federico Miyara
On 03/11/2022 16:09, Martijn van Beurden wrote:
Op do 3 nov. 2022 om 19:39 schreef Federico Miyara <fmiy...@fceia.unr.edu.ar>:
Martijn,
Currently FLAC already stores and restores most kinds of metadata corruption
without problems, so in most cases the conversion is already bit-accurate.
However, there are some kinds of corruption it cannot handle. These are the
kinds of corruption that invalidate your considerations. For example, when a
chunk length is incorrect, the location and length of the audio data is no
longer known. It is also possible the basic formatting information is invalid.
In this case, FLAC cannot compress the audio at all, not even without
considering foreign metadata, while general purpose compressors (who don't have
to discriminate between audio and metadata) have no problem compressing.
OK.
That's why I said
"as long as the audio data is consistent, with known format and correctly
located."
However, I think there are relatively few uncompressed formats, and probably
all of them share having large audio blocks. It would be feasible to think of
an algorithm that attempts to find audio alignment by iteratively testing a
short portion for different alignments (meaning different number of channels
and bit depths, little/big endianness, and a few other variants or combinations
of PCM) until the maximum compression is attained. Once located the optimum
alignment, the FLAC algorithm would provide bit-for-bit accuracy and maximum
compression, even in the absence of format-awareness. It would take a bit more
time to encode and could generate its internal header for direct playback
compatibility.
That would inevitably result in metadata getting stored as audio,
resulting in loud clicks and static, which I don't think is a good
idea.
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