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Converting to FLAC with active "Sample Rate Converter", always leads to broken output #639

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Switch123456789 opened this issue Oct 4, 2024 · 1 comment
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@Switch123456789
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To Reproduce

  1. Pick any audiofile as input & select FLAC format for output.
  2. Enable Sample Rate Converter and set a different rate as the the one from the input-file.
  3. Start the encoding process & get a broken file
  4. ALTERNATIVE: Enable processing during playback, hit play and notice nothing plays back.

System

  • Version: 1.1.7
  • OS: Arch Linux
  • CPU: AMD Ryzen 5 3600X
  • RAM: 16GiB

Additional context
I tried many combinations but only if I combine the Sample Rate Converter with FLAC as output-format things break.

@enzo1982
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Sorry for the late response!

This is most likely not a bug, but a side-effect of using the sample rate converter combined with a newly added feature in FLAC 1.4.0.

The sample rate converter operates on 32 bit samples, so any lower resolution samples are converted to 32 bit when performing a sample rate conversion. Before FLAC 1.4.0, the reference FLAC implementation was not able to handle 32 bit samples, so fre:ac automatically converted such samples to 24 bit when encoding in FLAC format.

As of fre:ac 1.1.7, this is no longer the case and fre:ac now creates 32 bit FLAC files. The drawback, however, is that most older implementations of FLAC cannot decode such files.

To work around this, either update your FLAC decoders to at least FLAC 1.4.0 or, if that is not possible, use the Sample Format Converter filter in fre:ac in conjunction with the Sample Rate Converter. The Sample Format Converter should come after the Sample Rate Converter in the list of selected filters and be configured to 16 or 24 bit integer format.

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