metadata_has_vorbis_comment is a FLAC__bool which defaults to false. There
should be no reason to modify stream_encoder.c, but just modify the caller.
The following command:
metaflac —remove —block-type=VORBIS_COMMENT —don’t-use-padding
… will remove Vorbis comments from existing files, so is must be legal without
modifying the library. metaflac can clearly create a new FLAC file without the
I use the flac command-line, and I never get Vorbis comments in the files that
I create. Perhaps you are using another tool which assumes Vorbis comments are
The FLAC algorithm is not dependent upon sample rate. AIFF has an 80-bit
floating point type for sample rate, so it should be able to handle 40 MHz. I
assume that any AIFF can be converted to FLAC losslessly, but I have not tested
whether certain sample rates are rejected. FLAC itself only supports sample
rates up to 655,350 Hz, so you may have a problem there unless you “lie” about
the sample rate when creating the file. Maybe you could just establish a
private convention to divide the sample rate by 100 to make yours fit. 40 MHz
would map to 400 kHz, 10 MHz would map to 100 kHz, and 5 MHz would map to 50
You’re probably asking for trouble if you try to reuse an encoder. It seems
like there would always be some risk that details from the previous file would
bleed through into the next. Have you benchmarked allocation and
initialization? Is it really that slow? In order to reuse an encoder, you’ll
need to overwrite all state variables, and I don’t see how that could be much
faster than simply allocating them anew. Perhaps you could allocate groups of
encoders at once, if that would speed the process.
On Feb 1, 2018, at 4:29 AM, Gabriel Corneanu <gabrielcorne...@gmail.com> wrote:
> Hello all
> I am using libFLAC in a corner application, compressing a lot of small
> First is a general question: in our application we have signals in range 5-10
> MHz, potentially 40MHz! Is there any potential problem with that?? The mac
> sample rate is limited in flac, but it doesn't really seem to be a problem.
> The output is stored as blob in a sqlite database, it never needs to be a
> valid audio file outside our application.
> In my tests, the signals are compressed very well, much better than general
> compression libraries like zlib, zstd, etc.
> Now other small issues; I also made some tickets about them, but I thought
> asking here might be better.
> 1. I would like to avoid saving vorbis comment, by default ~40 bytes. Right
> now the only option is to modify stream_encoder.c, see
> 2. Speed is very important, therefore I would like to reuse an encoder
> without re-initializing everything.
> Ideally I would like 2 (exported) functions: "flush" and "restart".
> "Flush" is self-explanatory, should properly end the encoding. I could split
> myself "flush" from "finish", it looks relatively simple.
> "Restart" should keep all current settings, generate a new stream header and
> clear everything for encoding a new signal.
> It' clear that current settings, re-creating windows, cpu-dependent
> functions, etc could be kept around.
> I was not quickly able to extract all the necessary initialization from
> "init_stream_internal_" into a new "FLAC__stream_encoder_restart" function.
> Gabriel Corneanu
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