[flac-dev] libFLAC optimizations request

[Gabriel Corneanu]

The problem is really as I wrote:
1. Metaflac is no option for me, I use libFLAC.dll
2. There is no way (at least how I read the code) to avoid saving
comment with libFLAC; I would appreciate an extra option to avoid it,
which can default to old behavior if compatibility is important.
3. I have a high speed application, where re-initializing an encoder
is really significant. On corner cases it causes an 25% overhead! Of
course I don't expect it to be that significant in normal cases.

Thanks for all replies, but I don't have the code at home.
I will create a patch with my changes for review.

Regards,
Gabriel

On Sun, Feb 4, 2018 at 9:57 AM, Brian Willoughby
<brianw at audiobanshee.com> wrote:
> Correction, the flac command-line does create a 40-byte Vorbis comment by default. I just never noticed it before. I’ve been using —no-padding all these years for minimal file sizes without realizing that I could save another 40 bytes.
>
> Anyway, since metaflac can remove the Vorbis comment using the standard library, then you should be able to create a solution without modifying libFLAC.
>
> Brian
>
>
> On Feb 4, 2018, at 12:43 AM, Brian Willoughby <brianw at audiobanshee.com> wrote:
>> Gabriel,
>>
>> 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 Vorbis comment.
>>
>> 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 needed.
>>
>>
>> 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 kHz.
>>
>>
>> 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 <gabrielcorneanu at gmail.com> wrote:
>>> Hello all
>>>
>>> I am using libFLAC in a corner application, compressing a lot of small signals.
>>> 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 "metadata_has_vorbis_comment".
>>>
>>> 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.
>>>
>>> Regards,
>>> Gabriel Corneanu
>