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1
General - (fb2k) / Re: If you're sick of unnecessary forced updates...
Last post by lvqcl -
I might indeed replace the beta expiration thing with something you can dismiss if you really don't want to update *now*, but I will have to make it reappear once in a while

Several times now I've been in the middle of doing other things and I attempt to pause or switch tracks (through the executable /pause switch (bound to a hotkey (linux...))) and instead I get the update warning, which means I have to drop everything I'm doing to deal with foobar2000 stuff.

Maybe it's better to disable update check if foobar2000 instance just sends a message to the main foobar2000?
2
General - (fb2k) / Re: If you're sick of unnecessary forced updates...
Last post by Xenolith0 -
Why hello there, this is Mr Dev Person speaking.
...snip...
That said, I might indeed replace the beta expiration thing with something you can dismiss if you really don't want to update *now*, but I will have to make it reappear once in a while, for the above reasons.

Hello Peter,

I understand the desire from a developer point-of-view about wanting all users running latest versions for support reasons. At the same time, from a user's perspective, purposely breaking software that otherwise, was working perfectly ten minutes ago because some timer rolled over, leaves a bad taste in my mouth.

For the beta's I can understand a repeating update nagger, but it would be really nice to not be forced to update immediately.

Several times now I've been in the middle of doing other things and I attempt to pause or switch tracks (through the executable /pause switch (bound to a hotkey (linux...))) and instead I get the update warning, which means I have to drop everything I'm doing to deal with foobar2000 stuff.

To the other posters: yes, updating is easy, but it totally breaks my focus on whatever else I happen to be working on... Hence my dirty clock hack, it was just one-interruption-too-many that day and I decided I didn't want to deal with the problem again.

4
Support - (fb2k) / Re: Question mark for duration on alot of AAC files
Last post by SilverDeath -
as i said - to much work - im not going to remux all my aac files (and do this for all future aac files) and handle duplicate files, etc. just because fb2k cant handle aac files correctly while virtually every other player can (just checked, even windows mediaplayer works).
guess ill just ignore it or switch back to winamp if theres no reasonable solution.
5
Support - (fb2k) / Re: Question mark for duration on alot of AAC files
Last post by Porcus -
Converting all affected files would be waaay to much work...

First, you should not "convert" in the "decode/encode" sense. You should remux (a.k.a. "re-encapsulate")
Like for example ffmpeg -i yourmusicfile.aac -acodec copy yourmusicfile.m4a [oh, btw: check if your ffmpeg version is willing to use ".m4a")

Second: you can run a FOR loop from your "top music directory", that will traverse all .aac files and put them in mp4 containers. Then you can afterwards copy tags if necessary, bit-compare to check that you have not lost any music, and then delete your .aac's - or tell fb2k to ignore them.
6
Support - (fb2k) / Re: Question mark for duration on alot of AAC files
Last post by SilverDeath -
Converting all affected files would be waaay to much work...

i tried the ffmpeg decoder:
configured it to handle aac files:

and set it to top of decoder invoking order:

but no difference at all - is there anything else i need to do, or a way to check if it is using ffmpeg?

EDIT:
just opened the console - it says:
Quote
Launching ffprobe:
"D:\Program Files\ffmpeg\bin\ffprobe.exe" -of xml -show_format -show_streams -show_chapters -hide_banner "<...path...>\TroyBoi - No Substitute ft. Y.A.S (256 kbit_s).aac"
Launching ffmpeg:
"D:\Program Files\ffmpeg\bin\ffmpeg.exe" -i "<...path...>\TroyBoi - No Substitute ft. Y.A.S (256 kbit_s).aac" -map 0:0 -f w64 -acodec pcm_f32le -
so ffmpeg seems to be running.
i quickly ran the ffprobe command via cmd which outputs:
Code: (xml) [Select]
<?xml version="1.0" encoding="UTF-8"?>
<ffprobe>
[aac @ 000001536d1bcb80] Estimating duration from bitrate, this may be inaccurate
Input #0, aac, from '<...path...>\TroyBoi - No Substitute ft. Y.A.S (256 kbit_s).aac':
  Duration: 00:04:03.79, bitrate: 246 kb/s
    Stream #0:0: Audio: aac (LC), 44100 Hz, stereo, fltp, 246 kb/s
    <streams>
        <stream index="0" codec_name="aac" codec_long_name="AAC (Advanced Audio Coding)" profile="LC" codec_type="audio" codec_time_base="1/44100" codec_tag_string="[0][0][0][0]" codec_tag="0x0000" sample_fmt="fltp" sample_rate="44100" channels="2" channel_layout="stereo" bits_per_sample="0" r_frame_rate="0/0" avg_frame_rate="0/0" time_base="1/28224000" duration_ts="6880770114" duration="243.791458" bit_rate="246339">
            <disposition default="0" dub="0" original="0" comment="0" lyrics="0" karaoke="0" forced="0" hearing_impaired="0" visual_impaired="0" clean_effects="0" attached_pic="0" timed_thumbnails="0"/>
        </stream>
    </streams>

    <chapters>
    </chapters>

    <format filename="<...path...>\TroyBoi - No Substitute ft. Y.A.S (256 kbit_s).aac" nb_streams="1" nb_programs="0" format_name="aac" format_long_name="raw ADTS AAC (Advanced Audio Coding)" duration="243.791458" size="7506918" bit_rate="246339" probe_score="51"/>
</ffprobe>
which seems to be also contradicting statements to the other players info... "Duration: 00:04:03.79, bitrate: 246 kb/s", 'duration="243.791458" bit_rate="246339"'.
However foobar2000/the ffmpeg component seems to ignore these infos anyway, as it still displays a question mark and no seekbar.
7
3rd Party Plugins - (fb2k) / Re: foo_volsync 0.11
Last post by le_bro -
Probably not that useful in Vista and newer, since recent foobar2000 synchronizes and controls its volume through per-application levels.
I would like to have foobar2000 volume able to synchronize to windows mixer master volume. Why? Because some ASIO drivers does not sync volume to windows mixer (for example E-MU and some of ESI drivers). PS: I don't use WASAPI, it doesn't suit me.
9
Support - (fb2k) / Re: How can I speed up autoplaylists initialization?
Last post by Daeron -
You might want to consider that something inherently flawed won't be so easily solved by just throwing more horsepower at it (it will help, but it might not grant you the magnitude of gains you are wishing for).

Comparing clock speeds between different architectures is somewhat pointless (4GHz on an old Pentium 4 is not the same as on a modern CPU), so you can't just assume higher is better. Maybe you meant single threaded performance?

Cores in general is a complicated topic and you shouldn't just be looking at what a single application will be doing. Surely you won't dedicate a high-end CPU to just create your autoplaylist faster? What else will you be doing on that PC? Gaming? Number crunching? How many of your important software support scaling on multiple cores properly? How many of them will be running concurrently? Did you look at the whole platform costs (motherboard, RAM, CPU cooler, potential upgrade path later)? These to me sound like much more important questions to answer. As far as handling your autoplaylists either of those CPUs will do just fine.

If they don't, you will really have to start addressing your setup in the first place. Probably by testing what happens if you only create autoplaylists one at a time on the fly (such as marc2003's method, or facets have queries you can save too). If that doesn't work, I'd reconsider how (and why) you are doing customdb fields in the first place. Third, the example query you have listed implies to me that there might be something specific about that group of elements (genres, genres families and the artists etc) you listed. If there is, you could tag them as such so you can retrieve them quickly later.

The bottom line is that to me your situation seems like a weird edge-case scenario of combination of things not working all that well together, so I wouldn't expect the devs to be able to give you much pointers without them replicating the exact scenario and starting to swap components in and out to see which mitigates it to what extent. The best candidate for that is - unfortunately - you.
10
AAC - General / Re: Is there any difference between AAC and MP3 for wireless use today?
Last post by saratoga -
Quote
I believe I have read that AAC is a good codec for wireless because it is natively supported with aptX

Aptx is an audio codec used by some Bluetooth hardware. AAC is a different audio codec sometimes used by Bluetooth. If you had hardware that could do AAC, and your files were the right bitrate, and your software could be configured to not transcode, then AAC might be preferred. If you're just using aptx it doesn't matter what the source is.
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