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Recent Posts
12
General - (fb2k) / Re: APT-X100 Component for Foobar 2.0 and later
Last post by Revup67 -
If you have a working component for 1.6.x, it's highly likely it works just fine with 32bit 2.1.2. I'm not saying you have to use 32bit if you don't want to but it seems like an easy choice if you really like/need that component.
I've avoided the 32 bit install this time around and went with the 64 bit install thus cannot find a current component for APTX100.. thanks for the insight
Description of foo_input_apt-x100
Quote
This plugin is capable to decode from *.AUD and *.AUE files of DTS Movie/Trailer Discs.
This plugin is capable to encode to *.AUD files of DTS Movie/Trailer Discs.
You shouldn't need it to decode DTS-wav. fb2k v2 has built-in DTS decoder already. Make sure that option "Enable additional decoding" (File->Preferences->Advanced->Playback->Enable additional decoding) is enabled. Also, remove obsolete foo_input_dts if you have it installed.
But if you sure you really need 64-bit version of foo_input_apt-x100 you can try to contact its author on Sourceforge - https://sourceforge.net/p/dvdadecoder/feature-requests/
Thanks, that check box was Enabled. No Foo DTS was loaded in 64 bit 2.12.  I read the ffMPEG now includes DTS as well.  APT-X100 is the only component missing from version 2.12 64 bit when compared with 1.61 (32 bit version).  The solution I just chose was to Convert the non listenable large WAV file to FLAC despite the file incompatibility,  I then loaded the FLAC version instead and this played fine, then removed the single WAV file which used DTS Coherent Acoustics as noted in 1.61 during playback.
13
General - (fb2k) / Re: APT-X100 Component for Foobar 2.0 and later
Last post by Revup67 -
If you have a working component for 1.6.x, it's highly likely it works just fine with 32bit 2.1.2. I'm not saying you have to use 32bit if you don't want to but it seems like an easy choice if you really like/need that component.
I've avoided the 32 bit install this time around and went with the 64 bit install thus cannot find a current component for APTX100.. thanks for the insight
14
foobar2000 mobile / Re: Foobar 2000 Android version - objects not found
Last post by racmegenz -
Hey! I am facing the same issue. I usually play albums, a few are in mp3 but occasionally replaygain scanner fails at generating the tag in one or two files per album. Just weird and annoying since this breaks the volume normalization (usually that file stays "loud").
15
General - (fb2k) / Re: APT-X100 Component for Foobar 2.0 and later
Last post by Bogozo -
Description of foo_input_apt-x100
Quote
This plugin is capable to decode from *.AUD and *.AUE files of DTS Movie/Trailer Discs.
This plugin is capable to encode to *.AUD files of DTS Movie/Trailer Discs.
You shouldn't need it to decode DTS-wav. fb2k v2 has built-in DTS decoder already. Make sure that option "Enable additional decoding" (File->Preferences->Advanced->Playback->Enable additional decoding) is enabled. Also, remove obsolete foo_input_dts if you have it installed.
But if you sure you really need 64-bit version of foo_input_apt-x100 you can try to contact its author on Sourceforge - https://sourceforge.net/p/dvdadecoder/feature-requests/
16
General Audio / Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy
Last post by timcupery -
I'd let LAME do it all in one shot.

Thanks Doug. By "let LAME do it all in one shot" do you mean
a) just let LAME default to reproducing the samplerate (at least with 48khz files)
b) set LAME itself to output 44.1khz files as part of the internal process.
If (b), do you know if there's a way to do this within the FB2k converter interface? I'm pretty sure there's a way to specify within command line, but I haven't used that in 15 years.

...I have some "ripped" DVD concerts that I've converted to MP3 and I've left them at 48kHz.  But I'm not trying to play them gapless

I discovered that my phone's music app didn't handle gapless on 48khz files through a similar situation. I had a live album that came with a concert DVD, and the DVD had a few songs that the live audio CD didn't. So I pulled those out and converted, initially to 48 khz for the files from the DVD. I re-did the songs from the DVD to 44.1 after noticing the handling-gapless problem with my phone player.
17
MP3 - General / Re: Resurrecting/Preserving the Helix MP3 encoder
Last post by Case -
There was actually a typo in the original sources. The code was always supposed to throw an error when sample rate was above 48 kHz, but the check was written to fail at 480 kHz.

Here's a test version without hardcoded limits to WAVE header size so the b) problem files can be encoded. MP3 file is no longer created when initialization fails. The error message telling that source file is unsupported will print additional information about each part that fails, be it number of channels, bit depth, WAVE type or sample rate.

No other fixes or improvements for now. I should be sleeping already.
18
General Audio / Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy
Last post by DVDdoug -
I'd let LAME do it all in one shot.

I just did a quick experiment with Kabuu Audio Converter and it handled the compression & resampling fine.   I know some resamplers "measure better" than others but I've never heard a difference, no matter what I was using, and you're converting to lossy anyway.

...I have some "ripped" DVD concerts that I've converted to MP3 and I've left them at 48kHz.  But I'm not trying to play them gapless and I don't have an iPhone.  (MP3 doesn't support 96kHz.)

...My compromise is to make a concert-length file plus separate song files with the applause/crowd noise faded in-and-out so I can play them individually and intermix with my other music.    
20
General Audio / preferred/best method for encoding 48khz FLAC --> 44.1khz lossy
Last post by timcupery -
Sometimes lossless files that I buy are 48 or even 96khz samplerate. I use FB2k to encode to LAME V3 for my listening library, and use the dBpoweramp/SSRC resampler (from the "available DSPs" menu in FB2k's converter) to get 44.1khz mp3s whenever I start with lossless at a higher samplerate.

I've wondered if there is a more efficient way to get from higher-samplerate lossless to 44.1 khz mp3. Perhaps
[highsamplerate FLAC --> resampler --> LAME]
has some downside to the two-step approach, and there's a way to do this as a single step internal to the LAME encoder.

I also have no reason to expect I could ABX any of the differences here, and am not particularly worried about this. But I'm asking this question because folks here (a) understand the math and process steps, and (b) are aware and honest of what can actually matter for differences being noticeable.

It's also possible that there's no good reason to go to 44.1 khz mp3 files. Given that a given lossy preset will have lowpass at a some frequency level, changing the official samplerate of the lossy file doesn't change the bitrate, and 48khz is normal enough that most hardware handles it and 44.1 just fine. The only reason I've stuck with 44.1 khz mp3 files is because gapless algorithm on my phone app (Musicolet) seems to handle live-concert or other continuous-play stuff better if it's all at 44.1 khz.
I also still use EAC's wav editor (which can only handle normal CD files, 44.1 khz stereo), although that doesn't matter in these cases. I can probably find something better and more flexible, but haven't put in the effort to figure out what such a thing would be, and switch.

Thanks for any feedback here!