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Recent Posts
11
WavPack / Re: playing wv (wavpack) files on ipod classic
Last post by HA-User -
i see.  the rip to wavpack has nothing to do with the project of listening on my ipod.  i want a permanent copy of my music.  and it is wonderful to be able to insert artwork and the rest of the metadata into it.

i would be doing that, irregardless of how i was gonna listen to it on my ipod.  for my portable listening, the lifetime goal is to have everything that i want to listen to on an ipod, converted to lossless m4a, and listen to it.  from my calculations, i think it is highly unlikely that i will ever have more songs than would fit on a terrabyte ipod.  but if i did, it wont be for a very LONG time !!
12
General - (fb2k) / Playback Interrupted When Changing Rating
Last post by Dryst -
This is an issue I recently noticed only when playing back APE files.  I do not see the issue at all when playing back FLAC, MP3 or OGG files.  When I make a change to the rating of a currently playing file, the audio gets interrupted briefly.  It is a short interruption, but it always happens.  Are there any changes I may make to help prevent this?  It's as if a process is blocking the audio file while making the change.  The rating change is taking too long.

I am using foo_input_monkey that I recently updated to the latest at the time-- 2.5

14
3rd Party Plugins - (fb2k) / Re: MathAudio Room EQ for Foobar2000
Last post by chr1 -
I have just started using the MathAudio RoomEQ plugin for Foobar on my PC, output going to Dx3Pro via USB. As the output level going to the DX3Pro is now significantly lower, I am wondering if it is OK to increase the signal to the DAC by upping the Preamp gain in the Foobar settings? The screenshot below shows Foobars VU meters playing at a very low level now. Should I just up the preamp settings such that the VU meters are at a more normal level. (Prior to the use of the MathAudio plugin, I had the preamp settings at - 6db as I read somewhere that this is best to avoid any chance of clipping.) Unfortunately my knowledge of digital clipping/gain matching between Foobar and DAC is minimal (at best!)...
Anyway, any help or advice would be most appreciated. Thanks in advance!
15
3rd Party Plugins - (fb2k) / Foobar2000 Preamp gain settings with MathAudio RoomEQ
Last post by chr1 -
I have just started using the MathAudio RoomEQ plugin for Foobar on my PC, output going to a Topping Dx3Pro DAC via USB. As the output level going to the DX3Pro is now significantly lower, I am wondering if it is OK to increase the signal to the DAC by upping the Preamp gain in the Foobar settings? The attached screenshot below shows Foobars VU meters playing at a very low level now. Should I just up the preamp settings such that the VU meters are at a more normal level. (Prior to the use of the MathAudio plugin, I had the preamp settings at - 6db as I read somewhere that this is best to avoid any chance of clipping.) Unfortunately my knowledge of digital clipping/gain matching between Foobar and DAC is minimal (at best!)...
Anyway, any help or advice would be most appreciated. Thanks in advance!
16
Support - (fb2k) / Re: Assign sequence of keyboard strokes to one key
Last post by ojdo -
What you describe cannot be accomplished from within fb2k. But there is AutoHotkey which was created for exactly you use case. It allows you to write short scripts that automate any sequence of mouse + keyboard actions, record macros and even interact with complex GUIs. Those scripts can then be mapped to any keyboard shortcut, which are either system-wide or application-specific. Yes, there is a learning curve, but there is an active community with plenty of documentation, support and examples.
17
AAC / Re: Container for existing raw aac files?
Last post by jaybeee -
As mentioned, ffmpeg will work, but I prefer to use MP4Box (downloads). I created a (windows) script a number of years ago to do this:

Code: [Select]
:: Name:     aac2m4a.cmd
:: Purpose:  Configures mp4box to package raw aac audio into m4a file
:: Author:   jaybeee
:: Revision: June 2021 - v2

@ECHO OFF

SETLOCAL ENABLEEXTENSIONS ENABLEDELAYEDEXPANSION

:: variables begin with v

:: set name of this script without file extension
SET vMe=%~n0

:: set name of the parent directory where this script resides
SET vParent=%~dp0

:: set location of mp4box ** CHANGE ME **
SET vmp4box="C:\Program Files\GPAC\mp4box.exe"

:: call mp4box to place all raw aac files in current directory in m4a container and save to m4a directory
MKDIR m4a
FOR %%f IN (%*) DO %vmp4box% -add %%f "m4a\%%~nf.m4a" -new

:: if you prefer to process the entire directory, then uncomment below command (remove ::), making sure to comment the above FOR command
::FOR %%f IN ("*.aac") DO %vmp4box% -add "%%f" "m4a\%%~nf.m4a" -new

:: Finish
ECHO Finished m4a creation

:: pause can be used to view the extraction details
PAUSE

:END
ENDLOCAL
ECHO ON
@EXIT /B 0

Copy and paste the above into a text file and rename it aac2m4a.cmd.
Then drag and drop the aac file(s) onto the aac2m4a.cmd file and they'll be individually (losslessly) remuxed into a m4a container and placed into a m4a directory.
Or, as you can see from the script, you can process an entire directory by double-clicking the aac2m4a.cmd after having made the changes to the script (uncomment one command and comment the other).

Simple but effective.
19
General - (fb2k) / Understanding FB2K Converter audio processing chain
Last post by Alexxander -
Hi,

As I’m looking to convert mainly from FLAC to AAC (qaac used) with single target level -14 LUFS, I’m looking to understand the audio processing chain that's behind the Converter Setup >> Processing screen.

I’ve read tens of posts and didn’t find a clear answer.

The Hydrogenaudio Wiki Foobar2000:Converter page states:

Quote
ReplayGain
    Select the relevant ReplayGain settings to apply to the converted content. This will permanently apply the change to the output file.
DSP
    Apply the effects of any available DSPs to the converted content. Again, this will permanently apply the change to the output file.

So it’s saying clearly that RG and active DSPs are applied to the AAC file.

However, I did some testing and my conclusion is the opposite: RG and DSP is applied before the encoding to AAC. If for example I set -20dB at preamp, the bitrate of the created AAC is lower than done without preamp.

On the other hand, if I scan and apply RG to an already existing AAC file, the average bitrate remains unchanged (as expected).

The audio processing chain of the Converter for example FLAC to AAC is like this?

    FLAC > WAV or PCM (not sure) > RG and/or Preamp > Active DSPs > Encode to AAC

As a side note, of RG and DSP applied before encoding, could encoding to AAC not cause clipping in some specific cases?

Many thanks for clarifying

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