Maybe Jac or someone else can help me with this issue with DCC-Studio I am having.
The software crashes when it starts writing the audio file to tape. Or playing it in Audio Editor.
This is what I am doing:
I have FLAC files from an album. I am doing the audioprocessing on Linux by the way, simply because that machine is a lot faster than the old WindowsMe machine I have which is running the DCC-Studio program and has the parallel port.
First I extract the FLAC files to WAV.
$ flac -d babylon.flac
This gives me the WAV file.
Then I use the (linux version of) mp2enc:
$ mp2enc -v 1 -b 384 -l 1 -r 48000 -o babylon.mp1 < babylon.wav
This leaves me with the MP1 file.
I use (under wine) dccu.exe to create the LVL TRK and MPP.
$ wine dccu.exe babylon.mp1
After this I have:
babylon.lvl
babylon.mpp
babylon.trk
Those 3 files I put in the c:/studio/audio/ directory of my WindowsME machine. When I open DCC-Studio - Audio Editor and then start playing the audiofile or write it to tape the program crashes with an âAssertion Violationâ.
Alternative step 2:
I also tried the new version of mp2enc Jac uploaded 18 days ago and per the README I used this commandline in step 2 instead:
$ wine mp2enc.exe -o babylon.trk < babylon.wav
but then I get no MPP and no LVL, so I can not import the audio into DCC-Studio.
So what could be the problem?
-edit- Oh, to be clear, the software does work correctly and writes tapes using other audio files, like the ones I capture (dump) from a DCC cassette.
Never mind, I think I figured it out. I was using a script I made that read the content of a directory and then processed all the files in it.
I preserved the (long) filenames, but then the files donât work with Studio, as stated above. I will do tests with more files, and improve my script, report back here and hopefully this thread can be closed
The DCCU program doesnât work correctly with long file names yet, I think. If you create a TRK file and it refers to an LVL file with a name it canât open (e.g because the LVL file has a long file name or doesnât exist), it crashes.
Another things for those reading along: I recently discovered that PASC regards the extra slots in padded MP1 frames in a 44.1 kHz file as âdummyâ. This doesnât apply to you @pvdm but if you convert a 44.1 kHz MP1 file with DCCU, it will cause problems in DCC-Studio: It will play with distortions and you canât seek in the file.
The recommended way to convert WAV files to DCC-Studio using the latest version of mp2enc:
mp2enc -a âArtist Nameâ -t âSong Titleâ -o â\path\to\outputfile.trkâ <â\path\to\input.wavâ
(The above is a single command).
You can add other parameters such as -r to override the sample rate if the program canât guess (it automatically converts 64 to 32, 88.2 to 44.1 and 96 to 48). If the input file is mono, youâll also have to add -s to convert to stereo.
If used with the above command format, mp2enc will generate DCC-Studio files, including a LVL file with actual VU meter values in it, and a TRK file that has the given artist name and song name, correctly cut off at 40 characters. The file names for the LVL and MPP files are short file names that it generates automatically based on the files that are already in the directory (you can override the number of the short file on the command line too; use mp2enc -h for more info).
I donât use Linux on any of my PCâs (maybe next year when itâs the year of the Linux desktop ) so I donât know if mp2enc works on Wine. I expect that it does.
With a smartly composed command (in Batch language or in Bash or whatever), the song names should be easy to generate.
Iâm working on adding PASC compatibility to other software such as the TwoLame library. Once thatâs done, youâll be able to convert any file thatâs supported by libsndfile to DCC-Studio. Stay tuned!
Iâm working on a modification for the TwoLame library to encode MP1. TwoLame is used by many projects (including FFMPEG) for MP2 encoding. If I can push the changes upstream, MP1 and PASC encoding will be featured in many pieces of software.
TwoLame includes a sample program (called âfrontendâ) which reads audio files using libsndfile. That makes it possible to load audio files of many formats including WAV, FLAC and others. Currently it has a problem that it reads the audio as 16 bit PCM so the sample program canât encode 24-bit files at the best quality, but that should be easy to fix.
Also the TwoLame sample program doesnât support sample rate conversion but in the process of my modifications, Iâll probably add an option to use libsrc to the sample program to do that.
Stay tuned to the forum and/or keep an eye on the OneLame repository.