Are there any audio formats that are stored in a plain text file? For me it is easier to understand how they are stored and how they are read, rather than reading documentation on binary formats.
No, there is not. You can however easily, using for example a Python interpreter, print out any binary file as numbers.
Just use a hex editor! Much of the relevant information is written in the file header in a human-readable form. Especially with WAV, AIFF or SD2 files.
Related
My output (csv/json) from my newly-created program (using .NET framework 4.6) need to be converted to a IBM-1027-codepage-binary-file (to be imported to Japanese client's IBM mainframe),
I've search the internet and know that Microsoft doesn't have equivalent to IBM-1027 code page.
So how could I output a IBM-1027-codepage-binary-file if I have an UTF-8 CSV/json file in my hand?
I'm asking around for other solutions, but for now, I think I'm going to have to suggest you do the conversion manually; I assume whichever language you're using allows you to do a hex conversion, at worst. For mainframes, the codepage is usually implicit in the dataset, it isn't something that is included in the file header.
So, what you can do is build a conversion table, from https://www.ibm.com/support/knowledgecenter/en/SSEQ5Y_5.9.0/com.ibm.pcomm.doc/reference/html/hcp_reference26.htm. Grab a character from your json/csv file, convert to the appropriate hex digits, and write those hex digits to a file. Repeat until EOF. (Note to actually write the hex data, not the ascii representation of the hex data.) Make sure that when the client transfers the file to their system, they perform a binary transfer.
If you wanted to get more complicated than that, you could look at enhancing/overriding part of the converter to CP500, which does exist on Microsoft Windows. One of the design points for EBCDIC was to make doing character conversions as simple as possible, so many of the CP500 characters hex representations are the same as the CP1027, with the exception of the Kanji characters.
This is a separate answer, from a colleague; I don't have the ability to validate it, I'm afraid.
transfer the file to the host in raw mode, just tag it as ccsid 1208
(edited)
for uss export _BPXK_AUTOCVT=ALL
oedit/obrowse handles it automatically.
sorry for this not being a programming question directly, but more indirectly as i try to batch convert audio files, which is proving difficult.
I have an audio file which i exported from a package. This audio file is of the RIFF WAVE format. As far as i have read up on headers, normal headers are 44 bytes long. Which contains the sub parts "fmt " and "data". However, this header shows all kind of weird junk, which i cannot actually place anywhere.
If anyone is an audio guru of sorts, please help me out on how to make this audio file accessible for most audio players? i do not care to lose some of the header data as long as it plays the actual content.
Here is a screenshot of my current header data unaltered:
Thanks in advance.
44Bytes is the size of a minimal Wav File header. The format allows for other data chunks in the header in addition to the Riff, fmt and data chunks.
It looks like you have some cue information in your file. This is not a problem, most audio players should accept a wav file with these chunks.
How to write cues/markers to a WAV file in .NET discusses how to add a cue chunk to a file.
http://www.sonicspot.com/guide/wavefiles.html covers some of the additional chunks a wav file can have.
Mike
Turns out this WAVE thing is just a container, and it actually contains a .ogg. I used ww2ogg 3rd party tool to get out these .ogg files as wave. Thanks for all the help though!
According to http://en.wikipedia.org/wiki/WAV there is a table of wave files with different comperssion. You can just investigate in HEX editor a value of AudioFormat field of fmt chunk, to get a list of most common codecs used for compression.
I would like to compress scanned text (monochrome or few colours) and store it in pdf (maybe djvu) files. I remember that I got very good results with Windows/Acrobat and "ZRLE" compressed monochrome tiff embedded into pdf. The algorithm was loossless as far as I remember. Now I search a way to obtain good results on linux. It should be storage saving and avoid loss (I do not mind loosing colours, but I do not want e.g. jpeg compression which would create noisy results for text scans). I need it for batch conversion, so I was thinking of the ImageMagick convert command. But which output format should I use so I get good results and to be able to embed it into pdf files (for example using pdflatex)? Or is it generally better to use djvu files?
jbig2enc encoder for images using jbig2 compression,
was originally written for GoogleBooks by Adam Langley
https://github.com/agl/jbig2enc
I forked to include latest improvements By Rubypdf and others
https://github.com/DingoDog/jbig2enc
I also built several binaries of jbig2enc for puppy linux (it can be working also on other distributions)
http://dokupuppylinux.info/programs:encoders
DJVU is not a bad choice, but if you want to stay in PDF for better compatibility you may want to look into lossless JBIG2 compression.
Quote from Wikipedia:
Overall, the algorithm used by JBIG2 to compress text is very similar
to the JB2 compression scheme used in the DjVu file format for coding
binary images.
I am looking for a way to automatically extract parts from audio files. Something like Imagemagick for audio files.
I only need to extract random parts of a fixed length from a large set of complete ogg-vorbis files. I easily know how to automatically interpret the output from a programm, so I would be able to write a small script if I had programs to do the following:
Get the length of the file
Extract parts of the given an offset in seconds and a length
Is there any program, which allows me to do this under linux? The files I am using are ogg vorbis files.
If there is a python library, which is able to do this, it would work as well.
You can use SoX (Sound eXchange) to do both.
I'm looking for a solution to this task: I want to open any audio file (MP3,FLAC,WAV), then proceed it to the extracted form and hash this data. The thing is: I don't know how to get this extracted audio data. DirectX could do the job, right? And also, I suppose if I have fo example two MP3 files, both 320kbps and only ID3 tags differ and there's a garbage inside on of the files mixed with audio data (MP3 format allows garbage to be inside) and I extract both files, I should get the exactly same audio data, right? I'd only differ if one file is 128 and the other 320, for example. Okay so, the question is, is there a way to use DirectX to get this extracted audio data? I imagine it'd be some function returning byte array or something. Also, it would be handy to just extract whole file without playback. I want to process hundreds of files so 3-10min/s each (if files have to be played at natural speed for decoding) is way worse that one second for each file (only extracting)
I hope my question is understandable.
Thanks a lot for answers,
Aaron
Use http://sox.sourceforge.net/ (multiplatform). It's faster than realtime as you'd like, and it's designed for batch mode much more than DirectX. For example, sox -r 48k -b 16 -L -c 1 in.mp3 out.raw. Loop that over your hundreds of files using whatever scripting language you like (bash, python, .bat, ...).