I've got a .wav file with a perfect square wave (PCM data with only "FF" or "00" bytes) and I'm positive there is some kind of encoded message in it.
I've tried everything I could think of to extract the encoded message in the file. From steganography to several different encoding schemes like NRZ, Manchester, Differential Manchester and got nothing.
I'm three days into analysis of the file and driving mad by now.
Can any of you think of some way (or software) that can perform any kind of analysis on uncompressed pcm data?
P.S.: The decoding of the file is part of a quiz with various steps and that is the reason I would rather not post the file or ask for a direct answer.
I want to get there myself, just needing someone pointing me in a new direction or fresh thoughts about the problem! :D
In te meanwhile I got some invaluable help and figured out it was a tape from a zx spectrum in form of a .wav file.
So all it took was downloading some emulator and loading the tape (.wav file)!
But thank you so much for helping Mike! :D
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I have some audio files that are in this mystery format I can't quite figure out. Here is a list of things I have tried:
Opening them up with a Hex Editor
Gary Kessler's File Signatures list
Audacity (opened as raw data)
Awave Studio
SoX (Sound eXchange)
TrID
DROID
Thanks to the answer by nneonneo here it sounds like my samples are somewhere between a typical compressed format and a speech format.
Here is an example, and this is what it should sound like. I'm at a loss for anything else I can try- most suggestions i've seen in my searches don't seem to work for headerless formats like this.
I would be delighted if anyone knows of any good tools or even if there was a paid service available for something like this.
How to create simple mp3 file? It can be just one tone 440hz playing for one minute for example. I know that there is the specification of mp3 but it's not so obvious and it is easy to do some mistake and receive incorrect format. May be someone already have any example of source code that create such simple mp3 file? Thank you :)
I've been asked to sample some data in a .wac file type. I'm not familiar with this standard and there is very little on the internet with regards to this format. I got given the .wav file but I don't think it was converted correctly, in that there was a none existent of the RIFF header so no .wav reader was able to read it.
Could anyone therefore shed some light into how I could possibly convert the .wac file into a .wav file? Doing some research, I cannot seem to find a converter tool on the internet, and, MatLab does not have a module for reading in .wac data.
NOTE: I've put the tag "game-engine" because according to this website: Here it is used in the infinity game engine.
I've come up with the following solution, however, massive thanks to #jpaari for his input.
Basically, I used sox:
sox -r 44100 -e unsigned -b 8 -c 1 input.raw output.wav
I was able to re-name the file to .raw and this worked. I'm going to update the Sample Rate to what #Aybe posted.
Try this http://www.shsforums.net/topic/39117-ps-gui-v304/
I think Audacity can do it aswell. Also the "unity3d" tag is not quite right.
I would like to create a utility in either PHP or Perl to convert an audio file created by the Nortel's Callpilot voice mail system into a wave file. The problem is that the format, which has the .vbk file extension, is unknown to virtually any audio player. To date, I have not found one that will play a .vbk file. I've looked at audio file conversion libraries in CPAN and tried many of them, they don't recognize the file. I was not successful with PHP's audio formats manipulation either. Nortel does provide a converter, however, it does not suite my needs. I would like to have this run via cron on a CentOS system. I don't know how to reverse engineer this format. There seems to be just scraps of info on this format on the web. This page indicates that it is "based on the H.232 format":
https://www.odesk.com/o/jobs/job/Reverse-Engineer-Nortel-VBK-Audio-Format_~~f501f11679f3f6bb/
I know this is a very old thread, but I've recently been looking into converting Nortel's vbk format as well. Importing the vbk files into Audacity with raw data option, Encoding: U-Law, Byte order: little-endian, Channels: 1 Channel (Mono), Sample rate: 8000 Hz. Not sure if they have multiple formats for their vbk files, but mine were from a BCM50 phone system.
Well, this is the joy of closed proprietary systems. But there is a chance they could play nice. Try to contact Callpilot and see if they'll give you the format specs. It's worth a shot.
As for reverse engineering, you need to be able to generate known content. Like a constant tone at 60Hz for exactly 1 second. Then at 50Hz. Then at 10 seconds. Compare them. Isolate the data from the metadata. There is going to be compression involved, so try a handful of common compression schemes, maybe research into Nortel's practices will probably tell you more. If you can feed that into a player and get a tone back out, you're on your way.
There's probably more informed and structured ways to go about reverse engineering, but from my experience it's a lot of trial and error.
i am wondering if there is any way to cycle through a .wav file to get the amplitude/DB of a specific point in the wav file. i am reading it into a byte array now but that has no help to me from what i can see.
i am using this in conjunction with some hardware i have developed that encodes light data into binary and outputs audio. i wont get into the details but i need to be able to do this in c# or c++. i cant find any info on this anywhere. i have never programmed anything relating to audio so excuse me if this is a very easy thing.
i dont have anything started since this is the starting point so if anybody can point me to some functions, libraries, or methods to being able to collect the amplitude of the wave at a specific time in the file, i would greatly appreciate it.
i hope this is enough info, and thank you in advance if you are kind enough to help.
It is possible and it is done in a straightforward way: the file with PCM audio contains one value for every channel, for every (1/sample-rate) of second.
The values however might vary: 8-bit, 16-bit, single precision floating point values. You certainly have to take this into account and this is the reason you cannot take the bytes from byte array directly.
The .WAV file also has a header preceding the actual payload.