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.
Related
I don't have an opportunity to buy Cubase, but my partner uses it a lot. I wanted to simplify his life and provide him with cpr projects instead of plain wav files, but no other software can open/save this format.
I looked at a sample cpr he sent me and it seems like the file does not contain audio data itself, it rather contains the mark-up and effects.
I wanted to know the following things:
Is it legal to try to reverse-engineer cpr files?
Is it difficult and who tried?
If someone knows other ways to transfer project files between Audacity/Rosegarden and Cubase? The main thing is the support of several tracks and their timing in one project, nothing fancy.
Cpr files comes from a proprietary format. You can have a look on this question.
I suppose it is pretty hard... and I didn't tried !
To my knowledge, there is no way to export/import a project between cubase and Audacity or Rosegarden. The OMF format which could be a good candidate, is not supported by Audacity or Rosegarden for now. You can still import/export the audio mix, the separated tracks, and the midi files separately. This method is really fastidious, but it probably provides the advantage to let you play and edit your projects in the next decades, that isn't obvious with project files.
guys!
I've been asked at work to prepare a large audio library for tests, which includes different files of HE-AACv1 & HE-AACv2. To be specific, all sample rates from 8000 up to 48000 and bit rates from 8 bit to 32 bit. But I've been digging the internet for a while and can't find the exact information. Is it even possible to make HE-AAC with 8KHz sample rate? And are there any tools that allow to directly set this specific values (bit rate and sample rate)? I've tried adobe audition, audacity and a bunch of something I found on google, but neither gives me opportunities that I need, or it's just me being dumb.
Please, if you know the way to create a HE-AACv1 and v2 from for example mp3 or wav, tell me how to do it, I would be very grateful.
Thx!
Adobe Audition can definitely create all WAVs that you need, look under "Edit->Convert sample type". You should first create your wav files, and then use some encoder to transcode to AAC.
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
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 looking to record voice in as compact a file format as possible for an ipad app, and not concerned about sound quality. I chose the ima4 format but don't really know much about audio, so am having trouble figuring out how to play back the produced file to test how it sounds. Is this a compressed format that I have to uncompress with some tool in order to just listen to it? Is this the right format if I want something compact and reasonably coherent but not worried about great quality?
Apparently, I had to save it as an .aif, .aiff, or .aifc file which then was playable by common players like iTunes.