So I've done some research and suppose, correct me if I'm wrong, a .wav file is basically I/Q(file) samples merged with extra information(header part).
What I'm trying to do is get that data from the .wav file(I already did that, also can convert bytes to samples).
My question here is how can I convert that .wav file to an I/Q file with the data/samples I have? is it possible? if it is, how?
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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 know the algorithm for converting .amr to .wav . What I have done till now is remove the .amr header and replace it with .wav header.The format gets changed but the audio plays some vague sound. I think this is because the raw data of .amr is different. How do I convert this to .wav?
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 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.
What is the difference between compressed and uncompressed .wav files?
The WAV format is a container format for audio files in Windows.
The WAV file consists of a header and the contents. The header contains information about the size, duration, sampling frequency, resolution, and other information about the audio contained in the WAV file. Generally, after the header is the actual audio data.
Since WAV is a container format, the data it contains can be stored in various formats. One of which is uncompressed PCM, but it can also store ADPCM, MP3 and other formats, and can be read and written if an audio codec for the format is available.
The difference between compressed and uncompressed WAV files is that the data contained within the WAV file is either uncompressed raw audio samples, or it is compressed using an audio codec, in which case, it must be decompressed before it can be played back.
Further reading:
Wikipedia: Audio compression (data)
Wikipedia: WAV
Wikipedia: Codec
There's a great explanation here. The basic difference is that an uncompressed wave file has just the raw bits in it as they "appear". There is nothing done to compress or shrink them. A compressed wave file uses some sort of codec to shrink down the data before putting it in the file.
The difference between these two things is basically in the size of object, the compressed one might have low size compared to uncompressed basically the content are the same.
You have to be very careful when using the word "uncompressed" when talking about media.
Basically ALL digital media is compressed in some way. Audio, or video. No matter what it is, it is compressed in some way. Its intrinsic to converting from analog to digital.
The problem isn't really technical, its lingual.
People think that uncompressed means "nothing done to it" when in reality there really isnt any way you can do this. There is always some kind of compression done when you convert the analog signal coming out of the mic and going into a file...Its essential.
What uncompressed means is very high quality. And different "Uncompressed" codecs do things differently.
I know more about video codecs, so i will base my example in those.
Black Magic (A company that makes video Out Cards) has an Uncompressed Codec. Its very good. Makes Beautiful images.. But its not really "uncompressed". Sure its big. But compare it to a DPX of TIFF image sequence...and it aint that big, and is quite compressed. Its only 10 bit, but something like an OpenEXR image sequence is like 32 bit...and coming from film, that is still technically compressed. It has to be.
Its just the nature of the beast.