I need to translate A-Law companded files to regular un-companded PCM. Is there any source code out there I can look at?
Look at AlawCodec.java.
The interesting things are in the static block of the class where the decoding tables are initialized and the read method.
Related
I am trying to parse very large gzip compressed (10+GB) file in python3. Instead of creating the parse tree, instead I used embedded actions based on the suggestions in this answer.
However, looking at the FileStream code it wants to read the entire file and then parse it. This will not work for big files.
So, this is a two part question.
Can ANTLR4 use a file stream, probably custom, that allows it to read chunks of the file at a time? What should the class interface look like?
Predicated on the above having "yes", would that class need to handle seek operations, which would be a problem if the underlying file is gzip compressed?
Short anser: no, not possible.
Long(er) answer: ANTLR4 can potentially use unlimited lookahead, so it relies on the stream to seek to any position with no delay or parsing speed will drop to nearly a hold. For that reason all runtimes use a normal file stream that reads in the entire file at once.
There were discussions/attempts in the past to create a stream that buffers only part of the input, but I haven't heard of anything that actually works.
I'm looking for method to process simple data into an audio output such as an mp3 file. The data is in the form of a two-column text file with a time signature in milliseconds and a level in millivolts.
Ideally the method would be script-able (with linux or unix tools). I have tried using Audacity to read raw data, however it seems to expect binary files, and doesn't seem to be flexible with sample rates etc.
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.
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, ...).