How do you programmatically compress a WAV file to another format (PCM, 11,025 KHz sampling rate, etc.)?
I'd look into audacity... I'm pretty sure they don't have a command line utility that can do it, but they may have a library...
Update:
It looks like they use libsndfile, which is released under the LGPL. I for one, would probably just try using that.
Use sox (Sound eXchange : universal sound sample translator) in Linux:
SoX is a command line program that can convert most popular audio files to most other popular audio file formats. It can optionally
change the audio sample data type and apply one or more sound effects to the file during this translation.
If you mean how do you compress the PCM data to a different audio format then there are a variety of libraries you can use to do this, depending on the platform(s) that you want to support. If you just want to change the sample rate of the PCM data then you need a sample rate conversion algorithm instead, which is a completely different problem. Can you be more specific in your requirements?
You're asking about resampling, and more specifically downsampling, not compression. While both processes are lossy (meaning that you will suffer loss of information), downsampling works on raw samples instead of in the frequency domain.
If you are interested in doing compression, then you should look into lame or OGG vorbis libraries; you are no doubt familiar with MP3 and OGG technology, though I have a feeling from your question that you are interested in getting back a PCM file with a lower sampling rate.
In that case, you need a resampling library, of which there are a few possibilites. The most widely known is libsamplerate, which I honestly would not recommend due to quality issues not only within the generated audio files, but also of the stability of the code used in the library itself. The other non-commercial possibility is sox, as a few others have mentioned. Depending on the nature of your program, you can either exec sox as a separate process, or you can call it from your own code by using it as a library. I personally have not tried this approach, but I'm working on a product now where we use sox (for upsampling, actually), and we're quite happy with the results.
The other option is to write your own sample rate conversion library, which can be a significant undertaking, but, if you only are interested in converting with an integer factor (ie, from 44.1kHz to 22kHz, or from 44.1kHz to 11kHz), then it is actually very easy, since you only need to strip out every Nth sample.
In Windows, you can make use of the Audio Compression Manager to convert between files (the acm... functions). You will also need a working knowledge of the WAVEFORMAT structure, and WAV file formats. Unfortunately, to write all this yourself will take some time, which is why it may be a good idea to investigate some of the open source options suggested by others.
I have written a my own open source .NET audio library called NAudio that can convert WAV files from one format to another, making use of the ACM codecs that are installed on your machine. I know you have tagged this question with C++, but if .NET is acceptable then this may save you some time. Have a look at the NAudioDemo project for an example of converting files.
Related
I'm not talking about any concrete language here. I want to analyse the MP3 file, so I want to get some information about sound from specific second (i don't know, tone/height/frequency of sound). How those data is stored in single file?
Unless you have weeks (months?) available to play with it, I would recommend using an existing MP3 decoding library to pull the decoded audio out of the file. In C/C++, there's libMAD or libmpg123, as well as the Windows components. In C#, you can use NAudio or NLayer.
Once you have the decoded data, you'll need to run a FFT, DFT, or DCT over it to convert to frequency & amplitude. The FFT is probably your best bet, though the DFT may give a less "noisy" analysis. YMMV.
Note that all three of the transforms provide amplitude values you can convert to decibel values.
there are some useful MP3 Librarys where you get information about your MP3 file.
If you use C# it could be NAudio.
http://naudio.codeplex.com/
I recommend the program xxd and google for the first steps.
First of all i would look into its binary code.
xxd -b file.mp3
Viewing it as ASCII also exposes some information.
xxd file.mp3
That was my first steps.
For a scientific project i need to compress video data. The video however doesn't contain natural video and the quality characteristics of the compression will be different than for natural footage (preservation of hard edges for example is more important than smooth gradients or color correctness).
I'm looking for a library that can be easily integrated in an existing c++ project and that let's me experiment with different video compression algorithms.
Any suggestions?
Look at FFmpeg. It is the the most mature open source tool for video compression and decompression. It comes with a command line tool, and with libraries for codecs and muxers/demuxers that can be statically or dynamically linked.
As satuon already answered, FFmpeg is the go-to solution for all things multimedia. However, I just wanted to suggest an easier path for you than trying to hook your program up to its libraries. It would probably be far easier for you to generate a sequence of raw RGB images within your program, dump each out to disc (perhaps using a ridiculously simple format like PPM), and then use FFmpeg from the command like to compress them into a proper movie.
This workflow might cut down on your prototyping and development time.
As for the specific video codec you will want to use, you have a plethora of options available to you. One of the most important considerations will be: Who needs to be able to play your video and what software will they have available?
How to write mp3 frames (not full mp3 files with ID3 etc) from PCM data?
I have something like PCM data (for ex 100mb) I want to create an array of mp3 frames from that data. How to perform such operation? (for ex with lame or any other opensource encoder)
What do I need:
Open Source Libs for encoding.
Tutorials and blog articles on How to do it, about etc.
You should be able to use LAME. It has a -t command line switch that turns off the INFO header in the output (otherwise present in frame 0). If that still leaves too much bookkeeping data, you should be able to write a separate tool to strip that away.
You are already on the right track: use LAME external executable, or any other shell-invoked encoder.
To build MP frames, were your layer of interest is 3, is not easy to do from scratch. There are compression steps, Fast-fourier transforms followed by quantization, which are of complex and tediously long explanation. The amount of work required for a developer to build it from scratch is very big.
There are programmatic C and C++ MP encoding libs, but you will be either asked for fees, be left with very limited support, or have very limited interfacing options.
Go LAME, study their wiki.
I've always been curious about audio conversion software, but I have never seen a proper explanation from a beginners point of view as to how to write a simple program that converts for example, a mp3 file to a wav. I'm not asking about any of the complex algorithms involved, just a small example using a simple library. Searching on SO, I came up with several names including:
Lame
The Synthesis Toolkit
OpenAL
DirectSound
But I'm unable to find a straightforward example of any of these libraries. Usually I don't mind wading through tons of code, but here I have absolutely no knowledge about the subject and so I always feel like I'm shooting in the dark.
Anyone here have a simple example / tutorial on converting a sound file using any of these libraries? My question is specifically directed towards C/C++ because those are the two languages I'm currently learning and so I'd like to continue to focus on them.
Edit: One thing I forgot to mention: I'm on a *NIX system.
Thanks everyone for the responses! I sort of cobbled them together to successfully make a small utility that converts a AIFF/WAV/etc file to an mp3 file. There seems to be some interest in this question, so here it what I did, step by step:
Step 1:
Download and install the libsndfile library as suggested by James Morris. This library is very easy to use – its only shortcoming is it won't work with mp3 files.
Step 2:
Look inside the 'examples' folder that comes with libsndfile and find generate.c. This gives a nice working example of converting any non-mp3 file to various file formats. It also gives a glimpse of the power behind libsndfile.
Step 3:
Borrowing code from generate.c, I created a c file that just converts an audio file to a .wav file. Here is my code: http://pastie.org/719546
Step 4:
Download and install the LAME encoder. This will install both the libmp3lame library and the lame command-line utility.
Step 5:
Now you can peruse LAME's API or just fork & exec a process to lame to convert your wav file to an mp3 file.
Step 6: Bring out the champagne and caviar!
If there is a better way (I'm sure there is) to do this, please let me know. I personally have never seen a step-by-step roadmap like this so I thought I'd put it out there.
For converting between various formats (except MP3) check libsndfile http://mega-nerd.com/libsndfile/
Libsndfile is a library designed to
allow the reading and writing of many
different sampled sound file formats
(such as MS Windows WAV and the
Apple/SGI AIFF format) through one
standard library interface.
During read and write operations,
formats are seamlessly converted
between the format the application
program has requested or supplied and
the file's data format. The
application programmer can remain
blissfully unaware of issues such as
file endian-ness and data format
It is also simple to use, with the API following the style of the Standard C library function names:
http://mega-nerd.com/libsndfile/api.html
And examples are included in the source distribution.
For actual audio output, another library will be needed, SDL as already mentioned might be a good place to start. While SDL can also read/write audio files, libsndfile is far superior.
If your curious about DSP and computers, take a look at the Synthesis Toolkit. It's sweet. It's designed for learning. The examples and tutorials they have on their website are straightforward and thorough. Keep in mind, the guys who wrote it, wrote it so they could create acoustic models of real instruments. As a result, they've included some instruments that are just plain wacky, but fun. It will give you a core understanding of processing PCM sound. And you'll probably be able to hack together some fun little noisemakers while your at it.
https://ccrma.stanford.edu/software/stk/
Check libmad http://mad.sourceforge.net " "M"peg "A"udio "D"ecoder library", should provide a good example.
Also for an easy cross-platform audio handling, check SDL http://www.libsdl.org/.
Hope that helps.
I want to decode JPEG files and obtain uncompressed decoded output in BMP/RGB format.I am using GNU/Linux, and C/C++.
I had a look at libjpeg, but there seemed not to be any good documentation available.
So my questions are:
Where is documentation on libjpeg?
Can you suggest other C-based jpeg-decompression libraries?
The documentation for libjpeg comes with the source-code. Since you haven't found it yet:
Download the source-code archive and open the file libjpeg.doc. It's a plain ASCII file, not a word document, so better open it in notepad or another ASCII editor.
There are some other .doc files as well. Most of them aren't that interesting though.
Unfortunately I cannot recommend any other library besides libjpeg. I tried a couple of alternatives, but Libjpeg always won. Is pretty easy to work with once you have the basics done. Also it's the most complete and most stable jpeg library out there.
MagickWand is the C API for ImageMagick:
http://imagemagick.org/script/magick-wand.php
I have not used it, but the documentation looks quite extensive.
You should check out Qt's QImage. It has a pretty easy interface that makes this task really easy. Setup is pretty simple for every platform.
If Qt is overkill, you can try Magick++ http://www.imagemagick.org/Magick++/. It supports similar operations and is also well suited for that sort of task. The last time I used it, I struggled a bit with dependencies for it on Windows, but don't recall much trouble on Linux.
For Magick++'s Image class, the function you probably want is getConstPixels.
I have code that you can copy ( or just use as a reference ) for loading a jpeg image using the libjpeg library.
You can browse the code here: http://code.google.com/p/kgui/source/browse/trunk/kguiimage.cpp
Just look for the function LoadJPGImage.
The code is setup to handle c++ binding of my DataHandle class to it for loading the image, that way the image can be a file or data already in memory or whatever.
A slightly out of the box solution is to acquire a copy of the netpbm tools, which transform images from pretty much any format to any other format via one of several very simple intermediate formats. They work well from the shell, and are most often used in pipes to read some arbitrary image, perform an operation on it, and write it out to some other format.
The pbm formats can be as simple as a plain ASCII header followed by the RGB data in ASCII or binary. They are intended to be simple enough to use without required a library to implement.
JPEG is supported in netpbm by read and write filters that are implemented on top of libjpeg.