I'm looking for an easy to use lib that will convert an MP3 file to a sequence of int values (and the reverse), preferable without having to dump them all into RAM. A "decode the next 16kB into this buffer" like API would be ideal.
I need C or simple C++ bindings.
A MP3<->RAW filter CLI tool would work but I'd rather not have to keep uncompressed files on disk.
Try libmad or ffmpeg's libavcodec. Both should meet your requirements. The ancient mp3lib which was originally derived from/part of (?) mpg123 has also been resurrected in mplayer with new development and perhaps has the best performance, but probably the ugliest code. :-)
I created a .NET wrapper for mpg123 for use in my projects, and posted it to SourceForge.
It is here.
http://sourceforge.net/projects/mpg123net/
So you say you need it for C/C++ - ok, i posted it so you can check out my sample, that has so little code that is enough for initializing decoder and putting it to work for you.
Related
So all I need is a simple function that sets it up (eating incoming PCM RATE (for example: rate near to 44100) It's channels (for example: 2) and -bits (for example: 16) and desirable 128 kb\s rate) and another one that takes PCM data and encodes it into pure MP3 frames.
I know it looks like a silly homework task but I assure you - it is not.
I hope it will be of help to all C++ developers starting with MP3s.
So can anybody please help me with that?
See the example I gave in your other question for the basic usage of Lame. It should contain everything you need.
It's a long time since i messed with this, but Lame lib contains all you need to do it, check out lame.h , there is some test code you could look into.
Personally, I'd be looking at ffmpeg's libavcodec. There's an example file containing an audio_encode_example which ought to be more or less what you're looking for.
See also this question.
Note that not all ffmpeg packagings include mp3 codec support (patent issues) by default, although there's usually some simple way of enabling it.
GStreamer should definitely be able to handle this.
If you are on Windows you can use the GStreamer Winbuilds to get started.
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.
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.
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.
This question has been in my mind for a few years and I never actually found the answer for this.
What I would like to do is extract the actual waveform/PCM of an MP3 file, so that I can play it using the soundcard (of course).
Ideally I would be experimenting some DSP effects.
My first step was to look into LAME, but I didn't find anything relevant about MP3 decoding in a program or stuff like that.
So I'm asking where I could find something like this.
What language should I use? I was thinking C, but maybe there are programming languages out there that would do the job more efficiently.
Thanks!
Guillaume.
The question boils down to: what are you trying to accomplish?
From the description of your question of decoding an MP3 and playing it on the sound card makes it sounds as if you are trying to make a media player.
However, if your intent is to play around with DSP effects, then it sounds like the question is more about processing the sound rather than decoding MP3s. if that's the case, probably looking into writing plug-ins for existing media players (such as Windows Media Player and Winamp) would be easiest path to what you're trying to accomplish.
Frankly, learning to write your own decoder from scratch is not just a programming problem but a mathematical one, so using existing libraries are the way to go. Talking to the operating system or libraries like DirectSound to output audio seems like unnecessary work if anything. I feel that working on plug-ins for existing players would be the way to go, unless your goal is to make your own media player.
If what you really want to accomplish is playing with audio data, then probably decoding an MP3 to uncompressed PCM using any MP3 decoder, then manipulating it in the language of your choice would accomplish your goal of dealing with effects with sound.
The language choice is going to depend on whether you are going to interact directly with MP3 decoding libraries, or whether you can just use raw audio input, which would allow you to use pretty much any language of your choice.
There was a similar question a while back, Getting started with programmatic audio, where I posted an answer on some basic ways to manipulate audio, such as amplification, changing playback speed, and doing some work with FFT.
libmpg123 should do the trick.
I have been using the Windows Media SDK, not for this purpose, but I am pretty sure there are hooks let that let you intercept the audio stream, or convert MP4 to uncompressed WAV. I used C++.
Lots:
http://www.mp3-tech.org/programmer/decoding.html
Pick your poison...
Also, LAME does decode MP3s (check out --decode option), so you might find something interesting in that source.
-Adam
It really depends what platform you are programming on and what you want to do with the code. If you are on Windows you should look at the windows media format sdk or DirectShow. They should both have the ability to decode mp3 files into the raw waveform. On the Mac, I would expect Quicktime to have this same ability. Others have already suggested source for Linux/open source code.
I would recommend looking at Cubase and Wavelab as both will convert MP3 to WAV etc and allow you to play around with the waveform