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.
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.
The Kinect OpenNI library uses a custom video file format to store videos that contain rgb+d information. These videos have the extension *.oni. I am unable to find any information or documentation whatsoever on the ONI video format.
I'm looking for a way to convert a conventional rgb video to a *.oni video. The depth channel can be left blank (ie zeroed out). For example purposes, I have a MPEG-4 encoded .mov file with audio and video channels.
There are no restrictions on how this conversion must be made, I just need to convert it somehow! Ie, imagemagick, ffmpeg, mencoder are all ok, as is custom conversion code in C/C++ etc.
So far, all I can find is one C++ conversion utility in the OpenNI sources. From the looks of it, I this converts from one *.oni file to another though. I've also managed to find a C++ script by a phd student that converts images from a academic database into a *.oni file. Unfortunately the code is in spanish, not one of my native languages.
Any help or pointers much appreciated!
EDIT: As my usecase is a little odd, some explanation may be in order. The OpenNI Drivers (in my case I'm using the excellent Kinect for Matlab library) allow you to specify a *.oni file when creating the Kinect context. This allows you to emulate having a real Kinect attached that is receiving video data - useful when you're testing / developing code (you don't need to have the Kinect attached to do this). In my particular case, we will be using a Kinect in the production environment (process control in a factory environment), but during development all I have is a video file :) Hence wanting to convert to a *.oni file. We aren't using the Depth channel at the moment, hence not caring about it.
I don't have a complete answer for you, but take a look at the NiRecordRaw and NiRecordSynthetic examples in OpenNI/Samples. They demonstrate how to create an ONI with arbitrary or modified data. See how MockDepthGenerator is used in NiRecordSynthetic -- in your case you will need MockImageGenerator.
For more details you may want to ask in the openni-dev google group.
Did you look into this command and its associated documentation
NiConvertXToONI --
NiConvertXToONI opens any recording, takes every node within it, and records it to a new ONI recording. It receives both the input file and the output file from the command line.
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?
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.
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.