Suppose I have .3g2 file. I noticed, they can contain audio track of different encoding (AAC, AMR).
Or, for example, an .m4a file can contain (AAC or ALAC) encoded audio track.
MediaInfo detects it pretty well, but I want to be able to do that using C++.
My question is, how can I detect the type of the audio track in a media file?
Thanks.
MediaInfo is also available with a C++ interface, just download MediaInfo library package, and here is a C++ example.
For getting the first audio track format: MediaInfo::Get(Stream_Audio, 0, "Format")
Related
I'm currently using SFML Audio for my visualizer and it works great but some of the limitations are causing problems. Currently my music library consists of MP3 and M4A files and I'm not keen on having to convert them every time I make additions to the library. I'm also looking for a way to read metadata from MP3 and M4A files so that I can list the title, album, and artist. I would you iTunes' exported playlist to get that info yet it only lists the title of the song. Since this is a visualizer I also need to be able to get at least 8192 samples per channel at the current audio position in order to produce the visualization. And lastly I'm hoping to be able to stream media instead of loading it all at once so that there's not a short pause when switching songs. What are some alternatives that support these features?
Must work with Windows.
C++ Library for loading/playing audio (I can make due with C if need be)
Supports MP3/M4A
Supports Reading metadata (Optionally on alternative library)
Supports getting sample data at the current position in the song (at least 8192 samples in advance)
Supports streaming media instead of loading it all at once to prevent lag on switching songs.
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.
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.
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