How can I, using a function, library, whatever I have to, concatenate two .wav files? The input should be the absolute paths, and the output an audio file created and placed (not just played) somewhere, it doesn't really matter where.
I am writing a Mac command line application in XCode 6.
The .wav file format is a very simple format, consisting of the fixed header that defines the audio file's properties; namely the endian-ness, the number of channels, and the sampling rate. Its documentation is widely defined on the intertubes.
Off the top of my head I don't recall if any common library offers a convenient way to do this (it's worth looking through libsndfile's API documentation, for something that would fit the bill).
In any case, it shouldn't be too tough to read the headers of both WAV files, to check their format, and then create the output file. If both WAV files have the same endian-ness, number of channels, and sampling rate, the procedure is trivial, otherwise you will have to resample/remix at least one of the files.
There is a very simple, lightweight and mature open source C API library for reading-writing several common audio file formats. I haven't worked with it for a while, if I remember well, it has routines for opening a sound file for writing, seeking the end, appending data from another file and updating the header. I hope this can help.
Related
I've seen a lot of examples of i/o with text files I'm just wondering if you can do the same with other file types like mp3's, jpg's, zip files, etc..?
Will iostream and fstream work for all of these or do I need another library? Do I need a new sdk?
It's all binary data so I'd think it would be that simple. But I've been unpleasently surprised before.
Could I convert all files to text or binary?
It depend on what you mean by "work"
You can think of those files as a book written in Greek.
If you want to just mess with binary representation (display text in Greek on screen) then yes, you can do that.
If you want to actually extract some info: edit video stream, remove voice from audio (actually understand what is written), then you would need to either parse file format yourself (learn Greek) or use some specialized library (hire a translator).
Either way, filestreams are suited to actually access those files data (and many libraries do use them under the hood)
You can work on binary streams by opening them with openmode binary :
ifstream ifs("mydata.mp3", ios_base::binary);
Then you read and write any binary content. However, if you need to generate or modify such content, play a video or display a piture, the you you need to know the inner details of the format you are using. This can be exremely complex, so a library would be recomended. And even with a library, advanced programming skills are required.
Examples of open source libraries: ffmpeg for usual audio/video format, portaudio for audio, CImg for image processing (in C++), libpng for png graphic format, lipjpeg for jpeg. Note that most libraries offer a C api.
Some OS also supports some native file types (example, windows bitmaps).
You can open these files using fstream, but the important thing to note is you must be intricately aware of what is contained within the file in order to process it.
If you just want to open it and spit out junk, then you can definitely just start at the first line of the file and exhaustively push all data into your console.
If you know what the file looks like on the inside, then you can process it just as you would any other file.
There may be specific libraries for processing specific files, but the fstream library will allow you to access any file you'd like.
All files are just bytes. There's nothing stopping you from reading/writing those bytes however you see fit.
The trick is doing something useful with those bytes. You could read the bytes from a .jpg file, for example, but you have to know what those bytes mean, and that's complicated. Usually it's best to use libraries written by people who know about the format in question, and let them deal with that complexity.
i'm a bit at loss here. My goal is to merge two video files (which might be of different file formats) and i'm already using libffmpeg for other simple tasks. I thought libffmpeg exposed some kind of function to merge files, but i can't find it.
I found these pages on the documentation that might be relevant: http://ffmpeg.org/doxygen/trunk/structConcatStream.html and http://ffmpeg.org/doxygen/trunk/group__lavf__encoding.html
I'm not sure if this is really relevant though? Can anybody point me in the right direction? Do i need to use FFmpeg muxing and manually joins streams? Is there any example that can explain to me what i should do? thanks!
For those looking for an example, i ended up using
How to use libavformat to concat 2 video files with same codec (re-muxing)?
there's a nice snippet and it works very well
Use ffmpeg to open file 1, start reading frames, converting to target format, and writing to the output file. When there are no more frames, close file 1 (leave output open). Open file 2, start reading frames, converting to target format, and writing to the output file. When there are no more frames, close file 2 and close output.
Merged and formats reconciled.
I've seen a lot of games use something similar to a .DAT file or a specific file type that the game has for itself. I'm just beginning with C++ and DirectX and I was interested in keeping my information in something similar to a .DAT.
My initial conception was that it would hold information on the files you wanted to store within the .DAT file. Something similar to a .RAR file. Unfortunately, my googleing skills did not help me in finding the answers.
Right now I'm simply loading textures and sound files from a folder called Data.
EDIT: While I understand that .DAT is short for data, and I've found that a .DAT file generally contains any assortment of information, I'm still unsure about how to go about doing something as packing images and sound files into any type of file and being able to read them.
I'm not sure about using fstreams to achieve my task, however I will look into streams related to storing data and how to properly read from that data. Meanwhile if anyone has another answer to offer based on this new information, it would be appreciated.
EDIT: Thanks to the answers, I stumbled across a similar question on stackoverflow and felt I'd share it here. Combining resources into a single binary file
I don't think there is really such thing as .dat file format. It's short for "data," and different applications just put in some proprietary stuff in it and call it ".dat." You can read up on fstream classes to do file IO in C++. See Input/Output with files.
What you then do is make up your own file format. For example, first 4 byte is int that indicates the number of blocks in the .dat and for each block, you have 4 byte indicating the length of each block, 4 byte indicating the type of the block, the variable length data itself .. something like that.
DAT obviously stands for data, and there is no real or de facto standard on what that extension actually refers to. Your decisions on the best file formats should be based on technical considerations, not pointless attempts at security through obscurity.
Professional games use a technique where they put all the needed resources (models, textures, sounds, ai, config, etc) zipped/packed into a single file thus making it faster to manage, harder to change (some even make use of a virtual filing system from what's inside the data file). Now, for what's inside the file is different depending on the needs of the game and the data structures that you use.
If you're just starting into gamedev, i recommend you stick with keeping all you assets separate and don't bother too much about packing them into a single file.
Now if you really want to start using a packed format here's a good pointer:
Creating a PAK File Format
Here's a link which claims that .dat is a movie format, 'DAT' being short for Digital Audio Tape.
I'm not sure I believe the link, but I do remember something about a Microsoft supported format called DAT, from long ago, when I used an earlier version of Windows.
It makes more sense as a logical extension for a DATA file of some kind.
.dat, as others have said, is literally just a data file. In reality, the file extension means nothing other than association with a program. For example, I could make a word processor that saves all the documents with the .mp3 file extension. These files wouldn't be playable in any media software, but the software might try. File extensions are used to help programs know what types of files they can and cannot open--however those rules don't have to be followed.
Anyway, you can dump any sort of information to a file. Programmers/software writers will often choose .dat as the extension of that file because it has become the standard to signify 'this file just holds a ton of data' and that the data doesn't necessarily hold any standardized headers, footers, or formatting.
A dat file could really contain anything. It might be as simple as a zip archive with the extension changed, or it could be a completely custom file type. If you're just starting out, you probably don't want to write your own file format, although doing so can be fun and educational. If you want to encapsulate your data files into some kind of container, you should probably go with a zip, paq, or maybe tar.gz.
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.
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.