I've just finished an assessment on Data Structures where one of the questions given was regarding MIDI files. In the descriptor (a criteria for assessors) it states the following:
"Candidates will need to demonstrate their knowledge... by showing they can describe:
At least two Standard File types for images, sounds, video or compression."
Does a MIDI file fall under any of these categories? From what I have searched online, it doesn't. So I am somewhat bemused at a question regarding MIDI files being in this assessment.
Cheers
Not really, no. MIDI tells a synthesizer how to make sounds, where what you can make is constrained by the synthesizer. It cannot be used to represent arbitrary sound data. Though it has been used in a way not originally intended to send small sound samples to a synthesizer.
Compressed sound formats are an entirely different thing from MIDI.
(By the way, the quoted question needs some commas to make sense.)
When I send a MIDI file to my synthesizer, I expect it to produce sounds.
Related
Can effects like fading, distortion, swing, reverb and sustain be added to the music extracted from a midi file on the computer? Or does midi allow us to incorporate these features during it's creation?
Going through the midi specification I couldn't find a possible way to do the above things. Am I correct or am I missing something?
Any suggestions?
MIDI files do not contain sounds; they are instructions for a synthesizer how to generate sounds.
The MIDI specification itself does not define any effects.
The General MIDI specification defines controller numbers for volume, reverb, and sustain effects; many synthesizers are General MIDI compatible.
While some synthesizers have distortion effects, there is no widely accepted standard.
You can simply add or modify controller change messages in the MIDI file. (Whether and how they work depends on your synthesizer.)
"Swing" is based on the timing of the notes. You'd have to change the timestamps of the note-on and note-off messages to affect this.
I want to start on a hobby project that focuses on displaying audio files in a folder in a certain fashion and has the ability to play such an audio file and shows basic control options for playing. However, i'm struggling to find a fit programming language for this.
The displaying part shouldn't be too hard and can probably be done in most of the programming languages. The audio part is what concerns me the most since it's not the main focus of the project and should only do limited things (so it shouldn't be too hard) and i do not know anything about sound support in the programming languages i currently know. (Java, C and C++)
Specifically i would like to be able to do these things:
Play a sound file
Stop/pause a playing song
Adjust volume
Show a bar that displays the current position in the song
Most files will be .mp3 files but being able to process other formats is certainly a plus. Since this is just a small project it's ok if it runs just on Windows. Scalabilty would be nice but not required.
It would be nice to have a small overview of audio support/audio libraries of programming languages (i'm always up for something new) that can accomplish these simple things, in a not too complicated way, aswell as personal experiences.
In this way i hope to create a better understanding of which programming language fits my project best. (i would very much like to not have to change language mid-way the project)
--
Edit:
This is only for a later stage of the project if the first part was successfull: i will want to change the file names of the audio files that are displayed. (to make them follow a specific format)
I haven't written audio processing programs much, but I know a lot of them exist for C and C++. For Java perhaps, too, but I don't know Java. I had used audio with SDL in a game, but that doesn't have that many features and I don't recommend it.
There's this question asking for a library in C, and there are a couple of similar questions that SO brings up on the side. You may want to take a look at those.
You would also need to look for a library that loads different file types. SDL at least, only opens .wav files, which I believe most of the playback libraries would support. For MP3, you will most likely need an additional library. I know Audacity uses LAME Mp3 so I'm guessing that should be good.
Some of the functionalities you want is also doable by yourself. For example, knowing the length of the music and the amount you have already read, you will know how far in the audio you are. Adjusting the volume is also a multiplication (in the simplest case) that you can do on the audio data if the library doesn't provide it.
A very good choice seems to be PortAudio which is used by Audacity, and also recommended in the accepted answer of the question I mentioned above.
I've done audio apps in both Java and C++. Java development goes way faster because it's a more powerful language and has garbage collection, but JavaSound is a pretty awful solution for audio. Of course, there are wrappers for FFMPEG and other stuff, so you can get a lot of things working. Here's an example of a Java audio app: http://www.indabamusic.com/help/mantis
OTOH, C++ gives you lots of control, low latency and wealth of libraries. (another answer mentioned Portaudio, which is, indeed, great.) But you will definitely find it also has a much longer development cycle.
You can certainly do everything you want to do with either language.
Is it possible rip game resources from a .smc file? Specifically art, music, sprites, etc. How does an emulator copy the system it emulates?
It's possible, in the sense that the information is all there in some manner. But an smc file is basically a compiled program with embedded resources, and there isn't even a standard compiler or standard format for storing the resources that you can start from.
And as far as image data goes, there is a good chance it will be in the palettized and tiled format used by the PPU, although it's also not unlikely that it will be compressed in some manner or another. But the palette will probably be almost impossible to find by static analysis, and the tile maps are probably generated from the level data rather than being explicitly stored anywhere. You may have better luck running it in an emulator and extracting the data from VRAM.
For music, the situation is even more discouraging. SNES audio is most akin to a MOD file: instruments are sampled, and then the individual samples are pitch-adjusted and mixed to generate the output sound. The SNES provides hardware to decode the instrument samples, manipulate the pitch, and mix them together, but no high-level program (i.e. no equivalent of a mod file "tracker") to play back actual songs. So you may be able to find the BRR-encoded instrument samples in the same manner you may be able to find the image tile data, but the song data can and will be formatted completely differently in different games. Again, your best luck may come from extracting the state of the APU as an SPC file and working with that.
As for your other question, see How do emulators work and how are they written? for a previous answer on that very topic.
I want to take a classical music piece in .mp3 (or other audio file if necessary) file and take the same music piece in *.midi file. then - I want to synchronize between them so as a result only the midi file would change and the timing of its beat would be synchronized with the .mp3. So lets say - if I would play them both on the same time they would play the same notes synchronizly.
How can I do so?
(I have cubase if the answer might be there...)
It's a tough task because general beat-tracking (follow tempo changes) hasn't yet been figured out.
There's at least one tool that does work though for matching an audio file to a midi file, assuming the audio file is almost identical to the midi file in terms of the score. But I can't remember it's named, never have used it. The place is to ask is the Music Information Retrieval community of scientists:
http://listes.ircam.fr/wws/info/music-ir
For manual mathcing, you can use modern DAW's like Logic, Pro Tools, etc, to help you with this by providing reasonably nice tools to build a detailed tempo-map of the audio file, and then the MIDI file would line right up with it, but it's a tedious task. You'll likely need tempo changes more often than every measure to get a nice alignment - it will be style-dependent.
You could use tools that already exist. For example, if you know the tempo of the mp3, then you could use this page to change the tempo on the midi file.
My question is not completely programming-related, but nevertheless I think SO is the right place to ask.
In my program I generate some audio data and save the track to a WAV file. Everything works fine with one sound generator. But now I want to add more generators and mix the generated audio data into one file. Unfortunately it is more complicated than it seems at first sight.
Moreover I didn't find much useful information on how to mix a set of audio samples.
So is there anyone who can give me advice?
edit:
I'm programming in C++. But it doesn't matter, since I was interested in the theory behind mixing two audio tracks. The problem I have is that I cannot just sum up the samples, because this often produces distorted sound.
I assume your problem is that for every audio source you're adding in, you're having to lower the levels.
If the app gives control to a user, just let them control the levels directly. Hotness is their responsibility, not yours. This is "summing."
If the mixing is automated, you're about to go on a journey. You'll probably need compression, if not limiting. (Limiting is an extreme version of compression.)
Note that anything you do to the audio (including compression and limiting) is a form of distortion, so you WILL have coloration of the audio. Your choice of compression and limiting algorithms will affect the sound.
Since you're not generating the audio in real time, you have the possibility of doing "brick wall" limiting. That's because you have foreknowledge of the levels. Realtime limiting is more limited because you can't know what's coming up--you have to be reactive.
Is this music, sound effects, voices, what?
Programmers here deal with this all the time.
Mixing audio samples means adding them together, that's all. Typically you do add them into a larger data type so that you can detect overflow and clamp the values before casting back into your destination buffer. If you know beforehand that you will have overflow then you can scale their amplitudes prior to addition - simply multiply by a floating point value between 0 and 1, again keeping in mind the issue of precision, perhaps converting to a larger data type first.
If you have a specific problem that is not addressed by this, feel free to update your original question.
dirty mix of two samples
mix = (a + b) - a * b * sign(a + b)
You never said what programming language and platform, however for now I'll assume Windows using C#.
http://www.codeplex.com/naudio
Great open source library that really covers off lots of the stuff you'd encounter during most audio operations.