I am trying to automate playing sounds using python's winsound module. I've come to the point where I can give the function any number of notes (whose names I've stored as frequencies), and a tempo in BPM. However, the default beep sound in winsound is very crude, and it spaces out in higher tempos (anything above 60 BPM).
Is there any way to use another sound instead of the default winsound.Beep function? I want to stick with generated sound through code (not mp3's or any downloaded files), because I have already stored various notes as frequencies, and have something to work with; all it needs is a better sound.
If winsound isn't the best option, is there any other way I could generate sound in python? I also have some experience with C++ and Java, (or at least enough for the purposes of automating the sounds), so are there any libraries to produce a better, and a more consistent sound?
Related
I'm looking for a program that is able to recognize individual audio samples from my computer and reroute them to trigger WAV files from a library. In my project, it would need to be realtime as the latency would not be a desired result. I tried using dictation software that would recognize words to trigger opening a file and that's the direction where I want to go, but instead of words I want it to be sounds and it would happen in realtime. I'm not sure where to go and am just looking for some guidance. Does anyone have any suggestions of what I should do?
That's a fairly broad question, but I can tell you how I would do it. (Hardly the only way, but where I would start.)
If you're looking for real time input, the Java Sound library (excellent tutorial here) allows for that. (Just note that microphone input from a web page is difficult on anything, due to major security concerns, so this would be a desktop application.)
If it needs to be real time, the first thing I would suggest is stream and multithread the hell out of it. I would suggest the Java 8 Stream API, but since you're looking for subsamples that match a specific pattern, then each data point will have to be aware of the state of its neighbors, and that isn't easy with streams.
You will probably want to know if a sound roughly resembles an audio profile, so for that, I would pick a tolerance on just how close you want it to be for a match (remembering that samples may not line up 100% anyway, so "exact" is not an option), and then look up Hidden Markov Models. I suggest these because they're what voice recognition software typically uses, and while your sounds may not be voices, it will give you an idea of what has already been done.
You'll also want to maintain a limited list of audio samples in memory. Specifically, you will likely need the most recent data, because an audio signal is a time-variant signal, and you can't get a match from just one point. I wouldn't make it much longer than the longest sample you're looking to recognize, as audio takes up a boatload of memory.
Lastly (for audio), I would recommend picking a standard format for comparison. Make it as good as gets you decent results, and start high. You will want to convert everything to that format before you compare it.
Once you recognize a specific sound, it's basically a Command Pattern. Specific sounds can be mapped, even with a java.util.HashMap, to specific files, which (if there are few enough) you might even have pre-loaded.
Lastly, it's worth looking at the Java Speech API. It's not part of the JDK and it's quite dated, but you might get some good advice from its implementation.
This is of course the advice of a Java-preferring programmer, but I imagine that there might be some decent libraries in Python and Ruby to help you as well; and of course there's something in C somewhere. This may sound like a lot, but most of the material is already implemented and ready-to-go.
Hopefully this helps, let's look forward to other answers.
I am using pygame to play .wav files and want to change the pitch of a particular .wav file as each level in my game progresses. To explain, my game is a near copy of the old Oric1 computer OricMunch Pacman game, where there are a few hundred pills to be munched on each level, and for every pill that is munched a short sound is played, with the pitch of the sound increasing slightly for each pill eaten/munched.
Now here is what I have tried:
1) I have used pythons wave module to create multiple copies of the sound file, each newly created file having a slight increase in pitch (by changing the 3rd parameter in params() the framerate, sometimes referred to as the sample frequency) for each cycle of a for loop. Having achieved this, I could then within the loop create multiple sound objects to add to a list, and then index through the list to play the sounds as each pill is eaten.
The problem is even though I can create hundreds of files (using the wave module) that play perfectly with their own unique pitches when played using windows media player, or even pythons winsound module, pygame does not seem to interpret the difference in pitch.
Now interestingly, I have downloaded the free trial version of Power Sound Editor which has the option to change the pitch, and so I’ve created just a few .wav files to test, and they clearly play with different pitches when played in pygame.
Observations:
From printing the params in my for loop, I can see that the framerate/frequency is changing as intended, and so obviously this is why the sounds play as intended through windows media player and winsound.
Within pygame I suspect the reason they don’t play with different pitches is because the frequency parameter is fixed, either to the default settings or via the use of pygame.mixer.pre_init, which I have indeed experimented with.
I then checked the params for each .wav file created by the Power Sound Editor, and noticed that even though the pitch sound was changing, the frequency stayed the same, which is not totally surprising since you have to select 1 of 3 options to save the files, either 22050, 44100 or 96000Hz
So now I thought time to check out the difference between pitch and frequency specifically in relation to sound, since I thought they were the same. What I found was it seems there are two principle aspects of sound waves: 1) The framerate/frequency And 2) The varying amplitude of multiple waves based on that frequency. Now I far from clearly understand this, but realise the Power Sound Editor must be altering the shape/pitch of the sound by manipulating the varying amplitude of multiple waves, point 2) above, and not by changing the fundamental frequency, point 1) above.
I am a beginner to python, pygame and programming in general, and have tried hard to find a simple way to change sound files to have gradually increasing pitches without changing the framerate/fundamental frequency. If there’s a module that I can import to help me change the pitch by manipulating the varying amplitude of mutiple waves (instead of changing the framerate/sample frequency which typically is either 22050 or 44100Hz), then it needs to take relatively no time at all if being done on the fly in order to not slow the game down. If the potential module opens, changes and then saves sound files, as opposed to altering them on the fly, then I guess it does not matter if it’s slow because I will just be creating the sound files so I can create sound objects from them in pygame to play.
Now if the only way to achieve no slow down in pygame is to create sound objects from sound files as I have already done, and then play them, then I need a way to manipulate the sound files like the Power Sound Editor (again I stress not by changing the framerate/sample frequency of typically 22050 or 44100) and then save the changed file.
I suppose in a nut shell, if I could magically automate Power Sound Editor to produce 3 to 4 hundred sound files without me having to click on the change pitch option and then save each time, this would be like having my own python way of doing it.
Conclusion:
Assuming creating sound objects from sound files is the only way not to slow my game down (as I suspect it might be) then I need the following:
An equivalent to the python wave module, but which changes the pitch like Power Sound Editor does, and not by changing the fundamental frequency like the wave module does.
Please can someone help me and let me know if there’s a way.
I am using python 3.2.3 and pygame 1.9.2
Also I’m just using pythons IDLE and I’m not familiar with using other editors.
Also I’m aware of Numpy and of various sound modules, but definitely don’t know how to use them. Also any potential modules would need to work with the above versions of python and pygame.
Thank you in advance.
Gary Townsend.
My Reply To The First Answer From Andbdrew Is Below:
Thank you for your assistance.
It does sound like changing the wave file data rather than the wave file parameters is what I need to do. For reference here is the code I have used to create the multiple files:
framerate = 44100 #Original .wav file framerate/sample frequency
for x in range(0, 25):
file = wave.open ('MunchEatPill3Amp.wav')
nFrames = file.getnframes()
wdata = file.readframes(nFrames)
params = file.getparams()
file.close()
n = list(params)
n[0] = 2
n[2] = framerate
framerate += 500
params = tuple(n)
name = 'PillSound' + str(x) + '.wav'
file = wave.open(name, 'wb')
file.setparams(params)
print(params)
file.writeframes(wdata)
file.close()
It sounds like writing different data would be equivalent or similar to how the Power Sound Editor is changing the pitch.
So please can you tell me if you know a way to modify/manipulate wdata to effectively change the pitch, rather than alter the sample rate in params(). Would this mean some relatively simple operation applied to wdata after it’s read from my .wav file. (I really hope so) I’ve heard of using numpy arrays, but I have no clue how to use these.
Please note that any .wav files modified in the above code, do indeed play in Python using winsound, or in windows media player, with the pitch increase sounding as intended. It’s only in Pygame that they don’t.
As I’ve mentioned, it seems because Pygame has a set frequency (I guess this frequency is also sample rate), that this might be the reason the pitch sounds the same, as if it wasn’t increased at all. Whereas when played with e.g. windows media player, the change in sample rate does result in a higher sounding pitch.
I suppose I just need to achieve the same increase in pitch sound by changing the file data, and not the file parameters, and so please can you tell me if you know a way.
Thank you again for helping with this.
To Summarise My Initial Question Overall, Here It Is Again:
How do you change the pitch of a .wav file without changing the framerate/sample frequency, by using the python programming language, and not some kind of separate software program such as Power Sound Editor?
Thank You Again.
You should change the frequency of the wave in your sample instead of changing the sample rate. It seems like python is playing back all of your wave files at the same sample rate (which is good), so your changes are not reflected.
Sample rate is sort of like meta information for a sound file. Read about it at http://en.m.wikipedia.org/wiki/Sampling_rate#mw-mf-search .
It tells you the amount of time between samples when you convert a continuous waveform into a discrete one. Although your (ab)use of it is cool, you would be better served by encoding different frequencies of sound in your different files all at the same sample rate.
I took a look at the docs for the wave module ( http://docs.python.org/3.3/library/wave.html ) and it looks like you should just write different data to your audio files when you call
Wave_write.writeframes(data)
That is the method that actually writes your audio data to your audio file.
The method you described is responsible for writing information about the audio file itself, not the content of the audio data.
Wave_write.setparams(tuple)
"... Where the tuple should be (nchannels, sampwidth, framerate, nframes, comptype, compname), with values valid for the set*() methods. Sets all parameters... " ( also from the docs )
If you post your code, maybe we can fix it.
If you just want to create multiple files and you are using linux, try SoX.
#!/bin/bash
for i in `seq -20 10 20`; do
sox 'input.wav' 'output_'$i'.wav' pitch $i;
done
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'd like to play a sound and have some way of reliably telling how much of it has thus far been played.
I've looked at several sound libraries but they are all horribly underdocumented and only seem to export a "PlaySound, no questions asked" routine.
I.e, I want this:
a = Sound(filename)
PlaySound(a);
while true:
print a.miliseconds_elapsed, a.length
sleep(1)
C, C++ or Python solutions preferred.
Thank you.
I use BASS Audio Library (http://www.un4seen.com/)
BASS is an audio library for use in Windows and Mac OSX software. Its purpose is to provide developers with powerful and efficient sample, stream (MP3, MP2, MP1, OGG, WAV, AIFF, custom generated, and more via add-ons), MOD music (XM, IT, S3M, MOD, MTM, UMX), MO3 music (MP3/OGG compressed MODs), and recording functions. All in a tiny DLL, under 100KB in size.*
A C program using BASS is as simple as
HSTREAM str;
BASS_Init(-1,44100,0,0,NULL);
BASS_Start();
str=BASS_StreamCreateFile(FALSE,filename,0,0,0);
BASS_ChannelPlay(str,FALSE);
while (BASS_ChannelIsActive(str)==BASS_ACTIVE_PLAYING) {
pos=BASS_ChannelGetPosition(str,BASS_POS_BYTE);
}
BASS_Stop();
BASS_Free();
This is most likely going to be both hardware-dependent (sound card etc) and OS-dependent (size of buffers used by OS etc).
Maybe it would help if you said a little more about what you're really trying to achieve and also whether we can make any assumptions about what hardware and OS this will run on ?
One possible solution: assume that the sound starts playing more or less immediately and then use a reasonably accurate timer to determine how much of the sound has played (since it will have a known, fixed sample rate).
I'm also looking for a nice Audiolibrary, where i can directly write on the Soundcards Buffer. I didn't have time yet to have a look at it myself, but pyAudio looks pretty nice. If you scroll down on the page you see an example similar like yours.
With help of the buffersize, number of channels and sample rate you can easily calculate the time each loop-step lasts and print it out.