Has anyone implemented the DIRAC2 library from http://www.dspdimension.com/technology-licensing/dirac2-iphone/ for real time pitch correction on the iPhone? The library doesn't appear to support real time processing but perhaps someone has done it?
Thx
I integrated DIRAC2 into an iPhone app so I could modify the playback speed, and it does indeed work in real-time on an iPhone 4. I had to use the lowest possible settings to keep the CPU usage down, but it does play without any skips and I am able to change the playback speed seamlessly.
Running the same project on a 3GS device yielded lesser results - namely that the audio had enough skipping that it wasn't really usable. One caveat to this, though, is that I was running my test on the free version of DIRAC2 which only supports a 44100 sample rate, which is much higher than I need. If the pro version is used and you slash the sample rate down to 22050 or lower, it might work on a 3GS, but don't quote me on that.
Anything older than a 3GS has absolutely no chance of real-time playback.
Hope this helps.
Confirmed from DSP Dimensions, that the current DIRAC2 library will not work in real time.
Related
I am working on an app that analyzes incoming audio from the built in microphone on iPhone/iPad using the iOS 6.0 SDK.
I have been struggling some time with very low levels of the lower frequencies (i.e. below 200 Hz) and I have (on the web) found others having the same problems without any answers to the problem.
Various companies working with audio tools for iOS states that there was (previous to iOS 6.0) a built in low-frequency rolloff filter that was causing these low signals on the lower frequencies BUT those sources also states that starting with iOS 6.0, it should be possible to turn off this automatic low-frequency filtering of the input audio signals.
I have gone through the audio unit header files, the audio documentation in Xcode as well as audio-related sample code without success. I have played with the different parameters and properties of the AudioUnit (which mentions low-pass filters and such) without solving the problem.
Does anybody know how to turn off the automatic low-frequency rolloff filter for RemoteIO input in iOS 6.0?
Under iOS 6.0 there is the possibility to set the current AVAudioSession to AVAudioSessionModeMeasurement like this:
[[AVAudioSession sharedInstance] setMode: AVAudioSessionModeMeasurement error:NULL];
This removes the low frequency filtering.
Link:
http://developer.apple.com/library/ios/#documentation/AVFoundation/Reference/AVAudioSession_ClassReference/Reference/Reference.html
I hope this helps.
I'm not sure if there is any way to ever accomplish this on these devices. Most microphones have difficulty with frequencies below 200 HZ (and above the 20 kHZ range as well). In fact, a lot of speakers can barely play audio at that range either. In order to get a clean signal at the <200 HZ range, you would require good enough hardware, which I think is a bit beyond the capabilities of the built in microphones of the iPhone/iPad. That's probably why Apple has filtered out these low frequency sounds, as they cannot guarantee a good enough recording, OR a good enough playback. Here's a link describing the situation better for the older devices (iPhone 4, iPhone 3GS, and iPad 1).
Apple is also very picky about what they will and won't let you play with. Even if you do find out where this filtering is taking place, interrupting that code will most likely result in your app being rejected by the app store. And due to hardware limitations, you probably wouldn't be able to achieve what you want to anyways.
Hope that Helps!
I want to do some DSP effect processing, create effect like flanger, echo, etc.
Could it be done via OpenAL? Or should I use enterely different framework/library?
Since iOS 5.0 some of the DSP effects are natively supported by OpenAL.
For example, reverb is supported with emulation for more than 10 different spaces (Small/Medium/Large Room, Medium/Large Hall, Plate, Medium/Large Chamber, Cathedral and several variations).
You can find a good reference implementation in the ObjectAL wrapper. The repository is available at https://github.com/kstenerud/ObjectAL-for-iPhone
Grab the source from this repository, load "ObjectAL.xcodeproj" and run the ObjectALDemo target on any iOS 5.0 device (should also work on the simulator). This will give you a good starting point and feeling of what the reverb effect is capable of. I personally recommend taking advantage of the ObjectAL library instead of working with OpenAL directly.
Good luck with your project!
Just write your own audio library. iOS devices don't have hardware acceleration for OpenAL. It isnt particularly difficult to do, and then you can also use apples audio units (some of which are hardware accelerated).
OK. If one uses the System Sounds Services for iPhone sound effects, there is no way to alter the level of the resulting sounds programmatically. Even worse, if one reduces the volume using the ringer volume control on the side of the iPhone to a very low level, then starts the app, the sound effects will be inaudible. On the other hand, if one increases the hardware level to the max before starting the app, sound effects will be uncomfortably loud. To all intents and purposes, this renders the System Sounds APIs useless (or at least ill-advised.) All of this is moot with regard to the iPod, as it does not show this behavior (after all, it isn't a phone.)
I decided to use an AVAudioplayer to play sound effects. Under iPhone SDK 3.1, my existing aiff files (mostly u-law format) work fine, but they won't play under SDK 3.0, and I get an error msg that the codec can't be found. According to Apple's documentation, I can use any supported file format under the caf umbrella, but there must be a codec available.
I have searched diligently, but although I have found lists of codecs available for Mac OS X, I can't find a list of codecs for the iPhone, particularly for SDK 3.0. Can anyone point me to such a list? I want my game (Imp or Oaf?) to work on OS 3.0 and later. I can use mp3 files, but there are latency problems there.
Thanks,
Dan
I can't point you to a list, but ima4 has worked for me since day 1. ima4 has a 1/4 compression ratio.
In Terminal, do this:
afconvert -f caff -d ima4 original_filename.wav
I want to analyze MIC audio on an ongoing basis (not just a snipper or prerecorded sample), and display frequency graph and filter out certain aspects of the audio. Is the iPhone powerful enough for that? I suspect the answer is a yes, given the Google and iPhone voice recognition, Shazaam and other music recognition apps, and guitar tuner apps out there. However, I don't know what limitations I'll have to deal with.
Anyone play around with this area?
Apple's sample code aurioTouch has a FFT implementation.
The apps that I've seen do some sort of music/voice recognition need an internet connection, so it's highly likely that these just so some sort of feature calculation on the audio and send these features via http to do the recognition on the server.
In any case, frequency graphs and filtering have been done before on lesser CPUs a dozen years ago. The iPhone should be no problem.
"Fast enough" may be a function of your (or your customer's) expectations on how much frequency resolution you are looking for and your base sample rate.
An N-point FFT is on the order of N*log2(N) computations, so if you don't have enough MIPS, reducing N is a potential area of concession for you.
In many applications, sample rate is a non-negotiable, but if it was, this would be another possibility.
I made an app that calculates the FFT live
http://www.itunes.com/apps/oscope
You can find my code for the FFT on GitHub (although it's a little rough)
http://github.com/alexbw/iPhoneFFT
Apple's new iPhone OS 4.0 SDK allows for built-in computation of the FFT with the "Accelerate" library, so I'd definitely start working with the new OS if it's a central part of your app's functionality.
You cant just port FFT code written in C into your app...there is the thumb compiler option that complicates floating point arithmetic. You need to put it in arm mode
I have a children's iPhone application that I am writing and I need to be able to shift the pitch of a sound sample using Core Audio. Does anyone have any example code I could look at where this is done. There are many music and game apps in the app store that do this so I know I am not the first one. However, I cannot find any examples of it being done.
you can use dirac-2 from dsp dimension for pitch shifting on the iphone. quote: -
"DIRAC2 is available as both a commercial object library offering unlimited sample rates and phase locked multichannel support and as a free single channel, 44.1/48kHz LE version."
use the soundtouch open source project to change pitch
Here is the link : http://www.surina.net/soundtouch/
Once you add soundtouch to your project, you have to give the input sound file path, output sound file path and pitch change as the input.
Since it takes more time to process your sound its better to modify soundtouch so that when you record the voice, directly give the data for processing. It will make your application better.
I know it's too late for the person who asked but it is really a valuable link (As I found) for any one else who is looking for the solution of the same problem.
So Here we have latest DIRAC3 with it's own audio player classes which will take care of run time pitch and speed(explore for god knows what more) shifting. Run the sample and have huge round of applause for that.
Try Dirac - it's the best technology out there and it's available on Win, Linux, MacOS X and iOS. We're using it in all our products (and a couple of others do as well, search for "Capo" on the App Store). They're at version 3 now which has seen a huge increase in performance since previous versions. Hope this helps.
See: Related question
How much control over pitch do you need... could you precalculate all the different sounds?
If the answer is yes, then you can just pick the right sounds and play them.
You could also use Audio Converter Services in conjunction with AVAudioPlayer, which will allow you to resample the audio (which will effectively repitch them, though they'll change duration).
Alternatively, as the related question points out, you could use OpenAL and AL_PITCH