I'm working on an app which has a requirement for running some basic audio filters (such as normalisation and reverb) on a file. The idea is to take an existing audio file, add the filters, and then write the data to a new file. Crucially, this must be done without any playback and should be fast (i.e. on a 60 second audio file I should be able to add reverb in under a second).
I've looked at several solutions such as The Amazing Audio Engine and AudioBox but these all seem to rely on you playing back any audio in realtime rather than writing it to a file.
Does anybody have examples, or can point me in the right direction, for simply taking a file and applying a basic audio filter without listening to it. I'm sure I must be missing something simple somewhere but my searches have turned up nothing.
In general,the steps are:
Set up an AUGraph like this - AudioFilePlayer -> Reverb/Limiter/EQ/etc. -> GenericOutput
Open the input file and schedule it on the AudioFilePlayer.
Create an output file and repeatedly call AudioUnitRender on the GenericOutput unit, writing the rendered buffers to the output file.
I'm not sure about the speed of this, but it should be acceptable.
There is a comprehensive example of offline rendering in this thread that covers the setup and rendering process.
Related
I'm working on a mobile application that can perform basic analysis on audio input from the microphone in real time. However, the usual way to do it using the 'AVAudioRecorder` as shown in this guide and the API requires you to save it to a file first.
Since the app is meant to stay on for a long time and be used multiple times a day, I want to avoid clutter the phone with many audio files or audio files that are too big. However, I can't seem to find the way around it. Searching for solutions on the internet always lead to solutions of how to save an audio to a file, instead of avoiding saving to a file and work with some kind of buffer.
Any pointers would be super helpful!
Both the iOS Audio Unit and the Audio Queue APIs allow one to process short buffers of audio input in real-time without saving to a file.
You can also use a tap on the AVAudioEngine. See Apple's documentation: https://developer.apple.com/library/ios/samplecode/AVAEMixerSample/Introduction/Intro.html
You can use /dev/null as path in the AVAudioRecorder instance. This way it will not save to a file, but just discard the data.
var url = NSUrl.FromString("/dev/null");
var recorder = new AVAudioRecorder(url, settings, out error);
my goal is to create a sampler instrument for iPhone/iOS.
The Instrument should play back sound files on different pitches/notes and it should have a volume envelope.
A volume envelope means, that the sounds volume is fading in when nit starts to play.
I tried countless way on creating that. The desired way is to use a AVAudioEngine's AVPlayerNode, then process the individual samples of that node in realtime.
Unfortunately I had no success on that approach so far. Could you give me some pointers on how this works in iOS?
Thanks,
Tobias
PS: I did not learn the Core Audio Framework. Maybe it is possible to access an AVAudioNodes Audio Unit to execute this job, but I had not the time to read into the Framework yet.
A more low-level way is to read the audio from the file and process the audio buffers.
You store the ADSR in an array or better, a mathematical function that outputs the envelope value based on the sound index you pass it (using interpolation). So the envelope maps to any sound's duration.
Then you multiply the audio sample with the returned envelope value to get the filtered sample.
One way would be to use the AVAudioNode and link a processing node to it.
I looked at another post of yours I think AUSampler - Controlling the Settings of the AUSampler in Real Time is what you're looking for.
I haven't yet used AVAudioUnitSampler, but I believe it is just a wrapper for the AUSampler. To configure an AUSampler you must first make and export a preset file on your mac using AULab. This file is a plist which contains file references and sampler decay volume pitch cutoff and all of the good stuff that the AUSampler is built for. Then this file is put into your app bundle. You then create a directory named "Sounds", copy of all of the referenced audio samples into that folder and put it in your app bundle as well (as a folder reference). Then you create your audioGraph (or in your case AVAudioEngine) and sampler and load the preset from the preset file in your app bundle. It's kind of a pain. These links I'm providing are what I used to get up and running, but they are a little dated, if I where to start now I would definitely look into the AVAudioUnitSampler first to see if there are easier ways.
To get AULab go to Apple's developer downloads, select "Audio Tools for Xcode". Once downloaded just open the DMG and drag the folder anywhere (I drag it to my Applications folder). Inside is The AULab.
Here is a technical note describing how to load presets, another technical note on how to change parameters (such as attack/decay) in real time, and here is a WWDC Video that walks you through the whole thing including the creation of your preset using AULab.
I am building a game that lets users remix songs. I have built a mixer (based upon the apple sample code MixerHost (creating an audioGraph with a mixer audioUnit), but expanded to load 12 tracks. everything is working fine, however it takes a really long time for the songs to load when the gamer selects the songs they want to remix. This is because the program has to load 12 separate mp4 files into memory before I can start playing the music.
I think what I need is to create a AUFilePlayer audioUnit that is in charge of loading the file into the mixer. If the AUFilePlayer can handle loading the file on the fly then the user will not have to wait for the files to load 100% into memory. My two questions are, 1. can an AUFilePlayer be used this way? 2. The documentation on AUFilePlayer is very very very thin. Where can I find some example code demonstrated how to implement a AUFilePlayer properly in IOS (not in MacOS)?
Thanks
I think you're right - in this case a 'direct-from-disk' buffering approach is probably what you need. I believe the correct AudioUnit subtype is AudioFilePlayer. From the documentation:
The unit reads and converts
audio file data into its own internal
buffers. It performs disk I/O on a
high-priority thread shared among all
instances of this unit within a process.
Upon completion of a disk read, the unit
internally schedules buffers for playback.
A working example of using this unit on Mac OS X is given in Chris Adamson's book Learning Core Audio. The code for iOS isn't much different, and is discussed in this thread on the CoreAudio-API mailing list. Adamson's working code example can be found here. You should be able to adapt this to your requirements.
I have run through an audio units tutorial for a sine wave generator and done a bit of reading, and I understand basically how it is working. What I would actually like to do for my app, is play a short sound file in response to some external event. These sounds would be about 1-2 seconds in duration and occur at a rate of about about 1-2 per second.
Basically where I am at right now is trying to figure out how to play an actual audio file using my audio unit, rather than generating a sine wave. So basically my question is, how do I get an audio unit to play an audio file?
Do I simply read bytes from the audio file into the buffer in the render callback?
(if so what class do I need to deal with to open / convert / decompress / read the audio file)
or is there some simpler method where I could maybe just hand off the entire buffer and tell it to play?
Any names of specific classes or APIs I will need to look at to accomplish this would be very helpful.
OK, check this:
http://developer.apple.com/library/ios/samplecode/MixerHost/Introduction/Intro.html
EDIT: That is a sample project. This page has detailed instructions with inline code to setup common configurations: http://developer.apple.com/library/ios/ipad/#DOCUMENTATION/MusicAudio/Conceptual/AudioUnitHostingGuide_iOS/ConstructingAudioUnitApps/ConstructingAudioUnitApps.html#//apple_ref/doc/uid/TP40009492-CH16-SW1
If you don't mind being tied to IOS 5+, you should look into AUFilePlayer. It is much easer then using the callbacks and you don't have to worry about setting up your own ring buffer (something that you would need to do if you want to avoid loading all of your audio data into memory on start-up)
I've searched around but haven't found any good examples or tutorials of saving audio out of a RemoteIO Audio Unit.
My setup: Using the MusicPlayer API, I have several AUSamplers -> MixerUnit -> RemoteIO
Audio playback works great. I would like to add functionality to save the audio output to a file. Would I do this in a render callback on the RemoteIO?
Any tips or pointers to example code much appreciated!
Due to the tight latency requirements of Audio Unit callbacks, one should not to do any synchronous file access (or any other calls that could potentially block, involve memory management or OS locking actions) inside the RemoteIO callback. Instead, just copy the audio data out to another buffer (a larger circular buffer for example), and set some state indicating how much data has been copied. Then, in another thread, when the amount of data is sufficient, write the contents of that buffer out to a file. This could be a raw PCM file, which can later be converted by AVAssetReader/Writer into another audio file type.