I have one sound and i need to change the pitch and tempo of the sound how can i achieve this using Open-AL and Core Audio.
i don't want to use any third party library for this purpose so can anyone help me to get started...
The kAudioUnitSubType_NewTimePitch iOS Audio Unit can independently change the pitch and tempo of Core Audio buffer streams, but it is not of very high quality (for instance, to the level of the best commercial solutions). You will have to know how to configure Audio Units and set up an AUGraph.
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My iOS project requires to retrieve some audio data (i.e. frequency , decibel) from an audio file.
By using AudioKit framework, I can get those data from the microphone by use AKFrequencyTracker, however, I am struggling on how to get the frequency straight away from the audio file without playing it. Because I need those data to plot some graphs (i.e. frequency vs. time, etc)
PS: I'm saving the record as an m4a format at the moment. (the format is optional)
Thanks in advance
You can use Accelerate framework FFT API's to get the frequency information from an audio file.
Here is a useful library to understand vDSP API usage.
https://github.com/tomer8007/real-time-audio-fft
What is best way for creating custom real time audio effect for audioEngine in iOS ?
I want to process audio at a low level, how to do it right? Be sure to use AudioUnitExtension? By simpler, I meant, is it possible to inherit from Audio Unit and using C code to change audio Data and send it back to the audioUnit connection chain in audioEngine?
You may find that the framework AudioKit will let you do what you want, the problem with manipulating audio at a low level is you have to deal with a lot of complex stuff tangental to what you are trying to achieve, just changing the playback rate of a sample means you have to deal with interpolation and antialias filters, AudioKit handles all of that for you but it may mean you have to change your way of thinking about what you are trying to.
https://github.com/AudioKit/AudioKit
Statement of Problem:
I have a collection of sound effects in my app stored as.m4a files (AAC format, 48 KHz, 16-bit) that I want to play at a variety of speeds and pitches, without having to pre-generate all the variants as separate files.
Although the .rate property of an AVAudioPlayer object can alter playback speed, it always maintains the original pitch, which is not what I want. Instead, I simply want to play the sound sample faster or slower and have the pitch go up or down to match — just like speeding up or slowing down an old-fashioned reel-to-reel tape recorder. In other words, I need some way to essentially alter the audio sample rate by amounts like +2 semitones (12% faster), –5 semitones (33% slower), +12 semitones (2x faster), etc.
Question:
Is there some way fetch the Linear PCM audio data from an AVAudioPlayer object, apply sample rate conversion using a different iOS framework, and stuff the resulting audio data into a new AVAudioPlayer object, which can then be played normally?
Possible avenues:
I was reading up on AudioConverterConvertComplexBuffer. In particular kAudioConverterSampleRateConverterComplexity_Mastering, and kAudioConverterQuality_Max, and AudioConverterFillComplexBuffer() caught my eye. So it looks possible with this audio conversion framework. Is this an avenue I should explore further?
Requirements:
I actually don't need playback to begin instantly. If sample rate conversion incurs a slight delay, that's fine. All of my samples are 4 seconds or less, so I would imagine that any on-the-fly resampling would occur quickly, on the order of 1/10 second or less. (More than 1/2 would be too much, though.)
I'd really rather not get into heavyweight stuff like OpenAL or Core Audio if there is a simpler way to do this using a conversion framework provided by iOS. However, if there is a simple solution to this problem using OpenAL or Core Audio, I'd be happy to consider that. By "simple" I mean something that can be implemented in 50–100 lines of code and doesn't require starting up additional threads to feed data to the a sound device. I'd rather just have everything taken care of automatically — which is why I'm willing to convert the audio clip prior to playing.
I want to avoid any third-party libraries here, because this isn't rocket science and I know it must be possible with native iOS frameworks somehow.
Again, I need to adjust the pitch and playback rate together, not separately. So if playback is slowed down 2x, a human voice would become very deep and slow-spoken. And if playback is sped up 2–3x, a human voice would sound like a fast-talking chipmunk. In other words, I absolutely do not want to alter the pitch while keeping the audio duration the same, because that operation results in an undesirably "tinny" sound when bending the pitch upward more than a couple semitones. I just want to speed the whole thing up and have the pitch go up as a natural side-effect, just like old-fashioned tape recorders used to do.
Needs to work in iOS 6 and up, although iOS 5 support would be a nice bonus.
The forum link Jack Wu mentions has one suggestion, which involves overriding the AIFF header data directly. This may work, but you will need to have AIFF files since it relies on a specific range of the AIFF header to write into. This also needs to be done before you create the AVAudioPlayer, which means that you can't modify the pitch once it is running.
If you are willing to go to the AudioUnits route, a complete simple solution is probably ~200 lines (note that this assumes the code style that has one function take up to 7 lines with one parameter on each line). There is an Varispeed AudioUnit, which does exactly what you want by locking pitch to rate. You would basically need to look at the API, docs and some sample AudioUnit code to get familiar and then:
create/init the audio graph and stream format (~100 lines)
create and add to the graph a RemoteIO AudioUnit (kAudioUnitSubType_RemoteIO) (this outputs to the speaker)
create and add a varispeed unit, and connect the output of the varispeed unit (kAudioUnitSubType_Varispeed) to the input of the RemoteIO Unit
create and add to the graph a AudioFilePlayer (kAudioUnitSubType_AudioFilePlayer) unit to read the file and connect it to the varispeed unit
start the graph to begin playback
when you want to change the pitch, do it via AudioUnitSetParameter, and the pitch and playback rate change will take effect while playing
Note that there is a TimePitch audio unit which allows independent control of pitch and rate, as well.
For iOS 7, you'd want to look at AVPlayerItem's time-pitch algorithm (audioTimePitchAlgorithm) called AVAudioTimePitchAlgorithmVarispeed. Unfortunately this feature is not available on early systems.
I want to record audio and apply custom-built sound effect filters, then play it back.
Are Audio Units and Audio Queue Services the API I'm looking for? Or are there other APIs which fit this purpose better?
Also, I've been told Audio Units can't be customized on iOS so there are just a few pre-made effects available. Is this true?
Audio Units is the most useful API for building effects processing under iOS. iOS 5 added several new types of filter and effect units. You can add your own custom DSP effects inside certain audio unit buffer callbacks.
Could someone explain to me how OpenAL fits in with the schema of sound on the iPhone?
There seem to be APIs at different levels for handling sound. The higher level ones are easy enough to understand.
But my understanding gets murky towards the bottom. There is Core Audio, Audio Units, OpenAL.
What is the connection between these? Is openAL the substratum, upon which rests Core Audio (which contains as one of its lower-level objects Audio Units) ?
OpenAL doesn't seem to be documented by Xcode, yet I can run code that uses its functions.
This is what I have figured out:
The substratum is Core Audio. Specifically, Audio Units.
So Audio Units form the base layer, and some low-level framework has been built on top of this. And the whole caboodle is termed Core Audio.
OpenAL is a multiplatform API -- the creators are trying to mirror the portability of OpenGL. A few companies are sponsoring OpenAL, including Creative Labs and Apple!
So Apple has provided this API, basically as a thin wrapper over Core Audio. I am guessing this is to allow developers to pull over code easily. Be warned, it is an incomplete implementation, so if you want OpenAL to do something that Core Audio can do, it will do it. But otherwise it won't.
Kind of counterintuitive -- just looking at the source, it looks as if OpenAL is lower level. Not so!
Core Audio covers a lot of things, such as reading and writing various file formats, converting between encodings, pulling frames out of streams, etc. Much of this functionality is collected as the "Audio Toolbox". Core Audio also offers multiple APIs for processing streams of audio, for playback, capture, or both. The lowest level one is Audio Units, which works with uncompressed (PCM) audio and has some nice stuff for applying effects, mixing, etc. Audio Queues, implemented atop Audio Units, are a lot easier because they work with compressed formats (not just PCM) and save you from some threading challenges. OpenAL is also implemented atop Audio Units; you still have to use PCM, but at least the threading isn't scary. Difference is that since it's not from Apple, its programming conventions are totally different from Core Audio and the rest of iOS (most obviously, it's a push API: if you want to stream with OpenAL, you poll your sources to see if they've exhausted their buffers and push in new ones; by contrast, Audio Queues and Audio Units are pull-based, in that you get a callback when new samples are needed for playback).
Higher level, as you've seen, is nice stuff like Media Player and AV Foundation. These are a lot easier if you're just playing a file, but probably aren't going to give you deep enough access if you want to do some kind of effects, signal processing, etc.