Changing sample rate of an AUGraph on iOS - ios

I've implemented an AUGraph similar to the one on the iOS Developer's Library. In my App, however, I need to be able to playback sound at different sample rates (probably two different ones).
I've been looking around Apple's documentation and haven't found a way to set the sample rate at runtime. I've been thinking of three possible work-arounds:
Re-initialize the AUGraph every time I need to change the sample rate;
Initialize a different AUGraph for each different sample rate;
Convert the sample rate of every sound before playing;
These methods all seem really clunky and heavy on the processor.
What is the best way of changing sample rate of an AUGraph at runtime?

typically #1 for continuous audio streaming scenarios.
your program may have a special need or benefit by using another approach you have listed:
#2: you need to process where reinitialization is not a concern.
#3: you need to mix and process two streams with different input sample rates together at the same time, particularly if you find yourself SRCing the signal multiple times.
but, if you just need playback with SRC and lowest latency is not a concern, you may want to try an AudioQueue instead.

I'm pretty sure that it can't be done in runtime. Solution #2 is your best bet, along with #3. For sample rate conversion, libsndfile can probably be adapted to your needs.

If you don't want latency from tearing down and rebuilding the audio graph, you may need to resample the sound data (for all but one sample rate).
You could either resample the sounds data before starting to play it, or run a real-time resampler as part of the audio graph. Many iOS music apps do the latter as part of a built-in sampler-based synth unit, so the device has plenty of compute power to do so.

Related

What kind of drum sampling options does Audiokit have?

Working in audio kit and I am looking to understand how people have incorporated drums. Obviously, the sampler is an option, but I am wondering if there is a built in option similar to some of the basic synthesis options.
There are a few options. I personally like the AppleSampler/MidiSampler like in the example but instead of using audio files you can create a EXS Sampler instrument in Logic where you can assign notes for different velocities. AppleSampler can also load AUPresets made in GarageBand and SoundFonts (SF2). The DunneAudioKit Sampler is an option if you are working with SFZ files, but I think that might be a work-in-progress in AudioKit 5. Loading WAV files directly into AppleSampler is also a good option if you just want one shot sounds.
I'm assuming you're mostly talking about playback of samples, not recording.
The best built-in option I've seen (other than AppleSampler/MidiSampler) is AudioPlayer, which lets you load in a sample and play it back on demand (from an on-screen pad, etc). MIDIListener can then help you respond to external MIDI events, etc. It works (I have a pretty big branch in my app where I tried it), but not sure it works well.
I wouldn't recommend DunneAudioKit Sampler for drums. There is no one-shot playback (so playing the same note in quick succession will cut off the previous note, even if you mess with the release). If you're trying to build a complex/realistic acoustic drum instrument, you'll also want round-robins so that variations of the same hit can be played, which Dunne also doesn't have. It can load SFZ files, but only a very limited subset of SFZ's opcodes (so again, it's missing things like round robins, mute groups, one-shot, etc).
Having gone down all those roads, I would suggest starting with AppleSampler, and I would build the EXS or aupreset file in Logic or Mainstage rather than trying to build something programmatically.
If your needs are really simple, the examples in AudioKit's recently released drum pad playground is a great place to start, loading single samples into a specific note on AppleSampler.

How to simulate audio and video calls in NS3?

I want to generate different types of traffic for analyzing OFDMA transmission in NS3. How can I simulate video and audio calls?
The first three options that come to mind are:
If you want to be as close to reality as possible, try out the Direct Code Execution (DCE) Module. I've never used it, so I'm not sure how well it's supported.
Use the OnOffApplication. The OnOffApplication allows you to set onTime, offTime, and a DataRate (among other variables). You can determine the rate at which data is sent for your audio or video program, and then provide those rates to the OnOffApplication. You may find the OnOffHelper convenient to set various parameters of an OnOffApplication.
Create your own Application. This option may be of particular interest since you could simulate variable bitrate audio/video calls. If you choose this option, I highly suggest you checkout the ns-3 tutorial for the walkthrough of fifth.cc to learn more about how to create your own Application.
The second option is probably easiest to use, but may not be as accurate as the first, or as flexible as the third.

How can I use AVAudioPlayer to play audio faster *and* higher pitched?

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.

Modifying Low, Mid, High Frequencies Core Audio IOS

I see that the only effect unit on iOS is the ipod EQ. Is there any other way to change the high, mid and low frequencies of an audio unit on iOS?
Unfortunately, the iPhone doesn't really allow custom AudioUnits (ie. an AudioUnit's ID cannot be registered for use by an AUGraph). What you can do is register a render callback and process the raw PCM data yourself. Sites like musicdsp.org have sample DSP code that you can utilize to implement any effect you can imagine.
Also, here is a similar StackOverflow question for reference: How to make a simple EQ AudioUnit
There are a bunch of built-in Audio Units including a set of filters, delay and even reverb. A good clue is to look in AUComponent.h. You will need to get their ABSD's properly setup otherwise they throw an error or produce silence. But they do work.

Virtual Instrument App Recording Functionality With RemoteIO

I'm developing a virtual instrument app for iOS and am trying to implement a recording function so that the app can record and playback the music the user makes with the instrument. I'm currently using the CocosDenshion sound engine (with a few of my own hacks involving fades etc) which is based on OpenAL. From my research on the net it seems I have two options:
Keep a record of the user's inputs (ie. which notes were played at what volume) so that the app can recreate the sound (but this cannot be shared/emailed).
Hack my own low-level sound engine using AudioUnits & specifically RemoteIO so that I manually mix all the sounds and populate the final output buffer by hand and hence can save said buffer to a file. This will be able to be shared by email etc.
I have implemented a RemoteIO callback for rendering the output buffer in the hope that it would give me previously played data in the buffer but alas the buffer is always all 00.
So my question is: is there an easier way to sniff/listen to what my app is sending to the speakers than my option 2 above?
Thanks in advance for your help!
I think you should use remoteIO, I had a similar project several months ago and wanted to avoid remoteIO and audio units as much as possible, but in the end, after I wrote tons of code and read lots of documentations from third party libraries (including cocosdenshion) I end up using audio units anyway. More than that, it's not that hard to set up and work with. If you however look for a library to do most of the work for you, you should look for one written a top of core audio not open al.
You might want to take a look at the AudioCopy framework. It does a lot of what you seem to be looking for, and will save you from potentially reinventing some wheels.

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