When to set kAudioUnitProperty_StreamFormat (and kAudioUnitProperty_SampleRate too)? For each AU in my AUGraph ? Or is it enough to set it jus for the AU Mixer ?
André
you set it on the inputs and outputs of each audiounit.
iphone only allows input signed ints. so don't bother with floats it just won't work.
you set the sample rates using
CAStreamBasicDesciption myDescription;
myDescription.mSampleRate = 44100.0f; // and do this for the other options such as mBitsPerChannel etc.
On the output of audiounits such as the mixer, it comes out as 8.24 fixed point format.
be aware of this when you're trying to create callbacks and using the audiounitrender function, the formats have to match and you can't change the output formats. (but you may still need to set it)
use printf("Mixer file format: "); myDescription.Print(); to get the format description. It will depend on where you put it in your initialization process.
In short, yes - for more detail on what you actually need to set on each unit, see Audio Unit Hosting Guide for iOS
Related
I would like to fill an array from a stream for around ten seconds.{I wish to do some processing on the data)So far I can:
(a) obtain the microphone stream using mediaRecorder
(b) use analyser and analyser.getFloatTimeDomainData(dataArray) to obtain an array but it is size limited to only a little over half a second of data.I can also successfully output the data after processing back onto a stream and to outDestination.
(c) I have also experimented with obtaining a 'chunks' array from mediaRecorder directly but the problem then is that I can't find any mime type that would give me a simple array of values - ie an uncompressed sample by sample single channel set of value - ie a longer version of 'dataArray' in (b).
I am wondering if I am missing a simple way round this problem?
Solutions I have seen tend to use step (b) and do regular polls then reassemble a longer array - however it seems the timing is a bit tricky ..
I'v also seen suggestions to use audio workouts - I might have to do this but would prefer a simpler solution!
Or again, if someone knows how to drive mediaRecorder to output the chunks array in a simple array format FLOAT32.of one channel.That would do the trick.
Or maybe I'm missing something simpler?
I have code showing those steps that have been successful and will upload if anyone requests.
I've hunted high and low and cannot find a solution to this problem. I am looking for a method to change the input/output devices which an AVAudioEngine will use on macOS.
When simply playing back an audio file the following works as expected:
var outputDeviceID:AudioDeviceID = xxx
let result:OSStatus = AudioUnitSetProperty(outputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &outputDeviceID, UInt32(MemoryLayout<AudioObjectPropertyAddress>.size))
if result != 0 {
print("error setting output device \(result)")
return
}
However if I initialize the audio input (with let input = engine.inputNode) then I get an error once I attempt to start the engine:
AVAEInternal.h:88 required condition is false: [AVAudioEngine.mm:1055:CheckCanPerformIO: (canPerformIO)]
I know that my playback code is OK since, if I avoid changing the output device then I can hear the microphone and the audio file, and if I change the output device but don't initialize the inputNode the file plays to the specified destination.
Additionally to this I have been trying to change the input device, I understood from various places that the following should do this:
let result1:OSStatus = AudioUnitSetProperty(inputUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Output, 0, &inputDeviceID, UInt32(MemoryLayout<AudioObjectPropertyAddress>.size))
if result1 != 0 {
print("failed with error \(result1)")
return
}
However, this doesn't work - in most cases it throws an error (10853) although if I select a sound card that has both inputs and outputs it succeeds - it appears that when I am attempting to set the output or the input node it is actually setting the device for both.
I would think that this meant that an AVAudioEngine instance can only deal with one device, however it is quite happy working with the default devices (mic and speakers/headphones) so I am confident that isn't the issue. Looking at some solutions I have seen online people simply change the default input, but this isn't a massively nice solution.
Does anyone have any ideas as to whether this is possible?
It's worth noting that kAudioOutputUnitProperty_CurrentDevice is the only property available, there is not an equivalent kAudioInputUnitProperty_CurrentDevice key, due to the fact that as I understand it both the inputNode and outputNode are classed as "Output Units" (as they both emit sound somewhere).
Any ideas would be much appreciated as this is very very frustrating!!
Thanks
So I filed a support request with apple on this and another issue and the response confirms that an AVAudioEngine can only be assigned to a single Aggregate device (that is, a device with both input and output channels) - the system default units create effectively an aggregate device internally which is why they work, although I've found an additional issue in that if the input device also has output capabilities (and you activate the inputNode) then that device has to be both the input and output device as otherwise the output appears not to work.
So answer is that I think there is no answer..
I currently am using Max/MSP to create an interactive system between lights and sound.
I am using Philips hue lighting which I have hooked up to Max/MSP and now I am wanting to trigger an increase in brightness/saturation on the input of a note from a Midi instrument. Does anyone have any ideas how this might be accomplished?
I have built this.
I used the shell object. And then feed an array of parameters into it via a javascipt file with the HUE API. There is a lag time of 1/6 of a second between commands.
Javascript file:
inlets=1;
outlets=1;
var bridge="192.168.0.100";
var hash="newdeveloper";
var bulb= 1;
var brt= 200;
var satn= 250;
var hcolor= 10000;
var bulb=1;
function list(bulb,hcolor,brt,satn,tran) {
execute('PUT','http://'+bridge+'/api/'+hash+'/lights/'+bulb+'/state', '"{\\\"on\\\":true,\\\"hue\\\":'+hcolor+', \\\"bri\\\":'+brt+',\\\"sat\\\":'+satn+',\\\"transitiontime\\\":'+tran+'}"');
}
function execute($method,$url,$message){
outlet(0,"curl --request",$method,"--data",$message,$url);
}
To control Philips Hue you need to issue calls to a restful http based api, like so: http://www.developers.meethue.com/documentation/core-concepts, using the [jweb] or [maxweb] objects: https://cycling74.com/forums/topic/making-rest-call-from-max-6-and-saving-the-return/
Generally however, to control lights you use DMX, the standard protocol for professional lighting control. Here is a somewhat lengthy post on the topic: https://cycling74.com/forums/topic/controlling-video-and-lighting-with-max/, scroll down to my post from APRIL 11, 2014 | 3:42 AM.
To change the bri/sat of your lights is explained in the following link (Registration/Login required)
http://www.developers.meethue.com/documentation/lights-api#16_set_light_state
You will need to know the IP Address of your hue hue bridge which is explained here: http://www.developers.meethue.com/documentation/getting-started and a valid username.
Also bear in mind the performance limitations. As a general rule you can send up to 10 lightstate commands per second. I would recommend having a 100ms gap between each one, to prevent flooding the bridge (and losing commands).
Are you interested in finding out details of who to map this data from a MIDI input to the phillips HUE lights within max? or are you already familiar with Max.
Using Tommy b's javascript (which you could put into a js object), You could for example scale the MIDI messages you want to use using midiin and borax objects and map them to the outputs you want using the scale object. Karlheinz Essl's RTC library is a good place to start with algorithmic composition if you want to transform the data at all http://www.essl.at/software.html
+1 for DMX light control via Max. There are lots of good max-to-dmx tutorials and USB-DMX hardware is getting pretty cheap. However, as someone who previously believed in dragging a bunch of computer equipment on stage just to control a light or two with an instrument, I'd recommend researching and purchasing a simple one channel "color organ" circuit kit (e.g., Velleman MK 110). Controlling a 120/240V light bulb via audio is easier than you might think; a computer for this type of application is usually overkill. Keep it simple and good luck!
i'm back with one more question related to BASS. I already had posted this question How Can we control bass of music in iPhone, but not get as much attention of your people as it should get. But now I have done some more search and had read the Core AUDIO. I got one sample code which i want to share with you people here is the link to download it iPhoneMixerEqGraphTest. Have a look on it in this code what i had seen is the developer had use preset Equalizer given by iPod in Apple. Lets see some code snippet too:----
// iPodEQ unit
CAComponentDescription eq_desc(kAudioUnitType_Effect, kAudioUnitSubType_AUiPodEQ, kAudioUnitManufacturer_Apple);
What kAudioUnitSubType_AUiPodEQ does is it get preset values from iPod's equalizer and return us in Xcode in an array which we can use in PickerView/TableView and can set any category like bass, rock, Dance etc. It is helpless for me as it only returns names of equalizer types like bass, rock, Dance etc. as i want to implement bass only and want to implement it on UISLider.
To implement Bass on slider i need values so that i can set minimum and maximum value so that on moving slider bass can be changed.
After getting all this i start reading Core Audio's Audio Unit framework's classes and got this
after that i start searching for bass control and got this
So now i need to implement this kAudioUnitSubType_LowShelfFilter. But now i don't know how to implement this enum in my code so that i can control the bass as written documentation. Even Apple had not write that how can we use it. kAudioUnitSubType_AUiPodEQ this category was returning us an array but kAudioUnitSubType_LowShelfFilter category is not returning any array. While using kAudioUnitSubType_AUiPodEQ this category we can use types of equalizer from an array but how can we use this category kAudioUnitSubType_LowShelfFilter. Can anybody help me regarding this in any manner? It would be highly appreciable.
Thanks.
Update
Although it's declared in the iOS headers, the Low Shelf AU is not actually available on iOS.
The parameters of the Low Shelf are different from the iPod EQ.
Parameters are declared and documented in `AudioUnit/AudioUnitParameters.h':
// Parameters for the AULowShelfFilter unit
enum {
// Global, Hz, 10->200, 80
kAULowShelfParam_CutoffFrequency = 0,
// Global, dB, -40->40, 0
kAULowShelfParam_Gain = 1
};
So after your low shelf AU is created, configure its parameters using AudioUnitSetParameter.
Some initial parameter values you can try would be 120 Hz (kAULowShelfParam_CutoffFrequency) and +6 dB (kAULowShelfParam_Gain) -- assuming your system reproduces bass well, your low frequency content should be twice as loud.
Can u tell me how can i use this kAULowShelfParam_CutoffFrequency to change the frequency.
If everything is configured right, this should be all that is needed:
assert(lowShelfAU);
const float frequencyInHz = 120.0f;
OSStatus result = AudioUnitSetParameter(lowShelfAU,
kAULowShelfParam_CutoffFrequency,
kAudioUnitScope_Global,
0,
frequencyInHz,
0);
if (noErr != result) {
assert(0 && "error!");
return ...;
}
Hi
After reading so much on the speak here apple's example, i couldnt understand what parts of it i need .
I only need to always get the audio data from mic , and check the power of it , in order to measure the audio input duration in ms .
I want to get to the place where i can get the audio level and write there my own objective c code, or send it to another .m file of my own.
Can someone direct me to that variable that holdes that data ?
and what are the things i can erase from it? (dont need openGL,quartz,play sounds,and view any graph )
Thanks a lot.