I am looking for advice on how to control 5 or more speakers with Matlab.
In an earlier thread I received advice on the hardware needed to control the speakers.
http://audio.stackexchange.com/questions/1541/easy-solution-for-controlling-5-or-more-speakers-from-a-single-computer
But I am still interested in advice on how to control these speakers. I tried posting on the audio forum.
http://audio.stackexchange.com/questions/1554/how-to-control-5-or-more-speakers-from-matlab
But 'Friend Of George' advised me to post here for better results.
I would appreciate your suggestions. Thank you for reading.
You will need to download a software interface to your audio hardware which enables Matlab to access multichannel audio driver, as the built-in Matlab audio only supports 2 channels.
I used this one in the past http://www.playrec.co.uk/, and it worked for me. It's not really a straight-forward "download and install" package, but the site has good documentation regarding how to make it work and use it.
If you want more options, search the web for "matlab multi channel audio"
Related
I am trying to encode a live stream into Apple HLS for iPhone on windows. I was looking at different options and wowza can do it, but doesn't support CDN distribution of HLS as far as I can see. Plus it costs a lot of money.
What I did find was this site: http://www.espend.de/artikel/iphone-ipad-ipod-http-streaming-segmenter-and-m3u8-windows.html
I can now set up a single bitrate stream easily, but my goal is an adapive multi-bitrate live stream. Is it possible? For VOD content it can easily be accomplished with creating the different qualities then linking to them in a new m3u8, but how would this be done in live?
I can of course set up three quality live streams and link to them in an m3u8, but how will I get them GOP-aligned in this case?
My initial thought was to have one ffmpeg instance create all qualities and re-stream those outputs to new ffmpeg-instances that just remux and pipe to the segmenter. But I would need some way of streaming locally between instances. Can that be done?
If anyone has a nice solution to this, or can link to other software capable of live HLS on windows, I would appreciate any input.
Have a great day!
Regards
Carl
Just to let people know, I ended up using http://www.ioncannon.net/projects/http-live-video-stream-segmenter-and-distributor/ on a linux virtualbox and it works great. I had trouble compiling it, but there were a couple of forks that fixed those problems.
I am developing an iPhone application (like Audio Processing). I have to give some effect to the audios.
If it is desktop app, many options are there. We can get good examples and full project like audacity. But I want to develop for iPhone.
I got an app with reverb option; (take a look at following link). Just I watch the "video", I did not test this application in my iPhone device.
http://www.appstorehq.com/reverb-iphone-89870/app
My question is; How can I develop the app with reverb functionality ? Is there any documentation for that ? If it is, just share with us.
NOTE: We can use AudioUnit to develop the app with reverb functionality (I am not clear with this.).
EDIT: I don't like to use any third party library.
If anybody having knowledge about this, please share with us.
Thanks.
if yourre targeting ios5 you can just the audio unit subtype kAudioUnitSubType_Reverb2 of the effect audio unit.
reverb unit
AudioComponentDescription auEffectUnitDescription;
auEffectUnitDescription.componentType = kAudioUnitType_Effect;
auEffectUnitDescription.componentSubType = kAudioUnitSubType_Reverb2;
auEffectUnitDescription.componentManufacturer = kAudioUnitManufacturer_Apple;
AUGraphAddNode(
processingGraph,
&auEffectUnitDescription,
&auEffectNode),
Failing that you could just write your own reverb code in the remoteio callback. A simple delay might be easier to do and would sound similar.
iOS 5.0 brings native OpenAL support, so it is now much easier - you don't have to code the algorithm yourself. It also bring support for a variety of reverb spaces:
Small Room
Medium Room
Large Room (2 configurations)
Medium Hall (3 configurations)
Large Hall (2 configurations)
Plate
Medium Chamber
Large Chamber
Cathedral
I suggest that you try the ObjectAL wrapper which already has a great support for the reverb effect:
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.
If you still don't to use any 3rd party library, you can just grab the relevant pieces from ObjectAL. Look for the reverb-related code in the following source files (and their corresponding headers):
https://github.com/kstenerud/ObjectAL-for-iPhone/blob/master/ObjectAL/ObjectAL/OpenAL/ALListener.m
https://github.com/kstenerud/ObjectAL-for-iPhone/blob/master/ObjectAL/ObjectAL/OpenAL/ALSource.m
https://github.com/kstenerud/ObjectAL-for-iPhone/blob/master/ObjectAL/ObjectAL/OpenAL/ALWrapper.m
Good luck with your project!
AUs are a good place to start.
write your own reverb AU which contains a reverb implementation. there are tons of ways to implement a reverb. a medium/long convolution reverb is much to ask from a phone, but something such as a FDN (feedback delay network) will not require a lot of memory or CPU.
both implementations are easy to implement, if you're familiar with audio programming and optimization. the tough part is actually making one that sounds very good and performs well.
if you're unable to write optimal low level code or you do not (presently) understand basic audio signal processing, then you'll have a few obstacles to overcome -- it may be a long road in that case.
Searching the iOS documentation for "reverb" produces a link to the Core Audio Overview, which references reverb as an "effect unit." Perhaps that's worth further study?
No good, I have attempted the audio unit approach and even though it is in the documentation it is "not" implemented yet by the apple engineers. Each time you call the function to set the reverb property you will only get failure status code. You would have to implement your own reverb effect. Try reading some DSP book and you might find a clue.
you need to learn some DSP-level coding, the DSP cookbook book is okay and there are others out there. But basically you need to be comfortable with handling audio signal in the frequency domain and things such as FFT's. Once you have that, implementing a reverb filter should be straight-forward.
This is an answer I've given before, but I believe it is relevant here. I am going to agree with the others and say that you are going to have to become a bit more familiar with core-audio if you want to do this properly.
I highly recommend this core-audio book. It will teach what you need to do this right and will save you a lot of frustration.
The chapter on audio effects has not been published yet, but if it is anything like the rest of the book it's worth the wait.
EDIT
You will most likely need to do this with an audio effect (which is a form of an audio unit).
Hi unfortunately I've not been able to figure out audio on the iPhone. The best I've come close to are the AVAudioRecorder/Player classes and I know that they are no good fo audio processing.
So i'm wondering if someone would be able to explain to me how to "listen" to the iPhone's mic input in chunks of say 1024 samples, analyse the samples and do stuff. And just keep going like that until my app terminates or tells it to stop. I'm not looking to save any data, all I want is to analyse the data in real time and do stuff in real time with it.
I've attempted to try and understand apples "aurioTouch" example but it's just way too complicated for me to understand.
So can someone explain to me how I should go about this?
If you want to analyze audio input in real-time, it doesn't get a lot simpler than Apple's aurioTouch iOS sample app with source code (there is also a mirror site). You can google a bit more info on using the Audio Unit RemoteIO API for recording, but you'll still have to figure out the real-time analysis DSP portion.
The Audio Queue API is a slight bit simpler for getting input buffers of raw PCM audio data from the mic, but not much simpler, and it has a higher latency.
Added later: There's also a version of aurioTouch converted to Swift here: https://github.com/ooper-shlab/aurioTouch2.0-Swift
AVAudioPlayer/Recorder class won't take you there if you wanna do any real time audio processing. The Audio Toolbox and Audio Unit frameworks are the way to go. Check here for apple's audio programming guide to see which framework suits your need. And believe me, these low level stuff is not easy and is poorly documented. CocoaDev has some tutorials where you can find sample codes. Also, there is an audio DSP library DIRAC I recently discovered for tempo and pitch manipulation. I haven't looked into it much but you might find it useful.
If all you want is samples with a minimum amount of processing by the OS, you probably want the Audio Queue API; see Audio Queue Services Programming Guide.
AVAudioRecorder is designed for recording to a file, and AudioUnit is more for "pluggable" audio processing (and on the Mac side of things, AU Lab is actually pretty cool).
I'm looking for a tutorial on how to listen to the microphone input while it is recording. I've been searching for a while but nothing really relevant comes up. Is this supported by the SDK or is it a bit of a hack to set up?
I've found this but I'd like to find something a little more educational.
Any tips?
Thanks!
It is supported, the framework is called Audio Sessions, the SDK guide is called "Audio Session Programming Guide", AudioSessionInitialize is probably a good starting point to learn the process.
I think there are a few decent example projects, aurioTouch I believe has some of the pieces you need.
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