guys.
I'm working on some audio services on iOS.
I trying to search any examples or tutorials about
how audio service or stream can read a existing audio file than
process something like filter, than write another file.
Is there any body who can help me?
Dirac3LE (by Stephan M. Bernsee) is a great library for this job.
There are examples and manual included in the download.
It is particulary inteded for time and pitch manipulation
but in your case you'll be interested in its EAFRead and EAFWrite
classes.
If you want to get familiar with the lower level library that you can also use for microphone input/sound output, and that you can get raw samples into and out of, I would suggest taking a look at Audio Queue Services.
I used it in my side project to get audio from the microphone, and I also wrote some code you might find useful to do fast vectorized, FFT based FIR filtering on input audio. You can find the code here https://github.com/jamescarlson/FreeAPRS
Related
After finally successfully finding a way to concatenate multiple voice files into one single audio file on the iPhone, I am am now trying to superimpose an audio file over the length of the voice file.
So basically I have two .m4a files:
voice.m4a which is about 10 seconds for example.
music.m4a which is about 5 seconds.
What I require is that two file be combined in such a manner that the resulting single audio file now contains the music in the background of the voice file for the length of it, so basically the resulting output should have the 10 seconds of voice and the 5seconds of music repeated twice. It is absolutely important to have a single file that contains all of this.
I am trying to get all of this done in an application on the iPhone.
Can anyone please help me out with this?
If you are looking to do that programmatically, you will need to go deeper down into CoreAudio. For a simpler solution you could use AudioQueues or for more fine grained control AudioUnits and an AUGraph. The MultiChannelMixer is the Audio Unit you are looking for. Unfortunately there is no space for an elaborate tutorial here (would take a couple of days to write just the tutorial itself), but I am hoping I could point you to the right direction.
If you decide to go down that path and want to do further audio programming then this one time simple example, then I strongly suggest you buy "Learning Core Audio, A Hands-on Guide to Audio Programming for Mac and iOS" - Chris Adamson, Kevin Avila. You can find it on Amazon, paperback or Kindle.
My aim is code a project which records human sound and changes it (with effects).
e.g : a person will record its sound over microphone (speak for a while) and than the program makes its like a baby sound.
This shall run effectively and fast (while recording the altering operation must run, too)
What is the optimum way to do it ?
Thanks
If you're looking for either XNA or DirectX to do this for you, I'm pretty sure you're going to be out of luck (I don't have much experience with DirectSound; maybe somebody can correct me). What it sounds like you want to do is realtime digital signal processing, which means that you're either going to need to write your own code to manipulate the raw waveform, or find somebody else who's already written the code for you.
If you don't have experience writing this sort of thing, it's probably best to use somebody else's signal processing library, because this sort of thing can quickly get complicated. Since you're developing for the PC, you're in luck; you can use any library you like using P/Invoke. You might try out some of the solutions suggested here and here.
MSDN has some info about the Audio namespace from XNA, and the audio recording introduced in version 4:
Working with Microphones
Recording Audio from a Microphone
Keep in mind that recorded data is returned in PCM format.
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.
Hi
I want to read and then want to perform some operations on raw audio. What is the best way to do this?
Audio File Services, Audio Converter Services, and Extended Audio File Services, all in Core Audio. AV Foundation + Core Media (specifically AVAssetReader) may also be an option, but it's really new, and therefore even less documented and less well understood than Core Audio at this point.
If you are looking for sample code, "Audio Graph" is a good starting point. The developer has provided a bit of his own documentation, that will help you quite a bit.
It will depend on the use for the audio. If latency is an issue, go for audio units. But if is not, a higher layer may be the one you require, such as AudioQueues.
I want to record two voices and compare them. I think there is some Apple sample code for voice recording. I have no idea about
comparing two audio files. What is the right approach for this? Is there any framework Apple provides for this purpose or is there any third party framework?
It's not in objective C, but it does contain some fantastic explanation about how audio is compared by Shazam, and includes sample code (and source for a working application) in Java:
Check this out
Additionally, This Question has a fantastic link to audio fingerprinting, which is essentially the same as the article above, but more in depth.
Hope this helps
I'm using Visqol for this purpose. If your audio files are generally not more than 10sek this could be something worth looking into. Also check ffmpeg library for converting the files into the desired format(Visqol will require certain sample rate depending if it is just music or speech).
https://github.com/google/visqol