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.
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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.
I'm trying to put together an open source library that allows iOS devices to play files with unsupported containers, as long as the track formats/codecs are supported. e.g.: a Matroska video (MKV) file with an H264 video track and an AAC audio track. I'm making an app that surely could use that functionality and I bet there are many more out there that would benefit from it. Any help you can give (by commenting here or—even better— collaborating with me) is much appreciated. This is where I'm at so far:
I did a bit of research trying to find out how players like AVPlayerHD or Infuse can play non-standard containers and still have hardware acceleration. It seems like they transcode small chunks of the whole video file and play those in sequence instead.
It's a good solution. But if you want to throw that video to an Apple TV, things don't work as planned since the video is actually a bunch of smaller chunks being played as a playlist. This site has way more info, but at its core streaming to Apple TV is essentially a progressive download of the MP4/MPV file being played.
I'm thinking a sort of streaming proxy is the way to go. For the playing side of things, I've been investigating AVSampleBufferDisplayLayer (more info here) as a way of playing the video track. I haven't gotten to audio yet. Things get interesting when you think about the AirPlay side of things: by having a "container proxy", we can make any file look like it has the right container without the file size implications of transcoding.
It seems like GStreamer might be a good starting point for the proxy. I need to read up on it; I've never used it before. Does this approach sound like a good one for a library that could be used for App Store apps?
Thanks!
Finally got some extra time to go over GStreamer. Especially this article about how it is already updated to use the hardware decoding provided by iOS 8. So no need to develop this; GStreamer seems to be the answer.
Thanks!
The 'chucked' solution is no longer necessary in iOS 8. You should simply set up a video decode session and pass in NALUs.
https://developer.apple.com/videos/wwdc/2014/#513
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
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.
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