I am building an App that allows our customers to record, save, and play recorded sound as a basic functionality. This should be straight using AVFoundation Framework. What I also allow users are
Fast Forward and Reverse functionality.
User should also able to manipulate the sound. I mean they allow to insert the sound in between their recorded sound later.
Could anyone please tell me how could I achieve these? Is there any good open-source library for this?
The AVAudioPlayer supports manipulating the playback speed via the rate and enableRate properties, but it only allows forward playing.
The MPMoviePlayerController conforms to the MPMediaPlayback protocol which allows you to specify any rate (even reverse). Though this method will result in choppy audio for some rates.
As far as merging audio files, I think your best bet is to convert your samples to linear PCM. Then you can insert additional samples anywhere in your stream.
Related
I have an audio app in which all of the sound generating work is accomplished by pure data (using libpd).
I've coded a special sequencer in swift which controls the start/stop playback of multiple sequences, played by the synth engines in pure data.
Until now, I've completely avoided using Core Audio or AVFoundation for any aspect of my app, because I know nothing about them, and they both seem to require C or Objective C coding, which I know nearly nothing about.
However, I've been told from a previous q&a on here, that I need to use Core Audio or AVFoundation to get accurate timing. Without it, I've tried everything else, and the timing is totally messed up (laggy, jittery).
All of the tutorials and books on Core Audio seem overwhelmingly broad and deep to me. If all I need from one of these frameworks is accurate timing for my sequencer, how do you suggest I achieve this as someone who is a total novice to Core Audio and Objective-C, but otherwise has a 95% finished audio app?
If your sequencer is Swift code that depends on being called just-in-time to push audio, it won't work with good timing accuracy. e.g. you can't get the timing you need.
Core Audio uses a real-time pull-model (which excludes Swift code of any interesting complexity). AVFoundation likely requires you to create your audio ahead of time, and schedule buffers. An iOS app needs to be designed nearly from the ground up for one of these two solutions.
Added: If your existing code can generate audio samples a bit ahead of time, enough to statistically cover using a jittery OS timer, you can schedule this pre-generated output to be played a few milliseconds later (e.g. when pulled at the correct sample time).
AudioKit is an open source audio framework that provides Swift access to Core Audio services. It includes a Core Audio based sequencer, and there is plenty of sample code available in the form of Swift Playgrounds.
The AudioKit AKSequencer class has the transport controls you need. You can add MIDI events to your sequencer instance programmatically, or read them from a file. You could then connect your sequencer to an AKCallbackInstrument which can execute code upon receiving MIDI noteOn and noteOff commands, which might be one way to trigger your generated audio.
I've been searching for a while and can't come to a good conclusion.
I am trying to create an app that can "record" beats that a user makes on a 4x4 button array. Each button has a sound tied to it and after they hit record, I want to mix the audio that gets played and save it to a file so they can listen to it and play over it later.
What makes this even trickier is that there will be a metronome playing and I do not want to mix the metronome sound into the audio that is getting saved.
From what I have found, the only way to go is Audio Units for these features, but I am reluctant to since it seems a little overkill and somewhat complicated to learn. Can Audio Toolbox make this any easier?
Thanks!
In generally, using a AudioToolBox easily implements.
more information, see below sample code. it's a lot of help.
MixerHost
I have a series of sounds that a user will play, rearrange, and edit etc. while using my app. When the user is finished, I want them to be able to save their work and record it to an mp3.
I don't want to play it through speakers and record it with the mic since that will result in low sound quality and interference. I cannot think of any ways of doing this that doesn't require extra hardware and/or a computer.
How can I do this using just their device?
Well, I would say it cant be done with AVFoundation.
My suggestion is to use Audio Units, and transform all your interactions to an audio graph. at some point you set a render notify on the RemoteIO so every time it renders sounds to the speakers you get a callback where you can write it down those frames/packets/data into a file.
I will probably suggest to use AAC(m4a) over MP3. I am not very fond of MP3, and to be honest as far as I know the sdk does not provide encoding to MP3, probably due to licensing issues. I could be wrong though. Check this sample code below, probably the best sample code you will ever find on Audio units on the web.
AudioGraph by Tom Zic
so I'm making an app and what I need to do is when for example someone starts talking I need to detect that there is a sound and then record it.
I found this tutorial http://mobileorchard.com/tutorial-detecting-when-a-user-blows-into-the-mic/ but it starts the recording on the beginning and then based on the recording it detects the sound.
Is there any other way to detect a sound without actually starting the recorder first? What I thought of would be having 2 recorders, one for detection and one for actually recording the sound. Another solution would be to edit (trim) the sound after it's recorded.
Are these approaches somehow standard or is there a better way to detect sound?
Thanks.
edit: if anyone ever reads this, I also found this http://bonkel.wordpress.com/2010/03/03/frequency-detection-using-fourier-transform/
If you don't mind getting a little dirty, you could go down to a lower level, to CoreAudio, and read data out of the input buffers until you see values exceeding your threshold, and start recording those input buffers, or triggering a high level recording call. You can similarly stop recording after a period of silence.
If you use CoreAudio, you have a lot of control over what you record. You could, pretty easily, filter out background noise, or add beeps to signify when the recording stopped due to silence, and even add markers to use later to match time to the recording.
CoreAudio does require you to do more work. You will have to read the microphone buffers on a timely basis and either save or discard the data pretty quickly in order not to drop any sound data. This isn't that hard, as the devices have plenty of CPU power to do that and other tasks at the same time - you just have to have a good grasp of CoreAudio.
There are plenty of Apple CoreAudio samples that can guide you. The WWDC 2010 and 2010 CoreAudio sessions are also a must-see.
You could use either the Audio Queue or the Core Audio (RemoteIO Audio Unit) API. Unless your app requires low latency, the Audio Queue API may be simpler to use.
You need to start the recording API to detect any sound, but you don't need to save everything you get from the recording callback to a file.
I want to record the sound of my iPhone-App. So like someone is playing something on a iPhone instrument and after that you can hear it.
Is it possible without the micro?
Do you mean an App you build yourself? If yes, you could just save the rendered waveform (maybe encoded/compressed to save space) for later playback. (see: Extended Audio File Services, it can write the same AudioBufferList to a file that you would render to the RemoteAudio Unit when playing audio in your Instrument-App)
[Edit: removed comments on recording third-party app audio output ...]
With the AVFoundation you are currently using, you're always working on the level of the sound files. Your code never sees the actual audio signal. Thus, you can't 'grab' the audio signal that your app generates when it is used. Also, AVAudioPlayer does not provide any means of getting to the final signal. If you're using multiple instances of AVAudio player to play multiple sounds at the same time you also wouldn't be able to get at the mixed signal.
Alas, you probably need to use CoreAudio which is a much more low level interface.
I'd like to suggest an alternative approach: Instead of recording the audio output, why not record the sequence of actions together with their time which lead to the audio being played? Write this sequence of events to a file and read it back in to reproduce the 'performance' - it's a bit like your own MIDI sequencer :)