From what I've found online so far, it seems as though the Spotify SDK does not allow developers to manipulate audio (by slowing down songs/changing their pitch); all it allows you to do is simply play the audio at its original pitch/speed.
What I'm wondering is how apps like the Amazing Slow Downer (https://apps.apple.com/us/app/amazing-slow-downer/id308998718) are able to manipulate audio pitch/tempo from Spotify/Apple Music?
I am trying to accomplish this for an iOS app I am building, but I have no idea where to start -- I would appreciate any help in pointing me in the right direction to learn how I can do this!
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I would like to modulate the signal from the mic input with a sine wave at 200HZ (FM only). Anyone know of any good tutorials/articles that will help get me started?
Any info is very welcome
Thanks
I suggest you start here Audio File Stream Services Reference
Here you can also find some basic tutorials: Getting Started with Audio & Video.
Especially the SpeakHere example app could be interesting
Hope that helps you
The standard way to do audio processing in iOS or OSX is Core Audio. Here's Apple's overview of the framework.
However, Core Audio has a reputation of being very difficult to learn, especially if you don't have experience with C. If you're still wanting to learn Core Audio, then this book is the way to go: Learning Core Audio.
There are simpler ways to work with audio on iOS and OSX, one of them being AudioKit, which was developed specifically so developers can quickly prototype audio without having to deal with lower-level memory management, buffers, and pointer arithmetic.
There are examples showing both FM synthesis and audio input via the microphone, so you should have everything you need :)
Full disclosure: I am one of the developers of AudioKit.
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
I am trying to create a video player for iOS, but with some additional audio track reading. I have been checking out MPVideoPlayerController, and also AVPlayer in the AV Foundation, but it's all kinda vague.
What I am trying to do is play a video (from a local .mp4), and while the movie is playing get the current audio buffer/frames, so I can do some calculations and other (not video/audio relevant) actions that depend on the currently played audio. This means that the video should keep on playing, with its audio tracks, but I also want the live raw audio data for calculations (like i.e.: getting the amplitude for certain frequency's).
Does anyone have an example or hints to do this ? Of-course I checked out Apple's AV Foundation library documentation, but it was not clear enough for me.
After a really (really) long time Googling, I found a blog post that describes MTAudioProcessingTap. Introduced in iOS 6.0, it solves my problem perfectly.
The how-to/blogpost can be found here : http://chritto.wordpress.com/2013/01/07/processing-avplayers-audio-with-mtaudioprocessingtap/
I Hope it helps anyone else now....... The only thing popping up for me Googling (with a lot of different terms) is my own post here. And as long as you don't know MTAudioProcessingTap exists, you don't know how to Google for it :-)
i'm working on creating an app that streams mp3s from a server. i'm using Matt Gallagher's AudioStreamer to accomplish this, but noticed that (especially on non-wifi) that it takes a few seconds to buffer and start streaming the audio. I'm looking to minimize this, à la Spotify, which is does it almost instantly.
What's the best way of doing this?
Spotify does a lot of clever stuff to ensure music streams instantly. I don't know how you're exactly implementing your streaming, but it's worth reading this article to get a handle on what exactly is going on when you 'stream' from Spotify:
http://pansentient.com/2011/04/spotify-technology-some-stats-and-how-spotify-works/
There's a lot of 'nifty' logic going on involving Spotify predicting which tracks you're likely to play in the future and pre-fetching them. Their mobile apps originally didn't take advantage of this to the same extent as the desktop, but I suspect as the apps have matured they've drip-fed some of the platform improvements down to mobile.
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.