How to use iOS AudioUnit render callback correctly

Question

I'm writing an iOS application that will play audio instructions as one of it's features.

Every time the application wants to play audio it reads from a non-standard file and puts the resulting PCM data for that audio in a buffer in memory.

Even though I have that buffer with the PCM data, I'm having trouble getting the application to actually play the sound. After searching the iOS documentation, I started implementing an AudioUnit. The problem with this AudioUnit is the use of a render callback (as far as I know, the only way to output sound). From Apple's developer documentation :

… render callbacks have a strict performance requirement that you must adhere to. A render callback lives on a real-time priority thread on which subsequent render calls arrive asynchronously. The work you do in the body of a render callback takes place in this time-constrained environment. If your callback is still producing sample frames in response to the previous render call when the next render call arrives, you get a gap in the sound. For this reason you must not take locks, allocate memory, access the file system or a network connection, or otherwise perform time-consuming tasks in the body of a render callback function

If I can't use locks inside the render callback method I can't be reading the buffer while writing in it. There is no opportunity to read the file and write to the buffer because the render callback will be accessing it constantly.

The only example I found actually generated the PCM data inside the render method, which I can't do.

Is this the only way of using AudioUnits (with an asynchronous render callback)?

Is there an alternative to playback PCM data from memory?

Answer 1

Using the RemoteIO Audio Unit might require having a separate data queue (fifo or circular buffer), outside the audio unit callback, which can pre-buffer up enough audio data from a file read, ahead of audio unit render callback, to meet worse case latencies. Then the render callback only needs to do a quick copy of the audio data, and then the update of a write-only flag that indicates that audio data was consumed.

An alternative built into iOS is to use the Audio Queue API, which does the pre-buffering for you. It allows your app to fill a number of larger audio buffers in the main run loop ahead of time. You still have to pre-buffer enough data to allow for the max of file, network, lock or other latencies.

Another strategy is to have alternative audio data to feed the real-time render callback if the file or network read didn't keep up, such as quickly creating an audio buffer that tapers to silence (and then un-tapering when real data starts arriving again).