Audio producer threads with OSX AudioComponent con

问题:

This question is not about a plugin, it's about a standalone application program design and is connected with few questions I've asked before.

I have to write a multi-threaded audio synthesizing function, whose amount of data crunching by far exceeds what can get accomodated on the CoreAudio render thread: several thousands of independent amplitude and phase interpolating sample-accurate sine-wave oscillators in real time. This requires more CPU power than any single processor core can bear, with all the optimizations available.

I'm doing my best to learn it, but it seems a wall, not a curve. The consumer thread may be a simple CA real-time priority render callback accepting AudioBufferList iodata, etc… …but what should be the producer thread(s)? If choosing another AudioComponent, it does no better than having it all on the output thread - it only gets more complicated and introduces additional latency.

If putting n parallel AudioComponents into a graph which feeds a ring buffer, which feeds the consumer thread, how can one guarantee it'll not end up on the same thread, stay in sync and sample-accurate?

If writing n traditional POSIX threads with joining outputs, how to achieve the CoreAudio pull-model would coexist with such a push model in real time?

Is there any such freely available example code ? Is there a reference, a textbook or a tutorial in writing such a code? I haven't found any publicly available information. It makes me kind of wonder that such question hadn't been asked before?

Thanks in advance!

回答1:

My strategy would be to have a separate thread (or threads) generating your LPCM audio in real-time and writing it to a suitably sized ring buffer. The core audio thread would read from this ring buffer.

If the reader (core audio thread) requests more audio than is available then you get a drop out (buffer underrun), because the reader cannot wait. This indicates that you need a larger buffer, or that your audio generation algorithm is not realtime and needs optimisation.

If the writer tries to write to the ring buffer when it is full, the oldest audio is overwritten and you have another glitch, this time via overrun. In this case your audio generation code is running too fast and needs to learn how to wait. Here it would be convenient to add write waitability, via a condition variable.

I know of two cool ringbuffer implementations that use virtual memory tricks to efficiently implement the modular memory access:

• Mike Ash's MAMirroredQueue
• Michael Tyson's TPCircularBuffer

If you're looking for something more portable, there's CBuffer for unix-like systems and Magic Ring Buffer for Windows.

However! You don't have to use tricky MMU ring buffers, you can use a plain old portable C version if you like.