Wow! After reading your reply, now I can better understand how FMOD Studio works. Thank you very much.
I think I’m almost there now.
Still lots of questions below, but this time it’ll be easier for you to answer, I guess.
Also, this is the last time. I promise.
Q1.
In this page, it says :
“Be aware that this will only happen if there are no pitch / frequency settings applied to the Channel, so this trick is often good for music.”
Why are pitch and frequency specifically mentioned here?
Is it because the two are the only processing which requires resampling under the hood, unlike other DSPs?
Q2.
As you know, in order to achieve low-latency on Android, it’s also important to set OpenSL ES’ buffer size properly using a value returned by getProperty(AudioManager.PROPERTY_OUTPUT_FRAMES_PER_BUFFER).
Now I know fmod.jar’s init() will adjust the sample rate automatically.
But what about buffer size? Will init() take care of it as well?
Q3.
It’s somewhat obscure whether FMOD_System_SetDSPBufferSize() sets FMOD System’s internal buffer or OpenSL ES’ (or AAudio’s).
I once wrote a Windows application with FMOD which used ASIO.
At that time, it crashed when unsupported values by my audio interface were set.
Also, one poster’s question here says only values supported by the audio interface worked.
But your document says
“Sets the buffer size settings for the FMOD software mixing engine.”
So this is confusing.
Is FMOD_System_SetDSPBufferSize() for the FMOD System only?
If yes, how can I set the OpenSL ES’ (or AAudio’s) buffer size?
Even aside from latency issue, setting proper buffersize is important because this accepted answer from an Android audio team developer says there’s a bug on Android M :
“Make your buffer size a multiple of AudioManager.getProperty(PROPERTY_OUTPUT_FRAMES_PER_BUFFER). Otherwise your callback will occasionally get two calls per timeslice rather than one.
… (On Android M, it is very important to use EXACTLY the system buffer size, due to a bug in the buffer handling code.)”