RemoteIO

I suggest to analyze the nature of the echo you are having in your system (for example test the echo path). Perhaps it is outside the capabilities of your built-in echo cancellation.

It turned out it was a programming error due to misunderstandings in Swift language. The line that was culprit was this:

-50 (kAudio_ParamError) means one of the parameters you passed is wrong.

A common error with AudioUnitRender is to pass an AudioBufferList with the wrong number of mNumberBuffers (you may be recording non-interleaved stereo) or the AudioBuffers themselves may be the wrong size or have the wrong number of channels.

I encounter this problem whenever I forget that the simulator and device remote audio units have different default stream formats and don't explicitly set them via

I have fixed this by changing the value of preferred IO buffer duration of Audio Session. Currently I am setting it to 0.01.

The audio latency seems to depend on the specific device model, audio session and options, requested sample rate and buffer size, and whether any other audio (background or recently closed app) is or has recently been playing or recording on the system. Whether or not the internal audio amplifier circuits (etc.) need to be powered up or are already turned on may make the biggest difference. Requesting certain sample rates seems to also cause extra time due to the buffering potentially needed by the OS resampling and mixer code.

So likely (2) or (3).

The best way to minimize latency when using RemoteIO is to request very short buffers (1 to 6 mS) in the audio session setup, start the audio session and Audio Unit way ahead of time (at app startup, view load, etc.), then fill the callback buffers with zeros (or discard recorded callback data) until you need sound.

It turns out the type of the Audio Unit was wrong.

I changed for kAudioUnitType_FormatConverter for the type and kept the same subtype.

Both get/set the tempo now works as expected.

I'm still unclear about why I didn't get any error, neither on setting up the audio unit, nor when setting the value for the rate.

The buffer duration is given to the RemoteIO callback in the form of the frameCount (proportional to the number of samples in the callback buffer) at a known sample rate. Any other notification would be asynchronous to this callback information, and thus possibly received at the wrong time compared to the actual change (which happens in the audio thread, not in the UI main run loop).

But your audio callback could change some visible state (global or in the parameter struct) which could be found by any other polling thread or consumer thread after the buffer duration update.

After wasting 2 days, I have found the issue is with not flushing the audio data using ExtAudioFileDispose call. As the final audio data was not flushed, the wav file was partial and corrupted. It was only played in VLC player and not any other players. After using the ExtAudioFileDispose, the Audio plays well with AVAudioPlayer also.