How to get smaller buffer size in multi-channel audio application with Oboe

Question

I'm using Oboe 1.2 in an audio android application. When I call getFramesPerBurst(), which gives the endpoint buffer size, I get expected results (240 frames) if the number of output channels is set to 2. However when I set 4 output channels, the value returned by getFramesPerBurst() is around 960 (!). Is that normal ? Is that a limitation of the hardware (I tested on 4 different devices though, with different os version) ? A limitation of Oboe ? I notice also that this value is different than the value given by the property PROPERTY_OUTPUT_FRAMES_PER_BUFFER of AudioManager from the AudioService.

oboe::AudioStreamBuilder builder;
if (!oboe::AudioStreamBuilder::isAAudioRecommended()){
    builder.setAudioApi(oboe::AudioApi::OpenSLES);
}
builder.setSharingMode(oboe::SharingMode::Exclusive);
builder.setFormat(oboe::AudioFormat::Float);
builder.setChannelCount(4);
builder.setCallback(&_oboeCallback);
builder.setPerformanceMode(oboe::PerformanceMode::LowLatency);

oboe::Result result = builder.openStream(&_stream);

if (result == oboe::Result::OK) {
    int framePerBurst = _stream->getFramesPerBurst(); // gives value around 960 for 4 channels, 240 for 2 channels
    _stream->setBufferSizeInFrames(2*framePerBurst);
}

Answer 1

Unless you are connecting to an audio device which actually has 4 independent channels (e.g. a USB audio interface or DJ controller like this one) then your 4 channel stream will need to be mixed into an N channel stream where N is the number of channels in your audio device. This could be 2 (stereo) for headphones or 1 (mono) for a built-in speaker.

The mixer introduces latency and larger buffer sizes. This is the difference in buffer sizes you see when you request a channel count of 2 vs 4.

For the lowest latency always leave the channel count unspecified when creating the stream, then do any channel count conversion inside your own app. There's an example of this here.