kAudioDevicePropertyBufferFrameSize replacement for iOS

问题

I was trying to set up an audio unit to render the music (instead of Audio Queue.. which was too opaque for my purposes).. iOS doesn't have this property kAudioDevicePropertyBufferFrameSize.. any idea how I can derive this value to set up the buffer size of my IO unit?

I found this post interesting.. it asks about the possibility of using a combination of kAudioSessionProperty_CurrentHardwareIOBufferDuration and kAudioSessionProperty_CurrentHardwareOutputLatency audio session properties to determine that value.. but there is no answer.. any ideas?

回答1:

You can use the kAudioSessionProperty_CurrentHardwareIOBufferDuration property, which represents the buffer size in seconds. Multiply this by the sample rate you get from kAudioSessionProperty_CurrentHardwareSampleRate to get the number of samples you should buffer.

The resulting buffer size should be a multiple of 2. I believe either 512 or 4096 are what you're likely to get, but you should always base it off of the values returned from AudioSessionGetProperty.

Example:

Float64 sampleRate;
UInt32 propSize = sizeof(Float64);
AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate, 
                        &propSize,
                        &sampleRate);

Float32 bufferDuration;
propSize = sizeof(Float32);
AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareIOBufferDuration, 
                        &propSize,
                        &bufferDuration);

UInt32 bufferLengthInFrames = sampleRate * bufferDuration;

The next step is to find out the input stream format of the unit you're sending audio to. Based on your description, I'm assuming that you're programmatically generating audio to send to the speakers. This code assumes unit is an AudioUnit you're sending audio to, whether that's the RemoteIO or something like an effect Audio Unit.

AudioStreamBasicDescription inputASBD;
UInt32 propSize = sizeof(AudioStreamBasicDescription);
AudioUnitGetProperty(unit,
                     kAudioUnitProperty_StreamFormat,
                     kAudioUnitScope_Input,
                     0,
                     &inputASBD,
                     &propSize);

After this, inputASBD.mFormatFlags will be a bit field corresponding to the audio stream format that unit is expecting. The two most likely sets of flags are named kAudioFormatFlagsCanonical and kAudioFormatFlagsAudioUnitCanonical. These two have corresponding sample types AudioSampleType and AudioUnitSampleType that you can base your size calculation off of.

As an aside, AudioSampleType typically represents samples coming from the mic or destined for the speakers, whereas AudioUnitSampleType is usually for samples that are intended to be processed (by an audio unit, for example). At the moment on iOS, AudioSampleType is a SInt16 and AudioUnitSampleType is fixed 8.24 number stored in a SInt32 container. Here's a post on the Core Audio mailing list explaining this design choice

The reason I hold back from saying something like "just use Float32, it'll work" is because the actual bit representation of the stream is subject to change if Apple feels like it.

回答2:

The audio unit itself decides on the actual buffer size, so the app's audio unit callback has to be able to handle any reasonable size given to it. You can suggest and poll the kAudioSessionProperty_CurrentHardwareIOBufferDuration property, but note that this value can while your app is running (especially during screen lock or call interruptions, etc.) outside of what the app can control.

来源：https://stackoverflow.com/questions/13157523/kaudiodevicepropertybufferframesize-replacement-for-ios

标签

ios

core-audio

audio-streaming

latency