Context:
I am currently developing an app using the Push-to-Talk (PTT) framework. I have reviewed both the PTT framework documentation and the CallKit demo project to better understand how to properly manage audio session activation and AVAudioEngine setup.
I am not activating the audio session manually. The audio session configuration is handled in the incomingPushResult or didBeginTransmitting callbacks from the PTChannelManagerDelegate.
I am using a single AVAudioEngine instance for both input and playback. The engine is started in the didActivate callback from the PTChannelManagerDelegate. When I receive a push in full duplex mode, I set the active participant to the user who is speaking.
Issue
When I attempt to talk while the other participant is already speaking, my input tap on the input node takes a few seconds to return valid PCM audio data. Initially, it returns an empty PCM audio block.
Details:
- The audio session is already active and configured with
.playAndRecord. - The input tap is already installed when the engine is started.
- When I talk from a neutral state (no one is speaking), the system plays the standard "microphone activation" tone, which covers this initial delay. However, this does not happen when I am already receiving audio.
Assumptions / Current Setup
- Because the audio session is active in play and record, I assumed that microphone input would be available immediately, even while receiving audio.
- However, there seems to be a delay before valid input is delivered to the tap, only occurring when switching from a receive state to simultaneously talking.
Questions
- Is this expected behavior when using the PTT framework in full duplex mode with a shared AVAudioEngine?
- Should I be restarting or reconfiguring the engine or audio session when beginning to talk while receiving audio?
- Is there a recommended pattern for managing microphone readiness in this scenario to avoid the initial empty PCM buffer?
- Would using separate engines for input and output improve responsiveness?
I would like to confirm the correct approach to handling simultaneous talk and receive in full duplex mode using PTT framework and AVAudioEngine. Specifically, I need guidance on ensuring the microphone is ready to capture audio immediately without the delay seen in my current implementation.
Relevant Code Snippets
Engine Setup
func setup() {
let input = audioEngine.inputNode
do {
try input.setVoiceProcessingEnabled(true)
} catch {
print("Could not enable voice processing \(error)")
return
}
input.isVoiceProcessingAGCEnabled = false
let output = audioEngine.outputNode
let mainMixer = audioEngine.mainMixerNode
audioEngine.connect(pttPlayerNode, to: mainMixer, format: outputFormat)
audioEngine.connect(beepNode, to: mainMixer, format: outputFormat)
audioEngine.connect(mainMixer, to: output, format: outputFormat)
// Initialize converters
converter = AVAudioConverter(from: inputFormat, to: outputFormat)!
f32ToInt16Converter = AVAudioConverter(from: outputFormat, to: inputFormat)!
audioEngine.prepare()
}
Input Tap Installation
func installTap() {
guard AudioHandler.shared.checkMicrophonePermission() else {
print("Microphone not granted for recording")
return
}
guard !isInputTapped else {
print("[AudioEngine] Input is already tapped!")
return
}
let input = audioEngine.inputNode
let microphoneFormat = input.inputFormat(forBus: 0)
let microphoneDownsampler = AVAudioConverter(from: microphoneFormat, to: outputFormat)!
let desiredFormat = outputFormat
let inputFramesNeeded = AVAudioFrameCount((Double(OpusCodec.DECODED_PACKET_NUM_SAMPLES) * microphoneFormat.sampleRate) / desiredFormat.sampleRate)
input.installTap(onBus: 0, bufferSize: inputFramesNeeded, format: input.inputFormat(forBus: 0)) { [weak self] buffer, when in
guard let self = self else { return }
// Output buffer: 1920 frames at 16kHz
guard let outputBuffer = AVAudioPCMBuffer(pcmFormat: desiredFormat, frameCapacity: AVAudioFrameCount(OpusCodec.DECODED_PACKET_NUM_SAMPLES)) else { return }
outputBuffer.frameLength = outputBuffer.frameCapacity
let inputBlock: AVAudioConverterInputBlock = { inNumPackets, outStatus in
outStatus.pointee = .haveData
return buffer
}
var error: NSError?
let converterResult = microphoneDownsampler.convert(to: outputBuffer, error: &error, withInputFrom: inputBlock)
if converterResult != .haveData {
DebugLogger.shared.print("Downsample error \(converterResult)")
} else {
self.handleDownsampledBuffer(outputBuffer)
}
}
isInputTapped = true
}
