feat(apple): gamepad UI v2 — controller settings + add host, aurora, macOS

Sources reorganized (client: Home/Session/Settings/Stores/Support/Trust; kit:
Audio/Connection/Gamepad/Input/Support/Video/Views) with the big files split
along the same seams.

The gamepad mode is couch-complete, and now on macOS too (the living-room
Mac case), not just iOS/iPadOS:

- GamepadSettingsView: a console-style, fully controller-navigable settings
  screen (X from the launcher) — up/down moves focus, left/right steps values
  (clamped, boundary thud), A cycles/toggles, B closes; the focused row shows a
  one-line description. Backed by GamepadMenuList, the vertical sibling of
  GamepadCarousel, and SettingsOptions — the option lists hoisted out of
  SettingsView statics and shared by the touch, tvOS and gamepad settings.
- GamepadAddHostView + GamepadKeyboard: register a host end to end with a pad
  — field rows open an on-screen controller keyboard (dpad grid, A types,
  X backspaces, B done); the launcher carousel ends in an Add Host tile, so
  the dead-end "add one with touch first" empty state is gone.
- Launcher polish: contextual hint bar with the pad's real button glyphs,
  controller name + battery chip, one shared console chrome.
- GamepadScreenBackground: an animated aurora (TimelineView-driven drifting
  blobs in the brand's violet family, breathing radii, slow hue shift,
  legibility scrim; freezes under Reduce Motion). Pure SwiftUI on purpose — a
  .metal library only bundles reliably in one of the two build systems (SPM vs
  the xcodeproj's synced folders) these sources compile under.
- macOS port: settings/add-host/library present as sized sheets (a macOS sheet
  takes its content's IDEAL size, and the GeometryReader-driven screens
  collapsed to nothing), NSScreen-based mode lists, scroll indicators .never
  (the "always show scroll bars" setting overrides .hidden), tray scrims so
  scrolled rows dim under the pinned title/hints, extra title clearance, and a
  PUNKTFUNK_FORCE_GAMEPAD_UI=1 dev hook — launcher/settings/add-host/keyboard/
  library render-verified live on a real Mac + LAN hosts.
- GamepadMenuInput: X button support, and (re)start now snapshots held buttons
  so a controller handoff press never fires twice (the B that closed the
  keyboard no longer also cancels the screen underneath).
- Cleanups: one "Connection failed" alert in ContentView instead of one per
  home screen; HostDiscovery.advertises/unsaved shared by both home screens.
- host: can_encode_444 stub for the non-Linux/Windows host build (the macOS
  synthetic-source loopback used by the Swift tests).

Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
This commit is contained in:
2026-07-02 11:05:10 +02:00
parent e925d00194
commit 133e25849d
84 changed files with 4231 additions and 2698 deletions
@@ -0,0 +1,88 @@
// CoreAudio HAL device enumeration for the Settings pickers. Devices are persisted by
// UID (stable across reboots/replugs AudioDeviceIDs are not); the empty UID means
// "system default", which additionally tracks default-device changes because we then
// never pin the engine to a concrete device.
#if os(macOS)
import CoreAudio
import Foundation
public struct AudioDevice: Hashable, Identifiable, Sendable {
public let uid: String
public let name: String
public var id: String { uid }
}
public enum AudioDevices {
/// Output-capable devices (speakers, headphones, multi-output).
public static func outputs() -> [AudioDevice] {
all().filter { hasStreams($0, scope: kAudioObjectPropertyScopeOutput) }
.compactMap(describe)
}
/// Input-capable devices (microphones, interfaces).
public static func inputs() -> [AudioDevice] {
all().filter { hasStreams($0, scope: kAudioObjectPropertyScopeInput) }
.compactMap(describe)
}
/// Resolve a persisted UID to the current AudioDeviceID nil when unplugged.
static func deviceID(forUID uid: String) -> AudioDeviceID? {
all().first { id in
stringProperty(id, kAudioDevicePropertyDeviceUID) == uid
}
}
private static func all() -> [AudioDeviceID] {
var address = AudioObjectPropertyAddress(
mSelector: kAudioHardwarePropertyDevices,
mScope: kAudioObjectPropertyScopeGlobal,
mElement: kAudioObjectPropertyElementMain)
var size: UInt32 = 0
guard AudioObjectGetPropertyDataSize(
AudioObjectID(kAudioObjectSystemObject), &address, 0, nil, &size) == noErr,
size > 0
else { return [] }
var ids = [AudioDeviceID](
repeating: 0, count: Int(size) / MemoryLayout<AudioDeviceID>.size)
guard AudioObjectGetPropertyData(
AudioObjectID(kAudioObjectSystemObject), &address, 0, nil, &size, &ids) == noErr
else { return [] }
return ids
}
private static func hasStreams(
_ id: AudioDeviceID, scope: AudioObjectPropertyScope
) -> Bool {
var address = AudioObjectPropertyAddress(
mSelector: kAudioDevicePropertyStreams,
mScope: scope,
mElement: kAudioObjectPropertyElementMain)
var size: UInt32 = 0
return AudioObjectGetPropertyDataSize(id, &address, 0, nil, &size) == noErr && size > 0
}
private static func describe(_ id: AudioDeviceID) -> AudioDevice? {
guard let uid = stringProperty(id, kAudioDevicePropertyDeviceUID),
let name = stringProperty(id, kAudioObjectPropertyName)
else { return nil }
return AudioDevice(uid: uid, name: name)
}
private static func stringProperty(
_ id: AudioDeviceID, _ selector: AudioObjectPropertySelector
) -> String? {
var address = AudioObjectPropertyAddress(
mSelector: selector,
mScope: kAudioObjectPropertyScopeGlobal,
mElement: kAudioObjectPropertyElementMain)
var ref: CFString?
var size = UInt32(MemoryLayout<CFString?>.size)
let status = withUnsafeMutablePointer(to: &ref) { p in
AudioObjectGetPropertyData(id, &address, 0, nil, &size, p)
}
guard status == noErr, let ref else { return nil }
return ref as String
}
}
#endif
@@ -0,0 +1,129 @@
import AVFoundation
import os
/// SPSC-ish jitter ring (interleaved float, `channels` per frame), drain thread render
/// callback. The unfair lock is held for microseconds; fine at render-callback rates. Priming:
/// reads return silence until enough is buffered (at least `prefill`, and at least one
/// packet more than the device's render quantum large-buffer devices would otherwise
/// chronically out-demand the prefill and oscillate prime dropout re-prime), and an
/// underrun re-primes, concealing jitter as one short dip instead of sustained crackle.
/// All counts stay whole frames (multiples of `channels`), so the interleave can never slip.
final class AudioRing: @unchecked Sendable {
private var buf: [Float]
private var readIdx = 0
private var writeIdx = 0
private var primed = false
private var renderQuantum = 0
private let prefill: Int
private let highWater: Int
private let channels: Int
private let lock = OSAllocatedUnfairLock()
/// `capacity`/`prefill` in samples (interleaved `channels` per frame, both whole frames).
init(capacity: Int, prefill: Int, channels: Int) {
buf = [Float](repeating: 0, count: capacity)
self.prefill = prefill
self.channels = channels
highWater = prefill * 4
}
func write(_ samples: UnsafePointer<Float>, count: Int) {
lock.lock()
defer { lock.unlock() }
let capacity = buf.count
// A single write larger than the whole ring would push readIdx PAST writeIdx below
// (inverting the valid range corruption). It never happens (one decoded packet is far
// under capacity), but guard rather than corrupt.
guard count <= capacity else { return }
if writeIdx + count - readIdx > capacity {
readIdx = writeIdx + count - capacity // overflow: drop oldest
}
for i in 0..<count {
buf[(writeIdx + i) % capacity] = samples[i]
}
writeIdx += count
// Latency clamp: both ends run at 48 kHz, so backlog from a network stall (or
// creeping host-vs-DAC clock skew) never drains on its own without this, one
// 300 ms hiccup leaves audio 300 ms behind video for the rest of the session.
// Shedding down to 2× prefill costs one audible blip instead.
if writeIdx - readIdx > highWater {
readIdx = writeIdx - prefill * 2
}
}
/// Fills `out` completely (silence beyond what's buffered).
func read(into out: UnsafeMutablePointer<Float>, count: Int) {
lock.lock()
defer { lock.unlock() }
renderQuantum = max(renderQuantum, count)
let available = writeIdx - readIdx
if !primed {
// One 5 ms host packet (240 frames × channels) of slack beyond the device's demand.
if available >= max(prefill, renderQuantum + 240 * channels) {
primed = true
} else {
for i in 0..<count { out[i] = 0 }
return
}
}
let n = min(available, count)
let capacity = buf.count
for i in 0..<n {
out[i] = buf[(readIdx + i) % capacity]
}
readIdx += n
if n < count {
for i in n..<count { out[i] = 0 }
primed = false // underrun re-prime before resuming
}
}
}
/// CoreAudio channel layout for the canonical wire order FL FR FC LFE RL RR [SL SR]. nil for
/// stereo (the standard layout is correct). For 5.1/7.1 we list explicit channel labels via
/// `kAudioChannelLayoutTag_UseChannelDescriptions` preset tags (DTS_5_1 etc.) don't reliably
/// match Moonlight's order. NB the 7.1 mapping (verified against the WASAPI 0x63F + SPA orderings):
/// wire idx 4-5 = RL/RR = the WAVE *back* pair LeftSurround/RightSurround; idx 6-7 = SL/SR = the
/// WAVE *side* pair LeftSurroundDirect/RightSurroundDirect. (Using RearSurround* for 6-7 would
/// swap side/back vs the Windows/Linux clients.)
func wireChannelLayout(channels: Int) -> AVAudioChannelLayout? {
let labels: [AudioChannelLabel]
switch channels {
case 6:
labels = [
kAudioChannelLabel_Left, kAudioChannelLabel_Right, kAudioChannelLabel_Center,
kAudioChannelLabel_LFEScreen, kAudioChannelLabel_LeftSurround,
kAudioChannelLabel_RightSurround,
]
case 8:
labels = [
kAudioChannelLabel_Left, kAudioChannelLabel_Right, kAudioChannelLabel_Center,
kAudioChannelLabel_LFEScreen,
kAudioChannelLabel_LeftSurround, kAudioChannelLabel_RightSurround, // wire RL/RR (back)
kAudioChannelLabel_LeftSurroundDirect, kAudioChannelLabel_RightSurroundDirect, // wire SL/SR (side)
]
default:
return nil
}
let size = MemoryLayout<AudioChannelLayout>.size
+ (labels.count - 1) * MemoryLayout<AudioChannelDescription>.stride
let raw = UnsafeMutableRawPointer.allocate(byteCount: size, alignment: 16)
defer { raw.deallocate() }
let layout = raw.bindMemory(to: AudioChannelLayout.self, capacity: 1)
layout.pointee.mChannelLayoutTag = kAudioChannelLayoutTag_UseChannelDescriptions
layout.pointee.mChannelBitmap = AudioChannelBitmap(rawValue: 0)
layout.pointee.mNumberChannelDescriptions = UInt32(labels.count)
// `mChannelDescriptions` is the C variable-length tail array (declared `[1]`, over-allocated
// above). Scope the pointer with `withUnsafeMutablePointer` taking `&mChannelDescriptions`
// inline yields a pointer valid only for that expression, so building a buffer from it that
// outlives the call is a dangling-pointer bug. Inside the closure it stays valid while we fill it.
withUnsafeMutablePointer(to: &layout.pointee.mChannelDescriptions) { tail in
let descs = UnsafeMutableBufferPointer(start: tail, count: labels.count)
for (i, lbl) in labels.enumerated() {
descs[i] = AudioChannelDescription(
mChannelLabel: lbl, mChannelFlags: AudioChannelFlags(rawValue: 0),
mCoordinates: (0, 0, 0))
}
}
return AVAudioChannelLayout(layout: layout)
}
@@ -0,0 +1,140 @@
// Opus PCM through CoreAudio's built-in codec (kAudioFormatOpus, macOS 10.13+ / iOS
// 11+) no bundled libopus. The host's audio plane is raw Opus packets (48 kHz stereo,
// one frame per packet); AVAudioConverter handles them as single-packet
// AVAudioCompressedBuffers with explicit packet descriptions.
//
// Both classes are single-threaded by contract (one per direction, owned by their
// drain/capture pipelines).
import AVFoundation
enum OpusCodecError: Error {
/// CoreAudio rejected the Opus stream format or had no converter for it.
case unavailable
case convertFailed(String)
}
/// 48 kHz stereo float32 interleaved the PCM side of both converters and the layout
/// of the playback ring buffer.
func opusPCMFormat() -> AVAudioFormat? {
AVAudioFormat(
commonFormat: .pcmFormatFloat32, sampleRate: 48_000, channels: 2, interleaved: true)
}
/// The compressed side: raw Opus, `framesPerPacket` nominal samples per packet at 48 kHz
/// (240 = the host's 5 ms audio plane; 960 = the 20 ms packets the encoder emits).
private func opusFormat(framesPerPacket: UInt32) -> AVAudioFormat? {
var desc = AudioStreamBasicDescription(
mSampleRate: 48_000,
mFormatID: kAudioFormatOpus,
mFormatFlags: 0,
mBytesPerPacket: 0,
mFramesPerPacket: framesPerPacket,
mBytesPerFrame: 0,
mChannelsPerFrame: 2,
mBitsPerChannel: 0,
mReserved: 0)
return AVAudioFormat(streamDescription: &desc)
}
final class OpusDecoder {
private let converter: AVAudioConverter
private let inBuf: AVAudioCompressedBuffer
private let opus: AVAudioFormat
let pcmFormat: AVAudioFormat
/// `framesPerPacket`: the sender's packet duration in samples (host audio = 240).
init(framesPerPacket: UInt32) throws {
guard let pcm = opusPCMFormat(), let opus = opusFormat(framesPerPacket: framesPerPacket),
let converter = AVAudioConverter(from: opus, to: pcm)
else { throw OpusCodecError.unavailable }
self.converter = converter
self.opus = opus
self.pcmFormat = pcm
inBuf = AVAudioCompressedBuffer(
format: opus, packetCapacity: 1, maximumPacketSize: 1500)
}
/// Decode one Opus packet into `out` (whose format must be `pcmFormat`); returns the
/// number of frames written. Empty packets (DTX) decode to 0 frames.
func decode(_ packet: Data, into out: AVAudioPCMBuffer) throws -> AVAudioFrameCount {
guard !packet.isEmpty else { return 0 }
guard packet.count <= Int(inBuf.maximumPacketSize) else {
throw OpusCodecError.convertFailed("packet larger than maximumPacketSize")
}
packet.withUnsafeBytes { raw in
inBuf.data.copyMemory(from: raw.baseAddress!, byteCount: raw.count)
}
inBuf.byteLength = UInt32(packet.count)
inBuf.packetCount = 1
inBuf.packetDescriptions![0] = AudioStreamPacketDescription(
mStartOffset: 0, mVariableFramesInPacket: 0, mDataByteSize: UInt32(packet.count))
out.frameLength = 0
var fed = false
var convError: NSError?
let status = converter.convert(to: out, error: &convError) { [inBuf] _, outStatus in
if fed {
outStatus.pointee = .noDataNow
return nil
}
fed = true
outStatus.pointee = .haveData
return inBuf
}
if status == .error {
throw OpusCodecError.convertFailed(convError?.localizedDescription ?? "decode")
}
return out.frameLength
}
}
final class OpusEncoder {
/// The encoder's packet duration: 960 samples = 20 ms, CoreAudio's default Opus
/// framing. The host's mic service decodes any Opus frame size up to 120 ms.
static let framesPerPacket: AVAudioFrameCount = 960
private let converter: AVAudioConverter
private let outBuf: AVAudioCompressedBuffer
let pcmFormat: AVAudioFormat
init() throws {
guard let pcm = opusPCMFormat(),
let opus = opusFormat(framesPerPacket: UInt32(Self.framesPerPacket)),
let converter = AVAudioConverter(from: pcm, to: opus)
else { throw OpusCodecError.unavailable }
converter.bitRate = 96_000
self.converter = converter
self.pcmFormat = pcm
outBuf = AVAudioCompressedBuffer(
format: opus, packetCapacity: 4, maximumPacketSize: 1500)
}
/// Encode exactly `framesPerPacket` frames of `pcmFormat` audio; returns the encoded
/// packets (normally one).
func encode(_ pcm: AVAudioPCMBuffer) throws -> [Data] {
outBuf.byteLength = 0
outBuf.packetCount = 0
var fed = false
var convError: NSError?
let status = converter.convert(to: outBuf, error: &convError) { _, outStatus in
if fed {
outStatus.pointee = .noDataNow
return nil
}
fed = true
outStatus.pointee = .haveData
return pcm
}
if status == .error {
throw OpusCodecError.convertFailed(convError?.localizedDescription ?? "encode")
}
guard let descs = outBuf.packetDescriptions else { return [] }
return (0..<Int(outBuf.packetCount)).map { i in
let d = descs[i]
return Data(
bytes: outBuf.data.advanced(by: Int(d.mStartOffset)),
count: Int(d.mDataByteSize))
}
}
}
@@ -0,0 +1,391 @@
// Session audio, both directions:
//
// host speaker: a drain thread pulls Opus packets (nextAudio, its own plane in the
// core), decodes via OpusDecoder, and writes PCM into a jitter ring; an
// AVAudioSourceNode pulls from the ring (silence on underrun with re-priming, so a
// network gap costs one dip, not permanent crackle).
//
// mic host: a second AVAudioEngine taps the input device, resamples to 48 kHz
// stereo, slices 20 ms chunks, Opus-encodes, and sendMic()s each packet the host
// feeds them into a virtual PipeWire source.
//
// Devices are chosen by UID ("" = system default: the engine is then never pinned to a
// concrete device and follows default-device changes). Two engines, not one a single
// AVAudioEngine ties input+output to one aggregate clock, separate engines keep
// arbitrary mic/speaker combinations trivial.
import AVFoundation
import os
private let log = Logger(subsystem: "io.unom.punktfunk", category: "audio")
/// Render-block-owned scratch storage: freed exactly when the closure (and thus the
/// last possible render call) is released never racing CoreAudio.
private final class ScratchBuffer {
// 8192 frames × up to 8 channels (7.1) the render block caps `frames` at 8192.
let ptr = UnsafeMutablePointer<Float>.allocate(capacity: 8192 * 8)
deinit { ptr.deallocate() }
}
public final class SessionAudio {
private let connection: PunktfunkConnection
private let flag = StopFlag()
private let drainDone = DispatchSemaphore(value: 0)
/// Owns the engine handles + drainStarted, paired with `flag`: stop() sets the flag
/// BEFORE taking the engines, every publisher re-checks the flag under this lock
/// after publishing-side work so a startCapture racing stop() (the mic-permission
/// callback arrives whenever the user clicks the prompt) can never leave a hot
/// microphone with no owner.
private let stateLock = NSLock()
private var playbackEngine: AVAudioEngine?
private var captureEngine: AVAudioEngine?
private var drainStarted = false
#if !os(macOS)
/// AVAudioSession `setCategory`/`setActive` are synchronous and block on the audio server, so
/// they must not run on the main thread (UI stall AVFoundation warns about it). PROCESS-WIDE
/// (static) so every SessionAudio shares one serial queue: the AVAudioSession is a process
/// singleton, and across a reconnect the old session's deactivate must be ordered before the
/// new session's activate (a per-instance queue would let them race and leave the new session's
/// audio deactivated). stop() enqueues its deactivate promptly so it lands before the next
/// session's activate.
private static let sessionQueue = DispatchQueue(label: "io.unom.punktfunk.audio.session")
#endif
public init(connection: PunktfunkConnection) {
self.connection = connection
}
/// Backstop for an owner dropping us without stop() unblocks the drain thread
/// (which captures the connection strongly, NOT self) within one poll timeout.
/// Engine teardown still belongs to stop().
deinit {
flag.stop()
}
/// Start playback (and, if enabled+authorized, the mic uplink). Empty UIDs = system default
/// device; on iOS the UIDs are ignored entirely (routes are AVAudioSession-managed). On macOS
/// the engines start synchronously on the caller's (main) thread. On iOS/tvOS start() is
/// ASYNCHRONOUS: it activates the AVAudioSession off the main thread, then starts the engines on
/// a later main-queue hop (gated by `!flag.isStopped`) so playback is live shortly after, not
/// on return. The mic may start later still if the permission prompt is pending.
public func start(speakerUID: String, micUID: String, micEnabled: Bool) {
#if os(macOS)
// No AVAudioSession on macOS start the engines directly (caller's thread, as before).
startEngines(speakerUID: speakerUID, micUID: micUID, micEnabled: micEnabled)
#else
// Configure + activate the session OFF the main thread (it blocks on the audio server),
// then start the engines back on the main thread once it's active engine routing/format
// depend on the active session. A stop() racing in between is caught by the flag guard.
Self.sessionQueue.async { [weak self] in
guard let self else { return }
self.activateAudioSession(micEnabled: micEnabled)
DispatchQueue.main.async { [weak self] in
guard let self, !self.flag.isStopped else { return }
self.startEngines(speakerUID: speakerUID, micUID: micUID, micEnabled: micEnabled)
}
}
#endif
}
#if !os(macOS)
/// Route + policy live in the session, not per-engine: stereo playback, mic capture when
/// enabled, Bluetooth allowed. Failure is non-fatal (defaults). Runs on `sessionQueue`.
private func activateAudioSession(micEnabled: Bool) {
let session = AVAudioSession.sharedInstance()
do {
#if os(iOS)
if micEnabled {
// .defaultToSpeaker: .playAndRecord otherwise routes to the iPhone EARPIECE; only
// affects the built-in route (headphones/BT still win).
try session.setCategory(
.playAndRecord, mode: .default,
options: [.allowBluetoothA2DP, .defaultToSpeaker])
} else {
try session.setCategory(.playback, mode: .default)
}
#else // tvOS no app-accessible mic
try session.setCategory(.playback, mode: .default)
#endif
try session.setActive(true)
} catch {
log.warning("AVAudioSession setup failed: \(error.localizedDescription)")
}
}
#endif
/// Build + start the playback engine (and the mic uplink when enabled + authorized). Main
/// thread (engine setup); on iOS/tvOS the session is already active by the time this runs.
private func startEngines(speakerUID: String, micUID: String, micEnabled: Bool) {
startPlayback(speakerUID: speakerUID)
#if os(tvOS)
// No app-accessible microphone input on tvOS playback only.
#else
guard micEnabled else { return }
switch AVCaptureDevice.authorizationStatus(for: .audio) {
case .authorized:
startCapture(micUID: micUID)
case .notDetermined:
AVCaptureDevice.requestAccess(for: .audio) { [weak self] granted in
DispatchQueue.main.async {
guard let self, granted, !self.flag.isStopped else { return }
self.startCapture(micUID: micUID)
}
}
default:
log.warning("microphone access denied — mic uplink disabled (System Settings → Privacy)")
}
#endif
}
/// Stop both directions. Safe from any thread; waits the drain thread out ( its
/// poll timeout) so the caller can close the connection right after.
public func stop() {
flag.stop() // before taking the engines see stateLock's comment
stateLock.lock()
let capture = captureEngine
captureEngine = nil
let playback = playbackEngine
playbackEngine = nil
let wasDraining = drainStarted
drainStarted = false
stateLock.unlock()
if let capture {
capture.inputNode.removeTap(onBus: 0)
capture.stop()
}
playback?.stop()
#if !os(macOS)
// Release the session so audio we interrupted (Music, podcasts) gets its resume cue. Like
// activation, setActive is synchronous/blocking run it on the shared serial session queue
// (off the main thread). Enqueued HERE engines already stopped, and BEFORE the drain wait
// below so across a reconnect it lands ahead of the next session's activate on the shared
// queue (otherwise a deferred deactivate could deactivate the new session). Fire-and-forget.
Self.sessionQueue.async {
do {
try AVAudioSession.sharedInstance().setActive(
false, options: .notifyOthersOnDeactivation)
} catch {
log.warning("AVAudioSession deactivation failed: \(error.localizedDescription)")
}
}
#endif
if wasDraining {
_ = drainDone.wait(timeout: .now() + .milliseconds(400))
}
}
// MARK: - Playback (host speaker)
private func startPlayback(speakerUID: String) {
// Build the playback layout from the host-RESOLVED channel count (never the request):
// 2 = stereo / 6 = 5.1 / 8 = 7.1, canonical wire order FL FR FC LFE RL RR SL SR.
let channels = Int(connection.resolvedAudioChannels)
// 1 s interleaved capacity, ~20 ms prefill (four 5 ms host packets of jitter absorption
// before the first sample plays), both scaled by the channel count.
let ring = AudioRing(
capacity: 48_000 * channels, prefill: 960 * channels, channels: channels)
let engine = AVAudioEngine()
#if os(macOS)
if !speakerUID.isEmpty {
if let dev = AudioDevices.deviceID(forUID: speakerUID),
let unit = engine.outputNode.audioUnit {
if !Self.setDevice(dev, on: unit) {
log.error("could not select speaker \(speakerUID) — using default")
}
} else {
log.warning("speaker \(speakerUID) not present — using default")
}
}
#endif
// Engine-native deinterleaved float; the render block deinterleaves from the ring. Surround
// uses an explicit wire-order channel layout; the mixer downmixes to the output device when
// it has fewer speakers (e.g. an iPhone's stereo built-ins). (Explicit if/else rather than
// map/flatMap so it's correct whether the channelLayout initializer is failable or not.)
var format: AVAudioFormat?
if channels == 2 {
format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channels: 2)
} else if let layout = wireChannelLayout(channels: channels) {
format = AVAudioFormat(standardFormatWithSampleRate: 48_000, channelLayout: layout)
}
guard let format else {
log.error("could not build \(channels)-channel audio format — audio disabled")
return
}
let scratch = ScratchBuffer() // block-owned; freed with the closure
let source = AVAudioSourceNode(format: format) { _, _, frameCount, abl -> OSStatus in
let frames = Int(frameCount)
guard frames <= 8192 else { return kAudioUnitErr_TooManyFramesToProcess }
ring.read(into: scratch.ptr, count: frames * channels)
let buffers = UnsafeMutableAudioBufferListPointer(abl)
// Deinterleave the wire-order interleaved ring into the engine's per-channel buses.
if buffers.count >= channels {
for ch in 0..<channels {
if let dst = buffers[ch].mData?.assumingMemoryBound(to: Float.self) {
for f in 0..<frames { dst[f] = scratch.ptr[f * channels + ch] }
}
}
}
return noErr
}
engine.attach(source)
engine.connect(source, to: engine.mainMixerNode, format: format)
engine.prepare()
do {
try engine.start()
} catch {
log.error("playback engine failed to start: \(error.localizedDescription)")
return
}
stateLock.lock()
if flag.isStopped {
stateLock.unlock()
engine.stop() // stop() already ran don't strand a started engine
return
}
playbackEngine = engine
stateLock.unlock()
startDrain(into: ring)
}
private func startDrain(into ring: AudioRing) {
stateLock.lock()
drainStarted = true
stateLock.unlock()
let thread = Thread { [connection, flag, drainDone] in
defer { drainDone.signal() }
// Decode happens IN-CORE (libopus multistream) AudioToolbox's Opus path is
// stereo-only and is handed back as interleaved f32 PCM in wire channel order.
while !flag.isStopped {
let pcm: PunktfunkConnection.AudioPCM?
do {
pcm = try connection.nextAudioPcm(timeoutMs: 100)
} catch {
break // session closed
}
guard let pcm, pcm.frameCount > 0 else { continue }
pcm.samples.withUnsafeBufferPointer { p in
if let base = p.baseAddress {
ring.write(base, count: pcm.frameCount * pcm.channels)
}
}
}
}
thread.name = "punktfunk-audio"
thread.qualityOfService = .userInteractive
thread.start()
}
// MARK: - Mic (mic host)
#if !os(tvOS)
private func startCapture(micUID: String) {
let engine = AVAudioEngine()
let input = engine.inputNode
#if os(macOS)
if !micUID.isEmpty {
if let dev = AudioDevices.deviceID(forUID: micUID), let unit = input.audioUnit {
if !Self.setDevice(dev, on: unit) {
log.error("could not select microphone \(micUID) — using default")
}
} else {
log.warning("microphone \(micUID) not present — using default")
}
}
#endif
let inFormat = input.outputFormat(forBus: 0)
guard inFormat.sampleRate > 0, inFormat.channelCount > 0 else {
log.error("no usable input device — mic uplink disabled")
return
}
guard let encoder = try? OpusEncoder(),
let resampler = AVAudioConverter(from: inFormat, to: encoder.pcmFormat),
let chunk = AVAudioPCMBuffer(
pcmFormat: encoder.pcmFormat, frameCapacity: OpusEncoder.framesPerPacket)
else {
log.error("Opus encoder unavailable — mic uplink disabled")
return
}
// Tap-thread-confined state: resample into `staging`, accumulate in `fifo`,
// slice 960-frame chunks for the encoder.
var fifo: [Float] = []
fifo.reserveCapacity(48_000)
var seq: UInt32 = 0
let connection = connection
let flag = flag
input.installTap(onBus: 0, bufferSize: 2048, format: inFormat) { buffer, _ in
if flag.isStopped { return }
let ratio = 48_000 / inFormat.sampleRate
let outCapacity = AVAudioFrameCount(
(Double(buffer.frameLength) * ratio).rounded(.up) + 64)
guard let staging = AVAudioPCMBuffer(
pcmFormat: encoder.pcmFormat, frameCapacity: outCapacity)
else { return }
var fed = false
var convError: NSError?
let status = resampler.convert(to: staging, error: &convError) { _, outStatus in
if fed {
outStatus.pointee = .noDataNow
return nil
}
fed = true
outStatus.pointee = .haveData
return buffer
}
guard status != .error, let p = staging.floatChannelData?[0] else { return }
fifo.append(contentsOf: UnsafeBufferPointer(
start: p, count: Int(staging.frameLength) * 2))
let samplesPerChunk = Int(OpusEncoder.framesPerPacket) * 2
while fifo.count >= samplesPerChunk {
chunk.frameLength = OpusEncoder.framesPerPacket
fifo.withUnsafeBufferPointer { src in
chunk.floatChannelData![0].update(
from: src.baseAddress!, count: samplesPerChunk)
}
fifo.removeFirst(samplesPerChunk)
guard let packets = try? encoder.encode(chunk) else { continue }
for packet in packets {
connection.sendMic(
packet, seq: seq, ptsNs: DispatchTime.now().uptimeNanoseconds)
seq &+= 1
}
}
}
engine.prepare()
do {
try engine.start()
} catch {
log.error("capture engine failed to start: \(error.localizedDescription)")
input.removeTap(onBus: 0)
return
}
stateLock.lock()
if flag.isStopped {
// stop() ran while we were starting (the permission prompt resolves at the
// user's leisure) tear the engine down ourselves, nobody else owns it now.
stateLock.unlock()
input.removeTap(onBus: 0)
engine.stop()
return
}
captureEngine = engine
stateLock.unlock()
log.info("mic uplink started (\(micUID.isEmpty ? "default input" : micUID))")
}
#endif
#if os(macOS)
private static func setDevice(_ id: AudioDeviceID, on unit: AudioUnit) -> Bool {
var dev = id
return AudioUnitSetProperty(
unit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0,
&dev, UInt32(MemoryLayout<AudioDeviceID>.size)) == noErr
}
#endif
}