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* Live push-to-talk voice for DMs: stream while you talk, voice note as fallback Holding the mic in a DM now streams AAC frames live over the Noise session (walkie-talkie style, ~0.5s mouth-to-ear at one hop) while recording the same audio as a normal voice note. On release the note ships through the existing fileTransfer pipeline; receivers that heard the live stream absorb it silently into the same bubble (matched by the burst ID embedded in the file name), so reliability comes for free and nobody sees duplicates. Protocol: - NoisePayloadType.voiceFrame = 0x08 carrying VoiceBurstPacket (burstID + seq + START/data/END/CANCELED, length-prefixed AAC frames) - 210-byte burst-content budget keeps each Noise packet inside the 256-byte padding bucket: one BLE frame, never the fragment scheduler - fire-and-forget: frames are dropped (never queued) without an established session; live is only offered when the peer is mesh-reachable Receive: - ChatLiveVoiceCoordinator assembles bursts (jitter-ordered, 0.5s gap skip, 3s idle end, flood/size caps), persists progressively as ADTS .aac so even a partial burst is a replayable bubble - live autoplay only when the conversation is on screen, app active, and the new app-info "live voice messages" toggle is on (also gates live sending) - one-playback-at-a-time via a shared ExclusivePlayback slot Capture: - PTTCaptureEngine taps AVAudioEngine, dual-encodes: live AAC frames + the finalized .m4a (same 16kHz/mono/16kbps settings as VoiceRecorder) - VoiceRecordingViewModel now drives a pluggable VoiceCaptureSession; the composer HUD shows a pulsing LIVE treatment when streaming Includes the push-to-talk design doc, 6 new localization keys across all 29 locales, and unit tests for framing, packetizer budget, ADTS output, codec round-trip, and the assembly/absorb lifecycle. Public-mesh PTT (MessageType 0x29) lands separately on top of this. Co-Authored-By: Claude Fable 5 <noreply@anthropic.com> * PTT follow-ups from review + field test: peer-ID normalization, toggle gates inbound, drop-path diagnostics Codex review fixes (#1403): - makeVoiceCaptureSession normalizes the selected peer with toShort() before the reachability/session checks and binds the send target to that same routing ID — a conversation selected under the stable 64-hex Noise key no longer silently falls back to a classic note while the short-ID session is established - the live-voice toggle now gates inbound bursts too: off means classic-notes-only in both directions (no live bubble, partial file, or early notification; the finalized note still arrives), with a test Field-test diagnostics (first device run: DM frames decrypted but no bubble appeared, with no log evidence of which guard dropped them): - coordinator logs undecodable frames (size + hex prefix) and blocked drops - makeAssembly logs directory/file-handle failures instead of returning nil silently - PTTLiveVoiceSession logs capture start and finish (packet/frame/duration counts); PTTCaptureEngine logs engine start success/failure with the input format; BLEService.sendVoiceFrame logs no-session drops Co-Authored-By: Claude Fable 5 <noreply@anthropic.com> * Fix iPhone live-capture failure: dead input unit (AURemoteIO -10851, 0 Hz) Field testing showed the phone's live capture failing at mic enable with AURemoteIO -10851 and an input format of 0 Hz / 2 ch — an input unit bound to an earlier (playback-only or settling) audio session. The Mac, which has no session lifecycle, captured fine, which is why public bursts from the Mac worked while phone-side sends degraded from working (first hold) to sporadic to dead across holds. Three layers of defense: - PTTCaptureEngine recreates its AVAudioEngine on every start(), after the session is configured, so the input unit binds to the session that is active now; a dead input (0 Hz or 0 channels) is now a distinct, logged error instead of a silent setup failure - PTTLiveVoiceSession retries the capture start once after a 150 ms route-settle pause - VoiceRecordingViewModel falls back to the classic VoiceRecorder within the same hold if the live engine still cannot start — a route glitch now costs the live stream, never the voice note Co-Authored-By: Claude Fable 5 <noreply@anthropic.com> --------- Co-authored-by: jack <jackjackbits@users.noreply.github.com> Co-authored-by: Claude Fable 5 <noreply@anthropic.com>
176 lines
6.6 KiB
Swift
176 lines
6.6 KiB
Swift
//
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// PTTAudioCodec.swift
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// bitchat
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//
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// This is free and unencumbered software released into the public domain.
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// For more information, see <https://unlicense.org>
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//
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import AVFoundation
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import BitLogger
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import Foundation
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/// Streaming PCM -> AAC-LC encoder for live voice. Stateful (the AAC encoder
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/// carries a bit reservoir across frames); one instance per burst.
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/// Not thread-safe — confine to one queue.
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final class PTTFrameEncoder {
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private let converter: AVAudioConverter
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private var pendingInput: [AVAudioPCMBuffer] = []
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init?() {
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guard let pcm = PTTAudioFormat.pcmFormat,
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let aac = PTTAudioFormat.aacFormat,
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let converter = AVAudioConverter(from: pcm, to: aac)
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else { return nil }
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converter.bitRate = PTTAudioFormat.bitRate
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self.converter = converter
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}
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/// Feeds PCM (16 kHz mono float) and returns every complete AAC frame the
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/// encoder produced. Frames come out ~130 bytes each at 16 kbps.
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func encode(_ buffer: AVAudioPCMBuffer) -> [Data] {
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pendingInput.append(buffer)
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return drainConverter()
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}
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private func drainConverter() -> [Data] {
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var frames: [Data] = []
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while true {
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let output = AVAudioCompressedBuffer(
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format: converter.outputFormat,
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packetCapacity: 8,
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maximumPacketSize: max(converter.maximumOutputPacketSize, 1)
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)
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var error: NSError?
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let status = converter.convert(to: output, error: &error) { [weak self] _, outStatus in
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guard let self, let next = self.pendingInput.first else {
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outStatus.pointee = .noDataNow
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return nil
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}
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self.pendingInput.removeFirst()
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outStatus.pointee = .haveData
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return next
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}
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if status == .error {
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SecureLogger.error("PTT encode failed: \(error?.localizedDescription ?? "unknown")", category: .session)
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return frames
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}
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frames.append(contentsOf: Self.extractPackets(from: output))
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// .haveData means the output buffer filled and more may be ready;
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// anything else means the converter wants more input.
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if status != .haveData { return frames }
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}
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}
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private static func extractPackets(from buffer: AVAudioCompressedBuffer) -> [Data] {
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guard buffer.packetCount > 0, let descriptions = buffer.packetDescriptions else { return [] }
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var frames: [Data] = []
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frames.reserveCapacity(Int(buffer.packetCount))
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for index in 0..<Int(buffer.packetCount) {
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let description = descriptions[index]
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guard description.mDataByteSize > 0 else { continue }
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let start = buffer.data.advanced(by: Int(description.mStartOffset))
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frames.append(Data(bytes: start, count: Int(description.mDataByteSize)))
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}
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return frames
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}
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}
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/// Streaming AAC-LC -> PCM decoder for live voice. Stateful; one instance per
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/// inbound burst. Not thread-safe — confine to one queue/actor.
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final class PTTFrameDecoder {
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private let converter: AVAudioConverter
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private let pcmFormat: AVAudioFormat
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private let aacFormat: AVAudioFormat
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init?() {
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guard let pcm = PTTAudioFormat.pcmFormat,
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let aac = PTTAudioFormat.aacFormat,
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let converter = AVAudioConverter(from: aac, to: pcm)
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else { return nil }
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self.converter = converter
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self.pcmFormat = pcm
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self.aacFormat = aac
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}
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/// Decodes one raw AAC frame to PCM. Returns nil for malformed input or
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/// while the decoder is still priming (the first frame of a stream).
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func decode(_ frame: Data) -> AVAudioPCMBuffer? {
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guard !frame.isEmpty, frame.count <= 8 * 1024 else { return nil }
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let input = AVAudioCompressedBuffer(format: aacFormat, packetCapacity: 1, maximumPacketSize: frame.count)
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frame.withUnsafeBytes { raw in
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guard let base = raw.baseAddress else { return }
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input.data.copyMemory(from: base, byteCount: frame.count)
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}
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input.byteLength = UInt32(frame.count)
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input.packetCount = 1
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input.packetDescriptions?.pointee = AudioStreamPacketDescription(
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mStartOffset: 0,
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mVariableFramesInPacket: 0,
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mDataByteSize: UInt32(frame.count)
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)
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guard let output = AVAudioPCMBuffer(
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pcmFormat: pcmFormat,
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frameCapacity: PTTAudioFormat.samplesPerFrame * 2
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) else { return nil }
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var consumed = false
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var error: NSError?
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let status = converter.convert(to: output, error: &error) { _, outStatus in
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if consumed {
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outStatus.pointee = .noDataNow
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return nil
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}
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consumed = true
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outStatus.pointee = .haveData
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return input
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}
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guard status != .error else {
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SecureLogger.debug("PTT decode failed: \(error?.localizedDescription ?? "unknown")", category: .session)
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return nil
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}
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return output.frameLength > 0 ? output : nil
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}
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}
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/// Sample-rate/channel converter from the microphone's native format to the
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/// 16 kHz mono processing format. Stateful; not thread-safe.
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final class PTTInputResampler {
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private let converter: AVAudioConverter
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private let outputFormat: AVAudioFormat
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private let ratio: Double
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init?(inputFormat: AVAudioFormat) {
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guard let pcm = PTTAudioFormat.pcmFormat,
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let converter = AVAudioConverter(from: inputFormat, to: pcm)
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else { return nil }
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self.converter = converter
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self.outputFormat = pcm
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self.ratio = PTTAudioFormat.sampleRate / inputFormat.sampleRate
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}
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func resample(_ buffer: AVAudioPCMBuffer) -> AVAudioPCMBuffer? {
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let capacity = AVAudioFrameCount(Double(buffer.frameLength) * ratio) + 64
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guard let output = AVAudioPCMBuffer(pcmFormat: outputFormat, frameCapacity: capacity) else { return nil }
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var consumed = false
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var error: NSError?
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let status = converter.convert(to: output, error: &error) { _, outStatus in
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if consumed {
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outStatus.pointee = .noDataNow
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return nil
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}
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consumed = true
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outStatus.pointee = .haveData
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return buffer
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}
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guard status != .error else {
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SecureLogger.debug("PTT resample failed: \(error?.localizedDescription ?? "unknown")", category: .session)
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return nil
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}
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return output.frameLength > 0 ? output : nil
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}
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}
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