Phases 2-6: detection, visual timeline, backend hand-off, voiceprints

Phase 2 (call detection): CallDetector using CoreAudio per-process mic
attribution (anarlog technique) — robust start+stop for Zoom/Teams/Signal/Meet,
ignoring our own recording; auto-record toggle. Built; pending live multi-app
confirmation by the user.

Phase 3 (visual timeline foundation): AppAdapter protocol + SpeakerObservation,
TimelineBuilder (hysteresis/overlap/self-merge/aliases), VisualTimeline (schema
1.1), TextRecognizer (Vision OCR), FrameSampler + GridCallAnalyzer (name OCR +
saturated-highlight active-speaker attribution), SignalAdapter, VisualObserver
(window capture; frames released, never saved; minimized->visual_gap, idle != gap).
Synthetic-frame tested; adapter geometry pending real Signal fixtures + live
VisualObserver validation.

Phase 5 (backend hand-off): SparkControlClient (multipart label-merge, sequential,
TLS-skip, 503 Retry-After/413), SessionPackager (chunk plan + WAV slice + timeline
slice/rebase), TranscriptAssembler + SpeakersFile, TranscriptPipeline. Validated
END-TO-END against the live backend (chunk -> label-merge -> speakers.json).

Phase 6 (voiceprints): VoiceprintStore (known_voiceprints, persist named
fingerprints, skip Unknown). Wired: 'Send to backend' button + transcript status,
auto-send toggle (default off) + self-name setting.

All adversarial-review findings fixed. App + XCTest suite build; tests pass.
This commit is contained in:
Grant Gilliam
2026-06-06 00:15:49 -05:00
parent fd7e1a5907
commit 863136aeec
27 changed files with 2108 additions and 22 deletions
@@ -0,0 +1,85 @@
import Foundation
import AVFoundation
/// Splits a long session into backend-sized chunks and produces, per chunk, the
/// sliced audio and the timeline rebased to chunk-local seconds.
///
/// The diarizer caps at 4 speakers/chunk and has request limits, so calls > ~3
/// min are chunked into ~23 min windows; names + voiceprints unify speakers
/// across chunks (handled in the pipeline).
enum SessionPackager {
struct PlannedChunk: Equatable {
let index: Int
let start: Double // global seconds
let end: Double
}
/// One chunk if short; otherwise even ~`chunkSeconds` windows.
static func planChunks(durationSec: Double,
chunkSeconds: Double = 150,
thresholdSec: Double = 180) -> [PlannedChunk] {
guard durationSec > thresholdSec else {
return [PlannedChunk(index: 0, start: 0, end: durationSec)]
}
var chunks: [PlannedChunk] = []
var start = 0.0
var index = 0
while start < durationSec - 0.001 {
let end = min(start + chunkSeconds, durationSec)
chunks.append(PlannedChunk(index: index, start: start, end: end))
start = end
index += 1
}
return chunks
}
/// Clip segments to `[start, end)` and rebase to chunk-local seconds, then
/// emit the flat `label-merge` array `[{start,end,name,confidence}]`.
static func rebasedTimelineData(_ segments: [VisualTimeline.Segment],
start: Double, end: Double) throws -> Data {
let flat: [[String: Any]] = segments.compactMap { seg in
let s = max(seg.start, start)
let e = min(seg.end, end)
guard e > s else { return nil }
return ["start": s - start, "end": e - start, "name": seg.name, "confidence": seg.confidence]
}
return try JSONSerialization.data(withJSONObject: flat, options: [])
}
/// Slice `[startSec, endSec)` of a 16 kHz mono WAV into `dest`.
static func sliceAudio(from source: URL, startSec: Double, endSec: Double, to dest: URL) throws {
let input = try AVAudioFile(forReading: source)
let sr = input.fileFormat.sampleRate
let startFrame = AVAudioFramePosition((startSec * sr).rounded())
let endFrame = min(input.length, AVAudioFramePosition((endSec * sr).rounded()))
guard endFrame > startFrame else { return }
let settings: [String: Any] = [
AVFormatIDKey: kAudioFormatLinearPCM,
AVSampleRateKey: sr,
AVNumberOfChannelsKey: 1,
AVLinearPCMBitDepthKey: 16,
AVLinearPCMIsFloatKey: false,
AVLinearPCMIsBigEndianKey: false,
]
let output = try AVAudioFile(forWriting: dest, settings: settings,
commonFormat: .pcmFormatFloat32, interleaved: false)
input.framePosition = startFrame
var remaining = AVAudioFrameCount(endFrame - startFrame)
let block: AVAudioFrameCount = 16_000
while remaining > 0 {
let n = min(block, remaining)
guard let buffer = AVAudioPCMBuffer(pcmFormat: input.processingFormat, frameCapacity: n) else { break }
try input.read(into: buffer, frameCount: n)
if buffer.frameLength == 0 { break }
try output.write(from: buffer)
remaining -= buffer.frameLength
}
}
/// Duration (seconds) of a WAV.
static func duration(of url: URL) -> Double {
guard let file = try? AVAudioFile(forReading: url), file.fileFormat.sampleRate > 0 else { return 0 }
return Double(file.length) / file.fileFormat.sampleRate
}
}