ci: retrigger build

- Set GOPRIVATE=git.daviestechlabs.io to bypass public Go module proxy
- Configure git URL insteadOf with DISPATCH_TOKEN for private repo access

.dockerignore

@@ -0,0 +1,9 @@
+.git
+.gitignore
+*.md
+LICENSE
+renovate.json
+*_test.go
+e2e_test.go
+__pycache__
+.env*

.gitea/workflows/build-push.yaml

@@ -8,6 +8,7 @@ on:
 
 env:
   NTFY_URL: http://ntfy.observability.svc.cluster.local:80
+  GOPRIVATE: git.daviestechlabs.io
   REGISTRY: gitea-http.gitea.svc.cluster.local:3000/daviestechlabs
   REGISTRY_HOST: gitea-http.gitea.svc.cluster.local:3000
   IMAGE_NAME: stt-module
@@ -20,20 +21,25 @@ jobs:
       - name: Checkout code
         uses: actions/checkout@v4
 
-      - name: Set up uv
+      - name: Set up Go
-        run: curl -LsSf https://astral.sh/uv/install.sh | sh && echo "$HOME/.local/bin" >> $GITHUB_PATH
+        uses: actions/setup-go@v5
+        with:
+          go-version-file: go.mod
+          cache: true
 
-      - name: Set up Python
+      - name: Configure private modules
-        run: uv python install 3.12
+        run: git config --global url."https://gitea-actions:${{ secrets.DISPATCH_TOKEN }}@git.daviestechlabs.io/".insteadOf "https://git.daviestechlabs.io/"
 
-      - name: Install dependencies
+      - name: Run go vet
-        run: uv sync --frozen --extra dev
+        run: go vet ./...
 
-      - name: Run ruff check
+      - name: Install golangci-lint
-        run: uv run ruff check .
+        run: |
+          curl -sSfL https://raw.githubusercontent.com/golangci/golangci-lint/HEAD/install.sh | sh -s -- -b "$(go env GOPATH)/bin"
+          echo "$(go env GOPATH)/bin" >> $GITHUB_PATH
 
-      - name: Run ruff format check
+      - name: Run golangci-lint
-        run: uv run ruff format --check .
+        run: golangci-lint run ./...
 
   test:
     name: Test
@@ -42,17 +48,23 @@ jobs:
       - name: Checkout code
         uses: actions/checkout@v4
 
-      - name: Set up uv
+      - name: Set up Go
-        run: curl -LsSf https://astral.sh/uv/install.sh | sh && echo "$HOME/.local/bin" >> $GITHUB_PATH
+        uses: actions/setup-go@v5
+        with:
+          go-version-file: go.mod
+          cache: true
 
-      - name: Set up Python
+      - name: Configure private modules
-        run: uv python install 3.12
+        run: git config --global url."https://gitea-actions:${{ secrets.DISPATCH_TOKEN }}@git.daviestechlabs.io/".insteadOf "https://git.daviestechlabs.io/"
 
-      - name: Install dependencies
+      - name: Verify dependencies
-        run: uv sync --frozen --extra dev
+        run: go mod verify
+
+      - name: Build
+        run: go build -v ./...
 
       - name: Run tests
-        run: uv run pytest -v
+        run: go test -v -race -coverprofile=coverage.out -covermode=atomic ./...
 
   release:
     name: Release
@@ -133,7 +145,18 @@ jobs:
 
       - name: Login to Gitea Registry
         run: |
-          echo "${{ secrets.REGISTRY_TOKEN }}" | docker login ${{ env.REGISTRY_HOST }} -u ${{ secrets.REGISTRY_USER }} --password-stdin
+          AUTH=$(echo -n "${{ secrets.REGISTRY_USER }}:${{ secrets.REGISTRY_TOKEN }}" | base64 -w0)
+          mkdir -p ~/.docker
+          cat > ~/.docker/config.json << EOF
+          {
+            "auths": {
+              "${{ env.REGISTRY_HOST }}": {
+                "auth": "$AUTH"
+              }
+            }
+          }
+          EOF
+          echo "Auth configured for ${{ env.REGISTRY_HOST }}"
 
       - name: Extract metadata
         id: meta

.gitea/workflows/update-dependency.yml

@@ -0,0 +1,60 @@
+name: Update handler-base
+
+on:
+  repository_dispatch:
+    types: [handler-base-release]
+
+env:
+  NTFY_URL: http://ntfy.observability.svc.cluster.local:80
+  GOPRIVATE: git.daviestechlabs.io
+
+jobs:
+  update:
+    name: Update handler-base dependency
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout code
+        uses: actions/checkout@v4
+        with:
+          token: ${{ secrets.DISPATCH_TOKEN }}
+
+      - name: Set up Go
+        uses: actions/setup-go@v5
+        with:
+          go-version-file: go.mod
+          cache: true
+
+      - name: Configure Git
+        run: |
+          git config user.name "gitea-actions[bot]"
+          git config user.email "actions@git.daviestechlabs.io"
+          git config --global url."https://gitea-actions:${{ secrets.DISPATCH_TOKEN }}@git.daviestechlabs.io/".insteadOf "https://git.daviestechlabs.io/"
+
+      - name: Update handler-base
+        run: |
+          VERSION="${{ gitea.event.client_payload.version }}"
+          echo "Updating handler-base to ${VERSION}"
+          go get git.daviestechlabs.io/daviestechlabs/handler-base@${VERSION}
+          go mod tidy
+
+      - name: Commit and push
+        run: |
+          VERSION="${{ gitea.event.client_payload.version }}"
+          if git diff --quiet go.mod go.sum; then
+            echo "No changes to commit"
+            exit 0
+          fi
+          git add go.mod go.sum
+          git commit -m "chore(deps): bump handler-base to ${VERSION}"
+          git push
+
+      - name: Notify
+        if: success()
+        run: |
+          VERSION="${{ gitea.event.client_payload.version }}"
+          curl -s \
+            -H "Title: 📦 Dep Update: ${{ gitea.repository }}" \
+            -H "Priority: default" \
+            -H "Tags: package,github" \
+            -d "handler-base updated to ${VERSION}" \
+            ${{ env.NTFY_URL }}/gitea-ci

.gitignore

@@ -9,3 +9,4 @@ dist/
 build/
 .env
 .env.local
+stt-module

Dockerfile

@@ -1,14 +1,23 @@
-FROM python:3.12-slim
+# Build stage — Go STT streaming service (HTTP variant)
+FROM golang:1.25-alpine AS builder
 
 WORKDIR /app
 
-# Copy requirements and install dependencies
+RUN apk add --no-cache ca-certificates
-COPY requirements.txt .
-RUN pip install --no-cache-dir -r requirements.txt
 
-# Copy application code
+COPY go.mod go.sum ./
-COPY stt_streaming.py .
+RUN go mod download
-COPY healthcheck.py .
 
-# Run the service
+COPY . .
-CMD ["python", "stt_streaming.py"]
+
+RUN CGO_ENABLED=0 GOOS=linux GOAMD64=v3 go build -ldflags="-w -s" -o /stt-module .
+
+# Runtime stage
+FROM scratch
+
+COPY --from=builder /etc/ssl/certs/ca-certificates.crt /etc/ssl/certs/
+COPY --from=builder /stt-module /stt-module
+
+USER 65534:65534
+
+ENTRYPOINT ["/stt-module"]

e2e_test.go

@@ -0,0 +1,278 @@
+package main
+
+import (
+	"bytes"
+	"encoding/binary"
+	"encoding/json"
+	"math"
+	"mime/multipart"
+	"net/http"
+	"net/http/httptest"
+	"strings"
+	"sync"
+	"testing"
+	"time"
+)
+
+// ────────────────────────────────────────────────────────────────────────────
+// E2E tests: audio buffer lifecycle + transcription pipeline
+// ────────────────────────────────────────────────────────────────────────────
+
+func TestAudioBufferE2E_FullLifecycle(t *testing.T) {
+	// Simulate a full session: start → chunks → process → end
+	ab := newAudioBuffer("e2e-session")
+
+	// Send 10 chunks of 50 KB each
+	for i := 0; i < 10; i++ {
+		ab.addChunk(make([]byte, 50000))
+	}
+
+	ab.mu.Lock()
+	if ab.totalBytes != 500000 {
+		t.Errorf("totalBytes = %d, want 500000", ab.totalBytes)
+	}
+	ab.mu.Unlock()
+
+	// Should process (meets 512KB threshold approximately)
+	if !ab.shouldProcess(512000, 5120000, 2.0) {
+		// Under threshold but check with voice activity
+		ab.hasVoiceActivity = true
+	}
+
+	// Get audio and verify concatenation
+	audio := ab.getAudio()
+	if len(audio) != 500000 {
+		t.Errorf("getAudio() len = %d, want 500000", len(audio))
+	}
+
+	// Clear and verify
+	ab.clear()
+	ab.mu.Lock()
+	seq := ab.sequence
+	total := ab.totalBytes
+	ab.mu.Unlock()
+	if seq != 1 {
+		t.Errorf("sequence = %d, want 1", seq)
+	}
+	if total != 0 {
+		t.Errorf("totalBytes after clear = %d", total)
+	}
+
+	// Mark complete
+	ab.markComplete()
+	ab.mu.Lock()
+	if !ab.isComplete {
+		t.Error("expected isComplete=true")
+	}
+	ab.mu.Unlock()
+}
+
+func TestAudioBufferE2E_InterruptDuringResponse(t *testing.T) {
+	ab := newAudioBuffer("interrupt-session")
+	ab.setState(stateResponding)
+
+	// Simulate loud speech during response (user interrupting)
+	loud := makeLoudAudio(500)
+
+	// First check: starts tracking interrupt timer
+	ab.checkInterrupt(loud, true, 0.01, 0.1)
+
+	// Sleep to exceed duration threshold
+	time.Sleep(150 * time.Millisecond)
+
+	// Second check: should now confirm interrupt
+	interrupted := ab.checkInterrupt(loud, true, 0.01, 0.1)
+	if !interrupted {
+		t.Error("expected interrupt after duration threshold")
+	}
+}
+
+func TestAudioBufferE2E_InterruptDisabled(t *testing.T) {
+	ab := newAudioBuffer("no-interrupt")
+	ab.setState(stateResponding)
+	loud := makeLoudAudio(500)
+
+	if ab.checkInterrupt(loud, false, 0.01, 0.0) {
+		t.Error("interrupt should not trigger when disabled")
+	}
+}
+
+func TestAudioBufferE2E_ConcurrentChunks(t *testing.T) {
+	ab := newAudioBuffer("concurrent")
+	var wg sync.WaitGroup
+	numGoroutines := 20
+	chunkSize := 1000
+
+	wg.Add(numGoroutines)
+	for i := 0; i < numGoroutines; i++ {
+		go func() {
+			defer wg.Done()
+			ab.addChunk(make([]byte, chunkSize))
+		}()
+	}
+	wg.Wait()
+
+	ab.mu.Lock()
+	total := ab.totalBytes
+	chunks := len(ab.chunks)
+	ab.mu.Unlock()
+
+	if total != numGoroutines*chunkSize {
+		t.Errorf("totalBytes = %d, want %d", total, numGoroutines*chunkSize)
+	}
+	if chunks != numGoroutines {
+		t.Errorf("chunks = %d, want %d", chunks, numGoroutines)
+	}
+
+	audio := ab.getAudio()
+	if len(audio) != numGoroutines*chunkSize {
+		t.Errorf("getAudio len = %d, want %d", len(audio), numGoroutines*chunkSize)
+	}
+}
+
+func TestTranscriptionE2E_MockWhisper(t *testing.T) {
+	// Full mock Whisper server that validates multipart form upload
+	whisperSrv := httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		if r.Method != http.MethodPost {
+			t.Errorf("expected POST, got %s", r.Method)
+		}
+		ct := r.Header.Get("Content-Type")
+		if ct == "" {
+			t.Error("missing Content-Type")
+		}
+
+		// Parse multipart form if present
+		if strings.HasPrefix(ct, "multipart/form-data") {
+			if err := r.ParseMultipartForm(32 << 20); err != nil {
+				t.Errorf("multipart parse: %v", err)
+			}
+			file, _, err := r.FormFile("file")
+			if err != nil {
+				t.Errorf("missing 'file' field: %v", err)
+			} else {
+				_ = file.Close()
+			}
+		}
+
+		w.Header().Set("Content-Type", "application/json")
+		_ = json.NewEncoder(w).Encode(map[string]string{"text": "hello world"})
+	}))
+	defer whisperSrv.Close()
+
+	// Build a proper multipart request like stt-module's transcribe() does
+	var buf bytes.Buffer
+	writer := multipart.NewWriter(&buf)
+	part, err := writer.CreateFormFile("file", "audio.wav")
+	if err != nil {
+		t.Fatal(err)
+	}
+	_, _ = part.Write(make([]byte, 8000)) // simulated audio
+	_ = writer.Close()
+
+	resp, err := http.Post(whisperSrv.URL+"/v1/audio/transcriptions", writer.FormDataContentType(), &buf)
+	if err != nil {
+		t.Fatal(err)
+	}
+	defer func() { _ = resp.Body.Close() }()
+	if resp.StatusCode != 200 {
+		t.Errorf("status = %d", resp.StatusCode)
+	}
+
+	var result map[string]string
+	_ = json.NewDecoder(resp.Body).Decode(&result)
+	if result["text"] != "hello world" {
+		t.Errorf("text = %q, want %q", result["text"], "hello world")
+	}
+}
+
+func TestAudioRMSE2E_RealisticSignal(t *testing.T) {
+	// Generate a sine wave at 440 Hz, 16kHz sample rate
+	sampleRate := 16000
+	duration := 0.1 // 100ms
+	numSamples := int(float64(sampleRate) * duration)
+	audio := make([]byte, numSamples*2)
+
+	amplitude := 16000.0
+	for i := 0; i < numSamples; i++ {
+		sample := int16(amplitude * math.Sin(2*math.Pi*440*float64(i)/float64(sampleRate)))
+		binary.LittleEndian.PutUint16(audio[i*2:], uint16(sample))
+	}
+
+	rms := calculateAudioRMS(audio)
+	// RMS of a sine wave = amplitude / sqrt(2) / 32768
+	expectedRMS := amplitude / math.Sqrt(2) / 32768.0
+	tolerance := 0.01
+	if math.Abs(rms-expectedRMS) > tolerance {
+		t.Errorf("RMS = %.4f, expected ~%.4f (±%.2f)", rms, expectedRMS, tolerance)
+	}
+
+	if !detectVoiceActivity(audio) {
+		t.Error("440 Hz sine at amplitude 16000 should be detected as voice")
+	}
+}
+
+// ────────────────────────────────────────────────────────────────────────────
+// Benchmarks
+// ────────────────────────────────────────────────────────────────────────────
+
+func BenchmarkAudioBufferAddChunk(b *testing.B) {
+	ab := newAudioBuffer("bench")
+	chunk := make([]byte, 4096)
+
+	b.ResetTimer()
+	for b.Loop() {
+		ab.addChunk(chunk)
+	}
+}
+
+func BenchmarkAudioBufferGetAudio(b *testing.B) {
+	ab := newAudioBuffer("bench")
+	for i := 0; i < 100; i++ {
+		ab.addChunk(make([]byte, 4096))
+	}
+
+	b.ResetTimer()
+	for b.Loop() {
+		_ = ab.getAudio()
+	}
+}
+
+func BenchmarkCalculateAudioRMS(b *testing.B) {
+	audio := make([]byte, 32000) // 1 second at 16kHz, 16-bit
+	for i := 0; i < 16000; i++ {
+		val := int16(16000 * math.Sin(float64(i)*0.1))
+		binary.LittleEndian.PutUint16(audio[i*2:], uint16(val))
+	}
+
+	b.ResetTimer()
+	for b.Loop() {
+		calculateAudioRMS(audio)
+	}
+}
+
+func BenchmarkDetectVoiceActivity(b *testing.B) {
+	audio := make([]byte, 8000) // 250ms at 16kHz
+	for i := 0; i < 4000; i++ {
+		val := int16(10000 * math.Sin(float64(i)*0.2))
+		binary.LittleEndian.PutUint16(audio[i*2:], uint16(val))
+	}
+
+	b.ResetTimer()
+	for b.Loop() {
+		detectVoiceActivity(audio)
+	}
+}
+
+func BenchmarkCheckInterrupt(b *testing.B) {
+	ab := newAudioBuffer("bench")
+	ab.setState(stateResponding)
+	audio := makeLoudAudio(500)
+
+	b.ResetTimer()
+	for b.Loop() {
+		ab.checkInterrupt(audio, true, 0.02, 999.0) // high duration so it never triggers
+		ab.mu.Lock()
+		ab.interruptStartTime = nil // reset for clean iteration
+		ab.mu.Unlock()
+	}
+}

go.mod

@@ -0,0 +1,41 @@
+module git.daviestechlabs.io/daviestechlabs/stt-module
+
+go 1.25.1
+
+require (
+	git.daviestechlabs.io/daviestechlabs/handler-base v0.1.3
+	github.com/nats-io/nats.go v1.48.0
+	github.com/vmihailenco/msgpack/v5 v5.4.1
+)
+
+require (
+	github.com/cenkalti/backoff/v5 v5.0.3 // indirect
+	github.com/cespare/xxhash/v2 v2.3.0 // indirect
+	github.com/fsnotify/fsnotify v1.9.0 // indirect
+	github.com/go-logr/logr v1.4.3 // indirect
+	github.com/go-logr/stdr v1.2.2 // indirect
+	github.com/google/uuid v1.6.0 // indirect
+	github.com/grpc-ecosystem/grpc-gateway/v2 v2.27.7 // indirect
+	github.com/klauspost/compress v1.18.0 // indirect
+	github.com/nats-io/nkeys v0.4.11 // indirect
+	github.com/nats-io/nuid v1.0.1 // indirect
+	github.com/vmihailenco/tagparser/v2 v2.0.0 // indirect
+	go.opentelemetry.io/auto/sdk v1.2.1 // indirect
+	go.opentelemetry.io/otel v1.40.0 // indirect
+	go.opentelemetry.io/otel/exporters/otlp/otlpmetric/otlpmetricgrpc v1.40.0 // indirect
+	go.opentelemetry.io/otel/exporters/otlp/otlptrace v1.40.0 // indirect
+	go.opentelemetry.io/otel/exporters/otlp/otlptrace/otlptracegrpc v1.40.0 // indirect
+	go.opentelemetry.io/otel/metric v1.40.0 // indirect
+	go.opentelemetry.io/otel/sdk v1.40.0 // indirect
+	go.opentelemetry.io/otel/sdk/metric v1.40.0 // indirect
+	go.opentelemetry.io/otel/trace v1.40.0 // indirect
+	go.opentelemetry.io/proto/otlp v1.9.0 // indirect
+	golang.org/x/crypto v0.47.0 // indirect
+	golang.org/x/net v0.49.0 // indirect
+	golang.org/x/sys v0.40.0 // indirect
+	golang.org/x/text v0.33.0 // indirect
+	google.golang.org/genproto/googleapis/api v0.0.0-20260128011058-8636f8732409 // indirect
+	google.golang.org/genproto/googleapis/rpc v0.0.0-20260128011058-8636f8732409 // indirect
+	google.golang.org/grpc v1.78.0 // indirect
+	google.golang.org/protobuf v1.36.11 // indirect
+)

go.sum

@@ -0,0 +1,81 @@
+git.daviestechlabs.io/daviestechlabs/handler-base v0.1.3 h1:uYog8B839ulqrWoht3qqCvT7CnR3e2skpaLZc2Pg3GI=
+git.daviestechlabs.io/daviestechlabs/handler-base v0.1.3/go.mod h1:M3HgvUDWnRn7cX3BE8l+HvoCUYtmRr5OoumB+hnRHoE=
+github.com/cenkalti/backoff/v5 v5.0.3 h1:ZN+IMa753KfX5hd8vVaMixjnqRZ3y8CuJKRKj1xcsSM=
+github.com/cenkalti/backoff/v5 v5.0.3/go.mod h1:rkhZdG3JZukswDf7f0cwqPNk4K0sa+F97BxZthm/crw=
+github.com/cespare/xxhash/v2 v2.3.0 h1:UL815xU9SqsFlibzuggzjXhog7bL6oX9BbNZnL2UFvs=
+github.com/cespare/xxhash/v2 v2.3.0/go.mod h1:VGX0DQ3Q6kWi7AoAeZDth3/j3BFtOZR5XLFGgcrjCOs=
+github.com/davecgh/go-spew v1.1.1 h1:vj9j/u1bqnvCEfJOwUhtlOARqs3+rkHYY13jYWTU97c=
+github.com/davecgh/go-spew v1.1.1/go.mod h1:J7Y8YcW2NihsgmVo/mv3lAwl/skON4iLHjSsI+c5H38=
+github.com/fsnotify/fsnotify v1.9.0 h1:2Ml+OJNzbYCTzsxtv8vKSFD9PbJjmhYF14k/jKC7S9k=
+github.com/fsnotify/fsnotify v1.9.0/go.mod h1:8jBTzvmWwFyi3Pb8djgCCO5IBqzKJ/Jwo8TRcHyHii0=
+github.com/go-logr/logr v1.2.2/go.mod h1:jdQByPbusPIv2/zmleS9BjJVeZ6kBagPoEUsqbVz/1A=
+github.com/go-logr/logr v1.4.3 h1:CjnDlHq8ikf6E492q6eKboGOC0T8CDaOvkHCIg8idEI=
+github.com/go-logr/logr v1.4.3/go.mod h1:9T104GzyrTigFIr8wt5mBrctHMim0Nb2HLGrmQ40KvY=
+github.com/go-logr/stdr v1.2.2 h1:hSWxHoqTgW2S2qGc0LTAI563KZ5YKYRhT3MFKZMbjag=
+github.com/go-logr/stdr v1.2.2/go.mod h1:mMo/vtBO5dYbehREoey6XUKy/eSumjCCveDpRre4VKE=
+github.com/golang/protobuf v1.5.4 h1:i7eJL8qZTpSEXOPTxNKhASYpMn+8e5Q6AdndVa1dWek=
+github.com/golang/protobuf v1.5.4/go.mod h1:lnTiLA8Wa4RWRcIUkrtSVa5nRhsEGBg48fD6rSs7xps=
+github.com/google/go-cmp v0.7.0 h1:wk8382ETsv4JYUZwIsn6YpYiWiBsYLSJiTsyBybVuN8=
+github.com/google/go-cmp v0.7.0/go.mod h1:pXiqmnSA92OHEEa9HXL2W4E7lf9JzCmGVUdgjX3N/iU=
+github.com/google/uuid v1.6.0 h1:NIvaJDMOsjHA8n1jAhLSgzrAzy1Hgr+hNrb57e+94F0=
+github.com/google/uuid v1.6.0/go.mod h1:TIyPZe4MgqvfeYDBFedMoGGpEw/LqOeaOT+nhxU+yHo=
+github.com/grpc-ecosystem/grpc-gateway/v2 v2.27.7 h1:X+2YciYSxvMQK0UZ7sg45ZVabVZBeBuvMkmuI2V3Fak=
+github.com/grpc-ecosystem/grpc-gateway/v2 v2.27.7/go.mod h1:lW34nIZuQ8UDPdkon5fmfp2l3+ZkQ2me/+oecHYLOII=
+github.com/klauspost/compress v1.18.0 h1:c/Cqfb0r+Yi+JtIEq73FWXVkRonBlf0CRNYc8Zttxdo=
+github.com/klauspost/compress v1.18.0/go.mod h1:2Pp+KzxcywXVXMr50+X0Q/Lsb43OQHYWRCY2AiWywWQ=
+github.com/nats-io/nats.go v1.48.0 h1:pSFyXApG+yWU/TgbKCjmm5K4wrHu86231/w84qRVR+U=
+github.com/nats-io/nats.go v1.48.0/go.mod h1:iRWIPokVIFbVijxuMQq4y9ttaBTMe0SFdlZfMDd+33g=
+github.com/nats-io/nkeys v0.4.11 h1:q44qGV008kYd9W1b1nEBkNzvnWxtRSQ7A8BoqRrcfa0=
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healthcheck.py

@@ -1,29 +0,0 @@
-#!/usr/bin/env python3
-"""
-Health check script for Kubernetes probes
-Verifies NATS connectivity
-"""
-
-import asyncio
-import os
-import sys
-
-import nats
-
-NATS_URL = os.environ.get("NATS_URL", "nats://nats.ai-ml.svc.cluster.local:4222")
-
-
-async def check_health():
-    """Check if service can connect to NATS."""
-    try:
-        nc = await asyncio.wait_for(nats.connect(NATS_URL), timeout=5.0)
-        await nc.close()
-        return True
-    except Exception as e:
-        print(f"Health check failed: {e}", file=sys.stderr)
-        return False
-
-
-if __name__ == "__main__":
-    result = asyncio.run(check_health())
-    sys.exit(0 if result else 1)

main.go

@@ -0,0 +1,510 @@
+package main
+
+import (
+	"bytes"
+	"context"
+	"encoding/binary"
+	"encoding/json"
+	"fmt"
+	"io"
+	"log/slog"
+	"math"
+	"mime/multipart"
+	"net/http"
+	"os"
+	"os/signal"
+	"strconv"
+	"strings"
+	"sync"
+	"syscall"
+	"time"
+
+	"github.com/nats-io/nats.go"
+	"github.com/vmihailenco/msgpack/v5"
+
+	"git.daviestechlabs.io/daviestechlabs/handler-base/config"
+	"git.daviestechlabs.io/daviestechlabs/handler-base/health"
+	"git.daviestechlabs.io/daviestechlabs/handler-base/messages"
+	"git.daviestechlabs.io/daviestechlabs/handler-base/natsutil"
+	"git.daviestechlabs.io/daviestechlabs/handler-base/telemetry"
+)
+
+// NATS subjects
+const (
+	streamSubjectPrefix        = "ai.voice.stream"
+	transcriptionSubjectPrefix = "ai.voice.transcription"
+)
+
+// Session states
+const (
+	stateListening  = "listening"
+	stateResponding = "responding"
+)
+
+// AudioBuffer manages audio chunks for a streaming session.
+type AudioBuffer struct {
+	mu                 sync.Mutex
+	sessionID          string
+	chunks             [][]byte
+	totalBytes         int
+	lastChunkTime      time.Time
+	isComplete         bool
+	sequence           int
+	state              string
+	speakerID          string
+	interruptStartTime *time.Time
+	hasVoiceActivity   bool
+}
+
+func newAudioBuffer(sessionID string) *AudioBuffer {
+	return &AudioBuffer{
+		sessionID:     sessionID,
+		lastChunkTime: time.Now(),
+		state:         stateListening,
+	}
+}
+
+func (ab *AudioBuffer) addChunk(audio []byte) {
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+	ab.chunks = append(ab.chunks, audio)
+	ab.totalBytes += len(audio)
+	ab.lastChunkTime = time.Now()
+	ab.hasVoiceActivity = detectVoiceActivity(audio)
+}
+
+func (ab *AudioBuffer) checkInterrupt(audio []byte, enableInterrupt bool, audioThreshold, interruptDuration float64) bool {
+	if !enableInterrupt {
+		return false
+	}
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+
+	if ab.state != stateResponding {
+		return false
+	}
+
+	rms := calculateAudioRMS(audio)
+	hasVoice := detectVoiceActivity(audio)
+
+	if rms >= audioThreshold && hasVoice {
+		if ab.interruptStartTime == nil {
+			now := time.Now()
+			ab.interruptStartTime = &now
+			slog.Info("potential interrupt detected", "session", ab.sessionID, "rms", rms)
+		}
+		elapsed := time.Since(*ab.interruptStartTime).Seconds()
+		if elapsed >= interruptDuration {
+			slog.Info("interrupt confirmed", "session", ab.sessionID, "elapsed", elapsed)
+			return true
+		}
+	} else {
+		ab.interruptStartTime = nil
+	}
+	return false
+}
+
+func (ab *AudioBuffer) setState(state string) {
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+	if state != stateListening && state != stateResponding {
+		return
+	}
+	if ab.state != state {
+		slog.Info("state changed", "session", ab.sessionID, "from", ab.state, "to", state)
+		ab.state = state
+		ab.interruptStartTime = nil
+	}
+}
+
+func (ab *AudioBuffer) shouldProcess(bufferSize, maxBufferSize int, chunkTimeout float64) bool {
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+
+	if !ab.hasVoiceActivity && ab.totalBytes < bufferSize &&
+		time.Since(ab.lastChunkTime).Seconds() < chunkTimeout {
+		return false
+	}
+	if ab.totalBytes >= bufferSize {
+		return true
+	}
+	if time.Since(ab.lastChunkTime).Seconds() > chunkTimeout && ab.totalBytes > 0 {
+		return true
+	}
+	return ab.totalBytes >= maxBufferSize
+}
+
+func (ab *AudioBuffer) getAudio() []byte {
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+	result := make([]byte, 0, ab.totalBytes)
+	for _, c := range ab.chunks {
+		result = append(result, c...)
+	}
+	return result
+}
+
+func (ab *AudioBuffer) clear() {
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+	ab.chunks = nil
+	ab.totalBytes = 0
+	ab.sequence++
+}
+
+func (ab *AudioBuffer) markComplete() {
+	ab.mu.Lock()
+	defer ab.mu.Unlock()
+	ab.isComplete = true
+}
+
+// calculateAudioRMS computes RMS level for 16-bit PCM audio, normalized to 0.0-1.0.
+func calculateAudioRMS(audio []byte) float64 {
+	if len(audio) < 2 {
+		return 0.0
+	}
+	numSamples := len(audio) / 2
+	var sumSquares float64
+	for i := 0; i < numSamples; i++ {
+		sample := int16(binary.LittleEndian.Uint16(audio[i*2 : i*2+2]))
+		sumSquares += float64(sample) * float64(sample)
+	}
+	rms := math.Sqrt(sumSquares / float64(numSamples))
+	return rms / 32768.0
+}
+
+// detectVoiceActivity uses RMS-based voice detection (pure Go, no cgo).
+func detectVoiceActivity(audio []byte) bool {
+	rms := calculateAudioRMS(audio)
+	return rms > 0.01 // reasonable threshold for speech
+}
+
+func main() {
+	slog.SetDefault(slog.New(slog.NewJSONHandler(os.Stdout, &slog.HandlerOptions{Level: slog.LevelInfo})))
+
+	cfg := config.Load()
+	cfg.ServiceName = "stt-streaming"
+
+	whisperURL := getEnv("WHISPER_URL", "http://whisper-predictor.ai-ml.svc.cluster.local")
+	bufferSizeBytes := getEnvInt("STT_BUFFER_SIZE_BYTES", 512000)
+	chunkTimeout := getEnvFloat("STT_CHUNK_TIMEOUT", 2.0)
+	maxBufferSize := getEnvInt("STT_MAX_BUFFER_SIZE", 5120000)
+	enableInterrupt := getEnvBool("STT_ENABLE_INTERRUPT_DETECTION", true)
+	audioLevelThreshold := getEnvFloat("STT_AUDIO_LEVEL_THRESHOLD", 0.02)
+	interruptDuration := getEnvFloat("STT_INTERRUPT_DURATION", 0.5)
+
+	ctx, cancel := context.WithCancel(context.Background())
+	defer cancel()
+
+	// Telemetry
+	tp, shutdown, err := telemetry.Setup(ctx, telemetry.Config{
+		ServiceName:      cfg.ServiceName,
+		ServiceVersion:   cfg.ServiceVersion,
+		ServiceNamespace: cfg.ServiceNamespace,
+		DeploymentEnv:    cfg.DeploymentEnv,
+		Enabled:          cfg.OTELEnabled,
+		Endpoint:         cfg.OTELEndpoint,
+	})
+	if err != nil {
+		slog.Error("telemetry setup failed", "error", err)
+	}
+	if shutdown != nil {
+		defer shutdown(ctx)
+	}
+	_ = tp
+
+	// NATS
+	natsOpts := []nats.Option{}
+	if cfg.NATSUser != "" && cfg.NATSPassword != "" {
+		natsOpts = append(natsOpts, nats.UserInfo(cfg.NATSUser, cfg.NATSPassword))
+	}
+	nc := natsutil.New(cfg.NATSURL, natsOpts...)
+	if err := nc.Connect(); err != nil {
+		slog.Error("NATS connect failed", "error", err)
+		os.Exit(1)
+	}
+	defer nc.Close()
+
+	// JetStream stream setup
+	js, err := nc.Conn().JetStream()
+	if err == nil {
+		_, err = js.AddStream(&nats.StreamConfig{
+			Name:      "AI_VOICE_STREAM",
+			Subjects:  []string{"ai.voice.stream.>", "ai.voice.transcription.>"},
+			Retention: nats.LimitsPolicy,
+			MaxAge:    5 * time.Minute,
+			Storage:   nats.MemoryStorage,
+		})
+		if err != nil {
+			slog.Info("JetStream stream setup", "msg", err)
+		} else {
+			slog.Info("created/updated JetStream stream AI_VOICE_STREAM")
+		}
+	}
+
+	httpClient := &http.Client{Timeout: 180 * time.Second}
+	running := true
+
+	// Session management
+	var sessionsMu sync.RWMutex
+	sessions := make(map[string]*AudioBuffer)
+
+	// Health server
+	healthSrv := health.New(cfg.HealthPort, cfg.HealthPath, cfg.ReadyPath, func() bool {
+		return running && nc.IsConnected()
+	})
+	healthSrv.Start()
+	defer healthSrv.Stop(ctx)
+
+	// Transcribe via Whisper HTTP
+	transcribe := func(audioData []byte) (string, error) {
+		var buf bytes.Buffer
+		w := multipart.NewWriter(&buf)
+		part, err := w.CreateFormFile("file", "audio.wav")
+		if err != nil {
+			return "", err
+		}
+		_, _ = part.Write(audioData)
+		_ = w.Close()
+
+		req, err := http.NewRequestWithContext(ctx, http.MethodPost, whisperURL+"/v1/audio/transcriptions", &buf)
+		if err != nil {
+			return "", err
+		}
+		req.Header.Set("Content-Type", w.FormDataContentType())
+
+		resp, err := httpClient.Do(req)
+		if err != nil {
+			return "", fmt.Errorf("whisper request: %w", err)
+		}
+		defer func() { _ = resp.Body.Close() }()
+		body, _ := io.ReadAll(resp.Body)
+
+		if resp.StatusCode >= 400 {
+			return "", fmt.Errorf("whisper %d: %s", resp.StatusCode, string(body))
+		}
+
+		var result struct {
+			Text string `json:"text"`
+		}
+		if err := json.Unmarshal(body, &result); err != nil {
+			return "", err
+		}
+		return result.Text, nil
+	}
+
+	// Process buffer for a session
+	processBuffer := func(sessionID string) {
+		sessionsMu.RLock()
+		buffer, ok := sessions[sessionID]
+		sessionsMu.RUnlock()
+		if !ok {
+			return
+		}
+
+		audioData := buffer.getAudio()
+		if len(audioData) == 0 {
+			return
+		}
+
+		buffer.mu.Lock()
+		seq := buffer.sequence
+		complete := buffer.isComplete
+		speakerID := buffer.speakerID
+		hasVoice := buffer.hasVoiceActivity
+		state := buffer.state
+		buffer.mu.Unlock()
+
+		slog.Info("processing buffer", "session", sessionID, "bytes", len(audioData), "seq", seq)
+
+		transcript, err := transcribe(audioData)
+		if err != nil {
+			slog.Error("transcription failed", "session", sessionID, "error", err)
+		}
+
+		if transcript != "" {
+			result := &messages.STTTranscription{
+				SessionID:        sessionID,
+				Transcript:       transcript,
+				Sequence:         seq,
+				IsPartial:        !complete,
+				IsFinal:          complete,
+				Timestamp:        time.Now().Unix(),
+				SpeakerID:        speakerID,
+				HasVoiceActivity: hasVoice,
+				State:            state,
+			}
+			packed, _ := msgpack.Marshal(result)
+			_ = nc.Conn().Publish(fmt.Sprintf("%s.%s", transcriptionSubjectPrefix, sessionID), packed)
+			slog.Info("published transcription", "session", sessionID, "seq", seq)
+		}
+
+		buffer.clear()
+
+		if complete {
+			slog.Info("session completed", "session", sessionID)
+			go func() {
+				time.Sleep(5 * time.Second)
+				sessionsMu.Lock()
+				delete(sessions, sessionID)
+				sessionsMu.Unlock()
+				slog.Info("cleaned up session", "session", sessionID)
+			}()
+		}
+	}
+
+	// Monitor goroutine for a session
+	monitorBuffer := func(sessionID string) {
+		for running {
+			sessionsMu.RLock()
+			buffer, ok := sessions[sessionID]
+			sessionsMu.RUnlock()
+			if !ok {
+				return
+			}
+			if buffer.shouldProcess(bufferSizeBytes, maxBufferSize, chunkTimeout) {
+				processBuffer(sessionID)
+			}
+			time.Sleep(100 * time.Millisecond)
+		}
+	}
+
+	// Handle incoming stream messages
+	handleStreamMsg := func(natMsg *nats.Msg) {
+		parts := strings.Split(natMsg.Subject, ".")
+		if len(parts) < 4 {
+			slog.Warn("invalid subject", "subject", natMsg.Subject)
+			return
+		}
+		sessionID := parts[3]
+
+		streamMsg, err := natsutil.Decode[messages.STTStreamMessage](natMsg.Data)
+		if err != nil {
+			slog.Error("decode error", "error", err)
+			return
+		}
+
+		switch streamMsg.Type {
+		case "start":
+			slog.Info("starting stream session", "session", sessionID)
+			buf := newAudioBuffer(sessionID)
+			if streamMsg.State != "" {
+				buf.setState(streamMsg.State)
+			}
+			if streamMsg.SpeakerID != "" {
+				buf.speakerID = streamMsg.SpeakerID
+			}
+			sessionsMu.Lock()
+			sessions[sessionID] = buf
+			sessionsMu.Unlock()
+			go monitorBuffer(sessionID)
+
+		case "state_change":
+			sessionsMu.RLock()
+			buffer, ok := sessions[sessionID]
+			sessionsMu.RUnlock()
+			if ok && streamMsg.State != "" {
+				buffer.setState(streamMsg.State)
+			}
+
+		case "end":
+			slog.Info("ending stream session", "session", sessionID)
+			sessionsMu.RLock()
+			buffer, ok := sessions[sessionID]
+			sessionsMu.RUnlock()
+			if ok {
+				buffer.markComplete()
+				buffer.mu.Lock()
+				hasData := buffer.totalBytes > 0
+				buffer.mu.Unlock()
+				if hasData {
+					processBuffer(sessionID)
+				}
+			}
+
+		case "chunk":
+			// Audio arrives as raw bytes — no base64 decode needed
+			if len(streamMsg.Audio) == 0 {
+				return
+			}
+
+			// Auto-create session if missing
+			sessionsMu.Lock()
+			buffer, ok := sessions[sessionID]
+			if !ok {
+				slog.Info("auto-creating session", "session", sessionID)
+				buffer = newAudioBuffer(sessionID)
+				sessions[sessionID] = buffer
+				go monitorBuffer(sessionID)
+			}
+			sessionsMu.Unlock()
+
+			// Check for interrupt
+			if buffer.checkInterrupt(streamMsg.Audio, enableInterrupt, audioLevelThreshold, interruptDuration) {
+				interruptMsg := &messages.STTInterrupt{
+					SessionID: sessionID,
+					Type:      "interrupt",
+					Timestamp: time.Now().Unix(),
+					SpeakerID: buffer.speakerID,
+				}
+				packed, _ := msgpack.Marshal(interruptMsg)
+				_ = nc.Conn().Publish(fmt.Sprintf("%s.%s", transcriptionSubjectPrefix, sessionID), packed)
+				slog.Info("published interrupt", "session", sessionID)
+				buffer.setState(stateListening)
+			}
+
+			buffer.addChunk(streamMsg.Audio)
+		}
+	}
+
+	if _, err := nc.Conn().Subscribe(streamSubjectPrefix+".>", handleStreamMsg); err != nil {
+		slog.Error("subscribe failed", "error", err)
+		os.Exit(1)
+	}
+	slog.Info("subscribed", "subject", streamSubjectPrefix+".>")
+	slog.Info("stt-streaming ready")
+
+	// Wait for shutdown
+	sigCh := make(chan os.Signal, 1)
+	signal.Notify(sigCh, syscall.SIGTERM, syscall.SIGINT)
+	<-sigCh
+
+	slog.Info("shutting down")
+	running = false
+	cancel()
+	slog.Info("shutdown complete")
+}
+
+// Helpers
+
+func getEnv(key, fallback string) string {
+	if v := os.Getenv(key); v != "" {
+		return v
+	}
+	return fallback
+}
+
+func getEnvInt(key string, fallback int) int {
+	if v := os.Getenv(key); v != "" {
+		if i, err := strconv.Atoi(v); err == nil {
+			return i
+		}
+	}
+	return fallback
+}
+
+func getEnvFloat(key string, fallback float64) float64 {
+	if v := os.Getenv(key); v != "" {
+		if f, err := strconv.ParseFloat(v, 64); err == nil {
+			return f
+		}
+	}
+	return fallback
+}
+
+func getEnvBool(key string, fallback bool) bool {
+	if v := os.Getenv(key); v != "" {
+		return strings.EqualFold(v, "true") || v == "1"
+	}
+	return fallback
+}

main_test.go

@@ -0,0 +1,200 @@
+package main
+
+import (
+	"encoding/binary"
+	"encoding/json"
+	"math"
+	"net/http"
+	"net/http/httptest"
+	"testing"
+	"time"
+)
+
+func TestCalculateAudioRMS(t *testing.T) {
+	// Silence: all zeros
+	silence := make([]byte, 200)
+	rms := calculateAudioRMS(silence)
+	if rms != 0.0 {
+		t.Errorf("silence RMS = %f, want 0.0", rms)
+	}
+
+	// Max amplitude: 16-bit samples at max value
+	loud := make([]byte, 200)
+	for i := 0; i < 100; i++ {
+		binary.LittleEndian.PutUint16(loud[i*2:], uint16(32767))
+	}
+	rms = calculateAudioRMS(loud)
+	if rms < 0.99 {
+		t.Errorf("max amplitude RMS = %f, want ~1.0", rms)
+	}
+
+	// Small input
+	if rms := calculateAudioRMS([]byte{0}); rms != 0.0 {
+		t.Errorf("single byte RMS = %f", rms)
+	}
+}
+
+func TestDetectVoiceActivity(t *testing.T) {
+	// Silence should not be detected as voice
+	silence := make([]byte, 200)
+	if detectVoiceActivity(silence) {
+		t.Error("silence detected as voice")
+	}
+
+	// Loud audio should be detected as voice
+	loud := make([]byte, 200)
+	for i := 0; i < 100; i++ {
+		val := int16(16000 * math.Sin(float64(i)*0.1))
+		binary.LittleEndian.PutUint16(loud[i*2:], uint16(val))
+	}
+	if !detectVoiceActivity(loud) {
+		t.Error("loud audio not detected as voice")
+	}
+}
+
+func TestAudioBufferBasic(t *testing.T) {
+	ab := newAudioBuffer("test-session")
+	if ab.sessionID != "test-session" {
+		t.Error("wrong session ID")
+	}
+	if ab.state != stateListening {
+		t.Error("initial state should be listening")
+	}
+
+	// Add chunk
+	chunk := make([]byte, 1000)
+	ab.addChunk(chunk)
+	if ab.totalBytes != 1000 {
+		t.Errorf("totalBytes = %d, want 1000", ab.totalBytes)
+	}
+
+	// Get audio
+	audio := ab.getAudio()
+	if len(audio) != 1000 {
+		t.Errorf("getAudio len = %d, want 1000", len(audio))
+	}
+
+	// Clear
+	ab.clear()
+	if ab.totalBytes != 0 {
+		t.Error("totalBytes should be 0 after clear")
+	}
+	if ab.sequence != 1 {
+		t.Errorf("sequence = %d, want 1", ab.sequence)
+	}
+}
+
+func TestAudioBufferStateChange(t *testing.T) {
+	ab := newAudioBuffer("s1")
+	ab.setState(stateResponding)
+	ab.mu.Lock()
+	if ab.state != stateResponding {
+		t.Error("state should be responding")
+	}
+	ab.mu.Unlock()
+
+	ab.setState("invalid")
+	ab.mu.Lock()
+	if ab.state != stateResponding {
+		t.Error("state should still be responding")
+	}
+	ab.mu.Unlock()
+}
+
+func TestAudioBufferShouldProcess(t *testing.T) {
+	ab := newAudioBuffer("s2")
+	// Empty buffer, recent time — should not process
+	if ab.shouldProcess(512000, 5120000, 2.0) {
+		t.Error("empty buffer should not process")
+	}
+
+	// Add enough data to meet threshold
+	ab.addChunk(make([]byte, 512000))
+	ab.hasVoiceActivity = true
+	if !ab.shouldProcess(512000, 5120000, 2.0) {
+		t.Error("full buffer should process")
+	}
+}
+
+func TestAudioBufferTimeout(t *testing.T) {
+	ab := newAudioBuffer("s3")
+	ab.addChunk(make([]byte, 100))
+	ab.hasVoiceActivity = true
+	// Simulate old lastChunkTime
+	ab.mu.Lock()
+	ab.lastChunkTime = time.Now().Add(-3 * time.Second)
+	ab.mu.Unlock()
+
+	if !ab.shouldProcess(512000, 5120000, 2.0) {
+		t.Error("timed-out buffer should process")
+	}
+}
+
+func TestAudioBufferCheckInterrupt(t *testing.T) {
+	ab := newAudioBuffer("s4")
+	// Not in responding state — no interrupt
+	loud := makeLoudAudio(100)
+	if ab.checkInterrupt(loud, true, 0.001, 0.0) {
+		t.Error("should not interrupt in listening state")
+	}
+
+	// Switch to responding
+	ab.setState(stateResponding)
+	// With 0 duration threshold, immediate interrupt
+	if !ab.checkInterrupt(loud, true, 0.001, 0.0) {
+		t.Error("should interrupt in responding state with loud audio")
+	}
+}
+
+func TestTranscribeHTTP(t *testing.T) {
+	ts := httptest.NewServer(http.HandlerFunc(func(w http.ResponseWriter, r *http.Request) {
+		if r.URL.Path != "/v1/audio/transcriptions" {
+			t.Errorf("unexpected path: %s", r.URL.Path)
+		}
+		if r.Method != http.MethodPost {
+			t.Errorf("expected POST, got %s", r.Method)
+		}
+		w.Header().Set("Content-Type", "application/json")
+		_ = json.NewEncoder(w).Encode(map[string]string{"text": "hello world"})
+	}))
+	defer ts.Close()
+
+	// Verify the mock responds correctly
+	resp, err := http.Post(ts.URL+"/v1/audio/transcriptions", "audio/wav", nil)
+	if err != nil {
+		t.Fatal(err)
+	}
+	defer func() { _ = resp.Body.Close() }()
+	if resp.StatusCode != 200 {
+		t.Errorf("status = %d", resp.StatusCode)
+	}
+}
+
+func TestHelpers(t *testing.T) {
+	t.Setenv("STT_TEST", "val")
+	if got := getEnv("STT_TEST", "x"); got != "val" {
+		t.Errorf("getEnv = %q", got)
+	}
+	t.Setenv("STT_PORT", "9090")
+	if got := getEnvInt("STT_PORT", 0); got != 9090 {
+		t.Errorf("getEnvInt = %d", got)
+	}
+	t.Setenv("STT_TIMEOUT", "1.5")
+	if got := getEnvFloat("STT_TIMEOUT", 0); got != 1.5 {
+		t.Errorf("getEnvFloat = %f", got)
+	}
+	t.Setenv("STT_FLAG", "true")
+	if got := getEnvBool("STT_FLAG", false); !got {
+		t.Error("getEnvBool should be true")
+	}
+}
+
+// makeLoudAudio creates a 16-bit PCM audio buffer with high amplitude.
+func makeLoudAudio(numSamples int) []byte {
+	buf := make([]byte, numSamples*2)
+	for i := 0; i < numSamples; i++ {
+		val := int16(20000 * math.Sin(float64(i)*0.3))
+		binary.LittleEndian.PutUint16(buf[i*2:], uint16(val))
+	}
+	return buf
+}

pyproject.toml

@@ -1,58 +0,0 @@
-[project]
-name = "stt-module"
-version = "1.0.0"
-description = "Streaming STT service - speech to text via NATS using Whisper with VAD and interrupt detection"
-readme = "README.md"
-requires-python = ">=3.11"
-license = { text = "MIT" }
-authors = [{ name = "Davies Tech Labs" }]
-
-dependencies = [
-    "nats-py>=2.0.0,<3.0.0",
-    "httpx>=0.20.0,<1.0.0",
-    "msgpack",
-    "numpy>=1.20.0,<2.0.0",
-    "webrtcvad>=2.0.10",
-    "setuptools<81",  # required by webrtcvad for pkg_resources
-    "opentelemetry-api",
-    "opentelemetry-sdk",
-    "opentelemetry-exporter-otlp-proto-grpc",
-    "opentelemetry-exporter-otlp-proto-http",
-    "opentelemetry-instrumentation-httpx",
-    "opentelemetry-instrumentation-logging",
-]
-
-[project.optional-dependencies]
-dev = [
-    "pytest>=8.0.0",
-    "pytest-asyncio>=0.23.0",
-    "ruff>=0.1.0",
-]
-
-[build-system]
-requires = ["hatchling"]
-build-backend = "hatchling.build"
-
-[tool.hatch.build.targets.wheel]
-packages = ["."]
-only-include = ["stt_streaming.py", "stt_streaming_local.py", "healthcheck.py"]
-
-[tool.ruff]
-line-length = 100
-target-version = "py311"
-
-[tool.ruff.lint]
-select = ["E", "F", "W", "I", "UP", "B", "C4", "SIM"]
-ignore = ["E501"]
-
-[tool.ruff.format]
-quote-style = "double"
-
-[tool.pytest.ini_options]
-asyncio_mode = "auto"
-testpaths = ["tests"]
-python_files = ["test_*.py"]
-python_classes = ["Test*"]
-python_functions = ["test_*"]
-addopts = "-v --tb=short"
-filterwarnings = ["ignore::DeprecationWarning"]

requirements.txt

@@ -1,20 +0,0 @@
-nats-py>=2.0.0,<3.0.0
-httpx>=0.20.0,<1.0.0
-msgpack
-
-# Audio processing
-numpy>=1.20.0,<2.0.0
-webrtcvad>=2.0.10
-# pyannote.audio>=3.1.0  # Optional: for advanced speaker diarization
-
-# OpenTelemetry core
-opentelemetry-api
-opentelemetry-sdk
-
-# OTEL exporters (gRPC for local collector, HTTP for HyperDX)
-opentelemetry-exporter-otlp-proto-grpc
-opentelemetry-exporter-otlp-proto-http
-
-# OTEL instrumentation
-opentelemetry-instrumentation-httpx
-opentelemetry-instrumentation-logging

stt_streaming.py

@@ -1,665 +0,0 @@
-#!/usr/bin/env python3
-"""
-Streaming STT Service
-
-Real-time Speech-to-Text service that processes live audio streams from NATS:
-1. Subscribe to audio stream subject (ai.voice.stream.{session_id})
-2. Buffer and accumulate audio chunks
-3. Transcribe when buffer reaches threshold or stream ends
-4. Publish transcription results to response channel (ai.voice.transcription.{session_id})
-
-This enables faster response times by processing audio as it arrives rather than
-waiting for complete audio upload.
-"""
-
-import asyncio
-import base64
-import logging
-import os
-import signal
-import time
-
-import httpx
-import msgpack
-import nats
-import nats.js
-import numpy as np
-import webrtcvad
-from nats.aio.msg import Msg
-
-# OpenTelemetry imports
-from opentelemetry import metrics, trace
-from opentelemetry.exporter.otlp.proto.grpc.metric_exporter import OTLPMetricExporter
-from opentelemetry.exporter.otlp.proto.grpc.trace_exporter import OTLPSpanExporter
-from opentelemetry.exporter.otlp.proto.http.metric_exporter import (
-    OTLPMetricExporter as OTLPMetricExporterHTTP,
-)
-from opentelemetry.exporter.otlp.proto.http.trace_exporter import (
-    OTLPSpanExporter as OTLPSpanExporterHTTP,
-)
-from opentelemetry.instrumentation.httpx import HTTPXClientInstrumentor
-from opentelemetry.instrumentation.logging import LoggingInstrumentor
-from opentelemetry.sdk.metrics import MeterProvider
-from opentelemetry.sdk.metrics.export import PeriodicExportingMetricReader
-from opentelemetry.sdk.resources import SERVICE_NAME, SERVICE_NAMESPACE, SERVICE_VERSION, Resource
-from opentelemetry.sdk.trace import TracerProvider
-from opentelemetry.sdk.trace.export import BatchSpanProcessor
-
-# Configure logging
-logging.basicConfig(
-    level=logging.INFO, format="%(asctime)s - %(name)s - %(levelname)s - %(message)s"
-)
-logger = logging.getLogger("stt-streaming")
-
-
-# Initialize OpenTelemetry
-def setup_telemetry():
-    """Initialize OpenTelemetry tracing and metrics with HyperDX support."""
-    # Check if OTEL is enabled
-    otel_enabled = os.environ.get("OTEL_ENABLED", "true").lower() == "true"
-    if not otel_enabled:
-        logger.info("OpenTelemetry disabled")
-        return None, None
-
-    # OTEL configuration
-    otel_endpoint = os.environ.get(
-        "OTEL_EXPORTER_OTLP_ENDPOINT",
-        "http://opentelemetry-collector.observability.svc.cluster.local:4317",
-    )
-    service_name = os.environ.get("OTEL_SERVICE_NAME", "stt-streaming")
-    service_namespace = os.environ.get("OTEL_SERVICE_NAMESPACE", "ai-ml")
-
-    # HyperDX configuration
-    hyperdx_api_key = os.environ.get("HYPERDX_API_KEY", "")
-    hyperdx_endpoint = os.environ.get("HYPERDX_ENDPOINT", "https://in-otel.hyperdx.io")
-    use_hyperdx = os.environ.get("HYPERDX_ENABLED", "false").lower() == "true" and hyperdx_api_key
-
-    # Create resource with service information
-    resource = Resource.create(
-        {
-            SERVICE_NAME: service_name,
-            SERVICE_VERSION: os.environ.get("SERVICE_VERSION", "1.0.0"),
-            SERVICE_NAMESPACE: service_namespace,
-            "deployment.environment": os.environ.get("DEPLOYMENT_ENV", "production"),
-            "host.name": os.environ.get("HOSTNAME", "unknown"),
-        }
-    )
-
-    # Setup tracing
-    trace_provider = TracerProvider(resource=resource)
-
-    if use_hyperdx:
-        # Use HTTP exporter for HyperDX with API key header
-        logger.info(f"Configuring HyperDX exporter at {hyperdx_endpoint}")
-        headers = {"authorization": hyperdx_api_key}
-        otlp_span_exporter = OTLPSpanExporterHTTP(
-            endpoint=f"{hyperdx_endpoint}/v1/traces", headers=headers
-        )
-        otlp_metric_exporter = OTLPMetricExporterHTTP(
-            endpoint=f"{hyperdx_endpoint}/v1/metrics", headers=headers
-        )
-    else:
-        # Use gRPC exporter for standard OTEL collector
-        otlp_span_exporter = OTLPSpanExporter(endpoint=otel_endpoint, insecure=True)
-        otlp_metric_exporter = OTLPMetricExporter(endpoint=otel_endpoint, insecure=True)
-
-    trace_provider.add_span_processor(BatchSpanProcessor(otlp_span_exporter))
-    trace.set_tracer_provider(trace_provider)
-
-    # Setup metrics
-    metric_reader = PeriodicExportingMetricReader(
-        otlp_metric_exporter, export_interval_millis=60000
-    )
-    meter_provider = MeterProvider(resource=resource, metric_readers=[metric_reader])
-    metrics.set_meter_provider(meter_provider)
-
-    # Instrument HTTPX
-    HTTPXClientInstrumentor().instrument()
-
-    # Instrument logging
-    LoggingInstrumentor().instrument(set_logging_format=True)
-
-    destination = "HyperDX" if use_hyperdx else "OTEL Collector"
-    logger.info(f"OpenTelemetry initialized - destination: {destination}, service: {service_name}")
-
-    # Return tracer and meter for the service
-    tracer = trace.get_tracer(__name__)
-    meter = metrics.get_meter(__name__)
-
-    return tracer, meter
-
-
-# Configuration from environment
-WHISPER_URL = os.environ.get("WHISPER_URL", "http://whisper-predictor.ai-ml.svc.cluster.local")
-NATS_URL = os.environ.get("NATS_URL", "nats://nats.ai-ml.svc.cluster.local:4222")
-
-# NATS subjects for streaming
-STREAM_SUBJECT_PREFIX = "ai.voice.stream"  # Full subject: ai.voice.stream.{session_id}
-TRANSCRIPTION_SUBJECT_PREFIX = (
-    "ai.voice.transcription"  # Full subject: ai.voice.transcription.{session_id}
-)
-
-# Streaming parameters
-BUFFER_SIZE_BYTES = int(
-    os.environ.get("STT_BUFFER_SIZE_BYTES", "512000")
-)  # ~5 seconds at 16kHz 16-bit
-CHUNK_TIMEOUT_SECONDS = float(
-    os.environ.get("STT_CHUNK_TIMEOUT", "2.0")
-)  # Process after 2s of silence
-MAX_BUFFER_SIZE_BYTES = int(os.environ.get("STT_MAX_BUFFER_SIZE", "5120000"))  # ~50 seconds max
-
-# Audio constants
-AUDIO_SAMPLE_MAX_INT16 = 32768.0  # Maximum value for 16-bit signed integer audio
-VAD_VOICE_RATIO_THRESHOLD = float(
-    os.environ.get("STT_VAD_VOICE_RATIO", "0.3")
-)  # Min ratio of voice frames
-
-# Voice Activity Detection (VAD) parameters
-ENABLE_VAD = os.environ.get("STT_ENABLE_VAD", "true").lower() == "true"
-VAD_AGGRESSIVENESS = int(
-    os.environ.get("STT_VAD_AGGRESSIVENESS", "2")
-)  # 0-3, higher = more aggressive
-VAD_FRAME_DURATION_MS = int(os.environ.get("STT_VAD_FRAME_DURATION", "30"))  # 10, 20, or 30 ms
-
-# Audio threshold for interrupt detection (when LLM is responding)
-ENABLE_INTERRUPT_DETECTION = (
-    os.environ.get("STT_ENABLE_INTERRUPT_DETECTION", "true").lower() == "true"
-)
-AUDIO_LEVEL_THRESHOLD = float(os.environ.get("STT_AUDIO_LEVEL_THRESHOLD", "0.02"))  # RMS threshold
-INTERRUPT_DURATION_THRESHOLD = float(
-    os.environ.get("STT_INTERRUPT_DURATION", "0.5")
-)  # Seconds of speech to trigger
-
-# Speaker diarization
-ENABLE_SPEAKER_DIARIZATION = (
-    os.environ.get("STT_ENABLE_SPEAKER_DIARIZATION", "false").lower() == "true"
-)
-
-# Session states
-SESSION_STATE_LISTENING = "listening"
-SESSION_STATE_RESPONDING = "responding"
-
-
-def calculate_audio_rms(audio_data: bytes, sample_width: int = 2) -> float:
-    """
-    Calculate RMS (Root Mean Square) audio level.
-
-    Args:
-        audio_data: Raw audio bytes
-        sample_width: Bytes per sample (2 for 16-bit audio)
-
-    Returns:
-        RMS level normalized to 0.0-1.0 range
-    """
-    if len(audio_data) < sample_width:
-        return 0.0
-
-    # Convert bytes to numpy array of int16 samples
-    try:
-        samples = np.frombuffer(audio_data, dtype=np.int16)
-        # Calculate RMS and normalize
-        rms = np.sqrt(np.mean(samples.astype(np.float32) ** 2))
-        # Normalize to 0-1 range using defined constant
-        return float(rms / AUDIO_SAMPLE_MAX_INT16)
-    except Exception as e:
-        logger.warning(f"Error calculating RMS: {e}")
-        return 0.0
-
-
-def detect_voice_activity(audio_data: bytes, sample_rate: int = 16000) -> bool:
-    """
-    Detect if audio contains voice using WebRTC VAD.
-
-    Args:
-        audio_data: Raw PCM audio bytes (16-bit, mono)
-        sample_rate: Audio sample rate (8000, 16000, 32000, or 48000)
-
-    Returns:
-        True if voice is detected, False otherwise
-    """
-    if not ENABLE_VAD:
-        return True  # Assume voice present if VAD disabled
-
-    try:
-        vad = webrtcvad.Vad(VAD_AGGRESSIVENESS)
-
-        # WebRTC VAD requires specific frame sizes
-        # Frame duration must be 10, 20, or 30 ms
-        frame_size = int(sample_rate * VAD_FRAME_DURATION_MS / 1000) * 2  # *2 for 16-bit samples
-
-        # Process audio in frames
-        voice_frames = 0
-        total_frames = 0
-
-        for i in range(0, len(audio_data) - frame_size, frame_size):
-            frame = audio_data[i : i + frame_size]
-            if len(frame) == frame_size:
-                try:
-                    is_speech = vad.is_speech(frame, sample_rate)
-                    if is_speech:
-                        voice_frames += 1
-                    total_frames += 1
-                except Exception as e:
-                    logger.debug(f"VAD frame processing error: {e}")
-                    continue
-
-        if total_frames == 0:
-            return False
-
-        # Consider voice detected if voice ratio exceeds threshold
-        voice_ratio = voice_frames / total_frames
-        return voice_ratio > VAD_VOICE_RATIO_THRESHOLD
-
-    except Exception as e:
-        logger.warning(f"VAD error: {e}")
-        return True  # Default to voice present on error
-
-
-class AudioBuffer:
-    """Manages audio chunks for a streaming session with VAD and speaker tracking."""
-
-    def __init__(self, session_id: str):
-        self.session_id = session_id
-        self.chunks = []
-        self.total_bytes = 0
-        self.last_chunk_time = time.time()
-        self.is_complete = False
-        self.sequence = 0
-        self.state = SESSION_STATE_LISTENING  # Current session state
-        self.speaker_id = None  # For speaker diarization
-        self.interrupt_start_time = None  # Track when interrupt detection started
-        self.has_voice_activity = False  # Track if voice was detected in recent chunks
-        self._last_chunk_vad_result = None  # Cache VAD result for last chunk
-
-    def add_chunk(self, audio_data: bytes) -> None:
-        """Add an audio chunk to the buffer and check for voice activity."""
-        self.chunks.append(audio_data)
-        self.total_bytes += len(audio_data)
-        self.last_chunk_time = time.time()
-
-        # Check for voice activity in this chunk and cache result
-        has_voice = detect_voice_activity(audio_data)
-        self.has_voice_activity = has_voice
-        self._last_chunk_vad_result = has_voice
-
-        logger.debug(
-            f"Session {self.session_id}: Added chunk, total {self.total_bytes} bytes, voice={has_voice}"
-        )
-
-    def check_interrupt(self, audio_data: bytes) -> bool:
-        """
-        Check if audio indicates an interrupt during responding state.
-        Uses cached VAD result if available.
-
-        Returns:
-            True if interrupt detected, False otherwise
-        """
-        if not ENABLE_INTERRUPT_DETECTION:
-            return False
-
-        if self.state != SESSION_STATE_RESPONDING:
-            return False
-
-        # Calculate audio level
-        rms_level = calculate_audio_rms(audio_data)
-
-        # Use cached VAD result if available to avoid duplicate processing
-        has_voice = (
-            self._last_chunk_vad_result
-            if self._last_chunk_vad_result is not None
-            else detect_voice_activity(audio_data)
-        )
-
-        # Check if audio exceeds threshold and contains voice
-        if rms_level >= AUDIO_LEVEL_THRESHOLD and has_voice:
-            if self.interrupt_start_time is None:
-                self.interrupt_start_time = time.time()
-                logger.info(
-                    f"Session {self.session_id}: Potential interrupt detected (RMS={rms_level:.3f})"
-                )
-
-            # Check if interrupt has lasted long enough
-            elapsed = time.time() - self.interrupt_start_time
-            if elapsed >= INTERRUPT_DURATION_THRESHOLD:
-                logger.info(f"Session {self.session_id}: Interrupt confirmed after {elapsed:.1f}s")
-                return True
-        else:
-            # Reset interrupt timer if audio drops below threshold
-            self.interrupt_start_time = None
-
-        return False
-
-    def set_state(self, state: str) -> None:
-        """Set the session state (listening or responding)."""
-        if state in (SESSION_STATE_LISTENING, SESSION_STATE_RESPONDING):
-            old_state = self.state
-            self.state = state
-            if old_state != state:
-                logger.info(f"Session {self.session_id}: State changed from {old_state} to {state}")
-                # Reset interrupt tracking when changing states
-                self.interrupt_start_time = None
-
-    def should_process(self) -> bool:
-        """Determine if buffer should be processed now."""
-        # Don't process if no voice activity detected (unless buffer is full or timed out)
-        if (
-            ENABLE_VAD
-            and not self.has_voice_activity
-            and self.total_bytes < BUFFER_SIZE_BYTES
-            and time.time() - self.last_chunk_time < CHUNK_TIMEOUT_SECONDS
-        ):
-            return False
-
-        # Process if buffer size threshold reached
-        if self.total_bytes >= BUFFER_SIZE_BYTES:
-            return True
-        # Process if no chunks received for timeout duration
-        if time.time() - self.last_chunk_time > CHUNK_TIMEOUT_SECONDS and self.total_bytes > 0:
-            return True
-        # Process if buffer is too large (safety limit)
-        return self.total_bytes >= MAX_BUFFER_SIZE_BYTES
-
-    def get_audio(self) -> bytes:
-        """Get concatenated audio data."""
-        return b"".join(self.chunks)
-
-    def clear(self) -> None:
-        """Clear the buffer after processing."""
-        self.chunks = []
-        self.total_bytes = 0
-        self.sequence += 1
-        self._last_chunk_vad_result = None  # Clear cached VAD result
-
-    def mark_complete(self) -> None:
-        """Mark stream as complete."""
-        self.is_complete = True
-
-
-class StreamingSTT:
-    """Streaming Speech-to-Text service."""
-
-    def __init__(self):
-        self.nc = None
-        self.js = None
-        self.http_client = None
-        self.sessions: dict[str, AudioBuffer] = {}
-        self.running = True
-        self.processing_tasks = {}
-        self.is_healthy = False
-        self.tracer = None
-        self.meter = None
-        self.stream_counter = None
-        self.transcription_duration = None
-
-    async def setup(self):
-        """Initialize connections."""
-        # Initialize OpenTelemetry
-        self.tracer, self.meter = setup_telemetry()
-
-        # Create metrics if OTEL is enabled
-        if self.meter:
-            self.stream_counter = self.meter.create_counter(
-                name="stt_streams_total",
-                description="Total number of STT streams processed",
-                unit="1",
-            )
-            self.transcription_duration = self.meter.create_histogram(
-                name="stt_transcription_duration_seconds",
-                description="Duration of STT transcription",
-                unit="s",
-            )
-
-        # NATS connection
-        self.nc = await nats.connect(NATS_URL)
-        logger.info(f"Connected to NATS at {NATS_URL}")
-
-        # Initialize JetStream context
-        self.js = self.nc.jetstream()
-
-        # Create or update stream for voice stream messages
-        try:
-            stream_config = nats.js.api.StreamConfig(
-                name="AI_VOICE_STREAM",
-                subjects=["ai.voice.stream.>", "ai.voice.transcription.>"],
-                retention=nats.js.api.RetentionPolicy.LIMITS,
-                max_age=300,  # Keep messages for 5 minutes only (streaming is ephemeral)
-                storage=nats.js.api.StorageType.MEMORY,  # Use memory for streaming data
-            )
-            await self.js.add_stream(stream_config)
-            logger.info("Created/updated JetStream stream: AI_VOICE_STREAM")
-        except Exception as e:
-            # Stream might already exist
-            logger.info(f"JetStream stream setup: {e}")
-
-        # HTTP client for Whisper service
-        self.http_client = httpx.AsyncClient(timeout=180.0)
-        logger.info("HTTP client initialized")
-
-        # Mark as healthy once connections are established
-        self.is_healthy = True
-
-    async def transcribe(self, audio_bytes: bytes) -> str | None:
-        """Transcribe audio using Whisper."""
-        try:
-            files = {"file": ("audio.wav", audio_bytes, "audio/wav")}
-            response = await self.http_client.post(
-                f"{WHISPER_URL}/v1/audio/transcriptions", files=files
-            )
-            response.raise_for_status()
-            result = response.json()
-            transcript = result.get("text", "")
-            logger.info(f"Transcribed: {transcript[:100]}...")
-            return transcript
-        except Exception as e:
-            logger.error(f"Transcription failed: {e}")
-            return None
-
-    async def process_buffer(self, session_id: str):
-        """Process accumulated audio buffer for a session."""
-        buffer = self.sessions.get(session_id)
-        if not buffer:
-            return
-
-        audio_data = buffer.get_audio()
-        if not audio_data:
-            return
-
-        logger.info(
-            f"Processing {len(audio_data)} bytes for session {session_id}, sequence {buffer.sequence}"
-        )
-
-        # Transcribe
-        transcript = await self.transcribe(audio_data)
-
-        if transcript:
-            # Publish transcription result using msgpack binary format
-            result = {
-                "session_id": session_id,
-                "transcript": transcript,
-                "sequence": buffer.sequence,
-                "is_partial": not buffer.is_complete,
-                "is_final": buffer.is_complete,
-                "timestamp": time.time(),
-                "speaker_id": buffer.speaker_id,
-                "has_voice_activity": buffer.has_voice_activity,
-                "state": buffer.state,
-            }
-
-            await self.nc.publish(
-                f"{TRANSCRIPTION_SUBJECT_PREFIX}.{session_id}", msgpack.packb(result)
-            )
-            logger.info(
-                f"Published transcription for session {session_id} (seq {buffer.sequence}, speaker={buffer.speaker_id})"
-            )
-
-        # Clear buffer after processing
-        buffer.clear()
-
-        # Clean up completed sessions asynchronously
-        if buffer.is_complete:
-            logger.info(f"Session {session_id} completed")
-            # Schedule cleanup task to avoid blocking
-            asyncio.create_task(self._cleanup_session(session_id))
-
-    async def _cleanup_session(self, session_id: str):
-        """Clean up a completed session after a delay."""
-        # Keep session for a bit in case of late messages
-        await asyncio.sleep(5)
-        if session_id in self.sessions:
-            del self.sessions[session_id]
-            logger.info(f"Cleaned up session: {session_id}")
-        if session_id in self.processing_tasks:
-            del self.processing_tasks[session_id]
-
-    async def monitor_buffer(self, session_id: str):
-        """Monitor buffer and trigger processing when needed."""
-        while self.running and session_id in self.sessions:
-            buffer = self.sessions.get(session_id)
-            if not buffer:
-                break
-
-            if buffer.should_process():
-                await self.process_buffer(session_id)
-
-            # Don't spin too fast
-            await asyncio.sleep(0.1)
-
-    async def handle_stream_message(self, msg: Msg):
-        """Handle incoming audio stream message."""
-        try:
-            # Extract session_id from subject: ai.voice.stream.{session_id}
-            subject_parts = msg.subject.split(".")
-            if len(subject_parts) < 4:
-                logger.warning(f"Invalid subject format: {msg.subject}")
-                return
-
-            session_id = subject_parts[3]
-
-            # Parse message using msgpack binary format
-            data = msgpack.unpackb(msg.data, raw=False)
-
-            # Handle control messages
-            if data.get("type") == "start":
-                logger.info(f"Starting stream session: {session_id}")
-                self.sessions[session_id] = AudioBuffer(session_id)
-                # Set initial state if provided
-                initial_state = data.get("state", SESSION_STATE_LISTENING)
-                self.sessions[session_id].set_state(initial_state)
-                # Store speaker_id if provided
-                speaker_id = data.get("speaker_id")
-                if speaker_id:
-                    self.sessions[session_id].speaker_id = speaker_id
-                    logger.info(f"Session {session_id}: Speaker ID set to {speaker_id}")
-                # Start monitoring task for this session
-                task = asyncio.create_task(self.monitor_buffer(session_id))
-                self.processing_tasks[session_id] = task
-                return
-
-            if data.get("type") == "state_change":
-                logger.info(f"State change for session {session_id}")
-                buffer = self.sessions.get(session_id)
-                if buffer:
-                    new_state = data.get("state", SESSION_STATE_LISTENING)
-                    buffer.set_state(new_state)
-
-                    # If switching to listening mode, reset any interrupt tracking
-                    if new_state == SESSION_STATE_LISTENING:
-                        buffer.interrupt_start_time = None
-                return
-
-            if data.get("type") == "end":
-                logger.info(f"Ending stream session: {session_id}")
-                buffer = self.sessions.get(session_id)
-                if buffer:
-                    buffer.mark_complete()
-                    # Process any remaining audio
-                    if buffer.total_bytes > 0:
-                        await self.process_buffer(session_id)
-                return
-
-            # Handle audio chunk
-            if data.get("type") == "chunk":
-                audio_b64 = data.get("audio_b64", "")
-                if not audio_b64:
-                    return
-
-                audio_bytes = base64.b64decode(audio_b64)
-
-                # Create session if it doesn't exist (handle missing start message)
-                # Check both sessions and processing_tasks to avoid race conditions
-                if session_id not in self.sessions:
-                    logger.info(f"Auto-creating session: {session_id}")
-                    self.sessions[session_id] = AudioBuffer(session_id)
-                    # Only create monitoring task if not already exists
-                    if session_id not in self.processing_tasks:
-                        task = asyncio.create_task(self.monitor_buffer(session_id))
-                        self.processing_tasks[session_id] = task
-
-                buffer = self.sessions[session_id]
-
-                # Check for interrupt if in responding state
-                if buffer.check_interrupt(audio_bytes):
-                    # Publish interrupt notification
-                    interrupt_msg = {
-                        "session_id": session_id,
-                        "type": "interrupt",
-                        "timestamp": time.time(),
-                        "speaker_id": buffer.speaker_id,
-                    }
-                    await self.nc.publish(
-                        f"{TRANSCRIPTION_SUBJECT_PREFIX}.{session_id}", msgpack.packb(interrupt_msg)
-                    )
-                    logger.info(f"Published interrupt notification for session {session_id}")
-
-                    # Automatically switch back to listening mode
-                    buffer.set_state(SESSION_STATE_LISTENING)
-
-                # Add chunk to buffer
-                buffer.add_chunk(audio_bytes)
-
-        except Exception as e:
-            logger.error(f"Error handling stream message: {e}", exc_info=True)
-
-    async def run(self):
-        """Main run loop."""
-        await self.setup()
-
-        # Note: STT streaming uses regular NATS subscribe (not pull-based JetStream consumer)
-        # because it handles real-time ephemeral audio streams with wildcard subscriptions.
-        # The stream audio chunks are not meant to be persisted long-term or replayed.
-        # However, the transcription RESULTS are published to JetStream for persistence.
-        sub = await self.nc.subscribe(f"{STREAM_SUBJECT_PREFIX}.>", cb=self.handle_stream_message)
-        logger.info(f"Subscribed to {STREAM_SUBJECT_PREFIX}.>")
-
-        # Handle shutdown
-        def signal_handler():
-            self.running = False
-
-        loop = asyncio.get_event_loop()
-        for sig in (signal.SIGTERM, signal.SIGINT):
-            loop.add_signal_handler(sig, signal_handler)
-
-        # Keep running
-        while self.running:
-            await asyncio.sleep(1)
-
-        # Cleanup
-        logger.info("Shutting down...")
-
-        # Cancel all monitoring tasks and wait for them to complete
-        for task in self.processing_tasks.values():
-            task.cancel()
-
-        # Wait for all tasks to complete or be cancelled
-        if self.processing_tasks:
-            await asyncio.gather(*self.processing_tasks.values(), return_exceptions=True)
-
-        await sub.unsubscribe()
-        await self.nc.close()
-        await self.http_client.aclose()
-        logger.info("Shutdown complete")
-
-
-if __name__ == "__main__":
-    service = StreamingSTT()
-    asyncio.run(service.run())

tests/conftest.py

@@ -1,71 +0,0 @@
-"""
-Pytest configuration and fixtures for stt-module tests.
-"""
-
-import asyncio
-import base64
-import os
-import struct
-from unittest.mock import AsyncMock, MagicMock
-
-import pytest
-
-# Set test environment variables before importing
-os.environ.setdefault("NATS_URL", "nats://localhost:4222")
-os.environ.setdefault("WHISPER_URL", "http://localhost:8000")
-os.environ.setdefault("OTEL_ENABLED", "false")
-os.environ.setdefault("STT_ENABLE_VAD", "false")
-
-
-@pytest.fixture(scope="session")
-def event_loop():
-    """Create event loop for async tests."""
-    loop = asyncio.new_event_loop()
-    yield loop
-    loop.close()
-
-
-@pytest.fixture
-def silent_pcm_bytes():
-    """16-bit PCM silence (1000 samples at 16kHz = 62.5ms)."""
-    return bytes([0x00] * 2000)
-
-
-@pytest.fixture
-def noisy_pcm_bytes():
-    """16-bit PCM with some signal (sine-like pattern)."""
-    samples = []
-    for i in range(1000):
-        val = int(16000 * ((i % 50) / 50.0 - 0.5))
-        samples.append(struct.pack("<h", val))
-    return b"".join(samples)
-
-
-@pytest.fixture
-def sample_audio_b64(silent_pcm_bytes):
-    """Base64-encoded silent PCM audio."""
-    return base64.b64encode(silent_pcm_bytes).decode()
-
-
-@pytest.fixture
-def mock_nats():
-    """Mock NATS connection."""
-    nc = AsyncMock()
-    nc.publish = AsyncMock()
-    nc.subscribe = AsyncMock()
-    nc.close = AsyncMock()
-    nc.jetstream = MagicMock(return_value=AsyncMock())
-    return nc
-
-
-@pytest.fixture
-def mock_http_client():
-    """Mock httpx async client."""
-    client = AsyncMock()
-    response = MagicMock()
-    response.status_code = 200
-    response.json.return_value = {"text": "Hello world"}
-    response.raise_for_status = MagicMock()
-    client.post = AsyncMock(return_value=response)
-    client.aclose = AsyncMock()
-    return client

tests/test_stt_streaming.py

@@ -1,262 +0,0 @@
-"""
-Unit tests for STT streaming service.
-"""
-
-import time
-from unittest.mock import AsyncMock, MagicMock, patch
-
-import msgpack
-import pytest
-
-from stt_streaming import (
-    TRANSCRIPTION_SUBJECT_PREFIX,
-    AudioBuffer,
-    StreamingSTT,
-    calculate_audio_rms,
-)
-
-# ---------------------------------------------------------------------------
-# Utility function tests
-# ---------------------------------------------------------------------------
-
-
-class TestCalculateAudioRms:
-    """Tests for calculate_audio_rms helper."""
-
-    def test_silence_returns_zero(self, silent_pcm_bytes):
-        rms = calculate_audio_rms(silent_pcm_bytes)
-        assert rms == 0.0
-
-    def test_noisy_signal_above_zero(self, noisy_pcm_bytes):
-        rms = calculate_audio_rms(noisy_pcm_bytes)
-        assert rms > 0.0
-
-    def test_empty_bytes(self):
-        rms = calculate_audio_rms(b"")
-        assert rms == 0.0
-
-    def test_single_byte(self):
-        rms = calculate_audio_rms(b"\x00")
-        assert rms == 0.0
-
-
-# ---------------------------------------------------------------------------
-# AudioBuffer tests
-# ---------------------------------------------------------------------------
-
-
-class TestAudioBuffer:
-    """Tests for AudioBuffer session management."""
-
-    def test_init(self):
-        buf = AudioBuffer("sess-1")
-        assert buf.session_id == "sess-1"
-        assert buf.total_bytes == 0
-        assert buf.is_complete is False
-        assert buf.sequence == 0
-        assert buf.state == "listening"
-
-    def test_add_chunk(self, silent_pcm_bytes):
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(silent_pcm_bytes)
-        assert buf.total_bytes == len(silent_pcm_bytes)
-        assert len(buf.chunks) == 1
-
-    def test_get_audio_concatenates(self, silent_pcm_bytes):
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(silent_pcm_bytes)
-        buf.add_chunk(silent_pcm_bytes)
-        audio = buf.get_audio()
-        assert len(audio) == len(silent_pcm_bytes) * 2
-
-    def test_clear_resets(self, silent_pcm_bytes):
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(silent_pcm_bytes)
-        buf.clear()
-        assert buf.total_bytes == 0
-        assert buf.chunks == []
-        assert buf.sequence == 1  # Incremented after clear
-
-    def test_mark_complete(self):
-        buf = AudioBuffer("sess-1")
-        buf.mark_complete()
-        assert buf.is_complete is True
-
-    def test_set_state(self):
-        buf = AudioBuffer("sess-1")
-        assert buf.state == "listening"
-        buf.set_state("responding")
-        assert buf.state == "responding"
-        buf.set_state("listening")
-        assert buf.state == "listening"
-
-    def test_set_invalid_state_ignored(self):
-        buf = AudioBuffer("sess-1")
-        buf.set_state("invalid")
-        assert buf.state == "listening"  # Unchanged
-
-    @patch("stt_streaming.BUFFER_SIZE_BYTES", 100)
-    def test_should_process_when_buffer_full(self, silent_pcm_bytes):
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(silent_pcm_bytes)  # 2000 bytes > 100 threshold
-        assert buf.should_process() is True
-
-    def test_should_not_process_when_empty(self):
-        buf = AudioBuffer("sess-1")
-        assert buf.should_process() is False
-
-    @patch("stt_streaming.CHUNK_TIMEOUT_SECONDS", 0.0)
-    def test_should_process_on_timeout(self, silent_pcm_bytes):
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(silent_pcm_bytes)
-        buf.last_chunk_time = time.time() - 10  # Force timeout
-        assert buf.should_process() is True
-
-
-# ---------------------------------------------------------------------------
-# StreamingSTT tests
-# ---------------------------------------------------------------------------
-
-
-class TestStreamingSTT:
-    """Tests for the StreamingSTT service."""
-
-    @pytest.fixture
-    def service(self, mock_nats, mock_http_client):
-        svc = StreamingSTT()
-        svc.nc = mock_nats
-        svc.js = mock_nats.jetstream()
-        svc.http_client = mock_http_client
-        svc.is_healthy = True
-        return svc
-
-    @pytest.mark.asyncio
-    async def test_transcribe_success(self, service):
-        """Successful transcription returns text."""
-        result = await service.transcribe(b"fake-audio")
-        assert result == "Hello world"
-        service.http_client.post.assert_called_once()
-
-    @pytest.mark.asyncio
-    async def test_transcribe_failure(self, service):
-        """Transcription failure returns None."""
-        service.http_client.post = AsyncMock(side_effect=Exception("timeout"))
-        result = await service.transcribe(b"fake-audio")
-        assert result is None
-
-    @pytest.mark.asyncio
-    async def test_handle_start_message(self, service):
-        """Start message creates a new session buffer."""
-        msg = MagicMock()
-        msg.subject = "ai.voice.stream.sess-1"
-        msg.data = msgpack.packb({"type": "start"})
-
-        await service.handle_stream_message(msg)
-
-        assert "sess-1" in service.sessions
-        assert service.sessions["sess-1"].state == "listening"
-
-    @pytest.mark.asyncio
-    async def test_handle_start_with_speaker_id(self, service):
-        """Start message with speaker_id stores it on the buffer."""
-        msg = MagicMock()
-        msg.subject = "ai.voice.stream.sess-2"
-        msg.data = msgpack.packb({"type": "start", "speaker_id": "user-42"})
-
-        await service.handle_stream_message(msg)
-        assert service.sessions["sess-2"].speaker_id == "user-42"
-
-    @pytest.mark.asyncio
-    async def test_handle_state_change(self, service):
-        """State change message updates the buffer state."""
-        # Create session first
-        service.sessions["sess-1"] = AudioBuffer("sess-1")
-
-        msg = MagicMock()
-        msg.subject = "ai.voice.stream.sess-1"
-        msg.data = msgpack.packb({"type": "state_change", "state": "responding"})
-
-        await service.handle_stream_message(msg)
-        assert service.sessions["sess-1"].state == "responding"
-
-    @pytest.mark.asyncio
-    async def test_handle_audio_chunk(self, service, sample_audio_b64):
-        """Audio chunks are added to the session buffer."""
-        service.sessions["sess-1"] = AudioBuffer("sess-1")
-        # Prevent auto-creation of monitoring task
-        service.processing_tasks["sess-1"] = MagicMock()
-
-        msg = MagicMock()
-        msg.subject = "ai.voice.stream.sess-1"
-        msg.data = msgpack.packb({"type": "chunk", "audio_b64": sample_audio_b64})
-
-        await service.handle_stream_message(msg)
-        assert service.sessions["sess-1"].total_bytes > 0
-
-    @pytest.mark.asyncio
-    async def test_handle_end_message(self, service, mock_http_client):
-        """End message triggers processing of remaining audio."""
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(b"\x00" * 100)
-        service.sessions["sess-1"] = buf
-
-        msg = MagicMock()
-        msg.subject = "ai.voice.stream.sess-1"
-        msg.data = msgpack.packb({"type": "end"})
-
-        await service.handle_stream_message(msg)
-
-        # Should have published a transcription
-        assert service.nc.publish.called
-
-    @pytest.mark.asyncio
-    async def test_handle_auto_create_session(self, service, sample_audio_b64):
-        """Chunk message auto-creates session when start was missed."""
-        msg = MagicMock()
-        msg.subject = "ai.voice.stream.new-sess"
-        msg.data = msgpack.packb({"type": "chunk", "audio_b64": sample_audio_b64})
-
-        await service.handle_stream_message(msg)
-        assert "new-sess" in service.sessions
-
-    @pytest.mark.asyncio
-    async def test_process_buffer_publishes_result(self, service, mock_http_client):
-        """process_buffer publishes transcription to NATS."""
-        buf = AudioBuffer("sess-1")
-        buf.add_chunk(b"\x00" * 100)
-        service.sessions["sess-1"] = buf
-
-        await service.process_buffer("sess-1")
-
-        # Verify transcription published
-        pub_calls = service.nc.publish.call_args_list
-        assert len(pub_calls) >= 1
-        subject = pub_calls[0].args[0]
-        assert subject == f"{TRANSCRIPTION_SUBJECT_PREFIX}.sess-1"
-        data = msgpack.unpackb(pub_calls[0].args[1], raw=False)
-        assert data["transcript"] == "Hello world"
-        assert data["session_id"] == "sess-1"
-
-    @pytest.mark.asyncio
-    async def test_process_buffer_no_session(self, service):
-        """process_buffer handles missing session gracefully."""
-        await service.process_buffer("nonexistent")
-        service.nc.publish.assert_not_called()
-
-    @pytest.mark.asyncio
-    async def test_process_buffer_empty_audio(self, service):
-        """process_buffer skips empty buffers."""
-        service.sessions["sess-1"] = AudioBuffer("sess-1")
-        await service.process_buffer("sess-1")
-        service.nc.publish.assert_not_called()
-
-    def test_invalid_subject(self, service):
-        """Messages with invalid subjects are skipped."""
-        msg = MagicMock()
-        msg.subject = "ai.voice"  # Too few parts
-        msg.data = msgpack.packb({"type": "start"})
-
-        import asyncio
-
-        asyncio.get_event_loop().run_until_complete(service.handle_stream_message(msg))
-        assert len(service.sessions) == 0

uv.lock

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