release: bump version to 0.0.4.7

- aria_bridge clean_text_for_tts: "0.1" / "0,5" / "1,25" wird jetzt als
  "null komma eins" / "null komma fuenf" / "eins komma zwei fuenf"
  ausgeschrieben. Lookahead verhindert Match auf IP-artige Strings.
- PcmStreamPlayer: 200ms Stille am Stream-Anfang, damit AudioTrack
  sauber anfaehrt und die ersten Worte nicht verschluckt werden.
  (XTTS-Warmup + play()-Startup-Latenz)

Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>

android/android/app/build.gradle

@@ -79,8 +79,8 @@ android {
         applicationId "com.ariacockpit"
         minSdkVersion rootProject.ext.minSdkVersion
         targetSdkVersion rootProject.ext.targetSdkVersion
-        versionCode 404
-        versionName "0.0.4.4"
+        versionCode 407
+        versionName "0.0.4.7"
         // Fallback fuer Libraries mit Product Flavors
         missingDimensionStrategy 'react-native-camera', 'general'
     }

android/android/app/src/main/java/com/ariacockpit/PcmStreamPlayerModule.kt

@@ -13,22 +13,29 @@ import com.facebook.react.bridge.ReactMethod
 import java.util.concurrent.LinkedBlockingQueue
 
 /**
- * Streamt PCM-s16le Audio direkt via AudioTrack MODE_STREAM.
+ * Streamt PCM-s16le Audio direkt via AudioTrack MODE_STREAM mit Pre-Roll.
+ *
+ * Pre-Roll: AudioTrack wird zwar direkt gebaut und gefuttert, aber play()
+ * wird erst aufgerufen wenn PREROLL_SECONDS Audio im Buffer ist. So hat
+ * der Stream Zeit einen Vorrat aufzubauen — wenn XTTS mit RTF>1 rendert
+ * (langsamer als Echtzeit), laeuft der Buffer trotzdem nicht leer.
  *
  * Flow:
- *   JS: start(sampleRate, channels) → öffnet AudioTrack und startet Writer-Thread
- *   JS: writeChunk(base64)           → dekodiert, queued, Writer schreibt non-blocking
- *   JS: end()                         → wartet bis Queue leer, schließt AudioTrack
- *   JS: stop()                        → Hart stoppen, Queue leeren (Cancel)
- *
- * Vorteil gegenüber Sound-File-Queue:
- *   - Keine Gap zwischen Chunks (AudioTrack puffert intern)
- *   - Erste Samples beginnen zu spielen sobald der erste Chunk da ist
- *   - Kein WAV-Header-Parsing pro Chunk
+ *   JS: start(sampleRate, channels) → öffnet AudioTrack (noch nicht play())
+ *   JS: writeChunk(base64)           → dekodiert, queued, Writer schreibt
+ *   Writer: spielt los sobald PREROLL erreicht ist
+ *   JS: end()                         → wartet bis Queue leer, schließt
+ *   JS: stop()                        → Hart stoppen (Cancel)
  */
 class PcmStreamPlayerModule(reactContext: ReactApplicationContext) : ReactContextBaseJavaModule(reactContext) {
     companion object {
         private const val TAG = "PcmStreamPlayer"
+        // Sekunden Audio die VOR play()-Start gepuffert sein muessen.
+        // 2.5s Vorrat = genug um XTTS-Render-Pausen zwischen Chunks zu puffern.
+        private const val PREROLL_SECONDS = 2.5
+        // Stille am Stream-Anfang, damit AudioTrack sauber anfaehrt und die
+        // ersten Samples nicht abgeschnitten werden (XTTS-Warmup + play()-Latenz).
+        private const val LEADING_SILENCE_SECONDS = 0.2
     }
 
     override fun getName() = "PcmStreamPlayer"
@@ -38,6 +45,10 @@ class PcmStreamPlayerModule(reactContext: ReactApplicationContext) : ReactContex
     private var writerThread: Thread? = null
     @Volatile private var writerShouldStop = false
     @Volatile private var endRequested = false
+    @Volatile private var prerollBytes: Int = 0
+    @Volatile private var playbackStarted = false
+    @Volatile private var bytesBuffered: Long = 0
+    @Volatile private var streamBytesPerFrame: Int = 2 // mono s16le default
 
     // ── Lifecycle ──
 
@@ -50,10 +61,14 @@ class PcmStreamPlayerModule(reactContext: ReactApplicationContext) : ReactContex
             val channelConfig = if (channels == 2) AudioFormat.CHANNEL_OUT_STEREO else AudioFormat.CHANNEL_OUT_MONO
             val encoding = AudioFormat.ENCODING_PCM_16BIT
             val minBuf = AudioTrack.getMinBufferSize(sampleRate, channelConfig, encoding)
-            // Grosszuegiger Buffer: 32x MinSize — tolerant gegen Netzwerk-Jitter und
-            // bursty XTTS-Delivery (Render dauert 1-3s, dann kommen alle Samples
-            // auf einmal). Bei 24kHz mono s16 entspricht 128KB ca. 2.7 Sekunden.
-            val bufferSize = (minBuf * 32).coerceAtLeast(128 * 1024)
+            val bytesPerSecond = sampleRate * channels * 2 // 16-bit = 2 bytes
+            // Buffer muss mindestens PREROLL + etwas Spielraum fassen.
+            val prerollTarget = (bytesPerSecond * PREROLL_SECONDS).toInt()
+            val bufferSize = (minBuf * 32).coerceAtLeast(prerollTarget * 2)
+            prerollBytes = prerollTarget
+            bytesBuffered = 0
+            playbackStarted = false
+            streamBytesPerFrame = channels * 2 // s16 = 2 bytes per sample
 
             val newTrack = AudioTrack.Builder()
                 .setAudioAttributes(
@@ -73,7 +88,7 @@ class PcmStreamPlayerModule(reactContext: ReactApplicationContext) : ReactContex
                 .setTransferMode(AudioTrack.MODE_STREAM)
                 .build()
 
-            newTrack.play()
+            // AudioTrack erstellen — play() wird erst aufgerufen wenn Pre-Roll erreicht.
             track = newTrack
             queue.clear()
             writerShouldStop = false
@@ -82,27 +97,83 @@ class PcmStreamPlayerModule(reactContext: ReactApplicationContext) : ReactContex
             writerThread = Thread({
                 val t = track ?: return@Thread
                 try {
+                    // Leading-Silence in den Buffer — gibt AudioTrack Zeit anzufahren.
+                    val silenceBytes = ((sampleRate * channels * 2) * LEADING_SILENCE_SECONDS).toInt() and 0x7FFFFFFE
+                    if (silenceBytes > 0) {
+                        val silence = ByteArray(silenceBytes)
+                        var silOff = 0
+                        while (silOff < silence.size && !writerShouldStop) {
+                            val w = t.write(silence, silOff, silence.size - silOff)
+                            if (w <= 0) break
+                            silOff += w
+                        }
+                        bytesBuffered += silence.size
+                    }
                     while (!writerShouldStop) {
                         val data = queue.poll(50, java.util.concurrent.TimeUnit.MILLISECONDS) ?: run {
-                            if (endRequested) return@Thread
+                            if (endRequested) {
+                                // Falls wir vor Pre-Roll enden (kurzer Text): trotzdem abspielen
+                                if (!playbackStarted) {
+                                    try { t.play() } catch (_: Exception) {}
+                                    playbackStarted = true
+                                }
+                                return@Thread
+                            }
                             null
                         } ?: continue
+
+                        // Pre-Roll Check: play() erst wenn genug gepuffert
+                        if (!playbackStarted && bytesBuffered + data.size >= prerollBytes) {
+                            try {
+                                t.play()
+                                playbackStarted = true
+                                Log.i(TAG, "Playback gestartet nach Pre-Roll ${bytesBuffered + data.size} Bytes")
+                            } catch (e: Exception) {
+                                Log.w(TAG, "play() failed: ${e.message}")
+                            }
+                        }
+
                         var offset = 0
                         while (offset < data.size && !writerShouldStop) {
                             val written = t.write(data, offset, data.size - offset)
                             if (written <= 0) break
                             offset += written
                         }
+                        bytesBuffered += data.size
                     }
                 } catch (e: Exception) {
                     Log.w(TAG, "Writer-Thread Fehler: ${e.message}")
                 } finally {
+                    // Warten bis alle geschriebenen Samples tatsaechlich abgespielt sind,
+                    // sonst cuttet t.release() die letzten Sekunden ab.
+                    try {
+                        val totalFrames = (bytesBuffered / streamBytesPerFrame).toInt()
+                        var lastPos = -1
+                        var stalledCount = 0
+                        while (!writerShouldStop) {
+                            val pos = t.playbackHeadPosition
+                            if (pos >= totalFrames) break
+                            // Safety: wenn Position 2s nicht mehr vorwaerts → AudioTrack hing
+                            if (pos == lastPos) {
+                                stalledCount++
+                                if (stalledCount > 40) {
+                                    Log.w(TAG, "playback stalled at $pos/$totalFrames — give up")
+                                    break
+                                }
+                            } else {
+                                stalledCount = 0
+                                lastPos = pos
+                            }
+                            Thread.sleep(50)
+                        }
+                        Log.i(TAG, "Playback fertig: frames=$totalFrames pos=${t.playbackHeadPosition}")
+                    } catch (_: Exception) {}
                     try { t.stop() } catch (_: Exception) {}
                     try { t.release() } catch (_: Exception) {}
                 }
             }, "PcmStreamWriter").apply { start() }
 
-            Log.i(TAG, "Stream gestartet: ${sampleRate}Hz ch=$channels buf=${bufferSize}B")
+            Log.i(TAG, "Stream gestartet: ${sampleRate}Hz ch=$channels buf=${bufferSize}B preroll=${prerollBytes}B")
             promise.resolve(true)
         } catch (e: Exception) {
             Log.e(TAG, "start fehlgeschlagen", e)

android/package.json

@@ -1,6 +1,6 @@
 {
   "name": "aria-cockpit",
-  "version": "0.0.4.4",
+  "version": "0.0.4.7",
   "private": true,
   "scripts": {
     "android": "react-native run-android",

bridge/aria_bridge.py

@@ -150,6 +150,15 @@ def _small_range_to_words(m):
     return f"{_num_to_words_de(a)} bis {_num_to_words_de(b)}"
 
 
+def _decimal_to_words(m):
+    """'0.1' / '0,1' → 'null komma eins', '1,25' → 'eins komma zwei fuenf'."""
+    int_part = int(m.group(1))
+    dec_part = m.group(2)
+    int_word = _num_to_words_de(int_part) if 0 <= int_part <= 59 else str(int_part)
+    dec_words = " ".join(_num_to_words_de(int(d)) for d in dec_part)
+    return f"{int_word} komma {dec_words}"
+
+
 _UNIT_WORDS = [
     (r'\bTB\b', 'Terabyte'),
     (r'\bGB\b', 'Gigabyte'),
@@ -236,6 +245,11 @@ def clean_text_for_tts(text: str) -> str:
     # Kleine Zahlen-Bereiche ohne "Uhr": "5-6" → "fuenf bis sechs"
     t = _re_tts.sub(r'\b(\d{1,2})\s*[-–]\s*(\d{1,2})\b', _small_range_to_words, t)
 
+    # Dezimalzahlen: "0.1" / "0,5" / "1,25" → "null komma eins" / "null komma fuenf" / ...
+    # Muss vor "Zahl+Einheit" laufen, sonst frisst die Unit-Regel den Nachkommaanteil.
+    # Lookahead verhindert Match auf IP-artigen Strings wie 192.168.1.1.
+    t = _re_tts.sub(r'\b(\d+)[.,](\d+)(?![.,\d])', _decimal_to_words, t)
+
     # Zahlen + Einheit: "22GB" → "22 Gigabyte" (Leerzeichen einfuegen)
     t = _re_tts.sub(r'(\d+)([A-Za-z]{1,4})\b', r'\1 \2', t)
 

xtts/bridge.js

@@ -138,31 +138,43 @@ async function _runTTSRequest(payload) {
     let chunkIndex = 0;
     let pcmMeta = null;
 
-    // EIN Request fuer den GANZEN Text — kein Gap zwischen Saetzen.
-    // XTTS rendert und wir streamen PCM sobald es reinkommt.
-    await streamXTTSAsPCM(
-      cleanText,
-      language || "de",
-      hasCustomVoice ? voiceSample : null,
-      (pcmBase64, meta) => {
-        if (!pcmMeta) pcmMeta = meta;
-        sendToRVS({
-          type: "audio_pcm",
-          payload: {
-            requestId: requestId || "",
-            messageId: messageId || "",
-            base64: pcmBase64,
-            format: "pcm_s16le",
-            sampleRate: meta.sampleRate,
-            channels: meta.channels,
-            voice: voice || "default",
-            chunk: chunkIndex++,
-            final: false,
-          },
-          timestamp: Date.now(),
-        });
-      },
-    );
+    const onChunk = (pcmBase64, meta) => {
+      if (!pcmMeta) pcmMeta = meta;
+      sendToRVS({
+        type: "audio_pcm",
+        payload: {
+          requestId: requestId || "",
+          messageId: messageId || "",
+          base64: pcmBase64,
+          format: "pcm_s16le",
+          sampleRate: meta.sampleRate,
+          channels: meta.channels,
+          voice: voice || "default",
+          chunk: chunkIndex++,
+          final: false,
+        },
+        timestamp: Date.now(),
+      });
+    };
+
+    // /tts_stream fuer echtes Streaming (funktioniert im XTTS local-Mode).
+    // Wenn Server im apiManual/api-Mode laeuft: 400 → Fallback auf /tts_to_audio/.
+    try {
+      await streamXTTSAsPCM(
+        cleanText,
+        language || "de",
+        hasCustomVoice ? voiceSample : null,
+        onChunk,
+      );
+    } catch (streamErr) {
+      log(`/tts_stream fehlgeschlagen (${streamErr.message.slice(0, 100)}) — Fallback /tts_to_audio/`);
+      await streamXTTSBatch(
+        cleanText,
+        language || "de",
+        hasCustomVoice ? voiceSample : null,
+        onChunk,
+      );
+    }
 
     // Am Ende: final-Flag damit App weiss "fertig" und Cache geschrieben werden kann
     if (pcmMeta) {
@@ -195,42 +207,48 @@ async function _runTTSRequest(payload) {
 }
 
 /**
- * Ruft /tts_stream auf — echter Streaming-Endpoint von daswer123.
- * Chunked Transfer-Encoding: Samples kommen WAEHREND XTTS rendert,
- * nicht erst am Ende. Time-to-first-audio ~300-500ms statt 2-3s.
- *
- * stream_chunk_size = Characters pro Render-Iteration.
- * 40 = guter Kompromiss zwischen Schnellstart und GPU-Effizienz.
+ * Ruft /tts_stream auf — echter Streaming-Endpoint bei daswer123.
+ * Schickt was der Server verlangt (allow: GET), aber mit JSON-Body
+ * als POST scheitert mit 405. Manche Versionen wollen GET + Query,
+ * andere POST + JSON. Testen was funktioniert.
  */
 function streamXTTSAsPCM(text, language, speakerWav, onPcmChunk) {
   return new Promise((resolve, reject) => {
-    const body = JSON.stringify({
-      text,
-      language,
-      speaker_wav: speakerWav || "",
-      stream_chunk_size: 40,
-    });
+    // Wichtig: speaker_wav MUSS als Query-Key dabei sein (Pydantic required) —
+    // auch bei default-voice mit leerem Wert. Sonst gibt's HTTP 422.
+    // stream_chunk_size=100: Kompromiss zwischen first-audio-latency und
+    // gap-risk. Bei RTX 3060 (RTF 1.48) ~3s bis erster Audio, Chunks gross
+    // genug dass der AudioTrack-Buffer (128KB ≈ 2.7s) zwischen Chunks nicht
+    // leerlauft.
+    const qs = new URLSearchParams();
+    qs.set("text", text);
+    qs.set("language", language || "de");
+    qs.set("speaker_wav", speakerWav || "");
+    qs.set("stream_chunk_size", "100");
 
-    const url = new URL(`${XTTS_API_URL}/tts_stream`);
+    const url = new URL(XTTS_API_URL);
+    const fullPath = `/tts_stream?${qs.toString()}`;
     const options = {
       hostname: url.hostname,
-      port: url.port,
-      path: url.pathname,
-      method: "POST",
-      headers: {
-        "Content-Type": "application/json",
-        "Content-Length": Buffer.byteLength(body),
-      },
+      port: url.port || 80,
+      path: fullPath,
+      method: "GET",
       timeout: 60000,
     };
 
+    log(`TTS GET /tts_stream?text=${text.slice(0, 30)}... (voice=${speakerWav ? "custom" : "default"})`);
+
     const req = http.request(options, (res) => {
       if (res.statusCode !== 200) {
         let body = "";
         res.on("data", (d) => { body += d.toString(); });
-        res.on("end", () => reject(new Error(`XTTS HTTP ${res.statusCode}: ${body.slice(0, 200)}`)));
+        res.on("end", () => {
+          log(`XTTS /tts_stream ${res.statusCode}: ${body.slice(0, 300)}`);
+          reject(new Error(`XTTS HTTP ${res.statusCode}: ${body.slice(0, 200)}`));
+        });
         return;
       }
+      log(`TTS stream verbunden, empfange PCM...`);
 
       let headerParsed = false;
       let sampleRate = 24000;
@@ -282,6 +300,76 @@ function streamXTTSAsPCM(text, language, speakerWav, onPcmChunk) {
 
     req.on("error", reject);
     req.on("timeout", () => { req.destroy(); reject(new Error("XTTS API Timeout (60s)")); });
+    req.end();
+  });
+}
+
+/**
+ * Fallback: /tts_to_audio/ (POST JSON) — rendert komplett, dann response.
+ * Kein echtes Streaming, aber stabil als Backup wenn /tts_stream nicht geht.
+ * Shared chunking-Logik mit streamXTTSAsPCM — parst WAV-Header, stueckelt PCM.
+ */
+function streamXTTSBatch(text, language, speakerWav, onPcmChunk) {
+  return new Promise((resolve, reject) => {
+    const body = JSON.stringify({
+      text,
+      language: language || "de",
+      speaker_wav: speakerWav || "",
+    });
+    const url = new URL(XTTS_API_URL);
+    const options = {
+      hostname: url.hostname,
+      port: url.port || 80,
+      path: "/tts_to_audio/",
+      method: "POST",
+      headers: {
+        "Content-Type": "application/json",
+        "Content-Length": Buffer.byteLength(body),
+      },
+      timeout: 60000,
+    };
+
+    const req = http.request(options, (res) => {
+      if (res.statusCode !== 200) {
+        let rb = "";
+        res.on("data", (d) => { rb += d.toString(); });
+        res.on("end", () => reject(new Error(`XTTS Batch HTTP ${res.statusCode}: ${rb.slice(0, 200)}`)));
+        return;
+      }
+      let headerParsed = false;
+      let sampleRate = 24000;
+      let channels = 1;
+      let leftover = Buffer.alloc(0);
+      let headerBuf = Buffer.alloc(0);
+      const HEADER_BYTES = 44;
+      const PCM_CHUNK_BYTES = 8192;
+
+      res.on("data", (chunk) => {
+        let data = chunk;
+        if (!headerParsed) {
+          headerBuf = Buffer.concat([headerBuf, data]);
+          if (headerBuf.length < HEADER_BYTES) return;
+          const header = headerBuf.slice(0, HEADER_BYTES);
+          try { channels = header.readUInt16LE(22); sampleRate = header.readUInt32LE(24); } catch (_) {}
+          headerParsed = true;
+          data = headerBuf.slice(HEADER_BYTES);
+        }
+        let combined = Buffer.concat([leftover, data]);
+        while (combined.length >= PCM_CHUNK_BYTES) {
+          const slice = combined.slice(0, PCM_CHUNK_BYTES);
+          combined = combined.slice(PCM_CHUNK_BYTES);
+          onPcmChunk(slice.toString("base64"), { sampleRate, channels });
+        }
+        leftover = combined;
+      });
+      res.on("end", () => {
+        if (leftover.length > 0) onPcmChunk(leftover.toString("base64"), { sampleRate, channels });
+        resolve();
+      });
+      res.on("error", reject);
+    });
+    req.on("error", reject);
+    req.on("timeout", () => { req.destroy(); reject(new Error("XTTS Batch Timeout (60s)")); });
     req.write(body);
     req.end();
   });

xtts/docker-compose.yml

@@ -33,6 +33,12 @@ services:
       - ./voices:/voices                        # Custom Voice Samples
     environment:
       - COQUI_TOS_AGREED=1
+      # Local-Modus statt default "apiManual": Modell bleibt im GPU-VRAM,
+      # Render startet sofort, /tts_stream funktioniert.
+      # Default-CMD des Images liest diese ENV: -ms ${MODEL_SOURCE:-"apiManual"}
+      - MODEL_SOURCE=local
+      # Speaker-Folder auf unsere gemounteten voices zeigen lassen
+      - EXAMPLE_FOLDER=/voices
     restart: unless-stopped
 
   # ─── XTTS Bridge (verbindet zu RVS) ───────────