fix(hermes-voice): finalize mobile audio and reduce latency

dockerfiles/hermes-webui-atlas-voice.js

@@ -223,8 +223,6 @@
       const mimeTypes=['audio/webm;codecs=opus','audio/ogg;codecs=opus','audio/mp4;codecs=mp4a.40.2','audio/mp4','audio/webm'];
       const mime=mimeTypes.find(function(value){return MediaRecorder.isTypeSupported(value);})||'';
       const chunks=[];
-      const preRoll=[];
-      let initialChunk=null;
       let heardSpeech=false;
       let voiceFrames=0;
       let noiseFloor=0.008;
@@ -235,13 +233,7 @@
       recorder.ondataavailable=function(event){
         if(!event.data||!event.data.size) return;
         if(event.data.type) recordedMime=event.data.type;
-        if(heardSpeech){chunks.push(event.data);return;}
-        // MediaRecorder's first timeslice owns the container initialization
-        // (EBML/Opus headers for WebM, and equivalent headers for Ogg/MP4).
-        // Keep it separately while bounding the actual audio pre-roll.
-        if(!initialChunk){initialChunk=event.data;return;}
-        preRoll.push(event.data);
-        while(preRoll.length>3) preRoll.shift();
+        chunks.push(event.data);
       };
       recorder.onstop=function(){
         if(vadTimer){clearInterval(vadTimer);vadTimer=null;}
@@ -254,7 +246,12 @@
         if(!heardSpeech||!chunks.length){restartSoon(token,300);return;}
         transcribe(new Blob(chunks,{type:recordedMime||'audio/webm'}),token);
       };
-      recorder.start(250);
+      // Ask the browser for one finalized container at stop. Android Chromium
+      // can emit timeslice fragments without a reusable EBML initialization
+      // header; concatenating those fragments made otherwise valid recordings
+      // intermittently unreadable by ffmpeg. A bounded 90-second Opus capture
+      // is small enough to retain as one browser-owned recording.
+      recorder.start();
       const silenceMs=Math.max(900,parseInt(localStorage.getItem('hermes-voice-silence-ms')||'1600',10)||1600);
       vadTimer=window.setInterval(function(){
         if(!active||token!==generation||!recorder||recorder.state==='inactive') return;
@@ -274,8 +271,6 @@
         if(!heardSpeech&&voiceFrames>=3){
           heardSpeech=true;
           lastSpeech=now;
-          if(initialChunk){chunks.push(initialChunk);initialChunk=null;}
-          while(preRoll.length) chunks.push(preRoll.shift());
         }else if(heardSpeech&&voiceNow){
           lastSpeech=now;
         }
@@ -352,7 +347,9 @@
     const text=cleanForSpeech(rows[rows.length-1].dataset.rawText||'');
     if(!text){restartSoon(token,250);return;}
     setState('speaking');
-    const chunks=typeof window._splitForTTS==='function'?window._splitForTTS(text,280):[text];
+    // Shorter first chunks lower time-to-first-audio; the next chunk is
+    // synthesized while the current one plays.
+    const chunks=typeof window._splitForTTS==='function'?window._splitForTTS(text,160):[text];
     try{
       let pending=fetchSpeech(chunks[0],language);
       for(let index=0;index<chunks.length;index+=1){

services/hermes/voice-deployment.yaml

@@ -132,7 +132,7 @@ spec:
             - {name: XDG_CACHE_HOME, value: /tmp/cache}
             - {name: HERMES_TTS_PORT, value: "9001"}
             - {name: HERMES_TTS_CACHE, value: /opt/models/piper}
-            - {name: HERMES_TTS_ONNX_THREADS, value: "2"}
+            - {name: HERMES_TTS_ONNX_THREADS, value: "4"}
           startupProbe:
             httpGet: {path: /health, port: http}
             periodSeconds: 5

testing/probes/hermes_voice_instrument_probe.js

@@ -310,10 +310,7 @@ async function normalMotionContract(scriptPath, mediaFixture) {
   assert.ok(probe.uploadedFile, 'transcription upload was not captured');
   assert.equal(probe.uploadedFile.name, 'voice-input.webm');
   assert.equal(probe.uploadedFile.type, mediaFixture.mime_type);
-  const expectedUpload = Buffer.concat([
-    mediaChunks[0],
-    ...mediaChunks.slice(preSpeechCount - 3),
-  ]);
+  const expectedUpload = Buffer.concat(mediaChunks);
   assert.deepEqual(Buffer.from(probe.uploadedFile.bytes), expectedUpload);
 
   probe.resolveTranscript({transcript: 'Hello Hermes'});

testing/tests/test_hermes_chat_quality.py

@@ -342,13 +342,14 @@ def test_chat_voice_uses_private_jetson_services_and_shared_auto_route():
     assert "/api/transcribe" in voice_script
     assert "/api/tts" in voice_script
     assert "speakResponse(generation)" in voice_script
-    assert "window._splitForTTS(text,280)" in voice_script
+    assert "window._splitForTTS(text,160)" in voice_script
     assert "pending=fetchSpeech(chunks[index+1],language)" in voice_script
     assert "restartSoon(token,450)" in voice_script
     assert "constraints.voiceIsolation=true" in voice_script
     assert "highpass.frequency.value=140" in voice_script
     assert "Math.max(0.04,noiseFloor*2.4+0.006)" in voice_script
-    assert "while(preRoll.length>3) preRoll.shift()" in voice_script
+    assert "recorder.start();" in voice_script
+    assert "recorder.start(250)" not in voice_script
 
     stt_server = (ROOT / "dockerfiles" / "hermes-jetson-stt-server.py").read_text()
     assert "def _repetitive_token" in stt_server
@@ -466,7 +467,7 @@ def test_voice_workloads_have_deliberate_xavier_placement():
     # Keep the deployment neutral so the old and new immutable images can
     # cross one Flux rollout safely; each image owns its compatible default.
     assert "HERMES_TTS_VOICE" not in tts_env
-    assert tts_env["HERMES_TTS_ONNX_THREADS"] == "2"
+    assert tts_env["HERMES_TTS_ONNX_THREADS"] == "4"
     assert tts["containers"][0]["resources"]["limits"]["cpu"] == "4"
     assert all("hostPath" not in volume for volume in stt["volumes"])
     assert all("hostPath" not in volume for volume in tts["volumes"])

testing/tests/test_hermes_handsfree_stt.py

@@ -92,14 +92,15 @@ def test_fixture_reproduces_the_current_ebml_header_failure(tmp_path: Path):
     assert "EBML header" in result.stderr
 
 
-def test_header_preserved_browser_webm_reaches_real_conversion_boundary(
+def test_complete_browser_recording_reaches_real_conversion_boundary(
     tmp_path: Path,
 ):
     module = _patched_transcription_module(tmp_path)
+    _fixture, chunks = _media_chunks()
     prepared, error = _convert(
         module,
         tmp_path,
-        _late_speech_upload(preserve_header=True),
+        b"".join(chunks),
         ".webm",
     )
 
@@ -164,8 +165,11 @@ def test_local_conversion_accepts_browser_fallback_containers(tmp_path: Path):
 def test_browser_uses_actual_recorder_mime_and_supported_extensions():
     source = VOICE_JS.read_text(encoding="utf-8")
 
-    assert "let initialChunk=null" in source
-    assert "if(initialChunk){chunks.push(initialChunk);initialChunk=null;}" in source
+    assert "recorder.start();" in source
+    assert "recorder.start(250)" not in source
+    assert "chunks.push(event.data)" in source
+    assert "initialChunk" not in source
+    assert "preRoll" not in source
     assert "let recordedMime=recorder.mimeType||mime||''" in source
     assert "if(event.data.type) recordedMime=event.data.type" in source
     assert "audio/mp4;codecs=mp4a.40.2" in source