dockerfiles/Dockerfile.hermes-webui

@@ -90,12 +90,14 @@ PY
 # Add the Atlas voice bridge as a narrow integration layer. It activates only
 # when a tenant's server-side STT capability reports the private Jetson route.
 COPY dockerfiles/hermes-webui-atlas-patch.py /tmp/hermes-webui-atlas-patch.py
+COPY dockerfiles/hermes-webui-stt-patch.py /tmp/hermes-webui-stt-patch.py
 COPY dockerfiles/hermes-webui-telegram-project-patch.py /tmp/hermes-webui-telegram-project-patch.py
 COPY dockerfiles/hermes-webui-atlas-voice.js /opt/hermes-webui/static/atlas-voice.js
 COPY dockerfiles/hermes-webui-atlas-voice.css /opt/hermes-webui/static/atlas-voice.css
 COPY dockerfiles/hermes-webui-router-patch.py /tmp/hermes-webui-router-patch.py
 COPY dockerfiles/hermes-webui-router.js /opt/hermes-webui/static/atlas-router.js
 RUN /opt/hermes/.venv/bin/python /tmp/hermes-webui-atlas-patch.py
+RUN /opt/hermes/.venv/bin/python /tmp/hermes-webui-stt-patch.py
 RUN /opt/hermes/.venv/bin/python /tmp/hermes-webui-telegram-project-patch.py
 RUN /opt/hermes/.venv/bin/python /tmp/hermes-webui-router-patch.py
 
@@ -109,6 +111,7 @@ RUN /opt/hermes/.venv/bin/python -c 'import cryptography, yaml' \
     && grep -Fq "'atlas/auto/maximum': 'Automatic · Maximum'" /opt/hermes-webui/static/panels.js \
     && grep -Fq 'Atlas Jetson (private)' /opt/hermes-webui/static/index.html \
     && grep -Fq 'HERMES_WEBUI_ATLAS_TTS_URL' /opt/hermes-webui/api/routes.py \
+    && grep -Fq 'Audio conversion failed: upload is invalid' /opt/hermes/tools/transcription_tools.py \
     && grep -Fq "capability.provider!=='local_command'" /opt/hermes-webui/static/atlas-voice.js \
     && grep -Fq 'prefers-reduced-motion: reduce' /opt/hermes-webui/static/atlas-voice.css \
     && grep -Fq 'id="voiceInstrumentStyles"' /opt/hermes-webui/static/index.html \
@@ -118,7 +121,8 @@ RUN /opt/hermes/.venv/bin/python -c 'import cryptography, yaml' \
     && grep -Fq 'explicit_reasoning_effort' /opt/hermes-webui/api/gateway_chat.py \
     && /opt/hermes/.venv/bin/python -m py_compile \
         /opt/hermes-webui/api/routes.py \
-        /opt/hermes-webui/api/gateway_chat.py
+        /opt/hermes-webui/api/gateway_chat.py \
+        /opt/hermes/tools/transcription_tools.py
 
 # Exercise the real server process in the target architecture before publish.
 RUN set -eu; \

dockerfiles/hermes-webui-atlas-voice.js

@@ -131,10 +131,17 @@
     if(typeof window.send==='function') window.send();
   }
 
+  function audioExtension(mimeType){
+    const normalized=String(mimeType||'').toLowerCase();
+    if(normalized.indexOf('ogg')>=0) return 'ogg';
+    if(normalized.indexOf('mp4')>=0) return 'mp4';
+    return 'webm';
+  }
+
   async function transcribe(blob, token){
     if(!active||token!==generation) return;
     setState('transcribing');
-    const ext=(blob.type||'').indexOf('ogg')>=0?'ogg':'webm';
+    const ext=audioExtension(blob.type);
     const form=new FormData();
     form.append('file',new File([blob],'voice-input.'+ext,{type:blob.type||'audio/'+ext}));
     try{
@@ -182,19 +189,26 @@
       audioContext.createMediaStreamSource(stream).connect(highpass);
       highpass.connect(analyser);
       const samples=new Uint8Array(analyser.fftSize);
-      const mimeTypes=['audio/webm;codecs=opus','audio/ogg;codecs=opus','audio/webm'];
+      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;
       let lastSpeech=Date.now();
       const started=Date.now();
       recorder=new MediaRecorder(stream,mime?{mimeType:mime}:undefined);
+      let recordedMime=recorder.mimeType||mime||'';
       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();
       };
@@ -207,7 +221,7 @@
         recorder=null;
         if(!active||token!==generation) return;
         if(!heardSpeech||!chunks.length){restartSoon(token,300);return;}
-        transcribe(new Blob(chunks,{type:mime||'audio/webm'}),token);
+        transcribe(new Blob(chunks,{type:recordedMime||'audio/webm'}),token);
       };
       recorder.start(250);
       const silenceMs=Math.max(900,parseInt(localStorage.getItem('hermes-voice-silence-ms')||'1600',10)||1600);
@@ -229,6 +243,7 @@
         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;

dockerfiles/hermes-webui-stt-patch.py

@@ -0,0 +1,37 @@
+#!/usr/bin/env python3
+"""Bound local-STT conversion errors in the pinned Hermes Agent source."""
+
+import os
+from pathlib import Path
+
+
+ROOT = Path(os.environ.get("HERMES_AGENT_PATCH_ROOT", "/opt/hermes"))
+
+
+def replace_exact(path: Path, before: str, after: str) -> None:
+    """Replace one exact upstream fragment, failing closed on image drift."""
+    source = path.read_text(encoding="utf-8")
+    if source.count(before) != 1:
+        raise SystemExit(f"Hermes STT patch context changed in {path}: {before[:80]!r}")
+    path.write_text(source.replace(before, after, 1), encoding="utf-8")
+
+
+transcription = ROOT / "tools/transcription_tools.py"
+replace_exact(
+    transcription,
+    """    except subprocess.CalledProcessError as e:
+        details = e.stderr.strip() or e.stdout.strip() or str(e)
+        logger.error("ffmpeg conversion failed for %s: %s", file_path, details)
+        return None, f"Failed to convert audio for local STT: {details}"
+""",
+    """    except subprocess.CalledProcessError as e:
+        details = e.stderr.strip() or e.stdout.strip() or str(e)
+        logger.error(
+            "ffmpeg conversion failed for %s: %s", file_path, details[-2000:]
+        )
+        return None, (
+            "Audio conversion failed: upload is invalid, incomplete, or uses an "
+            "unsupported codec"
+        )
+""",
+)

testing/fixtures/hermes-agent-9de9c25f/SOURCE.md

@@ -0,0 +1,12 @@
+# Hermes Agent STT fixture provenance
+
+This minimal module contains the exact `_prepare_local_audio` patch context
+extracted from the agent image inherited by `Dockerfile.hermes-webui`:
+
+- Image: `registry.bstein.dev/bstein/hermes-agent@sha256:81970563e542f0720773e72297810b3a844b83e381e278f25c0916c78d930107`
+- OCI platform manifest: `sha256:9b4c00a25fd08f0df3bcb800d755bd48576fb65af5ab0df40058fbb78a147e43`
+- Upstream Hermes revision: `9de9c25f620ff7f1ce0fd5457d596052d5159596`
+- Full upstream `tools/transcription_tools.py` SHA-256: `d7d7df56b98dfadc0a7c08d5addd02db4e3106e6fd3b753e9241c23145eb2951`
+
+Tests execute the production patcher against this source fragment and then run
+its real ffmpeg conversion boundary with browser-produced bytes.

testing/fixtures/hermes-agent-9de9c25f/tools/transcription_tools.py

@@ -0,0 +1,44 @@
+"""Exact pinned local-audio preparation fragment with its direct dependencies."""
+
+import os
+from pathlib import Path
+import shutil
+import subprocess
+from typing import Optional
+
+
+LOCAL_NATIVE_AUDIO_FORMATS = {".wav", ".aiff", ".aif"}
+logger = __import__("logging").getLogger(__name__)
+
+
+def _find_ffmpeg_binary() -> Optional[str]:
+    return shutil.which("ffmpeg")
+
+
+def windows_hide_flags() -> int:
+    return 0
+
+
+def _prepare_local_audio(file_path: str, work_dir: str) -> tuple[Optional[str], Optional[str]]:
+    """Normalize audio for local CLI STT when needed."""
+    audio_path = Path(file_path)
+    if audio_path.suffix.lower() in LOCAL_NATIVE_AUDIO_FORMATS:
+        return file_path, None
+
+    ffmpeg = _find_ffmpeg_binary()
+    if not ffmpeg:
+        return None, "Local STT fallback requires ffmpeg for non-WAV inputs, but ffmpeg was not found"
+
+    converted_path = os.path.join(work_dir, f"{audio_path.stem}.wav")
+    command = [ffmpeg, "-y", "-i", file_path, converted_path]
+
+    try:
+        subprocess.run(command, check=True, capture_output=True, text=True, timeout=300, stdin=subprocess.DEVNULL, creationflags=windows_hide_flags())
+        return converted_path, None
+    except subprocess.TimeoutExpired:
+        logger.error("ffmpeg conversion timed out for %s", file_path)
+        return None, "Audio conversion for local STT timed out"
+    except subprocess.CalledProcessError as e:
+        details = e.stderr.strip() or e.stdout.strip() or str(e)
+        logger.error("ffmpeg conversion failed for %s: %s", file_path, details)
+        return None, f"Failed to convert audio for local STT: {details}"

testing/fixtures/mediarecorder/SOURCE.md

@@ -0,0 +1,18 @@
+# Chromium MediaRecorder fixture provenance
+
+`chromium-webm-opus.webm` is the concatenation of ten successive 250 ms-ish
+data blobs emitted by a real Chromium `MediaRecorder` for
+`audio/webm;codecs=opus`; `chromium-webm-opus.json` records their exact byte
+lengths so tests can replay each browser event. The recording has leading
+silence followed by a 440 Hz signal, mirroring speech that begins after the
+hands-free VAD pre-roll is full.
+
+- Browser: Chromium 140.0.7339.16 (Playwright build v1187, arm64)
+- Generator: `testing/probes/generate_mediarecorder_fixture.js`
+- Timeslice request: 250 ms
+- Complete concatenated payload: WebM/Opus, accepted by ffmpeg 7.1.5
+- Complete payload SHA-256: `5585014ecad185511a727ccf47aa8b1ac70ea6f3efd7778309742bf88e6faf51`
+- First five chunks are treated as pre-speech by the test model; the production
+  pre-roll retains only the most recent three.
+
+The fixture is generated data, not recorded speech.

testing/fixtures/mediarecorder/chromium-webm-opus.json

@@ -0,0 +1,20 @@
+{
+  "mime_type": "audio/webm;codecs=opus",
+  "timeslice_ms": 250,
+  "pre_speech_chunk_count": 5,
+  "chunk_sizes": [
+    204,
+    70,
+    70,
+    70,
+    70,
+    3864,
+    5796,
+    4830,
+    4830,
+    1945
+  ],
+  "payload_file": "chromium-webm-opus.webm",
+  "browser": "140.0.7339.16",
+  "generator": "testing/probes/generate_mediarecorder_fixture.js"
+}

testing/probes/generate_mediarecorder_fixture.js

@@ -0,0 +1,84 @@
+'use strict';
+
+const fs = require('node:fs');
+const path = require('node:path');
+const { chromium } = require('playwright');
+
+async function main() {
+  const outputPath = process.argv[2];
+  if (!outputPath) throw new Error('usage: node generate_mediarecorder_fixture.js OUTPUT.json');
+
+  const browser = await chromium.launch({
+    headless: true,
+    args: ['--autoplay-policy=no-user-gesture-required'],
+  });
+  try {
+    const page = await browser.newPage();
+    const fixture = await page.evaluate(async () => {
+      const mimeType = 'audio/webm;codecs=opus';
+      if (!MediaRecorder.isTypeSupported(mimeType)) {
+        throw new Error(`${mimeType} is not supported by this Chromium build`);
+      }
+
+      const context = new AudioContext({sampleRate: 48000});
+      const destination = context.createMediaStreamDestination();
+      const oscillator = context.createOscillator();
+      const gain = context.createGain();
+      oscillator.frequency.value = 440;
+      gain.gain.value = 0;
+      oscillator.connect(gain).connect(destination);
+      oscillator.start();
+
+      const chunks = [];
+      const recorder = new MediaRecorder(destination.stream, {mimeType});
+      recorder.ondataavailable = event => {
+        if (event.data && event.data.size) chunks.push(event.data);
+      };
+      const stopped = new Promise(resolve => { recorder.onstop = resolve; });
+      const delay = milliseconds => new Promise(resolve => setTimeout(resolve, milliseconds));
+
+      recorder.start(250);
+      await delay(1400);
+      gain.gain.setValueAtTime(0.35, context.currentTime);
+      await delay(900);
+      gain.gain.setValueAtTime(0, context.currentTime);
+      await delay(500);
+      recorder.stop();
+      await stopped;
+
+      oscillator.stop();
+      await context.close();
+      const encoded = [];
+      for (const chunk of chunks) {
+        const bytes = new Uint8Array(await chunk.arrayBuffer());
+        let binary = '';
+        for (let index = 0; index < bytes.length; index += 0x8000) {
+          binary += String.fromCharCode(...bytes.subarray(index, index + 0x8000));
+        }
+        encoded.push(btoa(binary));
+      }
+      return {
+        mime_type: recorder.mimeType,
+        timeslice_ms: 250,
+        pre_speech_chunk_count: 5,
+        chunks_base64: encoded,
+      };
+    });
+    const chunks = fixture.chunks_base64.map(value => Buffer.from(value, 'base64'));
+    const payloadPath = outputPath.replace(/\.json$/i, '.webm');
+    fixture.chunk_sizes = chunks.map(chunk => chunk.length);
+    fixture.payload_file = path.basename(payloadPath);
+    delete fixture.chunks_base64;
+    fixture.browser = await browser.version();
+    fixture.generator = 'testing/probes/generate_mediarecorder_fixture.js';
+    fs.writeFileSync(payloadPath, Buffer.concat(chunks));
+    fs.writeFileSync(outputPath, `${JSON.stringify(fixture, null, 2)}\n`);
+  } finally {
+    await browser.close();
+  }
+}
+
+main().catch(error => {
+  console.error(error.stack || error);
+  process.exitCode = 1;
+});

testing/probes/hermes_voice_instrument_probe.js

@@ -2,6 +2,7 @@
 
 const assert = require('node:assert/strict');
 const fs = require('node:fs');
+const path = require('node:path');
 const vm = require('node:vm');
 
 class StyleDeclaration {
@@ -92,6 +93,7 @@ async function boot(scriptPath, reduced) {
   const reducedMotion = {matches: reduced, addEventListener() {}, removeEventListener() {}};
   let rejectNextCapture = false;
   let transcriptResolve;
+  let uploadedFile = null;
   let sent = 0;
   let lastAudio = null;
 
@@ -117,9 +119,10 @@ async function boot(scriptPath, reduced) {
   }
   class FakeMediaRecorder {
     static isTypeSupported() { return true; }
-    constructor(stream) {
+    constructor(stream, options) {
       this.stream = stream;
       this.state = 'inactive';
+      this.mimeType = options?.mimeType || 'audio/webm;codecs=opus';
       this.ondataavailable = null;
       this.onstop = null;
       recorders.push(this);
@@ -171,11 +174,18 @@ async function boot(scriptPath, reduced) {
       },
     },
   };
-  async function fetch(url) {
+  async function fetch(url, options) {
     if (url === '/api/transcribe/capability') {
       return {ok: true, json: async () => ({available: true, provider: 'local_command'})};
     }
     if (url === '/api/transcribe') {
+      const file = options?.body?.get('file');
+      assert.ok(file, 'transcription request did not carry a file');
+      uploadedFile = {
+        bytes: new Uint8Array(await file.arrayBuffer()),
+        name: file.name,
+        type: file.type,
+      };
       return new Promise(resolve => { transcriptResolve = resolve; });
     }
     if (url === '/api/tts') {
@@ -245,6 +255,7 @@ async function boot(scriptPath, reduced) {
     captures, recorders, analysers, assistantRows, toasts, reducedMotion,
     get sent() { return sent; },
     get lastAudio() { return lastAudio; },
+    get uploadedFile() { return uploadedFile; },
     set now(value) { now = value; },
     rejectCapture() { rejectNextCapture = true; },
     resolveTranscript(payload) {
@@ -261,7 +272,7 @@ async function boot(scriptPath, reduced) {
   };
 }
 
-async function normalMotionContract(scriptPath) {
+async function normalMotionContract(scriptPath, mediaFixture) {
   const probe = await boot(scriptPath, false);
   const originalStyleLink = probe.styleLink;
 
@@ -272,17 +283,38 @@ async function normalMotionContract(scriptPath) {
   await flush();
   assert.equal(probe.captures.length, 1);
 
+  const mediaChunks = mediaFixture.chunks;
+  const preSpeechCount = mediaFixture.pre_speech_chunk_count;
+  for (const chunk of mediaChunks.slice(0, preSpeechCount)) {
+    probe.recorders[0].ondataavailable({
+      data: new Blob([chunk], {type: mediaFixture.mime_type}),
+    });
+  }
+
   probe.analysers[0].level = 0.3;
   probe.runIntervals();
   probe.runIntervals();
   probe.runIntervals();
-  probe.recorders[0].ondataavailable({data: new Blob(['speech'], {type: 'audio/webm'})});
+  for (const chunk of mediaChunks.slice(preSpeechCount)) {
+    probe.recorders[0].ondataavailable({
+      data: new Blob([chunk], {type: mediaFixture.mime_type}),
+    });
+  }
   probe.now = 4000;
   probe.analysers[0].level = 0;
   probe.runIntervals();
   await flush();
+  await flush();
   assert.match(probe.indicator.className, /\btranscribing\b/);
   assert.equal(probe.label.textContent, 'Transcribing…');
+  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),
+  ]);
+  assert.deepEqual(Buffer.from(probe.uploadedFile.bytes), expectedUpload);
 
   probe.resolveTranscript({transcript: 'Hello Hermes'});
   await flush();
@@ -310,8 +342,21 @@ async function normalMotionContract(scriptPath) {
 
 async function main() {
   const scriptPath = process.argv[2];
-  assert.ok(scriptPath, 'usage: node hermes_voice_instrument_probe.js <atlas-voice.js>');
-  const first = await normalMotionContract(scriptPath);
+  const fixturePath = process.argv[3];
+  assert.ok(
+    scriptPath && fixturePath,
+    'usage: node hermes_voice_instrument_probe.js <atlas-voice.js> <media-fixture.json>',
+  );
+  const mediaFixture = JSON.parse(fs.readFileSync(fixturePath, 'utf8'));
+  const payload = fs.readFileSync(path.join(path.dirname(fixturePath), mediaFixture.payload_file));
+  let offset = 0;
+  mediaFixture.chunks = mediaFixture.chunk_sizes.map(size => {
+    const chunk = payload.subarray(offset, offset + size);
+    offset += size;
+    return chunk;
+  });
+  assert.equal(offset, payload.length, 'fixture chunk sizes do not cover the payload');
+  const first = await normalMotionContract(scriptPath, mediaFixture);
   const {probe, originalStyleLink, capturesBeforeSpeech} = first;
 
   assert.equal(probe.captures.length, capturesBeforeSpeech);

testing/tests/test_hermes_handsfree_stt.py

@@ -0,0 +1,203 @@
+"""Browser-to-ffmpeg contracts for Hermes hands-free transcription uploads."""
+
+from __future__ import annotations
+
+import importlib.util
+import json
+import os
+from pathlib import Path
+import shutil
+import subprocess
+import sys
+
+
+ROOT = Path(__file__).resolve().parents[2]
+VOICE_JS = ROOT / "dockerfiles/hermes-webui-atlas-voice.js"
+STT_PATCHER = ROOT / "dockerfiles/hermes-webui-stt-patch.py"
+AGENT_FIXTURE = ROOT / "testing/fixtures/hermes-agent-9de9c25f"
+MEDIA_FIXTURE = ROOT / "testing/fixtures/mediarecorder/chromium-webm-opus.json"
+SAFE_CONVERSION_ERROR = (
+    "Audio conversion failed: upload is invalid, incomplete, or uses an "
+    "unsupported codec"
+)
+
+
+def _media_chunks() -> tuple[dict[str, object], list[bytes]]:
+    fixture = json.loads(MEDIA_FIXTURE.read_text(encoding="utf-8"))
+    payload = MEDIA_FIXTURE.with_name(str(fixture["payload_file"])).read_bytes()
+    chunks = []
+    offset = 0
+    for size in fixture["chunk_sizes"]:
+        end = offset + int(size)
+        chunks.append(payload[offset:end])
+        offset = end
+    assert offset == len(payload)
+    return fixture, chunks
+
+
+def _late_speech_upload(*, preserve_header: bool) -> bytes:
+    fixture, chunks = _media_chunks()
+    count = int(fixture["pre_speech_chunk_count"])
+    retained = chunks[count - 3 :]
+    if preserve_header:
+        retained.insert(0, chunks[0])
+    return b"".join(retained)
+
+
+def _patched_transcription_module(tmp_path: Path):
+    target = tmp_path / "hermes-agent"
+    shutil.copytree(AGENT_FIXTURE, target)
+    env = os.environ.copy()
+    env["HERMES_AGENT_PATCH_ROOT"] = str(target)
+    subprocess.run(
+        [sys.executable, str(STT_PATCHER)],
+        cwd=ROOT,
+        env=env,
+        check=True,
+        capture_output=True,
+        text=True,
+    )
+    module_path = target / "tools/transcription_tools.py"
+    spec = importlib.util.spec_from_file_location(
+        "pinned_transcription_tools", module_path
+    )
+    assert spec and spec.loader
+    module = importlib.util.module_from_spec(spec)
+    spec.loader.exec_module(module)
+    return module
+
+
+def _convert(module, tmp_path: Path, payload: bytes, suffix: str):
+    source = tmp_path / f"voice-input{suffix}"
+    output_dir = tmp_path / f"converted-{suffix.removeprefix('.')}"
+    output_dir.mkdir()
+    source.write_bytes(payload)
+    return module._prepare_local_audio(str(source), str(output_dir))
+
+
+def test_fixture_reproduces_the_current_ebml_header_failure(tmp_path: Path):
+    """The red control is browser data after the old sliding pre-roll dropped chunk 0."""
+    broken = tmp_path / "current-late-speech.webm"
+    broken.write_bytes(_late_speech_upload(preserve_header=False))
+
+    result = subprocess.run(
+        ["ffmpeg", "-v", "error", "-y", "-i", str(broken), str(tmp_path / "bad.wav")],
+        check=False,
+        capture_output=True,
+        text=True,
+        timeout=15,
+    )
+
+    assert result.returncode != 0
+    assert "EBML header" in result.stderr
+
+
+def test_header_preserved_browser_webm_reaches_real_conversion_boundary(
+    tmp_path: Path,
+):
+    module = _patched_transcription_module(tmp_path)
+    prepared, error = _convert(
+        module,
+        tmp_path,
+        _late_speech_upload(preserve_header=True),
+        ".webm",
+    )
+
+    assert error is None
+    assert prepared is not None
+    output = Path(prepared)
+    assert output.is_file()
+    assert output.read_bytes().startswith(b"RIFF")
+
+
+def test_invalid_browser_upload_returns_bounded_error_without_ffmpeg_spam(
+    tmp_path: Path,
+):
+    module = _patched_transcription_module(tmp_path)
+    prepared, error = _convert(
+        module,
+        tmp_path,
+        _late_speech_upload(preserve_header=False),
+        ".webm",
+    )
+
+    assert prepared is None
+    assert error == SAFE_CONVERSION_ERROR
+    assert len(error) < 128
+    lowered = error.lower()
+    assert "ffmpeg version" not in lowered
+    assert "configuration:" not in lowered
+    assert "/tmp/" not in lowered
+
+
+def test_local_conversion_accepts_browser_fallback_containers(tmp_path: Path):
+    module = _patched_transcription_module(tmp_path)
+    formats = ((".ogg", "libopus"), (".mp4", "aac"))
+
+    for suffix, codec in formats:
+        source = tmp_path / f"source{suffix}"
+        subprocess.run(
+            [
+                "ffmpeg",
+                "-v",
+                "error",
+                "-y",
+                "-f",
+                "lavfi",
+                "-i",
+                "sine=frequency=440:duration=0.4",
+                "-c:a",
+                codec,
+                str(source),
+            ],
+            check=True,
+            capture_output=True,
+            text=True,
+            timeout=15,
+        )
+        prepared, error = _convert(module, tmp_path, source.read_bytes(), suffix)
+        assert error is None
+        assert prepared is not None
+        assert Path(prepared).read_bytes().startswith(b"RIFF")
+
+
+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 "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
+    assert "if(normalized.indexOf('ogg')>=0) return 'ogg'" in source
+    assert "if(normalized.indexOf('mp4')>=0) return 'mp4'" in source
+
+
+def test_stt_patch_is_built_fail_closed_into_the_webui_image(tmp_path: Path):
+    dockerfile = (ROOT / "dockerfiles/Dockerfile.hermes-webui").read_text(
+        encoding="utf-8"
+    )
+    assert "COPY dockerfiles/hermes-webui-stt-patch.py" in dockerfile
+    assert "python /tmp/hermes-webui-stt-patch.py" in dockerfile
+    assert "/opt/hermes/tools/transcription_tools.py" in dockerfile
+
+    drifted = tmp_path / "drifted-agent"
+    shutil.copytree(AGENT_FIXTURE, drifted)
+    transcription = drifted / "tools/transcription_tools.py"
+    source = transcription.read_text(encoding="utf-8")
+    transcription.write_text(
+        source.replace("Failed to convert audio for local STT", "upstream drift", 1),
+        encoding="utf-8",
+    )
+    env = os.environ.copy()
+    env["HERMES_AGENT_PATCH_ROOT"] = str(drifted)
+    result = subprocess.run(
+        [sys.executable, str(STT_PATCHER)],
+        cwd=ROOT,
+        env=env,
+        check=False,
+        capture_output=True,
+        text=True,
+    )
+    assert result.returncode != 0
+    assert "patch context changed" in result.stderr

testing/tests/test_hermes_voice_instrument.py

@@ -16,6 +16,9 @@ PATCHER = ROOT / "dockerfiles/hermes-webui-atlas-patch.py"
 VOICE_JS = ROOT / "dockerfiles/hermes-webui-atlas-voice.js"
 VOICE_CSS = ROOT / "dockerfiles/hermes-webui-atlas-voice.css"
 DOM_PROBE = ROOT / "testing/probes/hermes_voice_instrument_probe.js"
+MEDIARECORDER_FIXTURE = (
+    ROOT / "testing/fixtures/mediarecorder/chromium-webm-opus.json"
+)
 
 
 def _patched_fixture(tmp_path: Path) -> Path:
@@ -86,7 +89,7 @@ def test_visual_states_have_distinct_layers_finite_error_and_reduced_motion():
 
 def test_dom_probe_exercises_actual_injected_voice_script():
     result = subprocess.run(
-        ["node", str(DOM_PROBE), str(VOICE_JS)],
+        ["node", str(DOM_PROBE), str(VOICE_JS), str(MEDIARECORDER_FIXTURE)],
         cwd=ROOT,
         check=True,
         capture_output=True,