Suppress speaker echo on the microphone channel

CLAUDE.md

@@ -60,7 +60,9 @@ Handler registration lives in [src/main/ipc/](src/main/ipc/) - one file per doma
 
 ### Transcription and Suggestion Flow
 
-[src/main/services/transcript.service.ts](src/main/services/transcript.service.ts) is the central orchestrator. `transcriptService.ingest(channel, type, text)` merges both audio channels, deduplicates overlapping segments, and decides whether a final `Other` transcript is worth answering - with a `LIVE_SUGGESTION_GAP_MS` guard that suppresses the call if Self spoke recently.
+[src/main/services/transcript.service.ts](src/main/services/transcript.service.ts) is the central orchestrator. `transcriptService.ingest(channel, type, text)` merges both audio channels and decides whether a final `Other` transcript is worth answering - with a `LIVE_SUGGESTION_GAP_MS` guard that suppresses the call if Self spoke recently.
+
+**Nothing deduplicates.** `mergeAdjacentTranscripts` concatenates consecutive blocks from the *same* speaker that fall within `TRANSCRIPT_INTER_TRANSCRIPT_GAP_MS`, and no code anywhere compares the two channels against each other. This line claimed the opposite for a while, which is worth naming because the gap it papered over is the whole of #111: on speakers the microphone re-captures the interviewer, the same words arrive on `ch_0` and `ch_1`, and both are kept. See Headphones below.
 
 - `ch_0` = `Speaker.Other` (interviewer, captured via loopback audio)
 - `ch_1` = `Speaker.Self` (candidate, captured via microphone)
@@ -262,6 +264,24 @@ visible; the damaging half is not. The echo lands as a recent `Self` final, so
 suppresses the live suggestion **for the question that was just asked**, with no error anywhere.
 See #111 for the measurements and the longer-term suppression work.
 
+How much of the interviewer the microphone actually re-captures is a property of the machine, not
+something to reason about, and no constant in a future gate should be picked before it is measured.
+`test/manual/echo-probe.mjs` runs both captures through one worklet and reports the signed
+arrival-order delay, the correlation peak at that lag and the echo return loss, once a second - run
+by hand (`pnpm exec electron test/manual/echo-probe.mjs`), deliberately not in `test/run.mjs`, since
+it needs a desktop session, real speakers, and a person to play audio into them. `--no-aec`,
+`--no-ns` and `--no-agc` drive the A/B on the processing flags below.
+
+Those flags are stated rather than defaulted. Every `getUserMedia` in the app opens through
+`micConstraints()` in
+[live-transcription.service.ts](src/renderer/services/live-transcription.service.ts), which writes
+out `echoCancellation`, `noiseSuppression` and `autoGainControl`. Chromium already defaults all
+three to `true`, so this changes nothing today; the point is that they stop moving on their own
+under a version bump, and that there is one place to flip them from once the probe says which way
+they should go. `test/mic-constraints.test.mjs` fails on any capture that opens its own way instead,
+which is not hypothetical - two of them have already been added, one duplicating the flags and one
+opening with `audio: true`, and neither produced a conflict, a type error or a lint warning.
+
 [headphone-notice-dialog.tsx](src/renderer/components/custom/headphone-notice-dialog.tsx) is shown
 before every session until the user silences it, and it says what actually goes wrong rather than
 recommending headphones for "best results" - the cost of ignoring it is answers that never appear.

src/renderer/components/custom/settings/microphone-field.tsx

@@ -15,7 +15,7 @@ import { useAppState } from '@/hooks/use-app-state';
 import { useAudioInputDevices } from '@/hooks/use-audio-devices';
 import { useAudioInputDevice } from '@/hooks/use-audio-input-device';
 import { useConfigStore } from '@/hooks/use-config-store';
-import { resolveMicDeviceId } from '@/services/live-transcription.service';
+import { micConstraints, resolveMicDeviceId } from '@/services/live-transcription.service';
 import { RunningState } from '@/types/app-state';
 
 /**
@@ -91,8 +91,11 @@ export function MicrophoneField() {
     setTestStarting(true);
     try {
       const deviceId = await resolveMicDeviceId(deviceName);
+      // The same constraints a session opens with, so the level shown here is measured through
+      // the same processing chain the session will use. Opened as `true`, the test stream could
+      // run different gain and noise handling than the capture it is meant to predict.
       const stream = await navigator.mediaDevices.getUserMedia({
-        audio: deviceId ? { deviceId: { exact: deviceId } } : true,
+        audio: micConstraints(deviceId),
       });
       setTestStream(stream);
     } catch (e) {

src/renderer/services/live-transcription.service.ts

@@ -25,6 +25,43 @@ function buildStreamingUrl(language: Language): string {
   return `${STREAMING_URL}?language=${encodeURIComponent(language)}`;
 }
 
+/**
+ * Whether the microphone track runs Chromium's automatic gain control.
+ *
+ * Kept as a named constant rather than inlined because it is the flag most likely to move. AGC is
+ * the largest source of coupling-gain instability when the candidate is on speakers: it raises
+ * gain through quiet passages, which amplifies re-captured interviewer audio at exactly the moment
+ * an echo gate is trying to measure how much of it there is. The opposite pull is ASR accuracy for
+ * a quiet candidate. Measure with `test/manual/echo-probe.mjs` before changing it.
+ */
+const MIC_AUTO_GAIN_CONTROL = true;
+
+/**
+ * The constraints every microphone capture in the app opens with.
+ *
+ * The three processing flags are stated rather than left out. Chromium's defaults for an
+ * unspecified flag are already `true` for all three, so writing them changes nothing today - the
+ * point is that it stops changing on its own when Chromium's defaults move under a version bump,
+ * and that there is one place to flip them when the echo probe says which way they should go.
+ *
+ * An absent `deviceId` is the "system default microphone" case, and is deliberately expressed as
+ * an object with no `deviceId` key rather than as `audio: true` - `true` would drop the flags with
+ * it and put that user back on whatever Chromium currently defaults to.
+ *
+ * Exported because "one place" only holds if every caller uses it. The mock service and the
+ * settings microphone test open their own streams, and a second copy of these flags is the same
+ * drift this exists to stop - with the extra sting that the level the test meter shows would be
+ * measured through different processing than the session it is meant to predict.
+ */
+export function micConstraints(deviceId: string | null): MediaTrackConstraints {
+  return {
+    ...(deviceId ? { deviceId: { exact: deviceId } } : {}),
+    echoCancellation: true,
+    noiseSuppression: true,
+    autoGainControl: MIC_AUTO_GAIN_CONTROL,
+  };
+}
+
 // Inline AudioWorklet processor (runs off the main thread)
 const AUDIO_WORKLET_CODE = `
 class AudioSenderWorklet extends AudioWorkletProcessor {
@@ -487,7 +524,7 @@ class LiveTranscriptionService {
 
     const micDeviceId = await resolveMicDeviceId(audioInputDeviceName);
     this.micStream = await navigator.mediaDevices.getUserMedia({
-      audio: micDeviceId ? { deviceId: { exact: micDeviceId } } : true,
+      audio: micConstraints(micDeviceId),
       video: false,
     });
 
@@ -566,7 +603,7 @@ class LiveTranscriptionService {
 
     const deviceId = await resolveMicDeviceId(deviceName);
     const nextStream = await navigator.mediaDevices.getUserMedia({
-      audio: deviceId ? { deviceId: { exact: deviceId } } : true,
+      audio: micConstraints(deviceId),
       video: false,
     });
 

src/renderer/services/mock-transcription.service.ts

@@ -1,7 +1,7 @@
 import { getElectron } from '@/lib/utils';
 import { Language } from '@/types/language';
 
-import { AudioWsStream, resolveMicDeviceId } from './live-transcription.service';
+import { AudioWsStream, micConstraints, resolveMicDeviceId } from './live-transcription.service';
 
 /**
  * Microphone-only capture for a mock interview.
@@ -34,14 +34,7 @@ class MockTranscriptionService {
 
     const micDeviceId = await resolveMicDeviceId(audioInputDeviceName);
     this.micStream = await navigator.mediaDevices.getUserMedia({
-      audio: micDeviceId
-        ? {
-            deviceId: { exact: micDeviceId },
-            echoCancellation: true,
-            noiseSuppression: true,
-            autoGainControl: true,
-          }
-        : { echoCancellation: true, noiseSuppression: true, autoGainControl: true },
+      audio: micConstraints(micDeviceId),
       video: false,
     });