Record at the input device's native sample rate

- Recorder queries default_samplerate and streams/WAVs at that rate
  instead of forcing 16 kHz (helps devices that don't support it
  natively and keeps recordings at full fidelity)
- new resample_16k() linear resampler: the live rolling window is
  converted to Whisper's 16 kHz before being handed to the engines
- snapshot() now returns (16 kHz audio, absolute start time in s)
- final pass unchanged: faster-whisper (PyAV) and whisper.cpp
  (miniaudio) resample native-rate WAVs themselves
- GUI status shows the active rate; README updated

Validated with real 44.1 kHz and 48 kHz inputs: WAV headers match the
device rate, live-window resampling is length- and spectrum-correct.

README.md

@@ -16,12 +16,14 @@ With output stem `-o meeting`:
 
 | File               | Contents                                        |
 | ------------------ | ----------------------------------------------- |
-| `meeting.wav`      | raw recording (16 kHz mono WAV)                 |
+| `meeting.wav`      | raw recording (mono WAV at the input's native rate) |
 | `meeting.live.srt` | live rolling transcript (only with `--live > 0`) |
 | `meeting.txt`      | final transcript                                |
 | `meeting.srt`      | final transcript with timestamps                |
 | `meeting.json`     | final transcript, structured (language, segments) |
 
+Recording always uses the input device's native sample rate; audio is resampled to Whisper's 16 kHz automatically (both for the live window and the final pass).
+
 ## Requirements
 
 - Linux with an audio backend: **PipeWire** (or PulseAudio/ALSA). On Arch: `sudo pacman -S pipewire pipewire-pulse`

meetrec.py

@@ -8,12 +8,15 @@ printing a live rolling transcript. On Ctrl+C it stops cleanly and runs a
 final, higher-quality full transcription.
 
 Outputs (with -o meeting):
-    meeting.wav      raw recording, 16 kHz mono WAV
+    meeting.wav      raw recording, mono WAV at the input's native rate
     meeting.live.srt live rolling transcript (only when --live > 0)
     meeting.txt      final transcript
     meeting.srt      final transcript with timestamps
     meeting.json     final transcript, structured
 
+Audio is recorded at the input device's native sample rate; both the live
+window and the engines get it resampled to Whisper's 16 kHz internally.
+
 Setup on Arch:
     sudo pacman -S python pipewire pipewire-pulse
     python -m venv .venv && source .venv/bin/activate
@@ -75,6 +78,22 @@ def write_srt(path: str, items) -> None:
             f.write(f"{i}\n{fmt_srt(a)} --> {fmt_srt(b)}\n{text}\n\n")
 
 
+def resample_16k(audio: np.ndarray, rate: int) -> np.ndarray:
+    """Linearly resample mono float32 audio to 16 kHz (Whisper's rate).
+
+    Only needed for the live rolling buffer, which is fed to the engines as
+    a raw array (they assume 16 kHz there). WAV files are resampled by the
+    engines themselves.
+    """
+    audio = np.asarray(audio, dtype=np.float32)
+    if rate == SR or audio.size == 0:
+        return audio
+    n_out = max(1, int(round(audio.size * SR / rate)))
+    t_out = np.arange(n_out, dtype=np.float64) * (rate / SR)
+    return np.interp(t_out, np.arange(audio.size, dtype=np.float64),
+                     audio).astype(np.float32)
+
+
 def list_sources() -> None:
     print("PortAudio (PipeWire/Pulse/ALSA) input devices:\n")
     for i, d in enumerate(sd.query_devices()):
@@ -121,11 +140,15 @@ def resolve_source(name: str):
 # --------------------------------------------------------------------------
 
 class Recorder:
-    """Records mono 16 kHz float32 to a WAV file and keeps a rolling buffer."""
+    """Records mono float32 at the input's native rate to a WAV file and
+    keeps a rolling buffer for live transcription."""
 
     def __init__(self, device, wav_path: str):
         self.device = device
         self.wav_path = wav_path
+        dev = sd.query_devices(kind="input") if device is None \
+            else sd.query_devices(device)
+        self.rate = int(dev["default_samplerate"])   # input's native rate
         self.q: "queue.Queue" = queue.Queue()
         self.lock = threading.Lock()
         self.buf = np.zeros(0, dtype=np.float32)   # rolling window
@@ -146,7 +169,7 @@ class Recorder:
             block = indata[:, 0].astype(np.float32, copy=False)
             self.buf = np.append(self.buf, block)
             self.total += len(block)
-            limit = LIVE_WINDOW * SR
+            limit = LIVE_WINDOW * self.rate
             if len(self.buf) > limit:
                 drop = len(self.buf) - limit
                 self.buf = self.buf[drop:]
@@ -154,13 +177,13 @@ class Recorder:
 
     def duration(self) -> float:
         with self.lock:
-            return self.total / SR
+            return self.total / self.rate
 
     def _write(self):
         with wave.open(self.wav_path, "wb") as w:
             w.setnchannels(1)
             w.setsampwidth(2)
-            w.setframerate(SR)
+            w.setframerate(self.rate)
             while True:
                 block = self.q.get()
                 if block is None:
@@ -169,13 +192,13 @@ class Recorder:
                 w.writeframes(pcm.tobytes())
 
     def snapshot(self):
-        """Return (rolling audio, sample offset of its start)."""
+        """Return (rolling window resampled to 16 kHz, its start time in s)."""
         with self.lock:
-            return self.buf.copy(), self.base
+            return resample_16k(self.buf, self.rate), self.base / self.rate
 
     def start(self):
         self.stream = sd.InputStream(
-            samplerate=SR, channels=1, dtype="float32",
+            samplerate=self.rate, channels=1, dtype="float32",
             device=self.device, callback=self._callback)
         self.stream.start()
 
@@ -224,7 +247,7 @@ def live_worker(model, rec: Recorder, args, live_srt_path: str,
         stop.wait(args.live)
         if stop.is_set():
             break
-        buf, base = rec.snapshot()
+        buf, base_s = rec.snapshot()   # 16 kHz audio, absolute start time
         if len(buf) < 2 * SR:      # need at least 2 s of audio
             continue
         try:
@@ -234,7 +257,6 @@ def live_worker(model, rec: Recorder, args, live_srt_path: str,
             print(f"[live] transcription error: {e}", file=sys.stderr)
             continue
 
-        base_s = base / SR
         added = False
         for s in segs:
             a, b = base_s + s.start, base_s + s.end
@@ -538,7 +560,8 @@ def main():
 
     rec = Recorder(device, args.output + ".wav")
     rec.start()
-    print(f"recording to {args.output}.wav  (Ctrl+C to stop)", flush=True)
+    print(f"recording to {args.output}.wav at {rec.rate} Hz "
+          f"(Ctrl+C to stop)", flush=True)
 
     stop = threading.Event()
     worker = None

meetrec_gui.py

@@ -400,7 +400,9 @@ class MeetRecWindow(QWidget):
             name = sd.query_devices(idx)["name"]
         except Exception:
             name = self._args.source
-        self.status.setText(f"Recording from: {name}  →  {self._args.output}.wav")
+        self.status.setText(
+            f"Recording from: {name} @ {self.rec.rate} Hz  →  "
+            f"{self._args.output}.wav")
 
         if self._args.live:
             self.live_thread = threading.Thread(