feat(audio): v1 signature producer, bit-exact with the pipeline

`AudioSignature` is the DSP — band table, periodic Hann, radix-2 FFT,
band-mean peak, energy class, packing — and `AudioSignatureService` the
decode, running the FFmpeg binary `IMediaEncoder.EncoderPath` names. The
plugin gained no dependency.

The server specification's prose does not determine a byte stream, so the
parameters it leaves open are pinned by the fixture shared with the
extraction repo and restated at the top of `AudioSignature`: double
throughout, whole frames only, periodic Hann, unnormalised FFT, band mean
rather than sum, argmax ties to the lowest index.

`fixtures/audio/` holds the extraction repo's three files byte-identically
and the computed signature equals the recorded vector exactly. The binding
check regenerates the fixture PCM from `make_fixture.py`'s arithmetic and
verifies it against the recorded decode checksums, so it runs on a host
with no codec at all and a decode divergence stays distinguishable from a
DSP one; the two tests that drive real FFmpeg self-skip without a binary.

The workflow named "Test Plugin" until now only compiled one. A test that
is built and never run is not evidence, and a golden vector shared across
two repos exists precisely so CI fails when they drift.

TRACES: JR-042, JR-043 | SR-003

Jellyfin.Plugin.JRay.Tests/fixtures/audio/make_fixture.py

@@ -0,0 +1,114 @@
+#!/usr/bin/env python3
+"""Regenerate the JRay audio-signature golden fixture.
+
+    python3 make_fixture.py            # writes jray_audio_v1_tone.flac here
+
+This is the *source of truth* for the fixture media: `jray_audio_v1_tone.flac`
+is a lossless FLAC encoding of exactly the PCM this script emits, so any repo
+that wants to check its own audio-signature implementation against the golden
+vector in `jray_audio_v1_golden.json` can regenerate the input from scratch and
+confirm it is byte-identical (the golden file records `pcm_fnv1a64`, a hash of
+the decoded 16-bit samples).
+
+Deliberately dependency-free (no numpy) and written in plain arithmetic so it
+ports to any language in ~20 lines.
+
+Signal — 120.000 s, mono, 11025 Hz, 16-bit signed PCM:
+
+  * split into segments of 32768 samples (~2.97 s), 40.4 segments in total;
+  * segment `s` carries one sine at the geometric centre of log-band
+    `(s * 7) mod 32` of the 300-3000 Hz band, so all 32 bands are exercised;
+  * its amplitude walks a golden-ratio low-discrepancy sequence over
+    [10^-1.55, 10^-0.02] so frame energies spread continuously across ~1.5
+    decades and all four energy classes are exercised, without a dense cluster
+    of frames sitting on a class boundary;
+  * phase is carried across segment boundaries (no clicks);
+  * a constant, far quieter 777 Hz tone sits underneath so no frame is
+    degenerate;
+  * samples are quantised with floor(x * 32767 + 0.5).
+
+Why FLAC and not WAV: 120 s of 11025 Hz 16-bit PCM is 2.6 MB and does not
+compress in git. FLAC is lossless — FFmpeg decodes it to exactly the PCM
+written here — and is ~3.5x smaller. `--wav` writes the uncompressed original
+if you want to diff it.
+"""
+import math
+import struct
+import subprocess
+import sys
+import os
+
+SAMPLE_RATE = 11025
+DURATION_SEC = 120.0
+SEGMENT = 32768          # samples per tone segment
+BAND_STRIDE = 7          # coprime with 32 -> visits every band
+BAND_LO_HZ = 300.0
+BAND_HI_HZ = 3000.0
+NUM_BANDS = 32
+AMP_LOG_MIN = -1.55      # 10^-1.55 ~= 0.028
+AMP_LOG_SPAN = 1.53      # up to 10^-0.02 ~= 0.955
+PHI_FRAC = 0.6180339887498949
+BG_HZ = 777.0
+BG_AMP = 0.004
+
+OUT_FLAC = "jray_audio_v1_tone.flac"
+OUT_WAV = "jray_audio_v1_tone.wav"
+
+
+def generate():
+    """Return the 120 s signal as a list of int16 sample values."""
+    n = int(round(SAMPLE_RATE * DURATION_SEC))
+    out = [0] * n
+    phase = 0.0
+    two_pi = 2.0 * math.pi
+    for start in range(0, n, SEGMENT):
+        s = start // SEGMENT
+        end = min(n, start + SEGMENT)
+        band = (s * BAND_STRIDE) % NUM_BANDS
+        # geometric centre of log-band `band`
+        freq = BAND_LO_HZ * (BAND_HI_HZ / BAND_LO_HZ) ** ((band + 0.5) / NUM_BANDS)
+        amp = 10.0 ** (AMP_LOG_MIN + AMP_LOG_SPAN * ((s * PHI_FRAC) % 1.0))
+        step = two_pi * freq / SAMPLE_RATE
+        for k in range(end - start):
+            i = start + k
+            x = amp * math.sin(phase + step * k)
+            x += BG_AMP * math.sin(two_pi * BG_HZ * i / SAMPLE_RATE)
+            if x > 1.0:
+                x = 1.0
+            elif x < -1.0:
+                x = -1.0
+            out[i] = int(math.floor(x * 32767.0 + 0.5))
+        phase = (phase + step * (end - start)) % two_pi
+    return out
+
+
+def write_wav(path, samples):
+    data = struct.pack("<%dh" % len(samples), *samples)
+    hdr = b"RIFF" + struct.pack("<I", 36 + len(data)) + b"WAVE"
+    hdr += b"fmt " + struct.pack("<IHHIIHH", 16, 1, 1, SAMPLE_RATE,
+                                 SAMPLE_RATE * 2, 2, 16)
+    hdr += b"data" + struct.pack("<I", len(data))
+    with open(path, "wb") as fh:
+        fh.write(hdr + data)
+
+
+def main():
+    here = os.path.dirname(os.path.abspath(__file__))
+    samples = generate()
+    wav = os.path.join(here, OUT_WAV)
+    write_wav(wav, samples)
+    if "--wav" in sys.argv:
+        print("wrote", wav)
+        return
+    flac = os.path.join(here, OUT_FLAC)
+    # -compression_level 12 is deterministic for a given libFLAC/ffmpeg build;
+    # only the container bytes vary, never the decoded PCM.
+    subprocess.run(["ffmpeg", "-nostdin", "-v", "error", "-y", "-i", wav,
+                    "-c:a", "flac", "-compression_level", "12", flac],
+                   check=True)
+    os.remove(wav)
+    print("wrote", flac)
+
+
+if __name__ == "__main__":
+    main()