feat(morpheus,gateway-chat): streaming decode — TTFA ~4.5s -> ~0.8s
Wrapper gains POST /tts/stream: reads the vLLM token stream, decodes SNAC in WINDOWED
CHUNKS (every 6 frames, decode [2 ctx | 6 | 2 ctx] and emit only the middle 6 — context
both sides => seamless), and streams raw PCM16 (24kHz mono) as it generates. Windowed
(not per-frame) because per-frame CPU decode's per-call overhead x ~60 frames serialized
to ~7s (RTF 2.2); windowed keeps up (RTF ~0.97). Whole-clip /tts kept for non-browser use.
gateway-chat plays the stream via the Web Audio API (fetch reader -> int16->float32 ->
scheduled AudioBufferSourceNodes on a running clock; a new reply supersedes the prior
stream via a generation counter; 🔊 replays). Measured: TTFA 0.80s (was ~4.5s whole-clip),
RTF 0.97, full-duration match. CORS already covers the new route.
Deployed: tts rebuilt on irv-ml1, page pushed to ana-docker.
This commit is contained in:
@@ -144,33 +144,54 @@ function addMsg(who, cls){
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}
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function showErr(text){ const m = addMsg('error','err'); m.body.className = 'err'; m.body.textContent = text; }
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// --- mOrpheus TTS: auto-voice quoted text from a completed assistant reply ---
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let ttsAudio = null; // current clip, so a new reply interrupts the previous one
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// --- mOrpheus TTS: streaming-decode auto-voice of quoted text (Web Audio, low TTFA) ---
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let audioCtx = null; // shared AudioContext
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let ttsGen = 0; // generation counter — a newer reply's stream supersedes older ones
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function extractQuotes(text){
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const re = /“([^”]+)”|"([^"]+)"/g; // typographic "…" or straight "…"
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const out = []; let m;
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while ((m = re.exec(text)) !== null){ const q = (m[1] || m[2] || '').trim(); if (q) out.push(q); }
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return out;
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}
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async function streamSpeak(speech, tag){
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const mine = ++ttsGen; // cancels any in-flight stream
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if (!audioCtx) audioCtx = new (window.AudioContext || window.webkitAudioContext)();
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if (audioCtx.state === 'suspended') await audioCtx.resume();
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const url = $('ttsUrl').value.trim().replace(/\/+$/,'').replace(/\/tts$/,'') + '/tts/stream';
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const res = await fetch(url, {
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method:'POST', headers:{ 'Content-Type':'application/json' },
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body: JSON.stringify({ text: speech, voice: ($('ttsVoice').value.trim() || 'baddy'), max_tokens: 1200 })
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});
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if (!res.ok || !res.body){ tag.textContent = ' 🔇' + res.status; return; }
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const reader = res.body.getReader();
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let playhead = audioCtx.currentTime + 0.12, leftover = new Uint8Array(0);
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for(;;){
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const { done, value } = await reader.read();
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if (done || mine !== ttsGen) break; // stop if a newer reply took over
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let bytes = new Uint8Array(leftover.length + value.length);
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bytes.set(leftover); bytes.set(value, leftover.length);
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const n = bytes.length - (bytes.length % 2); // whole int16 samples only
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leftover = bytes.slice(n);
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if (!n) continue;
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const dv = new DataView(bytes.buffer, 0, n), N = n / 2, f32 = new Float32Array(N);
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for (let i = 0; i < N; i++) f32[i] = dv.getInt16(i * 2, true) / 32768;
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const abuf = audioCtx.createBuffer(1, N, 24000); abuf.getChannelData(0).set(f32);
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const src = audioCtx.createBufferSource(); src.buffer = abuf; src.connect(audioCtx.destination);
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if (playhead < audioCtx.currentTime) playhead = audioCtx.currentTime + 0.02; // underrun catch-up
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src.start(playhead); playhead += abuf.duration;
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}
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if (mine === ttsGen) tag.textContent = ' 🔈';
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}
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async function speakQuotes(text, msg){
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if (!$('ttsOn').checked) return;
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const quotes = extractQuotes(text);
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if (!quotes.length) return;
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const speech = quotes.join(' ');
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const tag = document.createElement('span'); tag.textContent = ' 🔊';
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tag.style.cursor = 'pointer'; tag.title = 'mOrpheus'; msg.wrap.querySelector('.who').append(tag);
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try {
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const res = await fetch($('ttsUrl').value.trim(), {
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method:'POST', headers:{ 'Content-Type':'application/json' },
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body: JSON.stringify({ text: quotes.join(' '), voice: ($('ttsVoice').value.trim() || 'baddy'), max_tokens: 1200 })
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});
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if (!res.ok){ tag.textContent = ' 🔇' + res.status; return; }
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const url = URL.createObjectURL(await res.blob());
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if (ttsAudio) ttsAudio.pause();
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ttsAudio = new Audio(url);
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tag.onclick = () => ttsAudio.play(); // click 🔊 to replay
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ttsAudio.onended = () => tag.textContent = ' 🔈';
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ttsAudio.play();
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} catch (e){ tag.textContent = ' 🔇'; }
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tag.style.cursor = 'pointer'; tag.title = 'mOrpheus — click to replay';
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tag.onclick = () => streamSpeak(speech, tag).catch(() => tag.textContent = ' 🔇');
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msg.wrap.querySelector('.who').append(tag);
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try { await streamSpeak(speech, tag); } catch (e){ tag.textContent = ' 🔇'; }
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}
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async function send(){
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@@ -11,11 +11,11 @@ pos2/3/5/6->L3, offsets k*4096). Named-speaker prompt: [SOH] "voice: text" [EOT]
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Zero-shot cloning: [BOS][SOH] ref_text [EOT][SOA][SOS] <ref audio tokens> [EOS_sp] then
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[SOH] text [EOT][SOA]. Generation stops at end-of-speech (128258).
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"""
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import os, io, base64
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import os, io, re, json, base64
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import numpy as np, torch, soundfile as sf, requests
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from scipy.signal import resample_poly
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from fastapi import FastAPI, HTTPException
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from fastapi.responses import Response
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from fastapi.responses import Response, StreamingResponse
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from fastapi.middleware.cors import CORSMiddleware
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from pydantic import BaseModel
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from transformers import AutoTokenizer
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@@ -128,3 +128,92 @@ def tts(req: TTSReq):
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sf.write(buf, audio, 24000, format="WAV", subtype="PCM_16")
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return Response(content=buf.getvalue(), media_type="audio/wav",
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headers={"X-Audio-Seconds": f"{len(audio)/24000:.2f}"})
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# --- streaming decode: emit raw PCM16 (24kHz mono) as the engine generates -----------
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# vLLM emits audio tokens as <custom_token_N> strings (token_id = 128256 + N). We decode in
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# WINDOWED CHUNKS: every CHUNK new frames, decode a small window [CTX left | CHUNK | CTX
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# right] and emit only the middle CHUNK frames (context on both sides => seamless, no
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# clicks). O(window) per call and few calls => decode keeps up with gen on CPU (per-frame
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# decode did NOT — per-call overhead x ~60 frames serialized to seconds). TTFA ~0.7s.
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CUSTOM_RE = re.compile(r"<custom_token_(\d+)>")
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FRAME = 2048 # audio samples per SNAC frame
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def _decode_frames(ids: list[int]):
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codes = [t - AUDIO_BASE for t in ids if AUDIO_BASE <= t <= AUDIO_MAX]
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n = len(codes) // 7
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if n == 0:
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return None
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l1, l2, l3 = [], [], []
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for i in range(n):
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f = codes[7 * i:7 * i + 7]
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c = [f[0], f[1] - 4096, f[2] - 8192, f[3] - 12288, f[4] - 16384, f[5] - 20480, f[6] - 24576]
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if any(x < 0 or x > 4095 for x in c):
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return None
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l1.append(c[0]); l2 += [c[1], c[4]]; l3 += [c[2], c[3], c[5], c[6]]
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ct = [torch.tensor(l1).unsqueeze(0).to(SNAC_DEVICE),
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torch.tensor(l2).unsqueeze(0).to(SNAC_DEVICE),
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torch.tensor(l3).unsqueeze(0).to(SNAC_DEVICE)]
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with torch.inference_mode():
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return snac_model.decode(ct).squeeze().cpu().numpy()
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def _pcm16(arr) -> bytes:
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return (np.clip(arr, -1, 1) * 32767).astype(np.int16).tobytes()
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@app.post("/tts/stream")
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def tts_stream(req: TTSReq):
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prompt_ids = _build_prompt(req)
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payload = {"model": VLLM_MODEL, "prompt": prompt_ids, "max_tokens": req.max_tokens,
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"temperature": req.temperature, "top_p": req.top_p, "skip_special_tokens": False,
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"stop_token_ids": [EOS_SP], "repetition_penalty": req.repetition_penalty, "stream": True}
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CTX, CHUNK = 2, 6 # context frames each side, frames emitted per decode
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def emit(emitted, total, final):
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# decode [emitted-CTX : end] with left/right context; return (pcm_bytes, new_emitted)
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end = total if final else emitted + CHUNK
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ws = max(0, emitted - CTX)
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audio = _decode_frames(buf[7 * ws:7 * (end + (0 if final else CTX))])
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if audio is None:
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return b"", end
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off = (emitted - ws) * FRAME
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chunk = audio[off:] if final else audio[off:off + CHUNK * FRAME]
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return (_pcm16(chunk) if len(chunk) else b""), end
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buf = []
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def gen():
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acc, cursor, emitted = "", 0, 0
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with requests.post(VLLM_URL, json=payload, stream=True, timeout=180) as r:
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for raw in r.iter_lines():
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if not raw:
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continue
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s = raw.decode("utf-8", "ignore")
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if not s.startswith("data:"):
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continue
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d = s[5:].strip()
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if d == "[DONE]":
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break
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try:
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j = json.loads(d)
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except Exception:
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continue
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acc += j["choices"][0].get("text", "")
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for m in CUSTOM_RE.finditer(acc, cursor):
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cursor = m.end()
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tid = 128256 + int(m.group(1))
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if AUDIO_BASE <= tid <= AUDIO_MAX:
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buf.append(tid)
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while (len(buf) // 7) - emitted >= CHUNK + CTX: # enough new frames + right ctx
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b, emitted = emit(emitted, len(buf) // 7, final=False)
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if b:
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yield b
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if (len(buf) // 7) > emitted: # flush the tail
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b, emitted = emit(emitted, len(buf) // 7, final=True)
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if b:
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yield b
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return StreamingResponse(gen(), media_type="application/octet-stream",
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headers={"Cache-Control": "no-cache", "X-Sample-Rate": "24000"})
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+37
-16
@@ -144,33 +144,54 @@ function addMsg(who, cls){
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}
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function showErr(text){ const m = addMsg('error','err'); m.body.className = 'err'; m.body.textContent = text; }
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// --- mOrpheus TTS: auto-voice quoted text from a completed assistant reply ---
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let ttsAudio = null; // current clip, so a new reply interrupts the previous one
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// --- mOrpheus TTS: streaming-decode auto-voice of quoted text (Web Audio, low TTFA) ---
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let audioCtx = null; // shared AudioContext
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let ttsGen = 0; // generation counter — a newer reply's stream supersedes older ones
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function extractQuotes(text){
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const re = /“([^”]+)”|"([^"]+)"/g; // typographic "…" or straight "…"
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const out = []; let m;
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while ((m = re.exec(text)) !== null){ const q = (m[1] || m[2] || '').trim(); if (q) out.push(q); }
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return out;
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}
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async function streamSpeak(speech, tag){
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const mine = ++ttsGen; // cancels any in-flight stream
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if (!audioCtx) audioCtx = new (window.AudioContext || window.webkitAudioContext)();
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if (audioCtx.state === 'suspended') await audioCtx.resume();
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const url = $('ttsUrl').value.trim().replace(/\/+$/,'').replace(/\/tts$/,'') + '/tts/stream';
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const res = await fetch(url, {
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method:'POST', headers:{ 'Content-Type':'application/json' },
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body: JSON.stringify({ text: speech, voice: ($('ttsVoice').value.trim() || 'baddy'), max_tokens: 1200 })
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});
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if (!res.ok || !res.body){ tag.textContent = ' 🔇' + res.status; return; }
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const reader = res.body.getReader();
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let playhead = audioCtx.currentTime + 0.12, leftover = new Uint8Array(0);
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for(;;){
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const { done, value } = await reader.read();
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if (done || mine !== ttsGen) break; // stop if a newer reply took over
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let bytes = new Uint8Array(leftover.length + value.length);
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bytes.set(leftover); bytes.set(value, leftover.length);
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const n = bytes.length - (bytes.length % 2); // whole int16 samples only
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leftover = bytes.slice(n);
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if (!n) continue;
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const dv = new DataView(bytes.buffer, 0, n), N = n / 2, f32 = new Float32Array(N);
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for (let i = 0; i < N; i++) f32[i] = dv.getInt16(i * 2, true) / 32768;
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const abuf = audioCtx.createBuffer(1, N, 24000); abuf.getChannelData(0).set(f32);
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const src = audioCtx.createBufferSource(); src.buffer = abuf; src.connect(audioCtx.destination);
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if (playhead < audioCtx.currentTime) playhead = audioCtx.currentTime + 0.02; // underrun catch-up
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src.start(playhead); playhead += abuf.duration;
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}
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if (mine === ttsGen) tag.textContent = ' 🔈';
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}
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async function speakQuotes(text, msg){
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if (!$('ttsOn').checked) return;
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const quotes = extractQuotes(text);
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if (!quotes.length) return;
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const speech = quotes.join(' ');
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const tag = document.createElement('span'); tag.textContent = ' 🔊';
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tag.style.cursor = 'pointer'; tag.title = 'mOrpheus'; msg.wrap.querySelector('.who').append(tag);
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try {
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const res = await fetch($('ttsUrl').value.trim(), {
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method:'POST', headers:{ 'Content-Type':'application/json' },
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body: JSON.stringify({ text: quotes.join(' '), voice: ($('ttsVoice').value.trim() || 'baddy'), max_tokens: 1200 })
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});
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if (!res.ok){ tag.textContent = ' 🔇' + res.status; return; }
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const url = URL.createObjectURL(await res.blob());
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if (ttsAudio) ttsAudio.pause();
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ttsAudio = new Audio(url);
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tag.onclick = () => ttsAudio.play(); // click 🔊 to replay
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ttsAudio.onended = () => tag.textContent = ' 🔈';
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ttsAudio.play();
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} catch (e){ tag.textContent = ' 🔇'; }
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tag.style.cursor = 'pointer'; tag.title = 'mOrpheus — click to replay';
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tag.onclick = () => streamSpeak(speech, tag).catch(() => tag.textContent = ' 🔇');
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msg.wrap.querySelector('.who').append(tag);
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try { await streamSpeak(speech, tag); } catch (e){ tag.textContent = ' 🔇'; }
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}
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async function send(){
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