v6.9: Per-line instruct on cloned voices
The clone server passed instruct=voice_cfg.get("instruct") at both call sites,
so a voice's delivery was fixed in voices.json for its whole lifetime. For an
audiobook that means a character can be consistent or expressive, not both:
VoiceDesign takes a per-request instruct but has no seed, so its identity is
re-sampled on every call, while VoiceClone is deterministic but had no way to
direct a line.
SpeechRequest now accepts instruct, merged identity-first via _merge_instruct()
— the voice's own description followed by the line's direction — so directing
one line cannot re-roll the character. Same rule run_voicedesign_server.py got
in v6.8.
This is safe on the clone path specifically because extract_embeddings.py writes
full ICL prompts (ref codec tokens + ref_text), not x-vector-only ones;
model.py warns that instruct is unreliable in the latter mode.
The change lives in patches/openai_server.patch, regenerated against the
pristine upstream blob and verified to apply cleanly. README documents the
request field and adds a section on the cast-in-VoiceDesign,
narrate-from-VoiceClone workflow.
Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>
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README.md
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README.md
@ -186,6 +186,39 @@ CustomVoice uses the model's built-in speaker names. Define the speaker IDs you
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Then call the CustomVoice service on port `8022`.
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## Consistent characters for long-form narration
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For an audiobook a character must sound identical across thousands of requests while still carrying emotion. Those two goals live in different services, so use each for what it is good at.
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**VoiceDesign has no seed.** `generate_voice_design()` accepts none, so a designed voice's identity is re-sampled from its prompt on every call. Low `temperature`/`top_p` narrows the spread but does not remove it. Treat VoiceDesign as a casting tool, not a narration engine.
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**VoiceClone is deterministic.** Every request seeds the RNG — from `seed` in `voices.json`, or a stable hash of the voice name when none is set — and the `.pt` prompt is a frozen recording of one specific reference. The same text yields the same audio across restarts.
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The workflow:
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1. **Cast** the character at port `8021` until a take sounds right. Generate the same line several times; the identity you hear wandering is what step 3 eliminates.
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2. **Freeze** the take you liked: save ~10–20 s of clean audio as the character's reference WAV with an exact transcript as `ref_text`. The character is now a VoiceClone voice.
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3. **Bake** the `.pt`. `extract_embeddings.py` writes a full ICL prompt (reference codec tokens plus transcript), which both pins the identity and puts you in the mode where instructions are followed reliably.
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4. **Pin the seed.** Audition with `find_best_seed.py` and `/seed-samples/{voice}`, then lock the winner with `POST /voice-seed`.
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5. **Narrate** at port `8020`, passing per-line `instruct` for emotion.
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Once a character's reference audio is chosen, treat it as immutable. Re-recording it invalidates the `.pt` (see v6.8) and shifts the identity, which means re-auditioning the seed and re-rendering everything already produced.
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### Directing a line
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`instruct` on a request is a *direction*, not a description. It is appended to the voice's own `instruct`, so keep the character in the voice and only the delivery in the request:
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```bash
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curl http://localhost:8020/v1/audio/speech \
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-H "Content-Type: application/json" \
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-d '{"input":"You should not have come here.",
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"voice":"DE_M_Zerwas",
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"instruct":"quietly, through gritted teeth"}' \
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--output line_0417.wav
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```
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Write `"whispering, barely audible"`, not `"a gravelly old man, whispering"` — the second re-describes the character and invites drift. Cast on a calm, level reference: a reference read with strong emotion bakes that emotion into the identity and fights every later direction. Test each character's most extreme line early; if a direction breaks the identity, the reference is too short or too expressive.
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## Streaming backend
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The streaming service on port `8023` uses the same generated `config/voices.json` and active VoiceClone reference voices as port `8020`, but returns WAV chunks while generation is still running. Use it when time-to-first-audio matters more than waiting for the complete WAV response.
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@ -213,7 +246,7 @@ The streaming service on port `8023` uses the same generated `config/voices.json
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| `response_format` | string | `wav` | `wav`, `pcm`, `mp3`, or `zip` (for timestamps) |
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| `speed` | float | 1.0 | Scales audio tempo via ffmpeg |
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| `language` | string | voice config | Per-request override for VoiceDesign/CustomVoice |
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| `instruct` | string | voice config | Per-request style override for VoiceDesign/CustomVoice |
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| `instruct` | string | voice config | Per-line delivery direction. Appended to the voice's own `instruct`, never replacing it. Supported by all three services |
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| `max_new_tokens` | int | server default | Per-request generation length override |
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WAV and PCM are streamed as audio is generated. MP3 is encoded after generation and returned as a complete response.
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@ -344,6 +377,13 @@ The first request after container startup can be slower because CUDA graph captu
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## Changelog
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### v6.9 — 2026-07-29
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**Feature: Per-Line Emotional Direction on Cloned Voices**
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- **Request-Level `instruct` for VoiceClone:** The clone server previously read `instruct` only from `voices.json`, so a character could have one fixed delivery for an entire book. `SpeechRequest` now accepts `instruct`, applied to both the streaming and non-streaming paths.
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- **Identity-First Merging:** A per-request `instruct` is appended to the voice's own description rather than replacing it, so directing one line's emotion cannot re-roll the character. Matches the behaviour added to the VoiceDesign server in v6.8.
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- **Why It Works Here:** Qwen3-TTS follows instructions reliably only in ICL mode, not with x-vector-only cloning. The `.pt` prompts written by `extract_embeddings.py` are full ICL prompts, so cloned voices qualify.
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- **Documentation:** New "Consistent characters for long-form narration" section covering the cast-in-VoiceDesign, narrate-from-VoiceClone workflow for audiobooks.
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### v6.8 — 2026-07-29
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**Fix: Stale Speaker Embeddings and Unregistered Voice Design Voices**
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- **Stale `.pt` Embeddings:** A speaker embedding is a cache baked from one specific pairing of reference audio and transcript. Re-recording a voice replaced those sources but left the old `.pt` in place, so the server kept cloning from an embedding whose audio tokens no longer matched its transcript — generation ignored the requested text and emitted unrelated filler. `generate_voices.py` now treats an embedding older than its reference audio or transcript as absent and regenerates it.
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@ -1,5 +1,5 @@
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diff --git a/examples/openai_server.py b/examples/openai_server.py
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index 2199e14..d10065f 100644
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index 2199e14..a65f81e 100644
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--- a/examples/openai_server.py
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+++ b/examples/openai_server.py
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@@ -36,6 +36,7 @@ API usage:
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@ -10,7 +10,7 @@ index 2199e14..d10065f 100644
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import io
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import json
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import logging
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@@ -66,9 +67,46 @@ app = FastAPI(title="faster-qwen3-tts OpenAI-compatible API")
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@@ -66,9 +67,66 @@ app = FastAPI(title="faster-qwen3-tts OpenAI-compatible API")
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tts_model = None
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voices: dict = {}
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@ -53,11 +53,31 @@ index 2199e14..d10065f 100644
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+ torch.manual_seed(seed)
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+ if torch.cuda.is_available():
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+ torch.cuda.manual_seed_all(seed)
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+
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+
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+def _merge_instruct(voice_cfg: dict, req_instruct: Optional[str]) -> Optional[str]:
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+ """Combine a voice's identity instruct with a per-request delivery direction.
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+
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+ A voice's instruct in voices.json describes WHO is speaking; an instruct
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+ sent with the request describes HOW this one line is delivered. Replacing
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+ the former with the latter would re-roll the character from a few words of
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+ direction, so combine them, identity first — the same rule the VoiceDesign
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+ server applies.
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+
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+ Instructions are only followed reliably in ICL mode. The .pt prompts written
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+ by extract_embeddings.py are full ICL prompts (ref codec tokens + ref_text),
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+ not x-vector-only, so this path qualifies.
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+ """
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+ base = str(voice_cfg.get("instruct") or "").strip()
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+ line = str(req_instruct or "").strip()
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+ if base and line and line != base:
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+ return f"{base} {line}"
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+ return line or base or None
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+
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# ---------------------------------------------------------------------------
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# Request / response models
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@@ -79,8 +117,8 @@ class SpeechRequest(BaseModel):
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@@ -79,8 +137,9 @@ class SpeechRequest(BaseModel):
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model: str = "tts-1"
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input: str
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voice: str = "alloy"
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@ -65,10 +85,11 @@ index 2199e14..d10065f 100644
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- speed: float = 1.0 # accepted but not yet applied
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+ response_format: str = "wav" # wav | pcm | mp3 | zip
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+ speed: float = 1.0 # scales audio tempo
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+ instruct: Optional[str] = None # per-line delivery direction, see _merge_instruct
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# ---------------------------------------------------------------------------
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@@ -145,6 +183,21 @@ def _to_mp3_bytes(pcm: np.ndarray, sample_rate: int) -> bytes:
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@@ -145,6 +204,21 @@ def _to_mp3_bytes(pcm: np.ndarray, sample_rate: int) -> bytes:
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def resolve_voice(voice_name: str) -> dict:
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"""Return voice config dict or fall back to default, else raise 400."""
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@ -90,7 +111,7 @@ index 2199e14..d10065f 100644
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if voice_name in voices:
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return voices[voice_name]
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if default_voice and default_voice in voices:
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@@ -168,7 +221,47 @@ def resolve_voice(voice_name: str) -> dict:
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@@ -168,7 +242,47 @@ def resolve_voice(voice_name: str) -> dict:
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# ---------------------------------------------------------------------------
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@ -135,11 +156,11 @@ index 2199e14..d10065f 100644
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+ return None
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+
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+
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+async def _stream_chunks(voice_cfg: dict, text: str, voice_name: str, speed: float) -> AsyncGenerator[bytes, None]:
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+async def _stream_chunks(voice_cfg: dict, text: str, voice_name: str, speed: float, instruct: Optional[str] = None) -> AsyncGenerator[bytes, None]:
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"""
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Run generate_voice_clone_streaming in a background thread and yield
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raw PCM bytes for each chunk as they arrive.
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@@ -177,21 +270,63 @@ async def _stream_chunks(voice_cfg: dict, text: str) -> AsyncGenerator[bytes, No
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@@ -177,21 +291,63 @@ async def _stream_chunks(voice_cfg: dict, text: str) -> AsyncGenerator[bytes, No
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_DONE = object()
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def producer():
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@ -179,7 +200,7 @@ index 2199e14..d10065f 100644
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ref_text=voice_cfg.get("ref_text", ""),
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chunk_size=voice_cfg.get("chunk_size", 12),
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- non_streaming_mode=False,
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+ instruct=voice_cfg.get("instruct"),
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+ instruct=_merge_instruct(voice_cfg, instruct),
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+ voice_clone_prompt=_load_voice_clone_prompt(voice_cfg, voice_name, tts_model),
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+ non_streaming_mode=True,
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+ temperature=voice_cfg.get("temperature", 0.8),
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@ -207,7 +228,7 @@ index 2199e14..d10065f 100644
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thread = threading.Thread(target=producer, daemon=True)
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thread.start()
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@@ -203,7 +338,7 @@ async def _stream_chunks(voice_cfg: dict, text: str) -> AsyncGenerator[bytes, No
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@@ -203,7 +359,7 @@ async def _stream_chunks(voice_cfg: dict, text: str) -> AsyncGenerator[bytes, No
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break
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if isinstance(item, Exception):
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raise item
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@ -216,7 +237,7 @@ index 2199e14..d10065f 100644
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# ---------------------------------------------------------------------------
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@@ -220,7 +355,8 @@ async def health():
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@@ -220,7 +376,8 @@ async def health():
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async def create_speech(req: SpeechRequest):
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if tts_model is None:
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raise HTTPException(status_code=503, detail="Model not loaded")
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@ -226,7 +247,7 @@ index 2199e14..d10065f 100644
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raise HTTPException(status_code=400, detail="'input' text is empty")
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voice_cfg = resolve_voice(req.voice)
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@@ -230,36 +366,77 @@ async def create_speech(req: SpeechRequest):
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@@ -230,36 +387,77 @@ async def create_speech(req: SpeechRequest):
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"wav": "audio/wav",
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"pcm": "audio/pcm",
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"mp3": "audio/mpeg",
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@ -255,7 +276,7 @@ index 2199e14..d10065f 100644
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- ref_audio=voice_cfg["ref_audio"],
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+ ref_audio=voice_cfg.get("ref_audio"),
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ref_text=voice_cfg.get("ref_text", ""),
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+ instruct=voice_cfg.get("instruct"),
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+ instruct=_merge_instruct(voice_cfg, req.instruct),
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+ voice_clone_prompt=_load_voice_clone_prompt(voice_cfg, req.voice, tts_model),
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+ non_streaming_mode=True,
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+ temperature=voice_cfg.get("temperature", 0.8),
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@ -305,11 +326,11 @@ index 2199e14..d10065f 100644
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if fmt == "wav":
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yield _wav_header(SAMPLE_RATE) # stream with unknown data length
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- async for raw_chunk in _stream_chunks(voice_cfg, req.input):
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+ async for raw_chunk in _stream_chunks(voice_cfg, req.input, req.voice, req.speed):
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+ async for raw_chunk in _stream_chunks(voice_cfg, req.input, req.voice, req.speed, req.instruct):
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yield raw_chunk
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return StreamingResponse(audio_stream(), media_type=content_type)
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@@ -306,16 +483,20 @@ def _parse_args():
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@@ -306,16 +504,20 @@ def _parse_args():
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p.add_argument("--host", default="0.0.0.0", help="Bind host (default: 0.0.0.0)")
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p.add_argument("--port", type=int, default=8000, help="Bind port (default: 8000)")
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p.add_argument("--device", default="cuda", help="Torch device (default: cuda)")
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@ -331,7 +352,7 @@ index 2199e14..d10065f 100644
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with open(args.voices) as f:
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voices = json.load(f)
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default_voice = next(iter(voices))
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@@ -344,6 +525,7 @@ def main():
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@@ -344,6 +546,7 @@ def main():
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args.model,
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device=args.device,
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dtype=torch.bfloat16,
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