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@@ -30,8 +30,6 @@ outputs/
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media-files/video/*.mp4
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media-files/video/*.mp4
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!media-files/video/*-480p.mp4
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!media-files/video/*-480p.mp4
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# Small demo evidence clips — keep committed so results are reviewable
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# Compressed exports of demucs stems — keep committed for review without
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# without local playback (unlike demucs's separated/ stems, which stay local)
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# local playback (the raw separated/*.wav stems stay local, per *.wav above)
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!tts/reference/*.wav
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!demucs/*.mp3
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!tts/output/
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!tts/output/*.wav
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@@ -23,10 +23,5 @@ Chain the three demos together on `media-files/` into an end-to-end pipeline:
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3. `tts` — clone the original speaker's voice (from the isolated vocal stem) and synthesize
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3. `tts` — clone the original speaker's voice (from the isolated vocal stem) and synthesize
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the translated text in that voice, then remix with the original instrumental stem
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the translated text in that voice, then remix with the original instrumental stem
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All three stages now have a working, if narrow, proof of concept:
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Stem separation and transcription are already working end to end (see `demucs/README.md` and
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`whisperx/README.md`); translation and voice-cloning TTS are not yet implemented.
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- Stem separation (`demucs/README.md`) and transcription (`whisperx/README.md`) run end to end
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on the full sample clip.
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- Translation + voice-cloning TTS (`tts/README.md`) works for a single short sentence
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(English → Spanish, cloned onto the isolated narrator voice via MOSS-TTS). Not yet wired up
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to run over the full transcript or to remix with the instrumental stem.
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@@ -17,6 +17,10 @@ Took ~12s for 11 minutes of audio. Outputs `vocals.wav` and `no_vocals.wav` (44.
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duration matches source) under `separated/htdemucs/<track-name>/` — gitignored, left on the GPU
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duration matches source) under `separated/htdemucs/<track-name>/` — gitignored, left on the GPU
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box at `/root/projects/audio-demo/repo/demucs/separated/`.
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box at `/root/projects/audio-demo/repo/demucs/separated/`.
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Full-length MP3 versions (128kbps, ~10MB each) are committed as `vocals.mp3` and
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`no_vocals.mp3` in this directory so both stems are reviewable directly from Gitea without
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needing local playback or GPU-box access.
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**Known gotcha:** demucs 4.1.0's package metadata doesn't declare `numpy` as a dependency even
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**Known gotcha:** demucs 4.1.0's package metadata doesn't declare `numpy` as a dependency even
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though `demucs/transformer.py` imports it directly — `pip install demucs` alone leaves the env
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though `demucs/transformer.py` imports it directly — `pip install demucs` alone leaves the env
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broken (`ModuleNotFoundError: No module named 'numpy'` at runtime). Install `numpy` explicitly
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broken (`ModuleNotFoundError: No module named 'numpy'` at runtime). Install `numpy` explicitly
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@@ -1,55 +1,3 @@
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# tts
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# tts
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Text-to-speech service demos and proofs-of-concept.
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Text-to-speech service demos and proofs-of-concept.
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## Voice-clone + translate demo (MOSS-TTS)
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`clone_and_translate_demo.py` translates a short English sentence spoken by the narrator in
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our sample clip to Spanish (via `Helsinki-NLP/opus-mt-en-es`), then uses
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[MOSS-TTS](https://github.com/OpenMOSS/MOSS-TTS) to synthesize the Spanish text in the
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narrator's cloned voice. The reference audio (`reference/narrator_ref_en.wav`) is trimmed
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from demucs's isolated vocals stem for the exact same sentence, so the English original and
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the Spanish clone can be compared directly — this is the first working slice of the full
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audio-translation pipeline sketched in the top-level README.
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### Setup
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MOSS-TTS ships as a full repo you install editable (not a plain `pip install`), and it isn't
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part of this git repo — it's vendored outside it since it's a large third-party project:
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```bash
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git clone --depth 1 https://github.com/OpenMOSS/MOSS-TTS.git /path/to/vendor/MOSS-TTS
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python3 -m venv .venv && . .venv/bin/activate
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pip install --extra-index-url https://download.pytorch.org/whl/cu128 \
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-e "/path/to/vendor/MOSS-TTS[torch-runtime]"
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pip install sentencepiece # for the MarianMT translation step
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```
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### Running
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```bash
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SITE=$(python3 -c "import site; print(site.getsitepackages()[0])")
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export LD_LIBRARY_PATH=$(find "$SITE/nvidia" -maxdepth 2 -type d -name lib | paste -sd: -)
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python3 clone_and_translate_demo.py
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```
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Output goes to `output/narrator_cloned_es.wav` (48kHz stereo, ~11s).
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### Gotchas found running this on an RTX 3090
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- **The flagship 8B `MOSS-TTS-v1.5` checkpoint OOMs on a 24GB 3090** — loading the model plus
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its audio tokenizer already uses ~22.5GB, leaving no room for generation. Use the 4B
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`MOSS-TTS-Local-Transformer-v1.5` checkpoint instead (also natively 48kHz stereo); it's
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the option the project itself recommends for smaller GPUs.
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- **`torchaudio.load()`/`.save()` fail with `Could not load libtorchcodec`** on this setup.
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The real cause (visible via `ctypes.CDLL` directly, not the wrapped exception) is a missing
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`libnppicc.so.12` — torchcodec links against NVIDIA's NPP library for its video-decode path,
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but `moss-tts`'s dependencies don't pull it in since MOSS-TTS only needs audio. Fix:
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`pip install nvidia-npp-cu12`, then add all `nvidia/*/lib` wheel directories to
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`LD_LIBRARY_PATH` before running (see the `SITE=...` line above) — the pip-installed CUDA
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libraries aren't on the default linker search path. This also affects code paths *inside*
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MOSS-TTS's own processor (it uses `torchaudio.load()` to read reference audio), so the fix
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has to be in place before the script imports it, not just around your own save calls.
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- Saving the generated audio with plain `soundfile.write()` instead of `torchaudio.save()`
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avoids depending on torchcodec at all for the output side, but doesn't help with the
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reference-audio loading path above.
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@@ -1,120 +0,0 @@
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"""
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Voice-clone + translate demo.
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Takes a short English sentence spoken by the narrator in our sample clip
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(the same sentence used as the reference audio, for an apples-to-apples
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comparison), machine-translates it to Spanish, then uses MOSS-TTS to
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synthesize the Spanish text in the narrator's cloned voice.
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Reference audio (tts/reference/narrator_ref_en.wav) was trimmed from
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demucs's isolated vocals stem for this same clip.
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"""
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import importlib.util
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from pathlib import Path
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import soundfile as sf
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import torch
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from transformers import AutoModel, AutoProcessor, MarianMTModel, MarianTokenizer
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HERE = Path(__file__).parent
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REFERENCE_AUDIO = HERE / "reference" / "narrator_ref_en.wav"
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OUTPUT_DIR = HERE / "output"
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TEXT_EN = (
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"We shall show you in this film typical examples of children who have "
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"been well brought up, the kind who grow up to be strong, healthy, and "
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"happy like these young folks."
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)
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device = "cuda" if torch.cuda.is_available() else "cpu"
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dtype = torch.bfloat16 if device == "cuda" else torch.float32
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def translate_to_spanish(text: str) -> str:
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model_name = "Helsinki-NLP/opus-mt-en-es"
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tokenizer = MarianTokenizer.from_pretrained(model_name)
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model = MarianMTModel.from_pretrained(model_name).to(device)
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batch = tokenizer([text], return_tensors="pt", padding=True).to(device)
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generated = model.generate(**batch, max_new_tokens=256)
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return tokenizer.decode(generated[0], skip_special_tokens=True)
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def resolve_attn_implementation() -> str:
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if (
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device == "cuda"
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and importlib.util.find_spec("flash_attn") is not None
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and dtype in {torch.float16, torch.bfloat16}
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):
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major, _ = torch.cuda.get_device_capability()
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if major >= 8:
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return "flash_attention_2"
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return "sdpa" if device == "cuda" else "eager"
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def main() -> None:
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text_es = translate_to_spanish(TEXT_EN)
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print(f"[EN] {TEXT_EN}")
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print(f"[ES] {text_es}")
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# Disable the broken cuDNN SDPA backend (per MOSS-TTS README).
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torch.backends.cuda.enable_cudnn_sdp(False)
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torch.backends.cuda.enable_flash_sdp(True)
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torch.backends.cuda.enable_mem_efficient_sdp(True)
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torch.backends.cuda.enable_math_sdp(True)
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# The flagship MOSS-TTS-v1.5 (8B) doesn't fit alongside its audio
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# tokenizer + generation overhead on a 24GB 3090 (OOMs at ~23.3GB used).
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# The 4B Local-Transformer variant is the recommended smaller-GPU option
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# and also natively outputs 48kHz stereo.
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pretrained = "OpenMOSS-Team/MOSS-TTS-Local-Transformer-v1.5"
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attn_implementation = resolve_attn_implementation()
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print(f"[INFO] attn_implementation={attn_implementation}")
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processor = AutoProcessor.from_pretrained(pretrained, trust_remote_code=True)
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processor.audio_tokenizer = processor.audio_tokenizer.to(device)
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model = AutoModel.from_pretrained(
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pretrained,
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trust_remote_code=True,
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attn_implementation=attn_implementation,
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dtype=dtype,
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).to(device)
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model.eval()
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conversation = [
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processor.build_user_message(
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text=text_es,
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language="Spanish",
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reference=[str(REFERENCE_AUDIO)],
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)
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]
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OUTPUT_DIR.mkdir(exist_ok=True, parents=True)
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with torch.no_grad():
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batch = processor([conversation], mode="generation")
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outputs = model.generate(
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input_ids=batch["input_ids"].to(device),
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attention_mask=batch["attention_mask"].to(device),
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max_new_tokens=4096,
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do_sample=True,
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audio_temperature=1.7,
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audio_top_p=0.8,
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audio_top_k=25,
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audio_repetition_penalty=1.0,
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)
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message = processor.decode(outputs)[0]
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audio = message.audio_codes_list[0]
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# torchaudio.save() depends on torchcodec, whose bundled shared libs
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# failed to load in this environment; soundfile (already a MOSS-TTS
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# dependency) sidesteps that entirely.
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data = audio.detach().cpu().to(torch.float32).numpy()
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if data.ndim == 2:
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data = data.T # soundfile wants (frames, channels), not (channels, frames)
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out_path = OUTPUT_DIR / "narrator_cloned_es.wav"
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sf.write(str(out_path), data, processor.model_config.sampling_rate)
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print(f"[INFO] Saved {out_path}")
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if __name__ == "__main__":
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main()
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