This commit is contained in:
Pascal
2026-05-10 16:43:58 +02:00
parent 8ca9622c70
commit 4feb286f04
9 changed files with 685 additions and 3 deletions
+4 -1
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@@ -43,13 +43,16 @@ sys.modules["qwen_tts"] = _qwen_pkg
_core_pkg = types.ModuleType("qwen_tts.core")
_core_pkg.__path__ = [os.path.join(UPSTREAM_ROOT, "qwen_tts", "core")]
# Inject the stubbed core module before any submodule import so the real
# qwen_tts/core/__init__.py never runs : it pulls the V1 25Hz tokenizer that
# imports whisper_encoder, which prints a flash-attn warning at module load.
sys.modules["qwen_tts.core"] = _core_pkg
from qwen_tts.core.tokenizer_12hz.configuration_qwen3_tts_tokenizer_v2 import Qwen3TTSTokenizerV2Config
from qwen_tts.core.tokenizer_12hz.modeling_qwen3_tts_tokenizer_v2 import Qwen3TTSTokenizerV2Model
_core_pkg.Qwen3TTSTokenizerV1Config = _StubV1Config
_core_pkg.Qwen3TTSTokenizerV1Model = _StubV1Model
_core_pkg.Qwen3TTSTokenizerV2Config = Qwen3TTSTokenizerV2Config
_core_pkg.Qwen3TTSTokenizerV2Model = Qwen3TTSTokenizerV2Model
sys.modules["qwen_tts.core"] = _core_pkg
from qwen_tts.core.models.modeling_qwen3_tts import Qwen3TTSForConditionalGeneration
from qwen_tts.core.models.configuration_qwen3_tts import Qwen3TTSConfig
+5 -2
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@@ -169,8 +169,11 @@ def main():
aa, ab = cc.pair("output-audio.bin", DUMP_CPP, DUMP_PT)
print(f"[Cossim] Audio cos: {cc.cos(aa, ab):.6f}")
n = min(audio_cpp.size, audio_pt.size)
print(f"[Cossim] WAV stft_cos: {cc.stft_cos(audio_cpp[:n], audio_pt[:n]):.6f} samples: {n}")
# STFT runs on the f32 bin dumps, not the WAV files : the C++ side writes
# PCM_16 which quantizes the very low amplitudes of greedy short outputs
# to zero, while the bin dumps preserve the raw float buffer.
n = min(aa.size, ab.size)
print(f"[Cossim] WAV stft_cos: {cc.stft_cos(aa.ravel()[:n], ab.ravel()[:n]):.6f} samples: {n}")
if __name__ == "__main__":
main()
+253
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@@ -0,0 +1,253 @@
#!/usr/bin/env python3
"""Cossim debug : C++ qwen-tts vs Python Qwen3-TTS on the Voice Clone Mode B
(ICL) path.
Inputs (relative to CWD = tests/) :
../examples/prompt.txt target text fed to both pipelines
../examples/freeman.wav reference audio for cloning
../examples/freeman.txt transcript of the reference audio
Default mode is greedy on both sides, non_streaming_mode=False which is
the ICL branch upstream : text + codec streams are aligned to the codec
length, the shorter one padded with tts_pad / truncated as needed.
Cote Python the speaker embedding is captured directly via
model.extract_speaker_embedding, and the reference codec frames via
model.speech_tokenizer.encode. Both intermediates land as speaker-emb.bin
and ref-codes.bin and are compared against the C++ side dumps emitted
by pipeline-tts.cpp when --ref-audio and --ref-text are set.
Dumps land in cpp/clone/ (C++) and python/clone/ (Python).
"""
import argparse
import os
import subprocess
import sys
import librosa
import numpy as np
import soundfile as sf
import torch
import cossim_common as cc
MODEL_T = "../models/qwen-talker-1.7b-base-{q}.gguf"
MODEL_CDC_T = "../models/qwen-tokenizer-12hz-{q}.gguf"
CKPT = "../checkpoints/Qwen3-TTS-12Hz-1.7B-Base"
DUMP_CPP = "cpp/clone"
DUMP_PT = "python/clone"
DEFAULT_REF_AUDIO = "../examples/freeman.wav"
DEFAULT_REF_TEXT = "../examples/freeman.txt"
# Mode B adds two pre-talker stages to the standard list : the speaker
# embedding extracted from the reference audio (ECAPA forward, projected to
# talker hidden), and the reference codec frames at 12.5 Hz.
STAGES_CLONE = cc.STAGES_STANDARD + [
("SpeakerEmb", "speaker-emb.bin"),
]
def main():
ap = argparse.ArgumentParser()
ap.add_argument("--prompt", default="../examples/prompt.txt")
ap.add_argument("--ref-audio", default=DEFAULT_REF_AUDIO,
help="reference WAV path for voice cloning")
ap.add_argument("--ref-text-file", default=DEFAULT_REF_TEXT,
help="path to a UTF-8 file with the transcript of ref-audio")
ap.add_argument("--seed", type=int, default=42)
ap.add_argument("--lang", default="english")
ap.add_argument("--quant", default="F32",
help="GGUF quantization suffix (F32, BF16, Q8_0, Q4_K_M)")
ap.add_argument("--out-pt", default=os.path.join(DUMP_PT, "clone-python.wav"))
ap.add_argument("--out-cpp", default=os.path.join(DUMP_CPP, "clone-cpp.wav"))
ap.add_argument("--max-new-tokens", type=int, default=64)
ap.add_argument("--trace", action="store_true",
help="print per sample u and idx for the first 32 samples")
args = ap.parse_args()
cc.ensure_dir(DUMP_PT)
cc.ensure_dir(DUMP_CPP)
os.makedirs(os.path.dirname(args.out_pt) or ".", exist_ok=True)
with open(args.prompt, "r", encoding="utf-8") as f:
text = f.read().strip()
with open(args.ref_text_file, "r", encoding="utf-8") as f:
ref_text = f.read().strip()
print(f"[Input] Prompt: {len(text)} chars: {text[:60]}{'...' if len(text) > 60 else ''}")
print(f"[Input] RefAudio: {args.ref_audio}")
print(f"[Input] RefText: {len(ref_text)} chars: {ref_text[:60]}{'...' if len(ref_text) > 60 else ''}")
print(f"[Input] Lang: {args.lang} Seed: {args.seed} MaxNewTokens: {args.max_new_tokens}")
print(f"[Input] Mode: greedy ICL")
torch.manual_seed(args.seed)
np.random.seed(args.seed)
cc.set_trace(args.trace)
cc.register_qwen3_tts()
device = "cuda" if torch.cuda.is_available() else "cpu"
print(f"[Python] Device: {device}")
model = cc.AutoModel.from_pretrained(
CKPT,
device_map=device,
dtype=torch.float32,
attn_implementation="eager",
).eval()
processor = cc.AutoProcessor.from_pretrained(CKPT, fix_mistral_regex=True)
# Load reference WAV. Resample to 24 kHz if needed since both the speaker
# encoder and the codec tokenizer expect 24 kHz mono input.
ref_wav, ref_sr = sf.read(args.ref_audio, always_2d=False)
if ref_wav.ndim > 1:
ref_wav = ref_wav[:, 0]
ref_wav = ref_wav.astype(np.float32)
target_sr = model.speaker_encoder_sample_rate
if ref_sr != target_sr:
ref_wav = librosa.resample(y=ref_wav, orig_sr=int(ref_sr), target_sr=int(target_sr))
ref_sr = target_sr
print(f"[Python] RefWav: {ref_wav.shape[0]} samples {ref_sr} Hz {ref_wav.shape[0]/ref_sr:.2f}s")
# Extract speaker embedding via ECAPA forward, projected to talker hidden.
spk_emb = model.extract_speaker_embedding(audio=ref_wav, sr=ref_sr)
print(f"[Python] SpeakerEmb shape: {tuple(spk_emb.shape)} dtype: {spk_emb.dtype}")
cc.save_dump(os.path.join(DUMP_PT, "speaker-emb.bin"), spk_emb)
# Encode the reference audio to 16 codebook codes at 12.5 Hz. The encode
# call returns shape [T_codec, K=16] after the internal transpose, while
# the C++ side dumps [K=16, T_codec] row major. We transpose here for a
# straight exact match comparison.
enc = model.speech_tokenizer.encode([ref_wav], sr=int(ref_sr))
ref_code_pt = enc.audio_codes[0]
ref_code_kt = ref_code_pt.transpose(0, 1).contiguous()
print(f"[Python] RefCodes shape: {tuple(ref_code_kt.shape)} (K, T_codec)")
cc.save_dump_i32(os.path.join(DUMP_PT, "ref-codes.bin"), ref_code_kt)
# Tokenize the utterance and the reference text.
assistant_text = f"<|im_start|>assistant\n{text}<|im_end|>\n<|im_start|>assistant\n"
inp_utt = processor(text=assistant_text, return_tensors="pt", padding=True)
input_ids = inp_utt["input_ids"].to(device)
if input_ids.dim() == 1:
input_ids = input_ids.unsqueeze(0)
print(f"[Python] InputIds shape: {tuple(input_ids.shape)}")
cc.save_dump_i32(os.path.join(DUMP_PT, "prompt-ids.bin"), input_ids[0])
ref_text_wrap = f"<|im_start|>assistant\n{ref_text}<|im_end|>\n"
inp_ref = processor(text=ref_text_wrap, return_tensors="pt", padding=True)
ref_ids = inp_ref["input_ids"].to(device)
if ref_ids.dim() == 1:
ref_ids = ref_ids.unsqueeze(0)
print(f"[Python] RefIds shape: {tuple(ref_ids.shape)}")
cc.save_dump_i32(os.path.join(DUMP_PT, "ref-ids.bin"), ref_ids[0])
cc.install_hooks(model, DUMP_PT)
# Custom subtalker_* kwargs are forwarded to talker.forward but not
# declared on GenerationMixin, so transformers 4.57 rejects them under
# the strict validator. Disable it on the talker only.
model.talker._validate_model_kwargs = lambda *a, **k: None
gen_kwargs = dict(
do_sample = False,
top_k = 1,
top_p = 1.0,
temperature = 1.0,
subtalker_dosample = False,
subtalker_top_k = 1,
subtalker_top_p = 1.0,
subtalker_temperature = 1.0,
repetition_penalty = 1.0,
)
# voice_clone_prompt dict mirrors what _prompt_items_to_voice_clone_prompt
# builds for a single ICL prompt item : ref_code is the [T_codec, K]
# tensor, ref_spk_embedding is the [hidden] tensor, x_vector_only=False
# and icl_mode=True together select the mode B branch upstream.
voice_clone_prompt_dict = dict(
ref_code = [ref_code_pt],
ref_spk_embedding = [spk_emb],
x_vector_only_mode = [False],
icl_mode = [True],
)
talker_codes_list, _ = model.generate(
input_ids=[input_ids],
ref_ids=[ref_ids],
voice_clone_prompt=voice_clone_prompt_dict,
languages=[args.lang],
non_streaming_mode=False,
max_new_tokens=args.max_new_tokens,
**gen_kwargs,
)
codes = talker_codes_list[0]
print(f"[Python] Codes shape: {tuple(codes.shape)} (T_frames, num_code_groups)")
cc.save_dump_i32(os.path.join(DUMP_PT, "codes-full.bin"), codes)
cc.save_dump_i32(os.path.join(DUMP_PT, "codes-step0.bin"), codes[0])
# The decode path prepends the reference codes and cuts the matching
# audio prefix afterwards, mirroring generate_voice_clone exactly so the
# produced WAV only covers the freshly generated portion.
cat_codes = torch.cat([ref_code_pt.to(codes.device), codes], dim=0)
wavs, fs = model.speech_tokenizer.decode([{"audio_codes": cat_codes}])
full_wav = np.asarray(wavs[0], dtype=np.float32)
ref_len = int(ref_code_pt.shape[0])
total_len = int(cat_codes.shape[0])
cut = int(ref_len / max(total_len, 1) * full_wav.shape[0])
audio_pt = full_wav[cut:]
sf.write(args.out_pt, audio_pt, fs, subtype="FLOAT")
cc.save_dump(os.path.join(DUMP_PT, "output-audio.bin"), audio_pt)
print(f"[Python] Audio: {audio_pt.shape[0]} samples {fs} Hz {audio_pt.shape[0]/fs:.2f}s -> {args.out_pt}")
if not os.path.isfile(cc.BIN):
print(f"[Cossim] FATAL: {cc.BIN} not found, build qwen-tts first")
sys.exit(1)
model_lm = MODEL_T.format(q=args.quant)
model_cdc = MODEL_CDC_T.format(q=args.quant)
for p in (model_lm, model_cdc):
if not os.path.isfile(p):
print(f"[Cossim] FATAL: GGUF not found: {p}")
sys.exit(1)
print(f"[Quant] {args.quant} -> {model_lm} + {model_cdc}")
del model
if device == "cuda":
torch.cuda.empty_cache()
cmd = [
cc.BIN,
"--model", model_lm,
"--codec", model_cdc,
"--seed", str(args.seed),
"--text", text,
"--ref-audio", args.ref_audio,
"--ref-text", ref_text,
"--lang", args.lang,
"--max-new", str(args.max_new_tokens),
"--dump", DUMP_CPP,
"-o", args.out_cpp,
"--greedy",
]
print(f"[GGML] Cmd: {' '.join(cmd[:6])} --text [...] --ref-audio {args.ref_audio} --ref-text [...] --lang {args.lang} --max-new {args.max_new_tokens} --dump {DUMP_CPP} -o {args.out_cpp} --greedy")
r = subprocess.run(cmd)
if r.returncode != 0:
sys.exit(r.returncode)
audio_cpp, sr = sf.read(args.out_cpp)
if audio_cpp.ndim > 1:
audio_cpp = audio_cpp[:, 0]
audio_cpp = audio_cpp.astype(np.float32)
print(f"[GGML] Audio: {audio_cpp.shape[0]} samples {sr} Hz {audio_cpp.shape[0]/sr:.2f}s -> {args.out_cpp}")
cc.compare_exact_i32("prompt-ids.bin", DUMP_CPP, DUMP_PT, "PromptIDs")
cc.compare_exact_i32("ref-codes.bin", DUMP_CPP, DUMP_PT, "RefCodes")
cc.compare_stages(STAGES_CLONE, DUMP_CPP, DUMP_PT)
cc.compare_exact_i32("codes-full.bin", DUMP_CPP, DUMP_PT, "CodesFull")
aa, ab = cc.pair("output-audio.bin", DUMP_CPP, DUMP_PT)
print(f"[Cossim] Audio cos: {cc.cos(aa, ab):.6f}")
n = min(aa.size, ab.size)
print(f"[Cossim] WAV stft_cos: {cc.stft_cos(aa.ravel()[:n], ab.ravel()[:n]):.6f} samples: {n}")
if __name__ == "__main__":
main()
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@@ -0,0 +1,8 @@
#!/bin/bash
for backend in CUDA0 Vulkan0 CPU; do
for quant in F32 BF16 Q8_0 Q4_K_M; do
GGML_BACKEND=$backend ./debug-clone-cossim.py --quant $quant \
2>&1 | tee clone-${backend}-${quant}.log
done
done
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@@ -0,0 +1,200 @@
#!/usr/bin/env python3
"""Cossim debug : C++ qwen-tts vs Python Qwen3-TTS on the CustomVoice 1.7B path.
Inputs (relative to CWD = tests/) :
../examples/prompt.txt target text fed to both pipelines
--speaker flag speaker preset key, default mirrors customvoice.sh
Default mode is greedy (do_sample=False on both sides). The speaker
preset is passed straight through model.generate as `speakers=[name]`
on the Python side, mirroring qwen_tts.inference.qwen3_tts_model.
generate_custom_voice. The speaker codec embedding row slips between
think_eos and codec_pad in the prefill, growing the prefill by one
codec vector. Cote C++ the same insertion happens inside prompt_builder.
Optional --instruct adds a style instruction in front of the prompt.
The 1.7B CustomVoice accepts it, the 0.6B does not.
Dumps land in cpp/customvoice/ (C++) and python/customvoice/ (Python).
"""
import argparse
import os
import subprocess
import sys
import numpy as np
import soundfile as sf
import torch
import cossim_common as cc
MODEL_T = "../models/qwen-talker-1.7b-customvoice-{q}.gguf"
MODEL_CDC_T = "../models/qwen-tokenizer-12hz-{q}.gguf"
CKPT = "../checkpoints/Qwen3-TTS-12Hz-1.7B-CustomVoice"
DUMP_CPP = "cpp/customvoice"
DUMP_PT = "python/customvoice"
DEFAULT_SPEAKER = "vivian"
def main():
ap = argparse.ArgumentParser()
ap.add_argument("--prompt", default="../examples/prompt.txt")
ap.add_argument("--speaker", default=DEFAULT_SPEAKER,
help="speaker preset key (lowercase), validated by the model")
ap.add_argument("--instruct", default="",
help="optional style instruction, empty disables the instruct prefix")
ap.add_argument("--seed", type=int, default=42)
ap.add_argument("--lang", default="english")
ap.add_argument("--quant", default="F32",
help="GGUF quantization suffix (F32, BF16, Q8_0, Q4_K_M)")
ap.add_argument("--out-pt", default=os.path.join(DUMP_PT, "customvoice-python.wav"))
ap.add_argument("--out-cpp", default=os.path.join(DUMP_CPP, "customvoice-cpp.wav"))
ap.add_argument("--max-new-tokens", type=int, default=64)
ap.add_argument("--trace", action="store_true",
help="print per sample u and idx for the first 32 samples")
args = ap.parse_args()
cc.ensure_dir(DUMP_PT)
cc.ensure_dir(DUMP_CPP)
os.makedirs(os.path.dirname(args.out_pt) or ".", exist_ok=True)
with open(args.prompt, "r", encoding="utf-8") as f:
text = f.read().strip()
print(f"[Input] Prompt: {len(text)} chars: {text[:60]}{'...' if len(text) > 60 else ''}")
print(f"[Input] Speaker: {args.speaker}")
if args.instruct:
print(f"[Input] Instruct: {args.instruct}")
print(f"[Input] Lang: {args.lang} Seed: {args.seed} MaxNewTokens: {args.max_new_tokens}")
print(f"[Input] Mode: greedy")
torch.manual_seed(args.seed)
np.random.seed(args.seed)
cc.set_trace(args.trace)
cc.register_qwen3_tts()
device = "cuda" if torch.cuda.is_available() else "cpu"
print(f"[Python] Device: {device}")
model = cc.AutoModel.from_pretrained(
CKPT,
device_map=device,
dtype=torch.float32,
attn_implementation="eager",
).eval()
processor = cc.AutoProcessor.from_pretrained(CKPT, fix_mistral_regex=True)
# Utterance text wrapped as assistant role.
assistant_text = f"<|im_start|>assistant\n{text}<|im_end|>\n<|im_start|>assistant\n"
inp_utt = processor(text=assistant_text, return_tensors="pt", padding=True)
input_ids = inp_utt["input_ids"].to(device)
if input_ids.dim() == 1:
input_ids = input_ids.unsqueeze(0)
print(f"[Python] InputIds shape: {tuple(input_ids.shape)}")
cc.save_dump_i32(os.path.join(DUMP_PT, "prompt-ids.bin"), input_ids[0])
# Optional instruct, None when empty so the talker forward keeps the
# standard CustomVoice prefill without any instruct prefix.
instruct_ids_arg = None
if args.instruct:
instruct_text = f"<|im_start|>user\n{args.instruct}<|im_end|>\n"
inp_ins = processor(text=instruct_text, return_tensors="pt", padding=True)
instruct_ids = inp_ins["input_ids"].to(device)
if instruct_ids.dim() == 1:
instruct_ids = instruct_ids.unsqueeze(0)
print(f"[Python] InstructIds shape: {tuple(instruct_ids.shape)}")
cc.save_dump_i32(os.path.join(DUMP_PT, "instruct-ids.bin"), instruct_ids[0])
instruct_ids_arg = [instruct_ids]
cc.install_hooks(model, DUMP_PT)
# Custom subtalker_* kwargs are forwarded to talker.forward but not
# declared on GenerationMixin, so transformers 4.57 rejects them under
# the strict validator. Disable it on the talker only.
model.talker._validate_model_kwargs = lambda *a, **k: None
gen_kwargs = dict(
do_sample = False,
top_k = 1,
top_p = 1.0,
temperature = 1.0,
subtalker_dosample = False,
subtalker_top_k = 1,
subtalker_top_p = 1.0,
subtalker_temperature = 1.0,
repetition_penalty = 1.0,
)
talker_codes_list, _ = model.generate(
input_ids=[input_ids],
instruct_ids=instruct_ids_arg,
languages=[args.lang],
speakers=[args.speaker],
non_streaming_mode=True,
max_new_tokens=args.max_new_tokens,
**gen_kwargs,
)
codes = talker_codes_list[0]
print(f"[Python] Codes shape: {tuple(codes.shape)} (T_frames, num_code_groups)")
cc.save_dump_i32(os.path.join(DUMP_PT, "codes-full.bin"), codes)
cc.save_dump_i32(os.path.join(DUMP_PT, "codes-step0.bin"), codes[0])
wavs, fs = model.speech_tokenizer.decode([{"audio_codes": codes}])
audio_pt = np.asarray(wavs[0], dtype=np.float32)
sf.write(args.out_pt, audio_pt, fs, subtype="FLOAT")
cc.save_dump(os.path.join(DUMP_PT, "output-audio.bin"), audio_pt)
print(f"[Python] Audio: {audio_pt.shape[0]} samples {fs} Hz {audio_pt.shape[0]/fs:.2f}s -> {args.out_pt}")
if not os.path.isfile(cc.BIN):
print(f"[Cossim] FATAL: {cc.BIN} not found, build qwen-tts first")
sys.exit(1)
model_lm = MODEL_T.format(q=args.quant)
model_cdc = MODEL_CDC_T.format(q=args.quant)
for p in (model_lm, model_cdc):
if not os.path.isfile(p):
print(f"[Cossim] FATAL: GGUF not found: {p}")
sys.exit(1)
print(f"[Quant] {args.quant} -> {model_lm} + {model_cdc}")
del model
if device == "cuda":
torch.cuda.empty_cache()
cmd = [
cc.BIN,
"--model", model_lm,
"--codec", model_cdc,
"--seed", str(args.seed),
"--text", text,
"--speaker", args.speaker,
"--lang", args.lang,
"--max-new", str(args.max_new_tokens),
"--dump", DUMP_CPP,
"-o", args.out_cpp,
"--greedy",
]
if args.instruct:
cmd[-1:-1] = ["--instruct", args.instruct]
print(f"[GGML] Cmd: {' '.join(cmd)}")
r = subprocess.run(cmd)
if r.returncode != 0:
sys.exit(r.returncode)
audio_cpp, sr = sf.read(args.out_cpp)
if audio_cpp.ndim > 1:
audio_cpp = audio_cpp[:, 0]
audio_cpp = audio_cpp.astype(np.float32)
print(f"[GGML] Audio: {audio_cpp.shape[0]} samples {sr} Hz {audio_cpp.shape[0]/sr:.2f}s -> {args.out_cpp}")
cc.compare_exact_i32("prompt-ids.bin", DUMP_CPP, DUMP_PT, "PromptIDs")
cc.compare_stages(cc.STAGES_STANDARD, DUMP_CPP, DUMP_PT)
cc.compare_exact_i32("codes-full.bin", DUMP_CPP, DUMP_PT, "CodesFull")
aa, ab = cc.pair("output-audio.bin", DUMP_CPP, DUMP_PT)
print(f"[Cossim] Audio cos: {cc.cos(aa, ab):.6f}")
n = min(aa.size, ab.size)
print(f"[Cossim] WAV stft_cos: {cc.stft_cos(aa.ravel()[:n], ab.ravel()[:n]):.6f} samples: {n}")
if __name__ == "__main__":
main()
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#!/bin/bash
for backend in CUDA0 Vulkan0 CPU; do
for quant in F32 BF16 Q8_0 Q4_K_M; do
GGML_BACKEND=$backend ./debug-customvoice-cossim.py --quant $quant \
2>&1 | tee customvoice-${backend}-${quant}.log
done
done
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#!/usr/bin/env python3
"""Cossim debug : C++ qwen-tts vs Python Qwen3-TTS on the VoiceDesign 1.7B path.
Inputs (relative to CWD = tests/) :
../examples/prompt.txt target text fed to both pipelines
--instruct flag style instruction, default mirrors tts.sh
Default mode is greedy (do_sample=False on both sides). Cote Python the
utterance and the instruction are tokenized separately and passed to
model.generate as input_ids and instruct_ids, mirroring exactly what
qwen_tts.inference.qwen3_tts_model.generate_voice_design does. Cote C++
both strings are passed through --text and --instruct, the prompt builder
wraps and tokenizes them in the same order.
Dumps land in cpp/tts/ (C++) and python/tts/ (Python).
"""
import argparse
import os
import subprocess
import sys
import numpy as np
import soundfile as sf
import torch
import cossim_common as cc
MODEL_T = "../models/qwen-talker-1.7b-voicedesign-{q}.gguf"
MODEL_CDC_T = "../models/qwen-tokenizer-12hz-{q}.gguf"
CKPT = "../checkpoints/Qwen3-TTS-12Hz-1.7B-VoiceDesign"
DUMP_CPP = "cpp/tts"
DUMP_PT = "python/tts"
DEFAULT_INSTRUCT = "male, young adult, moderate pitch"
def main():
ap = argparse.ArgumentParser()
ap.add_argument("--prompt", default="../examples/prompt.txt")
ap.add_argument("--instruct", default=DEFAULT_INSTRUCT,
help="natural language style instruction")
ap.add_argument("--seed", type=int, default=42)
ap.add_argument("--lang", default="english")
ap.add_argument("--quant", default="F32",
help="GGUF quantization suffix (F32, BF16, Q8_0, Q4_K_M)")
ap.add_argument("--out-pt", default=os.path.join(DUMP_PT, "tts-python.wav"))
ap.add_argument("--out-cpp", default=os.path.join(DUMP_CPP, "tts-cpp.wav"))
ap.add_argument("--max-new-tokens", type=int, default=64)
ap.add_argument("--trace", action="store_true",
help="print per sample u and idx for the first 32 samples")
args = ap.parse_args()
cc.ensure_dir(DUMP_PT)
cc.ensure_dir(DUMP_CPP)
os.makedirs(os.path.dirname(args.out_pt) or ".", exist_ok=True)
with open(args.prompt, "r", encoding="utf-8") as f:
text = f.read().strip()
print(f"[Input] Prompt: {len(text)} chars: {text[:60]}{'...' if len(text) > 60 else ''}")
print(f"[Input] Instruct: {args.instruct}")
print(f"[Input] Lang: {args.lang} Seed: {args.seed} MaxNewTokens: {args.max_new_tokens}")
print(f"[Input] Mode: greedy")
torch.manual_seed(args.seed)
np.random.seed(args.seed)
cc.set_trace(args.trace)
cc.register_qwen3_tts()
device = "cuda" if torch.cuda.is_available() else "cpu"
print(f"[Python] Device: {device}")
model = cc.AutoModel.from_pretrained(
CKPT,
device_map=device,
dtype=torch.float32,
attn_implementation="eager",
).eval()
processor = cc.AutoProcessor.from_pretrained(CKPT, fix_mistral_regex=True)
# Utterance text wrapped as assistant role, mirrors _build_assistant_text
# in qwen_tts.inference.qwen3_tts_model.
assistant_text = f"<|im_start|>assistant\n{text}<|im_end|>\n<|im_start|>assistant\n"
inp_utt = processor(text=assistant_text, return_tensors="pt", padding=True)
input_ids = inp_utt["input_ids"].to(device)
if input_ids.dim() == 1:
input_ids = input_ids.unsqueeze(0)
print(f"[Python] InputIds shape: {tuple(input_ids.shape)}")
cc.save_dump_i32(os.path.join(DUMP_PT, "prompt-ids.bin"), input_ids[0])
# Instruct text wrapped as user role, mirrors _build_instruct_text.
instruct_text = f"<|im_start|>user\n{args.instruct}<|im_end|>\n"
inp_ins = processor(text=instruct_text, return_tensors="pt", padding=True)
instruct_ids = inp_ins["input_ids"].to(device)
if instruct_ids.dim() == 1:
instruct_ids = instruct_ids.unsqueeze(0)
print(f"[Python] InstructIds shape: {tuple(instruct_ids.shape)}")
cc.save_dump_i32(os.path.join(DUMP_PT, "instruct-ids.bin"), instruct_ids[0])
cc.install_hooks(model, DUMP_PT)
# Custom subtalker_* kwargs are forwarded to talker.forward but not
# declared on GenerationMixin, so transformers 4.57 rejects them under
# the strict validator. Disable it on the talker only.
model.talker._validate_model_kwargs = lambda *a, **k: None
gen_kwargs = dict(
do_sample = False,
top_k = 1,
top_p = 1.0,
temperature = 1.0,
subtalker_dosample = False,
subtalker_top_k = 1,
subtalker_top_p = 1.0,
subtalker_temperature = 1.0,
repetition_penalty = 1.0,
)
talker_codes_list, _ = model.generate(
input_ids=[input_ids],
instruct_ids=[instruct_ids],
languages=[args.lang],
non_streaming_mode=True,
max_new_tokens=args.max_new_tokens,
**gen_kwargs,
)
codes = talker_codes_list[0]
print(f"[Python] Codes shape: {tuple(codes.shape)} (T_frames, num_code_groups)")
cc.save_dump_i32(os.path.join(DUMP_PT, "codes-full.bin"), codes)
cc.save_dump_i32(os.path.join(DUMP_PT, "codes-step0.bin"), codes[0])
wavs, fs = model.speech_tokenizer.decode([{"audio_codes": codes}])
audio_pt = np.asarray(wavs[0], dtype=np.float32)
sf.write(args.out_pt, audio_pt, fs, subtype="FLOAT")
cc.save_dump(os.path.join(DUMP_PT, "output-audio.bin"), audio_pt)
print(f"[Python] Audio: {audio_pt.shape[0]} samples {fs} Hz {audio_pt.shape[0]/fs:.2f}s -> {args.out_pt}")
if not os.path.isfile(cc.BIN):
print(f"[Cossim] FATAL: {cc.BIN} not found, build qwen-tts first")
sys.exit(1)
model_lm = MODEL_T.format(q=args.quant)
model_cdc = MODEL_CDC_T.format(q=args.quant)
for p in (model_lm, model_cdc):
if not os.path.isfile(p):
print(f"[Cossim] FATAL: GGUF not found: {p}")
sys.exit(1)
print(f"[Quant] {args.quant} -> {model_lm} + {model_cdc}")
del model
if device == "cuda":
torch.cuda.empty_cache()
cmd = [
cc.BIN,
"--model", model_lm,
"--codec", model_cdc,
"--seed", str(args.seed),
"--text", text,
"--instruct", args.instruct,
"--lang", args.lang,
"--max-new", str(args.max_new_tokens),
"--dump", DUMP_CPP,
"-o", args.out_cpp,
"--greedy",
]
print(f"[GGML] Cmd: {' '.join(cmd)}")
r = subprocess.run(cmd)
if r.returncode != 0:
sys.exit(r.returncode)
audio_cpp, sr = sf.read(args.out_cpp)
if audio_cpp.ndim > 1:
audio_cpp = audio_cpp[:, 0]
audio_cpp = audio_cpp.astype(np.float32)
print(f"[GGML] Audio: {audio_cpp.shape[0]} samples {sr} Hz {audio_cpp.shape[0]/sr:.2f}s -> {args.out_cpp}")
cc.compare_exact_i32("prompt-ids.bin", DUMP_CPP, DUMP_PT, "PromptIDs")
cc.compare_stages(cc.STAGES_STANDARD, DUMP_CPP, DUMP_PT)
cc.compare_exact_i32("codes-full.bin", DUMP_CPP, DUMP_PT, "CodesFull")
aa, ab = cc.pair("output-audio.bin", DUMP_CPP, DUMP_PT)
print(f"[Cossim] Audio cos: {cc.cos(aa, ab):.6f}")
n = min(aa.size, ab.size)
print(f"[Cossim] WAV stft_cos: {cc.stft_cos(aa.ravel()[:n], ab.ravel()[:n]):.6f} samples: {n}")
if __name__ == "__main__":
main()
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#!/bin/bash
for backend in CUDA0 Vulkan0 CPU; do
for quant in F32 BF16 Q8_0 Q4_K_M; do
GGML_BACKEND=$backend ./debug-tts-cossim.py --quant $quant \
2>&1 | tee tts-${backend}-${quant}.log
done
done