The upstream openai_server.py gained --max-seq-len after v5 was built.
Add it to the patch so every build gets explicit sequence length control.
Also add max_seq_len=args.max_seq_len to FasterQwen3TTS.from_pretrained()
so the argument is actually used.
Remove || true from git apply so patch failures fail the build instead
of silently producing a broken image.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
--max-seq-len is not in the current :latest image's openai_server.py (upstream
added it after v5 was built). Remove it from voiceclone command so the
container starts; upstream default (4096) applies. Will be re-added explicitly
once v6 image is built with current upstream.
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
The VoiceClone server was using non_streaming_mode=False, a mode designed
for streaming LLM->TTS pipelines. In that mode only one text token enters
the model's KV cache during prefill; the rest feed via trailing_text_hiddens
at one step per codec frame. For a 54-word paragraph this provides only ~4s
of text guidance for ~18s of speech — 77% generated with no text conditioning.
Without text context the model free-runs and drifts, sometimes changing gender.
Fix: switch to non_streaming_mode=True (already the default for VoiceDesign
and CustomVoice) so the full text is in the prefill throughout generation.
Also lower default temperature 0.9->0.8 and add top_p=0.9 to reduce
accumulated sampling noise over long runs. Temperature, top_k, and top_p
are now configurable per voice in voices.json.
- patches/openai_server.patch: updated for new upstream HEAD; both streaming
(WAV/PCM) and non-streaming (MP3) paths now use non_streaming_mode=True
- config/run_server.py: align warmup call to non_streaming_mode=True
- README.md: bump image tags v4->v5, add changelog section
- Version: v5
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
- Add run_customvoice_server.py: OpenAI-compatible TTS server for Qwen3-TTS
CustomVoice models with speaker-based voice selection and per-request
language/instruct/max_new_tokens overrides
- Add customvoice_voices.json: voice config for 9 built-in speakers across
English, Chinese, Japanese, and Korean
- Add benchmark_api.py: API benchmarking tool reporting TTFA, total time,
RTF, and speed multiplier across short/medium/long sentences
- Refactor run_voicedesign_server.py: extract _request_generation_params()
helper, add per-request language/instruct/max_new_tokens override support,
and wire DEFAULT_MAX_NEW_TOKENS to --max-seq-len arg
- Add faster-qwen3-tts.code-workspace for VS Code project config
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
- generate_voices.py: scan /config/speakers and /voices recursively,
support .ogg and .m4a (M4A auto-converted via ffmpeg), sanitise voice IDs
- auto_transcribe.py: scan both host paths recursively, support all formats,
use parakeet-asr on port 8010
- docker-compose.yml: mount /home/sparky/Projekte/TTS_Voices/speakers as
/voices, add faster-qwen3-tts-voicedesign service on port 8021
- run_voicedesign_server.py: OpenAI-compatible server for VoiceDesign model
- voicedesign_voices.json: 8 British/German VoiceDesign voices
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
- run_server.py: warm up CUDA graphs at server startup via lifespan
event so the first real request does not pay the 7-8s graph-
compilation penalty; uses modern lifespan API instead of deprecated
on_event
- generate_voices.py: set chunk_size=4 per voice so streaming clients
receive first audio after ~333ms instead of ~1s
- docker-compose.yml: add --max-seq-len 2048 (halves static KV cache,
reduces VRAM and graph-capture time)
- config/benchmark_api.py: new script to measure TTFA, RTF and speed
against the live API endpoint
Co-Authored-By: Claude Sonnet 4.6 <noreply@anthropic.com>
Run Qwen3-TTS with CUDA graph acceleration on NVIDIA DGX Spark
(ARM64 / SM 121 / CUDA 13) as an OpenAI-compatible TTS API server.
- Dockerfile targeting nvidia/cuda:13.0.2-base-ubuntu24.04 with ARM64 cu130 PyTorch wheels
- Patch for max-seq-len support to handle long reference audio without crashes
- OpenWebUI + SillyTavern compatible API endpoints (/v1/models, /v1/audio/voices, /speakers)
- Voice management: auto-generate voices.json from speaker reference audio files
- Auto-transcription helper script for generating reference text from audio
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
