The fused mode appended the codec stream tail to the predictor frame
graph, so one compute produced both a frame's codes and its 80 ms of
audio with no host round trip. The experiment applied to max_batch 1
with a streaming synthesis only, it cost throughput against the
buffered flush that stays the default, and it kept a second frame
graph, its ring inputs and an init flag alive for that single case. It
is not worth keeping.
Remove the tail helpers, the fused graph of CodePredGraphSet, the
codec_fused field of qt_init_params, the --codec-fused flag of both
tools and the harness switch that exercised it. The predictor frame
unroll and the in graph sampler are untouched.
One static frame graph per batch width replaces the per step chain:
prefill and the 15 acoustic steps run in a single backend compute.
This is the target architecture for the llama.cpp Qwen3-TTS port and
serves as its working GGML reference while under test.
Sampling is a plain op chain batched over slots: temperature, argsort
top_k (descending order is guaranteed on every backend, unlike top_k),
softmax, cumsum, cdf crossing against a per step philox uniform.
Greedy draws with u = 0 and lands on the argmax. Faster than the
fused sampling op under CUDA graph capture, greedy codes stay exact
against the Python reference on CPU, CUDA and Vulkan.
Opt in single slot latency mode (--codec-fused on qwen-tts and
tts-server, codec_fused in qt_init_params): the codec stream tail
joins the frame graph at T=1, codes read through a device view, one
80 ms chunk per compute with no host round trip.
Predictor 3.34 -> 3.11 ms/frame on CUDA, end to end -4%.
The left context of the buffered chunked decode is no longer a caller
knob: it derives from the codec's own sliding window (2x144 frames),
placing the default decode at the residual floor of the split.
codec_chunk_sec moves from qt_tts_params to qt_init_params, resolved
once to frames at load. The mid-struct removal bumps the ABI to a
closed range [QT_ABI_MIN_VERSION, QT_ABI_VERSION] = [4, 4]; the probe
asserts both bounds reject through the range check.
The CLI tool already exposes these flags for controlling the vocoder's
chunked-decode window. tts-server always used the hardcoded 24.0s
chunk / 2.0s left-context defaults, with no way to override them at
the server binary's CLI, unlike qwen-tts.
This matters on memory-constrained GPUs: utterances shorter than the
chunk duration decode in a single pass, which can OOM on a small GPU
shared with other processes. Tightening these values (e.g. 4.0/1.5)
forces genuine chunked decode with bounded peak memory per chunk.
The 15 predictor flavors build and allocate once at load, positions,
kv rows, and mask baked as never freed graph outputs, and replay
directly on the backend. The prefill slices the last position before
lm_head so every flavor reads one logits row at offset zero. Replaces
the per step graph rebuild, sched allocation, and debug prints.
The speech body accepts seed, max_new_tokens, temperature, top_k,
top_p, and repetition_penalty. Unset fields keep the engine defaults,
a temperature of zero selects greedy decoding, and the subtalker
mirrors the talker knobs. A fixed seed makes a request reproducible.
POST /v1/voices registers a voice from a WAV extracted server side
through qt_extract_voice_ref or from pre extracted .spk and .rvq
latents taken verbatim, DELETE drops it and GET lists it alongside the
model speakers. A registered voice wins over a speaker of the same
name and injects the reference latents into qt_tts_params, ref_text
present selects ICL clone mode. The registry lives in process RAM
under the synthesis mutex, so registration and lookups never race a
running synthesis. The audio and rvq readers gain buffer variants
factored from the file paths. The README and the architecture
document catch up on the streaming decode, the hidden bridge, and the
server endpoints.
Pascal
Gary
karlandPascal