The README still listed the settings UI as "todo" and its quick start told
users to download models into the working directory; phase 3 still described
the caption window's close button as "quit" and used the wrong binary name.
Bring the docs up to date and fill the gaps: a GUI guide (first run, settings
tabs, tray, download, verify), a full CLI option reference marking the
test-only flags, a troubleshooting page, the model registry's size/sha256
fields plus a licence/size table, the RTF measurement caveat (llama-cli 0.19
vs the in-process 0.56), and a note that the streaming-ONNX backend has its
own code path. phase0/phase1 experiment docs are marked as historical.
文档与现状对齐：README 的 Phase 4 从「待做」改为完成、快速开始不再让人把模型下到当前
目录；phase3 的 ✕ 说明改为「隐藏到托盘」、用法改用 live-caption；新增 GUI/CLI/排错三份
文档；asr-backends 补上注册表的 size/sha256 字段与模型许可/体积表；phase0 补上 RTF 口径
说明（llama-cli 的 0.19 与程序内的 0.56）；phase2 补上「流式 ONNX 走另一条分支」的说明。
The append mode used to feed chunks as successive conversation turns with
the context carried over. The model then copied the previous turn's text
verbatim instead of transcribing the new audio, so a continuous video
kept showing the same sentence forever. Transcribe each chunk
independently instead, matching sglang's segment streaming mode for
Qwen3-ASR; the fixed mode also stays flat at RTF 0.24 versus 0.57-1.08
for the incremental mode.
修复追加式的一个真 bug：原先把各段接成「携带上文的对话轮」，模型会抄上
一轮的文本 —— 连续视频里字幕永远重复同一句。改为每段独立转写（对应
sglang 给 Qwen3-ASR 的 segment 流式模式），内容随之正确变化；成本仍保持
平坦，RTF 0.24，而增量模式是 0.57~1.08。
Add an append-only streaming path that feeds each audio chunk as a new
conversation turn, so nothing is ever recomputed: the KV grows by
append, no prefix is re-decoded and no audio is re-encoded. Per-step
cost becomes flat instead of growing with the segment (RTF 1.06 -> 0.26
on 62.6s of continuous speech), but accuracy requires ~8s chunks because
the audio encoder needs its 8s attention window, so text only appears
every 8s. The incremental mode stays the default.
新增追加式流式：每块音频作为一个新的对话轮追加，前缀不重算、旧音频不重
编码，单步成本不再随段落增长（连续 62.6s 语音 RTF 从 1.06 降到 0.26）。
但准确率要求 chunk≈8s，否则会被截断，所以出字粒度是 8s；增量式仍为默认。
Add native streaming ASR: 8s-aligned audio slices are cached in the KV
cache, only new audio is encoded, and the previous text is fed back as
an assistant prefix with token rollback. Text matches offline results.
Streaming keeps each step within the 2s budget for segments up to 20s.
新增原生流式解码：8s 对齐的音频切片存入 KV 复用，只编码新音频，并把
上一轮文本作为 assistant 前缀回喂（带 token 回滚）。转写结果与离线
一致；20s 以内的段落每步计算都在 2s 预算内。
Add a C++/Qt6/CMake CLI prototype that captures the default sink
monitor through libpulse and transcribes it in-process with Qwen3-ASR
via libllama/libmtmd. Add phase 0/1 verification docs.
新增实时字幕命令行原型：libpulse 采集默认输出 monitor，进程内用
libllama/libmtmd 跑 Qwen3-ASR 推理。附阶段 0/1 验证文档。
blumia