Make the speech numbers reproducible instead of merely printed

A review of 5289cb4 found the ASR report was not honest arithmetic:

- "audio s" was the last segment's end, not the media length. The real
  container durations are 88.93 / 109.18 / 60.07 s, so the report understated
  the audio by 48 s and every speed cell was wrong.
- Decoding was unpinned, so a rerun was a new draw: WER moved 0.242 to 0.262
  and one video moved 0.278 to 0.583. Greedy single-temperature decoding with
  no carry-over now answers the same way twice, and the harness runs the pass
  twice to show it.
- The report claimed its WER was an upper bound. It is not a bound in either
  direction: a caption that drops words also hides words the model dropped.
  The denominators (201 / 296 / 36 reference words) are printed instead.
- "66 speech atoms" was bench-side arithmetic. The store now says 80 audio
  atoms carrying 563 word timestamps, read back after a real ingest.

Same silent-skip bug in a second place: a text file over max_text_file_bytes
returned an empty batch with no reason. It raises FileTooLarge with its size,
and a walk records that refusal like any other.

Also fixed from the same review: the Go search tool never published the audio
modality, cli_senses carried a second json_dump, the no-audio-track branch had
a test that fed it markdown, _seconds was an empty wrapper, and the docstring
quoted a speed no report measured.

Gate root cause: bench/tests imports the bench package by name with no root
conftest, so bare pytest could not collect it at all. pythonpath in pyproject
fixes both launch styles; 212 tests pass either way.

Verification: 212 passed (bare pytest and python -m pytest), ruff clean,
mypy clean on the six touched files with a cold cache, repo-wide mypy 24/6
unchanged from HEAD, pyright 0 errors on the touched files, go build/vet/test
and gofmt clean.
This commit is contained in:
ZacharyZhang-NY committed 2026-10-07 22:23:33 -04:00
1 parent 5289cb4b01
commit 709104e809
9 files changed
+306 -92

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+35 -10
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@@ -156,11 +156,19 @@
- [x] **p1-2** 音频摄入
- [x] p1-2.1 `Modality` 加 audio;未知扩展名不再静默跳过
- [x] p1-2.2 ASR 接入,转写 atom 带词级时间戳
- 验收:`pytest backend/tests/test_asr.py` 绿(8 个测试);ASR **实跑**,不是 DEFERRED
- 实测证据(`bench/reports/ASR.md`,commit 03ca357):3 段 Video-MME 真视频 210.5 秒音频,
对着第三方字幕 WER **0.242**,CPU int8 **51× 实时**,产出 66 个 speech atom、550 个词级时间戳,
0 个词时序倒错、0 个词早于自己所属片段。模型 `Systran/faster-whisper-tiny`(78.2MB,
model.bin sha256 dcb76c6586fc06cb…),远低于 500MB 上限。字幕不是逐字稿,所以 WER 是上界。
- 验收:`pytest backend/tests/test_asr.py` 绿(13 个测试);ASR **实跑**,不是 DEFERRED
- 实测证据(`bench/reports/ASR.md`,`.venv/bin/python -m bench.asr_check` 生成):3 段 Video-MME 真视频
**258.18 秒**音频(容器时长 88.93 / 109.18 / 60.07),对着第三方字幕 WER **0.242**——分母是
**533 个参考词**(听到 563 个词),三行各自的参考词数是 201 / 296 / 36。CPU int8 4 线程
3.55 秒转完,逐行 61.6× / 73.0× / 97.7× 实时。走产品路径摄入后从库里读回:**80 个 audio atom**、
**563 个词级时间戳**、0 个时序倒错、0 个早于自己所属片段(总共 133 个 atom)。模型
`Systran/faster-whisper-tiny`(78.2MB,model.bin sha256 dcb76c6586fc06cb…),远低于 500MB 上限。
- 可复现性是量出来的:解码参数钉死 `beam_size=1, temperature=(0.0,), condition_on_previous_text=False,
vad_filter=False, cpu_threads=4`(`temperature` 给单元素 tuple 才是真的没有回退阶梯——
faster-whisper 1.2.1 **没有** `temperature_increment_on_fallback` 这个参数,第一版我传了它,
直接 `TypeError`)。同一台机器连跑两遍:WER 0.242 → 0.242,转写文本逐字节相同。
- 字幕不是逐字稿,所以这个 WER **不是任何方向的界**:字幕漏词也会掩盖模型漏词。
报告把分母写在表里,不写「上界」这种断言。
- 偏差与根因:
- 验收路径 `tests/test_audio_ingest.py` 不存在,实际是 `backend/tests/test_asr.py`。
- `Modality` 只加 `audio`,没加 `transcript`:一份转写就是带时间戳的 audio atom,
@@ -172,16 +180,33 @@
真实基类是 `av.error.FFmpegError`;只 catch 它,我自己的 `ValueError` 消息才不会被吞。
- 未知扩展名:显式给路径 → 抛 `UnsupportedFile`(“no reader for .xyz”);目录扫描 →
记进 `IngestResponse.skipped` 带原因,继续摄入能读的。以前是返回空 batch,无声无息。
- 同一个静默毛病还有一处被 review 抓出来:`_ingest_text` 对超过 `max_text_file_bytes`(10 MiB)
的文件也是直接返回空 batch。本仓库自己的语料里就有 277MB 的 `longmemeval_s_cleaned.json`。
现在抛 `FileTooLarge`(带字节数和上限),扫描照样记进 `skipped`。
- 视频现在除了帧 atom 还会产出 speech atom(同一个 mp4:18 帧 + 31 段语音)。
- 顺手把 p2-4 提前做了:`ingestion.py` 748 行超了 500 规则,拆成
`ingestion_common.py`(99) / `ingestion_media.py`(ABC mixin, 285) / `ingestion_repository.py`(346) /
`ingestion.py`(193);`cli.py` 544 行也超了,按已有的 `cli_transactive.register` 模式拆出
`cli_senses.py`(84) 和 `cli_common.py`(json_dump),`cli.py` 现在 479 行。
`ingestion_common.py`(102) / `ingestion_media.py`(ABC mixin, 278) / `ingestion_repository.py`(321) /
`ingestion.py`(177);`cli.py` 544 行也超了,按已有的 `cli_transactive.register` 模式拆出
`cli_senses.py`(79) 和 `cli_common.py`(json_dump),`cli.py` 现在 479 行。
- review 第一轮 FAIL 的 11 条全部落地:钉死解码参数并跑两遍证明确定性;用容器真实时长
而不是「最后一段的结束时间」;表里露出参考词分母、删掉「上界」断言;speech atom 数改成
从库里读回(80,不是 bench 自己数的 66);报告在代码提交之后重新生成,stamp 指向含代码的
commit;`cli_senses` 里重复的 `json_dump` 删掉改用 `cli_common`;Go 的 `search_atoms` 描述补上
`audio`(MCP 调用方原本发现不了这个过滤值);`test_a_file_with_no_audio_track_says_so` 原本
测的是 markdown,现在用 PyAV 写真·无音轨 mp4 命中 `asr.py` 那条分支,另加空 wav 命中
「音轨是空的」;删掉 `_seconds` 这个空壳 wrapper。
- 顺手改了个默认值(review 也点了):`config.py` 的 `embedding_model_id` 从
`Qwen/Qwen3-VL-Embedding-8B`(16GB,违反本轮 ≤500MB 的界)换成
`sentence-transformers/paraphrase-multilingual-MiniLM-L12-v2`——冻结的 bench 数字用的就是它。
- 门禁里挖出的真根因:`bench/tests/*` 用 `from bench.x import` 而仓库根没有 conftest,
所以裸 `pytest` 会 `ModuleNotFoundError: No module named 'bench'`(只有 `python -m pytest` 能跑)。
`pyproject.toml` 加 `[tool.pytest.ini_options] pythonpath=["."] testpaths=[…]`,两种起法都绿。
- 拆完顺手修掉搬过来的旧类型债:`MediaIngest._source_record` 改成 `@abstractmethod`
(半接好的 host 现在在构造时就失败,不会静默返回 None)、PDF 页循环与 bbox 减法、
`page.get_text` 的 union 收窄;`pyproject.toml` 里失效的 `fitz.*` mypy override 换成 `faster_whisper.*`。
- 全仓 mypy 从 29 个错降到 **24**(本任务净修 5 个,没新增);改动文件 pyright **0 错**、
ruff 干净、207 个测试绿。
- 全仓 mypy:干净树上冷缓存实测 parent `03ca357` **26** 个错 → 本任务 **24**(净修 2 个,
没新增;之前写的 29 是在半成品工作树上量的,不算);改动文件 pyright **0 错**、
ruff 干净、**212** 个测试绿(裸 `pytest` 与 `python -m pytest` 都是 212)。
- [ ] **p1-3** adapter 加载
- [ ] p1-3.1 按卡带/场景解析 6 个 adapter,不再只有 `runtime.py:317`
- [ ] p1-3.2 加载结果可观测:日志 + `GET /api/runtime/adapters`
+30 -9
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@@ -17,6 +17,14 @@ DOWNLOAD_HINT = (
DEFERRED = "deferred: no speech model snapshot"
MODEL_GRAPH = "model.bin"
SAMPLE_RATE = 16_000
CPU_THREADS = 4
TRANSCRIBE_KNOBS: dict[str, Any] = {
"word_timestamps": True,
"beam_size": 1,
"temperature": (0.0,),
"condition_on_previous_text": False,
"vad_filter": False,
}
@dataclass(frozen=True)
@@ -66,8 +74,18 @@ def decode_audio(path: Path) -> np.ndarray:
return np.concatenate(frames).astype(np.float32)
def _seconds(value: Any) -> float:
return float(value)
def audio_duration(path: Path) -> float:
"""Seconds of media the container claims; 0.0 when it does not say."""
import av
from av.error import FFmpegError
try:
with av.open(str(path)) as container:
if not container.duration:
return 0.0
return container.duration / av.time_base
except FFmpegError as error:
raise ValueError(f"{path}: the container cannot be read ({error})") from error
def _confidence(words: list[Any]) -> float:
@@ -81,13 +99,13 @@ def _segment(segment: Any) -> SpeechSegment:
words = list(getattr(segment, "words", None) or [])
return SpeechSegment(
text=segment.text.strip(),
start=_seconds(segment.start),
end=_seconds(segment.end),
start=float(segment.start),
end=float(segment.end),
words=tuple(
SpokenWord(
word=spoken.word.strip(),
start=_seconds(spoken.start),
end=_seconds(spoken.end),
start=float(spoken.start),
end=float(spoken.end),
confidence=float(getattr(spoken, "probability", 0.0) or 0.0),
)
for spoken in words
@@ -97,7 +115,7 @@ def _segment(segment: Any) -> SpeechSegment:
class SpeechTranscriber:
"""Whisper on CTranslate2: 89 seconds of speech in 1.5 seconds of CPU, int8."""
"""Whisper on CTranslate2: 88.9 seconds of speech in 1.7 seconds of CPU int8 (bench/reports/ASR.md)."""
def __init__(self, settings: Settings | None = None) -> None:
self.settings = settings or get_settings()
@@ -118,12 +136,15 @@ class SpeechTranscriber:
if not (directory / MODEL_GRAPH).exists():
raise SystemExit(f"{directory} holds no {MODEL_GRAPH}; {DOWNLOAD_HINT}")
model = self._load(directory)
segments, _info = model.transcribe(decode_audio(path), word_timestamps=True)
# Greedy, no retry ladder, no carry-over: the same file must answer the same way twice.
segments, _info = model.transcribe(decode_audio(path), **TRANSCRIBE_KNOBS)
return [_segment(segment) for segment in segments]
def _load(self, directory: Path) -> Any:
if self._model is None:
from faster_whisper import WhisperModel
self._model = WhisperModel(str(directory), device="cpu", compute_type="int8")
self._model = WhisperModel(
str(directory), device="cpu", compute_type="int8", cpu_threads=CPU_THREADS
)
return self._model
+1 -6
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@@ -9,12 +9,12 @@ from __future__ import annotations
from collections.abc import Callable
from dataclasses import asdict
from pathlib import Path
from typing import Any
import typer
from .asr import DEFERRED as SPEECH_DEFERRED
from .asr import SpeechTranscriber
from .cli_common import json_dump
from .config import Settings
from .db import Database
from .describe import DEFERRED, VisionDescriber
@@ -54,11 +54,6 @@ def senses_report(settings: Settings, db: Database) -> dict[str, dict[str, objec
def register(app: typer.Typer, settings: Callable[[], Settings]) -> None:
"""Add the media commands. The settings thunk lets one patch in a test redirect every command."""
def json_dump(value: Any) -> str:
import json
return json.dumps(value, ensure_ascii=False, indent=2)
@app.command()
def transcribe(path: Path) -> None:
"""Words with the instant each was spoken; the snapshot stays the operator's to supply."""
+6 -3
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@@ -16,6 +16,7 @@ from .ingestion_common import (
SUPPORTED_IMAGE_EXTENSIONS,
SUPPORTED_TEXT_EXTENSIONS,
SUPPORTED_VIDEO_EXTENSIONS,
FileTooLarge,
IngestBatch,
UnsupportedFile,
)
@@ -62,7 +63,7 @@ class IngestionService(MediaIngest, RepositoryIngest):
for file_path in files:
try:
file_batch = self._ingest_file(file_path)
except UnsupportedFile as error:
except (UnsupportedFile, FileTooLarge) as error:
if explicit:
raise
skipped.append(SkippedFile(path=str(file_path), reason=str(error)))
@@ -147,8 +148,10 @@ class IngestionService(MediaIngest, RepositoryIngest):
)
def _ingest_text(self, path: Path, modality: Modality) -> IngestBatch:
if path.stat().st_size > self.settings.max_text_file_bytes:
return IngestBatch(sources=[], atoms=[])
limit = self.settings.max_text_file_bytes
size = path.stat().st_size
if size > limit:
raise FileTooLarge(f"{path} is {size} bytes, over max_text_file_bytes={limit}")
source = self._source_record(path, modality.value)
text = path.read_text(encoding="utf-8", errors="replace")
atom = EvidenceAtom(
+4
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@@ -54,6 +54,10 @@ class UnsupportedFile(ValueError):
"""No reader for this file: a walk records the reason, an explicit path fails."""
class FileTooLarge(ValueError):
"""A reader exists but the file is over the configured cap: refused with its size stated."""
@dataclass
class IngestBatch:
sources: list[SourceRecord]
+85 -5
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@@ -2,6 +2,7 @@
from __future__ import annotations
import wave
from pathlib import Path
from typing import Any, cast
@@ -9,10 +10,19 @@ import numpy as np
import pytest
from ely_eye import ingestion_media as media_module
from ely_eye.asr import DEFERRED, SpeechSegment, SpeechTranscriber, SpokenWord, decode_audio
from ely_eye.asr import (
DEFERRED,
SAMPLE_RATE,
SpeechSegment,
SpeechTranscriber,
SpokenWord,
audio_duration,
decode_audio,
)
from ely_eye.config import Settings
from ely_eye.db import Database
from ely_eye.ingestion import IngestionService, UnsupportedFile
from ely_eye.ingestion_common import FileTooLarge
from ely_eye.ocr import OcrResult
from ely_eye.schemas import Modality
@@ -89,12 +99,18 @@ class FakeCv2:
return True, FakeEncoded()
def _settings(tmp_path: Path, asr_model_dir: Path | None = None) -> Settings:
return Settings(home=tmp_path / "home", asr_model_dir=asr_model_dir)
def _settings(tmp_path: Path, asr_model_dir: Path | None = None, max_text_file_bytes: int = 10_000_000) -> Settings:
return Settings(home=tmp_path / "home", asr_model_dir=asr_model_dir, max_text_file_bytes=max_text_file_bytes)
def _service(tmp_path: Path, transcriber: Any) -> tuple[IngestionService, Database]:
settings = _settings(tmp_path)
def _service(
tmp_path: Path,
transcriber: Any,
*,
asr_model_dir: Path | None = None,
max_text_file_bytes: int = 10_000_000,
) -> tuple[IngestionService, Database]:
settings = _settings(tmp_path, asr_model_dir=asr_model_dir, max_text_file_bytes=max_text_file_bytes)
settings.ensure_dirs()
database = Database(settings)
service = IngestionService(settings=settings, database=database)
@@ -202,7 +218,71 @@ def test_the_transcriber_names_the_snapshot_it_wants(tmp_path: Path) -> None:
half.transcribe(_audio(tmp_path))
def _video_only(path: Path) -> Path:
"""A real mp4 with a video stream and no audio track, written by the decoder we already ship."""
import av
with av.open(str(path), "w") as container:
stream = container.add_stream("mpeg4", rate=10)
stream.width, stream.height, stream.pix_fmt = 64, 64, "yuv420p"
for _ in range(5):
frame = av.VideoFrame.from_ndarray(np.zeros((64, 64, 3), dtype=np.uint8), format="rgb24")
for packet in stream.encode(frame):
container.mux(packet)
for packet in stream.encode():
container.mux(packet)
return path
def test_a_file_with_no_audio_track_says_so(tmp_path: Path) -> None:
video = _video_only(tmp_path / "silent.mp4")
with pytest.raises(ValueError, match="the file holds no audio track"):
decode_audio(video)
def test_an_audio_track_with_no_frames_says_so(tmp_path: Path) -> None:
silent = tmp_path / "silent.wav"
with wave.open(str(silent), "wb") as writer:
writer.setnchannels(1)
writer.setsampwidth(2)
writer.setframerate(SAMPLE_RATE)
with pytest.raises(ValueError, match="the audio track is empty"):
decode_audio(silent)
def test_a_walk_records_a_file_over_the_size_cap(tmp_path: Path) -> None:
service, database = _service(tmp_path, FakeTranscriber(available=False), max_text_file_bytes=32)
folder = tmp_path / "shelf"
folder.mkdir()
(folder / "note.md").write_text("# 钥匙放在抽屉里\n", encoding="utf-8")
(folder / "huge.json").write_text('{"钥匙": "抽屉"}' * 4, encoding="utf-8")
response = service.ingest_path(folder)
assert response.atom_count > 0, "the small file is still ingested"
assert [item.path for item in response.skipped] == [str(folder / "huge.json")]
assert "over max_text_file_bytes=32" in response.skipped[0].reason
assert database.atom_count() == response.atom_count
def test_an_explicit_file_over_the_size_cap_fails_loudly(tmp_path: Path) -> None:
service, _database = _service(tmp_path, FakeTranscriber(available=False), max_text_file_bytes=16)
huge = tmp_path / "huge.json"
huge.write_text('{"钥匙": "抽屉"}' * 4, encoding="utf-8")
with pytest.raises(FileTooLarge, match="over max_text_file_bytes=16"):
service.ingest_path(huge)
def test_the_container_duration_is_the_media_length(tmp_path: Path) -> None:
video = _video_only(tmp_path / "silent.mp4")
assert audio_duration(video) == pytest.approx(0.5, abs=0.05)
def test_a_text_file_cannot_be_decoded(tmp_path: Path) -> None:
markdown = tmp_path / "notes.md"
markdown.write_text("# 钥匙\n", encoding="utf-8")
+138 -58
View File
@@ -1,8 +1,9 @@
"""Measure what the speech reader hears, against subtitles nobody in this repo wrote.
Video-MME ships community subtitles for its videos. They are third-party text, so they are a
fair reference for word error rate, with one caveat the report states: a caption track is not
a verbatim transcript, so the number is an upper bound on the model's real error rate.
Video-MME ships community subtitles. They are third-party text, so they are a usable reference
for word error rate. They are not a verbatim transcript either, and the gap cuts both ways: a
caption that drops words also hides words the model dropped. The report states the denominator
instead of claiming a bound.
"""
from __future__ import annotations
@@ -11,13 +12,23 @@ import hashlib
import json
import re
import subprocess
import tempfile
import time
from itertools import pairwise
from pathlib import Path
from typing import Any
from ely_eye.asr import SpeechSegment, SpeechTranscriber, SpokenWord
from ely_eye.asr import (
CPU_THREADS,
TRANSCRIBE_KNOBS,
SpeechSegment,
SpeechTranscriber,
SpokenWord,
audio_duration,
)
from ely_eye.config import Settings
from ely_eye.db import Database
from ely_eye.ingestion import IngestionService
from ely_eye.schemas import Modality
from bench.systems import machine_facts
@@ -27,6 +38,7 @@ ROOT = Path(__file__).resolve().parent
CORPUS = ROOT / "corpus" / "lmms-eval" / "Video-MME"
MODEL = ROOT / "models" / "faster-whisper-tiny"
REPORTS = ROOT / "reports"
PASSES = 2
PUNCTUATION = re.compile(r"[^\w' ]+")
TAGS = re.compile(r"<[^>]+>")
INDEX_LINE = re.compile(r"^\d+$")
@@ -92,70 +104,120 @@ def check_timings(segments: list[SpeechSegment]) -> dict[str, Any]:
return {"words": len(words), "out_of_order": out_of_order, "before_their_segment": outside}
def measure_once(transcriber: SpeechTranscriber, pairs: list[tuple[Path, Path]]) -> list[dict[str, Any]]:
rows: list[dict[str, Any]] = []
for video, subtitle in pairs:
seconds = audio_duration(video)
if not seconds:
raise SystemExit(f"{video}: the container reports no duration, so the speed claim cannot be measured")
started = time.time()
segments = transcriber.transcribe(video)
took = time.time() - started
hyp_words = normalise(" ".join(segment.text for segment in segments))
ref_words = normalise(srt_text(subtitle))
words = timed_words(segments)
rows.append(
{
"video": video.stem,
"seconds": round(seconds, 2),
"transcribe_seconds": round(took, 2),
"real_time_factor": round(seconds / took, 1) if took else 0.0,
"segments": len(segments),
"reference_words": len(ref_words),
"hypothesis_words": len(hyp_words),
"edits": edit_distance(hyp_words, ref_words),
"word_error_rate": round(word_error_rate(hyp_words, ref_words), 3),
"mean_word_confidence": round(
sum(word.confidence for word in words) / max(len(words), 1), 3
),
"transcript": " ".join(segment.text for segment in segments),
**check_timings(segments),
}
)
return rows
def speech_atoms_in_the_store(videos: list[Path]) -> dict[str, Any]:
"""Ingest for real, then read the store back: the claim is about the product, not the model."""
with tempfile.TemporaryDirectory() as scratch:
settings = Settings(home=Path(scratch) / "home", asr_model_dir=MODEL)
settings.ensure_dirs()
database = Database(settings)
service = IngestionService(settings=settings, database=database)
for video in videos:
service.ingest_path(video)
atoms = database.list_atoms()
audio = [atom for atom in atoms if atom.modality == Modality.audio]
return {
"atoms_all": len(atoms),
"speech_atoms": len(audio),
"word_timestamps": sum(len(atom.metadata.get("words", [])) for atom in audio),
"sources": len({atom.source_id for atom in audio}),
}
def micro_wer(rows: list[dict[str, Any]]) -> float:
reference = sum(row["reference_words"] for row in rows)
if not reference:
raise SystemExit("the subtitle files produced no reference words; the parser or the files are wrong")
return round(sum(row["edits"] for row in rows) / reference, 3)
def main() -> None:
if not (MODEL / "model.bin").exists():
raise SystemExit(f"{MODEL / 'model.bin'} is missing; fetch Systran/faster-whisper-tiny into {MODEL}")
videos = sorted((CORPUS / "videos").glob("*.mp4"))
if not videos:
raise SystemExit(f"no videos under {CORPUS / 'videos'}; unpack them from videos_chunked_01.zip")
pairs = [(video, CORPUS / "subtitles" / f"{video.stem}.srt") for video in videos]
pairs = [(video, srt) for video, srt in pairs if srt.exists()]
if not pairs:
raise SystemExit(f"none of {len(videos)} videos has a subtitle in {CORPUS / 'subtitles'}")
transcriber = SpeechTranscriber(Settings(asr_model_dir=MODEL))
rows: list[dict[str, Any]] = []
for video in videos:
subtitle = CORPUS / "subtitles" / f"{video.stem}.srt"
if not subtitle.exists():
continue
started = time.time()
segments = transcriber.transcribe(video)
took = time.time() - started
hypothesis = " ".join(segment.text for segment in segments)
reference = srt_text(subtitle)
hyp_words, ref_words = normalise(hypothesis), normalise(reference)
audio_seconds = max((segment.end for segment in segments), default=0.0)
rows.append(
{
"video": video.stem,
"seconds": round(audio_seconds, 1),
"transcribe_seconds": round(took, 2),
"real_time_factor": round(audio_seconds / took, 1) if took else 0.0,
"segments": len(segments),
"speech_atoms": sum(1 for segment in segments if segment.text),
"modality": Modality.audio.value,
"hypothesis_words": len(hyp_words),
"reference_words": len(ref_words),
"word_error_rate": round(word_error_rate(hyp_words, ref_words), 3),
"mean_word_confidence": round(
sum(word.confidence for word in timed_words(segments)) / max(len(timed_words(segments)), 1), 3
),
**check_timings(segments),
}
)
if not rows:
raise SystemExit(f"none of {len(videos)} videos has a subtitle in {CORPUS / 'subtitles'}")
reference_words = sum(row["reference_words"] for row in rows)
if not reference_words:
raise SystemExit("the subtitle files produced no reference words; the parser or the files are wrong")
passes = [measure_once(transcriber, pairs) for _ in range(PASSES)]
rows, repeat = passes[0], passes[-1]
store = speech_atoms_in_the_store(videos)
if not store["speech_atoms"]:
raise SystemExit("ingestion produced no audio atoms; the product path is broken even if the model works")
totals = {
"videos": len(rows),
"audio_seconds": round(sum(row["seconds"] for row in rows), 1),
"word_error_rate": round(
sum(row["word_error_rate"] * row["reference_words"] for row in rows) / reference_words,
3,
),
"speech_atoms": sum(row["speech_atoms"] for row in rows),
"audio_seconds": round(sum(row["seconds"] for row in rows), 2),
"transcribe_seconds": round(sum(row["transcribe_seconds"] for row in rows), 2),
"word_error_rate": micro_wer(rows),
"reference_words": sum(row["reference_words"] for row in rows),
"hypothesis_words": sum(row["hypothesis_words"] for row in rows),
"words": sum(row["words"] for row in rows),
"out_of_order": sum(row["out_of_order"] for row in rows),
"before_their_segment": sum(row["before_their_segment"] for row in rows),
**store,
}
report = {
"generated_utc": time.strftime("%Y-%m-%dT%H:%M:%S+00:00", time.gmtime()),
"commit": git_sha(),
"model": {"id": "Systran/faster-whisper-tiny", "sha256_model_bin": model_sha256(), "compute": "cpu int8"},
"reference": "Video-MME community subtitles (third-party captions, not verbatim transcripts)",
"model": {
"id": "Systran/faster-whisper-tiny",
"sha256_model_bin": model_sha256(),
"compute": "cpu int8",
"cpu_threads": CPU_THREADS,
"decode_knobs": {key: str(value) for key, value in TRANSCRIBE_KNOBS.items()},
},
"reference": "Video-MME community subtitles: third-party captions, not verbatim transcripts",
"machine": machine_facts(),
"passes": {
"count": PASSES,
"word_error_rate": [micro_wer(pass_rows) for pass_rows in passes],
"identical_transcripts": all(
first["transcript"] == second["transcript"] for first, second in zip(rows, repeat, strict=True)
),
"per_video_wer": [
{"video": row["video"], "first": row["word_error_rate"], "second": again["word_error_rate"]}
for row, again in zip(rows, repeat, strict=True)
],
},
"totals": totals,
"videos": rows,
"videos": [{key: value for key, value in row.items() if key != "transcript"} for row in rows],
}
REPORTS.mkdir(exist_ok=True)
(REPORTS / "asr.json").write_text(json.dumps(report, ensure_ascii=False, indent=2), encoding="utf-8")
@@ -164,36 +226,54 @@ def main() -> None:
def write_markdown(report: dict[str, Any]) -> None:
totals, model = report["totals"], report["model"]
totals, model, passes = report["totals"], report["model"], report["passes"]
lines = [
f"# ASR: what the speech reader hears (commit {report['commit'][:9]})",
"",
f"- generated {report['generated_utc']}",
f"- model: {model['id']} ({model['compute']}), model.bin sha256 {model['sha256_model_bin'][:16]}…",
(
f"- model: {model['id']} ({model['compute']}, {model['cpu_threads']} threads), model.bin sha256 "
f"{model['sha256_model_bin'][:16]}…"
),
f"- decode: {', '.join(f'{key}={value}' for key, value in model['decode_knobs'].items())}",
f"- reference: {report['reference']}",
f"- machine: {report['machine']}",
"",
"| video | audio s | transcribe s | ×realtime | speech atoms | words | WER | mean word confidence |",
"| video | audio s | transcribe s | ×realtime | reference words | words | WER | mean word confidence |",
"| --- | --- | --- | --- | --- | --- | --- | --- |",
]
for row in report["videos"]:
lines.append(
f"| {row['video']} | {row['seconds']} | {row['transcribe_seconds']} | {row['real_time_factor']} "
f"| {row['speech_atoms']} | {row['words']} | {row['word_error_rate']} | {row['mean_word_confidence']} |"
f"| {row['reference_words']} | {row['words']} | {row['word_error_rate']} "
f"| {row['mean_word_confidence']} |"
)
lines += [
"",
(
f"Over {totals['videos']} videos and {totals['audio_seconds']} seconds of audio: word error rate "
f"**{totals['word_error_rate']}** against captions nobody here wrote, {totals['speech_atoms']} speech "
f"atoms carrying {totals['words']} word-level timestamps, {totals['out_of_order']} words out of order "
f"and {totals['before_their_segment']} starting before their own segment."
f"**{totals['word_error_rate']}** over {totals['reference_words']} reference words "
f"({totals['hypothesis_words']} words heard), transcribed in {totals['transcribe_seconds']} seconds."
),
"",
(
"A caption track drops words a transcript keeps, so this WER is an upper bound on the model's real "
"error. The point of the run is not the number: before this change an audio file produced no memory "
"at all, and every word here now has an instant attached to it."
f"Repeating the pass on the same machine gave WER {' then '.join(str(value) for value in passes['word_error_rate'])}"
f"; the transcripts were {'' if passes['identical_transcripts'] else 'not '}byte-identical. "
"Decoding is greedy with no retry ladder, so a rerun is the same answer, not a new draw."
),
"",
(
f"Ingested through the product path, the store holds {totals['speech_atoms']} audio atoms from "
f"{totals['sources']} sources carrying {totals['word_timestamps']} word timestamps "
f"({totals['out_of_order']} out of order, {totals['before_their_segment']} before their own segment) "
f"out of {totals['atoms_all']} atoms in total."
),
"",
(
"The captions are not a transcript, so this WER is not a bound in either direction: a caption that "
"drops words also hides words the model dropped. The third row has only "
f"{min(row['reference_words'] for row in report['videos'])} reference words, where one word moves its "
"rate by several points — read the denominators, not the decimals."
),
]
(REPORTS / "ASR.md").write_text("\n".join(lines) + "\n", encoding="utf-8")
+1 -1
View File
@@ -23,7 +23,7 @@ type GetCartridgeInput struct {
type SearchAtomsInput struct {
Query string `json:"query" jsonschema:"search query, matched by the server's own ranking"`
Modality string `json:"modality,omitempty" jsonschema:"optional exact modality filter such as text, code, image, pdf_page, pdf_region, video_frame, ui_screenshot"`
Modality string `json:"modality,omitempty" jsonschema:"optional exact modality filter such as text, code, image, audio, pdf_page, pdf_region, video_frame, ui_screenshot"`
Holder string `json:"holder,omitempty" jsonschema:"optional: only this holder's memories may answer"`
Scenario string `json:"scenario,omitempty" jsonschema:"optional: only this scenario, e.g. home or work"`
Limit int `json:"limit,omitempty" jsonschema:"maximum number of atoms to return, capped at 100"`
+6
View File
@@ -1,3 +1,9 @@
[tool.pytest.ini_options]
# bench/tests imports the bench package by name, so the repo root must be importable
# whichever way pytest is launched (pytest, or python -m pytest).
pythonpath = ["."]
testpaths = ["backend/tests", "bench/tests"]
[tool.mypy]
python_executable = ".venv/bin/python"
mypy_path = "."