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에이전트 8종(대화형/프로그래머 C·S/디자이너/설계자/기획자/QA/PD/PM)에게 [공통 기본 능력 + 롤별 Specialty] 2층으로 지식을 주입하기 위한 재분류. 문서 내용·포맷은 무수정, 폴더 이동만 (6,372개 문서 수 보존 확인). - Topic_Programming → Domain_Programming (내부 구조 보존) - Topic_Graphic → Domain_Design - Topic_Business → Domain_Product - Topic_General → Domain_General - _Common 신설: Math(구 Topic_Math_Specialty), Reasoning(구 General/From_Thinking & Reasoning), Reasoning_Creativity(구 General/From_창의성), Communication(Poetic_Blog_Writing + From_writing) - 타 도메인의 From_* 폴더는 유지 (출처 표기일 뿐, 이미 도메인에 맞게 분류된 문서) - 빈 폴더 정리 (memory/procedures) - 에이전트→폴더 매핑은 workspace의 .astra/agent-knowledge-map.json (9개 에이전트) Co-Authored-By: Claude Fable 5 <noreply@anthropic.com>
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id, title, category, status, canonical_id, aliases, duplicate_of, source_trust_level, confidence_score, verification_status, tags, raw_sources, last_reinforced, github_commit, tech_stack
| id | title | category | status | canonical_id | aliases | duplicate_of | source_trust_level | confidence_score | verification_status | tags | raw_sources | last_reinforced | github_commit | tech_stack | |||||||||||||
|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|---|
| wiki-2026-0508-speech-recognition-foundations | Speech Recognition Foundations | 10_Wiki/Topics | verified | self |
|
none | A | 0.92 | applied |
|
2026-05-10 | pending |
|
Speech Recognition Foundations
매 한 줄
"매 acoustic signal → text token 의 sequence-to-sequence map". ASR 매 frontend (mel-spectrogram) + acoustic model (CTC/attention/RNNT) + language model (n-gram 또는 LM-fused) 의 stack. 매 2026 매 Whisper-large-v3 / Voxtral / Distil-Whisper / NVIDIA Canary-1B + Parakeet-TDT 의 production 의 표준.
매 핵심
매 frontend (signal processing)
- Resample: 16kHz mono 매 standard.
- STFT → Mel: 25ms window, 10ms hop, 80 mel bins.
- Log-mel + cepstral mean: 매 Whisper input.
- VAD (Silero, WebRTC): 매 silence 의 trim.
매 acoustic-model paradigm
- Hybrid HMM-DNN (Kaldi era): 매 deprecated 매 production 새 system.
- CTC (Connectionist Temporal Classification): monotonic alignment, blank-token. 매 streaming-friendly.
- Attention encoder-decoder (LAS, Whisper): non-monotonic, full-context. 매 high-accuracy offline.
- RNN-T / Transducer: streaming + accurate. 매 mobile/voice-assistant 의 dominant (Parakeet-TDT, Apple Siri).
매 modern open ASR (2026)
- Whisper-large-v3 (OpenAI 2023): 99 lang, multilingual + translation.
- Distil-Whisper-v3.5: 6× faster, 49% smaller, 1% WER drop.
- Voxtral (Mistral 2025): multilingual ASR + LLM-grade understanding.
- NVIDIA Canary-1B-Flash: 4-lang, top Open ASR Leaderboard (2025).
- Parakeet-TDT-1.1B (NVIDIA): streaming RNN-T + token-and-duration.
- SeamlessM4T v2 (Meta): speech + text bidirectional translation.
매 응용
- Meeting transcription (Otter, Granola, Fireflies).
- Captioning (live + offline).
- Voice agents (Pipecat, LiveKit).
- Call-center analytics.
- Accessibility (live caption OS-level).
💻 패턴
Whisper inference (HuggingFace)
from transformers import pipeline
import torch
asr = pipeline(
"automatic-speech-recognition",
model="openai/whisper-large-v3",
torch_dtype=torch.float16,
device="cuda",
return_timestamps=True,
)
out = asr("meeting.wav", chunk_length_s=30, batch_size=8)
print(out["text"])
for chunk in out["chunks"]:
print(chunk["timestamp"], chunk["text"])
faster-whisper (CTranslate2)
from faster_whisper import WhisperModel
model = WhisperModel("large-v3", device="cuda", compute_type="float16")
segments, info = model.transcribe(
"meeting.wav",
beam_size=5,
vad_filter=True,
vad_parameters=dict(min_silence_duration_ms=500),
language="ko",
)
for s in segments:
print(f"[{s.start:.2f}-{s.end:.2f}] {s.text}")
Streaming with Parakeet-TDT (NeMo)
import nemo.collections.asr as nemo_asr
m = nemo_asr.models.ASRModel.from_pretrained("nvidia/parakeet-tdt-1.1b")
m.change_decoding_strategy(None)
# Streaming buffered inference
from nemo.collections.asr.parts.utils.streaming_utils import CacheAwareStreamingAudioBuffer
buf = CacheAwareStreamingAudioBuffer(model=m, online_normalization=True)
buf.append_audio_file("call.wav", stream_id=-1)
for chunk in buf.iter_chunks():
text = m.transcribe([chunk])
yield text
MLX Whisper (Apple Silicon)
# pip install mlx-whisper
import mlx_whisper
result = mlx_whisper.transcribe(
"audio.mp3",
path_or_hf_repo="mlx-community/whisper-large-v3-turbo",
word_timestamps=True,
)
print(result["text"])
Custom mel feature extraction
import torch, torchaudio
def log_mel(wav: torch.Tensor, sr: int = 16000) -> torch.Tensor:
if sr != 16000:
wav = torchaudio.functional.resample(wav, sr, 16000)
spec = torchaudio.transforms.MelSpectrogram(
sample_rate=16000, n_fft=400, hop_length=160, n_mels=80,
)(wav)
return torch.log10(torch.clamp(spec, min=1e-10))
CTC decode with KenLM rescoring
from pyctcdecode import build_ctcdecoder
decoder = build_ctcdecoder(
labels=vocab,
kenlm_model_path="lm.arpa",
alpha=0.5, beta=1.5,
)
# logits: (T, V) numpy
text = decoder.decode(logits, beam_width=100)
Diarization + ASR (pyannote + Whisper)
from pyannote.audio import Pipeline
pipe = Pipeline.from_pretrained("pyannote/speaker-diarization-3.1")
diar = pipe("call.wav", num_speakers=2)
for turn, _, speaker in diar.itertracks(yield_label=True):
seg = wav[int(turn.start*sr):int(turn.end*sr)]
text = asr({"raw": seg, "sampling_rate": sr})["text"]
print(f"{speaker}: {text}")
WER evaluation
from jiwer import wer, cer, compute_measures
ref = "the quick brown fox"
hyp = "the quik brown fox"
print(wer(ref, hyp)) # 0.25
print(cer(ref, hyp))
print(compute_measures(ref, hyp)) # detailed I/D/S
매 결정 기준
| 상황 | Model |
|---|---|
| 99-lang offline | Whisper-large-v3 |
| Edge / 6× faster | Distil-Whisper-v3.5 |
| Apple Silicon laptop | MLX Whisper Turbo |
| Streaming voice-agent | Parakeet-TDT / Canary-1B-Flash |
| Multilingual + understanding | Voxtral |
| Speech-to-speech translation | SeamlessM4T v2 |
기본값: faster-whisper (large-v3) 매 batch; Parakeet-TDT 매 streaming.
🔗 Graph
- 부모: Sequence-to-Sequence-Models
- 응용: Whisper · Voice-Agent
- Adjacent: VAD · TTS
🤖 LLM 활용
언제: 매 transcript post-process (punctuation, summarization), 매 vocab/prompt biasing 매 domain term, 매 hallucination filter. 언제 X: 매 raw acoustic decode (use ASR model), 매 numerical WER eval (use jiwer).
❌ 안티패턴
- No VAD: 매 silence 매 hallucination ("Thanks for watching!"). 매 Whisper 의 known issue.
- Wrong sample rate: 매 8kHz / 44kHz 의 mismatch → 매 garbage output.
- chunk_length_s 너무 길게: 매 Whisper 30s 의 designed 의. 매 더 길면 truncation.
- Greedy decode in noisy: 매 beam search + LM rescore 의 use.
- No diarization in multi-speaker: 매 speaker turn 매 lost.
🧪 검증 / 중복
- Verified (Whisper paper arXiv:2212.04356; HuggingFace Open ASR Leaderboard 2025; NVIDIA NeMo docs).
- 신뢰도 A.
🕓 Changelog
| 날짜 | 변경 |
|---|---|
| 2026-05-08 | Phase 1 |
| 2026-05-10 | Manual cleanup — full content (Whisper/Parakeet/Voxtral + faster-whisper/MLX patterns) |