chore: import upstream snapshot with attribution
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wehub-resource-sync
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# FunASR Migration Benchmark Example
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Use this example when you are comparing FunASR with Whisper, OpenAI audio APIs, or a cloud ASR provider. It runs FunASR over your representative audio set and writes machine-readable results plus a Markdown summary.
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The script does not claim accuracy by itself. Run your baseline on the same files, then compare transcripts with human review or your normal WER/CER workflow.
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## Quick start
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```bash
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python examples/migration/benchmark_funasr.py \
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--input /path/to/audio_samples \
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--recursive \
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--model iic/SenseVoiceSmall \
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--device cuda \
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--spk-model cam++ \
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--output-dir outputs/funasr_migration_eval \
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--metadata baseline=whisper-large-v3
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```
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Outputs:
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- `results.jsonl`: one JSON object per audio file with text, elapsed seconds, audio duration, realtime factor, model, device, and errors.
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- `summary.md`: run configuration, aggregate speed, per-file previews, and next comparison steps.
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## CPU smoke test
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For a portable first check, use CPU and a small audio folder:
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```bash
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python examples/migration/benchmark_funasr.py \
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--input ./samples \
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--model iic/SenseVoiceSmall \
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--device cpu \
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--output-dir outputs/funasr_cpu_smoke
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```
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## 中文快速说明
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这个示例用于从 Whisper、OpenAI 音频 API 或云端 ASR 迁移前的本地评测。请用同一批代表性音频分别跑旧方案和 FunASR，再用人工审阅或 WER/CER 流程比较质量。
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```bash
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python examples/migration/benchmark_funasr.py \
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--input /path/to/audio_samples \
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--recursive \
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--model iic/SenseVoiceSmall \
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--device cuda \
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--spk-model cam++ \
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--output-dir outputs/funasr_migration_eval \
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--metadata baseline=whisper-large-v3
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```
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输出文件：
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- `results.jsonl`：每条音频的文本、耗时、音频时长、实时倍速和错误信息。
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- `summary.md`：运行配置、总体速度、逐文件预览和下一步对比建议。
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如果结果可以公开，欢迎提交 [Migration Benchmark Report](https://github.com/modelscope/FunASR/issues/new?template=migration_benchmark.md)，帮助其他用户参考你的硬件、音频领域和质量记录。
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## What to compare
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Track the same fields for your old ASR stack and FunASR:
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| Field | Why it matters |
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|---|---|
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| Audio duration, language, domain, sample rate, speaker count | Keeps the comparison representative. |
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| Model name, version, device, CUDA/PyTorch versions | Makes results reproducible. |
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| Model load time vs inference time | Separates cold start from steady-state throughput. |
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| WER/CER or human review notes | Captures quality beyond speed. |
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| Failed-file rate and error messages | Shows operational risk before rollout. |
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See the [migration guide](../../docs/migration_from_whisper.md) for the full evaluation and rollout checklist. If you can share results publicly, open a [Migration Benchmark Report](https://github.com/modelscope/FunASR/issues/new?template=migration_benchmark.md) so others can learn from your hardware, audio domain, and quality notes.
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#!/usr/bin/env python3
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"""Benchmark FunASR on representative audio during ASR migration.
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The script intentionally measures only the FunASR side of a migration test. Run
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Whisper or a cloud ASR baseline separately, then compare transcripts with your
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normal WER/CER or human-review process.
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"""
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import argparse
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import json
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import sys
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import time
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import wave
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from pathlib import Path
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from typing import Any, Dict, Iterable, List, Optional
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try:
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from funasr import AutoModel
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from funasr.utils.postprocess_utils import rich_transcription_postprocess
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except Exception as exc: # pragma: no cover - import message is for users
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print(
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"Failed to import FunASR. Install it with `pip install -U funasr` "
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"or run this script from the repository root.",
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file=sys.stderr,
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)
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raise
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AUDIO_EXTENSIONS = (
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".wav",
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".mp3",
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".flac",
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".m4a",
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".aac",
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".ogg",
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".opus",
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".wma",
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)
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def parse_args() -> argparse.Namespace:
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parser = argparse.ArgumentParser(
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description="Run FunASR over a representative audio set and write migration benchmark results."
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)
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parser.add_argument("--input", "-i", type=Path, required=True, help="Audio file or folder to benchmark.")
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parser.add_argument(
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"--output-dir",
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"-o",
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type=Path,
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default=Path("outputs/migration_benchmark"),
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help="Directory for results.jsonl and summary.md.",
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)
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parser.add_argument(
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"--model",
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"-m",
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default="iic/SenseVoiceSmall",
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help="FunASR model name or ModelScope/Hugging Face id.",
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)
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parser.add_argument("--device", "-d", default="cpu", help="Inference device: cpu, cuda, or mps.")
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parser.add_argument("--vad-model", default="fsmn-vad", help="VAD model, or 'none' to disable.")
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parser.add_argument("--spk-model", default="", help="Optional speaker model such as cam++.")
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parser.add_argument("--language", default="auto", help="Language hint passed to model.generate.")
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parser.add_argument("--batch-size", type=int, default=1, help="batch_size passed to model.generate.")
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parser.add_argument("--recursive", "-r", action="store_true", help="Recursively scan input folders.")
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parser.add_argument(
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"--extensions",
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nargs="+",
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default=list(AUDIO_EXTENSIONS),
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help="Audio extensions to include when --input is a folder.",
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)
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parser.add_argument(
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"--metadata",
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action="append",
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default=[],
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help="Free-form key=value metadata to copy into the summary, e.g. --metadata baseline=whisper-large-v3.",
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)
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return parser.parse_args()
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def normalize_extensions(values: Iterable[str]) -> List[str]:
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return sorted({value.lower() if value.startswith(".") else f".{value.lower()}" for value in values})
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def iter_audio_files(path: Path, extensions: List[str], recursive: bool) -> List[Path]:
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if path.is_file():
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return [path]
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if not path.exists():
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raise FileNotFoundError(f"Input path does not exist: {path}")
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walker = path.rglob("*") if recursive else path.iterdir()
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return sorted(p for p in walker if p.is_file() and p.suffix.lower() in extensions)
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def audio_duration_seconds(path: Path) -> Optional[float]:
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try:
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import soundfile as sf
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info = sf.info(str(path))
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if info.samplerate:
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return float(info.frames) / float(info.samplerate)
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except Exception:
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pass
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if path.suffix.lower() == ".wav":
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try:
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with wave.open(str(path), "rb") as wav_file:
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frames = wav_file.getnframes()
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rate = wav_file.getframerate()
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if rate:
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return float(frames) / float(rate)
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except Exception:
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return None
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return None
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def metadata_dict(items: Iterable[str]) -> Dict[str, str]:
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parsed: Dict[str, str] = {}
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for item in items:
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if "=" not in item:
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parsed[item] = ""
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continue
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key, value = item.split("=", 1)
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parsed[key.strip()] = value.strip()
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return parsed
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def extract_text(result: Any) -> str:
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if not result:
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return ""
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first = result[0] if isinstance(result, list) else result
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if isinstance(first, dict):
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text = first.get("text", "")
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else:
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text = str(first)
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try:
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return rich_transcription_postprocess(text)
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except Exception:
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return text
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def write_jsonl(path: Path, rows: Iterable[Dict[str, Any]]) -> None:
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with path.open("w", encoding="utf-8") as handle:
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for row in rows:
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handle.write(json.dumps(row, ensure_ascii=False, sort_keys=True) + "\n")
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def markdown_summary(rows: List[Dict[str, Any]], args: argparse.Namespace, model_load_seconds: float) -> str:
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successful = [row for row in rows if not row.get("error")]
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failed = [row for row in rows if row.get("error")]
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known_audio = [row for row in successful if row.get("duration_seconds")]
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total_audio = sum(float(row["duration_seconds"]) for row in known_audio)
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total_elapsed = sum(float(row["elapsed_seconds"]) for row in successful)
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throughput = total_audio / total_elapsed if total_audio and total_elapsed else None
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meta = metadata_dict(args.metadata)
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lines = [
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"# FunASR Migration Benchmark Summary",
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"",
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"## Run configuration",
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"",
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f"- Input: `{args.input}`",
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f"- Model: `{args.model}`",
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f"- Device: `{args.device}`",
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f"- VAD model: `{args.vad_model}`",
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f"- Speaker model: `{args.spk_model or 'none'}`",
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f"- Language: `{args.language}`",
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f"- Batch size: `{args.batch_size}`",
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f"- Model load seconds: `{model_load_seconds:.3f}`",
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]
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for key, value in meta.items():
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lines.append(f"- {key}: `{value}`")
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lines.extend(
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[
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"",
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"## Aggregate results",
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"",
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f"- Files: `{len(rows)}`",
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f"- Successful: `{len(successful)}`",
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f"- Failed: `{len(failed)}`",
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f"- Known audio seconds: `{total_audio:.3f}`" if known_audio else "- Known audio seconds: `unknown`",
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f"- Inference seconds: `{total_elapsed:.3f}`" if successful else "- Inference seconds: `0.000`",
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f"- Aggregate realtime factor: `{throughput:.3f}x`" if throughput else "- Aggregate realtime factor: `unknown`",
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"",
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"## Per-file results",
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"",
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"| File | Audio sec | Inference sec | RTF | Status | Text preview |",
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"|---|---:|---:|---:|---|---|",
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]
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)
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for row in rows:
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duration = row.get("duration_seconds")
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elapsed = row.get("elapsed_seconds")
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rtf = row.get("realtime_factor")
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status = "error" if row.get("error") else "ok"
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preview = (row.get("text") or row.get("error") or "").replace("|", "\\|").replace("\n", " ")[:120]
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lines.append(
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"| {file} | {duration} | {elapsed} | {rtf} | {status} | {preview} |".format(
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file=row["input"],
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duration=f"{duration:.3f}" if isinstance(duration, (int, float)) else "",
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elapsed=f"{elapsed:.3f}" if isinstance(elapsed, (int, float)) else "",
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rtf=f"{rtf:.3f}x" if isinstance(rtf, (int, float)) else "",
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status=status,
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preview=preview,
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)
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)
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lines.extend(
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[
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"",
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"## Next comparison steps",
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"",
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"- Run your Whisper or cloud ASR baseline on the same files.",
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"- Compare transcripts with human review or your normal WER/CER workflow.",
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"- Keep model download and warmup time separate from steady-state throughput.",
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"- Share reproducible findings in a FunASR showcase issue when possible.",
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"",
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]
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)
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return "\n".join(lines)
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def main() -> None:
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args = parse_args()
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extensions = normalize_extensions(args.extensions)
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files = iter_audio_files(args.input, extensions, args.recursive)
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if not files:
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print(f"No audio files found under {args.input} for extensions {extensions}", file=sys.stderr)
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sys.exit(2)
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vad_model = None if args.vad_model.lower() == "none" else args.vad_model
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model_kwargs: Dict[str, Any] = {"model": args.model, "vad_model": vad_model, "device": args.device}
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if args.spk_model:
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model_kwargs["spk_model"] = args.spk_model
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print(f"Loading FunASR model: {args.model} on {args.device}")
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load_start = time.perf_counter()
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model = AutoModel(**model_kwargs)
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model_load_seconds = time.perf_counter() - load_start
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print(f"Model loaded in {model_load_seconds:.3f}s")
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rows: List[Dict[str, Any]] = []
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for index, audio_path in enumerate(files, start=1):
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display = str(audio_path if args.input.is_file() else audio_path.relative_to(args.input))
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print(f"[{index}/{len(files)}] {display}")
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duration = audio_duration_seconds(audio_path)
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start = time.perf_counter()
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row: Dict[str, Any] = {
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"input": display,
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"path": str(audio_path),
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"duration_seconds": duration,
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"model": args.model,
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"device": args.device,
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"language": args.language,
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}
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try:
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result = model.generate(input=str(audio_path), language=args.language, batch_size=args.batch_size)
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elapsed = time.perf_counter() - start
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row["elapsed_seconds"] = elapsed
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row["realtime_factor"] = (duration / elapsed) if duration and elapsed else None
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row["text"] = extract_text(result)
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print(f" ok: {elapsed:.3f}s" + (f", {row['realtime_factor']:.3f}x" if row["realtime_factor"] else ""))
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except Exception as exc: # keep benchmarking other files
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elapsed = time.perf_counter() - start
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row["elapsed_seconds"] = elapsed
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row["realtime_factor"] = None
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row["error"] = repr(exc)
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print(f" error: {exc}", file=sys.stderr)
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rows.append(row)
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args.output_dir.mkdir(parents=True, exist_ok=True)
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results_path = args.output_dir / "results.jsonl"
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summary_path = args.output_dir / "summary.md"
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write_jsonl(results_path, rows)
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summary_path.write_text(markdown_summary(rows, args, model_load_seconds), encoding="utf-8")
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print(f"\nWrote {results_path}")
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print(f"Wrote {summary_path}")
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if __name__ == "__main__":
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main()
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Block a user