Evaluates automatic speech recognition systems on real-world meeting and close-talking microphone speech. It measures how well acoustic models trained on raw waveforms generalize to challenging multi-microphone environments compared to traditional feature-based baselines. Use when the user wants to benchmark on CHiME4, AMI, or asks about evaluating this task. Reports WER.
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---
name: chime4-ami-asr-eval
description: Evaluates automatic speech recognition systems on real-world meeting and close-talking microphone speech. It measures how well acoustic models trained on raw waveforms generalize to challenging multi-microphone environments compared to traditional feature-based baselines. Use when the user wants to benchmark on CHiME4, AMI, or asks about evaluating this task. Reports WER.
metadata:
skill_kind: dataset_eval
source_arxiv: 1906.11047
bibtex_key: [REDACTED]
confidence: high
---
# chime4-ami-asr-eval
> Multi-Span Acoustic Modelling using Raw Waveform Signals — Patrick von Platen, Chao Zhang, Philip Woodland (arXiv:1906.11047, 2019)
## What this evaluates
Evaluates automatic speech recognition systems on real-world meeting and close-talking microphone speech. It measures how well acoustic models trained on raw waveforms generalize to challenging multi-microphone environments compared to traditional feature-based baselines.
## Datasets
- **CHiME4** — total ?; splits: train (-1), dev (-1)
- **AMI** — total ?; splits: train (-1), dev (-1), eval (-1)
## Metrics
- `WER` **(primary)** — range: percent
- Word Error Rate: the minimum number of insertions, deletions, and substitutions of words required to transform the predicted transcript into the reference transcript, divided by the total number of words in the reference.
## Input / output format
**Input**: Raw waveform audio signals, shifted by 10ms (160 samples) per frame. Input is normalized to zero mean and unit variance (globally for CHiME4, per utterance and per meeting for AMI).
**Output**: Softmax probabilities over clustered triphone-states, decoded via Viterbi algorithm into word sequences using a language model.
## Scoring recipe
```python
def compute_wer(predictions, references):
total_errors = 0
total_words = 0
for pred, ref in zip(predictions, references):
edit_dist = levenshtein_distance(pred.split(), ref.split())
total_errors += edit_dist
total_words += len(ref.split())
return (total_errors / total_words) * 100 if total_words > 0 else 0.0
```
## Common pitfalls
- Microphone failure detection algorithm must be applied to select the correct channel for the CHiME4 dev set.
- Normalization differs between datasets: CHiME4 uses global normalization, while AMI uses per-utterance zero mean and per-meeting unit variance.
- Alignment is done at 10ms intervals to clustered triphone-states, not directly to words.
## Evidence (verbatim from paper)
> Experimental results on CHiME4 and AMI datasets show a 5% relative WER reduction over FBANK-based models, with evidence that smaller kernels (50 params) outperform larger ones (400 params), and that learned filters diverge significantly from traditional log-Mel filters, indicating effective feature learning beyond handcrafted designs.
## Citation
```bibtex
@misc{vonplaten2019multi,
title={Multi-Span Acoustic Modelling using Raw Waveform Signals},
author={Patrick von Platen, Chao Zhang, Philip Woodland},
year={2019},
note={arXiv:1906.11047}
}
```
- arXiv: 1906.11047