use jiwer instead of evaluate in benchmarks (#1159)

benchmark/evaluate_yt_commons.py

@@ -5,9 +5,9 @@ import os
 from io import BytesIO
 
 from datasets import load_dataset
-from evaluate import load
+from jiwer import wer
 from pytubefix import YouTube
-from torch.utils.data import DataLoader
+from pytubefix.exceptions import VideoUnavailable
 from tqdm import tqdm
 from transformers.models.whisper.english_normalizer import EnglishTextNormalizer
 
@@ -17,15 +17,19 @@ from faster_whisper import BatchedInferencePipeline, WhisperModel, decode_audio
 def url_to_audio(row):
     buffer = BytesIO()
     yt = YouTube(row["link"])
-    video = (
-        yt.streams.filter(only_audio=True, mime_type="audio/mp4")
-        .order_by("bitrate")
-        .desc()
-        .first()
-    )
-    video.stream_to_buffer(buffer)
-    buffer.seek(0)
-    row["audio"] = decode_audio(buffer)
+    try:
+        video = (
+            yt.streams.filter(only_audio=True, mime_type="audio/mp4")
+            .order_by("bitrate")
+            .desc()
+            .last()
+        )
+        video.stream_to_buffer(buffer)
+        buffer.seek(0)
+        row["audio"] = decode_audio(buffer)
+    except VideoUnavailable:
+        print(f'Failed to download: {row["link"]}')
+        row["audio"] = []
     return row
 
 
@@ -39,19 +43,12 @@ parser.add_argument(
 )
 args = parser.parse_args()
 
-# define the evaluation metric
-wer_metric = load("wer")
-
 with open(os.path.join(os.path.dirname(__file__), "normalizer.json"), "r") as f:
     normalizer = EnglishTextNormalizer(json.load(f))
 
 dataset = load_dataset("mobiuslabsgmbh/youtube-commons-asr-eval", streaming=True).map(
     url_to_audio
 )
-dataset = iter(
-    DataLoader(dataset["test"], batch_size=1, prefetch_factor=4, num_workers=2)
-)
-
 model = WhisperModel("large-v3", device="cuda")
 pipeline = BatchedInferencePipeline(model, device="cuda")
 
@@ -59,7 +56,9 @@ pipeline = BatchedInferencePipeline(model, device="cuda")
 all_transcriptions = []
 all_references = []
 # iterate over the dataset and run inference
-for i, row in tqdm(enumerate(dataset), desc="Evaluating..."):
+for i, row in tqdm(enumerate(dataset["test"]), desc="Evaluating..."):
+    if not row["audio"]:
+        continue
     result, info = pipeline.transcribe(
         row["audio"][0],
         batch_size=8,
@@ -77,7 +76,5 @@ all_transcriptions = [normalizer(transcription) for transcription in all_transcr
 all_references = [normalizer(reference) for reference in all_references]
 
 # compute the WER metric
-wer = 100 * wer_metric.compute(
-    predictions=all_transcriptions, references=all_references
-)
-print("WER: %.3f" % wer)
+word_error_rate = 100 * wer(hypothesis=all_transcriptions, reference=all_references)
+print("WER: %.3f" % word_error_rate)

benchmark/requirements.benchmark.txt

@@ -1,6 +1,5 @@
 transformers
 jiwer
-evaluate
 datasets
 memory_profiler
 py3nvml

benchmark/wer_benchmark.py

@@ -3,7 +3,7 @@ import json
 import os
 
 from datasets import load_dataset
-from evaluate import load
+from jiwer import wer
 from tqdm import tqdm
 from transformers.models.whisper.english_normalizer import EnglishTextNormalizer
 
@@ -25,9 +25,6 @@ model = WhisperModel(model_path, device="cuda")
 # load the dataset with streaming mode
 dataset = load_dataset("librispeech_asr", "clean", split="validation", streaming=True)
 
-# define the evaluation metric
-wer_metric = load("wer")
-
 with open(os.path.join(os.path.dirname(__file__), "normalizer.json"), "r") as f:
     normalizer = EnglishTextNormalizer(json.load(f))
 
@@ -58,7 +55,5 @@ all_transcriptions = [normalizer(transcription) for transcription in all_transcr
 all_references = [normalizer(reference) for reference in all_references]
 
 # compute the WER metric
-wer = 100 * wer_metric.compute(
-    predictions=all_transcriptions, references=all_references
-)
-print("WER: %.3f" % wer)
+word_error_rate = 100 * wer(hypothesis=all_transcriptions, reference=all_references)
+print("WER: %.3f" % word_error_rate)