Update README.md

README.md

@@ -51,7 +51,7 @@ To transcribe audio files the model can be used as a standalone acoustic model a
  ds = ds.map(map_to_array)
  
  # tokenize
- input_values = processor(ds["speech"][:2], return_tensors="pt", padding="longest").input_values  # Batch size 1
+ input_values = processor(ds[0]["audio"]["array"], return_tensors="pt", padding="longest").input_values  # Batch size 1
  
  # retrieve logits
  logits = model(input_values).logits
@@ -78,15 +78,8 @@ librispeech_eval = load_dataset("librispeech_asr", "clean", split="test")
 model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-100h").to("cuda")
 processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-base-100h")
 
-def map_to_array(batch):
-    speech, _ = sf.read(batch["file"])
-    batch["speech"] = speech
-    return batch
-
-librispeech_eval = librispeech_eval.map(map_to_array)
-
 def map_to_pred(batch):
-    input_values = processor(batch["speech"], return_tensors="pt", padding="longest").input_values
+    input_values = processor(batch["audio"]["array"], return_tensors="pt", padding="longest").input_values
     with torch.no_grad():
         logits = model(input_values.to("cuda")).logits