[Speech] Move all examples to new audio feature (#14045)

* up

* up

* up

* finish

examples/pytorch/speech-recognition/run_speech_recognition_ctc.py

@@ -342,7 +342,7 @@ def main():
 
     if data_args.audio_column_name not in raw_datasets["train"].column_names:
         raise ValueError(
-            f"--audio_column_name {data_args.audio_column_name} not found in dataset '{data_args.dataset_name}'. "
+            f"--audio_column_name '{data_args.audio_column_name}' not found in dataset '{data_args.dataset_name}'. "
             "Make sure to set `--audio_column_name` to the correct audio column - one of "
             f"{', '.join(raw_datasets['train'].column_names)}."
         )

setup.py

@@ -136,7 +136,7 @@ _deps = [
     "scikit-learn",
     "sentencepiece>=0.1.91,!=0.1.92",
     "sigopt",
-    "soundfile",
+    "librosa",
     "sphinx-copybutton",
     "sphinx-markdown-tables",
     "sphinx-rtd-theme==0.4.3",  # sphinx-rtd-theme==0.5.0 introduced big changes in the style.
@@ -254,7 +254,7 @@ extras["sigopt"] = deps_list("sigopt")
 extras["integrations"] = extras["optuna"] + extras["ray"] + extras["sigopt"]
 
 extras["serving"] = deps_list("pydantic", "uvicorn", "fastapi", "starlette")
-extras["audio"] = deps_list("soundfile")
+extras["audio"] = deps_list("librosa")
 extras["speech"] = deps_list("torchaudio") + extras["audio"]  # `pip install ".[speech]"` is deprecated and `pip install ".[torch-speech]"` should be used instead
 extras["torch-speech"] = deps_list("torchaudio") + extras["audio"]
 extras["tf-speech"] = extras["audio"]

src/transformers/dependency_versions_table.py

@@ -54,7 +54,7 @@ deps = {
     "scikit-learn": "scikit-learn",
     "sentencepiece": "sentencepiece>=0.1.91,!=0.1.92",
     "sigopt": "sigopt",
-    "soundfile": "soundfile",
+    "librosa": "librosa",
     "sphinx-copybutton": "sphinx-copybutton",
     "sphinx-markdown-tables": "sphinx-markdown-tables",
     "sphinx-rtd-theme": "sphinx-rtd-theme==0.4.3",

tests/test_modeling_flax_wav2vec2.py

@@ -356,21 +356,13 @@ class FlaxWav2Vec2ModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
-
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").filter(
+            lambda x: x["id"] in [f"1272-141231-000{i}" for i in range(num_samples)]
+        )[:num_samples]["audio"]
 
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
+        return [x["array"] for x in speech_samples]
-
-        return ds["speech"][:num_samples]
 
     def test_inference_ctc_robust_batched(self):
         model = FlaxWav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-large-960h-lv60-self", from_pt=True)

tests/test_modeling_hubert.py

@@ -613,21 +613,11 @@ class HubertModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
-
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").select(range(num_samples))[:num_samples]["audio"]
 
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
+        return [x["array"] for x in speech_samples]
-
-        return ds["speech"][:num_samples]
 
     def _load_superb(self, task, num_samples):
         from datasets import load_dataset

tests/test_modeling_sew.py

@@ -407,21 +407,13 @@ class SEWModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
+        ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").filter(
+            lambda x: x["id"] in [f"1272-141231-000{i}" for i in range(num_samples)]
+        )[:num_samples]["audio"]
 
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
+        return [x["array"] for x in speech_samples]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
-        ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
-
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
-
-        return ds["speech"][:num_samples]
 
     def test_inference_pretrained_batched(self):
         model = SEWModel.from_pretrained("asapp/sew-tiny-100k").to(torch_device)

tests/test_modeling_sew_d.py

@@ -428,21 +428,13 @@ class SEWDModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
+        ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").filter(
+            lambda x: x["id"] in [f"1272-141231-000{i}" for i in range(num_samples)]
+        )[:num_samples]["audio"]
 
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
+        return [x["array"] for x in speech_samples]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
-        ds = load_dataset("patrickvonplaten/librispeech_asr_dummy", "clean", split="validation")
-
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
-
-        return ds["speech"][:num_samples]
 
     def test_inference_pretrained_batched(self):
         model = SEWDModel.from_pretrained("asapp/sew-d-tiny-100k").to(torch_device)

tests/test_modeling_speech_to_text.py

@@ -715,18 +715,11 @@ class Speech2TextModelIntegrationTests(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
-        ds = ds.sort("id").select(range(num_samples)).map(map_to_array)
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").select(range(num_samples))[:num_samples]["audio"]
 
-        return ds["speech"][:num_samples]
+        return [x["array"] for x in speech_samples]
 
     def test_generation_librispeech(self):
         model = Speech2TextForConditionalGeneration.from_pretrained("facebook/s2t-small-librispeech-asr")

tests/test_modeling_tf_hubert.py

@@ -479,21 +479,13 @@ class TFHubertModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
-
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").filter(
+            lambda x: x["id"] in [f"1272-141231-000{i}" for i in range(num_samples)]
+        )[:num_samples]["audio"]
 
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
+        return [x["array"] for x in speech_samples]
-
-        return ds["speech"][:num_samples]
 
     def test_inference_ctc_normal(self):
         model = TFHubertForCTC.from_pretrained("facebook/hubert-large-ls960-ft")

tests/test_modeling_tf_wav2vec2.py

@@ -479,21 +479,13 @@ class TFWav2Vec2ModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
-
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").filter(
+            lambda x: x["id"] in [f"1272-141231-000{i}" for i in range(num_samples)]
+        )[:num_samples]["audio"]
 
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
+        return [x["array"] for x in speech_samples]
-
-        return ds["speech"][:num_samples]
 
     def test_inference_ctc_normal(self):
         model = TFWav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-base-960h")

tests/test_modeling_wav2vec2.py

@@ -900,21 +900,13 @@ class Wav2Vec2ModelIntegrationTest(unittest.TestCase):
     def _load_datasamples(self, num_samples):
         from datasets import load_dataset
 
-        import soundfile as sf
-
-        ids = [f"1272-141231-000{i}" for i in range(num_samples)]
-
-        # map files to raw
-        def map_to_array(batch):
-            speech, _ = sf.read(batch["file"])
-            batch["speech"] = speech
-            return batch
-
         ds = load_dataset("hf-internal-testing/librispeech_asr_dummy", "clean", split="validation")
+        # automatic decoding with librispeech
+        speech_samples = ds.sort("id").filter(
+            lambda x: x["id"] in [f"1272-141231-000{i}" for i in range(num_samples)]
+        )[:num_samples]["audio"]
 
-        ds = ds.filter(lambda x: x["id"] in ids).sort("id").map(map_to_array)
+        return [x["array"] for x in speech_samples]
-
-        return ds["speech"][:num_samples]
 
     def _load_superb(self, task, num_samples):
         from datasets import load_dataset