Create custom_inference.py

custom_inference.py

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+import torch
+from speechbrain.inference.interfaces import Pretrained
+
+
+class CustomEncoderWav2vec2Classifier(Pretrained):
+    """A ready-to-use class for utterance-level classification (e.g, speaker-id,
+    language-id, emotion recognition, keyword spotting, etc).
+    The class assumes that an self-supervised encoder like wav2vec2/hubert and a classifier model
+    are defined in the yaml file. If you want to
+    convert the predicted index into a corresponding text label, please
+    provide the path of the label_encoder in a variable called 'lab_encoder_file'
+    within the yaml.
+    The class can be used either to run only the encoder (encode_batch()) to
+    extract embeddings or to run a classification step (classify_batch()).
+    ```
+    Example
+    -------
+    >>> import torchaudio
+    >>> from speechbrain.pretrained import EncoderClassifier
+    >>> # Model is downloaded from the speechbrain HuggingFace repo
+    >>> tmpdir = getfixture("tmpdir")
+    >>> classifier = EncoderClassifier.from_hparams(
+    ...     source="speechbrain/spkrec-ecapa-voxceleb",
+    ...     savedir=tmpdir,
+    ... )
+    >>> # Compute embeddings
+    >>> signal, fs = torchaudio.load("samples/audio_samples/example1.wav")
+    >>> embeddings =  classifier.encode_batch(signal)
+    >>> # Classification
+    >>> prediction =  classifier .classify_batch(signal)
+    """
+
+    def __init__(self, *args, **kwargs):
+        super().__init__(*args, **kwargs)
+
+    def encode_batch(self, wavs, wav_lens=None, normalize=False):
+        """Encodes the input audio into a single vector embedding.
+        The waveforms should already be in the model's desired format.
+        You can call:
+        ``normalized = <this>.normalizer(signal, sample_rate)``
+        to get a correctly converted signal in most cases.
+        Arguments
+        ---------
+        wavs : torch.tensor
+            Batch of waveforms [batch, time, channels] or [batch, time]
+            depending on the model. Make sure the sample rate is fs=16000 Hz.
+        wav_lens : torch.tensor
+            Lengths of the waveforms relative to the longest one in the
+            batch, tensor of shape [batch]. The longest one should have
+            relative length 1.0 and others len(waveform) / max_length.
+            Used for ignoring padding.
+        normalize : bool
+            If True, it normalizes the embeddings with the statistics
+            contained in mean_var_norm_emb.
+        Returns
+        -------
+        torch.tensor
+            The encoded batch
+        """
+        # Manage single waveforms in input
+        if len(wavs.shape) == 1:
+            wavs = wavs.unsqueeze(0)
+
+        # Assign full length if wav_lens is not assigned
+        if wav_lens is None:
+            wav_lens = torch.ones(wavs.shape[0], device=self.device)
+
+        # Storing waveform in the specified device
+        wavs, wav_lens = wavs.to(self.device), wav_lens.to(self.device)
+        wavs = wavs.float()
+
+        # Computing features and embeddings
+        outputs = self.mods.wav2vec2(wavs)
+
+        # last dim will be used for AdaptativeAVG pool
+        outputs = self.mods.avg_pool(outputs, wav_lens)
+        outputs = outputs.view(outputs.shape[0], -1)
+        return outputs
+
+    def classify_batch(self, wavs, wav_lens=None):
+        """Performs classification on the top of the encoded features.
+        It returns the posterior probabilities, the index and, if the label
+        encoder is specified it also the text label.
+        Arguments
+        ---------
+        wavs : torch.tensor
+            Batch of waveforms [batch, time, channels] or [batch, time]
+            depending on the model. Make sure the sample rate is fs=16000 Hz.
+        wav_lens : torch.tensor
+            Lengths of the waveforms relative to the longest one in the
+            batch, tensor of shape [batch]. The longest one should have
+            relative length 1.0 and others len(waveform) / max_length.
+            Used for ignoring padding.
+        Returns
+        -------
+        out_prob
+            The log posterior probabilities of each class ([batch, N_class])
+        score:
+            It is the value of the log-posterior for the best class ([batch,])
+        index
+            The indexes of the best class ([batch,])
+        text_lab:
+            List with the text labels corresponding to the indexes.
+            (label encoder should be provided).
+        """
+        outputs = self.encode_batch(wavs, wav_lens)
+        outputs = self.mods.output_mlp(outputs)
+        out_prob = self.hparams.softmax(outputs)
+        score, index = torch.max(out_prob, dim=-1)
+        text_lab = self.hparams.label_encoder.decode_torch(index)
+        return out_prob, score, index, text_lab
+
+    def classify_file(self, path):
+        """Classifies the given audiofile into the given set of labels.
+        Arguments
+        ---------
+        path : str
+            Path to audio file to classify.
+        Returns
+        -------
+        out_prob
+            The log posterior probabilities of each class ([batch, N_class])
+        score:
+            It is the value of the log-posterior for the best class ([batch,])
+        index
+            The indexes of the best class ([batch,])
+        text_lab:
+            List with the text labels corresponding to the indexes.
+            (label encoder should be provided).
+        """
+        waveform = self.load_audio(path)
+        # Fake a batch:
+        batch = waveform.unsqueeze(0)
+        rel_length = torch.tensor([1.0])
+        outputs = self.encode_batch(batch, rel_length)
+        outputs = self.mods.output_mlp(outputs).squeeze(1)
+        out_prob = self.hparams.softmax(outputs)
+        score, index = torch.max(out_prob, dim=-1)
+        text_lab = self.hparams.label_encoder.decode_torch(index)
+        return out_prob, score, index, text_lab
+
+    def forward(self, wavs, wav_lens=None, normalize=False):
+        return self.encode_batch(
+            wavs=wavs, wav_lens=wav_lens, normalize=normalize
+        )