diff --git a/TTS/vocoder/models/wavernn.py b/TTS/vocoder/models/wavernn.py
index 9b0d6837..1977efb6 100644
--- a/TTS/vocoder/models/wavernn.py
+++ b/TTS/vocoder/models/wavernn.py
@@ -222,10 +222,7 @@ class Wavernn(BaseVocoder):
             samples at once. The Subscale WaveRNN produces 16 samples per step without loss of quality and offers an
             orthogonal method for increasing sampling efficiency.
         """
-        super().__init__()
-
-        self.args = config.model_params
-        self.config = config
+        super().__init__(config)
 
         if isinstance(self.args.mode, int):
             self.n_classes = 2 ** self.args.mode
@@ -572,8 +569,9 @@ class Wavernn(BaseVocoder):
 
     @torch.no_grad()
     def test_run(
-        self, ap: AudioProcessor, samples: List[Dict], output: Dict  # pylint: disable=unused-argument
+        self, assets: Dict, samples: List[Dict], output: Dict  # pylint: disable=unused-argument
     ) -> Tuple[Dict, Dict]:
+        ap = assets["audio_processor"]
         figures = {}
         audios = {}
         for idx, sample in enumerate(samples):
@@ -600,20 +598,21 @@ class Wavernn(BaseVocoder):
     def get_data_loader(  # pylint: disable=no-self-use
         self,
         config: Coqpit,
-        ap: AudioProcessor,
+        assets: Dict,
         is_eval: True,
         data_items: List,
         verbose: bool,
         num_gpus: int,
     ):
+        ap = assets["audio_processor"]
         dataset = WaveRNNDataset(
             ap=ap,
             items=data_items,
             seq_len=config.seq_len,
             hop_len=ap.hop_length,
-            pad=config.model_params.pad,
-            mode=config.model_params.mode,
-            mulaw=config.model_params.mulaw,
+            pad=config.model_args.pad,
+            mode=config.model_args.mode,
+            mulaw=config.model_args.mulaw,
             is_training=not is_eval,
             verbose=verbose,
         )
diff --git a/recipes/ljspeech/wavernn/train_wavernn.py b/recipes/ljspeech/wavernn/train_wavernn.py
index 8f138298..9777a985 100644
--- a/recipes/ljspeech/wavernn/train_wavernn.py
+++ b/recipes/ljspeech/wavernn/train_wavernn.py
@@ -1,7 +1,11 @@
 import os
 
-from TTS.trainer import Trainer, TrainingArgs, init_training
+
+from TTS.trainer import Trainer, TrainingArgs
+from TTS.utils.audio import AudioProcessor
 from TTS.vocoder.configs import WavernnConfig
+from TTS.vocoder.datasets.preprocess import load_wav_data
+from TTS.vocoder.models.wavernn import Wavernn
 
 output_path = os.path.dirname(os.path.abspath(__file__))
 config = WavernnConfig(
@@ -24,6 +28,24 @@ config = WavernnConfig(
     data_path=os.path.join(output_path, "../LJSpeech-1.1/wavs/"),
     output_path=output_path,
 )
-args, config, output_path, _, c_logger, dashboard_logger = init_training(TrainingArgs(), config)
-trainer = Trainer(args, config, output_path, c_logger, dashboard_logger, cudnn_benchmark=True)
+
+# init audio processor
+ap = AudioProcessor(**config.audio.to_dict())
+
+# load training samples
+eval_samples, train_samples = load_wav_data(config.data_path, config.eval_split_size)
+
+# init model
+model = Wavernn(config)
+
+# init the trainer and 🚀
+trainer = Trainer(
+    TrainingArgs(),
+    config,
+    output_path,
+    model=model,
+    train_samples=train_samples,
+    eval_samples=eval_samples,
+    training_assets={"audio_processor": ap},
+)
 trainer.fit()