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diff --git a/__output_audiosep/0004_0000/model_0059.pt b/__output_audiosep/0004_0000/model_0059.pt
new file mode 100644
index 0000000000000000000000000000000000000000..9df73d2b10711d150af3e7f6951ae74b19462e5b
--- /dev/null
+++ b/__output_audiosep/0004_0000/model_0059.pt
@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:ef62b0fdd9e9b81000da0db190be1df7e5451d70ab3c86609cab409ec3e38ab8
+size 34402
diff --git a/__output_audiosep/1004_0000/model_0119.pt b/__output_audiosep/1004_0000/model_0119.pt
new file mode 100644
index 0000000000000000000000000000000000000000..486897ba8edf9b6029db12850c946b9f2b1d7038
--- /dev/null
+++ b/__output_audiosep/1004_0000/model_0119.pt
@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:61768dbf9e81845dc656c5da3e8fe5ea053f73108f422caf7879b4ab55a0792a
+size 12755810
diff --git a/__output_audiosep/3001_0000/model_0199.pt b/__output_audiosep/3001_0000/model_0199.pt
new file mode 100644
index 0000000000000000000000000000000000000000..ab529c812b2c49e869d122738bde06f82e26f2dd
--- /dev/null
+++ b/__output_audiosep/3001_0000/model_0199.pt
@@ -0,0 +1,3 @@
+version https://git-lfs.github.com/spec/v1
+oid sha256:11341d63bad0a1ec15c5a94c5cc6f049720869fb9988da45ef7d07edd3c82e21
+size 12743211
diff --git a/app.py b/app.py
new file mode 100644
index 0000000000000000000000000000000000000000..c1ddedb381eb376aacb4eac782b82a861a23f980
--- /dev/null
+++ b/app.py
@@ -0,0 +1,7 @@
+import sys
+import os
+src_path = os.path.abspath(os.path.join(os.path.dirname(__file__), 'src'))
+os.sys.path.append(src_path)
+from gyraudio.audio_separation.visualization.interactive_audio import main as interactive_audio_main
+if __name__ == "__main__":
+    interactive_audio_main(sys.argv[1:])
diff --git a/audio_samples/0009/mix_snr_-1.wav b/audio_samples/0009/mix_snr_-1.wav
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diff --git a/audio_samples/0009/noise.wav b/audio_samples/0009/noise.wav
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diff --git a/audio_samples/0009/voice.wav b/audio_samples/0009/voice.wav
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diff --git a/requirements.txt b/requirements.txt
new file mode 100644
index 0000000000000000000000000000000000000000..d3e98baa6b00e88de5c4e4ea5c24a82246cc4738
--- /dev/null
+++ b/requirements.txt
@@ -0,0 +1,3 @@
+batch_processing
+interactive-pipe>=0.7.0
+torch>=2.0.0
diff --git a/src/gyraudio/__init__.py b/src/gyraudio/__init__.py
new file mode 100644
index 0000000000000000000000000000000000000000..39c149120fc0b690fbf4c5557336eaabac0a6e13
--- /dev/null
+++ b/src/gyraudio/__init__.py
@@ -0,0 +1,2 @@
+from pathlib import Path
+root_dir = Path(__file__).parent.parent.parent
diff --git a/src/gyraudio/audio_separation/architecture/building_block.py b/src/gyraudio/audio_separation/architecture/building_block.py
new file mode 100644
index 0000000000000000000000000000000000000000..82b791a79ecd847614f42e63daf0ba305c1c6b63
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/building_block.py
@@ -0,0 +1,51 @@
+import torch
+from typing import List
+
+
+class FilterBank(torch.nn.Module):
+    """Convolution filter bank (linear)
+    Serves as an embedding for the audio signal
+    """
+
+    def __init__(self, ch_in: int, out_dim=16, k_size=5, dilation_list: List[int] = [1, 2, 4, 8]):
+        super().__init__()
+        self.out_dim = out_dim
+        self.source_modality_conv = torch.nn.ModuleList()
+        for dilation in dilation_list:
+            self.source_modality_conv.append(
+                torch.nn.Conv1d(ch_in, out_dim//len(dilation_list), k_size, dilation=dilation, padding=(dilation*(k_size//2)))
+            )
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        out = torch.cat([conv(x) for conv in self.source_modality_conv], axis=1)
+        assert out.shape[1] == self.out_dim
+        return out
+
+
+class ResConvolution(torch.nn.Module):
+    """ResNet building block
+    https://paperswithcode.com/method/residual-connection
+    """
+
+    def __init__(self, ch, hdim=None, k_size=5):
+        super().__init__()
+        hdim = hdim or ch
+        self.conv1 = torch.nn.Conv1d(ch, hdim, k_size, padding=k_size//2)
+        self.conv2 = torch.nn.Conv1d(hdim, ch, k_size, padding=k_size//2)
+        self.non_linearity = torch.nn.ReLU()
+
+    def forward(self, x_in):
+        x = self.conv1(x_in)
+        x = self.non_linearity(x)
+        x = self.conv2(x)
+        x += x_in
+        x = self.non_linearity(x)
+        return x
+
+
+if __name__ == "__main__":
+    model = FilterBank(1, 16)
+    inp = torch.rand(2, 1, 2048)
+    out = model(inp)
+    print(model)
+    print(out[0].shape)
diff --git a/src/gyraudio/audio_separation/architecture/flat_conv.py b/src/gyraudio/audio_separation/architecture/flat_conv.py
new file mode 100644
index 0000000000000000000000000000000000000000..4ee9b47a838c7666e8d19dab0fa3b48c8d092905
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/flat_conv.py
@@ -0,0 +1,62 @@
+import torch
+from gyraudio.audio_separation.architecture.model import SeparationModel
+from typing import Tuple
+
+
+class FlatConvolutional(SeparationModel):
+    """Convolutional neural network for audio separation,
+    No decimation, no bottleneck, just basic signal processing
+    """
+
+    def __init__(self,
+                 ch_in: int = 1,
+                 ch_out: int = 2,
+                 h_dim=16,
+                 k_size=5,
+                 dilation=1
+                 ) -> None:
+        super().__init__()
+        self.conv1 = torch.nn.Conv1d(
+            ch_in, h_dim, k_size, 
+            dilation=dilation, padding=dilation*(k_size//2))
+        self.conv2 = torch.nn.Conv1d(
+            h_dim, h_dim, k_size,
+            dilation=dilation, padding=dilation*(k_size//2))
+        self.conv3 = torch.nn.Conv1d(
+            h_dim, h_dim, k_size,
+            dilation=dilation, padding=dilation*(k_size//2))
+        self.conv4 = torch.nn.Conv1d(
+            h_dim, h_dim, k_size,
+            dilation=dilation, padding=dilation*(k_size//2))
+        self.relu = torch.nn.ReLU()
+        self.encoder = torch.nn.Sequential(
+            self.conv1,
+            self.relu,
+            self.conv2,
+            self.relu,
+            self.conv3,
+            self.relu,
+            self.conv4,
+            self.relu
+        )
+        self.demux = torch.nn.Sequential(*(
+            torch.nn.Conv1d(h_dim, h_dim//2, 1),  # conv1x1
+            torch.nn.ReLU(),
+            torch.nn.Conv1d(h_dim//2, ch_out, 1),  # conv1x1
+        ))
+
+    def forward(self, mixed_sig_in: torch.Tensor) -> Tuple[torch.Tensor, torch.Tensor]:
+        """Perform feature extraction followed by classifier head
+
+        Args:
+            sig_in (torch.Tensor): [N, C, T]
+
+        Returns:
+            torch.Tensor: logits (not probabilities) [N, n_classes]
+        """
+        # Convolution backbone
+        # [N, C, T] -> [N, h, T]
+        features = self.encoder(mixed_sig_in)
+        # [N, h, T] -> [N, 2, T]
+        demuxed = self.demux(features)
+        return torch.chunk(demuxed, 2, dim=1)  # [N, 1, T], [N, 1, T]
diff --git a/src/gyraudio/audio_separation/architecture/model.py b/src/gyraudio/audio_separation/architecture/model.py
new file mode 100644
index 0000000000000000000000000000000000000000..ab38c4e89c40592b9179d8ba98302deecbfea33b
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/model.py
@@ -0,0 +1,28 @@
+import torch
+
+
+class SeparationModel(torch.nn.Module):
+    def __init__(self, *args, **kwargs) -> None:
+        super().__init__(*args, **kwargs)
+
+    def count_parameters(self) -> int:
+        """Count the total number of parameters of the model
+
+        Returns:
+            int: total amount of parameters
+        """
+        return sum(p.numel() for p in self.parameters() if p.requires_grad)
+
+    def receptive_field(self) -> int:
+        """Compute the receptive field of the model
+
+        Returns:
+            int: receptive field
+        """
+        input_tensor = torch.rand(1, 1, 4096, requires_grad=True)
+        out, out_noise = self.forward(input_tensor)
+        grad = torch.zeros_like(out)
+        grad[..., out.shape[-1]//2] = torch.nan  # set NaN gradient at the middle of the output
+        out.backward(gradient=grad)
+        self.zero_grad()  # reset to avoid future problems
+        return int(torch.sum(input_tensor.grad.isnan()).cpu())  # Count NaN in the input
diff --git a/src/gyraudio/audio_separation/architecture/neutral.py b/src/gyraudio/audio_separation/architecture/neutral.py
new file mode 100644
index 0000000000000000000000000000000000000000..b81c5319a173d5ce1fad5ef95713c3ed10e29309
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/neutral.py
@@ -0,0 +1,15 @@
+
+import torch
+from gyraudio.audio_separation.architecture.model import SeparationModel
+
+
+class NeutralModel(SeparationModel):
+    def __init__(self, *args, **kwargs) -> None:
+        super().__init__(*args, **kwargs)
+        self.fake = torch.nn.Conv1d(1, 1, 1, bias=False)
+
+    def forward(self, x: torch.Tensor) -> torch.Tensor:
+        """Identity function
+        """
+        n = self.fake(x)
+        return x, n
diff --git a/src/gyraudio/audio_separation/architecture/transformer.py b/src/gyraudio/audio_separation/architecture/transformer.py
new file mode 100644
index 0000000000000000000000000000000000000000..40467c6cf0b4697d00c85bd18d23ff511061d06b
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/transformer.py
@@ -0,0 +1,91 @@
+import torch
+from gyraudio.audio_separation.architecture.model import SeparationModel
+from gyraudio.audio_separation.architecture.building_block import FilterBank
+from typing import Optional
+
+
+class TransformerModel(SeparationModel):
+    """Transformer base model
+    =========================
+    - Embed signal with a filter bank
+    - No positional encoding (Potential  =add/concatenate positional encoding)
+    - `nlayers` * transformer blocks
+    """
+
+    def __init__(self,
+                 nhead: int = 8,  # H
+                 nlayers: int = 4,  # L
+                 k_size=5,
+                 embedding_dim: int = 64,  # D
+                 ch_in: int = 1,
+                 ch_out: int = 1,
+                 dropout: float = 0.,  # dr
+                 positional_encoding: str = None
+                 ) -> None:
+        """Transformer base model
+
+        Args:
+            nhead (int): number of heads in each of the MHA models
+            embedding_dim (int): D number of channels in the audio embeddings
+                = output of the filter bank
+                assume `embedding_dim` = `h_dim`
+                h_dim is the hidden dimension of the model.
+            nlayers (int): number of nn.TransformerEncoderLayer in nn.TransformerEncoder
+            dropout (float, optional): dropout value. Defaults to 0.
+        """
+        super().__init__()
+        self.model_type = "Transformer"
+        h_dim = embedding_dim  # use the same embedding & hidden dimensions
+
+        self.encoder = FilterBank(ch_in, embedding_dim, k_size=k_size)
+        if positional_encoding is None:
+            self.pos_encoder = torch.nn.Identity()
+        else:
+            raise NotImplementedError(
+                f"Unknown positional encoding {positional_encoding} - should be add/concat in future")
+        # self.pos_encoder = PositionalEncoding(h_dim, dropout=dropout)
+
+        encoder_layers = torch.nn.TransformerEncoderLayer(
+            d_model=h_dim,  # input dimension to the transformer encoder layer
+            nhead=nhead,  # number of heads for MHA (Multi-head attention)
+            dim_feedforward=h_dim,  # output dimension of the MLP on top of the transformer.
+            dropout=dropout,
+            batch_first=True
+        )  # we assume h_dim = d_model = dim_feedforward
+
+        self.transformer_encoder = torch.nn.TransformerEncoder(
+            encoder_layers,
+            num_layers=nlayers
+        )
+        self.h_dim = h_dim
+        self.target_modality_conv = torch.nn.Conv1d(h_dim, ch_out, 1)  # conv1x1 channel mixer
+        # Note: we could finish with a few residual conv blocks... this is pure signal processing
+
+    def forward(
+        self, src: torch.LongTensor,
+        src_mask: Optional[torch.FloatTensor] = None
+    ) -> torch.FloatTensor:
+        """Embdeddings, positional encoders, go trough `nlayers` of residual {multi (`nhead`) attention heads + MLP}.
+
+        Args:
+            src (torch.LongTensor): [N, 1, T] audio signal
+
+        Returns:
+            torch.FloatTensor: separated signal [N, 1, T]
+        """
+        src = self.encoder(src)  # [N, 1, T] -> [N, D, T]
+        src = src.transpose(-1, -2)  # [N, D, T] -> [N, T, D] # Transformer  expects (batch N, seq "T", features "D")
+        src = self.pos_encoder(src)  # -> [N, T, D]  - add positional encoding
+
+        output = self.transformer_encoder(src, mask=src_mask)  # -> [N, T, D]
+        output = output.transpose(-1, -2)  # -> [N, D, T]
+        output = self.target_modality_conv(output)  # -> [N, 1, T]
+        return output, None
+
+
+if __name__ == "__main__":
+    model = TransformerModel()
+    inp = torch.rand(2, 1, 2048)
+    out = model(inp)
+    print(model)
+    print(out[0].shape)
diff --git a/src/gyraudio/audio_separation/architecture/unet.py b/src/gyraudio/audio_separation/architecture/unet.py
new file mode 100644
index 0000000000000000000000000000000000000000..89bf30f2d2f360e14e91e2bdd18361984161506f
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/unet.py
@@ -0,0 +1,151 @@
+import torch
+from gyraudio.audio_separation.architecture.model import SeparationModel
+from gyraudio.audio_separation.architecture.building_block import ResConvolution
+from typing import Optional
+# import logging
+
+
+class EncoderSingleStage(torch.nn.Module):
+    """
+    Extend channels
+    Resnet
+    Downsample by 2
+    """
+
+    def __init__(self, ch: int, ch_out: int, hdim: Optional[int] = None, k_size=5):
+        # ch_out ~ ch_in*extension_factor
+        super().__init__()
+        hdim = hdim or ch
+        self.extension_conv = torch.nn.Conv1d(ch, ch_out, k_size, padding=k_size//2)
+        self.res_conv = ResConvolution(ch_out, hdim=hdim, k_size=k_size)
+        # warning on maxpooling jitter offset!
+        self.max_pool = torch.nn.MaxPool1d(kernel_size=2)
+
+    def forward(self, x):
+        x = self.extension_conv(x)
+        x = self.res_conv(x)
+        x_ds = self.max_pool(x)
+        return x, x_ds
+
+
+class DecoderSingleStage(torch.nn.Module):
+    """
+    Upsample by 2
+    Resnet
+    Extend channels
+    """
+
+    def __init__(self, ch: int, ch_out: int, hdim: Optional[int] = None, k_size=5):
+        """Decoder stage
+
+        Args:
+            ch (int): channel size (downsampled & skip connection have same channel size)
+            ch_out (int): number of output channels (shall match the number of input channels of the next stage)
+            hdim (Optional[int], optional): Hidden dimension used in the residual block. Defaults to None.
+            k_size (int, optional): Convolution size. Defaults to 5.
+        Notes:
+        ======
+        ch_out = 2*ch/extension_factor
+
+        self.scale_mixers_conv
+        - tells how lower decoded (x_ds) scale is merged with current encoded scale (x_skip)
+        - could be a pointwise aka conv1x1
+        """
+
+        super().__init__()
+        hdim = hdim or ch
+        self.scale_mixers_conv = torch.nn.Conv1d(2*ch, ch_out, k_size, padding=k_size//2)
+
+        self.res_conv = ResConvolution(ch_out, hdim=hdim, k_size=k_size)
+        # warning: Linear interpolation shall be "conjugated" with the skipping downsampling
+        # special care shall be taken care of regarding offsets
+        # https://arxiv.org/abs/1806.03185
+        self.upsample = torch.nn.Upsample(scale_factor=2, mode="linear", align_corners=True)
+        self.non_linearity = torch.nn.ReLU()
+
+    def forward(self, x_ds: torch.Tensor, x_skip: torch.Tensor) -> torch.Tensor:
+        """"""
+        x_us = self.upsample(x_ds)  # [N, ch, T/2] -> [N, ch, T]
+
+        x = torch.cat([x_us, x_skip], dim=1)  # [N, 2.ch, T]
+        x = self.scale_mixers_conv(x)  # [N, ch_out, T]
+        x = self.non_linearity(x)
+        x = self.res_conv(x)  # [N, ch_out, T]
+        return x
+
+
+class ResUNet(SeparationModel):
+    """Convolutional neural network for audio separation,
+
+    Decimation, bottleneck
+    """
+
+    def __init__(self,
+                 ch_in: int = 1,
+                 ch_out: int = 2,
+                 channels_extension: float = 1.5,
+                 h_dim=16,
+                 k_size=5,
+                 ) -> None:
+        super().__init__()
+        self.need_split = ch_out != ch_in
+        self.ch_out = ch_out
+        self.source_modality_conv = torch.nn.Conv1d(ch_in, h_dim, k_size, padding=k_size//2)
+        self.encoder_list = torch.nn.ModuleList()
+        self.decoder_list = torch.nn.ModuleList()
+        self.non_linearity = torch.nn.ReLU()
+
+        h_dim_current = h_dim
+        for _level in range(4):
+            h_dim_ds = int(h_dim_current*channels_extension)
+            self.encoder_list.append(EncoderSingleStage(h_dim_current, h_dim_ds, k_size=k_size))
+            self.decoder_list.append(DecoderSingleStage(h_dim_ds, h_dim_current, k_size=k_size))
+            h_dim_current = h_dim_ds
+        self.bottleneck = ResConvolution(h_dim_current, k_size=k_size)
+        self.target_modality_conv = torch.nn.Conv1d(h_dim, ch_out, 1)  # conv1x1 channel mixer
+
+    def forward(self, x_in):
+        # x_in (1, 2048)
+        x0 = self.source_modality_conv(x_in)
+        x0 = self.non_linearity(x0)
+        # x0 -> (16, 2048)
+
+        x1_skip, x1_ds = self.encoder_list[0](x0)
+        # x1_skip -> (24, 2048)
+        # x1_ds   -> (24, 1024)
+        # print(x1_skip.shape, x1_ds.shape)
+
+        x2_skip, x2_ds = self.encoder_list[1](x1_ds)
+        # x2_skip -> (36, 1024)
+        # x2_ds   -> (36, 512)
+        # print(x2_skip.shape, x2_ds.shape)
+
+        x3_skip, x3_ds = self.encoder_list[2](x2_ds)
+        # x3_skip -> (54, 512)
+        # x3_ds   -> (54, 256)
+        # print(x3_skip.shape, x3_ds.shape)
+
+        x4_skip, x4_ds = self.encoder_list[3](x3_ds)
+        # x4_skip -> (81, 256)
+        # x4_ds   -> (81, 128)
+        # print(x4_skip.shape, x4_ds.shape)
+
+        x4_dec = self.bottleneck(x4_ds)
+        x3_dec = self.decoder_list[3](x4_dec, x4_skip)
+        x2_dec = self.decoder_list[2](x3_dec, x3_skip)
+        x1_dec = self.decoder_list[1](x2_dec, x2_skip)
+        x0_dec = self.decoder_list[0](x1_dec, x1_skip)
+        demuxed = self.target_modality_conv(x0_dec)
+        # no relu
+        if self.need_split:
+            return torch.chunk(demuxed, self.ch_out, dim=1)
+        return demuxed, None
+
+
+if __name__ == "__main__":
+    model = ResUNet()
+    inp = torch.rand(2, 1, 2048)
+    out = model(inp)
+    print(model)
+    print(model.count_parameters())
+    print(out[0].shape)
diff --git a/src/gyraudio/audio_separation/architecture/wave_unet.py b/src/gyraudio/audio_separation/architecture/wave_unet.py
new file mode 100644
index 0000000000000000000000000000000000000000..b330c35ad80bb373e2a386355cd77aeb5153185c
--- /dev/null
+++ b/src/gyraudio/audio_separation/architecture/wave_unet.py
@@ -0,0 +1,163 @@
+import torch
+from gyraudio.audio_separation.architecture.model import SeparationModel
+from typing import Optional, Tuple
+
+
+def get_non_linearity(activation: str):
+    if activation == "LeakyReLU":
+        non_linearity = torch.nn.LeakyReLU()
+    else:
+        non_linearity = torch.nn.ReLU()
+    return non_linearity
+
+
+class BaseConvolutionBlock(torch.nn.Module):
+    def __init__(self, ch_in, ch_out: int, k_size: int, activation="LeakyReLU", dropout: float = 0, bias: bool = True) -> None:
+        super().__init__()
+        self.conv = torch.nn.Conv1d(ch_in, ch_out, k_size, padding=k_size//2, bias=bias)
+        self.non_linearity = get_non_linearity(activation)
+        self.dropout = torch.nn.Dropout1d(p=dropout)
+
+    def forward(self, x_in: torch.Tensor) -> torch.Tensor:
+        x = self.conv(x_in)  # [N, ch_in, T] -> [N, ch_in+channels_extension, T]
+        x = self.non_linearity(x)
+        x = self.dropout(x)
+        return x
+
+
+class EncoderStage(torch.nn.Module):
+    """Conv (and extend channels), downsample 2 by skipping samples
+    """
+
+    def __init__(self, ch_in: int, ch_out: int, k_size: int = 15, dropout: float = 0, bias: bool = True) -> None:
+
+        super().__init__()
+
+        self.conv_block = BaseConvolutionBlock(ch_in, ch_out, k_size=k_size, dropout=dropout, bias=bias)
+
+    def forward(self, x):
+        x = self.conv_block(x)
+
+        x_ds = x[..., ::2]
+        # ch_out = ch_in+channels_extension
+        return x, x_ds
+
+
+class DecoderStage(torch.nn.Module):
+    """Upsample by 2, Concatenate with skip connection, Conv (and shrink channels)
+    """
+
+    def __init__(self, ch_in: int, ch_out: int, k_size: int = 5, dropout: float = 0., bias: bool = True) -> None:
+        """Decoder stage
+        """
+
+        super().__init__()
+        self.conv_block = BaseConvolutionBlock(ch_in, ch_out, k_size=k_size, dropout=dropout, bias=bias)
+        self.upsample = torch.nn.Upsample(scale_factor=2, mode="linear", align_corners=True)
+
+    def forward(self, x_ds: torch.Tensor, x_skip: torch.Tensor) -> torch.Tensor:
+        """"""
+        x_us = self.upsample(x_ds)  # [N, ch, T/2] -> [N, ch, T]
+        x = torch.cat([x_us, x_skip], dim=1)  # [N, 2.ch, T]
+        x = self.conv_block(x)  # [N, ch_out, T]
+        return x
+
+
+class WaveUNet(SeparationModel):
+    """UNET in temporal domain (waveform)
+    = Multiscale convolutional neural network for audio separation
+    https://arxiv.org/abs/1806.03185
+    """
+
+    def __init__(self,
+                 ch_in: int = 1,
+                 ch_out: int = 2,
+                 channels_extension: int = 24,
+                 k_conv_ds: int = 15,
+                 k_conv_us: int = 5,
+                 num_layers: int = 6,
+                 dropout: float = 0.0,
+                 bias: bool = True,
+                 ) -> None:
+        super().__init__()
+        self.need_split = ch_out != ch_in
+        self.ch_out = ch_out
+        self.encoder_list = torch.nn.ModuleList()
+        self.decoder_list = torch.nn.ModuleList()
+        # Defining first encoder
+        self.encoder_list.append(EncoderStage(ch_in, channels_extension, k_size=k_conv_ds, dropout=dropout, bias=bias))
+        for level in range(1, num_layers+1):
+            ch_i = level*channels_extension
+            ch_o = (level+1)*channels_extension
+            if level < num_layers:
+                # Skipping last encoder since we defined the first one outside the loop
+                self.encoder_list.append(EncoderStage(ch_i, ch_o, k_size=k_conv_ds, dropout=dropout, bias=bias))
+            self.decoder_list.append(DecoderStage(ch_o+ch_i, ch_i, k_size=k_conv_us, dropout=dropout, bias=bias))
+        self.bottleneck = BaseConvolutionBlock(
+            num_layers*channels_extension,
+            (num_layers+1)*channels_extension,
+            k_size=k_conv_ds,
+            dropout=dropout,
+            bias=bias)
+        self.dropout = torch.nn.Dropout1d(p=dropout)
+        self.target_modality_conv = torch.nn.Conv1d(
+            channels_extension+ch_in, ch_out, 1, bias=bias)  # conv1x1 channel mixer
+
+    def forward(self, x_in: torch.Tensor) -> Tuple[torch.Tensor, Optional[torch.Tensor]]:
+        """Forward UNET pass
+
+        ```
+        (1  , 2048)----------------->(24 , 2048) > (1  , 2048)
+            v                            ^
+        (24 , 1024)----------------->(48 , 1024)
+            v                            ^
+        (48 , 512 )----------------->(72 , 512 )
+            v                            ^
+        (72 , 256 )----------------->(96 , 256 )
+            v                            ^
+        (96 , 128 )----BOTTLENECK--->(120, 128 )
+        ```
+
+        """
+        skipped_list = []
+        ds_list = [x_in]
+        for level, enc in enumerate(self.encoder_list):
+            x_skip, x_ds = enc(ds_list[-1])
+            skipped_list.append(x_skip)
+            ds_list.append(x_ds.clone())
+            # print(x_skip.shape, x_ds.shape)
+        x_dec = self.bottleneck(ds_list[-1])
+        for level, dec in enumerate(self.decoder_list[::-1]):
+            x_dec = dec(x_dec, skipped_list[-1-level])
+        # print(x_dec.shape)
+        x_dec = torch.cat([x_dec, x_in], dim=1)
+        # print(x_dec.shape)
+        x_dec = self.dropout(x_dec)
+        demuxed = self.target_modality_conv(x_dec)
+        # print(demuxed.shape)
+        if self.need_split:
+            return torch.chunk(demuxed, self.ch_out, dim=1)
+        return demuxed, None
+
+        # x_skip, x_ds
+        # (24, 2048), (24, 1024)
+        # (48, 1024), (48, 512 )
+        # (72, 512 ), (72, 256 )
+        # (96, 256 ), (96, 128 )
+
+        # (120, 128 )
+        # (96 , 256 )
+        # (72 , 512 )
+        # (48 , 1024)
+        # (24 , 2048)
+        # (25 , 2048) demuxed - after concat
+        # (1  , 2048)
+
+
+if __name__ == "__main__":
+    model = WaveUNet(ch_out=1, num_layers=9)
+    inp = torch.rand(2, 1, 2048)
+    out = model(inp)
+    print(model)
+    print(model.count_parameters())
+    print(out[0].shape)
diff --git a/src/gyraudio/audio_separation/data/__init__.py b/src/gyraudio/audio_separation/data/__init__.py
new file mode 100644
index 0000000000000000000000000000000000000000..1d20b74f53e8b855d1d4cebf2b8a05d3c0496938
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/__init__.py
@@ -0,0 +1,5 @@
+from gyraudio.audio_separation.data.mixed import MixedAudioDataset
+from gyraudio.audio_separation.data.remixed_fixed import RemixedFixedAudioDataset
+from gyraudio.audio_separation.data.remixed_rnd import RemixedRandomAudioDataset
+from gyraudio.audio_separation.data.single import SingleAudioDataset
+from gyraudio.audio_separation.data.dataloader import get_dataloader, get_config_dataloader
diff --git a/src/gyraudio/audio_separation/data/dataloader.py b/src/gyraudio/audio_separation/data/dataloader.py
new file mode 100644
index 0000000000000000000000000000000000000000..861c33a55ad41675fa3ba7db0cfd9823672e1a18
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/dataloader.py
@@ -0,0 +1,47 @@
+from torch.utils.data import DataLoader
+from gyraudio.audio_separation.data.mixed import MixedAudioDataset
+from typing import Optional, List
+from gyraudio.audio_separation.properties import (
+    DATA_PATH, AUGMENTATION, SNR_FILTER, SHUFFLE, BATCH_SIZE, TRAIN, VALID, TEST, AUG_TRIM
+)
+from gyraudio import root_dir
+RAW_AUDIO_ROOT = root_dir/"__data_source_separation"/"voice_origin"
+MIXED_AUDIO_ROOT = root_dir/"__data_source_separation"/"source_separation"
+
+
+def get_dataloader(configurations: dict, audio_dataset=MixedAudioDataset):
+    dataloaders = {}
+    for mode, configuration in configurations.items():
+        dataset = audio_dataset(
+            configuration[DATA_PATH],
+            augmentation_config=configuration[AUGMENTATION],
+            snr_filter=configuration[SNR_FILTER]
+        )
+        dl = DataLoader(
+            dataset,
+            shuffle=configuration[SHUFFLE],
+            batch_size=configuration[BATCH_SIZE],
+            collate_fn=dataset.collate_fn
+        )
+        dataloaders[mode] = dl
+    return dataloaders
+
+
+def get_config_dataloader(
+        audio_root=MIXED_AUDIO_ROOT,
+        mode: str = TRAIN,
+        shuffle: Optional[bool] = None,
+        batch_size: Optional[int] = 16,
+        snr_filter: Optional[List[float]] = None,
+        augmentation: dict = {}):
+    audio_folder = audio_root/mode
+    assert mode in [TRAIN, VALID, TEST]
+    assert audio_folder.exists()
+    config = {
+        DATA_PATH: audio_folder,
+        SHUFFLE: shuffle if shuffle is not None else (True if mode == TRAIN else False),
+        AUGMENTATION: augmentation,
+        SNR_FILTER: snr_filter,
+        BATCH_SIZE: batch_size
+    }
+    return config
diff --git a/src/gyraudio/audio_separation/data/dataset.py b/src/gyraudio/audio_separation/data/dataset.py
new file mode 100644
index 0000000000000000000000000000000000000000..904edbe33d2e712aad655426f98d5739f45a6407
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/dataset.py
@@ -0,0 +1,104 @@
+from torch.utils.data import Dataset
+from pathlib import Path
+from typing import Optional
+import torch
+from torch.utils.data import default_collate
+from typing import Tuple
+from functools import partial
+from gyraudio.audio_separation.properties import (
+    AUG_AWGN, AUG_RESCALE, AUG_TRIM, LENGTHS, LENGTH_DIVIDER, TRIM_PROB
+)
+
+
+class AudioDataset(Dataset):
+    def __init__(
+        self,
+        data_path: Path,
+        augmentation_config: dict = {},
+        snr_filter: Optional[float] = None,
+        debug: bool = False
+    ):
+        self.debug = debug
+        self.data_path = data_path
+        self.augmentation_config = augmentation_config
+        self.snr_filter = snr_filter
+        self.load_data()
+        self.length = len(self.file_list)
+        self.collate_fn = None
+        if AUG_TRIM in self.augmentation_config:
+            self.collate_fn = partial(collate_fn_generic,
+                                      lengths_lim=self.augmentation_config[AUG_TRIM][LENGTHS],
+                                      length_divider=self.augmentation_config[AUG_TRIM][LENGTH_DIVIDER],
+                                      trim_prob=self.augmentation_config[AUG_TRIM][TRIM_PROB])
+
+    def filter_data(self, snr):
+        if self.snr_filter is None:
+            return True
+        if snr in self.snr_filter:
+            return True
+        else:
+            return False
+
+    def load_data(self):
+        raise NotImplementedError("load_data method must be implemented")
+
+    def augment_data(self, mixed_audio_signal, clean_audio_signal, noise_audio_signal):
+        if AUG_RESCALE in self.augmentation_config:
+            current_amplitude = 0.5 + 1.5*torch.rand(1, device=mixed_audio_signal.device)
+            # logging.debug(current_amplitude)
+            mixed_audio_signal *= current_amplitude
+            noise_audio_signal *= current_amplitude
+            clean_audio_signal *= current_amplitude
+        if AUG_AWGN in self.augmentation_config:
+            # noise_std = self.augmentation_config[AUG_AWGN]["noise_std"]
+            noise_std = 0.01
+            current_noise_std = torch.randn(1) * noise_std
+            # logging.debug(current_noise_std)
+            extra_awgn = torch.randn(mixed_audio_signal.shape, device=mixed_audio_signal.device) * current_noise_std
+            mixed_audio_signal = mixed_audio_signal+extra_awgn
+            # Open question: should we add noise to the noise signal aswell?
+
+        return mixed_audio_signal, clean_audio_signal, noise_audio_signal
+
+    def __len__(self):
+        return self.length
+
+    def __getitem__(self, idx: int) -> torch.Tensor:
+        raise NotImplementedError("__getitem__ method must be implemented")
+
+
+def collate_fn_generic(batch, lengths_lim, length_divider=1024, trim_prob=0.5) -> Tuple[torch.Tensor, torch.Tensor]:
+    """Collate function to allow trimming (=crop the time dimension) of the signals in a batch.
+
+    Args:
+    batch (list): A list of tuples (triplets), where each tuple contain:
+    - mixed_audio_signal
+    - clean_audio_signal
+    - noise_audio_signal
+    lengths_lim (list) : A list of containing a minimum length (0) and a maximum length (1)
+    length_divider (int) : has to be a trimmed length divider
+    trim_prob (float) : trimming probability
+
+    Returns:
+    - Tensor: A batch of mixed_audio_signal, trimmed to the same length.
+    - Tensor: A batch of clean_audio_signal
+    - Tensor: A batch of noise_audio_signal
+    """
+
+    # Find the length of the shortest signal in the batch
+    mixed_audio_signal, clean_audio_signal, noise_audio_signal = default_collate(batch)
+    length = mixed_audio_signal[0].shape[-1]
+    min_length, max_length = lengths_lim
+    take_full_signal = torch.rand(1) > trim_prob
+    if not take_full_signal:
+        start = torch.randint(0, length-min_length, (1,))
+        trim_length = torch.randint(min_length, min(max_length, length-start-1)+1, (1,))
+        trim_length = trim_length-trim_length % length_divider
+        end = start + trim_length
+    else:
+        start = 0
+        end = length - length % length_divider
+    mixed_audio_signal = mixed_audio_signal[..., start:end]
+    clean_audio_signal = clean_audio_signal[..., start:end]
+    noise_audio_signal = noise_audio_signal[..., start:end]
+    return mixed_audio_signal, clean_audio_signal, noise_audio_signal
diff --git a/src/gyraudio/audio_separation/data/mixed.py b/src/gyraudio/audio_separation/data/mixed.py
new file mode 100644
index 0000000000000000000000000000000000000000..84e8b2c391da533b25d60e31a2330626d1505040
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/mixed.py
@@ -0,0 +1,40 @@
+from gyraudio.audio_separation.data.dataset import AudioDataset
+import logging
+import torch
+import torchaudio
+from typing import Tuple
+
+
+class MixedAudioDataset(AudioDataset):
+    def load_data(self):
+        self.folder_list = sorted(list(self.data_path.iterdir()))
+        self.file_list = [
+            [
+                list(folder.glob("mix*.wav"))[0],
+                folder/"voice.wav",
+                folder/"noise.wav"
+            ] for folder in self.folder_list
+        ]
+        snr_list = [float(file[0].stem.split("_")[-1]) for file in self.file_list]
+        self.file_list = [files for snr, files in zip(snr_list, self.file_list) if self.filter_data(snr)]
+        if self.debug:
+            logging.info(f"Available SNR {set(snr_list)}")
+            print(f"Available SNR {set(snr_list)}")
+            print("Filtered", len(self.file_list), self.snr_filter)
+        self.sampling_rate = None
+
+    def __getitem__(self, idx: int) -> Tuple[torch.Tensor, torch.Tensor, torch.Tensor]:
+        mixed_audio_path, signal_path, noise_path = self.file_list[idx]
+        assert mixed_audio_path.exists()
+        assert signal_path.exists()
+        assert noise_path.exists()
+        mixed_audio_signal, sampling_rate = torchaudio.load(str(mixed_audio_path))
+        clean_audio_signal, sampling_rate = torchaudio.load(str(signal_path))
+        noise_audio_signal, sampling_rate = torchaudio.load(str(noise_path))
+        self.sampling_rate = sampling_rate
+        mixed_audio_signal, clean_audio_signal, noise_audio_signal = self.augment_data(mixed_audio_signal, clean_audio_signal, noise_audio_signal)
+        if self.debug:
+            logging.debug(f"{mixed_audio_signal.shape}")
+            logging.debug(f"{clean_audio_signal.shape}")
+            logging.debug(f"{noise_audio_signal.shape}")
+        return mixed_audio_signal, clean_audio_signal, noise_audio_signal
diff --git a/src/gyraudio/audio_separation/data/remixed.py b/src/gyraudio/audio_separation/data/remixed.py
new file mode 100644
index 0000000000000000000000000000000000000000..8ba066e8a04c1ad0d1c7c1827c2b48adb4be585e
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/remixed.py
@@ -0,0 +1,53 @@
+from gyraudio.audio_separation.data.dataset import AudioDataset
+from typing import Tuple
+import logging
+from torch import Tensor
+import torch
+import torchaudio
+
+
+class RemixedAudioDataset(AudioDataset):
+    def generate_snr_list(self):
+        self.snr_list = None
+
+    def load_data(self):
+        self.folder_list = sorted(list(self.data_path.iterdir()))
+        self.file_list = [
+            [
+                folder/"voice.wav",
+                folder/"noise.wav"
+            ] for folder in self.folder_list
+        ]
+        self.sampling_rate = None
+        self.min_snr, self.max_snr = -4, 4
+        self.generate_snr_list()
+        if self.debug:
+            print("Not filtered", len(self.file_list), self.snr_filter)
+            print(self.snr_list)
+
+    def get_idx_noise(self, idx):
+        raise NotImplementedError("get_idx_noise method must be implemented")
+
+    def get_snr(self, idx):
+        raise NotImplementedError("get_snr method must be implemented")
+
+    def __getitem__(self, idx: int) -> Tuple[Tensor, Tensor, Tensor]:
+        signal_path = self.file_list[idx][0]
+        idx_noise = self.get_idx_noise(idx)
+        noise_path = self.file_list[idx_noise][1]
+
+        assert signal_path.exists()
+        assert noise_path.exists()
+        clean_audio_signal, sampling_rate = torchaudio.load(str(signal_path))
+        noise_audio_signal, sampling_rate = torchaudio.load(str(noise_path))
+        snr = self.get_snr(idx)
+        alpha = 10 ** (-snr / 20) * torch.norm(clean_audio_signal) / torch.norm(noise_audio_signal)
+        mixed_audio_signal = clean_audio_signal + alpha*noise_audio_signal
+        self.sampling_rate = sampling_rate
+        mixed_audio_signal, clean_audio_signal, noise_audio_signal = self.augment_data(
+            mixed_audio_signal, clean_audio_signal, noise_audio_signal)
+        if self.debug:
+            logging.debug(f"{mixed_audio_signal.shape}")
+            logging.debug(f"{clean_audio_signal.shape}")
+            logging.debug(f"{noise_audio_signal.shape}")
+        return mixed_audio_signal, clean_audio_signal, noise_audio_signal
diff --git a/src/gyraudio/audio_separation/data/remixed_fixed.py b/src/gyraudio/audio_separation/data/remixed_fixed.py
new file mode 100644
index 0000000000000000000000000000000000000000..78a7d5232053e22cfb8806d34088957029ea0fa2
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/remixed_fixed.py
@@ -0,0 +1,18 @@
+from gyraudio.audio_separation.data.remixed import RemixedAudioDataset
+import torch
+
+class RemixedFixedAudioDataset(RemixedAudioDataset):
+    def generate_snr_list(self) :
+        rnd_gen = torch.Generator()
+        rnd_gen.manual_seed(2147483647)
+        if self.snr_filter is None :
+            self.snr_list = self.min_snr + (self.max_snr - self.min_snr)*torch.rand(len(self.file_list), generator = rnd_gen)
+        else :
+            indices = torch.randint(0, len(self.snr_filter), (len(self.file_list),), generator=rnd_gen)
+            self.snr_list = [self.snr_filter[idx] for idx in indices]
+
+    def get_idx_noise(self, idx) :
+        return idx
+
+    def get_snr(self, idx) :
+        return self.snr_list[idx]
\ No newline at end of file
diff --git a/src/gyraudio/audio_separation/data/remixed_rnd.py b/src/gyraudio/audio_separation/data/remixed_rnd.py
new file mode 100644
index 0000000000000000000000000000000000000000..3a9cb822e62a2d7279f559a44af61eeb94625da8
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/remixed_rnd.py
@@ -0,0 +1,12 @@
+from gyraudio.audio_separation.data.remixed import RemixedAudioDataset
+from torch import rand, randint
+
+
+class RemixedRandomAudioDataset(RemixedAudioDataset):
+    def get_idx_noise(self, idx):
+        return randint(0, len(self.file_list)-1, (1,))
+
+    def get_snr(self, idx):
+        if self.snr_filter is None:
+            return self.min_snr + (self.max_snr - self.min_snr)*rand(1)
+        return self.snr_filter[randint(0, len(self.snr_filter), (1,))]
diff --git a/src/gyraudio/audio_separation/data/silence_detector.py b/src/gyraudio/audio_separation/data/silence_detector.py
new file mode 100644
index 0000000000000000000000000000000000000000..e69d35a952e9ca4ba7718eaa987e725675078815
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/silence_detector.py
@@ -0,0 +1,55 @@
+import torch
+import matplotlib.pyplot as plt
+import numpy as np
+
+
+def get_silence_mask(
+        sig: torch.Tensor, morph_kernel_size: int = 499, k_smooth=21, thresh=0.0001,
+        debug: bool = False) -> torch.Tensor:
+    with torch.no_grad():
+        smooth = torch.nn.Conv1d(1, 1, k_smooth, padding=k_smooth//2, bias=False).to(sig.device)
+        smooth.weight.data.fill_(1./k_smooth)
+        smoothed = smooth(torch.abs(sig))
+        st = 1.*(torch.abs(smoothed) < thresh*torch.ones_like(smoothed, device=sig.device))
+        sig_dil = torch.nn.MaxPool1d(morph_kernel_size, stride=1, padding=morph_kernel_size//2)(st)
+        sig_ero = -torch.nn.MaxPool1d(morph_kernel_size, stride=1, padding=morph_kernel_size//2)(-sig_dil)
+    if debug:
+        return sig_ero.squeeze(0), smoothed.squeeze(0), st.squeeze(0)
+    else:
+        return sig_ero
+
+
+def visualize_silence_mask(sig: torch.Tensor, silence_thresh: float = 0.0001):
+    silence_thresh = 0.0001
+    silence_mask, smoothed_amplitude, _ = get_silence_mask(
+        sig, k_smooth=21, morph_kernel_size=499, thresh=silence_thresh, debug=True
+    )
+    plt.figure(figsize=(12, 4))
+    plt.subplot(121)
+    plt.plot(sig.squeeze(0).cpu().numpy(), "k-", label="voice", alpha=0.5)
+    plt.plot(0.01*silence_mask.cpu().numpy(), "r-", alpha=1., label="silence mask")
+    plt.grid()
+    plt.legend()
+    plt.title("Voice and silence mask")
+    plt.ylim(-0.04, 0.04)
+
+    plt.subplot(122)
+    plt.plot(smoothed_amplitude.cpu().numpy(), "g--", alpha=0.5, label="smoothed amplitude")
+    plt.plot(np.ones(silence_mask.shape[-1])*silence_thresh, "c--", alpha=1., label="threshold")
+    plt.plot(-silence_thresh+silence_thresh*silence_mask.cpu().numpy(), "r-", alpha=1, label="silence mask")
+    plt.grid()
+    plt.legend()
+    plt.title("Thresholding mechanism")
+    plt.ylim(-silence_thresh, silence_thresh*10)
+    plt.show()
+
+
+if __name__ == "__main__":
+    from gyraudio.default_locations import SAMPLE_ROOT
+    from gyraudio.audio_separation.visualization.pre_load_audio import audio_loading
+    from gyraudio.audio_separation.properties import CLEAN, BUFFERS
+    sample_folder = SAMPLE_ROOT/"0009"
+    signals = audio_loading(sample_folder, preload=True)
+    device = "cuda" if torch.cuda.is_available() else "cpu"
+    sig_in = signals[BUFFERS][CLEAN].to(device)
+    visualize_silence_mask(sig_in)
diff --git a/src/gyraudio/audio_separation/data/single.py b/src/gyraudio/audio_separation/data/single.py
new file mode 100644
index 0000000000000000000000000000000000000000..307c289c40ae4994996e029370f87ea5fbdcf511
--- /dev/null
+++ b/src/gyraudio/audio_separation/data/single.py
@@ -0,0 +1,15 @@
+from gyraudio.audio_separation.data.dataset import AudioDataset
+import logging
+import torchaudio
+
+
+class SingleAudioDataset(AudioDataset):
+    def load_data(self):
+        self.file_list = sorted(list(self.data_path.glob("*.wav")))
+
+    def __getitem__(self, idx: int):
+        audio_path = self.file_list[idx]
+        assert audio_path.exists()
+        audio_signal, sampling_rate = torchaudio.load(str(audio_path))
+        logging.debug(f"{audio_signal.shape}")
+        return audio_signal
diff --git a/src/gyraudio/audio_separation/experiment_tracking/experiments.py b/src/gyraudio/audio_separation/experiment_tracking/experiments.py
new file mode 100644
index 0000000000000000000000000000000000000000..b11220c8dd7109aadc65daf7d2de3f4f6524d7c2
--- /dev/null
+++ b/src/gyraudio/audio_separation/experiment_tracking/experiments.py
@@ -0,0 +1,122 @@
+from gyraudio.default_locations import MIXED_AUDIO_ROOT
+from gyraudio.audio_separation.properties import (
+    TRAIN, TEST, VALID, NAME, EPOCHS, LEARNING_RATE,
+    OPTIMIZER, BATCH_SIZE, DATALOADER, AUGMENTATION,
+    SHORT_NAME, AUG_TRIM, TRIM_PROB, LENGTH_DIVIDER, LENGTHS, SNR_FILTER
+)
+from gyraudio.audio_separation.data.remixed_fixed import RemixedFixedAudioDataset
+from gyraudio.audio_separation.data.remixed_rnd import RemixedRandomAudioDataset
+from gyraudio.audio_separation.data import get_dataloader, get_config_dataloader
+from gyraudio.audio_separation.experiment_tracking.experiments_definition import get_experiment_generator
+import torch
+from typing import Tuple
+
+
+def get_experience(exp_major: int, exp_minor: int = 0, snr_filter_test=None, dry_run=False) -> Tuple[str, torch.nn.Module, dict, dict]:
+    """Get all experience details
+
+    Args:
+        exp_major (int): Major experience number
+        exp_minor (int, optional): Used for HP search. Defaults to 0.
+
+
+    Returns:
+        Tuple[str, torch.nn.Module, dict, dict]: short_name, model, config, dataloaders
+    """
+    model = None
+    config = {}
+    dataloader_name = "remix"
+    config = {
+        NAME: None,
+        OPTIMIZER: {
+            NAME: "adam",
+            LEARNING_RATE: 0.001
+        },
+        EPOCHS: 60,
+        DATALOADER: {
+            NAME: dataloader_name,
+        },
+        BATCH_SIZE: [16, 16, 16],
+        SNR_FILTER : snr_filter_test
+    }
+
+    model, config = get_experiment_generator(exp_major=exp_major)(config, no_model=dry_run, minor=exp_minor)
+    # POST PROCESSING
+    if isinstance(config[BATCH_SIZE], list) or isinstance(config[BATCH_SIZE], tuple):
+        config[BATCH_SIZE] = {
+            TRAIN: config[BATCH_SIZE][0],
+            TEST:  config[BATCH_SIZE][1],
+            VALID:  config[BATCH_SIZE][2],
+        }
+
+    if config[DATALOADER][NAME] == "premix":
+        mixed_audio_root = MIXED_AUDIO_ROOT
+        dataloaders = get_dataloader({
+            TRAIN: get_config_dataloader(
+                audio_root=mixed_audio_root,
+                mode=TRAIN,
+                shuffle=True,
+                batch_size=config[BATCH_SIZE][TRAIN],
+                augmentation=config[DATALOADER].get(AUGMENTATION, {})
+            ),
+            TEST: get_config_dataloader(
+                audio_root=mixed_audio_root,
+                mode=TEST,
+                shuffle=False,
+                batch_size=config[BATCH_SIZE][TEST],
+                snr_filter=config[SNR_FILTER]
+            )
+        })
+    elif config[DATALOADER][NAME] == "remix":
+        mixed_audio_root = MIXED_AUDIO_ROOT
+        aug_test = {}
+        if AUG_TRIM in config[DATALOADER].get(AUGMENTATION, {}):
+            aug_test = {
+                AUG_TRIM: {LENGTHS: [None, None], LENGTH_DIVIDER: config[DATALOADER][AUGMENTATION]
+                           [AUG_TRIM][LENGTH_DIVIDER], TRIM_PROB: -1.}
+            }
+        dl_train = get_dataloader(
+            {
+                TRAIN: get_config_dataloader(
+                    audio_root=mixed_audio_root,
+                    mode=TRAIN,
+                    shuffle=True,
+                    batch_size=config[BATCH_SIZE][TRAIN],
+                    augmentation=config[DATALOADER].get(AUGMENTATION, {})
+                )
+            },
+            audio_dataset=RemixedRandomAudioDataset
+        )[TRAIN]
+        dl_test = get_dataloader(
+            {
+                TEST: get_config_dataloader(
+                    audio_root=mixed_audio_root,
+                    mode=TEST,
+                    shuffle=False,
+                    batch_size=config[BATCH_SIZE][TEST]
+                )
+            },
+            audio_dataset=RemixedFixedAudioDataset
+        )[TEST]
+        dataloaders = {
+            TRAIN: dl_train,
+            TEST: dl_test
+        }
+    else:
+        raise NotImplementedError(f"Unknown dataloader {dataloader_name}")
+    assert config[NAME] is not None
+
+    short_name = f"{exp_major:04d}_{exp_minor:04d}"
+    config[SHORT_NAME] = short_name
+    return short_name, model, config, dataloaders
+
+
+if __name__ == "__main__":
+    from gyraudio.audio_separation.parser import shared_parser
+    parser_def = shared_parser()
+    args = parser_def.parse_args()
+
+    for exp in args.experiments:
+        short_name, model, config, dl = get_experience(exp)
+        print(short_name)
+        print(config)
diff --git a/src/gyraudio/audio_separation/experiment_tracking/experiments_decorator.py b/src/gyraudio/audio_separation/experiment_tracking/experiments_decorator.py
new file mode 100644
index 0000000000000000000000000000000000000000..82119f937b303f3795624e01b29ecf7dcee610b6
--- /dev/null
+++ b/src/gyraudio/audio_separation/experiment_tracking/experiments_decorator.py
@@ -0,0 +1,48 @@
+import torch
+from gyraudio.audio_separation.properties import (
+    NAME, ANNOTATIONS, NB_PARAMS, RECEPTIVE_FIELD
+)
+from typing import Optional
+REGISTERED_EXPERIMENTS_LIST = {}
+
+
+# def count_parameters(model: torch.nn.Module) -> int:
+#     """Count number of trainable parameters
+
+#     Args:
+#         model (torch.nn.Module): Pytorch model
+
+#     Returns:
+#         int: Number of trainable elements
+#     """
+#     return sum(p.numel() for p in model.parameters() if p.requires_grad)
+
+
+def registered_experiment(major: Optional[int] = None, failed: Optional[bool] = False) -> callable:
+    """Decorate and register an experiment
+    - Register the experiment in the list of experiments
+    - Count the number of parameters and add it to the config
+
+    Args:
+        major (Optional[int], optional): major id version = Number of the experiment. Defaults to None.
+        failed (Optional[bool], optional): If an experiment failed,
+        keep track of it but prevent from evaluating. Defaults to False.
+
+    Returns:
+        callable: decorator function
+    """
+    def decorator(func):
+        assert (major) not in REGISTERED_EXPERIMENTS_LIST, f"Experiment {major} already registered"
+
+        def wrapper(config, minor=None, no_model=False, model=torch.nn.Module()):
+            config, model = func(config, model=None if not no_model else model, minor=minor)
+            config[NB_PARAMS] = model.count_parameters()
+            config[RECEPTIVE_FIELD] = model.receptive_field()
+            assert NAME in config, "NAME not defined"
+            assert ANNOTATIONS in config, "ANNOTATIONS not defined"
+            return model, config
+        if not failed:
+            REGISTERED_EXPERIMENTS_LIST[major] = wrapper
+        return wrapper
+
+    return decorator
diff --git a/src/gyraudio/audio_separation/experiment_tracking/experiments_definition.py b/src/gyraudio/audio_separation/experiment_tracking/experiments_definition.py
new file mode 100644
index 0000000000000000000000000000000000000000..19afb4dc7e36aa9795a98121eff431a8b9caa54b
--- /dev/null
+++ b/src/gyraudio/audio_separation/experiment_tracking/experiments_definition.py
@@ -0,0 +1,320 @@
+from gyraudio.audio_separation.architecture.flat_conv import FlatConvolutional
+from gyraudio.audio_separation.architecture.unet import ResUNet
+from gyraudio.audio_separation.architecture.wave_unet import WaveUNet
+from gyraudio.audio_separation.architecture.neutral import NeutralModel
+from gyraudio.audio_separation.architecture.transformer import TransformerModel
+from gyraudio.audio_separation.properties import (
+    NAME, ANNOTATIONS, MAX_STEPS_PER_EPOCH, EPOCHS, BATCH_SIZE,
+    OPTIMIZER, LEARNING_RATE,
+    DATALOADER,
+    WEIGHT_DECAY,
+    LOSS, LOSS_L1,
+    AUGMENTATION, AUG_TRIM, AUG_AWGN, AUG_RESCALE,
+    LENGTHS, LENGTH_DIVIDER, TRIM_PROB,
+    SCHEDULER, SCHEDULER_CONFIGURATION
+)
+from gyraudio.audio_separation.experiment_tracking.experiments_decorator import (
+    registered_experiment, REGISTERED_EXPERIMENTS_LIST
+)
+
+
+@registered_experiment(major=9999)
+def neutral(config, model: bool = None, minor=None):
+    config[BATCH_SIZE] = [4, 4, 4]
+    config[EPOCHS] = 1
+    config[NAME] = "Neutral"
+    config[ANNOTATIONS] = "Neutral"
+    if model is None:
+        model = NeutralModel()
+    config[NAME] = "Neutral"
+    return config, model
+
+
+@registered_experiment(major=0)
+def exp_unit_test(config, model: bool = None, minor=None):
+    config[MAX_STEPS_PER_EPOCH] = 2
+    config[BATCH_SIZE] = [4, 4, 4]
+    config[EPOCHS] = 2
+    config[NAME] = "Unit Test - Flat Convolutional"
+    config[ANNOTATIONS] = "Baseline"
+    config[SCHEDULER] = "ReduceLROnPlateau"
+    config[SCHEDULER_CONFIGURATION] = dict(patience=5, factor=0.8)
+    if model is None:
+        model = FlatConvolutional()
+    return config, model
+
+# ---------------- Low Baseline -----------------
+
+
+def exp_low_baseline(
+    config: dict,
+    batch_size: int = 16,
+    h_dim: int = 16,
+    k_size: int = 9,
+    dilation: int = 0,
+    model: bool = None,
+    minor=None
+):
+    config[BATCH_SIZE] = [batch_size, batch_size, batch_size]
+    config[NAME] = "Flat Convolutional"
+    config[ANNOTATIONS] = f"Baseline H={h_dim}_K={k_size}"
+    if dilation > 1:
+        config[ANNOTATIONS] += f"_dil={dilation}"
+    config["Architecture"] = {
+        "name": "Flat-Conv",
+        "h_dim": h_dim,
+        "scales": 1,
+        "k_size": k_size,
+        "dilation": dilation
+    }
+    if model is None:
+        model = FlatConvolutional(k_size=k_size, h_dim=h_dim)
+    return config, model
+
+
+@registered_experiment(major=1)
+def exp_1(config, model: bool = None, minor=None):
+    config, model = exp_low_baseline(config, batch_size=32, k_size=5)
+    return config, model
+
+
+@registered_experiment(major=2)
+def exp_2(config, model: bool = None, minor=None):
+    config, model = exp_low_baseline(config, batch_size=32, k_size=9)
+    return config, model
+
+
+@registered_experiment(major=3)
+def exp_3(config, model: bool = None, minor=None):
+    config, model = exp_low_baseline(config, batch_size=32, k_size=9, dilation=2)
+    return config, model
+
+
+@registered_experiment(major=4)
+def exp_4(config, model: bool = None, minor=None):
+    config, model = exp_low_baseline(config, batch_size=16, k_size=9)
+    return config, model
+
+# ------------------ Res U-Net ------------------
+
+
+def exp_resunet(config, h_dim=16, k_size=5, model=None):
+    config[NAME] = "Res-UNet"
+    scales = 4
+    config[ANNOTATIONS] = f"Res-UNet-{scales}scales_h={h_dim}_k={k_size}"
+    config["Architecture"] = {
+        "name": "Res-UNet",
+        "h_dim": h_dim,
+        "scales": scales,
+        "k_size": k_size,
+    }
+    if model is None:
+        model = ResUNet(h_dim=h_dim, k_size=k_size)
+    return config, model
+
+
+@registered_experiment(major=2000)
+def exp_2000_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 60
+    config, model = exp_resunet(config)
+    return config, model
+
+
+@registered_experiment(major=2001)
+def exp_2001_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 60
+    config, model = exp_resunet(config, h_dim=32, k_size=5)
+    return config, model
+
+# ------------------ Wave U-Net ------------------
+
+
+def exp_wave_unet(config: dict,
+                  channels_extension: int = 24,
+                  k_conv_ds: int = 15,
+                  k_conv_us: int = 5,
+                  num_layers: int = 4,
+                  dropout: float = 0.0,
+                  bias: bool = True,
+                  model=None):
+    config[NAME] = "Wave-UNet"
+    config[ANNOTATIONS] = f"Wave-UNet-{num_layers}scales_h_ext={channels_extension}_k={k_conv_ds}ds-{k_conv_us}us"
+    if dropout > 0:
+        config[ANNOTATIONS] += f"-dr{dropout:.1e}"
+    if not bias:
+        config[ANNOTATIONS] += "-BiasFree"
+    config["Architecture"] = {
+        "k_conv_us": k_conv_us,
+        "k_conv_ds": k_conv_ds,
+        "num_layers": num_layers,
+        "channels_extension": channels_extension,
+        "dropout": dropout,
+        "bias": bias
+    }
+    if model is None:
+        model = WaveUNet(
+            **config["Architecture"]
+        )
+    config["Architecture"][NAME] = "Wave-UNet"
+    return config, model
+
+
+@registered_experiment(major=1000)
+def exp_1000_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 60
+    config, model = exp_wave_unet(config, model=model, num_layers=4, channels_extension=24)
+    # 4 layers, ext +24 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1001)
+def exp_1001_waveunet(config, model: bool = None, minor=None):
+    # OVERFIT 1M param ?
+    config[EPOCHS] = 60
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=16)
+    # 7 layers, ext +16 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1002)
+def exp_1002_waveunet(config, model: bool = None, minor=None):
+    # OVERFIT 1M param ?
+    config[EPOCHS] = 60
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=16)
+    config[DATALOADER][AUGMENTATION] = {
+        AUG_TRIM: {LENGTHS: [8192, 80000], LENGTH_DIVIDER: 1024, TRIM_PROB: 0.8},
+        AUG_RESCALE: True
+    }
+    # 7 layers, ext +16 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1003)
+def exp_1003_waveunet(config, model: bool = None, minor=None):
+    # OVERFIT 2.3M params
+    config[EPOCHS] = 60
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=24)
+    # 7 layers, ext +24 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1004)
+def exp_1004_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 120
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=28)
+    # 7 layers, ext +28 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1014)
+def exp_1014_waveunet(config, model: bool = None, minor=None):
+    # trained with min and max mixing snr hard coded between -2 and -1
+    config[EPOCHS] = 50
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=28)
+    # 7 layers, ext +28 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1005)
+def exp_1005_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 150
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=16)
+    config[DATALOADER][AUGMENTATION] = {
+        AUG_TRIM: {LENGTHS: [8192, 80000], LENGTH_DIVIDER: 1024, TRIM_PROB: 0.8},
+        AUG_RESCALE: True
+    }
+    # 7 layers, ext +16 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1006)
+def exp_1006_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 150
+    config, model = exp_wave_unet(config, model=model, num_layers=11, channels_extension=16)
+    config[DATALOADER][AUGMENTATION] = {
+        AUG_TRIM: {LENGTHS: [8192, 80000], LENGTH_DIVIDER: 4096, TRIM_PROB: 0.8},
+        AUG_RESCALE: True
+    }
+    # 11 layers, ext +16 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1007)
+def exp_1007_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 150
+    config, model = exp_wave_unet(config, model=model, num_layers=9, channels_extension=16)
+    config[DATALOADER][AUGMENTATION] = {
+        AUG_TRIM: {LENGTHS: [8192, 80000], LENGTH_DIVIDER: 4096, TRIM_PROB: 0.8},
+        AUG_RESCALE: True
+    }
+    # 11 layers, ext +16 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=1008)
+def exp_1008_waveunet(config, model: bool = None, minor=None):
+    # CHEAP BASELINE
+    config[EPOCHS] = 150
+    config, model = exp_wave_unet(config, model=model, num_layers=4, channels_extension=16)
+    config[DATALOADER][AUGMENTATION] = {
+        AUG_TRIM: {LENGTHS: [8192, 80000], LENGTH_DIVIDER: 1024, TRIM_PROB: 0.8},
+        AUG_RESCALE: True
+    }
+    # 4 layers, ext +16 - Nvidia T500 4Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=3000)
+def exp_3000_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 120
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=28, bias=False)
+    # 7 layers, ext +28 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    return config, model
+
+
+@registered_experiment(major=3001)
+def exp_3001_waveunet(config, model: bool = None, minor=None):
+    config[EPOCHS] = 200
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=28, bias=False)
+    # 7 layers, ext +28 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    config[SCHEDULER] = "ReduceLROnPlateau"
+    config[SCHEDULER_CONFIGURATION] = dict(patience=5, factor=0.8)
+    config[OPTIMIZER][LEARNING_RATE] = 0.002
+    return config, model
+
+
+@registered_experiment(major=3002)
+def exp_3002_waveunet(config, model: bool = None, minor=None):
+    # TRAINED WITH SNR -12db +12db (code changed manually!)
+    # See f910c6da3123e3d35cc0ce588bb5a72ce4a8c422
+    config[EPOCHS] = 200
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=28, bias=False)
+    # 7 layers, ext +28 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    config[SCHEDULER] = "ReduceLROnPlateau"
+    config[SCHEDULER_CONFIGURATION] = dict(patience=5, factor=0.8)
+    config[OPTIMIZER][LEARNING_RATE] = 0.002
+    return config, model
+
+
+@registered_experiment(major=4000)
+def exp_4000_bias_free_waveunet_l1(config, model: bool = None, minor=None):
+    # config[MAX_STEPS_PER_EPOCH] = 2
+    # config[BATCH_SIZE] = [2, 2, 2]
+    config[EPOCHS] = 200
+    config[LOSS] = LOSS_L1
+    config, model = exp_wave_unet(config, model=model, num_layers=7, channels_extension=28, bias=False)
+    # 7 layers, ext +28 - Nvidia RTX3060 6Gb RAM - 16 batch size
+    config[SCHEDULER] = "ReduceLROnPlateau"
+    config[SCHEDULER_CONFIGURATION] = dict(patience=5, factor=0.8)
+    config[OPTIMIZER][LEARNING_RATE] = 0.002
+    return config, model
+
+
+def get_experiment_generator(exp_major: int):
+    assert exp_major in REGISTERED_EXPERIMENTS_LIST, f"Experiment {exp_major} not registered"
+    exp_generator = REGISTERED_EXPERIMENTS_LIST[exp_major]
+    return exp_generator
+
+
+if __name__ == "__main__":
+    print(f"Available experiments: {list(REGISTERED_EXPERIMENTS_LIST.keys())}")
diff --git a/src/gyraudio/audio_separation/experiment_tracking/storage.py b/src/gyraudio/audio_separation/experiment_tracking/storage.py
new file mode 100644
index 0000000000000000000000000000000000000000..08439660827ff10d30b037851a2fade0aefe968c
--- /dev/null
+++ b/src/gyraudio/audio_separation/experiment_tracking/storage.py
@@ -0,0 +1,65 @@
+from gyraudio.audio_separation.properties import SHORT_NAME, MODEL, OPTIMIZER, CURRENT_EPOCH, CONFIGURATION
+from pathlib import Path
+from gyraudio.default_locations import EXPERIMENT_STORAGE_ROOT
+import logging
+import torch
+
+
+def get_output_folder(config: dict, root_dir: Path = EXPERIMENT_STORAGE_ROOT, override: bool = False) -> Path:
+    output_folder = root_dir/config["short_name"]
+    exists = False
+    if output_folder.exists():
+        if not override:
+            logging.info(f"Experiment {config[SHORT_NAME]} already exists. Override is set to False. Skipping.")
+        if override:
+            logging.warning(f"Experiment {config[SHORT_NAME]} will be OVERRIDDEN")
+            exists = True
+    else:
+        output_folder.mkdir(parents=True, exist_ok=True)
+        exists = True
+    return exists, output_folder
+
+
+def checkpoint_paths(exp_dir: Path, epoch=None):
+    if epoch is None:
+        checkpoints = sorted(exp_dir.glob("model_*.pt"))
+        assert len(checkpoints) > 0, f"No checkpoints found in {exp_dir}"
+        model_checkpoint = checkpoints[-1]
+        epoch = int(model_checkpoint.stem.split("_")[-1])
+        optimizer_checkpoint = exp_dir/model_checkpoint.stem.replace("model", "optimizer")
+    else:
+        model_checkpoint = exp_dir/f"model_{epoch:04d}.pt"
+        optimizer_checkpoint = exp_dir/f"optimizer_{epoch:04d}.pt"
+    return model_checkpoint, optimizer_checkpoint, epoch
+
+
+def load_checkpoint(model, exp_dir: Path, optimizer=None, epoch: int = None,
+                    device="cuda" if torch.cuda.is_available() else "cpu"):
+    config = {}
+    model_checkpoint, optimizer_checkpoint, epoch = checkpoint_paths(exp_dir, epoch=epoch)
+    model_state_dict = torch.load(model_checkpoint, map_location=torch.device(device))
+    model.load_state_dict(model_state_dict[MODEL])
+    if optimizer is not None:
+        optimizer_state_dict = torch.load(optimizer_checkpoint, map_location=torch.device(device))
+        optimizer.load_state_dict(optimizer_state_dict[OPTIMIZER])
+        config = optimizer_state_dict[CONFIGURATION]
+    return model, optimizer, epoch, config
+
+
+def save_checkpoint(model, exp_dir: Path, optimizer=None, config: dict = {}, epoch: int = None):
+    model_checkpoint, optimizer_checkpoint, epoch = checkpoint_paths(exp_dir, epoch=epoch)
+    torch.save(
+        {
+            MODEL: model.state_dict(),
+        },
+        model_checkpoint
+    )
+    torch.save(
+        {
+            CURRENT_EPOCH: epoch,
+            CONFIGURATION: config,
+            OPTIMIZER: optimizer.state_dict()
+        },
+        optimizer_checkpoint
+    )
+    print(f"Checkpoint saved:\n   - model: {model_checkpoint}\n   - checkpoint: {optimizer_checkpoint}")
diff --git a/src/gyraudio/audio_separation/infer.py b/src/gyraudio/audio_separation/infer.py
new file mode 100644
index 0000000000000000000000000000000000000000..1456afc9032b76d1dd7d286ace7ad49b528c6d7a
--- /dev/null
+++ b/src/gyraudio/audio_separation/infer.py
@@ -0,0 +1,202 @@
+from gyraudio.audio_separation.experiment_tracking.experiments import get_experience
+from gyraudio.audio_separation.parser import shared_parser
+from gyraudio.audio_separation.properties import TEST, NAME, SHORT_NAME, CURRENT_EPOCH, SNR_FILTER
+from gyraudio.default_locations import EXPERIMENT_STORAGE_ROOT
+from gyraudio.audio_separation.experiment_tracking.storage import load_checkpoint
+from gyraudio.audio_separation.experiment_tracking.storage import get_output_folder
+from gyraudio.audio_separation.metrics import snr
+from gyraudio.io.dump import Dump
+from pathlib import Path
+import sys
+import torch
+from tqdm import tqdm
+import torchaudio
+import pandas as pd
+from typing import List
+# Files paths
+DEFAULT_RECORD_FILE = "infer_record.csv"  # Store the characteristics of the inference record file
+DEFAULT_EVALUATION_FILE = "eval_df.csv"  # Store the characteristics of the inference record file
+# Record keys
+NBATCH = "nb_batch"
+BEST_SNR = "best_snr"
+BEST_SAVE_SNR = "best_save_snr"
+WORST_SNR = "worst_snr"
+WORST_SAVE_SNR = "worst_save_snr"
+RECORD_KEYS = [NAME, SHORT_NAME, CURRENT_EPOCH, NBATCH, SNR_FILTER, BEST_SAVE_SNR, BEST_SNR, WORST_SAVE_SNR, WORST_SNR]
+# Exaluation keys
+SAVE_IDX = "save_idx"
+SNR_IN = "snr_in"
+SNR_OUT = "snr_out"
+EVAL_KEYS = [SAVE_IDX, SNR_IN, SNR_OUT]
+
+
+def load_file(path: Path, keys: List[str]) -> pd.DataFrame:
+    if not (path.exists()):
+        df = pd.DataFrame(columns=keys)
+        df.to_csv(path)
+    return pd.read_csv(path)
+
+
+def launch_infer(exp: int, snr_filter: list = None, device: str = "cuda", model_dir: Path = None,
+                 output_dir: Path = EXPERIMENT_STORAGE_ROOT, force_reload=False, max_batches=None,
+                 ext=".wav"):
+    # Load experience
+    if snr_filter is not None:
+        snr_filter = sorted(snr_filter)
+    short_name, model, config, dl = get_experience(exp, snr_filter_test=snr_filter)
+    exists, exp_dir = get_output_folder(config, root_dir=model_dir, override=False)
+    assert exp_dir.exists(), f"Experiment {short_name} does not exist in {model_dir}"
+    model.eval()
+    model.to(device)
+    model, optimizer, epoch, config_checkpt = load_checkpoint(model, exp_dir, epoch=None, device=device)
+    # Folder creation
+    if output_dir is not None:
+        record_path = output_dir/DEFAULT_RECORD_FILE
+        record_df = load_file(record_path, RECORD_KEYS)
+
+        # Define conditions for filtering
+        exist_conditions = {
+            NAME: config[NAME],
+            SHORT_NAME: config[SHORT_NAME],
+            CURRENT_EPOCH: epoch,
+            NBATCH: max_batches,
+        }
+        # Create boolean masks and combine them
+        masks = [(record_df[key] == value) for key, value in exist_conditions.items()]
+        if snr_filter is None:
+            masks.append((record_df[SNR_FILTER]).isnull())
+        else:
+            masks.append(record_df[SNR_FILTER] == str(snr_filter))
+        combined_mask = pd.Series(True, index=record_df.index)
+        for mask in masks:
+            combined_mask = combined_mask & mask
+        filtered_df = record_df[combined_mask]
+
+        save_dir = output_dir/(exp_dir.name+"_infer" + (f"_epoch_{epoch:04d}_nbatch_{max_batches if max_batches is not None else len(dl[TEST])}")
+                               + ("" if snr_filter is None else f"_snrs_{'_'.join(map(str, snr_filter))}"))
+        evaluation_path = save_dir/DEFAULT_EVALUATION_FILE
+        if not (filtered_df.empty) and not (force_reload):
+            assert evaluation_path.exists()
+            print(f"Inference already exists, see folder {save_dir}")
+            record_row_df = filtered_df
+        else:
+            record_row_df = pd.DataFrame({
+                NAME: config[NAME],
+                SHORT_NAME: config[SHORT_NAME],
+                CURRENT_EPOCH: epoch,
+                NBATCH: max_batches,
+                SNR_FILTER: [None],
+            }, index=[0], columns=RECORD_KEYS)
+            record_row_df.at[0, SNR_FILTER] = snr_filter
+
+            save_dir.mkdir(parents=True, exist_ok=True)
+            evaluation_df = load_file(evaluation_path, EVAL_KEYS)
+            with torch.no_grad():
+                test_loss = 0.
+                save_idx = 0
+                best_snr = 0
+                worst_snr = 0
+                processed_batches = 0
+                for step_index, (batch_mix, batch_signal, batch_noise) in tqdm(
+                        enumerate(dl[TEST]), desc=f"Inference epoch {epoch}", total=max_batches if max_batches is not None else len(dl[TEST])):
+                    batch_mix, batch_signal, batch_noise = batch_mix.to(
+                        device), batch_signal.to(device), batch_noise.to(device)
+                    batch_output_signal, _batch_output_noise = model(batch_mix)
+                    loss = torch.nn.functional.mse_loss(batch_output_signal, batch_signal)
+                    test_loss += loss.item()
+
+                    # SNR stats
+                    snr_in = snr(batch_mix, batch_signal, reduce=None)
+                    snr_out = snr(batch_output_signal, batch_signal, reduce=None)
+                    best_current, best_idx_current = torch.max(snr_out-snr_in, axis=0)
+                    worst_current, worst_idx_current = torch.min(snr_out-snr_in, axis=0)
+                    if best_current > best_snr:
+                        best_snr = best_current
+                        best_save_idx = save_idx + best_idx_current
+                    if worst_current > worst_snr:
+                        worst_snr = worst_current
+                        worst_save_idx = save_idx + worst_idx_current
+
+                    # Save by signal
+                    batch_output_signal = batch_output_signal.detach().cpu()
+                    batch_signal = batch_signal.detach().cpu()
+                    batch_mix = batch_mix.detach().cpu()
+                    for audio_idx in range(batch_output_signal.shape[0]):
+                        dic = {SAVE_IDX: save_idx, SNR_IN: float(
+                            snr_in[audio_idx]), SNR_OUT: float(snr_out[audio_idx])}
+                        new_eval_row = pd.DataFrame(dic, index=[0])
+                        evaluation_df = pd.concat([new_eval_row, evaluation_df.loc[:]], ignore_index=True)
+
+                        # Save .wav
+                        torchaudio.save(
+                            str(save_dir/f"{save_idx:04d}_mixed{ext}"),
+                            batch_mix[audio_idx, :, :],
+                            sample_rate=dl[TEST].dataset.sampling_rate,
+                            channels_first=True
+                        )
+                        torchaudio.save(
+                            str(save_dir/f"{save_idx:04d}_out{ext}"),
+                            batch_output_signal[audio_idx, :, :],
+                            sample_rate=dl[TEST].dataset.sampling_rate,
+                            channels_first=True
+                        )
+                        torchaudio.save(
+                            str(save_dir/f"{save_idx:04d}_original{ext}"),
+                            batch_signal[audio_idx, :, :],
+                            sample_rate=dl[TEST].dataset.sampling_rate,
+                            channels_first=True
+                        )
+                        Dump.save_json(dic, save_dir/f"{save_idx:04d}.json")
+                        save_idx += 1
+                    processed_batches += 1
+                    if max_batches is not None and processed_batches >= max_batches:
+                        break
+            test_loss = test_loss/len(dl[TEST])
+            evaluation_df.to_csv(evaluation_path)
+
+            record_row_df[BEST_SAVE_SNR] = int(best_save_idx)
+            record_row_df[BEST_SNR] = float(best_snr)
+            record_row_df[WORST_SAVE_SNR] = int(worst_save_idx)
+            record_row_df[WORST_SNR] = float(worst_snr)
+            record_df = pd.concat([record_row_df, record_df.loc[:]], ignore_index=True)
+            record_df.to_csv(record_path, index=0)
+
+            print(f"Test loss: {test_loss:.3e}, \nbest snr performance: {best_save_idx} with {best_snr:.1f}dB, \nworst snr performance: {worst_save_idx} with {worst_snr:.1f}dB")
+
+    return record_row_df, evaluation_path
+
+
+def main(argv):
+    default_device = "cuda" if torch.cuda.is_available() else "cpu"
+    parser_def = shared_parser(help="Launch inference on a specific model"
+                               + ("\n<<<Cuda available>>>" if default_device == "cuda" else ""))
+    parser_def.add_argument("-i", "--input-dir", type=str, default=EXPERIMENT_STORAGE_ROOT)
+    parser_def.add_argument("-o", "--output-dir", type=str, default=EXPERIMENT_STORAGE_ROOT)
+    parser_def.add_argument("-d", "--device", type=str, default=default_device,
+                            help="Training device", choices=["cpu", "cuda"])
+    parser_def.add_argument("-r", "--reload", action="store_true",
+                            help="Force reload files")
+    parser_def.add_argument("-b", "--nb-batch", type=int, default=None,
+                            help="Number of batches to process")
+    parser_def.add_argument("-s",  "--snr-filter", type=float, nargs="+", default=None,
+                            help="SNR filters on the inference dataloader")
+    parser_def.add_argument("-ext", "--extension", type=str, default=".wav", help="Extension of the audio files to save",
+                            choices=[".wav", ".mp4"])
+    args = parser_def.parse_args(argv)
+    for exp in args.experiments:
+        launch_infer(
+            exp,
+            model_dir=Path(args.input_dir),
+            output_dir=Path(args.output_dir),
+            device=args.device,
+            force_reload=args.reload,
+            max_batches=args.nb_batch,
+            snr_filter=args.snr_filter,
+            ext=args.extension
+        )
+
+
+if __name__ == "__main__":
+    main(sys.argv[1:])
+
+# Example : python src\gyraudio\audio_separation\infer.py -i ./__output_audiosep -e 1002 -d cpu -b 2 -s 4 5 6
diff --git a/src/gyraudio/audio_separation/metrics.py b/src/gyraudio/audio_separation/metrics.py
new file mode 100644
index 0000000000000000000000000000000000000000..febb511003811418be8a30f1c389b3e19a79ed1b
--- /dev/null
+++ b/src/gyraudio/audio_separation/metrics.py
@@ -0,0 +1,101 @@
+from gyraudio.audio_separation.properties import SIGNAL, NOISE, TOTAL, LOSS_TYPE, COEFFICIENT, SNR
+import torch
+
+
+def snr(prediction: torch.Tensor, ground_truth: torch.Tensor, reduce="mean") -> torch.Tensor:
+    """Compute the SNR between two tensors.
+    Args:
+        prediction (torch.Tensor): prediction tensor
+        ground_truth (torch.Tensor): ground truth tensor
+    Returns:
+        torch.Tensor: SNR
+    """
+    power_signal = torch.sum(ground_truth**2, dim=(-2, -1))
+    power_error = torch.sum((prediction-ground_truth)**2, dim=(-2, -1))
+    eps = torch.finfo(torch.float32).eps
+    snr_per_element = 10*torch.log10((power_signal+eps)/(power_error+eps))
+    final_snr = torch.mean(snr_per_element) if reduce == "mean" else snr_per_element
+    return final_snr
+
+
+DEFAULT_COST = {
+    SIGNAL: {
+        COEFFICIENT: 0.5,
+        LOSS_TYPE: torch.nn.functional.mse_loss
+    },
+    NOISE: {
+        COEFFICIENT: 0.5,
+        LOSS_TYPE: torch.nn.functional.mse_loss
+    },
+    SNR: {
+        LOSS_TYPE: snr
+    }
+}
+
+
+class Costs:
+    """Keep track of cost functions.
+    ```
+    for epoch in range(...):
+        metric.reset_epoch()
+        for step in dataloader(...):
+            ... # forward
+            prediction = model.forward(batch)
+            metric.update(prediction1, groundtruth1, SIGNAL1)
+            metric.update(prediction2, groundtruth2, SIGNAL2)
+            loss = metric.finish_step()
+
+            loss.backward()
+            ... # backprop
+        metric.finish_epoch()
+        ... # log metrics
+    ```
+    """
+
+    def __init__(self, name: str, costs=DEFAULT_COST) -> None:
+        self.name = name
+        self.keys = list(costs.keys())
+        self.cost = costs
+
+    def __reset_step(self) -> None:
+        self.metrics = {key: 0. for key in self.keys}
+
+    def reset_epoch(self) -> None:
+        self.__reset_step()
+        self.total_metric = {key: 0. for key in self.keys+[TOTAL]}
+        self.count = 0
+
+    def update(self,
+               prediction: torch.Tensor,
+               ground_truth: torch.Tensor,
+               key: str
+               ) -> torch.Tensor:
+        assert key != TOTAL
+        # Compute loss for a single batch (=step)
+        loss_signal = self.cost[key][LOSS_TYPE](prediction, ground_truth)
+        self.metrics[key] = loss_signal
+
+    def finish_step(self) -> torch.Tensor:
+        # Reset current total
+        self.metrics[TOTAL] = 0.
+        # Sum all metrics to total
+        for key in self.metrics:
+            if key != TOTAL and self.cost[key].get(COEFFICIENT, False):
+                self.metrics[TOTAL] += self.cost[key][COEFFICIENT]*self.metrics[key]
+        loss_signal = self.metrics[TOTAL]
+        for key in self.metrics:
+            if not isinstance(self.metrics[key], float):
+                self.metrics[key] = self.metrics[key].item()
+            self.total_metric[key] += self.metrics[key]
+        self.count += 1
+        return loss_signal
+
+    def finish_epoch(self) -> None:
+        for key in self.metrics:
+            self.total_metric[key] /= self.count
+
+    def __repr__(self) -> str:
+        rep = f"{self.name}\t:\t"
+        for key in self.total_metric:
+            rep += f"{key}: {self.total_metric[key]:.3e} | "
+        return rep
diff --git a/src/gyraudio/audio_separation/parser.py b/src/gyraudio/audio_separation/parser.py
new file mode 100644
index 0000000000000000000000000000000000000000..85834acaee006a9879d9530b98e29bf2c5355693
--- /dev/null
+++ b/src/gyraudio/audio_separation/parser.py
@@ -0,0 +1,8 @@
+import argparse
+
+
+def shared_parser(help="Train models for audio separation"):
+    parser = argparse.ArgumentParser(description=help)
+    parser.add_argument("-e",  "--experiments", type=int, nargs="+", required=True,
+                        help="Experiment ids to be trained sequentially")
+    return parser
diff --git a/src/gyraudio/audio_separation/properties.py b/src/gyraudio/audio_separation/properties.py
new file mode 100644
index 0000000000000000000000000000000000000000..7704163f9af806dc9681f02c8e22988685cd62f9
--- /dev/null
+++ b/src/gyraudio/audio_separation/properties.py
@@ -0,0 +1,79 @@
+# Training modes (Train, Validation, Test)
+TRAIN = "train"
+VALID = "validation"
+TEST = "test"
+
+# Dataset properties (keys)
+DATA_PATH = "path"
+BATCH_SIZE = "batch_size"
+SHUFFLE = "shuffle"
+SNR_FILTER = "snr_filter"
+AUGMENTATION = "augmentation"
+DATALOADER = "dataloader"
+
+
+# Loss split
+SIGNAL = "signal"
+NOISE = "noise"
+TOTAL = "total"
+COEFFICIENT = "coefficient"
+
+
+# Augmentation types
+AUG_TRIM = "trim"  # trim batches to arbitrary length
+AUG_AWGN = "awgn"  # add white gaussian noise
+AUG_RESCALE = "rescale"  # rescale all signals arbitrarily
+
+# Trim types
+LENGTHS = "lengths"  # a list of min and max length
+LENGTH_DIVIDER = "length_divider"  # an int that divides the length
+TRIM_PROB = "trim_probability"  # a float in [0, 1] of trimming probability
+
+
+# Training configuration properties (keys)
+
+OPTIMIZER = "optimizer"
+LEARNING_RATE = "lr"
+WEIGHT_DECAY = "weight_decay"
+BETAS = "betas"
+EPOCHS = "epochs"
+BATCH_SIZE = "batch_size"
+MAX_STEPS_PER_EPOCH = "max_steps_per_epoch"
+
+
+# Properties for the model
+NAME = "name"
+ANNOTATIONS = "annotations"
+NB_PARAMS = "nb_params"
+RECEPTIVE_FIELD = "receptive_field"
+SHORT_NAME = "short_name"
+
+
+# Scheduler
+SCHEDULER = "scheduler"
+SCHEDULER_CONFIGURATION = "scheduler_configuration"
+
+# Loss
+LOSS = "loss"
+LOSS_L1 = "L1"
+LOSS_L2 = "MSE"
+LOSS_TYPE = "loss_type"
+SNR = "snr"
+
+# Checkpoint
+MODEL = "model"
+CURRENT_EPOCH = "current_epoch"
+CONFIGURATION = "configuration"
+
+
+# Signal names
+CLEAN = "clean"
+NOISY = "noise"
+MIXED = "mixed"
+PREDICTED = "predicted"
+
+
+# MISC
+PATHS = "paths"
+BUFFERS = "buffers"
+SAMPLING_RATE = "sampling_rate"
diff --git a/src/gyraudio/audio_separation/train.py b/src/gyraudio/audio_separation/train.py
new file mode 100644
index 0000000000000000000000000000000000000000..a45cb33c8d10edf1ac6dc27e0e6e31b9719a16ba
--- /dev/null
+++ b/src/gyraudio/audio_separation/train.py
@@ -0,0 +1,183 @@
+from gyraudio.audio_separation.experiment_tracking.experiments import get_experience
+from gyraudio.audio_separation.parser import shared_parser
+from gyraudio.audio_separation.properties import (
+    TRAIN, TEST, EPOCHS, OPTIMIZER, NAME, MAX_STEPS_PER_EPOCH,
+    SIGNAL, NOISE, TOTAL, SNR, SCHEDULER, SCHEDULER_CONFIGURATION,
+    LOSS, LOSS_L2, LOSS_L1, LOSS_TYPE, COEFFICIENT
+)
+from gyraudio.default_locations import EXPERIMENT_STORAGE_ROOT
+from torch.optim.lr_scheduler import ReduceLROnPlateau
+from gyraudio.audio_separation.experiment_tracking.storage import get_output_folder, save_checkpoint
+from gyraudio.audio_separation.metrics import Costs, snr
+# from gyraudio.audio_separation.experiment_tracking.storage import load_checkpoint
+from pathlib import Path
+from gyraudio.io.dump import Dump
+import sys
+import torch
+from tqdm import tqdm
+from copy import deepcopy
+import wandb
+import logging
+
+
+def launch_training(exp: int, wandb_flag: bool = True, device: str = "cuda", save_dir: Path = None, override=False):
+
+    short_name, model, config, dl = get_experience(exp)
+    exists, output_folder = get_output_folder(config, root_dir=save_dir, override=override)
+    if not exists:
+        logging.warning(f"Skipping experiment {short_name}")
+        return False
+    else:
+        logging.info(f"Experiment {short_name} saved in {output_folder}")
+
+    print(short_name)
+    print(config)
+    logging.info(f"Starting training for {short_name}")
+    logging.info(f"Config: {config}")
+    if wandb_flag:
+        wandb.init(
+            project="audio-separation",
+            entity="teammd",
+            name=short_name,
+            tags=["debug"],
+            config=config
+        )
+    training_loop(model, config, dl, wandb_flag=wandb_flag, device=device, exp_dir=output_folder)
+    if wandb_flag:
+        wandb.finish()
+    return True
+
+
+def update_metrics(metrics, phase, pred, gt, pred_noise, gt_noise):
+    metrics[phase].update(pred, gt, SIGNAL)
+    metrics[phase].update(pred_noise, gt_noise, NOISE)
+    metrics[phase].update(pred, gt, SNR)
+    loss = metrics[phase].finish_step()
+    return loss
+
+
+def training_loop(model: torch.nn.Module, config: dict, dl, device: str = "cuda", wandb_flag: bool = False,
+                  exp_dir: Path = None):
+    optim_params = deepcopy(config[OPTIMIZER])
+    optim_name = optim_params[NAME]
+    optim_params.pop(NAME)
+    if optim_name == "adam":
+        optimizer = torch.optim.Adam(model.parameters(), **optim_params)
+
+    scheduler = None
+    scheduler_config = config.get(SCHEDULER_CONFIGURATION, {})
+    scheduler_name = config.get(SCHEDULER, False)
+    if scheduler_name:
+        if scheduler_name == "ReduceLROnPlateau":
+            scheduler = ReduceLROnPlateau(optimizer, mode='max', verbose=True, **scheduler_config)
+            logging.info(f"Using scheduler {scheduler_name} with config {scheduler_config}")
+        else:
+            raise NotImplementedError(f"Scheduler {scheduler_name} not implemented")
+    max_steps = config.get(MAX_STEPS_PER_EPOCH, None)
+    chosen_loss = config.get(LOSS, LOSS_L2)
+    if chosen_loss == LOSS_L2:
+        costs = {TRAIN:  Costs(TRAIN), TEST: Costs(TEST)}
+    elif chosen_loss == LOSS_L1:
+        cost_init = {
+            SIGNAL: {
+                COEFFICIENT: 0.5,
+                LOSS_TYPE: torch.nn.functional.l1_loss
+            },
+            NOISE: {
+                COEFFICIENT: 0.5,
+                LOSS_TYPE: torch.nn.functional.l1_loss
+            },
+            SNR: {
+                LOSS_TYPE: snr
+            }
+        }
+        costs = {
+            TRAIN:  Costs(TRAIN, costs=cost_init),
+            TEST: Costs(TEST)
+        }
+    for epoch in range(config[EPOCHS]):
+        costs[TRAIN].reset_epoch()
+        costs[TEST].reset_epoch()
+        model.to(device)
+        # Training loop
+        # -----------------------------------------------------------
+
+        metrics = {TRAIN: {}, TEST: {}}
+        for step_index, (batch_mix, batch_signal, batch_noise) in tqdm(
+                enumerate(dl[TRAIN]), desc=f"Epoch {epoch}", total=len(dl[TRAIN])):
+            if max_steps is not None and step_index >= max_steps:
+                break
+            batch_mix, batch_signal, batch_noise = batch_mix.to(device), batch_signal.to(device), batch_noise.to(device)
+            model.zero_grad()
+            batch_output_signal, batch_output_noise = model(batch_mix)
+            loss = update_metrics(
+                costs, TRAIN,
+                batch_output_signal, batch_signal,
+                batch_output_noise, batch_noise
+            )
+            # costs[TRAIN].update(batch_output_signal, batch_signal, SIGNAL)
+            # costs[TRAIN].update(batch_output_noise, batch_noise, NOISE)
+            # loss = costs[TRAIN].finish_step()
+            loss.backward()
+            optimizer.step()
+        costs[TRAIN].finish_epoch()
+
+        # Validation loop
+        # -----------------------------------------------------------
+        model.eval()
+        torch.cuda.empty_cache()
+        with torch.no_grad():
+            for step_index, (batch_mix, batch_signal, batch_noise) in tqdm(
+                    enumerate(dl[TEST]), desc=f"Epoch {epoch}", total=len(dl[TEST])):
+                if max_steps is not None and step_index >= max_steps:
+                    break
+                batch_mix, batch_signal, batch_noise = batch_mix.to(
+                    device), batch_signal.to(device), batch_noise.to(device)
+                batch_output_signal, batch_output_noise = model(batch_mix)
+                loss = update_metrics(
+                    costs, TEST,
+                    batch_output_signal, batch_signal,
+                    batch_output_noise, batch_noise
+                )
+        costs[TEST].finish_epoch()
+        if scheduler is not None and isinstance(scheduler, ReduceLROnPlateau):
+            scheduler.step(costs[TEST].total_metric[SNR])
+        print(f"epoch {epoch}:\n{costs[TRAIN]}\n{costs[TEST]}")
+        wandblogs = {}
+        if wandb_flag:
+            for phase in [TRAIN, TEST]:
+                wandblogs[f"{phase} loss signal"] = costs[phase].total_metric[SIGNAL]
+                wandblogs[f"debug loss/{phase} loss signal"] = costs[phase].total_metric[SIGNAL]
+                wandblogs[f"debug loss/{phase} loss total"] = costs[phase].total_metric[TOTAL]
+                wandblogs[f"debug loss/{phase} loss noise"] = costs[phase].total_metric[NOISE]
+                wandblogs[f"{phase} snr"] = costs[phase].total_metric[SNR]
+                wandblogs["learning rate"] = optimizer.param_groups[0]['lr']
+            wandb.log(wandblogs)
+        metrics[TRAIN] = costs[TRAIN].total_metric
+        metrics[TEST] = costs[TEST].total_metric
+        Dump.save_json(metrics, exp_dir/f"metrics_{epoch:04d}.json")
+        save_checkpoint(model, exp_dir, optimizer, config=config, epoch=epoch)
+        torch.cuda.empty_cache()
+
+
+def main(argv):
+    default_device = "cuda" if torch.cuda.is_available() else "cpu"
+    parser_def = shared_parser(help="Launch training \nCheck results at: https://wandb.ai/teammd/audio-separation"
+                               + ("\n<<<Cuda available>>>" if default_device == "cuda" else ""))
+    parser_def.add_argument("-nowb", "--no-wandb", action="store_true")
+    parser_def.add_argument("-o", "--output-dir", type=str, default=EXPERIMENT_STORAGE_ROOT)
+    parser_def.add_argument("-f", "--force", action="store_true", help="Override existing experiment")
+
+    parser_def.add_argument("-d", "--device", type=str, default=default_device,
+                            help="Training device", choices=["cpu", "cuda"])
+    args = parser_def.parse_args(argv)
+    for exp in args.experiments:
+        launch_training(
+            exp, wandb_flag=not args.no_wandb, save_dir=Path(args.output_dir),
+            override=args.force,
+            device=args.device
+        )
+
+
+if __name__ == "__main__":
+    main(sys.argv[1:])
diff --git a/src/gyraudio/audio_separation/visualization/audio_player.py b/src/gyraudio/audio_separation/visualization/audio_player.py
new file mode 100644
index 0000000000000000000000000000000000000000..d8a9c0a5e9772fe25beca5f08ffd0eb3120ae007
--- /dev/null
+++ b/src/gyraudio/audio_separation/visualization/audio_player.py
@@ -0,0 +1,84 @@
+from gyraudio.audio_separation.properties import CLEAN, NOISY, MIXED, PREDICTED, SAMPLING_RATE
+from pathlib import Path
+from gyraudio.io.audio import save_audio_tensor
+from gyraudio import root_dir
+from interactive_pipe import Control, KeyboardControl
+from interactive_pipe import interactive
+import logging
+
+HERE = Path(__file__).parent
+MUTE = "mute"
+LOGOS = {
+    PREDICTED: HERE/"play_logo_pred.png",
+    MIXED: HERE/"play_logo_mixed.png",
+    CLEAN: HERE/"play_logo_clean.png",
+    NOISY: HERE/"play_logo_noise.png",
+    MUTE: HERE/"mute_logo.png",
+}
+ICONS = [it for key, it in LOGOS.items()]
+KEYS = [key for key, it in LOGOS.items()]
+
+ping_pong_index = 0
+
+
+@interactive(
+    player=Control(MUTE, KEYS, icons=ICONS))
+def audio_selector(sig, mixed, pred, global_params={}, player=MUTE):
+
+    global_params["selected_audio"] = player if player != MUTE else global_params.get("selected_audio", MIXED)
+    global_params[MUTE] = player == MUTE
+    if player == CLEAN:
+        audio_track = sig["buffers"][CLEAN]
+    elif player == NOISY:
+        audio_track = sig["buffers"][NOISY]
+    elif player == MIXED:
+        audio_track = mixed
+    elif player == PREDICTED:
+        audio_track = pred
+    else:
+        audio_track = mixed
+    return audio_track
+
+
+@interactive(
+    loop=KeyboardControl(True, keydown="l"))
+def audio_trim(audio_track, global_params={}, loop=True):
+    sampling_rate = global_params.get(SAMPLING_RATE, 8000)
+    if global_params.get("trim", False):
+        start, end = global_params["trim"]["start"], global_params["trim"]["end"]
+        remainder = (end-start) % 8
+        audio_trim = audio_track[..., start:end-remainder]
+        repeat_factor = int(sampling_rate*4./(end-start))
+        logging.debug(f"{repeat_factor}")
+        repeat_factor = max(1, repeat_factor)
+        if loop:
+            repeat_factor = 1
+        audio_trim = audio_trim.repeat(1, repeat_factor)
+        logging.debug(f"{audio_trim.shape}")
+    else:
+        audio_trim = audio_track
+    return audio_trim
+
+
+@interactive(
+    volume=(100, [0, 1000], "volume"),
+)
+def audio_player(audio_trim, global_params={}, volume=100):
+    sampling_rate = global_params.get(SAMPLING_RATE, 8000)
+    try:
+        if global_params.get(MUTE, True):
+            global_params["__stop"]()
+            print("mute!")
+        else:
+            ping_pong_path = root_dir/"__ping_pong"
+            ping_pong_path.mkdir(exist_ok=True)
+            global ping_pong_index
+            audio_track_path = ping_pong_path/f"_tmp_{ping_pong_index}.wav"
+            ping_pong_index = (ping_pong_index + 1) % 10
+            save_audio_tensor(audio_track_path, volume/100.*audio_trim,
+                              sampling_rate=global_params.get(SAMPLING_RATE, sampling_rate))
+            global_params["__set_audio"](audio_track_path)
+            global_params["__play"]()
+    except Exception as exc:
+        logging.warning(f"Exception in audio_player {exc}")
+        pass
diff --git a/src/gyraudio/audio_separation/visualization/interactive_audio.py b/src/gyraudio/audio_separation/visualization/interactive_audio.py
new file mode 100644
index 0000000000000000000000000000000000000000..aea428f3976f981a0621220869a5275d52aa6ebd
--- /dev/null
+++ b/src/gyraudio/audio_separation/visualization/interactive_audio.py
@@ -0,0 +1,392 @@
+from batch_processing import Batch
+import argparse
+from pathlib import Path
+from gyraudio.audio_separation.experiment_tracking.experiments import get_experience
+from gyraudio.audio_separation.experiment_tracking.storage import get_output_folder
+from gyraudio.default_locations import EXPERIMENT_STORAGE_ROOT
+from gyraudio.audio_separation.properties import (
+    SHORT_NAME, CLEAN, NOISY, MIXED, PREDICTED, ANNOTATIONS, PATHS, BUFFERS, SAMPLING_RATE, NAME
+)
+import torch
+from gyraudio.audio_separation.experiment_tracking.storage import load_checkpoint
+from gyraudio.audio_separation.visualization.pre_load_audio import (
+    parse_command_line_audio_load, load_buffers, audio_loading_batch)
+from gyraudio.audio_separation.visualization.pre_load_custom_audio import (
+    parse_command_line_generic_audio_load, generic_audio_loading_batch,
+    load_buffers_custom
+)
+from torchaudio.functional import resample
+from typing import List
+import numpy as np
+import logging
+from interactive_pipe.data_objects.curves import Curve, SingleCurve
+from interactive_pipe import interactive, KeyboardControl, Control
+from interactive_pipe.headless.pipeline import HeadlessPipeline
+from interactive_pipe.graphical.qt_gui import InteractivePipeQT
+from interactive_pipe.graphical.mpl_gui import InteractivePipeMatplotlib
+from gyraudio.audio_separation.visualization.audio_player import audio_selector, audio_trim, audio_player
+
+default_device = "cuda" if torch.cuda.is_available() else "cpu"
+LEARNT_SAMPLING_RATE = 8000
+
+
+@interactive(
+    idx=KeyboardControl(value_default=0, value_range=[
+                        0, 1000], modulo=True, keyup="8", keydown="2"),
+    idn=KeyboardControl(value_default=0, value_range=[
+                        0, 1000], modulo=True, keyup="9", keydown="3")
+)
+def signal_selector(signals, idx=0, idn=0, global_params={}):
+    if isinstance(signals, dict):
+        clean_sigs = signals[CLEAN]
+        clean = clean_sigs[idx % len(clean_sigs)]
+        if BUFFERS not in clean:
+            load_buffers_custom(clean)
+        noise_sigs = signals[NOISY]
+        noise = noise_sigs[idn % len(noise_sigs)]
+        if BUFFERS not in noise:
+            load_buffers_custom(noise)
+        cbuf, nbuf = clean[BUFFERS], noise[BUFFERS]
+        if clean[SAMPLING_RATE] != LEARNT_SAMPLING_RATE:
+            cbuf = resample(cbuf, clean[SAMPLING_RATE], LEARNT_SAMPLING_RATE)
+            clean[SAMPLING_RATE] = LEARNT_SAMPLING_RATE
+        if noise[SAMPLING_RATE] != LEARNT_SAMPLING_RATE:
+            nbuf = resample(nbuf, noise[SAMPLING_RATE], LEARNT_SAMPLING_RATE)
+            noise[SAMPLING_RATE] = LEARNT_SAMPLING_RATE
+        min_length = min(cbuf.shape[-1], nbuf.shape[-1])
+        min_length = min_length - min_length % 1024
+        signal = {
+            PATHS: {
+                CLEAN: clean[PATHS],
+                NOISY: noise[PATHS]
+
+            },
+            BUFFERS: {
+                CLEAN: cbuf[..., :1, :min_length],
+                NOISY: nbuf[..., :1, :min_length],
+            },
+            NAME: f"Clean={clean[NAME]} | Noise={noise[NAME]}",
+            SAMPLING_RATE: LEARNT_SAMPLING_RATE
+        }
+    else:
+        # signals are loaded in CPU
+        signal = signals[idx % len(signals)]
+        if BUFFERS not in signal:
+            load_buffers(signal)
+        global_params["premixed_snr"] = signal.get("premixed_snr", None)
+        signal[NAME] = f"File={signal[NAME]}"
+    global_params["selected_info"] = signal[NAME]
+    global_params[SAMPLING_RATE] = signal[SAMPLING_RATE]
+    return signal
+
+
+@interactive(
+    snr=(0., [-10., 10.], "SNR [dB]")
+)
+def remix(signals, snr=0., global_params={}):
+    signal = signals[BUFFERS][CLEAN]
+    noisy = signals[BUFFERS][NOISY]
+    alpha = 10 ** (-snr / 20) * torch.norm(signal) / torch.norm(noisy)
+    mixed_signal = signal + alpha * noisy
+    global_params["snr"] = snr
+    return mixed_signal
+
+
+@interactive(std_dev=Control(0., value_range=[0., 0.1], name="extra noise std", step=0.0001),
+             amplify=(1., [0., 10.], "amplification of everything"))
+def augment(signals, mixed, std_dev=0., amplify=1.):
+    signals[BUFFERS][MIXED] *= amplify
+    signals[BUFFERS][NOISY] *= amplify
+    signals[BUFFERS][CLEAN] *= amplify
+    mixed = mixed*amplify+torch.randn_like(mixed)*std_dev
+    return signals, mixed
+
+
+# @interactive(
+#     device=("cuda", ["cpu", "cuda"]) if default_device == "cuda" else ("cpu", ["cpu"])
+# )
+def select_device(device=default_device, global_params={}):
+    global_params["device"] = device
+
+
+@interactive(
+    model=KeyboardControl(value_default=0, value_range=[
+                          0, 99], keyup="pagedown", keydown="pageup")
+)
+def audio_sep_inference(mixed, models, configs, model: int = 0, global_params={}):
+    selected_model = models[model % len(models)]
+    config = configs[model % len(models)]
+    short_name = config.get(SHORT_NAME, "")
+    annotations = config.get(ANNOTATIONS, "")
+    device = global_params.get("device", "cpu")
+    with torch.no_grad():
+        selected_model.eval()
+        selected_model.to(device)
+        predicted_signal, predicted_noise = selected_model(
+            mixed.to(device).unsqueeze(0))
+        predicted_signal = predicted_signal.squeeze(0)
+    pred_curve = SingleCurve(y=predicted_signal[0, :].detach().cpu().numpy(),
+                             style="g-", label=f"predicted_{short_name} {annotations}")
+    return predicted_signal, pred_curve
+
+
+def compute_metrics(pred, sig, global_params={}):
+    METRICS = "metrics"
+    target = sig[BUFFERS][CLEAN]
+    global_params[METRICS] = {}
+    global_params[METRICS]["MSE"] = torch.mean((target-pred.cpu())**2)
+    global_params[METRICS]["SNR"] = 10. * \
+        torch.log10(torch.sum(target**2)/torch.sum((target-pred.cpu())**2))
+
+
+def get_trim(sig, zoom, center, num_samples=300):
+    N = len(sig)
+    native_ds = N/num_samples
+    center_idx = int(center*N)
+    window = int(num_samples/zoom*native_ds)
+    start_idx = max(0, center_idx - window//2)
+    end_idx = min(N, center_idx + window//2)
+    skip_factor = max(1, int(native_ds/zoom))
+    return start_idx, end_idx, skip_factor
+
+
+def zin(sig, zoom, center, num_samples=300):
+    start_idx, end_idx, skip_factor = get_trim(
+        sig, zoom, center, num_samples=num_samples)
+    out = np.zeros(num_samples)
+    trimmed = sig[start_idx:end_idx:skip_factor]
+    out[:len(trimmed)] = trimmed[:num_samples]
+    return out
+
+
+@interactive(
+    center=KeyboardControl(value_default=0.5, value_range=[
+                           0., 1.], step=0.01, keyup="6", keydown="4"),
+    zoom=KeyboardControl(value_default=0., value_range=[
+                         0., 15.], step=1, keyup="+", keydown="-"),
+    zoomy=KeyboardControl(
+        value_default=0., value_range=[-15., 15.], step=1, keyup="up", keydown="down")
+)
+def visualize_audio(signal: dict, mixed_signal, pred, zoom=1, zoomy=0., center=0.5, global_params={}):
+    """Create curves
+    """
+    zval = 1.5**zoom
+    start_idx, end_idx, _skip_factor = get_trim(
+        signal[BUFFERS][CLEAN][0, :], zval, center)
+    global_params["trim"] = dict(start=start_idx, end=end_idx)
+    selected = global_params.get("selected_audio", MIXED)
+    clean = SingleCurve(y=zin(signal[BUFFERS][CLEAN][0, :], zval, center),
+                        alpha=1.,
+                        style="k-",
+                        linewidth=0.9,
+                        label=("*" if selected == CLEAN else " ")+"clean")
+    noisy = SingleCurve(y=zin(signal[BUFFERS][NOISY][0, :], zval, center),
+                        alpha=0.3,
+                        style="y--",
+                        linewidth=1,
+                        label=("*" if selected == NOISY else " ") + "noisy"
+                        )
+    mixed = SingleCurve(y=zin(mixed_signal[0, :], zval, center), style="r-",
+                        alpha=0.1,
+                        linewidth=2,
+                        label=("*" if selected == MIXED else " ") + "mixed")
+    # true_mixed = SingleCurve(y=zin(signal[BUFFERS][MIXED][0, :], zval, center),
+    #                          alpha=0.3, style="b-", linewidth=1, label="true mixed")
+    pred.y = zin(pred.y, zval, center)
+    pred.label = ("*" if selected == PREDICTED else " ") + pred.label
+    curves = [noisy, mixed, pred, clean]
+    title = f"SNR  in {global_params['snr']:.1f} dB"
+    if "selected_info" in global_params:
+        title += f" | {global_params['selected_info']}"
+    title += "\n"
+    for metric_name, metric_value in global_params.get("metrics", {}).items():
+        title += f" | {metric_name} "
+        title += f"{metric_value:.2e}" if (abs(metric_value) < 1e-2 or abs(metric_value)
+                                           > 1000) else f"{metric_value:.2f}"
+    # if global_params.get("premixed_snr", None) is not None:
+    #     title += f"| Premixed SNR : {global_params['premixed_snr']:.1f} dB"
+    return Curve(curves, ylim=[-0.04 * 1.5 ** zoomy, 0.04 * 1.5 ** zoomy], xlabel="Time index", ylabel="Amplitude", title=title)
+
+
+def interactive_audio_separation_processing(signals, model_list, config_list):
+    sig = signal_selector(signals)
+    mixed = remix(sig)
+    # sig, mixed = augment(sig, mixed)
+    select_device()
+    pred, pred_curve = audio_sep_inference(mixed, model_list, config_list)
+    compute_metrics(pred, sig)
+    sound = audio_selector(sig, mixed, pred)
+    curve = visualize_audio(sig, mixed, pred_curve)
+    trimmed_sound = audio_trim(sound)
+    audio_player(trimmed_sound)
+    return curve
+
+
+def interactive_audio_separation_visualization(
+        all_signals: List[dict],
+        model_list: List[torch.nn.Module],
+        config_list: List[dict],
+        gui="qt"
+):
+    pip = HeadlessPipeline.from_function(
+        interactive_audio_separation_processing, cache=False)
+    if gui == "qt":
+        app = InteractivePipeQT(
+            pipeline=pip, name="audio separation", size=(1000, 1000), audio=True)
+    else:
+        logging.warning("No support for audio player with Matplotlib")
+        app = InteractivePipeMatplotlib(
+            pipeline=pip, name="audio separation", size=None, audio=False)
+    app(all_signals, model_list, config_list)
+
+
+def visualization(
+    all_signals: List[dict],
+    model_list: List[torch.nn.Module],
+    config_list: List[dict],
+    device="cuda"
+):
+    for signal in all_signals:
+        if BUFFERS not in signal:
+            load_buffers(signal, device="cpu")
+        clean = SingleCurve(y=signal[BUFFERS][CLEAN][0, :], label="clean")
+        noisy = SingleCurve(y=signal[BUFFERS][NOISY]
+                            [0, :], label="noise", alpha=0.3)
+        curves = [clean, noisy]
+        for config, model in zip(config_list, model_list):
+            short_name = config.get(SHORT_NAME, "unknown")
+            predicted_signal, predicted_noise = model(
+                signal[BUFFERS][MIXED].to(device).unsqueeze(0))
+            predicted = SingleCurve(y=predicted_signal.squeeze(0)[0, :].detach().cpu().numpy(),
+                                    label=f"predicted_{short_name}")
+            curves.append(predicted)
+        Curve(curves).show()
+
+
+def parse_command_line(parser: Batch = None, gradio_demo=True) -> argparse.ArgumentParser:
+    if gradio_demo:
+        parser = parse_command_line_gradio(parser)
+    else:
+        parser = parse_command_line_generic(parser)
+    return parser
+
+
+def parse_command_line_gradio(parser: Batch = None, gradio_demo=True) -> argparse.ArgumentParser:
+    if parser is None:
+        parser = parse_command_line_audio_load()
+    default_device = "cuda" if torch.cuda.is_available() else "cpu"
+    iparse = parser.add_argument_group("Audio separation visualization")
+    iparse.add_argument("-e",  "--experiments", type=int, nargs="+", default=3001,
+                        help="Experiment ids to be inferred sequentially")
+    iparse.add_argument("-p", "--interactive", default=True,
+                        action="store_true", help="Play = Interactive mode")
+    iparse.add_argument("-m", "--model-root", type=str,
+                        default=EXPERIMENT_STORAGE_ROOT)
+    iparse.add_argument("-d", "--device", type=str, default=default_device,
+                        choices=["cpu", "cuda"] if default_device == "cuda" else ["cpu"])
+    iparse.add_argument("-gui", "--gui", type=str,
+                        default="gradio", choices=["qt", "mpl", "gradio"])
+    pri
+    return parser
+
+
+def parse_command_line_generic(parser: Batch = None, gradio_demo=True) -> argparse.ArgumentParser:
+    if parser is None:
+        parser = parse_command_line_audio_load()
+    default_device = "cuda" if torch.cuda.is_available() else "cpu"
+    iparse = parser.add_argument_group("Audio separation visualization")
+    iparse.add_argument("-e",  "--experiments", type=int, nargs="+", required=True,
+                        help="Experiment ids to be inferred sequentially")
+    iparse.add_argument("-p", "--interactive",
+                        action="store_true", help="Play = Interactive mode")
+    iparse.add_argument("-m", "--model-root", type=str,
+                        default=EXPERIMENT_STORAGE_ROOT)
+    iparse.add_argument("-d", "--device", type=str, default=default_device,
+                        choices=["cpu", "cuda"] if default_device == "cuda" else ["cpu"])
+    iparse.add_argument("-gui", "--gui", type=str,
+                        default="qt", choices=["qt", "mpl", "gradio"])
+    return parser
+
+
+def main(argv: List[str]):
+    """Paired signals and noise in folders"""
+    batch = Batch(argv)
+    batch.set_io_description(
+        input_help='input audio files',
+        output_help=argparse.SUPPRESS
+    )
+    batch.set_multiprocessing_enabled(False)
+    parser = parse_command_line()
+    args = batch.parse_args(parser)
+    exp = args.experiments[0]
+    device = args.device
+    models_list = []
+    config_list = []
+    logging.info(f"Loading experiments models {args.experiments}")
+    for exp in args.experiments:
+        model_dir = Path(args.model_root)
+        short_name, model, config, _dl = get_experience(exp)
+        _, exp_dir = get_output_folder(
+            config, root_dir=model_dir, override=False)
+        assert exp_dir.exists(
+        ), f"Experiment {short_name} does not exist in {model_dir}"
+        model.eval()
+        model.to(device)
+        model, __optimizer, epoch, config = load_checkpoint(
+            model, exp_dir, epoch=None, device=args.device)
+        config[SHORT_NAME] = short_name
+        models_list.append(model)
+        config_list.append(config)
+    logging.info("Load audio buffers:")
+    all_signals = batch.run(audio_loading_batch)
+    if not args.interactive:
+        visualization(all_signals, models_list, config_list, device=device)
+    else:
+        interactive_audio_separation_visualization(
+            all_signals, models_list, config_list, gui=args.gui)
+
+
+def main_custom(argv: List[str]):
+    """Handle custom noise and custom signals
+    """
+    parser = parse_command_line()
+    parser.add_argument("-s", "--signal", type=str, required=True,
+                        nargs="+", help="Signal to be preloaded")
+    parser.add_argument("-n", "--noise", type=str, required=True,
+                        nargs="+", help="Noise to be preloaded")
+    args = parser.parse_args(argv)
+    exp = args.experiments[0]
+    device = args.device
+    models_list = []
+    config_list = []
+    logging.info(f"Loading experiments models {args.experiments}")
+    for exp in args.experiments:
+        model_dir = Path(args.model_root)
+        short_name, model, config, _dl = get_experience(exp)
+        _, exp_dir = get_output_folder(
+            config, root_dir=model_dir, override=False)
+        assert exp_dir.exists(
+        ), f"Experiment {short_name} does not exist in {model_dir}"
+        model.eval()
+        model.to(device)
+        model, __optimizer, epoch, config = load_checkpoint(
+            model, exp_dir, epoch=None, device=args.device)
+        config[SHORT_NAME] = short_name
+        models_list.append(model)
+        config_list.append(config)
+    all_signals = {}
+    for args_paths, key in zip([args.signal, args.noise], [CLEAN, NOISY]):
+        new_argv = ["-i"] + args_paths
+        if args.preload:
+            new_argv += ["--preload"]
+        batch = Batch(new_argv)
+        new_parser = parse_command_line_generic_audio_load()
+        batch.set_io_description(
+            input_help=argparse.SUPPRESS,  # 'input audio files',
+            output_help=argparse.SUPPRESS
+        )
+        batch.set_multiprocessing_enabled(False)
+        _ = batch.parse_args(new_parser)
+        all_signals[key] = batch.run(generic_audio_loading_batch)
+    interactive_audio_separation_visualization(
+        all_signals, models_list, config_list, gui=args.gui)
diff --git a/src/gyraudio/audio_separation/visualization/interactive_infer.py b/src/gyraudio/audio_separation/visualization/interactive_infer.py
new file mode 100644
index 0000000000000000000000000000000000000000..cafa00b67fffe1e9108c93ed9f45c9188436ca00
--- /dev/null
+++ b/src/gyraudio/audio_separation/visualization/interactive_infer.py
@@ -0,0 +1,117 @@
+from gyraudio.default_locations import EXPERIMENT_STORAGE_ROOT
+from gyraudio.audio_separation.parser import shared_parser
+from gyraudio.audio_separation.infer import launch_infer, RECORD_KEYS, SNR_OUT, SNR_IN, NBATCH, SAVE_IDX
+from gyraudio.audio_separation.properties import TEST, NAME, SHORT_NAME, CURRENT_EPOCH, SNR_FILTER 
+import sys
+import os
+from dash import Dash, html, dcc, callback, Output, Input, dash_table
+import plotly.express as px
+import plotly.graph_objects as go
+from plotly.subplots import make_subplots
+import pandas as pd
+from typing import List
+import torch
+from pathlib import Path
+DIFF_SNR = 'SNR out - SNR in'
+
+
+
+def get_app(record_row_dfs : pd.DataFrame, eval_dfs : List[pd.DataFrame]) :
+    app = Dash(__name__)
+    # names_options = [{'label' : f"{record[SHORT_NAME]} - {record[NAME]} epoch {record[CURRENT_EPOCH]:04d}", 'value' : record[NAME] } for idx, record in record_row_dfs.iterrows()]
+    app.layout = html.Div([
+        html.H1(children='Inference results', style={'textAlign':'center'}),
+        # dcc.Dropdown(names_options, names_options[0]['value'], id='exp-selection'),
+        # dcc.RadioItems(['scatter', 'box'], 'box', inline=True, id='radio-plot-type'),
+        dcc.RadioItems([SNR_OUT, DIFF_SNR], DIFF_SNR, inline=True, id='radio-plot-out'),
+        dcc.Graph(id='graph-content')
+    ])
+
+    @callback(
+        Output('graph-content', 'figure'),
+        # Input('exp-selection', 'value'),
+        # Input('radio-plot-type', 'value'),
+        Input('radio-plot-out', 'value'),
+    )
+    def update_graph(radio_plot_out) :
+        fig = make_subplots(rows = 2, cols = 1)
+        colors = px.colors.qualitative.Plotly
+        for id, record in record_row_dfs.iterrows() :
+            color = colors[id % len(colors)]
+            eval_df = eval_dfs[id].sort_values(by=SNR_IN)
+            eval_df[DIFF_SNR] = eval_df[SNR_OUT] - eval_df[SNR_IN]
+            legend = f'{record[SHORT_NAME]}_{record[NAME]}'
+            fig.add_trace(
+                go.Scatter(
+                    x=eval_df[SNR_IN], 
+                    y=eval_df[radio_plot_out], 
+                    mode="markers", marker={'color' : color}, 
+                    name=legend,
+                    hovertemplate = 'File : %{text}'+
+                        '<br>%{y}<br>',
+                    text = [f"{eval[SAVE_IDX]:.0f}" for idx, eval in eval_df.iterrows()]
+                    ),
+                row = 1, col = 1
+            )
+            eval_df_bins = eval_df
+            eval_df_bins[SNR_IN] = eval_df_bins[SNR_IN].apply(lambda snr : round(snr))
+            fig.add_trace(
+                go.Box(
+                    x=eval_df[SNR_IN], 
+                    y=eval_df[radio_plot_out], 
+                    fillcolor = color, 
+                    marker={'color' : color},
+                    name = legend
+                    ),
+                row = 2, col = 1
+            )
+
+        title = f"SNR performances"
+        fig.update_layout(
+            title=title,
+            xaxis2_title = SNR_IN,
+            yaxis_title = radio_plot_out,
+            hovermode='x unified'
+            )
+        return fig
+
+        
+    
+    return app
+
+    
+def main(argv):
+    default_device = "cuda" if torch.cuda.is_available() else "cpu"
+    parser_def = shared_parser(help="Launch training \nCheck results at: https://wandb.ai/balthazarneveu/audio-sep"
+                               + ("\n<<<Cuda available>>>" if default_device == "cuda" else ""))
+    parser_def.add_argument("-i", "--input-dir", type=str, default=EXPERIMENT_STORAGE_ROOT)
+    parser_def.add_argument("-o", "--output-dir", type=str, default=EXPERIMENT_STORAGE_ROOT)
+    parser_def.add_argument("-d", "--device", type=str, default=default_device,
+                            help="Training device", choices=["cpu", "cuda"])
+    parser_def.add_argument("-b", "--nb-batch", type=int, default=None,
+                    help="Number of batches to process")
+    parser_def.add_argument("-s",  "--snr-filter", type=float, nargs="+", default=None,
+                    help="SNR filters on the inference dataloader")
+    args = parser_def.parse_args(argv)
+    record_row_dfs = pd.DataFrame(columns = RECORD_KEYS)
+    eval_dfs = []
+    for exp in args.experiments:
+        record_row_df, evaluation_path = launch_infer(
+                exp,
+                model_dir=Path(args.input_dir),
+                output_dir=Path(args.output_dir),
+                device=args.device,
+                max_batches=args.nb_batch,
+                snr_filter=args.snr_filter
+            )
+        eval_df = pd.read_csv(evaluation_path)
+        # Careful, list order for concat is important for index matching eval_dfs list
+        record_row_dfs = pd.concat([record_row_dfs.loc[:], record_row_df], ignore_index=True)
+        eval_dfs.append(eval_df)
+    app = get_app(record_row_dfs, eval_dfs)
+    app.run(debug=True)
+
+
+if __name__ == '__main__':
+    os.environ["KMP_DUPLICATE_LIB_OK"] = "True"
+    main(sys.argv[1:])
\ No newline at end of file
diff --git a/src/gyraudio/audio_separation/visualization/mute_logo.png b/src/gyraudio/audio_separation/visualization/mute_logo.png
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diff --git a/src/gyraudio/audio_separation/visualization/play_logo_clean.png b/src/gyraudio/audio_separation/visualization/play_logo_clean.png
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diff --git a/src/gyraudio/audio_separation/visualization/play_logo_mixed.png b/src/gyraudio/audio_separation/visualization/play_logo_mixed.png
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diff --git a/src/gyraudio/audio_separation/visualization/play_logo_noise.png b/src/gyraudio/audio_separation/visualization/play_logo_noise.png
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diff --git a/src/gyraudio/audio_separation/visualization/play_logo_pred.png b/src/gyraudio/audio_separation/visualization/play_logo_pred.png
new file mode 100644
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diff --git a/src/gyraudio/audio_separation/visualization/pre_load_audio.py b/src/gyraudio/audio_separation/visualization/pre_load_audio.py
new file mode 100644
index 0000000000000000000000000000000000000000..c6a9a1b36eebe27cad611255879591569a403c5f
--- /dev/null
+++ b/src/gyraudio/audio_separation/visualization/pre_load_audio.py
@@ -0,0 +1,71 @@
+from batch_processing import Batch
+import argparse
+import sys
+from pathlib import Path
+from gyraudio.audio_separation.properties import CLEAN, NOISY, MIXED, PATHS, BUFFERS, NAME, SAMPLING_RATE
+from gyraudio.io.audio import load_audio_tensor
+
+
+def parse_command_line_audio_load() -> argparse.ArgumentParser:
+    parser = argparse.ArgumentParser(description='Batch audio processing',
+                                     formatter_class=argparse.ArgumentDefaultsHelpFormatter)
+
+    parser.add_argument("-preload", "--preload", action="store_true", help="Preload audio files")
+    return parser
+
+
+def outp(path: Path, suffix: str, extension=".wav"):
+    return (path.parent / (path.stem + suffix)).with_suffix(extension)
+
+
+def load_buffers(signal: dict, device="cpu") -> None:
+    clean_signal, sampling_rate = load_audio_tensor(signal[PATHS][CLEAN], device=device)
+    noisy_signal, sampling_rate = load_audio_tensor(signal[PATHS][NOISY], device=device)
+    mixed_signal, sampling_rate = load_audio_tensor(signal[PATHS][MIXED], device=device)
+    signal[BUFFERS] = {
+        CLEAN: clean_signal,
+        NOISY: noisy_signal,
+        MIXED: mixed_signal
+    }
+    signal[SAMPLING_RATE] = sampling_rate
+
+
+def audio_loading(input: Path, preload: bool) -> dict:
+    name = input.name
+    clean_audio_path = input/"voice.wav"
+    noisy_audio_path = input/"noise.wav"
+    mixed_audio_path = list(input.glob("mix*.wav"))[0]
+    signal = {
+        NAME: name,
+        PATHS: {
+            CLEAN: clean_audio_path,
+            NOISY: noisy_audio_path,
+            MIXED: mixed_audio_path
+        }
+    }
+    signal["premixed_snr"] = float(mixed_audio_path.stem.split("_")[-1])
+    if preload:
+        load_buffers(signal)
+    return signal
+
+
+def audio_loading_batch(input: Path, args: argparse.Namespace) -> dict:
+    """Wrapper to load audio files from a directory using batch_processing
+    """
+    return audio_loading(input, preload=args.preload)
+
+
+def main(argv):
+    batch = Batch(argv)
+    batch.set_io_description(
+        input_help='input audio files',
+        output_help=argparse.SUPPRESS
+    )
+    parser = parse_command_line_audio_load()
+    batch.parse_args(parser)
+    all_signals = batch.run(audio_loading_batch)
+    return all_signals
+
+
+if __name__ == "__main__":
+    main(sys.argv[1:])
diff --git a/src/gyraudio/audio_separation/visualization/pre_load_custom_audio.py b/src/gyraudio/audio_separation/visualization/pre_load_custom_audio.py
new file mode 100644
index 0000000000000000000000000000000000000000..e4c08865cca223ee7ce5f835d25f26d7212ef2aa
--- /dev/null
+++ b/src/gyraudio/audio_separation/visualization/pre_load_custom_audio.py
@@ -0,0 +1,53 @@
+from batch_processing import Batch
+import argparse
+import sys
+from pathlib import Path
+from gyraudio.audio_separation.properties import PATHS, BUFFERS, NAME, SAMPLING_RATE
+from gyraudio.io.audio import load_audio_tensor
+
+
+def parse_command_line_generic_audio_load() -> argparse.ArgumentParser:
+    parser = argparse.ArgumentParser(description='Batch audio loading',
+                                     formatter_class=argparse.ArgumentDefaultsHelpFormatter)
+
+    parser.add_argument("-preload", "--preload", action="store_true", help="Preload audio files")
+    return parser
+
+
+def load_buffers_custom(signal: dict, device="cpu") -> None:
+    generic_signal, sampling_rate = load_audio_tensor(signal[PATHS], device=device)
+    signal[BUFFERS] = generic_signal
+    signal[SAMPLING_RATE] = sampling_rate
+
+
+def audio_loading(input: Path, preload: bool) -> dict:
+    name = input.parent.name + "/" + input.stem
+    signal = {
+        NAME: name,
+        PATHS: input,
+    }
+    if preload:
+        load_buffers_custom(signal)
+    return signal
+
+
+def generic_audio_loading_batch(input: Path, args: argparse.Namespace) -> dict:
+    """Wrapper to load audio files from a directory using batch_processing
+    """
+    return audio_loading(input, preload=args.preload)
+
+
+def main(argv):
+    batch = Batch(argv)
+    batch.set_io_description(
+        input_help='input audio files',
+        output_help=argparse.SUPPRESS
+    )
+    parser = parse_command_line_generic_audio_load()
+    batch.parse_args(parser)
+    all_signals = batch.run(generic_audio_loading_batch)
+    return all_signals
+
+
+if __name__ == "__main__":
+    main(sys.argv[1:])
diff --git a/src/gyraudio/default_locations.py b/src/gyraudio/default_locations.py
new file mode 100644
index 0000000000000000000000000000000000000000..4b3201a8cb33924a00ac89451759a8a34d84021b
--- /dev/null
+++ b/src/gyraudio/default_locations.py
@@ -0,0 +1,5 @@
+from gyraudio import root_dir
+RAW_AUDIO_ROOT = root_dir/"__data_source_separation"/"voice_origin"
+MIXED_AUDIO_ROOT = root_dir/"__data_source_separation"/"source_separation"
+EXPERIMENT_STORAGE_ROOT = root_dir/"__output_audiosep"
+SAMPLE_ROOT = root_dir/"audio_samples"
diff --git a/src/gyraudio/io/audio.py b/src/gyraudio/io/audio.py
new file mode 100644
index 0000000000000000000000000000000000000000..e97121a3fafcadb9324ed5636f5c3e57320a0eb0
--- /dev/null
+++ b/src/gyraudio/io/audio.py
@@ -0,0 +1,29 @@
+import scipy
+import numpy as np
+import torchaudio
+import torch
+from pathlib import Path
+from typing import Tuple
+
+
+def load_raw_audio(path: str) -> np.array:
+    assert path.exists(), f"Audio path {path} does not exist"
+    rate, signal = scipy.io.wavfile.read(path)
+    return rate, signal
+
+
+def load_audio_tensor(path: Path, device=None) -> Tuple[torch.Tensor, int]:
+    assert path.exists(), f"Audio path {path} does not exist"
+    signal, rate = torchaudio.load(str(path))
+    if device is not None:
+        signal = signal.to(device)
+    return signal, rate
+
+
+def save_audio_tensor(path: Path, signal: torch.Tensor, sampling_rate: int):
+    torchaudio.save(
+        str(path),
+        signal.detach().cpu(),
+        sample_rate=sampling_rate,
+        channels_first=True
+    )
diff --git a/src/gyraudio/io/dump.py b/src/gyraudio/io/dump.py
new file mode 100644
index 0000000000000000000000000000000000000000..4c7453339493a47ac10cb0266cd42e69056584f3
--- /dev/null
+++ b/src/gyraudio/io/dump.py
@@ -0,0 +1,54 @@
+from pathlib import Path
+import json
+import logging
+import pickle
+YAML_SUPPORT = True
+YAML_NOT_DETECTED_MESSAGE = "yaml is not installed, consider installing it by pip install PyYAML"
+try:
+    import yaml
+    from yaml.loader import SafeLoader, BaseLoader
+except ImportError as e:
+    YAML_SUPPORT = False
+    logging.warning(f"{e}\n{YAML_NOT_DETECTED_MESSAGE}")
+
+
+class Dump:
+    @staticmethod
+    def load_yaml(path: Path, safe_load=True) -> dict:
+        assert YAML_SUPPORT, YAML_NOT_DETECTED_MESSAGE
+        with open(path) as file:
+            params = yaml.load(
+                file, Loader=SafeLoader if safe_load else BaseLoader)
+        return params
+
+    @staticmethod
+    def save_yaml(data: dict, path: Path, **kwargs):
+        path.parent.mkdir(parents=True, exist_ok=True)
+        assert YAML_SUPPORT, YAML_NOT_DETECTED_MESSAGE
+        with open(path, 'w') as outfile:
+            yaml.dump(data, outfile, **kwargs)
+
+    @staticmethod
+    def load_json(path: Path,) -> dict:
+        with open(path) as file:
+            params = json.load(file)
+        return params
+
+    @staticmethod
+    def save_json(data: dict, path: Path):
+        path.parent.mkdir(parents=True, exist_ok=True)
+        with open(path, 'w') as outfile:
+            json.dump(data, outfile)
+
+    @staticmethod
+    def load_pickle(path: Path,) -> dict:
+        with open(path, "rb") as file:
+            unpickler = pickle.Unpickler(file)
+            params = unpickler.load()
+            # params = pickle.load(file)
+        return params
+
+    @staticmethod
+    def save_pickle(data: dict, path: Path):
+        with open(path, 'wb') as outfile:
+            pickle.dump(data, outfile)
diff --git a/src/gyraudio/io/imu.py b/src/gyraudio/io/imu.py
new file mode 100644
index 0000000000000000000000000000000000000000..aac673cb1756b350eda4f9bb424e307d83106774
--- /dev/null
+++ b/src/gyraudio/io/imu.py
@@ -0,0 +1,76 @@
+from pathlib import Path
+import gpmf
+import numpy as np
+from gyraudio.properties import GYRO_KEY, ACCL_KEY
+def extract_imu_blocks(stream, key=GYRO_KEY):
+    """ Extract imu data blocks from binary stream
+
+    This is a generator on lists `KVLItem` objects. In
+    the GPMF stream, imu data comes into blocks of several
+    different data items. For each of these blocks we return a list.
+
+    Parameters
+    ----------
+    stream: bytes
+        The raw GPMF binary stream
+
+    Returns
+    -------
+    imu_items_generator: generator
+        Generator of lists of `KVLItem` objects
+    """
+    for s in gpmf.parse.filter_klv(stream, "STRM"):
+        content = []
+        is_imu = False
+        for elt in s.value:
+            content.append(elt)
+            if elt.key == key:
+                is_imu = True
+        if is_imu:
+            yield content
+
+
+def parse_imu_block(imu_block, key=GYRO_KEY):
+    """Turn imu data blocks into `imuData` objects
+
+    Parameters
+    ----------
+    imu_block: list of KVLItem
+        A list of KVLItem corresponding to a imu data block.
+
+    Returns
+    -------
+    imu_data: imuData
+        A imuData object holding the imu information of a block.
+    """
+    block_dict = {
+        s.key: s for s in imu_block
+    }
+
+    imu_data = block_dict[key].value * 1.0 / block_dict["SCAL"].value
+    return {
+        "timestamp": block_dict["TSMP"],
+        key: imu_data,
+    }
+
+
+def get_imu_data(pth: Path) -> np.array:
+    """Extract imu data from GPMF metadata from a Gopro video file
+
+    Parameters
+    ----------
+    path: str
+        Path to the GPMF file
+
+    Returns
+    -------
+    imu_data: list of imuData
+        List of imuData objects holding the imu information of the file.
+    """
+    stream = gpmf.io.extract_gpmf_stream(pth)
+    imu_data_dict = {}
+    for key in [GYRO_KEY, ACCL_KEY]:
+        imu_blocks = extract_imu_blocks(stream, key=key)
+        imu_data = [parse_imu_block(imu_block, key=key) for imu_block in imu_blocks]
+        imu_data_dict[key] = np.vstack([np.array(imu[key]) for imu in imu_data])
+    return imu_data_dict[GYRO_KEY], imu_data_dict[ACCL_KEY]
diff --git a/src/gyraudio/properties.py b/src/gyraudio/properties.py
new file mode 100644
index 0000000000000000000000000000000000000000..56b5e8608c78ce733ce99c45f99909bd8715c2ee
--- /dev/null
+++ b/src/gyraudio/properties.py
@@ -0,0 +1,6 @@
+GYRO = "gyro"
+ACCL = "accelerometer"
+AUDIO = "audio"
+AUDIO_RATE = "audio_rate"
+GYRO_KEY = "GYRO"
+ACCL_KEY = "ACCL"