mirror of
https://github.com/ml-explore/mlx-examples.git
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* Add MusicGen model * add benchmarks * change to from_pretrained * symlinks * add readme and requirements * fix readme * readme
742 lines
24 KiB
Python
742 lines
24 KiB
Python
# Copyright © 2024 Apple Inc.
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import functools
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import json
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import math
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from pathlib import Path
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from types import SimpleNamespace
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from typing import List, Optional, Tuple, Union
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import mlx.core as mx
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import mlx.nn as nn
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import numpy as np
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_lstm_kernel = mx.fast.metal_kernel(
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name="lstm",
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input_names=["x", "h_in", "cell", "hidden_size", "time_step", "num_time_steps"],
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output_names=["hidden_state", "cell_state"],
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header="""
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template <typename T>
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T sigmoid(T x) {
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auto y = 1 / (1 + metal::exp(-metal::abs(x)));
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return (x < 0) ? 1 - y : y;
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}
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""",
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source="""
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uint b = thread_position_in_grid.x;
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uint d = hidden_size * 4;
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uint elem = b * d + thread_position_in_grid.y;
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uint index = elem;
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uint x_index = b * num_time_steps * d + time_step * d + index;
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auto i = sigmoid(h_in[index] + x[x_index]);
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index += hidden_size;
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x_index += hidden_size;
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auto f = sigmoid(h_in[index] + x[x_index]);
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index += hidden_size;
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x_index += hidden_size;
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auto g = metal::precise::tanh(h_in[index] + x[x_index]);
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index += hidden_size;
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x_index += hidden_size;
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auto o = sigmoid(h_in[index] + x[x_index]);
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cell_state[elem] = f * cell[elem] + i * g;
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hidden_state[elem] = o * metal::precise::tanh(cell_state[elem]);
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""",
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)
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def lstm_custom(x, h_in, cell, time_step):
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assert x.ndim == 3, "Input to LSTM must have 3 dimensions."
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out_shape = cell.shape
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return _lstm_kernel(
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inputs=[x, h_in, cell, out_shape[-1], time_step, x.shape[-2]],
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output_shapes=[out_shape, out_shape],
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output_dtypes=[h_in.dtype, h_in.dtype],
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grid=(x.shape[0], h_in.size // 4, 1),
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threadgroup=(256, 1, 1),
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)
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class LSTM(nn.Module):
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def __init__(
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self,
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input_size: int,
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hidden_size: int,
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bias: bool = True,
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):
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super().__init__()
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self.hidden_size = hidden_size
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self.Wx = mx.zeros((4 * hidden_size, input_size))
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self.Wh = mx.zeros((4 * hidden_size, hidden_size))
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self.bias = mx.zeros((4 * hidden_size,)) if bias else None
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def __call__(self, x, hidden=None, cell=None):
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if self.bias is not None:
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x = mx.addmm(self.bias, x, self.Wx.T)
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else:
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x = x @ self.Wx.T
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all_hidden = []
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B = x.shape[0]
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cell = cell or mx.zeros((B, self.hidden_size), x.dtype)
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for t in range(x.shape[-2]):
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if hidden is None:
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hidden = mx.zeros((B, self.hidden_size * 4), x.dtype)
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else:
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hidden = hidden @ self.Wh.T
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hidden, cell = lstm_custom(x, hidden, cell, t)
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all_hidden.append(hidden)
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return mx.stack(all_hidden, axis=-2)
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class EncodecConv1d(nn.Module):
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"""Conv1d with asymmetric or causal padding and normalization."""
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def __init__(
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self,
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config,
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in_channels: int,
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out_channels: int,
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kernel_size: int,
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stride: int = 1,
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dilation: int = 1,
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):
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super().__init__()
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self.causal = config.use_causal_conv
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self.pad_mode = config.pad_mode
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self.norm_type = config.norm_type
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self.conv = nn.Conv1d(
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in_channels, out_channels, kernel_size, stride, dilation=dilation
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)
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if self.norm_type == "time_group_norm":
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self.norm = nn.GroupNorm(1, out_channels, pytorch_compatible=True)
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self.stride = stride
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# Effective kernel size with dilations.
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self.kernel_size = (kernel_size - 1) * dilation + 1
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self.padding_total = kernel_size - stride
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def _get_extra_padding_for_conv1d(
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self,
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hidden_states: mx.array,
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) -> mx.array:
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length = hidden_states.shape[1]
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n_frames = (length - self.kernel_size + self.padding_total) / self.stride + 1
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n_frames = int(math.ceil(n_frames)) - 1
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ideal_length = n_frames * self.stride + self.kernel_size - self.padding_total
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return ideal_length - length
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def _pad1d(
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self,
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hidden_states: mx.array,
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paddings: Tuple[int, int],
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mode: str = "zero",
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value: float = 0.0,
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):
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if mode != "reflect":
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return mx.pad(
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hidden_states, paddings, mode="constant", constant_values=value
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)
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length = hidden_states.shape[1]
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prefix = hidden_states[:, 1 : paddings[0] + 1][:, ::-1]
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suffix = hidden_states[:, max(length - (paddings[1] + 1), 0) : -1][:, ::-1]
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return mx.concatenate([prefix, hidden_states, suffix], axis=1)
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def __call__(self, hidden_states):
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extra_padding = self._get_extra_padding_for_conv1d(hidden_states)
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if self.causal:
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# Left padding for causal
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hidden_states = self._pad1d(
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hidden_states, (self.padding_total, extra_padding), mode=self.pad_mode
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)
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else:
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# Asymmetric padding required for odd strides
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padding_right = self.padding_total // 2
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padding_left = self.padding_total - padding_right
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hidden_states = self._pad1d(
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hidden_states,
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(padding_left, padding_right + extra_padding),
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mode=self.pad_mode,
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)
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hidden_states = self.conv(hidden_states)
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if self.norm_type == "time_group_norm":
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hidden_states = self.norm(hidden_states)
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return hidden_states
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class EncodecConvTranspose1d(nn.Module):
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"""ConvTranspose1d with asymmetric or causal padding and normalization."""
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def __init__(
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self,
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config,
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in_channels: int,
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out_channels: int,
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kernel_size: int,
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stride: int = 1,
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):
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super().__init__()
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self.causal = config.use_causal_conv
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self.trim_right_ratio = config.trim_right_ratio
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self.norm_type = config.norm_type
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self.conv = nn.ConvTranspose1d(in_channels, out_channels, kernel_size, stride)
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if config.norm_type == "time_group_norm":
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self.norm = nn.GroupNorm(1, out_channels, pytorch_compatible=True)
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self.padding_total = kernel_size - stride
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def __call__(self, hidden_states):
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hidden_states = self.conv(hidden_states)
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if self.norm_type == "time_group_norm":
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hidden_states = self.norm(hidden_states)
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if self.causal:
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padding_right = math.ceil(self.padding_total * self.trim_right_ratio)
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else:
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padding_right = self.padding_total // 2
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padding_left = self.padding_total - padding_right
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end = hidden_states.shape[1] - padding_right
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hidden_states = hidden_states[:, padding_left:end, :]
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return hidden_states
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class EncodecLSTM(nn.Module):
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def __init__(self, config, dimension):
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super().__init__()
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self.lstm = [LSTM(dimension, dimension) for _ in range(config.num_lstm_layers)]
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def __call__(self, hidden_states):
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h = hidden_states
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for lstm in self.lstm:
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h = lstm(h)
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return h + hidden_states
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class EncodecResnetBlock(nn.Module):
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"""
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Residual block from SEANet model as used by EnCodec.
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"""
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def __init__(self, config, dim: int, dilations: List[int]):
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super().__init__()
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kernel_sizes = (config.residual_kernel_size, 1)
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if len(kernel_sizes) != len(dilations):
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raise ValueError("Number of kernel sizes should match number of dilations")
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hidden = dim // config.compress
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block = []
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for i, (kernel_size, dilation) in enumerate(zip(kernel_sizes, dilations)):
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in_chs = dim if i == 0 else hidden
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out_chs = dim if i == len(kernel_sizes) - 1 else hidden
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block += [nn.ELU()]
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block += [
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EncodecConv1d(config, in_chs, out_chs, kernel_size, dilation=dilation)
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]
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self.block = block
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if getattr(config, "use_conv_shortcut", True):
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self.shortcut = EncodecConv1d(config, dim, dim, kernel_size=1)
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else:
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self.shortcut = nn.Identity()
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def __call__(self, hidden_states):
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residual = hidden_states
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for layer in self.block:
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hidden_states = layer(hidden_states)
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return self.shortcut(residual) + hidden_states
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class EncodecEncoder(nn.Module):
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"""SEANet encoder as used by EnCodec."""
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def __init__(self, config):
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super().__init__()
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model = [
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EncodecConv1d(
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config, config.audio_channels, config.num_filters, config.kernel_size
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)
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]
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scaling = 1
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for ratio in reversed(config.upsampling_ratios):
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current_scale = scaling * config.num_filters
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for j in range(config.num_residual_layers):
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model += [
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EncodecResnetBlock(
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config, current_scale, [config.dilation_growth_rate**j, 1]
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)
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]
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model += [nn.ELU()]
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model += [
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EncodecConv1d(
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config,
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current_scale,
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current_scale * 2,
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kernel_size=ratio * 2,
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stride=ratio,
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)
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]
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scaling *= 2
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model += [EncodecLSTM(config, scaling * config.num_filters)]
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model += [nn.ELU()]
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model += [
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EncodecConv1d(
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config,
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scaling * config.num_filters,
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config.hidden_size,
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config.last_kernel_size,
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)
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]
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self.layers = model
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def __call__(self, hidden_states):
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for layer in self.layers:
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hidden_states = layer(hidden_states)
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return hidden_states
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class EncodecDecoder(nn.Module):
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"""SEANet decoder as used by EnCodec."""
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def __init__(self, config):
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super().__init__()
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scaling = int(2 ** len(config.upsampling_ratios))
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model = [
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EncodecConv1d(
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config,
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config.hidden_size,
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scaling * config.num_filters,
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config.kernel_size,
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)
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]
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model += [EncodecLSTM(config, scaling * config.num_filters)]
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for ratio in config.upsampling_ratios:
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current_scale = scaling * config.num_filters
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model += [nn.ELU()]
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model += [
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EncodecConvTranspose1d(
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config,
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current_scale,
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current_scale // 2,
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kernel_size=ratio * 2,
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stride=ratio,
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)
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]
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for j in range(config.num_residual_layers):
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model += [
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EncodecResnetBlock(
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config, current_scale // 2, (config.dilation_growth_rate**j, 1)
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)
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]
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scaling //= 2
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model += [nn.ELU()]
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model += [
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EncodecConv1d(
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config,
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config.num_filters,
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config.audio_channels,
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config.last_kernel_size,
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)
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]
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self.layers = model
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def __call__(self, hidden_states):
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for layer in self.layers:
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hidden_states = layer(hidden_states)
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return hidden_states
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class EncodecEuclideanCodebook(nn.Module):
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"""Codebook with Euclidean distance."""
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def __init__(self, config):
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super().__init__()
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self.embed = mx.zeros((config.codebook_size, config.codebook_dim))
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def quantize(self, hidden_states):
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embed = self.embed.T
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scaled_states = hidden_states.square().sum(axis=1, keepdims=True)
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dist = -(
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scaled_states
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- 2 * hidden_states @ embed
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+ embed.square().sum(axis=0, keepdims=True)
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)
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embed_ind = dist.argmax(axis=-1)
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return embed_ind
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def encode(self, hidden_states):
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shape = hidden_states.shape
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hidden_states = hidden_states.reshape((-1, shape[-1]))
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embed_ind = self.quantize(hidden_states)
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embed_ind = embed_ind.reshape(*shape[:-1])
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return embed_ind
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def decode(self, embed_ind):
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return self.embed[embed_ind]
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class EncodecVectorQuantization(nn.Module):
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"""
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Vector quantization implementation. Currently supports only euclidean distance.
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"""
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def __init__(self, config):
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super().__init__()
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self.codebook = EncodecEuclideanCodebook(config)
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def encode(self, hidden_states):
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return self.codebook.encode(hidden_states)
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def decode(self, embed_ind):
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return self.codebook.decode(embed_ind)
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class EncodecResidualVectorQuantizer(nn.Module):
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"""Residual Vector Quantizer."""
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def __init__(self, config):
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super().__init__()
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self.codebook_size = config.codebook_size
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hop_length = np.prod(config.upsampling_ratios)
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self.frame_rate = math.ceil(config.sampling_rate / hop_length)
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self.num_quantizers = int(
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1000 * config.target_bandwidths[-1] // (self.frame_rate * 10)
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)
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self.layers = [
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EncodecVectorQuantization(config) for _ in range(self.num_quantizers)
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]
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def get_num_quantizers_for_bandwidth(
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self, bandwidth: Optional[float] = None
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) -> int:
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"""Return num_quantizers based on specified target bandwidth."""
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bw_per_q = math.log2(self.codebook_size) * self.frame_rate
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num_quantizers = self.num_quantizers
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if bandwidth is not None and bandwidth > 0.0:
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num_quantizers = int(max(1, math.floor(bandwidth * 1000 / bw_per_q)))
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return num_quantizers
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def encode(
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self, embeddings: mx.array, bandwidth: Optional[float] = None
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) -> mx.array:
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"""
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Encode a given input array with the specified frame rate at the given
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bandwidth. The RVQ encode method sets the appropriate number of
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quantizers to use and returns indices for each quantizer.
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"""
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num_quantizers = self.get_num_quantizers_for_bandwidth(bandwidth)
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residual = embeddings
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all_indices = []
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for layer in self.layers[:num_quantizers]:
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indices = layer.encode(residual)
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quantized = layer.decode(indices)
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residual = residual - quantized
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all_indices.append(indices)
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out_indices = mx.stack(all_indices, axis=1)
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return out_indices
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def decode(self, codes: mx.array) -> mx.array:
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"""Decode the given codes to the quantized representation."""
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quantized_out = None
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for i, indices in enumerate(codes.split(codes.shape[1], axis=1)):
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layer = self.layers[i]
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quantized = layer.decode(indices.squeeze(1))
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if quantized_out is None:
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quantized_out = quantized
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else:
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quantized_out = quantized + quantized_out
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return quantized_out
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|
|
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class EncodecModel(nn.Module):
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def __init__(self, config):
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super().__init__()
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self.config = config
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self.encoder = EncodecEncoder(config)
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self.decoder = EncodecDecoder(config)
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self.quantizer = EncodecResidualVectorQuantizer(config)
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def _encode_frame(
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self, input_values: mx.array, bandwidth: float, padding_mask: mx.array
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) -> Tuple[mx.array, Optional[mx.array]]:
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"""
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Encodes the given input using the underlying VQVAE.
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"""
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length = input_values.shape[1]
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duration = length / self.config.sampling_rate
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if (
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self.config.chunk_length_s is not None
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and duration > 1e-5 + self.config.chunk_length_s
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):
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raise RuntimeError(
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f"Duration of frame ({duration}) is longer than chunk {self.config.chunk_length_s}"
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)
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scale = None
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if self.config.normalize:
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# if the padding is non zero
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input_values = input_values * padding_mask[..., None]
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mono = mx.sum(input_values, axis=2, keepdims=True) / input_values.shape[2]
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scale = mono.square().mean(axis=1, keepdims=True).sqrt() + 1e-8
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input_values = input_values / scale
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embeddings = self.encoder(input_values)
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codes = self.quantizer.encode(embeddings, bandwidth)
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return codes, scale
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def encode(
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self,
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input_values: mx.array,
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padding_mask: mx.array = None,
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bandwidth: Optional[float] = None,
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) -> Tuple[mx.array, Optional[mx.array]]:
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"""
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Encodes the input audio waveform into discrete codes.
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Args:
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input_values (mx.array): The input audio waveform with shape
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``(batch_size, channels, sequence_length)``.
|
|
padding_mask (mx.array): Padding mask used to pad the ``input_values``.
|
|
bandwidth (float, optional): The target bandwidth. Must be one of
|
|
``config.target_bandwidths``. If ``None``, uses the smallest
|
|
possible bandwidth. bandwidth is represented as a thousandth of
|
|
what it is, e.g. 6kbps bandwidth is represented as bandwidth == 6.0
|
|
|
|
Returns:
|
|
A list of frames containing the discrete encoded codes for the
|
|
input audio waveform, along with rescaling factors for each chunk
|
|
when ``config.normalize==True``. Each frame is a tuple ``(codebook,
|
|
scale)``, with ``codebook`` of shape ``(batch_size, num_codebooks,
|
|
frames)``.
|
|
"""
|
|
|
|
if bandwidth is None:
|
|
bandwidth = self.config.target_bandwidths[0]
|
|
if bandwidth not in self.config.target_bandwidths:
|
|
raise ValueError(
|
|
f"This model doesn't support the bandwidth {bandwidth}. "
|
|
f"Select one of {self.config.target_bandwidths}."
|
|
)
|
|
|
|
_, input_length, channels = input_values.shape
|
|
|
|
if channels < 1 or channels > 2:
|
|
raise ValueError(
|
|
f"Number of audio channels must be 1 or 2, but got {channels}"
|
|
)
|
|
|
|
chunk_length = self.chunk_length
|
|
if chunk_length is None:
|
|
chunk_length = input_length
|
|
stride = input_length
|
|
else:
|
|
stride = self.chunk_stride
|
|
|
|
if padding_mask is None:
|
|
padding_mask = mx.ones(input_values.shape[:2], dtype=mx.bool_)
|
|
encoded_frames = []
|
|
scales = []
|
|
|
|
step = chunk_length - stride
|
|
if (input_length % stride) != step:
|
|
raise ValueError(
|
|
"The input length is not properly padded for batched chunked "
|
|
"encoding. Make sure to pad the input correctly."
|
|
)
|
|
|
|
for offset in range(0, input_length - step, stride):
|
|
mask = padding_mask[:, offset : offset + chunk_length].astype(mx.bool_)
|
|
frame = input_values[:, offset : offset + chunk_length]
|
|
encoded_frame, scale = self._encode_frame(frame, bandwidth, mask)
|
|
encoded_frames.append(encoded_frame)
|
|
scales.append(scale)
|
|
|
|
encoded_frames = mx.stack(encoded_frames)
|
|
|
|
return (encoded_frames, scales)
|
|
|
|
@staticmethod
|
|
def _linear_overlap_add(frames: List[mx.array], stride: int):
|
|
if len(frames) == 0:
|
|
raise ValueError("`frames` cannot be an empty list.")
|
|
|
|
dtype = frames[0].dtype
|
|
N, frame_length, C = frames[0].shape
|
|
total_size = stride * (len(frames) - 1) + frames[-1].shape[1]
|
|
|
|
time_vec = mx.linspace(0, 1, frame_length + 2, dtype=dtype)[1:-1]
|
|
weight = 0.5 - (time_vec - 0.5).abs()
|
|
|
|
weight = weight[:, None]
|
|
sum_weight = mx.zeros((total_size, 1), dtype=dtype)
|
|
out = mx.zeros((N, total_size, C), dtype=dtype)
|
|
offset = 0
|
|
|
|
for frame in frames:
|
|
frame_length = frame.shape[1]
|
|
out[:, offset : offset + frame_length] += weight[:frame_length] * frame
|
|
sum_weight[offset : offset + frame_length] += weight[:frame_length]
|
|
offset += stride
|
|
|
|
return out / sum_weight
|
|
|
|
def _decode_frame(
|
|
self, codes: mx.array, scale: Optional[mx.array] = None
|
|
) -> mx.array:
|
|
embeddings = self.quantizer.decode(codes)
|
|
outputs = self.decoder(embeddings)
|
|
if scale is not None:
|
|
outputs = outputs * scale
|
|
return outputs
|
|
|
|
@property
|
|
def channels(self):
|
|
return self.config.audio_channels
|
|
|
|
@property
|
|
def sampling_rate(self):
|
|
return self.config.sampling_rate
|
|
|
|
@property
|
|
def chunk_length(self):
|
|
if self.config.chunk_length_s is None:
|
|
return None
|
|
else:
|
|
return int(self.config.chunk_length_s * self.config.sampling_rate)
|
|
|
|
@property
|
|
def chunk_stride(self):
|
|
if self.config.chunk_length_s is None or self.config.overlap is None:
|
|
return None
|
|
else:
|
|
return max(1, int((1.0 - self.config.overlap) * self.chunk_length))
|
|
|
|
def decode(
|
|
self,
|
|
audio_codes: mx.array,
|
|
audio_scales: Union[mx.array, List[mx.array]],
|
|
padding_mask: Optional[mx.array] = None,
|
|
) -> Tuple[mx.array, mx.array]:
|
|
"""
|
|
Decodes the given frames into an output audio waveform.
|
|
|
|
Note that the output might be a bit bigger than the input. In that
|
|
case, any extra steps at the end should be trimmed.
|
|
|
|
Args:
|
|
audio_codes (mx.array): Discret code embeddings of shape
|
|
``(batch_size, nb_chunks, chunk_length)``.
|
|
audio_scales (mx.array): Scaling factor for each input.
|
|
padding_mask (mx.array): Padding mask.
|
|
"""
|
|
chunk_length = self.chunk_length
|
|
if chunk_length is None:
|
|
if audio_codes.shape[1] != 1:
|
|
raise ValueError(f"Expected one frame, got {len(audio_codes)}")
|
|
audio_values = self._decode_frame(audio_codes[:, 0], audio_scales[0])
|
|
else:
|
|
decoded_frames = []
|
|
|
|
for frame, scale in zip(audio_codes, audio_scales):
|
|
frames = self._decode_frame(frame, scale)
|
|
decoded_frames.append(frames)
|
|
|
|
audio_values = self._linear_overlap_add(
|
|
decoded_frames, self.chunk_stride or 1
|
|
)
|
|
|
|
# truncate based on padding mask
|
|
if padding_mask is not None and padding_mask.shape[1] < audio_values.shape[1]:
|
|
audio_values = audio_values[:, : padding_mask.shape[1]]
|
|
return audio_values
|
|
|
|
@classmethod
|
|
def from_pretrained(cls, path_or_repo: str):
|
|
from huggingface_hub import snapshot_download
|
|
|
|
path = Path(path_or_repo)
|
|
if not path.exists():
|
|
path = Path(
|
|
snapshot_download(
|
|
repo_id=path_or_repo,
|
|
allow_patterns=["*.json", "*.safetensors", "*.model"],
|
|
)
|
|
)
|
|
|
|
with open(path / "config.json", "r") as f:
|
|
config = SimpleNamespace(**json.load(f))
|
|
|
|
model = EncodecModel(config)
|
|
model.load_weights(str(path / "model.safetensors"))
|
|
processor = functools.partial(
|
|
preprocess_audio,
|
|
sampling_rate=config.sampling_rate,
|
|
chunk_length=model.chunk_length,
|
|
chunk_stride=model.chunk_stride,
|
|
)
|
|
mx.eval(model)
|
|
return model, processor
|
|
|
|
|
|
def preprocess_audio(
|
|
raw_audio: Union[mx.array, List[mx.array]],
|
|
sampling_rate: int = 24000,
|
|
chunk_length: Optional[int] = None,
|
|
chunk_stride: Optional[int] = None,
|
|
):
|
|
r"""
|
|
Prepare inputs for the EnCodec model.
|
|
|
|
Args:
|
|
raw_audio (mx.array or List[mx.array]): The sequence or batch of
|
|
sequences to be processed.
|
|
sampling_rate (int): The sampling rate at which the audio waveform
|
|
should be digitalized.
|
|
chunk_length (int, optional): The model's chunk length.
|
|
chunk_stride (int, optional): The model's chunk stride.
|
|
"""
|
|
if not isinstance(raw_audio, list):
|
|
raw_audio = [raw_audio]
|
|
|
|
raw_audio = [x[..., None] if x.ndim == 1 else x for x in raw_audio]
|
|
|
|
max_length = max(array.shape[0] for array in raw_audio)
|
|
if chunk_length is not None:
|
|
max_length += chunk_length - (max_length % chunk_stride)
|
|
|
|
inputs = []
|
|
masks = []
|
|
for x in raw_audio:
|
|
length = x.shape[0]
|
|
mask = mx.ones((length,), dtype=mx.bool_)
|
|
difference = max_length - length
|
|
if difference > 0:
|
|
mask = mx.pad(mask, (0, difference))
|
|
x = mx.pad(x, ((0, difference), (0, 0)))
|
|
inputs.append(x)
|
|
masks.append(mask)
|
|
return mx.stack(inputs), mx.stack(masks)
|