Blog Archive

Wednesday, October 2, 2024

如何在中国使用Fidelity Cash Management VISA Debit card


Fidelity Cash Management VISA Debit card  

申请链接:

https://fidelity.app.link/e/rGWgIGAznNb


特色

  1. 全球任何一家ATM取钱都免手续费!!有时ATM那边也会收取一定手续费,但是 Fidelity 会给报销掉这部分费用,也就是真的保证了一点手续费都没有!!
  2. 存的是美元,取款时取出的是当地货币,自动会按照当天汇率计算,非常方便!
  3. 零月费,没有最低存款要求。
  4. $0 incoming wire transfer fee,国内和国际汇款都是如此。
  5. $0 outgoing wire transfer fee,国内和国际汇款都是如此!这一特性相当少见,其他银行outgoing wire fee基本上都要$25-$30的!
  6. 你将可以选择把这个账户里的余额自动存入货币基金SPAXX(2024年4月写文时利率高达4.95%,这个利率基本上就是随着美联储的利率而变)。在这之前此账户的利率只有弱鸡的2.x%。SPAXX的利率已经在各种活期存款账户里算是最能打的之一了,虽然以前也能在Fidelity的brokerage账户里买到,但是毕竟checking账户用起来更方便,增加了这个feature之后让Fidelity这个账户吸引力大增!
  7. 此卡是 Visa 卡,几乎全世界的 ATM 都可以刷。

缺点

  1. 每日ATM取钱限额:$500。
  2. 第一次往里转钱可能会很麻烦,关联银行的话可能需要等一封纸质的信验证你的地址信息,所以一定要留有提前量不要临出国了再办结果最后来不及往里转钱。
  3. 往里转钱可能最终会需要3~5个工作日才能取(其实所有银行都这样 到账需要一定时间),所以出国前一定要规划好留有提前量。

中国ATM机器上使用注意事项

(2024年10月1日最新验证)
找到又VISA标志的ATM机器(比如:建行),标识有“可受理外卡取款”,插卡输密码(密码是4位数字pin码),输完4位密码直接确认就好,语言选择中文或者中英文对照都可以,账户选择“默认账户”,然后选择“取款”(不要查询,查询会报错,报错后就会自动退款),输入要取的金额,不能超过卡内可取金额,否则会失败,取款无论成功与否,持卡人都会收到提示短信


Charles Schwab Checking 银行账户:全球ATM无手续费取现的 Debit Card

操作步骤:

开设账户:schwab 并同时申请 Debit Card.

https://www.schwab.com/client-referral?refrid=REFERVBDKWAV2

开户送1000美元,且能获得在中国的ATM上无手续费去人民币现金


特色

  1. (开户奖励实际上是跟着 Brokerage Account 走的,要求存入的金额必须存入 Brokerage Account 而放在 Checking Account 的钱不计入,因此请移步至 Charles Schwab Brokerage Account 查看。)
  2. 全球任何一家ATM取钱都免手续费!有时ATM那边也会收取一定手续费,但是 Charles Schwab 会在月末时给报销掉这部分费用,也就是真的保证了一点手续费都没有!!
  3. 存的是美元,取款时取出的是当地货币,自动会按照当天汇率计算,非常方便!
  4. $0 月费,没有最低存款要求。
  5. $0 incoming wire transfer fee,适合接从国内打过来的汇款。
  6. 此卡是 Visa 卡,几乎全世界的ATM都可以刷。

缺点

  1. 申请这个账户会在 Equifax (EQ) 留下一个 Hard Pull (HP)。(Brokerage+checking 只有一个 HP。)【更新】根据最新的数据点,开Schwab的账户终于不再有hard pull了!
  2. 每日ATM取钱限额:$1000。
  3. 往里转钱可能最终会需要3~5个工作日才能取(其实所有银行都这样 到账需要一定时间),所以出国前一定要规划好留有提前量。
  4. 这个 Checking Account 是和其投资账户关联的,所以开户时必须也同时开一个投资账户。不过都是免费的,所以就开了就好了。

参考信息:
另外一张广受欢迎的卡是: Fidelity Cash Management VISA Debit card
具体申请和使用请参见:

Here is free $30 for your food

Here is the link:  

https://drd.sh/uzcj2Xh8rlu0ByUo


When you use the above link to sign up the DoorDash account, you can get $30 off ($10 off each of your first 3 orders).


And I will also get $20 in credits for my food. Thanks.


Monday, September 16, 2024

[paper+code] Conformer and Zipformer for ASR

Efficient conformer-based speech recognition with linear attention

Paper: https://arxiv.org/pdf/2104.06865

code: https://github.com/Alex2135/ASR-proto?tab=readme-ov-file


Zipformer: A faster and better encoder for automatic speech recognition

https://arxiv.org/abs/2310.11230

code: https://github.com/k2-fsa/icefall

Sunday, August 25, 2024

Transformer studying notes

 1. Paper.

https://www.kdocs.cn/l/cqkBn5wZWxXs

2. Blog:

2.1) https://blog.csdn.net/Magical_Bubble/article/details/89083225

2.2)  https://ugirc.blog.csdn.net/article/details/120394042

https://ugirc.blog.csdn.net/article/details/120394042?spm=1001.2101.3001.6661.1&utm_medium=distribute.pc_relevant_t0.none-task-blog-2%7Edefault%7ECTRLIST%7EPaidSort-1-120394042-blog-89083225.235%5Ev43%5Epc_blog_bottom_relevance_base4&depth_1-utm_source=distribute.pc_relevant_t0.none-task-blog-2%7Edefault%7ECTRLIST%7EPaidSort-1-120394042-blog-89083225.235%5Ev43%5Epc_blog_bottom_relevance_base4&utm_relevant_index=1&ydreferer=aHR0cHM6Ly9ibG9nLmNzZG4ubmV0L01hZ2ljYWxfQnViYmxlL2FydGljbGUvZGV0YWlscy84OTA4MzIyNQ%3D%3D



3. Video

https://www.bilibili.com/video/BV1vf4y1n7k2/



4. Code:

# ======================================
# === Pytorch手写Transformer完整代码
# ======================================
"""
code by Tae Hwan Jung(Jeff Jung) @graykode, Derek Miller @dmmiller612, modify by shwei
Reference: https://github.com/jadore801120/attention-is-all-you-need-pytorch
           https://github.com/JayParks/transformer
"""
# ====================================================================================================
# 数据构建
import math
import torch
import numpy as np
import torch.nn as nn
import torch.optim as optim
import torch.utils.data as Data

device = 'cpu'
# device = 'cuda'

# transformer epochs
epochs = 100
# epochs = 1000

# 这里我没有用什么大型的数据集,而是手动输入了两对中文→英语的句子
# 还有每个字的索引也是我手动硬编码上去的,主要是为了降低代码阅读难度
# S: Symbol that shows starting of decoding input
# E: Symbol that shows starting of decoding output
# P: Symbol that will fill in blank sequence if current batch data size is short than time steps

# 训练集
sentences = [
    # 中文和英语的单词个数不要求相同
    # enc_input                dec_input           dec_output
    ['我 有 一 个 好 朋 友 P', 'S i have a good friend .', 'i have a good friend . E'],
    ['我 有 零 个 女 朋 友 P', 'S i have zero girl friend .', 'i have zero girl friend . E']
]

# 测试集(希望transformer能达到的效果)
# 输入:"我 有 一 个 女 朋 友"
# 输出:"i have a girlfriend"

# 中文和英语的单词要分开建立词库
# Padding Should be Zero
src_vocab = {'P': 0, '我': 1, '有': 2, '一': 3, '个': 4, '好': 5, '朋': 6, '友': 7, '零': 8, '女': 9}
src_idx2word = {i: w for i, w in enumerate(src_vocab)}
src_vocab_size = len(src_vocab)

tgt_vocab = {'P': 0, 'i': 1, 'have': 2, 'a': 3, 'good': 4, 'friend': 5, 'zero': 6, 'girl': 7, 'S': 8, 'E': 9, '.': 10}
idx2word = {i: w for i, w in enumerate(tgt_vocab)}
tgt_vocab_size = len(tgt_vocab)

src_len = 8  # (源句子的长度)enc_input max sequence length
tgt_len = 7  # dec_input(=dec_output) max sequence length

# Transformer Parameters
d_model = 512  # Embedding Size(token embedding和position编码的维度)
d_ff = 2048  # FeedForward dimension (两次线性层中的隐藏层 512->2048->512,线性层是用来做特征提取的),当然最后会再接一个projection层
d_k = d_v = 64  # dimension of K(=Q), V(Q和K的维度需要相同,这里为了方便让K=V)
n_layers = 6  # number of Encoder of Decoder Layer(Block的个数)
n_heads = 8  # number of heads in Multi-Head Attention(有几套头)


# ==============================================================================================
# 数据构建


def make_data(sentences):
    """把单词序列转换为数字序列"""
    enc_inputs, dec_inputs, dec_outputs = [], [], []
    for i in range(len(sentences)):
        enc_input = [[src_vocab[n] for n in sentences[i][0].split()]]  # [[1, 2, 3, 4, 0], [1, 2, 3, 5, 0]]
        dec_input = [[tgt_vocab[n] for n in sentences[i][1].split()]]  # [[6, 1, 2, 3, 4, 8], [6, 1, 2, 3, 5, 8]]
        dec_output = [[tgt_vocab[n] for n in sentences[i][2].split()]]  # [[1, 2, 3, 4, 8, 7], [1, 2, 3, 5, 8, 7]]

        enc_inputs.extend(enc_input)
        dec_inputs.extend(dec_input)
        dec_outputs.extend(dec_output)

    return torch.LongTensor(enc_inputs), torch.LongTensor(dec_inputs), torch.LongTensor(dec_outputs)


enc_inputs, dec_inputs, dec_outputs = make_data(sentences)


class MyDataSet(Data.Dataset):
    """自定义DataLoader"""

    def __init__(self, enc_inputs, dec_inputs, dec_outputs):
        super(MyDataSet, self).__init__()
        self.enc_inputs = enc_inputs
        self.dec_inputs = dec_inputs
        self.dec_outputs = dec_outputs

    def __len__(self):
        return self.enc_inputs.shape[0]

    def __getitem__(self, idx):
        return self.enc_inputs[idx], self.dec_inputs[idx], self.dec_outputs[idx]


loader = Data.DataLoader(MyDataSet(enc_inputs, dec_inputs, dec_outputs), 2, True)


# ====================================================================================================
# Transformer模型

class PositionalEncoding(nn.Module):
    def __init__(self, d_model, dropout=0.1, max_len=5000):
        super(PositionalEncoding, self).__init__()
        self.dropout = nn.Dropout(p=dropout)

        pe = torch.zeros(max_len, d_model)
        position = torch.arange(0, max_len, dtype=torch.float).unsqueeze(1)
        div_term = torch.exp(torch.arange(0, d_model, 2).float() * (-math.log(10000.0) / d_model))
        pe[:, 0::2] = torch.sin(position * div_term)
        pe[:, 1::2] = torch.cos(position * div_term)
        pe = pe.unsqueeze(0).transpose(0, 1)
        self.register_buffer('pe', pe)

    def forward(self, x):
        """
        x: [seq_len, batch_size, d_model]
        """
        x = x + self.pe[:x.size(0), :]
        return self.dropout(x)


def get_attn_pad_mask(seq_q, seq_k):
    # pad mask的作用:在对value向量加权平均的时候,可以让pad对应的alpha_ij=0,这样注意力就不会考虑到pad向量
    """这里的q,k表示的是两个序列(跟注意力机制的q,k没有关系),例如encoder_inputs (x1,x2,..xm)和encoder_inputs (x1,x2..xm)
    encoder和decoder都可能调用这个函数,所以seq_len视情况而定
    seq_q: [batch_size, seq_len]
    seq_k: [batch_size, seq_len]
    seq_len could be src_len or it could be tgt_len
    seq_len in seq_q and seq_len in seq_k maybe not equal
    """
    batch_size, len_q = seq_q.size()  # 这个seq_q只是用来expand维度的
    batch_size, len_k = seq_k.size()
    # eq(zero) is PAD token
    # 例如:seq_k = [[1,2,3,4,0], [1,2,3,5,0]]
    pad_attn_mask = seq_k.data.eq(0).unsqueeze(1)  # [batch_size, 1, len_k], True is masked
    return pad_attn_mask.expand(batch_size, len_q, len_k)  # [batch_size, len_q, len_k] 构成一个立方体(batch_size个这样的矩阵)


def get_attn_subsequence_mask(seq):
    """建议打印出来看看是什么的输出(一目了然)
    seq: [batch_size, tgt_len]
    """
    attn_shape = [seq.size(0), seq.size(1), seq.size(1)]
    # attn_shape: [batch_size, tgt_len, tgt_len]
    subsequence_mask = np.triu(np.ones(attn_shape), k=1)  # 生成一个上三角矩阵
    subsequence_mask = torch.from_numpy(subsequence_mask).byte()
    return subsequence_mask  # [batch_size, tgt_len, tgt_len]


# ==========================================================================================
class ScaledDotProductAttention(nn.Module):
    def __init__(self):
        super(ScaledDotProductAttention, self).__init__()

    def forward(self, Q, K, V, attn_mask):
        """
        Q: [batch_size, n_heads, len_q, d_k]
        K: [batch_size, n_heads, len_k, d_k]
        V: [batch_size, n_heads, len_v(=len_k), d_v]
        attn_mask: [batch_size, n_heads, seq_len, seq_len]
        说明:在encoder-decoder的Attention层中len_q(q1,..qt)和len_k(k1,...km)可能不同
        """
        scores = torch.matmul(Q, K.transpose(-1, -2)) / np.sqrt(d_k)  # scores : [batch_size, n_heads, len_q, len_k]
        # mask矩阵填充scores(用-1e9填充scores中与attn_mask中值为1位置相对应的元素)
        scores.masked_fill_(attn_mask, -1e9)  # Fills elements of self tensor with value where mask is True.

        attn = nn.Softmax(dim=-1)(scores)  # 对最后一个维度(v)做softmax
        # scores : [batch_size, n_heads, len_q, len_k] * V: [batch_size, n_heads, len_v(=len_k), d_v]
        context = torch.matmul(attn, V)  # context: [batch_size, n_heads, len_q, d_v]
        # context:[[z1,z2,...],[...]]向量, attn注意力稀疏矩阵(用于可视化的)
        return context, attn


class MultiHeadAttention(nn.Module):
    """这个Attention类可以实现:
    Encoder的Self-Attention
    Decoder的Masked Self-Attention
    Encoder-Decoder的Attention
    输入:seq_len x d_model
    输出:seq_len x d_model
    """
    def __init__(self):
        super(MultiHeadAttention, self).__init__()
        self.W_Q = nn.Linear(d_model, d_k * n_heads, bias=False)  # q,k必须维度相同,不然无法做点积
        self.W_K = nn.Linear(d_model, d_k * n_heads, bias=False)
        self.W_V = nn.Linear(d_model, d_v * n_heads, bias=False)
        # 这个全连接层可以保证多头attention的输出仍然是seq_len x d_model
        self.fc = nn.Linear(n_heads * d_v, d_model, bias=False)

    def forward(self, input_Q, input_K, input_V, attn_mask):
        """
        input_Q: [batch_size, len_q, d_model]
        input_K: [batch_size, len_k, d_model]
        input_V: [batch_size, len_v(=len_k), d_model]
        attn_mask: [batch_size, seq_len, seq_len]
        """
        residual, batch_size = input_Q, input_Q.size(0)
        # 下面的多头的参数矩阵是放在一起做线性变换的,然后再拆成多个头,这是工程实现的技巧
        # B: batch_size, S:seq_len, D: dim
        # (B, S, D) -proj-> (B, S, D_new) -split-> (B, S, Head, W) -trans-> (B, Head, S, W)
        #           线性变换               拆成多头

        # Q: [batch_size, n_heads, len_q, d_k]
        Q = self.W_Q(input_Q).view(batch_size, -1, n_heads, d_k).transpose(1, 2)
        # K: [batch_size, n_heads, len_k, d_k] # K和V的长度一定相同,维度可以不同
        K = self.W_K(input_K).view(batch_size, -1, n_heads, d_k).transpose(1, 2)
        # V: [batch_size, n_heads, len_v(=len_k), d_v]
        V = self.W_V(input_V).view(batch_size, -1, n_heads, d_v).transpose(1, 2)

        # 因为是多头,所以mask矩阵要扩充成4维的
        # attn_mask: [batch_size, seq_len, seq_len] -> [batch_size, n_heads, seq_len, seq_len]
        attn_mask = attn_mask.unsqueeze(1).repeat(1, n_heads, 1, 1)

        # context: [batch_size, n_heads, len_q, d_v], attn: [batch_size, n_heads, len_q, len_k]
        context, attn = ScaledDotProductAttention()(Q, K, V, attn_mask)
        # 下面将不同头的输出向量拼接在一起
        # context: [batch_size, n_heads, len_q, d_v] -> [batch_size, len_q, n_heads * d_v]
        context = context.transpose(1, 2).reshape(batch_size, -1, n_heads * d_v)

        # 这个全连接层可以保证多头attention的输出仍然是seq_len x d_model
        output = self.fc(context)  # [batch_size, len_q, d_model]
        return nn.LayerNorm(d_model).to(device)(output + residual), attn


# Pytorch中的Linear只会对最后一维操作,所以正好是我们希望的每个位置用同一个全连接网络
class PoswiseFeedForwardNet(nn.Module):
    def __init__(self):
        super(PoswiseFeedForwardNet, self).__init__()
        self.fc = nn.Sequential(
            nn.Linear(d_model, d_ff, bias=False),
            nn.ReLU(),
            nn.Linear(d_ff, d_model, bias=False)
        )

    def forward(self, inputs):
        """
        inputs: [batch_size, seq_len, d_model]
        """
        residual = inputs
        output = self.fc(inputs)
        return nn.LayerNorm(d_model).to(device)(output + residual)  # [batch_size, seq_len, d_model]


class EncoderLayer(nn.Module):
    def __init__(self):
        super(EncoderLayer, self).__init__()
        self.enc_self_attn = MultiHeadAttention()
        self.pos_ffn = PoswiseFeedForwardNet()

    def forward(self, enc_inputs, enc_self_attn_mask):
        """E
        enc_inputs: [batch_size, src_len, d_model]
        enc_self_attn_mask: [batch_size, src_len, src_len]  mask矩阵(pad mask or sequence mask)
        """
        # enc_outputs: [batch_size, src_len, d_model], attn: [batch_size, n_heads, src_len, src_len]
        # 第一个enc_inputs * W_Q = Q
        # 第二个enc_inputs * W_K = K
        # 第三个enc_inputs * W_V = V
        enc_outputs, attn = self.enc_self_attn(enc_inputs, enc_inputs, enc_inputs,
                                               enc_self_attn_mask)  # enc_inputs to same Q,K,V(未线性变换前)
        enc_outputs = self.pos_ffn(enc_outputs)
        # enc_outputs: [batch_size, src_len, d_model]
        return enc_outputs, attn


class DecoderLayer(nn.Module):
    def __init__(self):
        super(DecoderLayer, self).__init__()
        self.dec_self_attn = MultiHeadAttention()
        self.dec_enc_attn = MultiHeadAttention()
        self.pos_ffn = PoswiseFeedForwardNet()

    def forward(self, dec_inputs, enc_outputs, dec_self_attn_mask, dec_enc_attn_mask):
        """
        dec_inputs: [batch_size, tgt_len, d_model]
        enc_outputs: [batch_size, src_len, d_model]
        dec_self_attn_mask: [batch_size, tgt_len, tgt_len]
        dec_enc_attn_mask: [batch_size, tgt_len, src_len]
        """
        # dec_outputs: [batch_size, tgt_len, d_model], dec_self_attn: [batch_size, n_heads, tgt_len, tgt_len]
        dec_outputs, dec_self_attn = self.dec_self_attn(dec_inputs, dec_inputs, dec_inputs,
                                                        dec_self_attn_mask)  # 这里的Q,K,V全是Decoder自己的输入
        # dec_outputs: [batch_size, tgt_len, d_model], dec_enc_attn: [batch_size, h_heads, tgt_len, src_len]
        dec_outputs, dec_enc_attn = self.dec_enc_attn(dec_outputs, enc_outputs, enc_outputs,
                                                      dec_enc_attn_mask)  # Attention层的Q(来自decoder) 和 K,V(来自encoder)
        dec_outputs = self.pos_ffn(dec_outputs)  # [batch_size, tgt_len, d_model]
        return dec_outputs, dec_self_attn, dec_enc_attn  # dec_self_attn, dec_enc_attn这两个是为了可视化的


class Encoder(nn.Module):
    def __init__(self):
        super(Encoder, self).__init__()
        self.src_emb = nn.Embedding(src_vocab_size, d_model)  # token Embedding
        self.pos_emb = PositionalEncoding(d_model)  # Transformer中位置编码时固定的,不需要学习
        self.layers = nn.ModuleList([EncoderLayer() for _ in range(n_layers)])

    def forward(self, enc_inputs):
        """
        enc_inputs: [batch_size, src_len]
        """
        enc_outputs = self.src_emb(enc_inputs)  # [batch_size, src_len, d_model]
        enc_outputs = self.pos_emb(enc_outputs.transpose(0, 1)).transpose(0, 1)  # [batch_size, src_len, d_model]
        # Encoder输入序列的pad mask矩阵
        enc_self_attn_mask = get_attn_pad_mask(enc_inputs, enc_inputs)  # [batch_size, src_len, src_len]
        enc_self_attns = []  # 在计算中不需要用到,它主要用来保存你接下来返回的attention的值(这个主要是为了你画热力图等,用来看各个词之间的关系
        for layer in self.layers:  # for循环访问nn.ModuleList对象
            # 上一个block的输出enc_outputs作为当前block的输入
            # enc_outputs: [batch_size, src_len, d_model], enc_self_attn: [batch_size, n_heads, src_len, src_len]
            enc_outputs, enc_self_attn = layer(enc_outputs,
                                               enc_self_attn_mask)  # 传入的enc_outputs其实是input,传入mask矩阵是因为你要做self attention
            enc_self_attns.append(enc_self_attn)  # 这个只是为了可视化
        return enc_outputs, enc_self_attns


class Decoder(nn.Module):
    def __init__(self):
        super(Decoder, self).__init__()
        self.tgt_emb = nn.Embedding(tgt_vocab_size, d_model)  # Decoder输入的embed词表
        self.pos_emb = PositionalEncoding(d_model)
        self.layers = nn.ModuleList([DecoderLayer() for _ in range(n_layers)])  # Decoder的blocks

    def forward(self, dec_inputs, enc_inputs, enc_outputs):
        """
        dec_inputs: [batch_size, tgt_len]
        enc_inputs: [batch_size, src_len]
        enc_outputs: [batch_size, src_len, d_model]   # 用在Encoder-Decoder Attention层
        """
        dec_outputs = self.tgt_emb(dec_inputs)  # [batch_size, tgt_len, d_model]
        dec_outputs = self.pos_emb(dec_outputs.transpose(0, 1)).transpose(0, 1).to(
            device)  # [batch_size, tgt_len, d_model]
        # Decoder输入序列的pad mask矩阵(这个例子中decoder是没有加pad的,实际应用中都是有pad填充的)
        dec_self_attn_pad_mask = get_attn_pad_mask(dec_inputs, dec_inputs).to(device)  # [batch_size, tgt_len, tgt_len]
        # Masked Self_Attention:当前时刻是看不到未来的信息的
        dec_self_attn_subsequence_mask = get_attn_subsequence_mask(dec_inputs).to(
            device)  # [batch_size, tgt_len, tgt_len]

        # Decoder中把两种mask矩阵相加(既屏蔽了pad的信息,也屏蔽了未来时刻的信息)
        dec_self_attn_mask = torch.gt((dec_self_attn_pad_mask + dec_self_attn_subsequence_mask),
                                      0).to(device)  # [batch_size, tgt_len, tgt_len]; torch.gt比较两个矩阵的元素,大于则返回1,否则返回0

        # 这个mask主要用于encoder-decoder attention层
        # get_attn_pad_mask主要是enc_inputs的pad mask矩阵(因为enc是处理K,V的,求Attention时是用v1,v2,..vm去加权的,要把pad对应的v_i的相关系数设为0,这样注意力就不会关注pad向量)
        #                       dec_inputs只是提供expand的size的
        dec_enc_attn_mask = get_attn_pad_mask(dec_inputs, enc_inputs)  # [batc_size, tgt_len, src_len]

        dec_self_attns, dec_enc_attns = [], []
        for layer in self.layers:
            # dec_outputs: [batch_size, tgt_len, d_model], dec_self_attn: [batch_size, n_heads, tgt_len, tgt_len], dec_enc_attn: [batch_size, h_heads, tgt_len, src_len]
            # Decoder的Block是上一个Block的输出dec_outputs(变化)和Encoder网络的输出enc_outputs(固定)
            dec_outputs, dec_self_attn, dec_enc_attn = layer(dec_outputs, enc_outputs, dec_self_attn_mask,
                                                             dec_enc_attn_mask)
            dec_self_attns.append(dec_self_attn)
            dec_enc_attns.append(dec_enc_attn)
        # dec_outputs: [batch_size, tgt_len, d_model]
        return dec_outputs, dec_self_attns, dec_enc_attns


class Transformer(nn.Module):
    def __init__(self):
        super(Transformer, self).__init__()
        self.encoder = Encoder().to(device)
        self.decoder = Decoder().to(device)
        self.projection = nn.Linear(d_model, tgt_vocab_size, bias=False).to(device)

    def forward(self, enc_inputs, dec_inputs):
        """Transformers的输入:两个序列
        enc_inputs: [batch_size, src_len]
        dec_inputs: [batch_size, tgt_len]
        """
        # tensor to store decoder outputs
        # outputs = torch.zeros(batch_size, tgt_len, tgt_vocab_size).to(self.device)

        # enc_outputs: [batch_size, src_len, d_model], enc_self_attns: [n_layers, batch_size, n_heads, src_len, src_len]
        # 经过Encoder网络后,得到的输出还是[batch_size, src_len, d_model]
        enc_outputs, enc_self_attns = self.encoder(enc_inputs)
        # dec_outputs: [batch_size, tgt_len, d_model], dec_self_attns: [n_layers, batch_size, n_heads, tgt_len, tgt_len], dec_enc_attn: [n_layers, batch_size, tgt_len, src_len]
        dec_outputs, dec_self_attns, dec_enc_attns = self.decoder(dec_inputs, enc_inputs, enc_outputs)
        # dec_outputs: [batch_size, tgt_len, d_model] -> dec_logits: [batch_size, tgt_len, tgt_vocab_size]
        dec_logits = self.projection(dec_outputs)
        return dec_logits.view(-1, dec_logits.size(-1)), enc_self_attns, dec_self_attns, dec_enc_attns


model = Transformer().to(device)
# 这里的损失函数里面设置了一个参数 ignore_index=0,因为 "pad" 这个单词的索引为 0,这样设置以后,就不会计算 "pad" 的损失(因为本来 "pad" 也没有意义,不需要计算)
criterion = nn.CrossEntropyLoss(ignore_index=0)
optimizer = optim.SGD(model.parameters(), lr=1e-3, momentum=0.99)  # 用adam的话效果不好

# ====================================================================================================
for epoch in range(epochs):
    for enc_inputs, dec_inputs, dec_outputs in loader:
        """
        enc_inputs: [batch_size, src_len]
        dec_inputs: [batch_size, tgt_len]
        dec_outputs: [batch_size, tgt_len]
        """
        enc_inputs, dec_inputs, dec_outputs = enc_inputs.to(device), dec_inputs.to(device), dec_outputs.to(device)
        # outputs: [batch_size * tgt_len, tgt_vocab_size]
        outputs, enc_self_attns, dec_self_attns, dec_enc_attns = model(enc_inputs, dec_inputs)
        loss = criterion(outputs, dec_outputs.view(-1))  # dec_outputs.view(-1):[batch_size * tgt_len * tgt_vocab_size]
        print('Epoch:', '%04d' % (epoch + 1), 'loss =', '{:.6f}'.format(loss))

        optimizer.zero_grad()
        loss.backward()
        optimizer.step()


def greedy_decoder(model, enc_input, start_symbol):
    """贪心编码
    For simplicity, a Greedy Decoder is Beam search when K=1. This is necessary for inference as we don't know the
    target sequence input. Therefore we try to generate the target input word by word, then feed it into the transformer.
    Starting Reference: http://nlp.seas.harvard.edu/2018/04/03/attention.html#greedy-decoding
    :param model: Transformer Model
    :param enc_input: The encoder input
    :param start_symbol: The start symbol. In this example it is 'S' which corresponds to index 4
    :return: The target input
    """
    enc_outputs, enc_self_attns = model.encoder(enc_input)
    dec_input = torch.zeros(1, 0).type_as(enc_input.data)  # 初始化一个空的tensor: tensor([], size=(1, 0), dtype=torch.int64)
    terminal = False
    next_symbol = start_symbol
    while not terminal:
        # 预测阶段:dec_input序列会一点点变长(每次添加一个新预测出来的单词)
        dec_input = torch.cat([dec_input.to(device), torch.tensor([[next_symbol]], dtype=enc_input.dtype).to(device)],
                              -1)
        dec_outputs, _, _ = model.decoder(dec_input, enc_input, enc_outputs)
        projected = model.projection(dec_outputs)
        prob = projected.squeeze(0).max(dim=-1, keepdim=False)[1]
        # 增量更新(我们希望重复单词预测结果是一样的)
        # 我们在预测是会选择性忽略重复的预测的词,只摘取最新预测的单词拼接到输入序列中
        next_word = prob.data[-1]  # 拿出当前预测的单词(数字)。我们用x'_t对应的输出z_t去预测下一个单词的概率,不用z_1,z_2..z_{t-1}
        next_symbol = next_word
        if next_symbol == tgt_vocab["E"]:
            terminal = True
        # print(next_word)

    # greedy_dec_predict = torch.cat(
    #     [dec_input.to(device), torch.tensor([[next_symbol]], dtype=enc_input.dtype).to(device)],
    #     -1)
    greedy_dec_predict = dec_input[:, 1:]
    return greedy_dec_predict


# ==========================================================================================
# 预测阶段
# 测试集
sentences = [
    # enc_input                dec_input           dec_output
    ['我 有 零 个 女 朋 友 P', '', '']
]

enc_inputs, dec_inputs, dec_outputs = make_data(sentences)
test_loader = Data.DataLoader(MyDataSet(enc_inputs, dec_inputs, dec_outputs), 2, True)
enc_inputs, _, _ = next(iter(test_loader))

print()
print("="*30)
print("利用训练好的Transformer模型将中文句子'我 有 零 个 女 朋 友' 翻译成英文句子: ")
for i in range(len(enc_inputs)):
    greedy_dec_predict = greedy_decoder(model, enc_inputs[i].view(1, -1).to(device), start_symbol=tgt_vocab["S"])
    print(enc_inputs[i], '->', greedy_dec_predict.squeeze())
    print([src_idx2word[t.item()] for t in enc_inputs[i]], '->',
          [idx2word[n.item()] for n in greedy_dec_predict.squeeze()])


Thursday, August 1, 2024

[info] what dose it mean when asterisk (*) appear in python function signature?

In Python, an asterisk (*) in a function signature can have several meanings depending on its context. Here are the main uses:

  1. Variable Positional Arguments (*args): When a single asterisk precedes a parameter, it collects all additional positional arguments passed to the function into a tuple.


    def my_function(*args): print(args) my_function(1, 2, 3) # Output: (1, 2, 3)
  2. Keyword-Only Arguments: When an asterisk is placed in the function signature without a variable name, it indicates that all subsequent parameters must be specified as keyword arguments.


    def my_function(a, b, *, c): print(a, b, c) my_function(1, 2, c=3) # Valid my_function(1, 2, 3) # Invalid: TypeError
  3. Variable Keyword Arguments (**kwargs): When a double asterisk precedes a parameter, it collects all additional keyword arguments passed to the function into a dictionary.


    def my_function(**kwargs): print(kwargs) my_function(a=1, b=2) # Output: {'a': 1, 'b': 2}
  4. Unpacking Arguments: An asterisk can also be used in the function call to unpack a list or tuple into positional arguments, or a dictionary into keyword arguments.

    def my_function(a, b, c):
    print(a, b, c) args = (1, 2, 3) my_function(*args) # Output: 1 2 3 kwargs = {'a': 1, 'b': 2, 'c': 3} my_function(**kwargs) # Output: 1 2 3

Each of these uses helps make functions more flexible and allows for more dynamic handling of arguments.

Monday, February 19, 2024

How to build a realtime ASR with opensource Whisper

Building a real-time Automatic Speech Recognition (ASR) system using OpenAI's Whisper involves several steps, including setting up your environment, downloading Whisper models, and creating a real-time audio processing pipeline. Here's a general approach to get you started:

Step 1: Install Whisper

First, you need to ensure that Python is installed on your system. Whisper can be installed via pip. In your terminal, run:

bash
pip install git+https://github.com/openai/whisper.git

Make sure your Python environment has the necessary dependencies for Whisper.

Step 2: Real-Time Audio Processing Setup

For real-time ASR, you need to capture audio from your microphone, process it in chunks, and then feed these chunks to Whisper for transcription. Python libraries such as sounddevice and numpy can be helpful for capturing and processing audio in real-time.

bash
pip install sounddevice numpy

Step 3: Implementing Real-Time Transcription

The following Python script provides a basic structure for implementing real-time ASR with Whisper. This script captures audio from the microphone, processes it in real-time, and prints the transcriptions:

```
import sounddevice as sd
import numpy as np
import whisper

# Load Whisper model
model = whisper.load_model("base")  # You can choose different model sizes

def audio_callback(indata, frames, time, status):
    # This function will be called for each audio block captured by the microphone
    audio_np = np.array(indata[:, 0], dtype=np.float32)  # Convert to numpy array
    # Here, you could buffer the audio and then process it with Whisper
    
    # Example: Passing the audio to Whisper directly for real-time transcription
    # might not be efficient or feasible due to the model's expectations for input length and format.
    # Instead, consider buffering the audio and processing it in larger chunks.
    
    # Print the status for debugging
    print("Capturing audio...", status)

# Start streaming from microphone
with sd.InputStream(callback=audio_callback):
    sd.sleep(10000)  # Keep capturing for a certain time or implement a loop for continuous capture

```

Notes:

  • This script sets up a basic framework. You need to adapt the audio_callback function to buffer audio data and then periodically send this data to the Whisper model for transcription. Whisper typically requires a longer audio segment (several seconds) to produce accurate transcriptions, so real-time performance may vary based on how you implement buffering and processing.
  • Consider the limitations and performance of the Whisper model you choose (base, small, medium, large). Larger models offer better accuracy but require more computational resources.

Step 4: Post-Processing and Displaying Transcriptions

After receiving the transcription from Whisper, you can display it in real-time or use it as needed in your application. Be mindful of the latency involved in processing and transcription when designing your real-time ASR system.

Final Thoughts:

  • Real-time ASR with Whisper involves handling streaming audio data, which can be complex. Performance and accuracy will depend on your implementation details and the computational resources available.
  • Test with different Whisper models and audio processing strategies to find the best balance between latency and transcription accuracy for your use case.