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import numpy as np
import torch
import time

def generate_embeddings(model, tokenizer, text, bicodec, prompt_text=None, prompt_audio=None):
    """
    为 Spark LLM 生成预测所需的输入嵌入
    
    Args:
        model: Spark LLM 模型
        tokenizer: 文本分词器
        text: 要生成语音的文本
        bicodec: BiCodecTokenizer 实例
        prompt_text: 提示文本(可选)
        prompt_audio: 提示音频数组(可选)
    
    Returns:
        dict: 包含 input_embs 的字典,用于模型预测
    """
    device = next(model.parameters()).device
    
    with torch.no_grad():
        # 1. 处理提示音频,提取 global_tokens 和 semantic_tokens
        if prompt_audio is not None:
            # 确保音频数据是 float32 类型
            audio_data = np.array(prompt_audio, dtype=np.float32)
            target_sample_rate = bicodec.config['sample_rate']
            
            # 检查是否需要重采样
            # 注意:这里假设 prompt_audio 已经是从 soundfile 加载的,采样率信息在外部处理
            # BiCodecTokenizer 期望 16kHz 采样率的音频
            print(f"BiCodecTokenizer 期望的采样率: {target_sample_rate}Hz")
            print(f"音频数据形状: {audio_data.shape}")
            
            # 使用 BiCodec 提取 tokens (返回顺序: global_tokens, semantic_tokens)
            global_tokens, semantic_tokens = bicodec.tokenize(audio_data)
            global_tokens = global_tokens.squeeze(0).squeeze(0).detach().cpu().tolist()
            semantic_tokens = semantic_tokens.squeeze(0).squeeze(0).detach().cpu().tolist()
        else:
            global_tokens = []
            semantic_tokens = []
        
        # 2. 处理文本
        if prompt_text is not None:
            # 连接提示文本和目标文本
            full_text = prompt_text + text
            # 初始的 semantic tokens 等于 prompt_audio 提取的 semantic tokens
            initial_semantic_tokens = semantic_tokens.copy()
        else:
            full_text = text
            initial_semantic_tokens = []
        
        # 3. 获取文本 tokens
        text_tokens = tokenizer.encode(full_text, add_special_tokens=False)
        
        # 4. 转换为张量
        text_tokens_tensor = torch.tensor(text_tokens, dtype=torch.long, device=device)
        global_tokens_tensor = torch.tensor(global_tokens, dtype=torch.long, device=device)
        semantic_tokens_tensor = torch.tensor(initial_semantic_tokens, dtype=torch.long, device=device)
        
        # 5. 获取嵌入
        text_embs = model.text_embedder(text_tokens_tensor)
        global_embs = model.global_embedder(global_tokens_tensor)
        semantic_embs = model.model.embeddings(semantic_tokens_tensor)
        
        # 6. 获取特殊标记嵌入
        tag_0_emb = model.tts_tag_embedder(torch.tensor([0], dtype=torch.long, device=device))
        tag_1_emb = model.tts_tag_embedder(torch.tensor([1], dtype=torch.long, device=device))
        tag_2_emb = model.tts_tag_embedder(torch.tensor([2], dtype=torch.long, device=device))
        
        # 7. 连接嵌入
        input_embs = torch.cat([
            tag_2_emb, 
            text_embs, 
            tag_0_emb, 
            global_embs, 
            tag_1_emb, 
            semantic_embs
        ], dim=0)
        
        # 8. 添加批次维度
        input_embs = input_embs.unsqueeze(0)  # [1, seq_len, hidden_size]
        
        return {
            "input_embs": input_embs,
            "global_tokens": global_tokens_tensor,
        }

def generate_embeddings_batch(model, tokenizer, texts, bicodec, prompt_text=None, prompt_audio=None):
    """
    为 Spark LLM 批量生成预测所需的输入嵌入,支持多个文本的并行处理
    
    Args:
        model: Spark LLM 模型
        tokenizer: 文本分词器
        texts: 要生成语音的文本列表
        bicodec: BiCodecTokenizer 实例
        prompt_text: 提示文本(可选)
        prompt_audio: 提示音频数组(可选)
    
    Returns:
        tuple: (embeddings_dict, attention_mask) 包含批量 input_embs 的字典和注意力掩码
    """
    device = next(model.parameters()).device
    dtype = next(model.parameters()).dtype
    batch_size = len(texts)
    
    with torch.no_grad():
        # 1. 处理提示音频,提取 global_tokens 和 semantic_tokens
        if prompt_audio is not None:
            # 确保音频数据是 float32 类型
            audio_data = np.array(prompt_audio, dtype=np.float32)
            target_sample_rate = bicodec.config['sample_rate']
            
            print(f"BiCodecTokenizer 期望的采样率: {target_sample_rate}Hz")
            print(f"音频数据形状: {audio_data.shape}")
            
            # 使用 BiCodec 提取 tokens (返回顺序: global_tokens, semantic_tokens)
            global_tokens, semantic_tokens = bicodec.tokenize(audio_data)
            global_tokens = global_tokens.squeeze(0).squeeze(0).detach().cpu().tolist()
            semantic_tokens = semantic_tokens.squeeze(0).squeeze(0).detach().cpu().tolist()
        else:
            global_tokens = []
            semantic_tokens = []
        
        # 2. 处理所有文本,获取每个样本的嵌入组件
        all_input_embs = []
        all_attention_masks = []
        
        for text in texts:
            # 处理单个文本
            if prompt_text is not None:
                full_text = prompt_text + text
                initial_semantic_tokens = semantic_tokens.copy()
            else:
                full_text = text
                initial_semantic_tokens = []
            
            # 获取文本 tokens
            text_tokens = tokenizer.encode(full_text, add_special_tokens=False)
            
            # 转换为张量
            text_tokens_tensor = torch.tensor(text_tokens, dtype=torch.long, device=device)
            global_tokens_tensor = torch.tensor(global_tokens, dtype=torch.long, device=device)
            semantic_tokens_tensor = torch.tensor(initial_semantic_tokens, dtype=torch.long, device=device)
            
            # 获取嵌入
            text_embs = model.text_embedder(text_tokens_tensor)
            global_embs = model.global_embedder(global_tokens_tensor)
            semantic_embs = model.model.embeddings(semantic_tokens_tensor)
            
            # 获取特殊标记嵌入
            tag_0_emb = model.tts_tag_embedder(torch.tensor([0], dtype=torch.long, device=device))
            tag_1_emb = model.tts_tag_embedder(torch.tensor([1], dtype=torch.long, device=device))
            tag_2_emb = model.tts_tag_embedder(torch.tensor([2], dtype=torch.long, device=device))
            
            # 连接嵌入
            input_embs = torch.cat([
                tag_2_emb, 
                text_embs, 
                tag_0_emb, 
                global_embs, 
                tag_1_emb, 
                semantic_embs
            ], dim=0)  # [seq_len, hidden_size]
            
            all_input_embs.append(input_embs)
            all_attention_masks.append(torch.ones(input_embs.shape[0], dtype=torch.long, device=device))
        
        # 3. 找到最大序列长度
        max_seq_len = max(emb.shape[0] for emb in all_input_embs)
        hidden_size = all_input_embs[0].shape[1]
        
        # 4. 创建批量张量,使用 left padding 和零填充
        batch_input_embs = torch.zeros(batch_size, max_seq_len, hidden_size, device=device, dtype=dtype)
        batch_attention_mask = torch.zeros(batch_size, max_seq_len, dtype=torch.long, device=device)
        
        for i, (input_embs, attention_mask) in enumerate(zip(all_input_embs, all_attention_masks)):
            seq_len = input_embs.shape[0]
            # Left padding: 将序列放在右侧,左侧填充零
            batch_input_embs[i, -seq_len:, :] = input_embs
            batch_attention_mask[i, -seq_len:] = attention_mask
        
        # 5. 创建 global_tokens 的批量版本
        global_tokens_tensor = torch.tensor(global_tokens, dtype=torch.long, device=device, requires_grad=False)
        batch_global_tokens = global_tokens_tensor.unsqueeze(0).expand(batch_size, -1)
        
        return {
            "input_embs": batch_input_embs,
            "global_tokens": batch_global_tokens,
        }, batch_attention_mask

# Repetition Aware Sampling in VALL-E 2
def ras_sampling(weighted_scores, decoded_tokens, top_p=0.8, top_k=25, win_size=10, tau_r=0.1):
    top_ids = nucleus_sampling(weighted_scores, top_p=top_p, top_k=top_k)
    rep_num = (torch.tensor(decoded_tokens[-win_size:]).to(weighted_scores.device) == top_ids).sum().item()
    if rep_num >= win_size * tau_r:
        top_ids = random_sampling(weighted_scores)
    return top_ids


def nucleus_sampling(weighted_scores, top_p=0.8, top_k=25):
    prob, indices = [], []
    cum_prob = 0.0
    sorted_value, sorted_idx = weighted_scores.softmax(dim=0).sort(descending=True, stable=True)
    for i in range(len(sorted_idx)):
        # sampling both top-p and numbers.
        if cum_prob < top_p and len(prob) < top_k:
            cum_prob += sorted_value[i]
            prob.append(sorted_value[i])
            indices.append(sorted_idx[i])
        else:
            break
    prob = torch.tensor(prob).to(weighted_scores)
    indices = torch.tensor(indices, dtype=torch.long).to(weighted_scores.device)
    top_ids = indices[prob.multinomial(1, replacement=True)]
    return top_ids

def random_sampling(weighted_scores):
    top_ids = weighted_scores.softmax(dim=0).multinomial(1, replacement=True)
    return top_ids

def generate(model,
             inputs_embeds, 
             attention_mask,
             new_max_tokens,
             top_k,
             top_p,
             temperate,
             eos_token_id,
             pad_token_id,
             past_key_values
             ):
    """
    seperate two stages of generation:
    1. prefill
    2. decode
    we will measure the time of each stage and the total time
    """
    start_time = time.time()
    model.eval()
    batch_size = inputs_embeds.shape[0]
    decoded_tokens = [[] for _ in range(batch_size)]
    is_decoding = [True for _ in range(batch_size)]
    with torch.no_grad():
        # 1. prefill
        outputs = model.model.forward(
            attention_mask=attention_mask,
            inputs_embeds=inputs_embeds,
            past_key_values=past_key_values,
            use_cache=True,
            output_attentions=False,
            output_hidden_states=True,
            return_dict=False
        )
        hidden_states = outputs[0]
        past_key_values = outputs[1]
    prefill_time = time.time() - start_time
    tokens = attention_mask.shape[0]*attention_mask.shape[1]
    print(f"Prefill time: {prefill_time} seconds, all tokens is {tokens}, speed is {tokens/prefill_time} tokens/s ")

    # 2. decode
    start_time = time.time()
    #sampling the logits using top_k, top_p, temperature
    decoded_tokens_size = 0
    while True:
        logits = model.lm_head(hidden_states)
        last_time_decoded = []
        logits = logits[:, -1, :]
        continue_decoding = False
        for i in range(batch_size):
            if is_decoding[i]:
                logits_i = logits[i, :]
                top_ids = ras_sampling(logits_i, decoded_tokens[i], top_p=top_p, top_k=top_k).item()
                decoded_tokens[i].append(top_ids)
                last_time_decoded.append([top_ids])
                if top_ids == eos_token_id:
                    is_decoding[i] = False
                else:
                    continue_decoding = True
                decoded_tokens_size += 1
            else:
                decoded_tokens[i].append(pad_token_id)
                last_time_decoded.append([pad_token_id])
        if not continue_decoding:
            break
        last_time_decoded = torch.tensor(last_time_decoded, dtype=torch.long, device=device)
        lm_input = model.get_input_embeddings()(last_time_decoded)
        outputs = model.model.forward(
            inputs_embeds=lm_input,
            past_key_values=past_key_values,
            use_cache=True,
            output_attentions=False,
            output_hidden_states=True,
            return_dict=False
        )
        hidden_states = outputs[0]
        past_key_values = outputs[1]
    decode_time = time.time() - start_time
    print(f"Decode time: {decode_time} seconds, all tokens is {decoded_tokens_size}, speed is {decoded_tokens_size/decode_time} tokens/s ")
    print(f"decoded_tokens: {decoded_tokens}")
    return decoded_tokens, past_key_values

if __name__ == "__main__":
    import os
    import sys
    current_dir = os.path.dirname(os.path.abspath(__file__))
    print('add current dir to sys.path', current_dir)
    sys.path.append(current_dir)
    device = 'cuda:2'
    from sparktts.models.audio_tokenizer import BiCodecTokenizer
    from transformers import AutoTokenizer, AutoModelForCausalLM 
    import soundfile as sf
    audio_tokenizer = BiCodecTokenizer(model_dir=current_dir, device=device)

    print(audio_tokenizer)

    tokenizer = AutoTokenizer.from_pretrained(current_dir, trust_remote_code=True)
    model = AutoModelForCausalLM.from_pretrained(current_dir, trust_remote_code=True)
    print(tokenizer)
    print(model)

    model = model.bfloat16().to(device)
    model.eval()
    prompt_text = "我们并不是通过物理移动手段找到星河的。"
    prompt_audio_file = os.path.join(current_dir, 'kafka.wav')
    prompt_audio, sampling_rate = sf.read(prompt_audio_file)

    print(f"Loaded prompt audio from {prompt_audio_file}")
    print(f"Original sampling rate: {sampling_rate}Hz")
    print(f"Audio shape: {prompt_audio.shape}")
    target_sample_rate = audio_tokenizer.config['sample_rate']
    if sampling_rate != target_sample_rate:
        print(f"Resampling from {sampling_rate}Hz to {target_sample_rate}Hz...")
        from librosa import resample
        prompt_audio = resample(prompt_audio, orig_sr=sampling_rate, target_sr=target_sample_rate)
        prompt_audio = np.array(prompt_audio, dtype=np.float32)
        print(f"Resampled audio shape: {prompt_audio.shape}")
    else:
        print(f"Audio sampling rate already matches target ({target_sample_rate}Hz)")
    texts = ["为了点燃青少年对科技的热情,培养他们的创新思维与动手能力,杏花岭区巨轮街道社区教育学校携手中车社区教育分校,与太原市科学技术协会联手,于暑期精心策划了一场别开生面的青少年数智技术服务港探索之旅,吸引了众多社区青少年的积极参与。"]
    eos_token_id = model.config.vocab_size - 1
    print(f"EOS token ID: {eos_token_id}")
    # 生成输入嵌入
    embeddings,attention_mask = generate_embeddings_batch(
        model=model,
        tokenizer=tokenizer,
        texts=texts,
        bicodec=audio_tokenizer,
        prompt_text=prompt_text,
        prompt_audio=prompt_audio
    )
    input_embs = embeddings['input_embs']
    print(f"input_embs shape: {input_embs.shape}")
    print(f"attention_mask shape: {attention_mask.shape}")
    print(f"input_embs dtype: {input_embs.dtype}")
    print(f"attention_mask dtype: {attention_mask.dtype}")
    print(f"input_embs: {input_embs}")
    print(f"attention_mask: {attention_mask}")
    print(f"input_embs: {input_embs}")
    with torch.no_grad():
        generate(model,
                 input_embs,
                 attention_mask,
                 new_max_tokens=1000,
                 top_k=25,
                 top_p=0.95,
                 temperate=1.0,
                 eos_token_id=eos_token_id,
                 pad_token_id=tokenizer.pad_token_id if hasattr(tokenizer, 'pad_token_id') else tokenizer.eos_token_id,
                 past_key_values=None)
    
    with torch.no_grad():
        audio_tokens,past_key_values = generate(model,
                 input_embs,
                 attention_mask,
                 new_max_tokens=1000,
                 top_k=50,
                 top_p=0.8,
                 temperate=1.0,
                 eos_token_id=eos_token_id,
                 pad_token_id=tokenizer.pad_token_id if hasattr(tokenizer, 'pad_token_id') else tokenizer.eos_token_id,
                 past_key_values=None)
    audio_tokens = torch.tensor(audio_tokens, dtype=torch.long, device=device)
    audio_tokens = audio_tokens[:,:-1]
    print(f"audio_tokens: {audio_tokens}")
    print(f"past_key_values: {past_key_values}")
    global_tokens = embeddings['global_tokens']
    print(f"global_tokens shape: {global_tokens.shape}")
    print(f"audio_tokens shape: {audio_tokens.shape}")
    with torch.no_grad():
        wav = audio_tokenizer.detokenize(global_tokens, audio_tokens)
    print(f"wav shape: {wav.shape}")
    sf.write('test.wav', wav, audio_tokenizer.config['sample_rate'])