import network
import usocket as socket
import ustruct
import time
import machine
import urequests
from machine import Pin, I2S
import ubinascii
import urandom

# I2S 引脚配置
BCLK_PIN = 13
WS_PIN = 12
SD_PIN = 14
# 增益控制引脚
GAIN_PIN = 15

# 初始 I2S 配置，后续根据文件实际参数调整
SAMPLE_RATE = 16000
BITS_PER_SAMPLE = 16  # 修改为 16 位
CHANNELS = 2
BUFFER_SIZE = 8192

# 初始化I2S配置（用于音频输出）
def init_i2s():
    global BCLK_PIN, WS_PIN, SD_PIN, GAIN_PIN, SAMPLE_RATE, BITS_PER_SAMPLE, CHANNELS, BUFFER_SIZE
    gain_pin = machine.Pin(GAIN_PIN, machine.Pin.OUT)
    gain_pin.value(1)
    # 使用GPIO14 (BCLK), GPIO15 (LRC), GPIO32 (DIN)
    i2s = I2S(0,
              sck=Pin(BCLK_PIN),  # Serial clock output
              ws=Pin(WS_PIN),   # Word clock output
              sd=Pin(SD_PIN),   # Serial data output
              mode=I2S.TX,  # 使用发送模式
              bits=BITS_PER_SAMPLE,  # 修改为 16 位数据
              format=I2S.MONO if CHANNELS == 1 else I2S.STEREO,  # 修正声道判断逻辑
              rate=SAMPLE_RATE,    # 采样率
              ibuf=BUFFER_SIZE)    # 输入缓冲区大小
    return i2s


class InterphoneHandler:

    def play_wav_file(self, data):
        try:
            # 要播放的本地 WAV 文件
            WAV_FILE = data['file']
            # 初始化 I2S 接口
            audio_out = init_i2s()

            # 打开 WAV 文件
            with open(WAV_FILE, 'rb') as f:
                # 跳过 WAV 文件头部（通常为 44 字节）
                f.seek(44)
                while True:
                    # 读取音频数据块
                    data = f.read(BUFFER_SIZE)
                    if len(data) == 0:
                        break
                    # 将音频数据写入 I2S 接口
                    print('将音频数据写入 I2S 接口')
                    audio_out.write(data)

        except Exception as e:
            print(f"播放音频时出错: {e}")
        finally:
            if 'audio_out' in locals():
                audio_out.deinit()

    def play_audio_stream(self, i2s, url):
        try:
            # 添加请求头信息
            headers = {
                'User-Agent': 'MicroPython v1.0.0'
            }
            # 发送 HTTP 请求，获取流式响应
            response = urequests.get(url, headers=headers, stream=True)

            # 检查响应状态码
            if response.status_code != 200:
                print(f"请求失败，状态码: {response.status_code}")
                response.close()
                return

            # 解析 WAV 头部
            header = response.raw.read(44)

            if len(header) != 44 or header[0:4] != b'RIFF' or header[8:12] != b'WAVE' or header[12:16] != b'fmt ':
                print(header)
                raise ValueError("Not a valid WAV file")

            # 提取采样率、位深、声道数
            global SAMPLE_RATE, BITS_PER_SAMPLE, CHANNELS
            SAMPLE_RATE = int.from_bytes(header[24:28], 'little')
            BITS_PER_SAMPLE = int.from_bytes(header[34:36], 'little')
            CHANNELS = int.from_bytes(header[22:24], 'little')

            # 重新初始化 I2S 输出
            audio_out = init_i2s()

            # 开始流式播放
            chunk_size = 1024

            while True:
                chunk = response.raw.read(chunk_size)
                if not chunk:
                    break
                audio_out.write(chunk)

        except Exception as e:
            print("Playback error:", e)
        finally:
            if 'audio_out' in locals():
                audio_out.deinit()
            if 'response' in locals():
                response.close()

    def stream_and_play(self, data):
        i2s = init_i2s()
        url = "http://iot.julecn.com/interphone/get_voice?name=" + data['name']   # 替换为音频文件的URL
        print(url)
        self.play_audio_stream(i2s, url)


def generate_random_hex():
    # 初始化一个空的字节数组，用于存储随机字节
    random_bytes = bytearray()

    # 循环 4 次，每次生成 32 位（4 字节）的随机数
    for _ in range(4):
        # 生成 32 位随机数
        rand_32_bits = urandom.getrandbits(32)
        # 将 32 位随机数转换为 4 字节，并添加到字节数组中
        random_bytes.extend(rand_32_bits.to_bytes(4, 'big'))

    # 将随机字节转换为十六进制字符串
    hex_string = ubinascii.hexlify(random_bytes).decode()
    return hex_string