while(snr_t < 46 ) %迭代训练 s_left_0 = [fill,soud_left_d,fill]; s_right_0 = [fill,soud_right_d,fill]; s_left_1 = ResBlock(s_left_0,conv_t(2,:),conv_t(3,:)); s_right_1 = ResBlock(s_right_0,conv_t(2,:),conv_t(3,:)); s_left_1 = conv(s_left_1,conv_t(4,:)); s_right_1 = conv(s_right_1,conv_t(4,:)); s_left_2 = s_left_0(1+wid:end-wid) + s_left_1(1+4*wid:end-4*wid) * w_t; s_right_2 = s_right_0(1+wid:end-wid) + s_right_1(1+4*wid:end-4*wid) * w_t; s_left_2 = PixelShuffle(s_left_2,conv_t(6:21,:)); s_right_2 = PixelShuffle(s_right_2,conv_t(6:21,:)); s_left_2 = conv(s_left_2,conv_t(5,:)); s_right_2 = conv(s_right_2,conv_t(5,:)); s_left_2 = s_left_2(1+wid:end-wid); s_right_2 = s_right_2(1+wid:end-wid); power_s = Count(s_left_2(1,:)); power_mult = power_s / power_xr; s_left_2 = s_left_2 / (power_mult)^0.5; s_right_2 = s_right_2 / (power_mult)^0.5; temp = s_left_2 - soud_left_t; loss_num = LossCount(temp,soud_left); diff = Variance(temp,soud_left); snr = psnr(temp,soud_left);

这段代码看起来像是一段音频信号处理的代码，其中包括了一些卷积运算、残差块、像素重排等操作。代码中使用了一些函数，如 Count、LossCount、Variance 和 psnr，这些函数可能是自己实现的，用于计算信号的功率、损失、方差和信噪比等指标。整个代码似乎是一个循环，其中的变量 snr_t 作为循环条件，当其小于 46 时进行迭代训练。在每次迭代中，会对两个音频信号进行处理，并计算出它们之间的损失和信噪比等指标。

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while(snr_t < 46 ) %迭代训练 % s_left_0 = conv(soud_left_d,conv_t(1,:)); % s_right_0 = conv(soud_right_d,conv_t(1,:)); s_left_0 = [fill,soud_left_d,fill]; s_right_0 = [fill,soud_right_d,fill]; s_left_1 = ResBlock(s_left_0,conv_t(2,:),conv_t(3,:)); s_right_1 = ResBlock(s_right_0,conv_t(2,:),conv_t(3,:)); s_left_1 = conv(s_left_1,conv_t(4,:)); s_right_1 = conv(s_right_1,conv_t(4,:));

这段代码是一个迭代训练的过程，其中使用了卷积操作（conv）和残差块（ResBlock）。具体地，代码中将输入信号soud_left_d和soud_right_d进行填充（fill），然后分别通过卷积模板（conv_t）进行卷积操作。接着，将卷积输出通过残差块进行处理，最后再次进行卷积操作，并计算信噪比（snr_t）。当信噪比达到一定阈值（46）时，迭代训练结束。

while(snr_t < 46 ) %迭代训练 s_left_0 = [fill,soud_left_d,fill]; s_right_0 = [fill,soud_right_d,fill]; s_left_1 = ResBlock(s_left_0,conv_t(2,:),conv_t(3,:)); s_right_1 = ResBlock(s_right_0,conv_t(2,:),conv_t(3,:)); s_left_1 = conv(s_left_1,conv_t(4,:)); s_right_1 = conv(s_right_1,conv_t(4,:)); s_left_2 = s_left_0(1+wid:end-wid) + s_left_1(1+4wid:end-4wid) * w_t; s_right_2 = s_right_0(1+wid:end-wid) + s_right_1(1+4wid:end-4wid) * w_t; s_left_2 = PixelShuffle(s_left_2,conv_t(6:21,:)); s_right_2 = PixelShuffle(s_right_2,conv_t(6:21,:)); s_left_2 = conv(s_left_2,conv_t(5,:)); s_right_2 = conv(s_right_2,conv_t(5,:)); s_left_2 = s_left_2(1+wid:end-wid); s_right_2 = s_right_2(1+wid:end-wid); power_s = Count(s_left_2(1,:)); power_mult = power_s / power_xr; s_left_2 = s_left_2 / (power_mult)^0.5; s_right_2 = s_right_2 / (power_mult)^0.5; temp = s_left_2 - soud_left_t; loss_num = LossCount(temp,soud_left); diff = Variance(temp,soud_left); snr = psnr(temp,soud_left);

这段代码的作用是对语音信号进行迭代训练和处理，希望获得更高的信噪比（SNR）。具体来说，它首先将输入的左右声道信号进行填充，然后通过ResBlock函数进行残差块的处理。接着，通过卷积操作对信号进行处理，并使用PixelShuffle函数对信号进行下采样。最后，通过计算信号的功率、多项式系数等参数，对信号进行归一化处理，并计算损失函数、方差和信噪比等指标。这个过程会不断迭代，直到信噪比达到指定的阈值。