AEC3 Extracted From WebRTC
visual studio 2015
see demo.cc
AEC3 Extracted From WebRTC
你好!我下载了你裁剪出来的AEC3的代码,有个疑问:audio_processing/audio_buffer.cc 这个文件中,void AudioBuffer::SplitIntoFrequencyBands() 函数中的 "if (num_bands_ > 1) {...}" 这个判断条件,是你自己加上去的,还是在某一版本的WebRTC的源码中就已经存在呢?谢谢!
I see that you're using visual studio. But I know that AEC3 is cross platform. Do you think there's many changes necessary to get it working on Mac ?
作者你好,感谢你的开源代码,请问有没有基于APM实现的aec3的例子或者相关的资料
AEC3用的是 kalman + PBFDAK 的双滤波器形式,我用AEC3跑了很多的音频之后发现:它在双工部分有较多的残留回声,更多的还是靠NLP来压。这样不就完全没有体现到kalman滤波器的优势吗?kalman滤波器在双工的时候表现应该远比这好才对。总的来说,它的双工性能似乎不好,是它在计算上的精简导致的吗?
近端录音是很干净的声音,远端有人说话,通过这个算法后,近端处理后的声音有小回声,AEC3算法本身的问题吗
Hi,
I am not much familiar with the C projects but I had to select this project as I am working on this area. I tried to build this project with Visual Studio 2022 and ran into a bunch of errors. Can you please show a little detailed path to how can I build and run this.? should I must use Visual Studi 2015? and then should set any extra settings before running? Thanks.
hi,我看了你在 CSDN 的文章,其中写到:“通过与kalman滤波器仔细对比可以发现,其实这个滤波器是kalman滤波器的精简版,其更新公式与kalman滤波器是一样的,只不过精简了系数误差协方差的计算,计算的是滤波器抽头系数误差协方差的平均,而不是计算每块滤波器抽头系数的误差协方差”。
我想知道这个“滤波器抽头系数误差协方差”指的是 Erl(t) = sum{H(n)*H(n)} 还是 E2(t) 呢(因为mu的更新公式中出现了 N * E2,我猜可能就是块平均)?
我对C++工程不是很熟,找了挺久没有找到对应的精简计算在哪里,可以帮我解释一下或者贴出来精简部分的代码在哪里吗
該如何將這個專案編譯成可匯入到android專案中的library?
如题
Could you please add a link to my Qt adaptation?
It uses Cmake and can be very useful for who wants a realtime example with microphone and speakers
我愿称之为好活!
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