When readframes() is used in python, the online documention says sampling frequency is returned it looks it returns 2 bytes. I think there are 4 byte on each frame:
I´m trying to simulate death over 7 years with the cumulative probability as follows: tab <- data.frame(id=1:1000,char=rnorm(1000,7,4))
What is the simplest way to capture audio from the built in audio input and be able to read the raw sampled values (as in a .wav) in real time as they come in when requested, like reading from a socke
Is there an algorithm for upsampling PCMU without开发者_StackOverflow社区 having to convert it to PCM first?
Decoded audio using FFmpeg (function avcodec_decode_audio3) and try to reproduce it through Audiotrack in Android. When playing I hear some growling. No music. On the forums advised that the problem w
I have 8k16bit pcm audio and I want to upsample it to 16k16bit. I have to do this manually. Can someone tell me the algorithm for linear interpolation? Should I interpolate between each two bytes?
What is the difference between Linear Quantization and Non-linear Quantization ? I\'m talking with regard to PCM samples.
How can I upsample AMR audio data. The am开发者_Python百科r file consists of 6 bytes header - \"!#AMR\".getBytes() and after that there are frames 32bytes each with 1 byte header and 31bytes audio. Ho
I try to implement Hampel tanh estimators to normalize highly asymmetric data. In order to do this, I need to perform the following calculation:
I am wondering on the relationship between a block of samples and its time equivalent. Given my rough idea so 开发者_StackOverflow社区far:
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