I\'ve written a program that transmits a PCM stream from my pc to another pc or my android(using an AudioTrack). It uses java sound 开发者_JS百科and takes the target and source lines from the Stereo M
I have 16 bit PCM audio and I want to convert it to 8bit PCMU. As far as I know 16bit to 8 bit conversion is easy - just drop the last 8 bits from each sample.
i\'m really sorry if i\'m asking something already asked, but i looked everywhere and nothing came up.
I have pcm 开发者_StackOverflow社区audio stored in a byte array. It is 16 bits per sample. I want to make it 8 bit per sample audio.
I\'ve never been able to understand how audio data is开发者_StackOverflow中文版 stored.However, I\'d like to know a way to find the pitch of PCM data.Let\'s say, for example, that I recorded a single
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.
I read that signal power = signal * signal. What is that? what is signal? how do we obtain it? I\'m progra开发者_如何学Gomming in C(if that\'s necessary to post)Signal is the amplitude of whatever inp
Next iteration of my question: Thank you for your inputs, it has helped me to understa开发者_C百科nd a little bit more about the Frame and inputSamples utility.
I want to rea开发者_运维问答d pcm samples from a file using fread and want to determine the signal strength of the samples.How do I go about it?
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