MP3 ENCODER LIBRARY HOW TO
MP3 ENCODER LIBRARY PDF
MP3 ENCODER LIBRARY CODE

vmsg uses the latest version of the LAME library ( 3.100 from October 2017)īut encoding mp3 is very fast on modern CPUs – even when using the asm.js version of the LAME mp3 encoder – so choosing between Web Audio Recorder and vmsg comes down to ease of use, browser support, whether or not the library is still maintained and personal preference.

vmsg uses the same C based LAME mp3 encoding library but compiled to a newer WebAssembly library which, in theory, should be faster than the asm.js one and smaller (73KB vs 114 KB gzipped).

MP3 ENCODER LIBRARY CODE

Web Audio Recorder used a JavaScript library compiled from LAME’s C code to a subset of JavaScript called asm.js.

Vmsg is very similar to WebAudioRecorder.js in that it uses a compiled version of the LAME mp3 encoding library but they differ in the way that library was compiled:

MP3 ENCODER LIBRARY HOW TO

We’ve previously covered how to use Recorder.js to capture uncompressed pcm audio and WebAudioRecorder.js to capture pcm, mp3 and Vorbis audio in the browser.

MP3 ENCODER LIBRARY PDF

See, for example, Wikipedia page on Oversampling, or many vendors' application notes (each word a different link, the Microchip one goes in some theoretical detail), see also the relevant chapter in Lyons's ' Understanding Digital Signal Processing' (on Amazon, or maybe some pdf might be found floating in the net.).This is the 3rd post in our new recording audio in HTML5 series. In general, for n bits, 2^2n (=4^n) oversampling is needed. double it), you need 4× oversampling, as you gain only half bit (Sqrt(2)) of S/n every time you double the samples.

So, to increase the real resolution by 1 bit, (i.e. This is, in fact, an old analogue trick you could sometimes find paralleled transistors in (pre)amplifiers: the increase in signal was larger than the one in noise. OTOH, when two you sum the signal part, very correlated voltages, the result is that the average amplitude is twice the original one. the average amplitude increased by Sqrt(2). When you sum two noises together, as they are random, the resulting signal will have double the power of the original signals, i.e. Sorry, I was possibly too terse, if you are not familiar with the topic - not that I pretend to be an expert. Does this make sense? If so, it is quite simple nothing much 'real time' especially if DMA is used to read the ADC. The FLASH writing speed is 30kbytes/sec which seems plenty fast enough. So the encoder would run periodically (before the input buffer overflows only a few k is needed if processing it every 100ms) and each time it is run, one would write the data to a file (FatFS SPI FLASH). the ADC data ('PCM', which sounds like amplitude-compressed ADC data) is provided as a block of RAM, and the output is similarly a block of RAM. A quick look at the ST library (not had time to read all 200+MB of it) shows this which suggests that is exactly what they intend i.e. This is because the sampling obviously has to be at regular intervals, and one doesn't want to run the encoder, and the file system, in a timer ISR. I posted the outline here Looking at this in more detail, and having looked at the ST mp3 library, it seems to me that the best way to implement this is by driving the ADC with a timer and using DMA to store the values in RAM, and then running the encoder as an RTOS thread, every 100ms or so. It is a sort of an automatic sound recorder. I am looking at implementing an mp3 encoder which, commencing at power-up, uses the 12 bit ADC and writes out an mp3 file at power-down.

Mp3 encoder library