On 08/19/2013 08:34 PM, keith gould wrote: > Been reading through codec2.c in the newest release, and focusing on > codec2_encode_1200. > > The comment above the method: > " Encodes 320 speech samples (40ms of speech) into 48 bits. " > > I would like to verify the following from you guys: > > 1) The 40ms of speech is only true at 8kHz. If we pass in recorded > audio at a higher rate then the 320 samples would equate to some > smaller amount of time > > 2) The buffer (of 320 samples) does not change in quantity as the rate > of the source file changes. The buffer size is constant (320 samples) > > 3) These 320 samples will always output 48 bits, even if those 48 bits > corresponds to varying lengths of time due to different sample lengths > for different files. > > Hope this makes sense. I'm pretty sure this is true, but I don't yet > fully understand the codec's internals. > If you get to understand the codec's internals you will find that feeding the codec at rates other than 8k samples/second is a very poor choice.
Regards, Steve ------------------------------------------------------------------------------ Get 100% visibility into Java/.NET code with AppDynamics Lite! It's a free troubleshooting tool designed for production. Get down to code-level detail for bottlenecks, with <2% overhead. Download for free and get started troubleshooting in minutes. http://pubads.g.doubleclick.net/gampad/clk?id=48897031&iu=/4140/ostg.clktrk _______________________________________________ Freetel-codec2 mailing list [email protected] https://lists.sourceforge.net/lists/listinfo/freetel-codec2
