search for: qauliti

Hi All, Need Help ! I am interested testing opus codec encoding decoding qaulity. for this have complied opus code codec from souce. After compiling i got opus_demo app. for Encoding i followed below steps: 1) Reference file used music_orig.wav (http://www.opus-codec.org/examples/samples/music_orig.wav) Number of samples : 4358219 (90.8 s) 2015-04-13 10:40:07 UTC Sampling

Hi, The following may sound stupid so please forgive my stupidness. I have a question which I don't know how to name it so I have to start from the beginning. In an attempt to gain better understand how a photochemical air qaulity model works, I plotted hourly ozone concentration contour from the model's standard output to observe the changes in simulation. Because the model actually

Hi JM, I am using narrowband mode with speex qaulity set to 0 and complexity set to 4. The code works fine on the 3.2GHz processor. The test program speexenc takes 7ms to encode 5sec worth of voice on the 3.2Ghz processor. The same takes 700ms on the ARM5E intel Xscale. The code that I wrote puts 10 speex frames in one RTP packet. When exexcuted on ARM I get lots of underrruns hence

Hi Sakharam, I see 2 potential issues with what you are doing. 1. ./opus_demo -e voip 48000 2 16 music_orig.wav testcase30.opus in above command, "16" for bits/sec seems too low. I'm no audio expert, but just cant convince myself you can get any reasonable audio data with 16 bits/sec. FYI, I was able to encode and decode with 16 bits/sec, but when I played the decode file with

Hi JM, Thanks for the earlier replies. I have tested it with quality 1,2&3 the result is not much different. Also I must tell you that I am using the armbe-linux-gcc compiler which does not suppoort arm5e-asm optimisations. I get lots of assember related errors. Is there any alternative? Will moving on to a board supported by arm-linux-gcc compiler ,to enable arm5e optimisation, help?

Monkey's Audio lossless compressor (currently win32 only, free but not open-source except decoder) author is thinking to implement a kind of audiophile-quality lossy compression which would filter "noise bits" that are hard to encode lossless but which are (or should be) inaudible and thus improve lossless compression (avg. 300-450kbps). I think that implementing something like this

Hi Can someone please help with the following, We are using asterisk@home 1.5 and SIP trunks to communicate to the PSTN network. We are having some problems with the call quality. Although we can hear the other person's voice quite clear when making or receiving a call, we get complaints from the people on the other end saying that our voices sound very unclear, low and that the voice