MisterH Posted October 30, 2009 Share Posted October 30, 2009 Hello- I'm testing Quorum right now and am wondering how to improve the sound quality when using the web interface for audio I/O. I've been testing on a local network and it is pretty terrible. I had planned on being able to accommodate VOIP users and I realize there are a lot of variables that affect audio quality in streaming applications. However, since my use so far is strictly local with plenty of bandwidth available, why does it sound as awful as a 4kHz sampling rate mp3 file? This can't be as good as it gets can it? Link to comment Share on other sites More sharing options...
pythonpoole Posted November 6, 2009 Share Posted November 6, 2009 Well It shouldn't sound like 4 kHz, but I would expect it to sound like 8 kHz as this is the telephony standard for now (except for some proprietary VoIP services like Skype). If you have a 22 or 44 kHz sound file and you down-sample it to 8 kHz, I find that it sounds a lot worse than if the original composition was made for 8 kHz playback (This is also true for text to speech and why many TTS companies will offer both a 22 and 8 kHz version of their voices). Lastly, if you are using the GSM codec, you will get cell-phone like quality & audio compression. You will find that using the G.711 codec which is basically uncompressed, provides much better quality audio. There is a trade-off however... If I remember correctly, GSM only uses about 5 kbps of bandwidth whereas G.711 uses something like 80 kbps (and I think that's just 1 way, x2 for up & down bandwidth). Link to comment Share on other sites More sharing options...
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