
Copyright 2013 Simon Brown
7.3 Audio Codec
This software uses the Opus Interactive Audio Codec to reduce network bandwidth when
sending audio between the server and the console. For more information about Opus visit
http://www.opus-codec.org/.
There are various options which can be set in the encoder (runs on the server), these are
described below.
The opus encoder supports three coding modes (the default used in the server is
OPUS_APPLICATION_AUDIO):
VOIP gives best quality at a given bitrate for voice signals. It enhances the input
signal by high-pass filtering and emphasizing formants and harmonics. Optionally it
includes in-band forward error correction to protect against packet loss. Use this
mode for typical VoIP applications. Because of the enhancement, even at high
bitrates the output may sound different from the input.
Audio gives best quality at a given bitrate for most non-voice signals like music. Use
this mode for music and mixed (music/voice) content, broadcast, and applications
requiring less than 15 ms of coding delay.
The encoder supports a hint which helps the encoder's mode selection (the default used is
OPUS_AUTO):
Auto (default)
Voice Bias thresholds towards choosing LPC or Hybrid modes.
Kommentare zu diesen Handbüchern