PCoIP Audio Features

• 2 channel Stereo audio
• Automatic quality adjustment to fit inside the available PCoIP session bandwidth
• Audio In and Audio Out via the HP Anyware Audio Driver included in the HP Anyware or ALC888 equivalent audio in the PCoIP Remote Workstation Card.

Codecs

There are 2 audio codecs in use in the PCoIP environment, ADPCM and OPUS. Both codes offer sound quality from compressed mono through to high quality stereo depending on the available bandwidth. To determine which codec is in use, refer to this table.

*OPUS codec used for Audio out from the host to client only. Microphone / Audio In is the ADPCM codec.

OPUS

With the introduction of the OPUS codec, bandwidth required for high quality audio has been significantly reduced. The maximum audio quality is achieved with a bit rate of around 256 kbit/s + an additional 20 -40% network and error correction overhead, freeing up bandwidth for other data. 

PCoIP monitors the bandwidth available for audio and selects the audio compression algorithm that provides the best quality possible for the available bandwidth. Maximum audio quality with a bit rate of 256 kbit/s is used when PCoIP detects that the total available bandwidth for the session is greater than 10 mbit/s. Below 10 mbit/s PCoIP dynamically adjusts the audio bit rate between 256 kbit/s and 32 bit/s depending on the available PCoIP bandwidth.

Error correction is used with OPUS to minimize the pops, crackles and silences that can be experienced when streaming audio enhancing the user experience. Even with error correction, too much data loss between the host and client will still result in poor audio. When a audio packet is lost it is not retransmitted as the time for the audio to be played has passed. As PCoIP is a real time protocol, audio cannot be retransmitted as the audio and imaging must happen as quickly as possible after the user's input.

ADPCM

ADPCM is used for all audio from client to host as well as host to client on older platforms. 

As with OPUS, the bandwidth used by audio is dynamically adjusted based on the PCoIP bandwidth available. Unlike OPUS, the audio bandwidth bit rate is set and will jump between the different audio qualites based on the availalbe PCoIP network bandwidth.

Audio bandwidth limit

Setting the audio bandwidth limit from host to client

When using audio on a constrained link it may be desirable to limit the bandwidth in use. This can be done via the PCoIP Group Policy setting Configure the PCoIP session audio bandwidth limit.

Considerations when setting the Group Policy setting

• The audio bandwidth limit only applies between the host and client. There is no ability to limit the client to host.
• The default limit for the OPUS codec is 256 kbit/s + overhead
• The default limit for ADPCM is 500Kbit/s
• Setting the limit below 32 for OPUS or below 60 for ADPCM will result in no audio playback
• The audio limit takes effect immediately
• The bandwidth limit applies to both OPUS and ADPCM codecs. If you are using a mixed environment with older clients, consider the ADPCM bandwidth requirements when configuring the setting or update the clients.

Setting the bandwidth limit for client to host audio

There is currently no direct configuration that allows for limiting the audio bandwidth between the client and host. 

Workaround: When using a Zero Client, the bandwidth limit for upload can be set on the zero client in some circumstances. 

• When USB is not in use for anything not locally terminated (keyboard, mouse, local audio).
• Teradici Management Console is managing the Zero Client, or the AWI is accessible.

To set the upload bandwidth limit, set the device bandwidth limit. This sets the total PCoIP bandwidth from the client to host. The bandwidth limit impacts, audio, keyboard, mouse, USB and Imaging acknowledgement data. Set the limit to the just below the available network bandwidth for the next higher audio quality in the ADPCM codec table.

ADPCM technical details