The following screenshot shows the Jamulus general settings window.
Sound card device
The ASIO driver (sound card) can be selected using Jamulus under the Windows operating system. If the selected ASIO driver is not valid an error message is shown and the previous valid driver is selected. Under the Mac operating system the input and output hardware can be selected.
Input/output channel mapping
In case the selected sound card device offers more than one input or output channel, the Input Channel Mapping and Ouptut Channel Mapping settings are visible.
For each Jamulus input/output channel (left and right channel) a different actual sound card channel can be selected.
The buffer delay setting is a fundamental setting of the Jamulus software. This setting has influence on many connection properties.
Three buffer sizes are supported:
- 128 samples: This is the preferred setting since it gives lowest latency but does not work with all sound cards.
- 256 samples: This setting should work on most of the available sound cards.
- 512 samples: This setting should only be used if only a very slow computer or a slow internet connection is available.
Some sound card driver do not allow the buffer delay to be changed from within the Jamulus software. In this case the buffer delay setting is disabled. To change the actual buffer delay, this setting has to be changed in the sound card driver. On Windows, press the ASIO Setup button to open the driver settings panel.
On Linux, use the Jack configuration tool to change the buffer size.
The actual buffer delay has influence on the connection status, the current upload rate and the overall delay. The lower the buffer size, the higher the probability of red light in the status indicator (drop outs) and the higher the upload rate and the lower the overall delay.
The buffer setting is therefore a trade-off between audio quality and overall delay.
Jitter buffer with buffer status indicator
The jitter buffer compensates for network and sound card timing jitters. The size of this jitter buffer has therefore influence on the quality of the audio stream (how many dropouts occur) and the overall delay (the longer the buffer, the higher the delay).
The jitter buffer size can be manually chosen for the local client and the remote server. For the local jitter buffer, dropouts in the audio stream are indicated by the light on the bottom of the jitter buffer size faders. If the light turns to red, a buffer overrun/underrun took place and the audio stream is interrupted.
The jitter buffer setting is therefore a trade-off between audio quality and overall delay.
An auto setting of the jitter buffer size setting is available. If the check Auto is enabled, the jitter buffers of the local client and the remote server are set automatically based on measurements of the network and sound card timing jitter. If the Auto check is enabled, the jitter buffer size faders are disabled (they cannot be moved with the mouse).
Select the number of audio channels to be used. There are three modes available. The mono and stereo modes use one and two audio channels respectively. In the mono-in/stereo-out mode the audio signal which is sent to the server is mono but the return signal is stereo. This is useful for the case that the sound card puts the instrument on one input channel and the microphone on the other channel. In that case the two input signals can be mixed to one mono channel but the server mix can be heard in stereo.
Enabling the stereo streaming mode will increase the stream data rate. Make sure that the current upload rate does not exceed the available bandwidth of your internet connection.
In case of the stereo streaming mode, no audio channel selection for the reverberation effect will be available on the main window since the effect is applied on both channels in this case.
Select the desired audio quality. A low, normal or high audio quality can be selected. The higher the audio quality, the higher the audio stream data rate. Make sure that the current upload rate does not exceed the available bandwidth of your internet connection.
New client level
The new client level setting defines the fader level of a new connected client in percent. I.e. if a new client connects to the current server, it will get the specified initial fader level if no other fader level of a previous connection of that client was already stored.
If enabled, a fancy skin will be applied to the main window.
Central server address
The central server address is the IP address or URL of the central server at which the server list of the connection dialog is managed. If the Default check box is checked, the default central server address is shown read-only.
Current connection status parameter
The ping time is the time required for the audio stream to travel from the client to the server and backwards. This delay is introduced by the network. This delay should be as low as 20-30 ms. If this delay is higher (e.g., 50-60 ms), your distance to the server is too large or your internet connection is not sufficient.
The overall delay is calculated from the current ping time and the delay which is introduced by the current buffer settings.
The upstream rate depends on the current audio packet size and the audio compression setting. Make sure that the upstream rate is not higher than the available rate (check the upstream capabilities of your internet connection by, e.g., using speedtest.net).