eeva/jamulus
1
0
Fork 0
Code Issues Pull requests Releases Wiki Activity

Update README.md

Browse source 
finished the settings window description
This commit is contained in:
corrados 2019-06-16 11:08:08 +02:00 committed by GitHub
parent b1d043301b
commit 7fb632a209
No known key found for this signature in database
GPG key ID: 4AEE18F83AFDEB23
1 changed files with 86 additions and 9 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

95
README.md
View file

@ -43,7 +43,7 @@ ASIO support, you can try out the following alternative:
[ASIO4ALL - Universal ASIO Driver For WDM Audio](http://www.asio4all.com)
The ASIO buffer size should be selected as low as possible to get
the minimum audio latency (a good choice is 128 samples).
the minimum audio latency (a good choice is 128 samples).
Help (Software Manual)
@ -75,7 +75,7 @@ audio stream is interrupted caused by one of the following problems:
![Input level](src/res/homepage/inputlevel.jpg)
The input level indicators show the input level of the two stereo channels of the current selected audio input.
Make sure not to clip the input signal to avoid distortions of the audio signal.
Make sure not to clip the input signal to avoid distortions of the audio signal.
#### Chat button opens the Chat dialog
@ -83,7 +83,7 @@ Make sure not to clip the input signal to avoid distortions of the audio signal.
Press the Chat button to open the Chat dialog. The chat text entered in that dialog is transmitted to
all connected clients. If a new chat message arrives and the Chat dialog is not already open, it will
be opened automatically at all clients.
be opened automatically at all clients.
#### My Profile button opens the Musician Profile dialog
@ -113,7 +113,7 @@ address is specified in the settings, no server list will be available.
Alternatively, you can enter an IP address or URL of the server running the Jamulus server in the server address
field. An optional port number can be added after the IP address or URL using a comma as a separator, e.g,
jamulus.dyndns.org:22124. A list of the most recent used server IP addresses or URLs is available for selection.
jamulus.dyndns.org:22124. A list of the most recent used server IP addresses or URLs is available for selection.
#### Reverberation effect
@ -126,7 +126,7 @@ to right and move the fader upwards until the desired reverberation level is rea
The reverberation effect requires significant CPU so that it should only be used on fast PCs. If the reverberation
level fader is set to minimum (which is the default setting), the reverberation effect is switched off and does
not cause any additional CPU usage.
not cause any additional CPU usage.
#### Local audio input fader
@ -135,7 +135,7 @@ not cause any additional CPU usage.
With the audio fader, the relative levels of the left and right local audio channels can be changed. For a mono signal
it acts like a panning between the two channels. If, e.g., a microphone is connected to the right input channel and
an instrument is connected to the left input channel which is much louder than the microphone, move the audio fader
in a direction where the label above the fader shows L -x, where x is the current attenuation indicator.
in a direction where the label above the fader shows L -x, where x is the current attenuation indicator.
#### Server audio mixer
@ -158,7 +158,7 @@ be set to solo which means that all other channels except of the current channel
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.
system the input and output hardware can be selected.
#### Input/output channel mapping
@ -166,7 +166,7 @@ system the input and output hardware can be selected.
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.
a different actual sound card channel can be selected.
#### Buffer delay
@ -194,7 +194,84 @@ higher the upload rate and the lower the overall delay.
![Buffer delay dependencies](src/res/homepage/bufferdelaydependency.jpg)
The buffer setting is therefore a trade-off between audio quality and overall delay.
The buffer setting is therefore a trade-off between audio quality and overall delay.
#### Jitter buffer with buffer status indicator
![Jitter buffer](src/res/homepage/jitterbuffer.png)
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).
#### Audio channels
![Audio channels](src/res/homepage/audiochannels.png)
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.
#### Audio quality
![Audio quality](src/res/homepage/audioquality.png)
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
![New client level](src/res/homepage/newclientlevel.png)
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.
#### Fancy skin
![Fancy skin](src/res/homepage/fancyskin.png)
If enabled, a fancy skin will be applied to the main window.
#### Central server address
![Central server address](src/res/homepage/centralserveraddress.png)
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
![Indicators](src/res/homepage/indicators.png)
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](http://speedtest.net)).
Compilation and Development