jamulus/ChangeLog

3.1.0

- support for MacOS (using CoreAudio audio interface)

- sound card frame size support for frame sizes other than 128, 256 and 512
  samples

- improvement of network buffer (jitter buffer) in case of small buffer sizes

- all available ASIO sample formats supported

- bug fix for M-Audio Delta ASIO driver


3.0.3

- accessibility improvements

- show number of connected clients in window title (and therefore in OS task
  bar)

- added "Whats this" help text to the GUI controls in the general settings
  dialog, added Tool Tips to some GUI controls

- server logging history grid lines of weekend days are now plotted with
  different line width

- removed ALSA support since the ALSA interface implementation in llcon was
  buggy and will not be maintained in the future

- bug fix: fix for server logging history graph x-axis

- bug fix: fix for disconnect issue at the server


3.0.2

- new server features: for chat messages the time stamp is also shown, parsing
  of existing log file supported

- updates for help texts

- new design for fader tag

- change for Jack Linux audio interface: ports are only registered and
  connected once when the software is started up

- bug fix: under bad network conditions chat messages were randomly repeated

- bug fix: in case the server was shutdown and restarted during a connection,
  the channel name was not updated correctly at the server


3.0.1

- use low complexity CELT encoder profile, this lowers audio dropout
  probability on slow computers

- in case "Open Chat on New Message" is not enabled, a hint in the status bar
  is shown when a message is received

- bug fix: buzzing occurred when audio stream was interrupted (e.g. in case
  of network trouble)


3.0.0

- introduced new audio codec "CELT", not compatible to old versions

- only the sound card frame sizes 128, 256 and 512 is allowed (since other
  frame sizes require additional conversion buffers which introduce delay)

- IMA-ADPCM, MS-ADPCM removed

- since CELT works on 48 kHz sample rate, resampling was removed

- various bug fixes (e.g. disconnecting did not work reliably)


2.3.0

- new system sample rate of 33 kHz to improve audio quality, not compatible
  to old versions

- added history graph for server logging

- added command line argument for connecting a server at software start-up


2.2.2

- "Mute" and "Solo" check boxes for each connected client fader

- store previous server URLs

- changes to the main GUI (grouped "local" and "server" settings)

- LED status lights and LED input level meter

- better behaviour on disconnection (introduced disconnection protocol message)


2.2.1

- bug fix and improvements for automatic jitter buffer size setting


2.2.0

- added Jack audio interface (Linux)

- simplified settings dialog, complete redesign (removed sound card
  buffer settings, network block sizes settings, added upload rate display)

- improved audio stability (audio interface is not callback based, removed
  intermediate audio buffers, client audio buffer size equals network
  buffer size now)

- added upload rate limitation for server (server decides which network
  parameters to use depending on the upload limit and the number of connected
  clients)


2.1.4

- added automatic jitter buffer size setting

- speed optimizations to improve audio interface stability

- new default settings (e.g. turn off Reverb by default since it requires
  significant CPU)


2.1.3

- added sound card selection

- improved ASIO configuration

- added total delay display


2.1.2

- audio compression type can be selected (IMA ADPCM, MS ADPCM, None)

- security checks for protocol messages (wrong messages could crash the
  software)

- bug fix: ping time measurement problems on Windows OS fixed


2.1.1

- added a chat window

- the client can select a port number of server additional to the URL

- the server can be operated on an arbitrary port number

- ping time information in general settings dialog added


2.1.0

- ASIO support


2.0.0

- first QT4 compatible release


0.9.9

- new client settings dialog

- at each client a separate audio mix can be generated for all connected clients
  at the server


0.9.4

- added a protocol mechanism, now it is possible to set the jitter buffer in the
  server according to the setting in the client (they are coupled now)

- removed sample rate offset estimation since it was not used anyway

- internal audio processing in the server is now based on the minimum block
  size (improves latency)


0.9.1

- initial version