160 lines
3.6 KiB
Text
160 lines
3.6 KiB
Text
3.0.2
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- fix for Jack Linux audio interface: ports are only once registered and
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connect when the software is started up
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- bug fix: under bad network conditions chat messages were randomly repeated
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- bug fix: in case the server was shutdown and restarted during a connection,
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the channel name was not updated correctly at the server
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- updates for help texts
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- new design for fader tag
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- new server features: for chat messages the time stamp is also shown, parsing
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of existing log file supported
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3.0.1
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- bug fix: buzzing occurred when audio stream was interrupted (e.g. in case
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of network trouble)
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- in case "Open Chat on New Message" is not enabled, a hint in the status bar
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is shown when a message is received
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- use low complexity CELT encoder profile, this lowers audio dropout
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probability on slow computers
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3.0.0
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- introduced new audio codec "CELT", not compatible to old versions
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- only the sound card frame sizes 128, 256 and 512 is allowed (since other
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frame sizes require additional conversion buffers which introduce delay)
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- IMA-ADPCM, MS-ADPCM removed
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- since CELT works on 48 kHz sample rate, resampling was removed
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- various bug fixes (e.g. disconnecting did not work reliably)
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2.3.0
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- new system sample rate of 33 kHz to improve audio quality, not compatible
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to old versions
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- added history graph for server logging
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- added command line argument for connecting a server at software start-up
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2.2.2
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- "Mute" and "Solo" check boxes for each connected client fader
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- changes to the main GUI (grouped "local" and "server" settings)
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- LED status lights and LED input level meter
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- better behaviour on disconnection (introduced disconnection protocol message)
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- store previous server URLs
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2.2.1
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- bug fix and improvements for automatic jitter buffer size setting
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2.2.0
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- added Jack audio interface (Linux)
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- simplified settings dialog, complete redesign (removed sound card
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buffer settings, network block sizes settings, added upload rate display)
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- improved audio stability (audio interface is not callback based, removed
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intermediate audio buffers, client audio buffer size equals network
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buffer size now)
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- added upload rate limitation for server (server decides which network
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parameters to use depending on the upload limit and the number of connected
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clients)
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2.1.4
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- added automatic jitter buffer size setting
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- speed optimizations to improve audio interface stability
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- new defaults (e.g. turn off Reverb by default since it requires significant
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CPU)
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2.1.3
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- added sound card selection
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- added total delay display
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- improved ASIO configuration
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2.1.2
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- audio compression type can be selected (IMA ADPCM, MS ADPCM, None)
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- ping time measurement problems on Windows OS fixed
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- security checks for protocol messages (wrong messages could crash the
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software)
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2.1.1
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- chat window
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- server can be started on different port
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- client can select port of server
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- ping time information in general settings dialog
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2.1.0
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- ASIO support
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2.0.0
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- first QT4 compatible release
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0.9.9
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- new client settings dialog
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- at each client a separate audio mix can be generated for all connected clients
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at the server
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0.9.4
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- implemented protocol, now it is possible to set the jitter buffer in the
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server according to the setting in the client (they are coupled now)
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- removed sample rate offset estimation since it was not used anyway
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- internal audio processing in the server is now based on minimum block
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size (improves latency)
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0.9.1
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- initial version
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