Issue identified with Patch Group management and circular dependencies that can develop because of stale patch groups. Refactored all patch group management into a new PatchGroupManager. Patch groups are now automatically deleted if we don't receive any update messages within 5 seconds. The PatchGroupAlias was also updated to not roll-up attributes from subordinate patched group aliases when those aliases are also patch group aliases -- this eliminates the circular dependency chaining.
Added IOverflowListener interface and updated Sources to support overflow listeners. Updated ChannelState to register as an overflow listener so that it can visually depict to the user when the channel enters/exits source overflow state. Added new color pattern for OVERFLOW with yellow text on red.
Changes to model, view and controller components and updates all gui
editors. Removes the configuration panel and tree and replaces with 3
new panels. Updates playlist.xml to a new playlist_v2.xml format.
-Removed channel collection panel since panels will only have 1 channel
now
-Resolved issue with channel list and channel selection events
-Fixed some of the classes from the rebase to master branch
-Started work on new channel editor channel viewer/controller
-Removed a number of unused buffer package classes
-Moved automatic frequency control to be a module and removed special
handling
-Added IUnfilteredRealBufferListener & Provider interfaces
-Completely removed the ResourceManager class
component.
-Airspy supported with 10 and 2.5 MSps sample rates. Additional
decimation may be added later.
-Spectral display updated to include smoothing and filtering options.
Note: the decimated output of a tuner channel from the airspy is
48,076.9 samples per second, versus the normal 48 kHz decimated rate.
This is 1.0015 times the normal rate and doesn't appear to impact the
decoders or audio processing. Therefore, no fractional resampling has
been added.
from Module so that Decoder can function independent of DecoderSTate and
DecoderState can more cleanly interface with other modules.
Resolves issue with LTR-type decoders where the AudioModule was
filtering demodulated audio buffers before the decoder received the
audio - demodulated audio buffers are now copied before creating
AudioPackets.
Note: Several of the decoders are currently non-functioning in this
commit. If you want the latest functioning codebase, please checkout
the commit prior to this one. All decoders will eventually be updated
with this new architecture.
Currently broken:
- Logging and recording
- Aux decoders (Fleetsync/MDC/LoJack/Tait)
- LTR/Passport decoders
Changes in this commit:
- Module architecture for decoders and components with central
provisioning by the DecoderFactory
- Transition from Complex sample stream processing to float buffer
processing for improved memory management
- Redesigned filter structures - to support buffers and streams
- Resolved several memory leak issues
- Universal sharing of the thread pool manager
- Added centralized audio management with call/user prioritization and
new gui with mute/volume controls
- Embedded audio metadata for eventual streaming support
- Redesigned ProcessingChain to support independent re-use
- Added new TrafficChannelManager to handle traffic channel allocations
- Traffic Channel pool size and Call timeout available in P25 and
MPT1327 configs
- Redesigned channel and decoder gui components
Changed C4FM symbol filter to reset frequency tracking values when the tuner's frequency or PPM values change.
Moved all frequency control pieces into the source.tuner.frequency package.
The CQPSK decoder is starting to take shape, but needs more tweaking and testing. It's not keeping a good lock on strong P25 signals. I have to add some more features to the intrumentation package to allow me to visualize the internals of the CQPSK decoder so that I can get a better sense of what needs fine tuning.
The CQPSK decoder may work for anyone that wants to sync the repo and run the build script.
Converted to use float primitives and to keep samples in buffer format until the sample rate is decimated to 48kHz.
Added new non-recursive, CIC decimation filter -- untested
Updated sound card/mixer input source management. Now testing for mono/stereo support for each card and limiting channel choices accordingly. Separated the mixersource into a ComplexMixer and and SimplexMixerSource. The ComplexMixer is used for sound card based tuners like the funcube. The SimplexMixerSource is used for demodulated sound card inputs.
Interfaces for sample handling - standardized on using 'Complex' to indicate I/Q samples and 'Simplex' to indicate demodulated float samples. The processing chain now accepts either a decimated (48k) list of complex samples from a tuner channel source, or a 48k simplex buffer (array of demodulated float samples).
Resolved issue with HackRF tuner not applying tuner configuration information.
