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.
Potential existed for the audio output class to become unsynchronized
with the underlying source data line and attempt to perform a blocking
write to the data line when the data line doesn't have the capacity to
accept the full buffer, and the data line is not started. Thus,
playback could potentially stay in a persitent blocked state.
Refactored Mono and Stereo audio output implementations to push the
audio buffer queue processing into the parent abstract class.
AudioManager was updated to remove the last time active timestamp out
of the audio manager and down into the audio output to remove duplicate
timestamp accounting.
and constraints on scheduled thread pool saturation. Adds a new
scheduled thread pool for each processing chain that the components in
the chain can share, versus the single shared pool exposed by the
threadpoolmanager class.
-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
AudioManager connection class monitoring of the playback state. Adds a
line listener to monitor when the line is started/stopped and responds
accordingly.
Removes squelch control from mono and stereo audio playback components
which was a holdover -- squelch state is now managed within the
audio component for each processing chain.
with TO/FROM identifiers displayed, but no audio playing. This occurred
when the decoder sent a few audio buffers which caused the display to
update from the metadata, however the decoder stopped sending buffers
and the audio start threshold is never met. Clear display is invoked
upon audio stop, so the panel was never started or stopped or cleared
until the next call came in.
Changed to store the incoming metadata and only display it once the
audio playback starts and reset the display once the audio stops.
starting/stopping the monitor for each call to simply leaving the
monitor task running continuously. Was causing issue with scheduled
thread executor not shutting down the task in a timely manner -- new
approach should avoid start/stop related issues altogether.
Updates P25DecoderState with reordering of timing of when we send
updates for the TO/FROM identifiers so that the audio manager receives
up-to-date metadata and hopefully resolves issue with 'Do Not Monitor'
talkgroups being monitored by the audio manager.
Updates the AudioManager with a try/catch block around the processing
task so that we can catch any audio processing exceptions that occur
during the timer task and might get suppressed by the scheduled thread
executor.
Resolves issue in tuner controller where controller was not releasing
sourced tuner channels, preventing re-tuning to other frequencies.
Resolves issue with channel timeout on MPT1327 monitored traffic
channels
Redesigned audio manager handling of audio outputs to ensure that only
one channel's audio can be routed to an audio output at a time.
Changed audio outputs to use the shared thread pool manager for
scheduled tasks, instead of creating a thread pool for each audio
output.
while the buffer processor thread is running.
Added check for 0 space available in data line buffers to avoid trying
to write a zero-length audio chunk to the data line.
Added thread sleep for 100 milliseconds when we detect that the data
line buffer is full, to allow time for the data line to drain before we
check it again. Hopefully this should minimize the chattering during
playback when the code was repeatedly hitting the available() method to
check for space availability.
audio samples are produced to cause the audio channel to start playback
(doesn't exceed threshold), causing the audio channel to become locked
until there was new audio from the same talkgroup.
Updated p25 decoder state to better track calls and call detects from
a trunked channel
Fixed issue with false detects on LDU2 encrypted call messages
Fixed Airspy channel allocation ... will allocate channels across the
entire bandwidth, minus 50 kHz on either end, and will not adjust the
center frequency if it's not needed.
Fixed traffic channel teardown -- traffic channels are immediately
removed now, without a fade delay.
AudioMetadata - updated flag is now reset upon copying the metadata
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
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.
