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.
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
