Resolves issue with MP3 framing where the remote server was disconnecting us due to bad MP3 sync. The LAME encoder was sending partial 20 byte MP3 frames initiall, then following with a 144 byte array which contained the remainder of the first frame and the header of the next frame. Adjusted the MP3SilenceGenerator to only emit complete MP3 frames.
This should be the final commit for this feature, pending merge back into master branch.
Note: Shoutcast V1 streaming to a Shoutcast V2.5.x server seems to be problematic. The newer server tracks MP3 boundaries in order to insert advertising and this appears to have problems keeping sync with the MP3 stream. I spent significant time eliminating MP3 framing errors in my code. You can see these 'Bad frame' errors in the sc_serv.log and the shoutcast server will eventually reset the connection after a period of time or after a number of framing errors occur. I anticipate that streaming to a Shoutcast V2 server using the ultravox (ie V2.x) protocol may resolve this issue.
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.
functionality so that the editor starts up without a channel displayed
and as each channel row is selected in the channels table, the editor is
updated with the channel configuration.
Likewise, the channel editor is now a channel event listener on the
channel model and will maintain channel state if the user changes
channel configuration in any of the external channel controllers like
the spectral display or the channel list.
-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
bug where Linux runs out of FileDescriptors, even though we're
explicitly closing out file channels. Error handler sets flag to
prevent any further audio wave recorders from being started and logs the
same.
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.
per-call recording and accurate state tracking.
Resolved issue with ThreadPoolManager limiting total number of scheduled
tasks (removed limiter - original intent is now OBE)
Updated ChannelState to better handle auxiliary decoder decode events,
so that when a fleetsync ident is at the end of the transmission, the
decode doesn't kick the channel back into a call state after it has
faded, causing static bursts and additional empty recordings.
Updated recording file name to include millisecond level precision, to
further avoid duplicate recording names
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
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.
