bad data.
2. Fix a bug that could allow a Koetter-Vardy false decode instead of
a valid Berlekamp-Massey decode, sometimes leading to program crash.
3. Many more edits in the User's Guide, *.adoc files.
git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/branches/wsjtx@3664 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
* Moved doc/source/*.txt to AsciiDoc ext source/*.adoc
- Correctly identifies AsciiDoc files
* wsjtx-main.adoc
- Updated links and include:: for *.adoc name change
* Added and updated rig-config-* files.
* Added inital draft of quick-reference.adoc
* build-doc.sh
- Updated the script to build new files and .adoc name change
* Removed:
- yaesu.txt rigtemplate.txt rig-configuration.txt
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smaller files, by section and sub-section.
Also extensive editing up through section 7. Sections 6.2, 8, and
beyond definitely need work. Other polishing is also desirable, and
maybe also some additions and/or changes.
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applicable, JT65v2 otherwise. Thanks to DL9RDZ!
2. Fix a bug in flat3.f90, evident on OSX systems. Thanks to G4KLA!
3. The Makefile now copy the jt9code executable into the destination
directory for to-be-packaged executables.
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Added code to flush any queued spots to PSKReporter prior to a band change. This should prevent spots being reported on a wrong band.
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Add -DUNIX flag to FFLAGS in Makefile.linux.
Tweak the use of framesAfterDownsample in Detector.cpp.
git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/branches/wsjtx@3631 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
Windows Vista has a broken rate converter which gets invoked when an
input audio stream at 48kHz sampel rate is requested. I've no idea why
our application can't get exclusive access to the audio input device
and have a unconverted stream direct at 48kHz.
To get around this our down sampling filter for audio input from 48kHz
to 12kHz is disaabled by default on Windows Vista, instead we request
a 12kHz stream and process it directly.
This default behviour can be overriden by specifying the following
settings value:
[Tune]
Audio\DisableInputResampling=false
This settings value defaults to true on Windows Vista and false
everywhere else so normally needn't be present.
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The ".quit" file that is used to communicate shutdown to the "jt9"
sub-process was not getting deleted under some obscure
circumstances. If "wsjtx" is started with the ".quit" file in place;
"jt9" starts and stops immediately rather than going into it's normal
wait state. Probably some sort of race condition between the two
processes dying and file and/or shared memory locks.
Rather than track the issue down I have added code to ensure that
".quit" is removed before starting "jt9".
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The filter used for 4 times down-sampling cannot handle sample streams
where the hardware or drivers deliver chunks of data that are not
multiples of 4 frames long. This seems to be prevalent on some Linux
platforms. Also de-interleaving of single channel audio from stereo
streams was no longer supported.
I have changed the input strategy to de-interleave the incoming
sample stream into an intermediate buffer large enough to hold all the
samples required for a single unit of processing (one basic waterfall
interval) and apply the down-sampling filter to the whole intermediate
buffer just prior dispatch to the FFT generator.
This now means that we are now using the ubiquitous 48kHz hardware
sample rate for both input and output of audio across all platforms
and decoding a single channel of a stereo stream is again
supported. The down-sampling to 12kHz is done with a high quality FIR
49-tap low pass filter specifically designed by Joe (K1JT) for
operation in a 4kHz bandwidth.
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2. Removed commented code left over from previous edits in MainWindow::doubleClickOnCall()
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48kHz to 12kHz down sampler routine.
Added assert to disallow stereo processing of i/p samples
until fil4.f90 can deal with interleaved stereo streams.
Added QProcess error to jt9 error handler, not that anything
is done with the error code yet but at least it can be examined
in the debugger if required.
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Please note that I removed the option "-mno-stack-arg-probe" from CFLAGS.
Was there any good reason for it still being there?
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By adding the following section to the initialisation file the audio buffer
sizes and audio thread priority may be adjusted.
[Tune]
Audio\InputBufferFrames=1200
Audio\OutputBufferMs=1000
Audio\ThreadPriority=4
The values above are the program defaults that will be used if the
initialisation parameters are omitted.
Thread prioritis are the QThread::Priority enumumeration values.
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2. Clear the "Name" field after logging a QSO.
3. Add fil4.f90, a subroutine for downsampling from 48000 Hz to 12000 Hz
sample rate. (Not yet incorporated in WSJT-X, but intended to be called
from arounf line 51 in Detector.cpp.)
4. Minor updates to User's Guide.
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Greg Beam
Joe Taylor
Bill Somerville
Edson W. R. Pereira
Murray Curtis
John Nelson