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.
git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/branches/wsjtx@3588 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
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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2. The main window can be expanded wider to allow those with large screens to use larger fonts
3. New class DecodedText handles the formatted text from decoder.f90.
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Change source URLs in teh CMake scripts for the kvasd binaries.
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initialisation list to avoid a null pointer issue in restoring the
settings on startup.
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2. Remove offending redefinition of "struct timezone" in timeval.h, it's
no longer needed. (Let me know if this breaks anyone's build procedure!!)
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It's necessary to convey "appDir" from the GUI to jt9, so that jt9
can find the .lock and .quit files.
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2. Moved some of the text handling code from MainWindow to Displaytext
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Make the two sliders on main window have similar appearance.
Remove unused #ifdef block.
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2. The available character count is now updated by catching resizeEvents
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Detector no longer collects samples when transmitting or not
monitoring. This was causingissues with the waterfall after tx periods
and was a waste of CPU cycles.
Correct the missing window title on the waterfall window.
Stop and restart the output audio stream if the configuration of
device or channels changes in the setup dialog. This defect was
causing spurious transmissions after changing configuration.
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Audio input can be mono, left of stereo pair or, right of stereo
pair. Audio output can be mono, left of stereo pair, right of stereo
pair or, both of stereo pair (the same output goes to both channels in
both mode). Settings are remembered between sessions.
Stream channel suport is implemented mainly in the new AudioDevice
class which is now the base class of Modulator and Detector.
Audio channels are selected on the configuration screen. Only
supported channel configurations per device can be selected.
Audio output volume (actually attenuation) is now possible from the
GUI. I have added a slider control to the main window; I don't
necessarily propose this as a final release location for the widget as
I understand that changes to the main screen are sensitive. This
location is just a starting suggestion for a trial. The volume
(attenuation) setting is remembered between sessions and is not device
dependent. This addresses all issues of volume setting on *nix
versions since there is no need to use pavucontrol to set audio
levels. The volume (attenuation) action is logarithmic.
Shaped CW keying has been implemented in Modulator although it is
currently disabled as I am not 100% happy wth the implementation. If
you want to try it define the C++ preprocessor macro WSJT_SOFT_KEYING
in your build.
The Modulator instance has been moved to the same thread as the
SoundOutput instance as it should have been since the output callback
already operates in that thread. Cross thread slots are now correctly
called in a thread safe way as a result.
A number of files where in the SVN repository with DOS line endings
which I have removed. SVN users on Windows need set the config for
native line endings so that DOS line endings are automatically
stripped on checkin.
The DevSetup class now holds it's UI o the heap to reduce imapact on
build dependencies.
The application settings are now passed to objects from the main.cpp
file. Management of settings are moved to the responsible classes (top
level windows). This has involved a few settings moving groups so
users will see some settings reverting to default values on the first
run of an update.
Persistance of top level windows geometry and position is now handled
in the recommened manner (constructor for load, closeEvent for store
in modal windows and, hideEvent for store in modeless dialogs).
The MainWindow class now holds its children as members rather than
global variables.
The LogQSO class now hides its implementation and takes responsibility
for its own settings and widows rendering parameters. A new settings
file group is implemented to persist the LogQSO class settings.
The WideGraph class now hides its implementation and manages its own
settings and window rendering parameters.
--This line, and those below, will be ignored--
M Modulator.cpp
M rigclass.cpp
M widegraph.cpp
M signalmeter.cpp
M soundin.cpp
M soundout.cpp
M mainwindow.h
M main.cpp
M meterwidget.h
M devsetup.cpp
M mainwindow.ui
M Detector.cpp
M logqso.h
M rigclass.h
M mainwindow.cpp
M meterwidget.cpp
M soundin.h
M devsetup.ui
M wsjtx.pro
M devsetup.h
M logqso.cpp
M Modulator.hpp
M psk_reporter.cpp
M killbyname.cpp
M Detector.hpp
M signalmeter.h
M widegraph.h
M psk_reporter.h
M soundout.h
M PSKReporter.h
M lib/afc65b.f90
M lib/gran.c
M lib/usleep.c
M lib/afc9.f90
M lib/wrapkarn.c
A AudioDevice.hpp
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2. Fix jt9 so that decoding starts at fStart (which now may be non-zero).
3. Improved or new Tool Tips for some GUI controls.
4. Updates to User's Guide.
git-svn-id: svn+ssh://svn.code.sf.net/p/wsjt/wsjt/branches/wsjtx@3540 ab8295b8-cf94-4d9e-aec4-7959e3be5d79
-changed log to hash table for faster lookup
-improved ADIF compatibility
mainwindow.ui
-fixed typo in menu entry
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2. Remove (or comment out) most qDebug() statements.
3. Swap positions of labels 2 and 3 on status bar.
4. In "Split Tx" mode, transmit audio always between 1500 and 2000 Hz.
5. Rearrange positions of WideGraph controls.
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