- Fixes BFont::GetHasGlyphs, the "empty square" which was returned led
it to think the font had glyphs for everything
- This means the "no character" empty square will not be drawn anymore,
if we want it back, we will need to rework the implementation a bit
more (either request it explicitly when there is a missing glyph, or
return it as it was before but including an info that it is the
"missing glyph")
- Maybe GetHasGlyphs should also bypass the font fallback system, and
return what's actually in the requested font only.
This also fixes a locking problem in the GlyphLayoutEngine, the code
didn't handle the read/write lock properly and tried to ReadUnlock from
a place where only the write lock was held.
- Interface Kit tests use a View class which is not BView, and has no
SetViewUIColor.
- Libexpat is now in a separate package.
- Shouldn't we add this to the buildbots so we can cath the breakage?
Avoids showing the same language multiple times in the list, for example
when there are cyrillic and latin variants. It is still possible to pick
one of the variants, as they are also added as country-specific entries.
Fixes#9144.
- Don't crash when a [.svn] cache directory is missing
- Ignores the cache directories of the main VCS : bazaar, git, mercurial, svn
Signed-off-by: Adrien Destugues <pulkomandy@pulkomandy.tk>
Fixes#11532.
url and email label are now marked as a link and open the address
in the browser/mail-app on click
Signed-off-by: Adrien Destugues <pulkomandy@pulkomandy.tk>
Fixes#3825
A few extra changes from the original patch:
- Remove "smart" parsing (detection of "ftp", etc) as it could lead to
ftp://ftp://url or other strangeness,
- Add gopher protocol, because mmu_man may have an home page there.
A combination of two problems made things go wrong with the timestamp of
decoded audio.
1) The output buffer size is too small to hold the complete input.
swresample handles this by buffering the input for use the next time it
is called, however repeatedly doing this results in lots of buffering,
and our way to compute the output timestamp from the input does not take
it into account so it does weird things. Moreover, we would need to
empty the buffer by calling swr_convert with NULL input in that case.
Fix: make sure to not feed more data to swr_convert than it can output
in our buffer. This way, no buffering occurs, only the matrixing
conversion.
2) When using planar audio, the "frame size" is a bit different. Instead
of adding sample size * channel count to 1 pointer, we need to add
sample size * 1 to each channel buffer.
Fix: add the "fInputFrameSize" which takes this into account, instead of
misusing fOutputFrameSize for the input.
Fixes#12460.
* While on my system it looked like to run well, it introduced
a regression on some hardware. I'm reverting it as I don't have
a better solution, with the difference that we will do that only
for perfomance time events.
* Fixes#12624.
* Messages that we don't have info for during synchronization are now
collected, and synchronized once the folder state is available,
instead of waiting for that particular entry to become available
(without proper means to wait).
* The counter did not take continuation commands into account, so that
it would never reach zero again.
* Optimized IMAPConnectionWorker::_WaitForCommands() such that it will
always acquire all available "slots".
* Since the BMediaRecorder have an it's own estabilished policy
relating releasing the producer node we will not interfere with
it except when we are controlled from Cortex.
* SoundRecorder can now be connected and disconnected safely,
due to some unknown reason, probably format negotiation problems,
ATM we need a system mixer between.
* The normal behavior is unchanged, when the record button is pressed
it will record from the preferred audio interface and disconnect on
stop. But when the connection is made manually it will stay connected
until the user explictly disconnect it.
* To make it work, instantiate a system mixer, connect the output to
SoundRecorder, instantiate a node like the ToneProducer, and connect
it's output to the mixer input. Press the record button and check
the Mixer has started from the Cortex transport, if not, start it.
* This required to review various parts of the code, and
isn't probably still perfect. The main problem was an attributes
hell where redondance created a lot of problems, all this data
is now controlled mostly by the node.
* Header indentation changes needed too.
* The WaitForMessage is moved at the end of the loop, and
the check for quitting the loop is moved as it's condition,
this makes us to be sure that once a Quit() request is performed
we will not handle events anymore.
* Other minor style and initialization fixes.
* This is the only solution that allowed to use the best
of both ways to do this calculus. I've also tested it
with a modified sound player that snoozed every time
the buffer should be handled, and found that neither
of the lateness calculus I tested (including enqueue_time)
really solve all problems. That's why I've tried to find
an average solution. There's still room for improvements
eventually.
