For a path "D:/foo/bar-$Y/$M/$T.flv" this will return at least "D:/foo"
(and create "D:/foo" if the directory doesn't exist) or at most
"D:/foo/bar-$Y/$M" with $Y and $M expanded, if those directories exist
This allows better backward compatibility with older versions. This
will keep the current scene collection set to default if the user ever
decides to downgrade OBS versions for whatever reason.
Fixes issues where the desktop device would drop and reacquire then stop providing audio due to no audio being played, causing OBS to freeze and break horribly.
This changes the semantics of SaveToFile from "disable all file output
when disabled" to "automatically save stream to file when enabled",
which allows the "Start Recording" button to be enabled even when
SaveToFile is disabled.
Having the "Start Recording" button enabled now depends on having a
non-empty SavePath
BufferSize settings in the config files will be ignored as long as
the checkbox is unchecked (UseBufferSize is unset or 0), to make
the contents of the settings dialog reflect actual encoder behavior
Caused us no end of pain. Users clicked it so much that it made us all
snap. Now, if a user really, REALLY wants to use it, they'll have to
manually set the ini file variable in their profile ini, under [Audio],
InputDevicesForDesktopSound=1
- Add an option in game capture to use SetWindowsHookEx instead of
CreateRemoteThread to inject the library.
- Add logging of what process game capture is trying to capture.
- Rename "audio time offset" where it was used to "audio sync offset"
- Adde logging for audio sync offsets
- Update version to 0.62b test 1
This should alleviate the impact of enabling "Use Input devices ..." in
the advanced config
Also add a log message to signify that "Use Input devices ..." is enabled
This will automatically update profiles to the new encoder selection
according to the old setting resolution, i.e. UseQSV && UseNVENC results
in QSV being selected.
This should free up some space to allow more encoder troubleshooting and
diagnostics in the settings window (relevant QSV changes coming soon ...)
Also hopefully made supported encoder settings a bit more clear by
disabling more settings if the selected encoder doesn't support them
Applies only when in file only mode or with stream paused while file
output is enabled
Still need to evaluate not closing the file prematurely while stream is
active with a large number of buffered frames in the encoder
These complement the new start/stop recording events as they are called
as soon as a network connection is initiated/severed regardless of
previous encoder state
This enforces the warning that settings will not be applied until
streaming is restarted from the settings dialog for all profile settings
Also fixes a bug that would prevent start streaming/recording buttons
from being properly enabled/disabled when switching from a
recording/streaming only profile to a streaming/recording only profile
on stream stop
Additionally, the microphone volume meter got disabled when switching
from a profile with microphone enabled to a profile with microphone
disabled while the microphone was still active
Move the keep recording setting from global.ini to Broadcast Settings
(profile dependent), make it default to off, comment out the old
dashboard link code for now.
- I needed to change the parameter for my hack, so I'm just going to
create a new function instead, QueryAudio2. C++ mangling and API
breakage makes me most displeased.
- Woops, don't use global audio time for sorting, use the latest audio
timestamp from that specific device.
- Also, made it so lastSentTimestamp is only done when data is actually
used, accidentally put it in the wrong scope.
- Fixed an issue where audio data that came in bursts would cause
desyncs. Basically, if data came in too late, it would continue to
buffer little by little, causing progressive desync with certain
microphones and capture devices (avermedia stream engine for the live
gamer portable especially). Also seemed to happen with HDMI data
from the newer game consoles, like xbox one apparently, though I
can't be too sure.
Now, it queries the mic and auxilary sound sources until sound
buffers are depleted. After doing so, it then "sorts" the audio
packets timestamps backwards from the most recent packet to the
oldest audio packet. By doing this, it compensates for burst, and
ensures that all audio data is seamless. New burst data coming in
will then line up properly with the older data via the sort function.
NOTE: This needs testing
