- Fix a bug where the initial audio data insertion would cause all
audio data to unintentionally clear (mixed up < and > operators, damn
human error)
- Fixed a potential interdependant lock scenario with channel mutex
locks and graphics mutex locks. The main video thread could lock the
graphics mutex and then while in the graphics mutex could lock the
channels mutex. Meanwhile in another thread, the channel mutex could
get locked, and then the graphics mutex would get locked, causing a
deadlock.
The best way to deal with this is to not let mutexes lock within
other mutexes, but sometimes it's difficult to avoid such as in the
main video thread.
- Audio devices should now be functional, and the devices in the audio
settings can now be changed as desired.
Modify the obs_display API so that it always uses an orthographic
projection that is the size of the display, rather than OBS' base size.
Having it do an orthographic projection to OBS' base size was silly
because it meant that everything would be skewed if you wanted to draw
1:1 in the display. This deoes mean that the callbacks must handle
resizing the images, but it's worth it to ensure 1:1 draw sizes.
As for the preview widget, instead of making some funky widget within
widget that resizes, it's just going to be a widget within the entire
top layout. Also changed the preview padding color to gray.
- Implement a means of obtaining default settings for an
input/output/encoder. obs_source_defaults for example will return
the default settings for a particular source type.
- Because C++ doesn't have designated initializers, use functions in
the WASAPI plugin to register the sources instead.
- Implement windows monitor capture (code is so much cleaner than in
OBS1). Will implement duplication capture later
- Add GDI texture support to d3d11 graphics library
- Fix precision issue with sleep timing, you have to call
timeBeginPeriod otherwise windows sleep will be totally erratic.
- Add WASAPI audio capture for windows, input and output
- Check for null pointer in os_dlopen
- Add exception-safe 'WinHandle' and 'CoTaskMemPtr' helper classes that
will automatically call CloseHandle on handles and call CoTaskMemFree
on certain types of memory returned from windows functions
- Changed the wide <-> MBS/UTF8 conversion functions so that you use
buffers (like these functions are *supposed* to behave), and changed
the ones that allocate to a different naming scheme to be safe
- Split input and output audio captures so that they're different
sources. This allows easier handling and enumeration of audio
devices without having to do some sort of string processing.
This way the user interface code can handle this a bit more easily,
and so that it doesn't confuse users either. This should be done for
all audio capture sources for all operating systems. You don't have
to duplicate any code, you just need to create input/output wrapper
functions to designate the audio as input or output before creation.
- Make it detect soundflower and wavtap devices as mac "output" devices
(even though they're actually input) for the mac output capture, and
make it so that users can select a default output capture and
automatically use soundflower or wavtap.
I'm not entirely happy about having to do this, but because mac is
designed this way, this is really the only way to handle it that
makes it easier for users and UI code to deal with.
Note that soundflower and wavtap are still also designated as input
devices, so will still show up in input device enumeration.
- Remove pragma messages because they were kind polluting the other
compiler messages and just getting in the way. In the future we can
just do a grep for TODO to find them.
- Redo list property again, this time using a safer internal array,
rather than requiring sketchy array inputs. Having functions handle
everything behind the scenes is much safer.
- Remove the reference counter debug log code, as it was included
unintentionally in a commit.
Categories added an unnecessary complexity to making properties, and
would very likely almost never be used in most cases, and were more of a
display feature. The main issue is that it made property data more
complex to work with, and I just didn't feel comfortable with that.
Also, added a function to allow you to retrieve a porperty just by its
name.
When a source/output/etc has a property of a 'list' type, there was no
way to get the names associated with its values. That, and it only
supported lists of either text, or enums (0..[value] only).
Now, you can associate translated names with those values, and use
integer, float, or string values. Put it all in to one function as well
to simplify its usage.
I plan on using this to help get enumerations from devices/etc for
certain types of sources. For example, if I get the properties of an
audio source, I'd like to have a list of available devices with it as
well.
- Signals and dynamic callbacks now require declarations to be made
before being used. What this does is allows us to get information
about the functions dynamically which can be relayed to the user and
plugins for future extended usage (this should have big implications
later for scripting in particular, hopefully).
- Reduced the number of types calldata uses from "everything I could
think of" to simply integer, float, bool, pointer/object, string.
Integer data is now stored as long long. Floats are now stored as
doubles (check em).
- Use a more consistent naming scheme for lexer error/warning macros.
- Fixed a rather nasty bug where switching to an existing scene would
cause it to increment sourceSceneRefs, which would mean that it would
never end up never properly removing the source when the user clicks
removed (stayed in limbo, obs_source_remove never got called)
See, it can sometimes be a bit confusing. These functions should
definitely not fail under normal circumstances, and these errors may
affect the user and/or application in some way.
LOG_ERROR should be used in places where though recoverable (or at least
something that can be handled safely), was unexpected, and may affect
the user/application.
LOG_WARNING should be used in places where it's not entirely unexpected,
is recoverable, and doesn't really affect the user/application.
- Add CoreAudio device input capture for mac audio capturing. The code
should cover just about everything for capturing mac input device
audio. Because of the way mac audio is designed, users may have no
choice but to obtain the open source soundflower software to capture
their mac's desktop audio. It may be necessary for us to distribute
it with the program as well.
- Hide event backend
- Use win32 events for windows
- Allow timed waits for events
- Fix a few warnings
the signals for scenes could have potentially conflicted with default
source signals. "remove" should be used for source removal, for
example. Changed the scene signals to "item-add" and "item-remove" for
its items.
Split off activate to activate and show callbacks, and split off
deactivate to deactivate and hide callbacks. Sources didn't previously
have a means to know whether it was actually being displayed in the main
view or just happened to be visible somewhere. Now, for things like
transition sources, they have a means of knowing when they have actually
been "activated" so they can initiate their sequence.
A source is now only considered "active" when it's being displayed by
the main view. When a source is shown in the main view, the activate
callback/signal is triggered. When it's no longer being displayed by
the main view, deactivate callback/signal is triggered.
When a source is just generally visible to see by any view, the show
callback/signal is triggered. If it's no longer visible by any views,
then the hide callback/signal is triggered.
Presentation volume will now only be active when a source is active in
the main view rather than also in auxilary views.
Also fix a potential bug where parents wouldn't properly increment or
decrement all the activation references of a child source when a child
was added or removed.
Implement a few audio options in to the user interface as well as a few
inline audio functions in audio-io.h.
Make it so ffmpeg plugin automatically converts to the desired format.
Use regular interleaved float internally for audio instead of planar
float.
This allows the changing of bideo settings without having to completely
reset all graphics data. Will recreate internal output/conversion
buffers and such and reset the main preview.
Make it so obs_data settings input in to *_update are applied to the
existing settings rather than fully replace the existing settings. That
way you can update with only certain specific settings, leaving other
settings untouched. Of course if you're already using the original
settings pointer in the first place then you've already done that, so
it'll just ignore it because you've already applied them.
- Remove obs_source::type because it became redundant now that the
type is always stored in the obs_source::info variable.
- Apply presentation volumes of 1.0 and 0.0 to sources when they
activate/deactivate, respectively. It also applies that presentation
volume to all sub-sources, with exception of transition sources.
Transition sources must apply presentation volume manually to their
sub-sources with the new transition functions below.
- Add a "transition_volume" variable to obs_source structure, and add
three functions for handling volume for transitions:
* obs_transition_begin_frame
* obs_source_set_transition_vol
* obs_transition_end_frame
Because the to/from targets of a transition source might both contain
some of the same sources, handling the transitioning of volumes for
that specific situation becomes an issue.
So for transitions, instead of modifying the presentation volumes
directly for both sets of sources, we do this:
- First, call obs_transition_begin_frame at the beginning of each
transition frame, which will reset transition volumes for all
sub-sources to 0. Presentation volumes remain unchanged.
- Call obs_source_set_transition_vol on each sub-source, which will
then add the volume to the transition volume for each source in
that source's tree. Presentation volumes still remain unchanged.
- Then you call obs_trandition_end_frame when complete, which will
then finally set the presentation volumes to the transition
volumes.
For example, let's say that there's one source that's within both the
"transitioning from" sources and "transition to" sources. It would
add both the fade in and fade out volumes to that source, and then
when the frame is complete, it would set the presentation volume to
the sum of those two values, rather than set the presentation volume
for that same source twice which would cause weird volume jittering
and also set the wrong values.
Now sources will be properly activated and deactivated when they are in
use or not in use.
Had to figure out a way to handle child sources, and children of
children, just ended up implementing simple functions that parents use
to signal adding/removal to help with hierarchial activation and
deactivation of child sources.
To prevent the source activate/deactivate callbacks from being called
more than once, added an activation reference counter. The first
increment will call the activate callback, and the last decrement will
call the deactivate callback.
Added "source-activate" and "source-deactivate" signals to the main obs
signal handler, and "activate" and "deactivate" to individual source
signal handlers.
Also, fixed the main window so it properly selects a source when the
current active scene has been changed.
Added a "master" volume for the entire audio subsystem.
Also, added a "presentation" volume for both the master volume and for
each invidiaul source. The presentation volume is used to control
things like transitioning volumes, preventing sources from outputting
any audio when they're inactive, as well as some other uses in the
future.
If audio was under, it originally did a full reset of the audio timing.
However, resetting the audio timing when this happens is kind of a bad
thing. It's better just to clamp the value to the expected timestamp to
ensure seamless audio output.
Also, implement audio timestamp smoothing to ensure audio tries to be as
seamless as possible.
I actually did compile that last commit and misread the failed projects
as 0. I'm just going to put the conversion stuff in video-io.h stuff
because it requires it anyway, and video-scaler.h already depends on
video-io.h for the video_format enum anyway.
Add a scaler interface (defaults to swscale), and if a separate output
wants to use a different scale or format than the default output format,
allow a scaler instance to be created automatically for that output,
which will then receive the new scaled output.
If there are for example more than one audio outputs and they have
different sample rates or channels and such, this will allow automatic
conversion of that audio to the request formats/channels/rates (but only
if requested).
- Changed glMapBuffer to glMapBufferRange to allow invalidation. Using
just glMapBuffer alone was causing some unacceptable stalls.
- Changed dynamic buffers from GL_DYNAMIC_WRITE to GL_STREAM_WRITE
because I had misunderstood the OpenGL specification
- Added _OPENGL and _D3D11 builtin preprocessor macros to effects to
allow special processing if needed
- Added fmod support to shaders (NOTE: D3D and GL do not function
identically with negative numbers when using this. Positive numbers
however function identically)
- Created a planar conversion shader that converts from packed YUV to
planar 420 right on the GPU without any CPU processing. Reduces
required GPU download size to approximately 37.5% of its normal rate
as well. GPU usage down by 10 entire percentage points despite the
extra required pass.
There were a *lot* of warnings, managed to remove most of them.
Also, put warning flags before C_FLAGS and CXX_FLAGS, rather than after,
as -Wall -Wextra was overwriting flags that came before it.
jp9000
fryshorts