MagickCore is provided here as an alternative to FFmpeg in case FFmpeg
is not easily supported (for example, debian-based linux distros).
NOTE: Cmake configuration needs to be changed in order to allow
MagickCore image support.
NOTE: In texture_setimage, I had to move variables to the top of the
scope because microsoft's C compiler will give the legacy C90 error of:
'illegal use of this type as an expression'.
To sum it up, microsoft's C compiler is still utter garbage.
Similar to the shader functions, the effect parameter functions take
the effect as a parameter. However, the effect parameter is pretty
pointless, because the effect parameter.. parameter stores the effect
pointer interally.
...I'm actually concerned that I went a bit overkill trying to prevent
backwards compatibility issues with this abstraction design, because
this is a large number of files that have to be modified just to add a
single graphics subsystem export. Someone's going to strangle me, and
when you know that someone might strangle you, that means that you did
something wrong. We'll have to look in to simplifying this in the
future without killing backward compatibility safety.
The module callback obs_module_set_locale will be called after loading
the module, and any time the locale is manually changed via core API.
When this function is called, the module is expected to load new text
lookup values for all the text it uses based upon the current locale.
The locale parameter was a mistake, because it puts extra needless
burden upon the module developer to have to handle this variable for
each and every single callback function. The parameter is being removed
in favor of a single centralized module callback function that
specifically updates locale information for a module only when needed.
Having the value stored here is somewhat pointless, so this is one step
in fixing the locale handling. Locale should be handled by the modules
themselves with their own loaded locale lookup information.
This API is used to set the current locale for libobs, which it will set
for all modules when a module is loaded or specifically when the locale
is manually changed.
These functions were mostly related to being able to set true fullscreen
mode -- however, this has no place for our purposes, and these functions
were just sitting empty and unused, so they should be removed.
Besides, fullscreen mode only applies to the windows operating system.
This variable is currently somewhat pointless, I was originally going to
use it to tell the graphics subsystem to completely rebuild the internal
vertex buffers, but it would be bad/inefficient to allow that
functionality.
These are meant to reflect auto-detection configuration changes that
should not be written to the config, for example, frame rate changes
for a camera where the (user-/config-file-)configured frame rate isn't
available but a similar frame rate can be automatically chosen
Default values are now permanently stored in the obs_data_items and
can be accessed via the new get_default functions
Also default values are no longer serialized to JSON to ease transition
to new default values
This allows, for example, disconnected devices for dshow/avcapture to
be listed (and to stay selected in the user config) while making them
unavailable for selection in new sources
The 'initialize' callback is used before the encoders/output start up so
it can adjust encoder settings to required values if needed.
Also added the function 'obs_encoder_active' that returns true or false
depending on whether that encoder is active or not.
Structures with anonymous unions would a warning when you do a brace
assignment on them.
Also fixed some unused parameters and removed some unused variables.
So, scene editing was interesting (and by interesting I mean
excruciating). I almost implemented 'manipulator' visuals (ala 3dsmax
for example), and used 3 modes for controlling position/rotation/size,
but in a 2D editing, it felt clunky, so I defaulted back to simply
click-and-drag for movement, and then took a similar though slightly
different looking approach for handling scaling and reszing.
I also added a number of menu item helpers related to positioning,
scaling, rotating, flipping, and resetting the transform back to
default.
There is also a new 'transform' dialog (accessible via menu) which will
allow you to manually edit every single transform variable of a scene
item directly if desired.
If a scene item does not have bounds active, pulling on the sides of a
source will cause it to resize it via base scale rather than by the
bounding box system (if the source resizes that scale will apply). If
bounds are active, it will modify the bounding box only instead.
How a source scales when a bounding box is active depends on the type of
bounds being used. You can set it to scale to the inner bounds, the
outer bounds, scale to bounds width only, scale to bounds height only,
and a setting to stretch to bounds (which forces a source to always draw
at the bounding box size rather than be affected by its internal size).
You can also set it to be used as a 'maximum' size, so that the source
doesn't necessarily get scaled unless it extends beyond the bounds.
Like in OBS1, objects will snap to the edges unless the control key is
pressed. However, this will now happen even if the object is rotated or
oriented in any strange way. Snapping will also occur when stretching
or changing the bounding box size.
There are a ridiculous number of features related to scaling and
positioning due to requests by a number of people who complained that
they hated the way that OBS1 would always resize their sources when the
source's base size changed. There were also people who wanted more
control for how the resizing was handled, or the ability to completely
prevent resizing entirely if desired. So I made it so that you can
optionally use a 'bounds' system, which allows you to specify different
styles of controlling resizing.
If disabled, the source will always automatically resize and only the
base scale is applied. If enabled, you have a variety of different ways
to limit/control how it can resize within the bounds, or make it so it
can't resize at all. You can also control alignment within that
bounding box, so you can make it so that a source always aligns to a
side or corner of the box.
I also added an alignment value which changes how the source is oriented
relative to the position of the scene item. For example, setting
bottom-right alignment will make it so that the position of the item is
the bottom right corner of the source. When the source resizies, it
will resize leftward and upward in that case, which solves the problem
of how a source resizes relative to a desired position.
I encountered a situation where I wanted to delete a callback for a
signal while inside of that signal. However it would hard lock, and
even after that, it would mess up the loop for the callback list.
So, change the mutex of the individual signals to a recursive-style
mutex, and then if a callback of a signal is deleted while currently in
that signal, just mark it for deletion, which will happen after the
signal is complete.
This replaces the older code which simply queried the max volume level
value for any given audio.
I'm still not 100% sure on if this is how I want to approach the
problem, particularly, whether this should be done in obs_source or in
audio_line, but it can always be moved later if needed.
This uses the calculations by the awesome Bill Hamilton that OBS1 used
for its volume levels. It calculates the current max (level),
magnitude, and current peak. This data then can be used to create
awesome volume meter controls later on.
NOTE: Will probably need optimization, does one float at a time right
now.
Also, change some of the naming conventions. I actually need to change
a lot of the naming conventions in general so that all words are
separated by underscores. Kind of a bad practice there on my part.
There was a fundamental flaw with the string type conversion functions
where the sizes were not being properly accounted for. They were using
the 'len' value as a value for the output rather than only for the
input, which was bad because the output could have more or less
characters than the input.
When a source's private data is being created by a module, it wasn't
able to call most source functions because most functions rely on the
obs_source_info part of the context to be set. This fixes that issue.
It was strange that this wasn't already the case because the other
context types already did the same thing.
This uses the reverse planar YUV 4:2:0 conversion shader to output a YUV
texture without having to convert it via CPU. Again, this will reduce
video upload bandwidth usage to 37.5% of the original rate. I suspect
this will be particularly useful for when an FFmpeg or libav input
plugin for playing videos is made.
NOTE: There's an issue with certain texture sizes right now I haven't
been able to identify, if the full size of texture data divided by the
base texture width is an uneven number, the V chroma plane seems like it
can potentially shift, though I only had this happen with 160x90
resolution C920. Almost all resolutions tend to be even. Needs further
testing with more devices that support planar YUV 4:2:0 output.
This adds button support to properties, which will allow a properties
pane to let the user click a button to activate something with a
particular obs context. When pressed, the button will execute the
callback given, with the context's private data as a parameter.
Character conversion functions did not previously ask for a maximum
buffer size for their 'dst' parameter, it's unsafe to assume some given
destination buffer may have enough size to accommodate a conversion.
Added github gist API uploading to the help menu to help make problems a
bit easier to debug in the future. It's somewhat vital that this
functionality be implemented before any release in order to analyze any
given problem a user may be experiencing.
This fixes an issue reported by valgrind where overlapping memory
was copied with memcpy.
This also removes a redundant assignment where the array size was
explicitly set to zero when it was already zero.
This doesn't add FLV file output to the user interface yet, but we'll
get around to that eventually. This just adds an FLV output type.
Also, removed ftello/fseeko because off_t is a really annoying data
type, and I'd rather have a firm int64_t for large sizes, so I named it
to os_fseeki64 and os_ftelli64 instead, and changed the file size
function to return an int64_t.
Not entirely sure how this happened but I *think* that a null source was
somehow being added to the list of user sources for one particular user,
and then I noticed this code does not check to see whether the source is
null or not.
Just use platform-nix.c code for general stuff that mac is compliant
with, and put a define around everything else. Take that code out of
platform-cocoa.m.
Added os_opendir, os_readdir, and os_closedir to be able to query
available files within a directory.
First, if the private data of the source fails to be created, then do
not destroy the source. If the source is destroyed, all the user's data
associated with that source is lost, which could end up being a
potential problem. Instead, let it linger as a 'dead' source until the
user chooses to fix the problem (though this should never really happen,
the source module functions should be programmed to handle this
scenario)
Secondly, rename new_frame_ready to ready_async_frame, and fix a
potential memory leak with it.
obs_source_output_video can cause cached frames to be freed twice if
called with a partially destroyed source, among other undesirable
effects; freeing the source private data right after the destroy signal
has been processed ensures proper behavior
- Add volume control
These volume controls are basically nothing more than sliders. They
look terrible and hopefully will be as temporary as they are
terrible.
- Allow saving of specific non-user sources via obs_load_source and
obs_save_source functions.
- Save data of desktop/mic audio sources (sync data, volume data, etc),
and load the data on startup.
- Make it so that a scene is created by default if first time using the
application. On certain operating systems where supported, a default
capture will be created. Desktop capture on mac, particularly. Not
sure what to do about windows because monitor capture on windows 7 is
completely terrible and is bad to start users off with.
If a source with async video wasn't currently active, it would endlessly
buffer the video data, which would cause memory to grow endlessly until
available memory was extinguished.
This really needs to be replaced with a proper caching mechanism at some
point.
This saves scenes/sources from json on exit, and properly loads it back
up when starting up the program again, as well as the currently active
scene.
I had to add a 'load' and 'save' callback to the source interface
structure because I realizes that certain sources (such as scenes)
operate different with their saved data; scenes for example would have
to keep track of their settings information constantly, and that was
somewhat unacceptable to make it functional.
The optional 'load' callback will be called only after having loaded
setttings specifically from file/imported data, and the 'save' function
will be called only specifically when data actually needs to be saved.
I also had to adjust the obs_scene code so that it's a regular input
source type now, and I also modified it so that it doesn't have some
strange custom creation code anymore. The obs_scene_create function is
now simply just a wrapper for obs_source_create. You could even create
a scene with obs_source_create manually as well.
The 'wait' constant was a terrible means of trying to ensure that the
packets were interleaved. Instead, calculate the current highest
timestamps of each encoder that's present in the interleaved buffer, and
use that as a means of detecting whether the current packet should be
sent off. This will guarantee sorting without relying on some arbirary
constant that 'assumes' that it'll be interleaved. It also reduces
buffering any more than what is needed to interleave.
- Updated the services API so that it links up with an output and
the output gets data from that service rather than via settings.
This allows the service context to have control over how an output is
used, and makes it so that the URL/key/etc isn't necessarily some
static setting.
Also, if the service is attached to an output, it will stick around
until the output is destroyed.
- The settings interface has been updated so that it can allow the
usage of service plugins. What this means is that now you can create
a service plugin that can control aspects of the stream, and it
allows each service to create their own user interface if they create
a service plugin module.
- Testing out saving of current service information. Saves/loads from
JSON in to obs_data_t, seems to be working quite nicely, and the
service object information is saved/preserved on exit, and loaded
again on startup.
- I agonized over the settings user interface for days, and eventually
I just decided that the only way that users weren't going to be
fumbling over options was to split up the settings in to simple/basic
output, pre-configured, and then advanced for advanced use (such as
multiple outputs or services, which I'll implement later).
This was particularly painful to really design right, I wanted more
features and wanted to include everything in one interface but
ultimately just realized from experience that users are just not
technically knowledgable about it and will end up fumbling with the
settings rather than getting things done.
Basically, what this means is that casual users only have to enter in
about 3 things to configure their stream: Stream key, audio bitrate,
and video bitrate. I am really happy with this interface for those
types of users, but it definitely won't be sufficient for advanced
usage or for custom outputs, so that stuff will have to be separated.
- Improved the JSON usage for the 'common streaming services' context,
I realized that JSON arrays are there to ensure sorting, while
forgetting that general items are optimized for hashing. So
basically I'm just using arrays now to sort items in it.
Add API for streaming services. The services API simplifies the
creation of custom service features and user interface.
Custom streaming services later on will be able to do things such as:
- Be able to use service-specific APIs via modules, allowing a more
direct means of communicating with the service and requesting or
setting service-specific information
- Get URL/stream key via other means of authentication such as OAuth,
or be able to build custom URLs for services that require that sort
of thing.
- Query information (such as viewer count, chat, follower
notifications, and other information)
- Set channel information (such as current game, current channel title,
activating commercials)
Also, I reduce some repeated code that was used for all libobs objects.
This includes the name of the object, the private data, settings, as
well as the signal and procedure handlers.
I also switched to using linked lists for the global object lists,
rather than using an array of pointers (you could say it was..
pointless.) ..Anyway, the linked list info is also stored in the shared
context data structure.
Just wanted the ability to be able to add private data to the properties
data. Makes it a little easier to manage data if you get updates from
controls.
Before, async video sources would flicker because they were only being
drawn when they were updated. So when updated, they'd draw that frame,
then it would stop drawing it until it updated again. This fixes that
issue and they should now draw properly.
Also, fix a few other minor bugs and issues relating to async video,
and make it so that non-async video filters can be properly applied to
them.
For the purposes of testing, change the 'test-random' source to an async
video source that updates every quarter of a second with a new random
face.
Also fix a bug where non-async video sources wouldn't have filter
effects applied properly.
A little bit of history about frame dropping:
I did a large number of experiments with frame dropping in old versions
of OBS1, and it's not an easy thing to deal with. I tried just about
everything from standard i-frame delay, to large buffers, to dumping
packets, to super-unnecessarily-complex things that just ended up
causing more problems than they was worth.
When I did my experiments, I found that the most ideal frame drop system
(in terms of reducing the amount of total data that needed to be
dropped) was in the 0.4xx days where I had a 3 second frame-drop buffer
where I could calculate the actual buffer size in bytes, and then
intellgently choose packets in that buffer to trim it down to a specific
size while minimizing the number of p-frames and i-frames dropped, and
preventing the actual impact of dropped frames on the stream. The
downside of it was that it required too much extra latency, and far too
many people complained about it, so it was removed in favor of the
current system.
The current system I just refer to just as 'packet dumping', which when
combined with low keyframe intervals (like most services use these
days), is the next-best method from my experience. Just dump the buffer
when you reach a threshold of buffering (which I prefer to measure with
time rather than in size), then wait for a new i-frame. Simple,
effective, and reduces the risk of consecutive buffering, while still
having fairly low impact on the stream output due to the low keyframe
interval of services.
By the way, audio will not (and should not ever) be dropped, lest you
end up with syncing issues (among other nasty things) specific to server
implementation.
- Fix an issue that could occur when using more than one video encoder.
Audio/video would not sync up correctly because they were expected to
be paired with a particular encoder. This simply adds a little
helper variable to encoder packets that specifies the system time in
microseconds. We then use that system time to sync
- Fix an issue with x264 with fractional FPS rates (29.97 and 59.94
particularly) where it would create ridiculously large stream
outputs. The problem was that you shouldn't set the timebase_*
variables in the x264 params manually, let x264 handle the default
values for it and leave them at 0.
- Make x264 use CFR output, because there's no reason to ever use VFR
in this case.
- Implement the RTMP output module. This time around, we just use a
simple FLV muxer, then just write to the stream with RTMP_Write.
Easy and effective.
- Fix the FLV muxer, the muxer now outputs proper FLV packets.
- Output API:
* When using encoders, automatically interleave encoded packets
before sending it to the output.
* Pair encoders and have them automatically wait for the other to
start to ensure sync.
* Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop'
because it was a bit confusing, and doing this makes a lot more
sense for outputs that need to stop suddenly (disconnections/etc).
- Encoder API:
* Remove some unnecessary encoder functions from the actual API and
make them internal. Most of the encoder functions are handled
automatically by outputs anyway, so there's no real need to expose
them and end up inadvertently confusing plugin writers.
* Have audio encoders wait for the video encoder to get a frame, then
start at the exact data point that the first video frame starts to
ensure the most accrate sync of video/audio possible.
* Add a required 'frame_size' callback for audio encoders that
returns the expected number of frames desired to encode with. This
way, the libobs encoder API can handle the circular buffering
internally automatically for the encoder modules, so encoder
writers don't have to do it themselves.
- Fix a few bugs in the serializer interface. It was passing the wrong
variable for the data in a few cases.
- If a source has video, make obs_source_update defer the actual update
callback until the tick function is called to prevent threading
issues.
I was getting cases where the CPU cache was causing issues with the
allocation counter, for the longest time I thought I was doing something
wrong, but when the allocation counter went below 0, I realized it was
because I didn't use atomics for incrementing/decrementing the
allocation counter variable. The allocation counter now always should
have the correct value.
- Add interleaving of video/audio packets for outputs that are encoded
and expect both video and audio data, sorting the packets and sending
them to the output when both video and audio is received.
- Combine create and initialize callbacks for the encoder API callback
interface.
Improve the properties API so that it can actually respond somewhat to
user input. Maybe later this might be further improved or replaced with
something script-based.
When creating a property, you can now add a callback to that property
that notifies when the property has been changed in the user interface.
Return true if you want the properties to be refreshed, or false if not.
Though now that I think about it I doubt there would ever be a case
where you would have this callback and *not* refresh the properties.
Regardless, this allows functions to change the values of properties or
settings, or enable/disable/hide other property controls from view
dynamically.
- Add start/stop code to obs-output module
- Use a circular buffer for the buffered encoder packets instead of a
dynamic array
- Add pthreads.lib as a dependency to obs-output module on windows in
visual studio project files
- Fix an windows export bug for avc parsing functions on windows.
Also, rename those functions to be more consistent with each other.
- Make outputs use a single function for encoded data rather than
multiple functions
- Add the ability to make 'text' properties be passworded
- obs-outputs module: Add preliminary code to send out data, and add
an FLV muxer. This time we don't really need to build the packets
ourselves, we can just use the FLV muxer and send it directly to
RTMP_Write and it should automatically parse the entire stream for us
without us having to do much manual code at all. We'll see how it
goes.
- libobs: Add AVC NAL packet parsing code
- libobs/media-io: Add quick helper functions for audio/video to get
the width/height/fps/samplerate/etc rather than having to query the
info structures each time.
- libobs (obs-output.c): Change 'connect' signal to 'start' and 'stop'
signals. 'start' now specifies an error code rather than whether it
simply failed, that way the client can actually know *why* a failure
occurred. Added those error codes to obs-defs.h.
- libobs: Add a few functions to duplicate/free encoder packets
The serializer code is meant to be used as a means of reading/writing
data from any arbitrary type of input/output.
The array output serializer makes it so we can stream data to a dynamic
array on the fly.
- Add dummy GL texture support to allow libobs texture references to be
created for GL without
- Add a texture_getobj function to allow the retrieval of the
context-specific object, such as the D3D texture pointer, or the
OpenGL texture object handle.
- Also cleaned up the export stuff. I realized it was all totally
superfluous. Kind of a dumb moment, but nice to clean it up
regardless.
- Make it so that encoders can be assigned to outputs. If an encoder
is destroyed, it will automatically remove itself from that output.
I specifically didn't want to do reference counting because it leaves
too much potential for unchecked references and it just felt like it
would be more trouble than it's worth.
- Add a 'flags' value to the output definition structure. This lets
the output specify if it uses video/audio, and whether the output is
meant to be used with OBS encoders or not.
- Remove boilerplate code for outputs. This makes it easier to program
outputs. The boilerplate code involved before was mostly just
involving connecting to the audio/video data streams directly in each
output plugin.
Instead of doing that, simply add plugin callback functions for
receiving video/audio (either encoded or non-encoded, whichever it's
set to use), and then call obs_output_begin_data_capture and
obs_output_end_data_capture to automatically handle setting up
connections to raw or encoded video/audio streams for the plugin.
- Remove 'active' function from output callbacks, as it's no longer
really needed now that the libobs output context automatically knows
when the output is active or not.
- Make it so that an encoder cannot be destroyed until all data
connections to the encoder have been removed.
- Change the 'start' and 'stop' functions in the encoder interface to
just an 'initialize' callback, which initializes the encoder.
- Make it so that the encoder must be initialized first before the data
stream can be started. The reason why initialization was separated
from starting the encoder stream was because we need to be able to
check that the settings used with the encoder *can* be used first.
This problem was especially annoying if you had both video/audio
encoding. Before, you'd have to check the return value from
obs_encoder_start, and if that second encoder fails, then you
basically had to stop the first encoder again, making for
unnecessary boilerplate code whenever starting up two encoders.
- Add a properties window for sources so that you can now actually edit
the settings for sources. Also, display the source by itself in the
window (Note: not working on mac, and possibly not working on linux).
When changing the settings for a source, it will call
obs_source_update on that source when you have modified any values
automatically.
- Add a properties 'widget', eventually I want to turn this in to a
regular nice properties view like you'd see in the designer, but
right now it just uses a form layout in a QScrollArea with regular
controls to display the properties. It's clunky but works for the
time being.
- Make it so that swap chains and the main graphics subsystem will
automatically use at least one backbuffer if none was specified
- Fix bug where displays weren't added to the main display array
- Make it so that you can get the properties of a source via the actual
pointer of a source/encoder/output in addition to being able to look
up properties via identifier.
- When registering source types, check for required functions (wasn't
doing it before). getheight/getwidth should not be optional if it's
a video source as well.
- Add an RAII OBSObj wrapper to obs.hpp for non-reference-counted
libobs pointers
- Add an RAII OBSSignal wrapper to obs.hpp for libobs signals to
automatically disconnect them on destruction
- Move the "scale and center" calculation in window-basic-main.cpp to
its own function and in its own source file
- Add an 'update' callback to WASAPI audio sources
Microsoft's garbage compiler just doesn't even.. read the names of
enums. It sees an enum and goes "durr, that's an int" without even
properly evaluating it. Just total garbage, as per usual.
Also, rename atomic functions to be consistent with the rest of the
platform/threading functions, and move atomic functions to threading*
files rather than platform* files
- Implement OBS encoder interface. It was previously incomplete, but
now is reaching some level of completion, though probably should
still be considered preliminary.
I had originally implemented it so that encoders only have a 'reset'
function to reset their parameters, but I felt that having both a
'start' and 'stop' function would be useful.
Encoders are now assigned to a specific video/audio media output each
rather than implicitely assigned to the main obs video/audio
contexts. This allows separate encoder contexts that aren't
necessarily assigned to the main video/audio context (which is useful
for things such as recording specific sources). Will probably have
to do this for regular obs outputs as well.
When creating an encoder, you must now explicitely state whether that
encoder is an audio or video encoder.
Audio and video can optionally be automatically converted depending
on what the encoder specifies.
When something 'attaches' to an encoder, the first attachment starts
the encoder, and the encoder automatically attaches to the media
output context associated with it. Subsequent attachments won't have
the same effect, they will just start receiving the same encoder data
when the next keyframe plays (along with SEI if any). When detaching
from the encoder, the last detachment will fully stop the encoder and
detach the encoder from the media output context associated with the
encoder.
SEI must actually be exported separately; because new encoder
attachments may not always be at the beginning of the stream, the
first keyframe they get must have that SEI data in it. If the
encoder has SEI data, it needs only add one small function to simply
query that SEI data, and then that data will be handled automatically
by libobs for all subsequent encoder attachments.
- Implement x264 encoder plugin, move x264 files to separate plugin to
separate necessary dependencies.
- Change video/audio frame output structures to not use const
qualifiers to prevent issues with non-const function usage elsewhere.
This was an issue when writing the x264 encoder, as the x264 encoder
expects non-const frame data.
Change stagesurf_map to return a non-const data type to prevent this
as well.
- Change full range parameter of video scaler to be an enum rather than
boolean
...The reason why audio didn't work was because I overwrote the bitrate
values.
As for semaphores, mac doesn't support unnamed semaphores without using
mach semaphores. So, I just implemented a semaphore wrapper for each
OS.
Ensure that a source has a valid name. Duplicates aren't a big deal
internally, but sources without a name are probably something that
should be avoided. Made is so that if a source is programmatically
created without a name, it's assigned an index based name.
In the main basic-mode window, made it check to make sure the name was
valid as well.
- Add some temporary streaming code using FFmpeg. FFmpeg itself is not
very ideal for streaming; lack of direct control of the sockets and
no framedrop handling means that FFmpeg is definitely not something
you want to use without wrapper code. I'd prefer writing my own
network framework in this particular case just because you give away
so much control of the network interface. Wasted an entire day
trying to go through FFmpeg issues.
There's just no way FFmpeg should be used for real streaming (at
least without being patched or submitting some sort of patch, but I'm
sort of feeling "meh" on that idea)
I had to end up writing multiple threads just to handle both
connecting and writing, because av_interleaved_write_frame blocks
every call, stalling the main encoder thread, and thus also stalling
draw signals.
- Add some temporary user interface for streaming settings. This is
just temporary for the time being. It's in the outputs section of
the basic-mode settings
- Make it so that dynamic arrays do not free all their data when the
size just happens to be reduced to 0. This prevents constant
reallocation when an array keeps going from 1 item to 0 items. Also,
it was bad to become dependent upon that functionality. You must now
always explicitly call "free" on it to ensure the data is free, and
that's how it should be. Implicit functionality can lead to
confusion and maintainability issues.
- 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.
Originally, the rendering system was designed to only display sources
and such, but I realized there would be a flaw; if you wanted to render
the main viewport in a custom way, or maybe even the entire application
as a graphics-based front end, you wouldn't have been able to do that.
Displays have now been separated in to viewports and displays. A
viewport is used to store and draw sources, a display is used to handle
draw callbacks. You can even use displays without using viewports to
draw custom render displays containing graphics calls if you wish, but
usually they would be used in combination with source viewports at
least.
This requires a tiny bit more work to create simple source displays, but
in the end its worth it for the added flexibility and options it brings.
The API used to be designed in such a way to where it would expect
exports for each individual source/output/encoder/etc. You would export
functions for each and it would automatically load those functions based
on a specific naming scheme from the module.
The idea behind this was that I wanted to limit the usage of structures
in the API so only functions could be used. It was an interesting idea
in theory, but this idea turned out to be flawed in a number of ways:
1.) Requiring exports to create sources/outputs/encoders/etc meant that
you could not create them by any other means, which meant that
things like faruton's .net plugin would become difficult.
2.) Export function declarations could not be checked, therefore if you
created a function with the wrong parameters and parameter types,
the compiler wouldn't know how to check for that.
3.) Required overly complex load functions in libobs just to handle it.
It makes much more sense to just have a load function that you call
manually. Complexity is the bane of all good programs.
4.) It required that you have functions of specific names, which looked
and felt somewhat unsightly.
So, to fix these issues, I replaced it with a more commonly used API
scheme, seen commonly in places like kernels and typical C libraries
with abstraction. You simply create a structure that contains the
callback definitions, and you pass it to a function to register that
definition (such as obs_register_source), which you call in the
obs_module_load of the module.
It will also automatically check the structure size and ensure that it
only loads the required values if the structure happened to add new
values in an API change.
The "main" source file for each module must include obs-module.h, and
must use OBS_DECLARE_MODULE() within that source file.
Also, started writing some doxygen documentation in to the main library
headers. Will add more detailed documentation as I go.
The signature detection code when reading UTF-8 files was causing the
UTF-8 strings read from file to allocate more data than they were
supposed to, causing the last 3 bytes to be garbage
- Fill in the rest of the FFmpeg test output code for testing so it
actually properly outputs data.
- Improve the main video subsystem to be a bit more optimal and
automatically output I420 or NV12 if needed.
- Fix audio subsystem insertation and byte calculation. Now it will
seamlessly insert new audio data in to the audio stream based upon
its timestamp value. (Be extremely cautious when using floating
point calculations for important things like this, and always round
your values and check your values)
- Use 32 byte alignment in case of future optimizations and export a
function to get the current alignment.
- Make os_sleepto_ns return true if slept, false if the time has
already been passed before the call.
- Fix sinewave output so that it actually properly calculates a middle
C sinewave.
- Change the use of row_bytes to linesize (also makes it a bit more
consistent with FFmpeg's naming as well)
- Add planar audio support. FFmpeg and libav use planar audio for many
encoders, so it was somewhat necessary to add support in libobs
itself.
- Improve/adjust FFmpeg test output plugin. The exports were somewhat
messed up (making me rethink how exports should be done). Not yet
functional; it handles video properly, but it still does not handle
audio properly.
- Improve planar video code. The planar video code was not properly
accounting for row sizes for each plane. Specifying row sizes for
each plane has now been added. This will also make it more compatible
with FFmpeg/libav.
- Fixed a bug where callbacks wouldn't create properly in audio-io and
video-io code.
- Implement 'blogva' function to allow for va_list usage with libobs
logging.
- Implement texture scaling/conversion/downloading for the main view so
we can finally start getting data to output.
Also, redesign how it works a bit, it will now properly wait one full
frame for each step in the process: rendering the main texture,
scaling the main texture to an output texture, staging/downloading the
ouput texture, and then outputting that staged data. This way, the
GPU will have more than enough time to fully complete each step.
- Fix a bug with OpenGL plugin's texture staging function. Was using
glBindBuffer instead of what should have been used: glBindTexture.
- Change the naming scheme of the variables in default.effect. It's now
named with the idea of just "color matrix" in mind instead of "yuv
matrix", and instead of DrawRGBToYUV, it's now just DrawMatrix.
- Implemented better C++ classes for handling scenes/sources/items in
obs.hpp, allowing them to automatically increment and decrement the
references of each, as well as assign them to QVariants.
- Because QVariants are now using the C++ classes, remove the pointer
QVariant wrapper.
- Use the new C++ classes with the QVariant user data of list box items,
both for the sake of thread safety and to ensure that the data
referenced is not freed until removed. NOTE: still might need some
testing.
- Implemented a source-remove signal from libobs, and start using that
signal instead of the source-destroy signal for signalling item
removal.
- Add property list callbacks to sources/outputs/encoders so that if
necessary user interface can be automatically generated or perhaps a
property list widget can be used for them.
- Change some of the property API names. obs_property_list_t felt a bit
awkward when actually using it, so I just renamed it to
obs_properties_t.
- Removed the getdata/setdata nad getparam/setparam functions from
sources/services, they will be superseded by the dynamic procedure
call API.
Implement a properties definition interface to allow modules to export
general properties associated with objects of libobs.
The properties definition interface allows the option for automatic
settings UI generation (which will make simple plugins easier to develop
without the need for user interface), as well as allow real-time
property editing of values of things like sources/outputs/etc without
having to open property dialogs. More property types can be added in
the future as needed as well.
Reduce and simplify the UI export interface. Having to export functions
with designated names was a bit silly for this case, it makes more sense
for inputs/outputs/etc because they have more functions associated with
them, but in this case the callback can be retrieved simply through the
enumeration exports. Makes it a bit easier and a little less awkward
for this situation.
Also, changed the exports and names to be a bit more consistent,
labelling them both as either "modal" or "modeless", and changed the UI
function calls to obs_exec_ui and obs_create_ui to imply modal/modeless
functionality a bit more.
I realized that I had intended modeless UI to be usable by plugins, but
it had been pointed out to me that modeless really needs to return a
pointer/handle to the user interface object that was created.
The ui_enum function gets a const struct obs_ui_info **, which basically
means it expects static data to be used. I originally had it the other
way around, but yea, it's probably not a good idea, so I'm going to
revert back to the original code instead, which doesn't rely on the data
being static.
Made it so that enum_ui returns a const pointer to a structure rather
than require an actual structure.
Changed a few of the descriptions that I missed.
Add the ability to be able to call and use toolkit-specific or
program-specific user interface in modules.
User interface code can be either bundled with the module, or 'split'
out in to separate libraries (recommended).
There are three reasons why splitting is recommended:
1.) It allows plugins to be able to create custom user interface for
each toolkit if desired.
2.) Often, UI will be programmed in one language (the language of the
toolkit), and core logic may be programmed in another. This
allows plugins to keep the languages separated if necessary.
3.) It prevents direct linkage of UI toolkits libraries with core
module logic.
Splitting is not required, though is recommended if you want your plugin
to be more flexible with other user interface toolkits or programs.
Will implement a generic properties lookup next, which will be used for
automatic UI handling so that plugin UI isn't necessarily required.
Scene items previously were removed by calling obs_sceneitem_destroy,
but this proved to be a potential race condition where two different
threads could try to destroy the same scene item at the same time.
Instead of doing that, reference counting is now used on scene items,
and an explicit obs_sceneitem_remove function is used instead for item
removal, which sets a 'removed' variable to ensure it can only be called
exactly one time.
The previous commit used the scene as a parameter to check to see if
the scene item was still present within the scene before destroying, but
this was actually unnecessary because the fault was because the destroy
signal was being triggered *before* the scene's mutex locked, thus
causing a race condition. I changed the code so that it signals after
the lock instead of before, so the scene parameter should no longer be
necessary.
Fixes a deadlock when trying to remove a source from the GUI. The scene
item signal handlers would mark the source as removed which results in
the video thread also trying to run obs_sceneitem_destroy thereby
deadlocking the video thread (and the GUI thread)
- Add 'set_default' functions to obs-data.*. These functions ensure
that a paramter exists and that the parameter is of a specific type.
If not, it will create or overwrite the value with the default setting
instead.
These functions are meant to be explicitly called before using any of
the 'get' functions. The reason why it was designed this way is to
encourage defaults to be set in a single place/function.
For example, ideal usage is to create one function for your data,
"set_my_defaults(obs_data_t data)", set all the default values within
that function, and then call that function on create/update, that way
all defaults are centralized to a single place.
- Ensure that data passed to sources/encoders/outputs/etc is always
valid, and not a null value.
- While I'm remembering, fix a few defaults of the main program config
file data.
Add a fairly easy to use settings interface that can be passed to
plugins, and replaced the old character string system that was being
used before. The new data interface allows for an easier method of
getting/altering settings for plugins, and is built to be serializable
to/from JSON.
Also, removed another wxFormBuilder file that was no longer in use.
I'm doing this because I might create another data structure called
obs_data for a different purpose. That and obs_program_data feels a bit
less vague for what it does.
- Move over the last of the original settings dialog code to QT. It was
actually a bit easier to write in the QT version. wxWidgets was
definitely not ideal for that because the pages would fully
create/destroy every time.
- [Win32] Fix os_dlopen so that it only appends .dll if not present
- [MacOS] Fix name dialog text edit widget issue (it would be better if
we could just use the list widget for editing labels, will have to
look in to that in the future)
- Tweak the settings UI a bit more and make 30 FPS default
- Add a macro to convert a QString to a UTF-8 const char * string
- Rename build/plugins to build/obs-plugins
- Remove the last of the wxWidgets code
Fixed a few files that went over 80 columns, mostly just a nitpack on my
part.
libobs/obs-nix.c had a rather bad case of leading whitespace.
Also, fixed the x86 obs-studio project files so that it would properly
output to the right directory. It couldn't find libobs.lib because
obs-studio's project settings had it outputting to a different place
than the rest of the projects.
- I seem to have fixed ths issues with the main preview widget. It
seems you just need to set the right window attributes to stop it from
breaking. Though when opengl is enabled, there appears to be a weird
background glitch in the Qt stuff -- I'm not entirely sure what's
going on. Bug in Qt?
Also fixed the layout issues, and the widget now properly resizes and
centers in to its parent widget.
- Prevent the render loop from accessing data if the data isn't valid.
Because obs->data is freed before the graphics stuff, it can cause
the graphics to keep trying to query the obs->data.displays_mutex
after it had already been destroyed.
- Added some code for FFmpeg output that I'm still playing around with.
Right now I'm just trying to get it to output to file and try to
understand the FFmpeg/libav APIs. Hopefully in the future this plugin
can be used for any sort of output to FFmpeg.
- Fixed a cast warning in audio-io.c with size_t -> uint32_t
- Renamed the 'video_info' and 'audio_info' structures to
'video_conver_info' and 'audio_convert_info' to better represent their
actual purpose, and to avoid confusion with 'audio_output_info' and
'video_output_info' structures.
- Removed a few macros from obs-def.h that were at one point going to be
used but no longer going to be used (at least for now)
Just a minor fix mostly because I noticed that I kept accidentally
forgetting to add checks to the code properly. This is one of those
cases where macros come in useful, as macros can automate the process
and help prevent these mistakes from happening by accident.
Changed the comments to properly reflect the new callbacks, as I had
forgotten to update the comments for them both.
Also, changed "setbitrate" and "request_keyframe" return values to be
boolean.
- First, I redid the output interface for libobs. I feel like it's
going in a pretty good direction in terms of design.
Right now, the design is so that outputs and encoders are separate.
One or more outputs can connect to a specific encoder to receive its
data, or the output can connect directly to raw data from libobs
output itself, if the output doesn't want to use a designated encoder.
Data is received via callbacks set when you connect to the encoder or
raw output. Multiple outputs can receive the data from a single
encoder context if need be (such as for streaming to multiple channels
at once, and/or recording with the same data).
When an encoder is first connected to, it will connect to raw output,
and start encoding. Additional connections will receive that same
data being encoded as well after that. When the last encoder has
disconnected, it will stop encoding. If for some reason the encoder
needs to stop, it will use the callback with NULL to signal that
encoding has stopped. Some of these things may be subject to change
in the future, though it feels pretty good with this design so far.
Will have to see how well it works out in practice versus theory.
- Second, Started adding preliminary RTMP/x264 output plugin code.
To speed things up, I might just make a direct raw->FFmpeg output to
create a quick output plugin that we can start using for testing all
the subsystems.
Completely revamped the entire media i/o data and handlers. The
original idea was to have a system that would have connecting media
inputs and outputs, but at a certain point I realized that this was an
unnecessary complexity for what we wanted to do. (Also, it reminded me
of directshow filters, and I HATE directshow with a passion, and
wouldn't wish it upon my greatest enemy)
Now, audio/video outputs are connected to directly, with better callback
handlers, and will eventually have the ability to automatically handle
conversions such as 4:4:4 to 4:2:0 when connecting to an input that uses
them. Doing this will allow the video/audio i/o handlers to also
prevent duplicate conversion, as well as make it easier/simple to use.
My true goal for this is to make output and encoder plugins as simple to
create as possible. I want to be able to be able to create an output
plugin with almost no real hassle of having to worry about image
conversions, media inputs/outputs, etc. A plugin developer shouldn't
have to handle that sort of stuff when he/she doesn't really need to.
Plugins will be able to simply create a callback via obs_video() and/or
obs_audio(), and they will automatically receive the audio/video data in
the formats requested via a simple callback, without needing to do
almost anything else at all.
When the first async video frame is used it would not set audio timing,
moved that code into obs_source_getframe. Also, might consider renaming
obs_source_getframe. "Query frame" instead perhaps? Will sleep on it,
might not even bother.
- Add preliminary (yet to be tested) handling of timestamp invalidation
issues that can happen with specific devices, where timestamps can
reset or go backward/forward in time with no rhyme or reason. Spent
the entire day just trying to figure out the best way to handle this.
If both audio and video are present, it will increment a reference
counter if video timestamps invalidate, and decrement the reference
counter when the audio timestamps invalidate. When the reference
counter is not 0, it will not send audio as the audio will have
invalid timing. What this does is it ensures audio data will never go
out of bounds in relation to the video, and waits for both audio and
video timestamps to "jump" together before resuming audio.
- Moved async video frame timing adjustment code into
obs_source_getframe instead so it's automatically handled whenever
called.
- Removed the 'audio wait buffer' as it was an unnecessary complexity
that could have had problems in the future. Instead, audio will not
be added until video starts for sources that have both async
audio/video. Audio could have buffered for too long of a time anyway,
who knows what devices are going to do.
- Fixed a minor conversion warning in audio-io.c
- In the audio I/O code, if there's a pause in the program or its
threads (especially the audio thread), it'll cause it to sample too
much data, and increase line->base_timestamp to a potentially higher
value than the next audio timestamp that may be added to the line.
This would cause it to crash originally, because it expects audio
data that is within the designated buffering limit.
Because that audio data cannot be filled by that data anyway, just
ignore the audio data until it goes back to the right timing (which
it will as long as the code that is using the line accounts for its
current system time)
- Often, timestamps will go "back" in time with certain.. terrible
devices that no one should use. When this occurs, timing is now
reset so that the new audio comes in directly after the old audio
seamlessly.
- Audio data was just being popped to the "front" of the mix buffer, so
instead it now properly pops into the correct position in the mix
buffer (proper mixing still needs to be implemented)
- Added a test audio sinewave test source that should just play a sine
wave of the middle C note. Using unsigned 8 bit mono to test
ffmpeg's audio resampler, seems to work pretty good.
- Fixed a boolean trap in threading.h for the event_init function, it
now uses enum event_type, which can be EVENT_TYPE_MANUAL or
EVENT_TYPE_AUTO, to specify whether the event is automatically reset
or not.
- Changed display names of test sources to something a little less
vague.
- Removed te whole "if timestamp is 0 just use current system time"
when outputting source audio, if you want to use system time you
should just use system time yourself. Using 0 as some sort of
"indicator" like that just makes things confusing, and prevents you
from legitimately using 0 as a timestamp for your audio data.
- Circular buffer code wasn't correctly handling the splitting of
newly placed data segments, the code was untested and turned out to
just be backwards. It now copied the data to the back and front of
the buffer properly.
For one, I added a new member gs_window for future use.
The member is "display" which represents our connection to X11.
Ideally, we should use this specific connection to deal with our Window.
For now, it's disabled. Read comment for more information.
Secondly, wxGTK apparently doesn't map our window in some cases.
This causes the window ID passed to be bad and will stop (or segfault)
our program. This might be related to the first commit above.
For now, all this commit does is realize the window manually.
- Mixing still isn't implemented, but the audio system should be able
to start up, and mix at least once audio line for the time being.
Will have to write some test audio sources to verify things are
working properly, and build the rest of the output functionality.
- Using a recursive mutex fixes issues where objects need to enter the
main libobs sources mutex while already within the mutex in the same
thread. Otherwise it would keep getting locked on itself on
destruction.
- Apply the volume specified with the audio data packet before
inserting the audio data into the circular buffer. Added functions
for multiplying the volume with all the different audio bit depths.
(Could probably be greatly optmimized later)
BtbN
jp9000
Palana
Danni
fryshorts
Zachary Lund