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.
jp9000
Danni
Palana
BtbN
fryshorts