(This commit also modifies the decklink, linux-v4l2, mac-avcapture,
obs-ffmpeg, and win-dshow modules)
Originally, async buffering for sources was supposed to be a
user-controllable flag. However, that turned out to be less than ideal
because sources (such as the win-dshow plugin) were programmed with
automatic control over their buffering (such as automatically detecting
USB 2.0 capture devices and then enabling in those cases).
The fact that it was a flag caused a design flaw to where buffering
values would be overwritten when a source is loaded from save data.
Because of that, this flag is being deprecated and replaced with a
specific function to enable unbuffered mode instead.
Adds functions to turn on audio monitoring to allow the user to hear
playback of an audio source over the user's speaker. It can be set to
turn off monitoring and only output to stream, or it can be set to
output only to monitoring, or it can be set to both.
On windows, audio monitoring uses WASAPI. Windows also is capable of
syncing the audio to the video according to when the video frame itself
was played.
On mac, it uses AudioQueue.
On linux, it's not currently implemented and won't do anything (to be
implemented).
Allows getting the current active framerate that the core is rendering
with. This takes in to account any rendering lag or stalls that may be
occurring.
Allows the ability to use scale filters such as point, bicubic, lanczos
on specific scene items, disabled by default. When using one of the
latter two options, if the item's scale is under half of the source's
original size, it uses the bilinear low resolution downscale shader
instead.
(Note: this commit also modifies the obs-filters and test-input modules)
Changes the obs_source_process_filter_begin return type so that it
returns true/false to indicate that filter processing should or should
not continue (for example if the filter is bypassed or if there's some
other sort of issue that causes the filtering to fail)
This is a band-aid solution to be able to create temporary services
without logging them and keep them out of enumeration functions.
This is a band-aid solution -- 'master obs context lists' should not be
kept by the core. Logging of object creation/destruction should also be
controlled by the front-end instead of the core.
Determines whether an obs object was created successfully. If a plugin
that's used for a saved object is removed (third party plugins), its
data will become invalid, but the objects can often still be created for
the sake of preserving user settings, but sometimes these objects can
cause problems if they're actually used (such as using them for
transitions).
Adds deinterlacing API functions. Both standard and 2x variants are
supported. Deinterlacing is set via obs_source_set_deinterlace_mode and
obs_source_set_deinterlace_field_order.
This was implemented in to the core itself because deinterlacing should
happen before effect filters are processed, but after async filters are
processed. If this were added as a filter, there is the possibility
that a different filter is processed before deinterlacing, which could
mess with the result. It was also a bit easier to implement this way
due to the fact that that deinterlacing may need to have access to the
previous async frame.
Effects were split in to separate files to reduce load time (especially
for yadif shaders which take a significant amount of time to compile).
(Note: This commit also changes the UI)
Changed:
-------------------
void obs_load_sources(obs_data_array_t *sources_list);
To:
-------------------
void obs_load_sources(obs_data_array_t *sources_list,
obs_source_load_cb callback, void *private_data);
Signals should really never be required to use to make some function
work properly. The "source_load" signal was required for the
obs_load_sources function, but it's meant more for loading private data
in the settings, not for general loading of sources.
This changes it so that a callback is explicitly required to load the
sources.
(Note: This commit also modifies the UI)
Allows the ability to duplicate sources fully copied, and/or have the
scene and its duplicates be private sources
Mostly only used for transitions with the intention of automatically
creating transitions which don't require configuration, returns whether
the source has any properties or not (whether it's configurable)
(Note: This commit also modifies UI)
Instead of using signals, use designated callback lists for audio
capture and audio control helpers. Signals aren't suitable here due to
the fact that signals aren't meant for things that happen every frame or
things that happen every time audio/video is received. Also prevents
audio from being allocated every time these functions are called due to
the calldata structure.
Transition sources are implemented by registering a source type as
OBS_SOURCE_TYPE_TRANSITION. They're automatically marked as video
composite sources, and video_render/audio_render callbacks must be set
when registering the source. get_width and get_height callbacks are
unused for these types of sources, as transitions automatically handle
width/height behind the scenes with the transition settings.
In the video_render callback, the helper function
obs_transition_video_render is used to assist in automatically
processing and rendering the audio. A render callback is passed to the
function, which in turn passes to/from textures that are automatically
rendered in the back-end.
Similarly, in the audio_render callback, the helper function
obs_transition_audio_render is used to assist in automatically
processing and rendering the audio. Two mix callbacks are used to
handle how the source/destination sources are mixed together. To ensure
the best possible quality, audio processing is per-sample.
Transitions can be set to automatically resize, or they can be set to
have a fixed size. Sources within transitions can be made to scale to
the transition size (with or without aspect ratio), or to not scale
unless they're bigger than the transition. They can have a specific
alignment within the transition, or they just default to top-left.
These features are implemented for the purpose of extending transitions
to also act as "switch" sources later, where you can switch to/from two
different sources using the transition animation.
Planned (but not yet implemented and lower priority) features:
- "Switch" transitions which allow the ability to switch back and forth
between two sources with a transitioning animation without discarding
the references
- Easing options to allow the option to transition with a bezier or
custom curve
- Manual transitioning to allow the front-end/user to manually control
the transition offset
(Note: test and UI are also modified by this commit)
API Changed (removed "enum obs_source_type type" parameter):
-------------------------
obs_source_get_display_name
obs_source_create
obs_get_source_output_flags
obs_get_source_defaults
obs_get_source_properties
Removes the "type" parameter from these functions. The "type" parameter
really doesn't serve much of a purpose being a parameter in any of these
cases, the type is just to indicate what it's used for.
The new audio subsystem fixes two issues:
- First Primary issue it fixes is the ability for parent sources to
intercept the audio of child sources, and do custom processing on
them. The main reason for this was the ability to do custom
cross-fading in transitions, but it's also useful for things such as
side-chain effects, applying audio effects to entire scenes, applying
scene-specific audio filters on sub-sources, and other such
possibilities.
- The secondary issue that needed fixing was audio buffering.
Previously, audio buffering was always a fixed buffer size, so it
would always have exactly a certain number of milliseconds of audio
buffering (and thus output delay). Instead, it now dynamically
increases audio buffering only as necessary, minimizing output delay,
and removing the need for users to have to worry about an audio
buffering setting.
The new design makes it so that audio from the leaves of the scene graph
flow to the root nodes, and can be intercepted by parent sources. Each
audio source handles its own buffering, and each audio tick a specific
number of audio frames are popped from the front of the circular buffer
on each audio source. Composite sources (such as scenes) can access the
audio for child sources and do custom processing or mixing on that
audio. Composite sources use the audio_render callback of sources to do
synchronous or deferred audio processing per audio tick. Things like
scenes now mix audio from their sub-sources.
(Note: This commit breaks libobs compilation. Skip if bisecting)
Removes audio lines and stores the circular buffer for the audio on the
source itself.
Before if a source was set to invisible it would still be considered
active. This changes it so that the source is deactivated when the
source is invisible to reduce needless resource usage or capturing.
Renames:
----------------------------------------
obs_source_add_child
obs_source_remove_child
obs_source_enum_sources
obs_source_enum_tree
obs_source_info::enum_sources
To:
----------------------------------------
obs_source_add_active_child
obs_source_remove_active_child
obs_source_enum_active_sources
obs_source_enum_active_tree
obs_source_info::enum_active_sources
These functions/callbacks had misleading names: they originally implied
any child sources, when they actually meant active child sources that
are being used to render video or audio. It's important that the
function names represent their actual purpose.
(Note: This commit breaks UI compilation. Skip if bisecting)
Adds a means of saving specific sources that the front-end chooses,
rather than being forced to use the now-removed "user list".
(Note: This commit breaks UI compilation. Skip if bisecting)
API Removed:
------------------------
obs_add_source
API Changed:
------------------------
obs_source_remove: Now just marks/signals a source for removal
The concept of "user sources" is flawed: it was something that the
front-end was forced to deal with if it wanted to automate source
saving/loading, and often it had to code around it. That's not how
saving/loading should work, a front-end should be allowed to manage
lists of sources in the way it explicitly chooses, and it should be able
to choose which sources it wants to save/load.
This function was removed even though the browser plugin was using this
function on mac, so this is being put back in temporarily while the
browser plugin is modified to remove this function.
API removed:
--------------------
gs_effect_t *obs_get_default_effect(void);
gs_effect_t *obs_get_default_rect_effect(void);
gs_effect_t *obs_get_opaque_effect(void);
gs_effect_t *obs_get_solid_effect(void);
gs_effect_t *obs_get_bicubic_effect(void);
gs_effect_t *obs_get_lanczos_effect(void);
gs_effect_t *obs_get_bilinear_lowres_effect(void);
API added:
--------------------
gs_effect_t *obs_get_base_effect(enum obs_base_effect effect);
Summary:
--------------------
Combines multiple near-identical functions into a single function with
an enum parameter.
This feature allows a user to delay an output (as long as the output
itself supports it). Needless to say this intended for live streams,
where users may want to delay their streams to prevent stream sniping,
cheating, and other such things.
The design this time was a bit more elaborate, but still simple in
design: the user can now schedule stops/starts without having to wait
for the stream itself to stop before being able to take any action.
Optionally, they can also forcibly stop stream (and delay) in case
something happens which they might not want to be streamed.
Additionally, a new option was added to preserve stream cutoff point on
disconnections/reconnections, so that if you get disconnected while
streaming, when it reconnects, it will reconnect right at the point
where it left off. This will probably be quite useful for a number of
applications in addition to regular delay, such as setting the delay to
1 second and then using this feature to minimize, for example, a
critical stream such as a tournament stream from getting any of its
stream data cut off. However, using this feature will of course cause
the stream data to buffer and increase delay (and memory usage) while
it's in the process of reconnecting.
API Changed:
---------------------------
From:
- bool obs_startup(const char *locale, profiler_name_store_t *store);
To:
- bool obs_startup(const char *locale, const char *module_config_path,
profiler_name_store_t *store);
Summary:
---------------------------
This allows plugin modules to store plugin-specific configuration data
(rather than only allowing objects to store configuration data). This
will be useful for things like caching data, for example looking up and
storing ingests from remote (rather than storing locally), or caching
font data (so it doesn't have to build a font cache each time), among
other things.
Also adds a module-specific directory for the UI
Due to all the threads in libobs it wouldn't be safe to make that
parameter reconfigurable after libobs is initialized without adding
even more synchronization. On the other hand, adding a function to set
the name store before calling obs_startup would solve the problem of
passing a name store into libobs, but it can lead to more complicated
semantics for obs_get_profiler_name_store (e.g., should it always return
the current name store even if libobs isn't initialized until someone
calls set_name_store(NULL)? should obs_shutdown call
set_name_store(NULL)? Passing it as obs_startup parameter avoids
these (and hopefully other) potential misunderstandings
(Non-compiling commit: windowless-context branch)
API Changed:
---------------------
Removed functions:
- obs_add_draw_callback
- obs_remove_draw_callback
- obs_resize
- obs_preview_set_enabled
- obs_preview_enabled
Removed member variables from struct obs_video_info:
- window_width
- window_height
- window
Summary:
---------------------
Changes the core libobs API to not be dependent upon a main window/view.
If you wish to draw to a window/view, use an obs_display object to
handle it.
This allows the use of libobs without requiring a window to be present
on the system. This is also prunes code that had to be needlessly
duplicated to handle the "main" window.
Intentionally breaks compilation when trying to compile the specific
merged commits within the windowless-context branch. This is meant to
be used in conjunction with a merge commit so that bisecting will never
see any non-compiling commits.
In case the encoder has to use a different sample rate (due to the
sample rate being unsupported), we need an API function for the encoder
to get the sample rate that the encoder is actually running at.
Allows the ability to hint at encoders what format should be used.
This is particularly useful if libobs is currently operating in planar
4:4:4, but you want to force an encoder used for streaming to convert to
NV12 to prevent streaming issues.
If you don't need to see what's displayed, then this is particularly
useful for two reasons:
1. It reduces the number of draw/present calls
2. It can prevent issues with certain hardware setups where rendering on
a monitor hooked up to a separate card can experience slowdowns
obs_source_process_filter tried to do everything in a single function,
but the problem is that effect parameters would not properly be
accounted for due to the way it internally draws, therefore it was
necessary to split the functions in to two, you first call
obs_source_process_filter_begin, then you set your effect parameters,
then you finally call obs_source_process_filter_end. This ensures that
when the filter is drawn, that the effect parameters are set.
These functions are primarily for use with filters, filters need to be
able to get the width/height of a target source without it necessarily
getting the post-filtered dimensions.
When a frame is processed by a filter, it comes directly from the
source's video frame cache. However, if a filter is using or processing
those frames for whatever reason, there would be no guarantee that the
frames would persist during processing, and frames could eventually be
deallocated unexpected, for example when the resolution or format
changes.
So the solution is to implement simple reference counting for the frames
so that the frames will exist until they have been released by any
source or filter that's using them.
Core API functions changed:
-----------------------------
EXPORT bool obs_reset_audio(struct audio_output_info *aoi);
EXPORT bool obs_get_audio_info(struct audio_output_info *aoi);
To:
-----------------------------
EXPORT bool obs_reset_audio(const struct obs_audio_info *oai);
EXPORT bool obs_get_audio_info(struct obs_audio_info *oai);
Core structure added:
-----------------------------
struct obs_audio_info {
uint32_t samples_per_sec;
enum speaker_layout speakers;
uint64_t buffer_ms;
};
Non-interleaved (planar) floating point output is standard with audio
filtering, so to prevent audio filters from having to worry about
different audio format implementations and for the sake consistency
between user interfaces, make it so that audio is always set to
non-interleaved floating point output.
API changed from:
------------------------
EXPORT void obs_service_apply_encoder_settings(obs_service_t *service,
obs_encoder_t *video_encoder,
obs_encoder_t *audio_encoder);
void obs_service_info::apply_encoder_settings(void *data
obs_encoder_t *video_encoder,
obs_encoder_t *audio_encoder);
To:
------------------------
EXPORT void obs_service_apply_encoder_settings(obs_service_t *service,
obs_data_t *video_encoder_settings,
obs_data_t *audio_encoder_settings);
void obs_service_info::apply_encoder_settings(void *data
obs_data_t *video_encoder_settings,
obs_data_t *audio_encoder_settings);
These changes make it so that instead of an encoder potentially being
updated more than once with different settings, that these functions
will be called for the specific settings being used, and the settings
will be updated according to what's required by the service.
This fixes that design flaw and ensures that there's no case where
obs_encoder_update is called where the settings might not have
service-specific settings applied.
The obs_source_get_width and obs_source_get_height functions need to
account for filters that may change their width/height such as a crop
filter or something similar.
As a side effect of this commit, because these functions need to lock
the filter mutex, these functions can no longer be used with a
const-qualified obs_source_t pointer.
obs_source_inc_showing and obs_source_dec_showing are used to indicate
that a source is showing or no longer being shown in a particular area
of the program outside from the main render view.
One could use an obs_view, but I felt like it was unnecessary because
using an obs_view just to display a single source feels somewhat
superfluous.
Instead of having services automatically apply encoder settings on
initialization (whether the output wants to or not), instead make it
something that must be explicitly called by the developer. There are
cases where the developer may not wish to apply the service-specific
settings, or may wish to override them for whatever reason.
Before: After:
obs_service_gettype obs_service_get_type
It seems there was an API function that was missed when we were doing
our big API consistency update. Unsquishes obs_service_gettype to
obs_service_get_type.
API changed:
--------------------------
void obs_output_set_audio_encoder(
obs_output_t *output,
obs_encoder_t *encoder);
obs_encoder_t *obs_output_get_audio_encoder(
const obs_output_t *output);
obs_encoder_t *obs_audio_encoder_create(
const char *id,
const char *name,
obs_data_t *settings);
Changed to:
--------------------------
/* 'idx' specifies the track index of the output */
void obs_output_set_audio_encoder(
obs_output_t *output,
obs_encoder_t *encoder,
size_t idx);
/* 'idx' specifies the track index of the output */
obs_encoder_t *obs_output_get_audio_encoder(
const obs_output_t *output,
size_t idx);
/* 'mixer_idx' specifies the mixer index to capture audio from */
obs_encoder_t *obs_audio_encoder_create(
const char *id,
const char *name,
obs_data_t *settings,
size_t mixer_idx);
Overview
--------------------------
This feature allows multiple audio mixers to be used at a time. This
capability was able to be added with surprisingly very little extra
overhead. Audio will not be mixed unless it's assigned to a specific
mixer, and mixers will not mix unless they have an active mix
connection.
Mostly this will be useful for being able to separate out specific audio
for recording versus streaming, but will also be useful for certain
streaming services that support multiple audio streams via RTMP.
I didn't want to use a variable amount of mixers due to the desire to
reduce heap allocations, so currently I set the limit to 4 simultaneous
mixers; this number can be increased later if needed, but honestly I
feel like it's just the right number to use.
Sources:
Sources can now specify which audio mixers their audio is mixed to; this
can be a single mixer or multiple mixers at a time. The
obs_source_set_audio_mixers function sets the audio mixer which an audio
source applies to. For example, 0xF would mean that the source applies
to all four mixers.
Audio Encoders:
Audio encoders now must specify which specific audio mixer they use when
they encode audio data.
Outputs:
Outputs that use encoders can now support multiple audio tracks at once
if they have the OBS_OUTPUT_MULTI_TRACK capability flag set. This is
mostly only useful for certain types of RTMP transmissions, though may
be useful for file formats that support multiple audio tracks as well
later on.
The comment says "these are different", but doesn't state why.
Actually, I should really rename the output flags so they're not flags,
but instead just "caps", because that's really all that they are.
If an async video source stops video for whatever reason, it would get
stuck on the last frame that was played. This was particularly awkward
when I wanted to give the user the ability to deactivate a source such
as a webcam because it would get stuck on the last frame.
This changes the way source volume handles transitioning between being
active and inactive states.
The previous way that transitioning handled volume was that it set the
presentation volume of the source and all of its sub-sources to 0.0 if
the source was inactive, and 1.0 if active. Transition sources would
then also set the presentation volume for sub-sources to whatever their
transitioning volume was. However, the problem with this is that the
design didn't take in to account if the source or its sub-sources were
active anywhere else, so because of that it would break if that ever
happened, and I didn't realize that when I was designing it.
So instead, this completely overhauls the design of handling
transitioning volume. Each frame, it'll go through all sources and
check whether they're active or inactive and set the base volume
accordingly. If transitions are currently active, it will actually walk
the active source tree and check whether the source is in a
transitioning state somewhere.
- If the source is a sub-source of a transition, and it's not active
outside of the transition, then the transition will control the
volume of the source.
- If the source is a sub-source of a transition, but it's also active
outside of the transition, it'll defer to whichever is louder.
This also adds a new callback to the obs_source_info structure for
transition sources, get_transition_volume, which is called to get the
transitioning volume of a sub-source.
Changed the design from using obs_source::enum_refs to just simply
preventing infinite source recursion in general, rather than allowing it
through the enum_refs variable. obs_source_add_child has been changed
so that it now returns a boolean, and if the function fails, it means
that the child cannot be added due to that potential recursion.
obs_encoder_getdisplayname declaration was not changed to match the
definition (obs_encoder_get_display_name) when the API consistency
update occurred.
This adds bicubic and lanczos scaling capability to libobs to improve
scaling quality and sharpness when the output resolution has to be
scaled relative to the base resolution. Bilinear is also available,
although bilinear has rather poor quality and causes scaling to appear
blurry.
If the output resolution is close to the base resolution, then bilinear
is used instead as an optimization, as there's no need to use these
shaders if scaling is not in use.
The Bicubic and Lanczos effects are also exposed via exported function
to allow the ability to use those shaders in plugin modules if desired.
The API change adds a variable 'scale_type' to the obs_video_info
structure that allows the user interface to choose what type of scaling
filter should be used.
This was an important change because we were originally using an
hard-coded 709/partial range color matrix for the output, which was
causing problems for people wanting to use different formats or color
spaces. This will now automatically generate the color matrix depending
on the format, color space, and range, or use an identity matrix if the
video format is RGB instead of YUV.
This adds a volume meter object to libobs that can be used by the GUI
or plugins to convert the raw audio level data from sources to values
that can easily be used to display the audio data.
The volume meter object will use the same mapping functions as the
fader object to map dB levels to a scale.
This adds a new library of audio control functions mainly for the use in
GUIS. For now it includes an implementation of a software fader that can
be attached to sources in order to easily control the volume.
The fader can translate between fader-position, volume in dB and
multiplier with a configurable mapping function.
Currently only a cubic mapping (mul = fader_pos ^ 3) is included, but
different mappings can easily be added.
Due to libobs saving/restoring the source volume from the multiplier,
the volume levels for existing source will stay the same, and live
changing of the mapping will work without changing the source volume.
If you look at the previous commits, you'll see I had added
obs_source_draw before. For custom drawn sources in particular, each
time obs_source_draw was called, it would restart the effect and its
passes for each draw call, which was not optimal. It should really use
the effect functions for that. I'll have to add a function to simplify
effect usage.
I also realized that including the color matrix parameters in
obs_source_draw made the function kind of messy to use; instead,
separating the color matrix stuff out to
obs_source_draw_set_color_matrix feels a lot more clean.
On top of that, having the ability to set the position would be nice to
have as well, rather than having to mess with the matrix stuff each
time, so I also added that for the sake of convenience.
obs_source_draw will draw a texture sprite, optionally of a specific
size and/or at a specific position, as well as optionally inverted. The
texture used will be set to the 'image' parameter of whatever effect is
currently active.
obs_source_draw_set_color_matrix will set the color matrix value if the
drawing requires color matrices. It will set the 'color_matrix',
'color_range_min', and 'color_range_max' parameters of whatever effect
is currently active.
Overall, these feel much more clean to use than the previous iteration.
This function simplifies drawing textures for sources in order to help
reduce boilerplate code. If a source is a custom drawn source, it will
automatically set up the effect to draw the sprite. If it's not a
custom drawn source, it will simply draw the sprite as per normal. If
the source uses a specific color matrix, it will also handle that as
well.
This moves the 'flags' variable from the obs_source_frame structure to
the obs_source structure, and allows user flags to be set for a specific
source. Having it set on the obs_source_frame structure didn't make
much sense.
OBS_SOURCE_UNBUFFERED makes it so that the source does not buffer its
async video output in order to try to play it on time. In other words,
frames are played as soon as possible after being received.
Useful when you want a source to play back as quickly as possible
(webcams, certain types of capture devices)
This reverts commit c3f4b0f018.
The obs_source_frame should not need to take flags to do this. This
shouldn't be a setting associated with the frame, but rather a setting
associated with the source itself. This was the wrong approach to
solving this particular problem.
Add 'flags' member variable to obs_source_frame structure.
The OBS_VIDEO_UNBUFFERED flags causes the video to play back as soon as
it's received (in the next frame playback), causing it to disregard the
timestamp value for the sake of video playback (however, note that the
video timestamp is still used for audio synchronization if audio is
present on the source as well).
This is partly a convenience feature, and partly a necessity for certain
plugins (such as the linux v4l plugin) where timestamp information for
the video frames can sometimes be unreliable.
obs_source_update_properties should be called by sources when property
values change, e.g. a capture device source would use this when it
detects a new capture device (in case its properties contain a list of
available capture devices or similar)
This Fixes a minor flaw with the API where data had to always be mutable
to be usable by the API.
Functions that do not modify the fundamental underlying data of a
structure should be marked as constant, both for safety and to signify
that the parameter is input only and will not be modified by the
function using it.
Typedef pointers are unsafe. If you do:
typedef struct bla *bla_t;
then you cannot use it as a constant, such as: const bla_t, because
that constant will be to the pointer itself rather than to the
underlying data. I admit this was a fundamental mistake that must
be corrected.
All typedefs that were pointer types will now have their pointers
removed from the type itself, and the pointers will be used when they
are actually used as variables/parameters/returns instead.
This does not break ABI though, which is pretty nice.
API functions added:
-----------------------------------------------
obs_output_set_preferred_size
obs_output_get_width
obs_output_get_height
obs_encoder_set_scaled_size
obs_encoder_get_width
obs_encoder_get_height
These functions allow for easier means of setting a custom resolution on
an output or encoder.
If an output uses an encoder and you set the preferred width/height
using the output, then the output will attempt to set the scaled
width/height for the encoder it's currently using.
Outputs and encoders now should use these functions to determine the
width/height of the raw frame data instead of using the video-io
functions.
Instead of having functions like obs_signal_handler() that can fail to
properly specify their actual intent in the name (does it signal a
handler, or does it return a signal handler?), always prefix functions
that are meant to get information with 'get' to make its functionality
more explicit.
Previous names: New names:
-----------------------------------------------------------
obs_audio obs_get_audio
obs_video obs_get_video
obs_signalhandler obs_get_signal_handler
obs_prochandler obs_get_proc_handler
obs_source_signalhandler obs_source_get_signal_handler
obs_source_prochandler obs_source_get_proc_handler
obs_output_signalhandler obs_output_get_signal_handler
obs_output_prochandler obs_output_get_proc_handler
obs_service_signalhandler obs_service_get_signal_handler
obs_service_prochandler obs_service_get_proc_handler
API Removed:
- graphics_t obs_graphics();
Replaced With:
- void obs_enter_graphics();
- void obs_leave_graphics();
Description:
obs_graphics() was somewhat of a pointless function. The only time
that it was ever necessary was to pass it as a parameter to
gs_entercontext() followed by a subsequent gs_leavecontext() call after
that. So, I felt that it made a bit more sense just to implement
obs_enter_graphics() and obs_leave_graphics() functions to do the exact
same thing without having to repeat that code. There's really no need
to ever "hold" the graphics pointer, though I suppose that could change
in the future so having a similar function come back isn't out of the
question.
Still, this at least reduces the amount of unnecessary repeated code for
the time being.
Changed:
- obs_source_gettype
To:
- enum obs_source_type obs_source_get_type(obs_source_t source);
- const char *obs_source_get_id(obs_source_t source);
This function was inconsistent for a number of reasons. First, it
returns both the ID and the type of source (input/transition/filter),
which is inconsistent with the name of "get type". Secondly, the
'squishy' naming convention which has just turned out to be bad
practice and causes inconsistencies. So it's now replaced with two
functions that just return the type and the ID.
Prefix with obs_ for the sake of consistency
Renamed enums:
- order_movement (now obs_order_movement)
Affected functions:
- obs_source_filter_setorder
- obs_sceneitem_setorder
Renamed functions:
- obs_source_getframe (rename to obs_source_get_frame)
- obs_source_releaseframe (rename to obs_source_release_frame)
For the sake of consistency and helping to get rid of the "squishy
function name" issue
Changed API:
- char *obs_find_plugin_file(const char *sub_path);
Changed to: char *obs_module_file(const char *file);
Cahnge it so you no longer need to specify a sub-path such as:
obs_find_plugin_file("module_name/file.ext")
Instead, now automatically handle the module data path so all you need
to do is:
obs_module_file("file.ext")
- int obs_load_module(const char *name);
Changed to: int obs_open_module(obs_module_t *module,
const char *path,
const char *data_path);
bool obs_init_module(obs_module_t module);
Change the module loading API so that if the front-end chooses, it can
load modules directly from a specified path, and associate a data
directory with it on the spot.
The module will not be initialized immediately; obs_init_module must
be called on the module pointer in order to fully initialize the
module. This is done so a module can be disabled by the front-end if
the it so chooses.
New API:
- void obs_add_module_path(const char *bin, const char *data);
These functions allow you to specify new module search paths to add,
and allow you to search through them, or optionally just load all
modules from them. If the string %module% is included, it will
replace it with the module's name when that string is used as a
lookup. Data paths are now directly added to the module's internal
storage structure, and when obs_find_module_file is used, it will look
up the pointer to the obs_module structure and get its data directory
that way.
Example:
obs_add_module_path("/opt/obs/my-modules/%module%/bin",
"/opt/obs/my-modules/%module%/data");
This would cause it to additionally look for the binary of a
hypthetical module named "foo" at /opt/obs/my-modules/foo/bin/foo.so
(or libfoo.so), and then look for the data in
/opt/obs/my-modules/foo/data.
This gives the front-end more flexibility for handling third-party
plugin modules, or handling all plugin modules in a custom way.
- void obs_find_modules(obs_find_module_callback_t callback, void
*param);
This searches the existing paths for modules and calls the callback
function when any are found. Useful for plugin management and custom
handling of the paths by the front-end if desired.
- void obs_load_all_modules(void);
Search through the paths and both loads and initializes all modules
automatically without custom handling.
- void obs_enum_modules(obs_enum_module_callback_t callback,
void *param);
Enumerates currently opened modules.
The version macro that modules use to compile versus the actual core
version that may be in use may be different, so this is a way to compare
them to check for compatibility issues later on.
Changed API functions:
libobs: obs_reset_video
Before, video initialization returned a boolean, but "failed" is too
little information, if it fails due to lack of device capabilities or
bad video device parameters, the front-end needs to know that.
The OBS Basic UI has also been updated to reflect this API change.
This function is used to simplify the process when using the default
locale handling for modules. It will automatically search in the plugin
data directory associated with the specific module specified, load the
default locale text (for example english if its default language is
english), and then it will load the set locale on top of the default
locale, which will cause text to use the default locale if the desired
locale text is not found.
Total bytes, total frames, and frames dropped. Total frames is
generated automatically, but total bytes and total dropped frames are
returned via callbacks.
The core itself now provides reconnection options (enabled by default, 2
second timeout between reconnects, 20 retries max until actual
disconnection occurs). This will make things easier for both module
developers and UI developers.
Reconnecting treats the stream as though it were still active, and
signals are sent when reconnecting and upon successful reconnection.
Need to implement user interface information for reconnections.
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.
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.
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.
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.
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.
- 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.
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.
- 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.
- 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.
- 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
- 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
- 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
- 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.
- 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.
- 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.
- 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.
- 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.
- 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.
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.
