Adds a cmake variable (DEBUG_FFMPEG_MUX) which enables FFmpeg debug
output in the ffmpeg-mux subprocess, and if on Windows, shows the
console window of the ffmpeg-mux subprocess so the current output can be
seen.
Add util/sse2neon.h to libobs_util_HEADERS in CMakelists.txt. The
previous patch #3180 for arm/rpi support was missing this in
libobs/CMakeLists.txt, and as a result plugins such as obs-websocket,
obs-ndi fail to compile on arm/rpi as the header is not copied with the
install.
As of 3.17 using find_package_handle_standard_args checks that the name
of the FindXXX file and the first argument are the same case.
Some modules used non-standard variables or the old singular variables
instead of plurals. This normalizes variable usage to the new-style.
Some CMakeLists.txt did custom error checking instead of propagating
find_package errors. These were changes to call find_package with
REQUIRED or without QUIET where needed and shortens the custom status
messages. This helps users who want to enable that functionality see
what precisely wasnt found.
-DNO_WARN_X86_INTRINSICS -mvsx flags are necessary on PPC64(LE), but
those flags haven't been added yet to libobs.pc, making some plugins
fail to build.
As os_gettime_ns() gets large the current scaling methods, mostly by casting
to uint64_t, may lead to numerical overflows. Sweep the code and use
util_mul_div64() where applicable.
Signed-off-by: Hans Petter Selasky <hps@selasky.org>
As of 3.17 using find_package_handle_standard_args checks that the name
of the FindXXX file and the first argument are the same case.
Some modules used non-standard variables or the old singular variables
instead of plurals. This normalizes variable usage to the new-style.
Some CMakeLists.txt did custom error checking instead of propagating
find_package errors. These were changes to call find_package with
REQUIRED or without QUIET where needed and shortens the custom status
messages. This helps users who want to enable that functionality see
what precisely wasnt found.
Instead of erroring out completely when it can't determine if the
certificate is valid, proceed anyway. This matches how web browsers
treat failed cert revocation checks. schannel just has somewhat
paranoid defaults.
SIMDE was introduced for aarch64 support, however, the library itself
supports non-SIMD fallback, which allows us provide support to other
platforms without code changes.
There is another world beyond x86. So we can simply enable SIMDE for
processors without SSE2 support.
Signed-off-by: Jiaxun Yang <jiaxun.yang@flygoat.com>
In 'libobs/CMakeLists.txt', use '${CMAKE_INSTALL_LIBDIR}' instead of
'${CMAKE_INSTALL_PREFIX}/lib', as the latter results into 'libobs.pc'
being installed under '/lib' when '/lib64' would be more appropriate.
In 'libobs/libobs.pc.in', use '@CMAKE_INSTALL_FULL_LIBDIR@' for
'libdir', '@CMAKE_INSTALL_FULL_INCLUDEDIR@' for 'includedir',
and '@CMAKE_INSTALL_PREFIX@' for 'prefix'.
Gentoo-Bug: https://bugs.gentoo.org/644538
Allows the ability to encode by passing NV12 textures. This uses a
separate thread for texture-based encoders with a small queue of
textures. An output texture with a keyed mutex shared texture is locked
between OBS and each encoder. A new encoder callback and capability
flag is used to encode with textures.
The cmake script code used to find and depend on pulseaudio for audio
monitoring on linux was unintentionally being triggered on mac. This
patch makes it so that the pulseaudio cmake code is ignored on mac.
Certain PSAPI functions moved to KERNEL32 after Vista, setting
PSAPI_VERSION 1 uses the old PSAPI exports for compatibility. Vista
support was unintentionally broken by a6318ff.
With certain combinations of CMake (<3.9) and ImageMagick (>=7),
configure will silently fail setting up support for ImageMagick,
even if the user requested for it, and the build will move on
(as long as FFMPEG_AVCODEC is found). This commit adds a check
that will stop configure in such cases.
Closesjp9000/obs-studio#1018
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).
A linker error for XGetXCBConnection started occurring after the
frontend API was added, the fix is simply to make sure libobs is
properly linking to X11_XCB.
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).
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
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.
