When a source window was not available, a red background was shown
instead. This was undesirable, and expected behavior would be for the
background to be transparent, enabling what exists behind the source
to be shown.
These functions were mistakenly not marked as static. They are not used
outside of their compiled object module files, therefore there's no
reason for them not to be static.
Apple wants to get people to move over to their own crypto APIs instead
of using OpenSSL, so disable the warning in the files where OpenSSL is
used for the time being.
If the media source is set to restart on activation, it also shuts down
when not active. However, it would *always* start regardless of
active/inactive when the source is first created. It shouldn't do that,
it should start up only when it becomes active.
Certain types of sources (display captures, game captures, audio
device captures, video device captures) should not be duplicated. This
capability flag hints that the source prefers references over full
duplication.
Darkest dungeon uses an unusual technique for drawing its frames: a
fixed 1920x1080 frame buffer used in place of the backbuffer, which is
then stretched to fit the size of the screen (whether the screen is
bigger or smaller than the actual texture).
The custom frame would cause glReadBuffer to initially fail with an
error. When this happens, their custom frame buffer is in use, so all
that needs to be done is simply reset the capture and force the current
output size to 1920x1080 while that custom frame is in use.
They presumably did this in order to ensure the game looks the same at
any resolution. Instead of having to use power-of-two sprites and
mipmaps for every single game sprite and stretch/skew each of them
(which would risk the final output "not looking quite right" at
different resolutions), they simply use non-pow-2 sprites with no
mipmaps and render them all on to one texture of a fixed size and then
stretch that final output texture. That ensures that the actual
composite of the game still looks the same at any resolution, while
reducing texture memory by not requiring each sprite to use a
power-of-two texture and mipmaps.
Adds the option of making the media file restart when the source becomes
active (such as switching to a scene with it).
Due to lack of libff features to start/stop/pause/seek media files,
currently this just destroys the demuxer and recreates it. Ideally,
libff should have some functions to allow a more optimal means of doing
those things.
Reactors a bit of code related to starting up FFmpeg and makes it so the
initial view for the media source's properties displays the most
commonly desired settings.
Instead of the media source properties showing the URL mode by default
along with a whole bunch of properties that are confusing to most users,
starts on file mode and changes defaults to be a bit more sensible
related to file input.
Also, as a temporary measure for fixing color format issues (some video
files would display their color information incorrectly), forced format
conversion is now enabled by default, and has been moved to advanced
settings. Ideally, the actual bug causing color format issues in either
media-io or libff should be fixed at some point.
When browsing for a file, it would also just use *.* for the file
filter, which is a pain to use. This has been changed to use a
reasonable file filter related to common video/audio files so you don't
have to wade through non-media files just to select a media file. A
filter to show all files is still available as well.
Prevents it from loading the entire image in the graphics thread, and
allows for animated gifs in the filter (not that anyone would ever do
that. ..right?)
Fixes what is arguably the most annoying feature of the mask/blend
filter, the fact that the image always stretches to the entire source.
It now centers and preserves aspect ratio by default, with an option to
make it stretch and discard aspect ratio to make it operate as it did
before.
Images continually loading/unloading every time transitioning occurs
adds a lot of unnecessary transition lag. The user can always change
this value manually and/or use scene collections, so change the default
setting to make it not unload/reload by default feels a bit more safe.
This is probably not necessary but might fix an issue where errors pass
through to other parts of the program, possibly causing the crash on
exit related to the xcomposite capture.
Some games don't catch GL errors via glGetError, so there's a
possibility that an error will pass through to the capture calls,
causing a false failure.
The most simple solution is to just clear the error flag on each capture
call.
To first render the filter, the width/height values must be set, but
currently they're only set in the render function, which means that the
crop filter can never be rendered when the program first starts up.
This would cause the filter to fail to render at all under those
circumstances.
This patch moves the calculations from render to tick to ensure that
they're always called and the values are always set.
The virtual address table values for Reset/ResetEx can sometimes point
to functions that are in libraries outside of D3D8.dll and D3D9.dll, and
will cause a crash if used. Instead, just hook Reset/ResetEx when one
of the Present* functions are called.
Not calling recv when data is received will accumulate data in the
internal recveive buffer until it's full, in which case is will stop
acknowledging. This can lead to unjustified disconnections.
If there was an attempt to destroy the rtmp-stream output while it was
already connecting and stopping at the same time, it would try to join
with the stop thread rather than with the connect thread. The connect
thread would then continue past destruction.
Sometimes stopping a connection can lock up due to data that still
remains to be sent, and this would lock up the thread requesting the
stop (typically the UI thread). So instead of locking up the calling
thread, spawn a new thread specifically for stopping so the calling
thread can continue uninterrupted. If the user attempts to reconnect,
it will wait for the stop thread to complete in the connect thread
before attempting to connect.
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.
Use explicit UTF-8 byte sequence for the "no-break space" character.
Prevents issues with certain editors, and fixes the following compiler
warning on Visual C++:
warning C4819: The file contains a character that cannot be represented
in the current code page (X). Save the file in Unicode format to prevent
data loss
This replaces the name-based detection of the 4K intensity pro, and
allows other devices to be able to use the BGRA pixel format, if the
user so chooses.
Another thread could be manipulating the active_log_contexts array while the current thread is trying to read it, resulting in an uninitialized memory crash as the da_push_back call was not protected by the mutex.
Polls for file changes like the text plugin does. This is an interim
solution; both the text plugin and image source should use a file
monitoring API, preferably implemented through libobs.
Closesjp9000/obs-studio#482
When using a text file with the source and the font face is changed, it
would cause it to fail to update the glyphs and text accordingly. It
would trigger an error jump at line 392 of text-freetype2.c, ultimately
resulting in the text to render garbled after that.
How to reproduce:
Set the source to get text from a file, then just change the font face
(but not the size or anything else).
When updating text from file periodically, newer glyphs that weren't
already cached would not end up being rendered. This fixes the issue by
calling cache_glyphs after the file has been updated.
How to reproduce the original issue:
Set a text-freetype2 source to load an english-only text file. Then
overwrite the text in the file with non-english characters. The
non-english characters will then fail to render.
Reported at https://obsproject.com/mantis/view.php?id=336Closesjp9000/obs-studio#481
Instead of using shell functions to get the windows system directory,
use the kernel32 functions (GetSystemDirectory and
GetSystemWow64Directory). Reduces a bit of unnecessary overhead.
This caches the font list data to a file to minimize load times. Font
data will be refreshed when any font files are added/removed, based upon
a checksum of the font file names and dates (if available).
Microphones and other input devices can often have bad or erroneous
timestamps. Although we handle bad timestamps much better in
obs-studio, there are still lingering issues that can crop up from time
to time with device QPC timestamps that leads to mic data not playing
back properly. It's best if it be off by default rather than on, which
will now cause it to use system timestamps for input devices by default.
This changes it to the same handling as OBS1 for this case.
The new 'offset' value was not being passed back to the caller, which
caused the caller to continue to use the old value and thus would cause
an invalid hook and crash.
The call to CoInitializeEx in the win-mf module caused some sort of
conflict with the decklink module, causing the decklink module to crash
on exit. Instead, let libobs handle COM initialization.
If the GL capture part of the game capture hook fails to initialized for
whatever reason, it will go in to an infinite reacquire loop. If it
fails to initialize shared texture capture, try shared memory capture
instead.
For game capture, if a game is running at for example 800 FPS and limit
capture framerate is off, it would try to capture all 800 of those
frames, dramatically reducing performance more than what would ever be
necessary.
When limit capture framerate is off, instead of capturing all frames,
capture frames at an interval of twice the OBS FPS, identical to how
OBS1 works by default. This should greatly increase performance under
that circumstance.
This also adds the ability to detect whether it stopped due to lack of
space or not -- particularly useful for the FFmpeg output due to
lossless file format support.
For the FFmpeg output, the encoder ids are sort of superfluous. They
really should be optional. If they're not set, it should use the
encoder name string instead to determine the ids automatically.
It seems that certain encoders (quicksync) do not have proper back-end
support in the windows media foundation libraries for certain CPUs.
Quicksync doesn't appear to support CPUs that are not haswell (4xxx) or
above. It's really annoying, but there's not much we can do about it
until we implement our own custom quicksync implementation.
This check simply makes it attempt to spawn an encoder to check to see
whether the encoder can actually be created before registering an
encoder.
The previous commit (672378d20) was supposed to fix issues with the
encoder releasing while data was still being processed, but did not
account for when the encoder has never started up. That was my fault.
Furthermore, the way in which it was waiting to drain events was
incorrect. The encoder may still be active even though there aren't any
events queued. The proper way to wait for an async encoder to finish up
is to process output samples until it requests more input samples.
After I made it so that the encoder internal data gets destroyed when
all outputs stop using it (fa7286f8), the media foundation h264 encoder
started having crashes on shutdown. After a lot of testing, I realized
that the reason it started happening is almost assuredly because active
encoding events had not yet been completed.
After making it wait on those events by calling DrainEvents(true), the
crashes stopped. So asynchronous actions were clearly still occurring
and it was shutting down while data was still being processed, thus
leading to a crash.
Adds a VideoToolbox based H264 encoder for OSX, which most notably
allows the use of hardware encoding (Quicksync).
NOTES:
- Hardware encoding is handled by Apple itself internally. The plugin
itself has little control over many details due to the way that Apple
designed the VideoToolbox interface. Generally however, quicksync is
used if available on the CPU, and quicksync is almost always available
due to the fact that macs are exclusively Intel.
- The VideoToolbox does not seem to implement CBR, so it won't be
available. These encoders are generally not recommended for
streaming.
Implements hardware encoders through the Media Foundation interface
provided by Microsoft.
Supports:
- Quicksync (Intel)
- VCE (AMD)
- NVENC (NVIDIA, might only be supported through MF on Windows 10)
Notes:
- NVENC and VCE do not appear to have proper CBR implementations. This
isn't a fault of our code, but the Media Foundation libraries.
Quicksync however appears to be fine.
API changed from:
obs_source_info::get_name(void)
obs_output_info::get_name(void)
obs_encoder_info::get_name(void)
obs_service_info::get_name(void)
API changed to:
obs_source_info::get_name(void *type_data)
obs_output_info::get_name(void *type_data)
obs_encoder_info::get_name(void *type_data)
obs_service_info::get_name(void *type_data)
This allows the type data to be used when getting the name of the
object (useful for plugin wrappers primarily).
NOTE: Though a parameter was added, this is backward-compatible with
older plugins due to calling convention. The new parameter will simply
be ignored by older plugins, and the stack (if used) will be cleaned up
by the caller.
This is used by some muxers that set AVFMT_NOFILE and doesn't seem to
hurt muxers that don't set it; notable this makes the hls muxer output
its m3u8 playlist with the proper filename in the proper directory
cleaning up my previous commit a bit. we can just keep the
appropriate BMDPixelFormat as a data member and keep StartCapture() a
bit clearer.
this might also be helpful if (when?) the detection code needs to be
more robust or configurable
detect the device type when initializing the device instance and
determine whether to capture YUV or RGB. tested with a Blackmagic
Intensity Pro and a Blackmagic Intensity Pro 4K in the same machine,
capturing at the same time, on Linux
For both cases the cur_level calculations were "wrong". For one channel
case, I assume that was only an oversight, as for two channels case
cur_level "calculation", getting the level from downmixing to mono will
result in an attenuated level than expected. One solution is to use the
highest level of both channels to drive the gate.
..This is rather embarrassing. I used the parameter variable and the
actual variable that I wanted to used went completely unused. Would
static analysis catch something like this, I wonder? Would probably
have to be really good static analysis.
YouTube Gaming is live since today (26 August 2015) and people will ask
for it.
This makes it a bit clearer that YouTube and YouTube Gaming
(which share the same ingestion system) work with OBS MP.
This will use the services.json file present in the cache, or if it has
the wrong format version or is corrupted for whatever reason, uses the
local version instead.
Also a minor refactor, makes it so that you call the open_services_file
function to get the services array, rather than having to get the file
name each time.
This reverts commit 74354dc4cf. I really
shouldn't have modified this, especially not in this way. Was the wrong
approach. The thing I was trying to fix was very rare as well.
When a window being captured is closed, it never tries to reacquire.
This just searches for the window in video_tick and reacquires if the
currently set window is found again.
Closesjp9000/obs-studio#465
I made the rather tough call of not showing all services by default; I
didn't want to have to do this, but too many services are asking to be
put in to the program, and any time I add a service in to the list, I
feel uncomfortable because I feel like I'm potentially advertising them,
and/or they're using our program to advertise as well. Some of these
services are particularly bad at policing illegal/copyrighted content,
host content that I personally find distasteful or incredibly stupid
(what the heck is up with these "vaping" streams?), or are just fairly
terrible websites in general that I just feel uncomfortable with showing
by default.
However, I do not really want to reject anyone either, I want to let
their users be able to use our program with relative ease, but more than
anything I just simple don't want to be seen as "endorsing" some of
these websites (more than others in particular). I know that a "show
all services" checkbox is probably pretty pointless/superfluous thing to
do, but I feel like it's at the very least a means of saying "hey, I
don't really endorse these guys," or "use at your own risk," or
"warning: this website is incredibly terrible."
Honestly, I couldn't really think of any better solution that would
a.) still list all services without outright censoring them, and
b.) prevent us from being seen as "endorsing" all services.
(Although maybe this whole thing feels a bit.. passive aggressive. I
feel like I'm tipping over someone's garden gnome in the middle of the
night while they're sleeping. Still, it's something.)
NOTE: This code is backward compatible; i.e., if you previously had a
service selected that's not common but don't have the "show all"
checkbox checked, it'll still show that service for convenience.
Services almost always recommend this be enabled, and I generally want
to make configuration easier for users; with CBR they don't have to set
things like the CRF value.
The single darray solution was potentially unsafe since you're not
allowed to modify the (encode) buffer between calls to
complex_input_data_proc which is potentially violated if the darray
had to be resized due to capacity being < 2 * in_bytes_required
This is my fault; I made an idiotic assumption about the data and it
ended up causing the plugin to crash. This is definitely one of my more
embarrassing moments.
This just changes the x264 encoder settings; it doesn't actually change
the framerate of OBS. OBS will always output at a constant framerate
regardless of whether this option is on or off; this just changes how
the encoder encodes the data.
With no stream key, no streams were actually being created.
This is a crazy configuration anyway, but it resulted in OBS getting
stuck in the "Connecting" state with no way to cancel.
We now just use the blank key and hope for the best.
Reinstate flag checks in RTMP_Close that were erroneously removed.
Clear out the Link state before we establish a new connection. There is
too much state carried around during authentication that has no good
place to clear it in librtmp, which assumes a clean structure when the
connection is initially established.
Adds Microsoft Media Foundation AAC Encoder that supports
96k to 192k bitrates. This plugin is only enabled on Microsoft
Windows 8+ due to performance issues found on Windows 7.
Authentication code has been updated as per the changes to support
multiple streams.
Authentication is now also enabled by default, and should be a no-op
if the server does not request authentication or username and password
details are not provided.
I've come to realize that it's probably not wise to deviate from the
original version's functionality due to the fact that the original
version works without issues. I'm wondering if some of the capture
problems have been due to the fact that the direct hook method (via
CreateRemoteThread) was removed, so I put it back in, made it default,
and added an option to use anti-cheat compatibility just like in the
original version.
This particularly affected audio encoding, audio encoding previously
would count samples and use it to create an encoding timestamp, but
because I was using a standard integer (which is 32bit by default on
x86), it would max out at about 0x7FFFFFFF samples, which is about 12
hours of samples at 48000 sample rate. After that, it would start going
into negative territory (overflowing). By changing it to int64_t, it
will make it so that audio at 48000 samples per second would only be
able to overflow after about.. 6.09 million years. In other words,
this should fix the issue for good.
Livecoding.tv (coding), gaminglive.tv (gaming), and beam.pro
(gaming/music)
I really don't see any problems with adding these particular services to
the local list while the actual remote ingest lookup code has yet to be
even started yet (as of this writing). They seem to be harmless
services that are dedicated to specific types of content (stated above).
When hooking 64bit functions, sometimes the offset between the function
being hooked and the hook itself can be large enough to where it
requires a 64bit offset to be used. However, because a 64bit jump
requires overwriting so many code instructions in the function, it can
sometimes overwrite code in to an adjacent function, thereby causing a
crash.
The 64bit hook bounce (created by R1CH) is designed to prevent using
very long jumps in the target by creating executable memory within a
32bit offset of that target, and then writing it with the 64bit long
jump instruction instead. Then in the target function, it will jump to
that memory instead, thus forcing the actual hooked function to use a
32bit hook instead of a 64bit hook, and using at most 5 bytes for the
actual hook, preventing any likelihood of it overwriting an adjacent
function.
System timestamps were being used instead of timestamps from the
audio/video input. This would cause potential desync as well as
incremental buffering when using devices with the blackmagic video
source. Using the timestamps direct from the SDK itself fixes those
issues, and causes audio/video to play back properly and in sync.
This filter simply modifies the volume of the signal as a convenient way
of modifying the volume before other filters, or amplify the volume
without having to mess with advanced audio properties.
In addition to the flv file format, this allows the ability to save to
container formats such as mp4, ts, mkv, and any other containers that
support the current codecs being used.
It pipes the encoded data to the ffmpeg-mux process, which then safely
muxes the file from the encoded data. If the main program unexpectedly
terminates, the ffmpeg-mux piped program will safely close the file and
write trailer data, preventing file corruption.
Instead of using system timestamps for playback, use the timestamps
directly from the video/audio data to ensure all the data is synced up
using the obs_source back-end.
I think the original misconception when this was written was that OBS
would not handle timestamp resets/loops, which isn't the case; it will
actually handle all timestamp resets and abnormalities. It's always
best to use the obs_source back-end for all playback and syncing.
When the bitrate was set to 64 CoreAudio would call
complex_input_data_proc more than once, which in turn would cause
consumed bytes in the input buffer to be "freed" more than once (once
for every additional call of complex_input_data_proc and once in
aac_encode)
This allows the ability to output the audio of the device as desktop
audio (via the WaveOut or DirectSound audio renderers) instead of
capturing the audio only.
In the future, we'll implement audio monitoring which will make this
feature obsolete, but for the time being I decided to add this option as
a temporary measure to allow users to play the audio from their devices
via the DirectShow output.
Found via UBSan, actual (sample) error:
"plugins/text-freetype2/text-functionality.c:284:26: runtime error: left
shift of 194 by 24 places cannot be represented in type 'int'"
Add the include directories found by cmake to the jack plugin.
This allows for the plugin to compile when the jack headers were
found in a directory that is not normally in the search path of the
compiler (e.g. /usr/local/include)
Add the include directories found by cmake to the v4l2 plugin.
This allows for the plugin to compile when the vl42 headers were
found in a directory that is not normally in the search path of the
compiler (e.g. /usr/local/include)
Add the include directories found by cmake to the pulseaudio plugin.
This allows for the plugin to compile when the pulseaudio headers were
found in a directory that is not normally in the search path of the
compiler (e.g. /usr/local/include)
Fix build errors for older versions of the api where
VIDIOC_ENUM_DV_TIMINGS was defined but V4L2_IN_CAP_DV_TIMINGS was not.
I was under the impression that they were added at the same time, but
apparently i was wrong there.
Thanks to kmoore@FreeBSD.org for spotting this on FreeBSD.
In the settings if you select default container then the
format becomes null. If null then audio/video codec ids should
not be set on the output format as they will both be
AV_CODEC_ID_NONE causing a context with no streams specified
to be created (error).
Add compatibility with older versions of the api by not failing to
build when the VIDIOC_ENUM_DV_TIMINGS is missing. In older versions
of the api there was a different system to get dv-timing presets, which
was replaced by the current enumeration system with Linux 3.4.
This will allow for the plugin to be built against older versions of the
api by disabling the enumeration support, thus reducing the
functionality for some devices.
Improve compatibility with older versions of the api by not requiring
V4L2_CAP_DEVICE_CAPS. If we don't have this, we fall back to using the
capabilities member for the whole device instead of the device_caps
member for the currently selected subdevice. Just like we would do if
the device would not support this.
The new device_caps field was introduced with Linux 3.3.
Add BGRX and BGRA as supported video formats, since obs can handle them
directly. I unfortunately missed those when i initially wrote this
mapping due to my webcam not offering those formats.
API Changed (in struct obs_encoder_info):
----------------------------------------
bool (*get_audio_info)(void *data, struct audio_convert_info *info);
bool (*get_video_info)(void *data, struct video_scale_info *info);
To:
----------------------------------------
void (*get_audio_info)(void *data, struct audio_convert_info *info);
void (*get_video_info)(void *data, struct video_scale_info *info);
The encoder video/audio information callbacks no longer need to manually
query the libobs video/audio information, that information is now passed
via the parameter, which the callbacks can modify.
The refactor that reduces boilerplate in the encoder video/audio
information callbacks also removes the need for their return values, so
change the return types to void.
I'm putting this option in due to the fact that there are legitimate
cases where a device may flip the output unexpectedly (such as the
Datapath VisionDVI-DL running in RGB video format), and that a user may
want to be able to view the source in a projector or source properties
without the image being inverted.
My original line of thinking was that they can just use a transform to
flip the image, but I felt this problem impacts rendering everywhere,
such as in the projector and in the source properties, so having it as
an option in the source itself feels like the best way to ensure that a
user can get it to render everywhere properly.
Check the actual name of the codec before applying an x264-specific
preset so we don't encounter an "Invalid argument" error when using
other h264 encoders in FFmpeg (such as NVEnc).
Closesjp9000/obs-studio#412
Changes:
- Prevent concurrent calls to EnumDevices (resolves a crash with
some device filters (like the XCAPTURE-1) with multiple active
dshow sources)
Adds the following changes:
- Prioritize YUV formats over non-YUV formats for performance and to
prevent intermediary filters
- Directly connect filters when possible to avoid intermediary filters
When frames were dropped, it would also drop I-frames, which can mess
with the keyframe calculation of certain services that depend on
I-frames in their output protocol (such as HLS).
The kCGDisplayStreamShowCursor option used with the dictionary does not
work if you assign @true or @false to it. After some testing, it needs
to point to the id cast of either kCFBooleanTrue or kCFBooleanFalse in
order for it to work properly.
If it doesn't use either of those values, the display stream seems to
use its internal default, which on 10.8 and 10.9 is visible, and 10.10+
is invisible, which would explain why people on 10.10 couldn't get the
cursor to capture.
Some formats (like WMV) would send out audio packets that
had channels set but did not specify a channel layout.
Solution is to no longer rely on channel layout to get the
channels and just get the channel count directly off the
FFmpeg audio frame.
If capture starts too quickly, the file mapping will return 2, which
means file not found, and it would then reset the capture and try again.
Sometimes this would result in long intervals where it wouldn't capture.
This fixes the issue by simply making game capture retry if file mapping
returns error number 2.
This allows applying a mask based upon the chroma value (U/V) of a
specific color in YUV color space. Commonly used to mask out
greenscreens and bluescreens in live video.
Allows any source to be cropped, though note that this renders to
texture first, so for more optimal results, cropping values should
probably be put in to capture sources that can be cropped as they're
actually rendered by the source.
This filter allows the ability to use a texture to modify the video
output of a source, the ability to:
- apply a color-based mask (dark = transparent, light = opaque)
- apply a mask based upon the alpha of an image
- blend an image via subtraction, addition, or multiplication
Adds a filter for delaying asynchronous video/audio data, for example
from webcams, video devices, or media playback sources. Mainly intended
to allow users to sync up their webcams to other devices.
If the settings are reset to defaults or if the settings are just bad,
the video would get stuck on the last frame that was displayed, which
feels a bit awkward. Best to make it stop video output entirely rather
than get stuck on the last video frame.
Certain devices (particularly certain mixers and soundflower 64ch) would
have an arbitrary number of channels, and wouldn't really be mappable to
a specific speaker layout supported by libobs.
So to fix this issue, if the channel count is above 8, force the data to
stereo to ensure playback can still occur, rather than cause it to just
fail.
This fixes an issue primarily with filter rendering: when capturing
windows and displays, their alpha channel is almost always 0, causing
the image to be completely invisible unintentionally. The original fix
for this for many sources was just to turn off the blending, which would
be fine if you're not rendering any filters, but filters will render to
render targets first, and that lack of alpha will end up carrying over
in to the final image.
This doesn't apply to any mac captures because mac actually seems to set
the alpha channel to 1.
The code specific to Windows: helps convert `BSTR` instances to
`std::string`s; provides a Windows COM-specific implementation of
`CreateDeckLinkDiscoveryInstance`; aliases CFUUIDGetUUIDBytes,
CFUUIDBytes, and IUnknownUUID (the Linux SDK does this, but for some
reason the Windows SDK does not).
Some changes were made to the stock DeckLink SDK: removed all references
to legacy API headers in DeckLinkAPI.idl; removed all instances of
`importlib("stdole2.tlb");`.
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.
If this option is on, the image will unload when the image isn't
displayed anywhere, and then reload it when it's displayed again to
reduce the amount of memory images take up on VRAM at any given time.
If this option is off, then it's always loaded in VRAM regardless of
whether it's displayed or not.
Closes Pull Request #380
(edited by Jim)
Add a source property to enable buffering of frames, which is enabled by
default. This replaces the old "Use system timing" option by setting the
unbuffered source flag instead of using different timestamps.
While similar in intentions to the old option, this method should reduce
latency even more.
Fix bug 0000151: File loading not properly handled.
Link to bug: https://obsproject.com/mantis/view.php?id=151
A newly selected font is not loaded properly if "read from file" is
active without a valid file. Old error handling lead to random memory
being displayed.
Closes Pull Request #390
(message edited by Jim)
By default, video plays back based upon the timestamp for each frame,
and buffers the frames as needed to ensure that they play back at the
expected timing.
However, this can add some minor additional delay to the video, and may
not be ideal for certain devices such as webcams and generally any
device that has minimal latency. However, because those are the only
type of devices that typically have drivers, there's no real need to
have it on by default.
This adds an option to use buffering, and leaves it off by default.
Closes pull request #384
(message added by jim)
Use the macro from the mac capture plugin to convert the fourcc integer
value to a string. This makes the debug statement for the pixel format
slightly more readable for the casual developer.
Remove the "Leave Unchanged" option for the input and video format
select.
This option was primarily added for cases in which the
resolution and framerate are set by another program or the capture
device itself and the values are not directly supported by the plugin.
One major usecase here would be capture devices for tv signals which
might be set to a spcific resolution and refresh rate, and might fail
to initialize in case any other combination of those settings is forced.
In case of the input this option did not make much sense, as the first
input is probably the best default option in most cases.
For the video format this default is even bad in some cases. If an
format emulated by libv4l2 is selected for example, this will usually
configure the device to use mjpeg with libv4l2 converting it to some
format obs can use. When obs or the source is then restarted and the
"Leave Unchanged" is enabled the plugin will fail, because it will only
notice that the device is set to mjpeg, without any knowledge about the
possibility of letting libv4l2 handle the conversion.
Using the first available and supported format is not nescessarily the
best choice, but still preferable to some random format that will
cause the plugin to not capture at all. Forcing a choice here will
hopefully also make the plugin behaviour more predicatable for the user.
Remove the constraint for device inputs to be of the type "CAMERA".
This was added under the false assumption that inputs of the type
"TUNER" are only used for control purposes.
This was caused to do the new RTMP code that added support for multiple
streams; the stream index needs to be reset on RTMP_Close otherwise it
will keep using the wrong stream information.
I had this issue where IDXGISwapChain::ResizeBuffers would fail in the
hooks, causing games to crash when they resized their backbuffers
because ResizeBuffers would return an 'invalid call' HRESULT value. In
the ResizeBuffers documentation it says that it will only happen if a
backbuffer currently has any outstanding references, but there's no way
this would happen unless ResizeBuffers internally calls Present or vise
versa.
After ResizeBuffers has been called, the very first call to Present will
somehow seemingly invalidate and/or destroy the current backbuffer.
It's very strange, but that seems to be what's going on, at least for
the game I was testing. So if you are performing a post-overlay
capture, then you must ignore the capture on the very first call to
Present.
It's Microsoft's code so you can't really know what's going on, you just
have to work around these strange issues seemingly in the dark.
Martell changed this function without realizing that this was calling a
function below it, not recursively calling itself. The reason why he
got the warning was because there was no forward declaration of the
function that was being called; I think he's used to C where only one
function definition can exist with the same name. In this case, it was
another function with the same name but with different parameters,
something that's permitted in C++. I wish I had realized this sooner.
This fixes the crashes people have been having with devices.
Apparently someone dumb (aka me) neglected to properly handle the inline
graphics hook API functions. You're not supposed to 'extern' inline
functions, they need to be defined for each file when ever they're used.
Apparently neglected to use the reference operator. I think this may
partially be one of the reasons why many developers still choose to use
pointers instead of references, but fortunately an actual GOOD compiler
warns about this (aka anything but vc)
Clears up a warning (to prevent && and || confusion), and clarifies what
specifically the if statement is trying to accomplish (check to see if
the capture is valid)
On windows, for whatever reason sockets use the SOCKET type which is not
a signed integer. Still, even though it's not a signed integer, -1 is
used to indicate an invalid socket, but the way you use it is via
microsoft's fabulously dumb little INVALID_SOCKET define, so we have to
make librtmp use that instead.
The HWND type is a void pointer, but HWND values are global and always
32bit despite, so casting to 32bit can cause cast warnings on actual
good compilers like gcc via mingw. This change correctly handles the
casting to 32bits without producing unwanted warnings or errors on
mingw.
win-capture should not postfix .lib to psapi.
The graphics hook also requires psapi when linking.
Also change some link libs as mingw-w64 libraries are not postfixed
.lib.
Hopefully we can get this function merged for mingw-w64 4.1. As for the
4.0 release, adding a new header is a big change, it'll have to wait for
the next version.
Remove the .lib postfix from strmiids
ksuser provides KSCATEGORY_ENCODER and similar GUIDS used
wmcodecdspuuid provides MEDIASUBTYPE_H264 MEDIASUBTYPE_RAW_AAC1 and
MEDIASUBTYPE_I420 so no need to define them in dshow-formats. The
submodule will have to be updated to support this change.
I feel like due to lack of user understanding, it's important to specify
that the higher the preset is (veryfast/superfast/ultrafast) the less
CPU that the encoder will use
Gol-D-Ace
Charles Ray Shisler III
jp9000
Radzaquiel
yogpstop
dodgepong
Lucian Poston
Palana
Richard Stanway
Aarni Koskela
Copy Liu
kc5nra
Brian S. Stephan
Mathias Panzenböck
hwdro
Gökberk Yaltıraklı
Marc Chambers
Haden F
Jess Mayo
Taylor Blau
CommanderRoot
Night
Bl00drav3n
fryshorts
Kris Moore
GoaLitiuM
vividnightmare
Ethan Lee
Skyler Lipthay
Christoph Hohmann
Manuel Kroeber
Martell Malone