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
Add CBR, CRF to properties so that it can be changed by the user. If
CBR is on, CRF will be disabled. Also added a 'Use Custom Buffer Size'
option to make it so that the buffer size will automatically be set to
the bitrate if its value is false. This is primarily a convenience
feature for users.
Certain RTMP services will support multi audio tracks via RTMP. This
updates librtmp with custom code that enables multiple streams per
connection to be used; each subsequent stream typically containing extra
audio tracks. The audio encoder names are used to indicate the names of
tracks, and the name of the tracks are used for the stream keys for
those subsequent tracks.
This makes FFmpeg usable as an output, and removes or changes most of
the code that was originally intended for testing purposes.
Changes the settings for the FFmpeg output to the following:
* url: Sets the output URL or file path
* video_bitrate: Sets the video bitrate
* audio_bitrate: Sets the audio bitrate
* video_encoder: Sets the video encoder (by name, blank for default)
* audio_encoder: Sets the audio encoder (by name, blank for default)
* video_settings: Sets custom video encoder FFmpeg settings
* audio_settings: Sets custom audio encoder FFmpeg settings
* scale_width: Image scale width (0 if none)
* scale_height: Image scale height (0 if none)
The reason why scale_width and scale_height are provided is because it
may internally convert formats, and it may be a bit more optimal to use
that scaler instead of the pre-output scaler just in case it already has
to convert formats internally anyway (though you can do it either way
you wish).
Video format handling has also changed; it will now attempt to use the
closest format to the current format if available for a given video
codec.
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.
Add a check to the cursor render function to ensure the cursor texture
exists. It seems like it is very unlikely but still possible, that the
first tick which should set the texture might fail. In that case obs
would crash in the render function.
jp9000
Gol-D-Ace
kc5nra
Skyler Lipthay
Radzaquiel
Christoph Hohmann
fryshorts
Manuel Kroeber
Martell Malone
Bl00drav3n
dodgepong
Palana
Bernd Buschinski
Akagi201