Adds functions to allow sources to inform the UI whether the audio is
currently active or not. Allows the ability to turn on/off the items in
the mixer.
If for whatever reason the format is swapped between BGRA/BGRX in an
async filter, swap the texture to compensate for that. This allows
plugins to change the format if necessary.
Originally when the audio_submix function was created, it used all mixer
tracks, but at a certain point that was removed because it can only use
the first track, so some older code was unintentionally left over,
causing the same code to be executed 6 times mistakenly. This cleans
that up by removing the unnecessary function parameter and for loop.
This fixes a race condition where the audio/video backends/threads may
start using sources before their obs_source_info::create function has
been called.
Adds the "audio_line" internal source type as a bare source type for the
sole purpose of outputting audio, and the obs_source_info::audio_mix
callback which allows mixing of those audio lines, which is then treated
as normal audio for the source. Audio line objects should be added as
sub-sources when multiple audio lines from a single source are needed,
then mixed together with the audio_mix callback.
The difference between the new obs_source_info::audio_mix callback and
obs_source_info::audio_render is that obs_source_info::audio_mix (along
with the audio_line source) are only one track, and it outputs audio to
the source automatically via obs_source_output_audio() when the call
completes. This allows the mixed audio to be treated like a normal
source's audio, in that you can filter it, change its volume, or monitor
it.
This change was necessary because the CEF (used with the browser source)
outputs multiple audio streams at once to a single browser source, so
it's the program's responsibility to mix those streams together itself.
The shaders to unpack YUV information from the same texture were rather
complicated. Breaking them up into separate textures makes the shaders
much simpler, and we can remove the PRECISION_OFFSET hack.
Performance also gets a nice boost on Intel for planar textures.
Intel GPA, SetStablePowerState, Intel HD Graphics 530, 1920x1080
UYVY: 473 us -> 457 us
YUY2: 492 us -> 422 us
YVYU: 491 us -> 441 us
I420: 1637 us -> 505 us
I422: 1644 us -> 482 us
I444: 1653 us -> 504 us
NV12: 1656 us -> 369 us
Y800 (limited): 270 us -> 277 us
Y800 (full): 263 us -> 289 us
RGB (limited): 341 us -> 411 us
BGR3 (limited): 512 us -> 509 us
BGR3 (full): 527 us -> 534 us
The video format is not updated if switching between cache-compatible
formats, e.g. YUY2 and YVYU, resulting in the wrong conversion technique
being used. This change ensures the format is always up-to-date.
Unlike get_properties, there is not reason to not call get_defaults if it is
given in addition to get_defaults2. Additonally this fixes the bug with
'init_encoder' which would only ever call get_defaults, resulting in broken
encoders if those used get_defaults2.
If an audio source does not provide enough data at a steady pace, the
timestamp update does not happen, and buffering increases until it
maxes out. To counteract this, update the timestamp anyway.
Another issue for decoupled audio sources is that timing is not
adjusted for divergence from system time. Making this adjustment is
better for timing stability.
5+ hours of stable audio without any buffering on my GV-USB2 where it
used to add 21ms every 5 mintues or so.
Fixes https://obsproject.com/mantis/view.php?id=1269
Fix ternary test to use BGRX render targets for YUV to RGB
conversions. The previous behavior may have been fine though since
the shaders fill the alpha channel with 1.0 anyway.
Code submissions have continually suffered from formatting
inconsistencies that constantly have to be addressed. Using
clang-format simplifies this by making code formatting more consistent,
and allows automation of the code formatting so that maintainers can
focus more on the code itself instead of code formatting.
The cache coherency of rasterization for full-screen passes is better
using an oversized triangle that is clipped rather than two triangles.
Traversal order of rasterization is GPU-specific, but will almost
certainly be better using an undivided primitive.
A smaller benefit is that quads along the diagonal are not evaluated
multiple times, but that's minor in comparison.
Redo format shaders to bypass vertex buffer, and input layout. Add
global shader bool "obs_glsl_compile" to make API-specific decisions,
i.e. handle upside-down UVs. gl_ortho is not needed for format
conversion because the vertex shader does not use ViewProj anymore.
This can be applied to more situations, but start small first.
Testbed full screen passes, Intel HD Graphics 530:
RGBA -> UYVX: 467 -> 439 us, ~6% savings
UYVX -> uv: 295 -> 239 us, ~19% savings
A previous refactoring to make DrawMatrix unnecessary has left behind
unreachable YUV conversions. Even if this code was somehow reachable,
DrawMatrix for YUV -> RGB doesn't exist anymore, so they would render
incorrectly anyway.
This commit adds a function to forcefully stop a transition, and to
increment/decrement the showing counter for a source with the MAIN_VIEW
type.
These functions are needed for the transition previews to work as
intended.
libobs: Add support for limited to full color range conversions when
using RGB or Y800 formats, and move RGB converison for Y800 formats to
the GPU.
decklink: Stop hiding color space/range properties for RGB formats, and
remove "YUV" from "YUV Color Space" and "YUV Color Range".
win-dshow: Remove "YUV" from "YUV Color Space" and "YUV Color Range".
UI: Remove "YUV" from "YUV Color Space" and "YUV Color Range".
Fixes handling of the `obs_source_frame::full_range` member variable,
which is often set to false by default by many plugins even when using
RGB, which would cause RGB to be marked as "partial range". This change
is crucial for when partial range RBG support is implemented.
Adds `obs_source_frame2` structure that replaces the `full_range` member
variable with a `range` variable, which uses the `video_range_type` enum
to allow handling default range values. This member variable treats
VIDEO_RANGE_DEFAULT as full range if the format is RGB, and partial
range if the format is YUV.
Also adds `obs_source_output_video2` and `obs_source_preload_video2`
functions which use the `obs_source_frame2` structure instead of the
`obs_source_frame` structure.
When using the original `obs_source_frame`, `obs_source_output_video`,
and `obs_source_preload_video` functions, RGB will always be full range
by default for backward compatibility purposes.
Currently several shaders need "DrawMatrix" techniques to support the
possibility that the input texture is a "YUV" format. Also, "DrawMatrix"
is overloaded for translation in both directions when it is written for
RGB to "YUV" only.
A cleaner solution is to handle "YUV" to RGB up-front as part of format
conversion, and ensure only RGB inputs reach the other shaders. This is
necessary to someday perform correct scale filtering without the cost of
redundant "YUV" conversions per texture tap.
A necessary prerequisite for this is to add conversion support for
VIDEO_FORMAT_I444, and that is now in place. There was already a hack in
place to cover VIDEO_FORMAT_Y800. All other "YUV" formats already have
conversion functions.
"DrawMatrix" has been removed from shaders that only supported "YUV" to
RGB conversions. It still exists in shaders that perform RGB to "YUV"
conversions, and the implementations have been sanitized accordingly.
Add D3D/GL debug markers to make RenderDoc captures easier to tranverse.
Also add obs_source_get_name_no_null() to avoid boilerplate for safe
string formatting.
Closesobsproject/obs-studio#1799
Currently SrcBlendAlpha and DestBlendAlpha are both ONE, and can
combine together to form two. This is not a noticeable problem for
UNORM targets because the channels are clamped, but it will likely
become a problem if FLOAT targets are more widely used.
This change switches DestBlendAlpha to INVSRCALPHA, and starts
backgrounds as opaque black instead of transparent black. The blending
behavior of stacked transparents is preserved without overflowing the
alpha channel.
If audio monitoring is enabled and set to output only, the
obs_source_show_preloaded_video function would still incorrectly set the
current source audio output timestamp to the current system time, which
would cause audio to use an incorrect starting point from long ago for
the first starting audio segment if audio monitoring is then turned off
some time after having started the source.
Instead, the starting timestamp should be set to 0 if audio monitoring
is enabled with no output to stream, so that if/when audio monitoring is
disabled, it recalculates the starting timestamp of the first audio
packet on the spot again.
Closesobsproject/obs-studio#1522
Don't allow calling a source's custom get_width or get_height callbacks
unless the context is valid. Fixes a bug where a null pointer could be
passed to those functions.
The filters array should not be accessed without locking the filter
mutex. This locks the filter mutex, grabs a reference to the first
filter, unlocks the mutex, renders the filter, and then releases the
reference.
Closesobsproject/obs-studio#1215
Solves crash when a user tries to paste filters without selecting a
source or when pasting filters from a deleted source. This commit
checks if both sources are still valid before pasting.
This addresses Mantis Issue 1220
(https://obsproject.com/mantis/view.php?id=1220).
If a filter's implementation (its plugin for example) no longer exists,
it would cause the source to stop rendering if that filter was present
on the source. Instead, just bypass the filter to ensure that the
source continues to render.
The memset in custom_audio_render() did not clear all audio buffers when
the number of output channels was less then 8. This caused wrong audio
output on mixes that did not get cleared.
Closesjp9000/obs-studio#1123
Because it would be troublesome to add the ability to remove source
types (in case for example a script fails to reload), instead make it so
source types can be temporarily disabled while the program is running.
Decoupling the audio from the video causes the audio to be played right
when it's received rather than attempt to sync up to the video frames.
This is useful with certain async sources/devices when the audio/video
timestamps are not reliable.
Naturally because it plays audio right when it's received, this should
only be used when the async source is operating in unbuffered mode,
otherwise the video frame timing will be out of sync by the amount of
buffering the video currently has.
When an async video frame comes in and it sets the timing_adjust value
(used to sync audio to video based upon their timestamps), it would use
os_gettime_ns as a base. Instead, it should use OBS' current video
frame time so that the audio and video playback is as accurate as
possible relative to the actual exact timestamp of the video frame.
(Results are almost insignificant, but it's nice to be as precise as
possible)
Fixes a potential issue where copying filters from one source to another
might add filters from the old source that are not compatible with the
new source.
(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.
Eventually, most things should be replaced with Load where applicable
(though in some cases sub-pixel sampling is desired).
This commit also fixes a bug where NV12 async sources wouldn't render
correctly.
(Note: This commits also modifies the linux-pulseaudio, mac-capture, and
win-wasapi plugins)
Do not prevent the targeted output device from being monitored if the
selected monitor output device is a different one.
Closesjp9000/obs-studio#872
This change prevents source flags from being incorrectly overwritten and
set to 0. Eventually flags need to be separated from source settings
and this should be reverted, but for now this solves an issue where
buffering would be enabled on async video sources regardless of whether
the user disabled it or not on the source.
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).
Fixes a bug that would allow possible infinite recursion within a source
tree. To fix this, inactive sources must be enumerated as well in order
to prevent infinite recursion.
Commit 53955301a2 introduced a async source texture copy bug due to
creating a new case in a switch without adding a break to the one above
it, causing it to execute both cases by mistake.
Because D3D11 specifically does not support an L8 texture format (you
have to use a shader swizzle), manually convert Y800 signals to RGBX
instead. This also fixes a bug where Y800 signals will render red.
Closesjp9000/obs-studio#718
If an async source is cropped on one side, then when the program is
restarted and the source is loaded from file, the async source will
start out with a width/height of zero. This will cause the async source
to not be drawn if cropping or scale filtering is added to the scene
item, because it has to be rendered to a texture first. However, the
source cannot reset its size until it's drawn, so it leaves it in
perpetual state of having a 0x0 size.
This fixes that problem by ensuring that the async source size is always
reset even when not being rendered.
Closejp9000/obs-studio#686
When a scene is duplicated the filters on the scene were not copied to
the new scene. This causes that a temporary copy of a scene renders
differently in the program than in the preview when using studio mode.
(Note: This commit also modifies coreaudio-encoder, win-capture, and
win-mf modules)
This reduces logging to the user's log file. Most of the things
specified are not useful for examining log files, and make reading log
files more painful.
The things that are useful to log should be up to the front-end to
implement. The core and core plugins should have minimal mandatory
logging.
The active_refs and show_refs variable would only increment/decrement
their children if their values were 1 and 0, which means that in the
case of scenes within scenes, sub-sources of scenes within scenes would
end up having the wrong ref values.
When using GPU conversion for 4:2:0 frames on async video sources, it
would create a texture bigger than necessary and try to copy too much
data from the frame, resulting in a crash.
When a transition is a sub-source of another source, it would not call
the transition's active source enum function, meaning that any sources
the transition had would not increment their active/showing refs (it
would only be called when activating the transition directly before).
That would result in negative/invalid active/showing refs on its
sub-sources, causing them to become permanently active/inactive and/or
permanently showing/hidden.
(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)
The source shouldn't be inserted into obs->data.first_audio_source until it's
fully initialized, or other threads will access source->control and
dereference an uninitialized pointer.
This patch fixes a specific crash where if the user named a filter the
same name as an input source that already existed in the system, scene
item loading code could find the filter with the same name instead of
the source, and mistakenly use it as the scene item's source directly.
This would cause a crash when trying to render that filter as a regular
source.
Marking filters as private is a temporary and simple workaround to the
solution. Filters are currently not meant to be found via the main
enumeration/search functions, which is a design flaw (lack of
consistency). In future major API revisions of libobs, filters should
be reworked to act as sources, with the sources they filter as
sub-sources ideally.
Additionally, the concept of "private context objects" and "primary
lists of context objects" in the back-end should probably also be
removed, allowing the font-end (or optional separate API layers) to
control all primary lists of obs context objects. These minor issues
that occur ultimately stem from API design flaws which need to be
corrected.
This crash happened when a filter was mistakenly used as a regular
source due to an unrelated bug in filter code and scene loading code.
The filter and the source it belongs to both had the same names, and the
source loading code found the filter and mistakenly used it as the
source instead of the actual source with the same name.
(Note: Also modified the obs-ffmpeg plugin module)
Allows the ability for frame data to pass 8-bit grayscale images (Y800
color format).
Closesjp9000/obs-studio#515
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).
Instead of just updating the async texture variables directly in the
source, allow the ability to pass the async texture variables via
function parameters to allow the ability to parse more than one frame to
more than one texture.
This code is primarily intended to be used to upload/convert the
"previous" async frame for the deinterlacer (if necessary).
(Note: This commit also modifies obs-filters and text-freetype2)
This simplifies writing of effects. DrawMatrix is no longer necessary
because there are no sources that require drawing with a color matrix
other than async sources, and async sources are automatically processed
and don't defer their initial render stage to filters.
Under certain circumstances, the timing_adjust variable would cause line
1161 to continually trigger over and over again. The "loop detection"
code incorrectly made it so that any timestamp that was just simply
below the expected value would be seen as a jump. After that, the
timing_adjust variable would be set for the frame again, and then the
audio would see it as a jump again after that, and those two things
would continue endlessly. This would cause stuttering particularly with
certain devices (particularly elgato/lgp/hdpvr) where the audio/video
data are decoded and sent at varying/different/unpredictable times.
To fix this issue, it should not detect values below as jumps, but
instead should only do it for values that exceed the MAX_TS_VAR (maximum
timestamp variance) value.
If obs_source::audio_ts is set to 0 (such as by discard_if_stopped in
obs-audio.c), but the push_back variable in the source_output_audio_data
function in obs-source.c was being set to true (meaning it's within the
seamless audio smoothing threshold), it would cause it to never reset
the obs_source::audio_ts value, and thus all audio data from the source
would become perpetually ignored by the audio subsystem until there was
finally some sort of timestamp jump that caused it to call
source_output_audio_place, and thus reset obs_source::audio_ts.
obs_source::audio_ts is only reset in source_output_audio_place, not in
source_output_audio_push_back, so the most simple solution is to just
call source_output_audio_push_back is obs_source::audio_ts is 0.
There's technically no need to clear the audio data here, nor is there
any need to try to trick the timestamp in to a different position. It
can simple just reset the audio timing.
Prevents a possible case where audio data might be deleted when it's not
necessary to delete any.
This variable is used to detect whether audio has stopped -- if audio
stops, it detects that no new data is coming in, and resets the audio
position so that it eliminates the chance of causing the audio buffering
to go haywire if audio starts up again. However, this variable was not
being reset every time the value changes, which it should.
The seamless audio looping code would erroneously trigger for things
that weren't loops, causing the audio data to continually push back and
ignore timestamps, thus going out of sync.
There does need to be loop handling code, but due to the fact that other
things may need to trigger this code, it's best just to clear the audio
data and start from a fresh sync point. Unfortunately for the case of
loops, this means the window in which audio data loops and video frames
loop need to be muted.
This fixes an age-old issue where audio samples could be lost or audio
could temporarily go out of sync in the case of looping videos. When
audio/video data is looping, there's a window between when the audio
data resets its timestamp value and when the video data resets its
timestamp value. This method simply pushes back the audio data while in
that window and does not modify sync, and when it detects that its out
of the loop window it simply forces a resync of the audio data in the
circular buffer.
This ensures that minimal audio data is lost in the loop process, and
minimizes the likelihood of any sort of sync issues associated with
looping.
Instead of applying the resampler offset right away (to each audio
packet), apply the resampler offset when the timestamps are converted to
system timestamps. This fixes an issue where if audio timestamps reset
to 0 (for whatever reason), the offset would cause the timestamp to go
in to the negative.
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)
Adds a "composite" source type which is used for sources that composite
one or more sub-sources. The audio_render callback is called for
composite sources to allow those types of sources to do custom
processing of the audio of its sub-sources.
(Note: This commit breaks libobs compilation. Skip if bisecting)
This variable is somewhat redundant. Volume is already known/accessible
to front-ends.
(Note: This commit breaks libobs compilation. Skip if bisecting)
Removes audio lines and stores the circular buffer for the audio on the
source itself.
(Note: This commit breaks libobs compilation. Skip if bisecting)
The mixers that a source was assigned to were originally stored in the
audio line. This will store it in the sources themselves instead.
Originally this was programmed to call the recursive height/width
functions if the source type was an input with the intention of not
calling it on filters, but instead of doing that just program it to do
just that: only call the recursive height/width functions if it's not a
filter.
This was originally used for calculating audio volume if transitions
were active, but transitions won't work that way so tracking the active
transitions is no longer needed.
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)
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 shouldn't happen anymore because crop was fixed, but if a filter
returns 0x0 size and is invalid it shouldn't stop the filter chain.
Instead, it should just be skipped.
When an async video source is about to be rendered, the async texture
should be updated before any effect filtering occurs, rather than right
when it's about to render.
Fixes a few bugs:
- If the async texture hadn't drawn for its first time, and the source
has an effect filter, it would never end up rendering the first
frame due to the fact that it would fail on obs-source.c:2434 for the
first filter, causing it to never actually render the source, and thus
never get to a point in which it could call set_async_texture_size to
establish the async texture width/height for the first time.
- Any time the async texture size changed, it would only update the
async texture size at the end of the filter loop, which means that the
first frame after a size change would use the old size for the filters
rather than update to the new size right away.
VodBox
Exeldro
jp9000
jpark37
SoraYuki
James Park
Michael Fabian 'Xaymar' Dirks
wangrui05
cg2121
Bas van Meel
Alex Anderson
Christoph Hohmann
Shaolin
SuslikV
Zachary Lund
Richard Stanway
jpk
John Bradley