(Jim) For some reason a character outside of the standard ASCII
character set was used in this source file. "Eight Spoked Asterisk",
Unicode character 0x02733. Non-ASCII characters should probably not be
used directly in source files due to potential character encoding
issues that can occur depending on the operating system and compiler.
Closesjp9000/obs-studio#485
When an encoder has been removed (such as CoreAudio) and the audio
bitrates currently configured no longer are available to the current
audio encoders anymore, it would cause GetAACEncoderForBitrate to return
false with no encoder available.
To fix the issue, instead just choose the closest bitrate relative to
the current bitrate (rounded up).
So certain high-profile individuals were complaining that it was
difficult to configure recording settings for quality in OBS. So, I
decided to add a very easy-to-use auto-configuration for high quality
encoding -- including lossless encoding. This feature will
automatically configure ideal recording settings based upon a specified
quality level.
Recording quality presets added to simple output:
- Same as stream: Copies the encoded streaming data with no extra usage
hit.
- High quality: uses a higher CRF value (starting at 23) if using x264.
- Indistinguishable quality: uses a low CRF value (starting at 16) if
using x264.
- Lossless will spawn an FFmpeg output that uses huffyuv encoding. If a
user tries to select lossless, they will be warned both via a dialog
prompt and a warning message in the settings window to ensure they
understand that it requires tremendous amounts of free space. It will
always use the AVI file format.
Extra Notes:
- When High/Indistinguishable quality is set, it will allow you to
select the recording encoder. Currently, it just allows you to select
x264 (at either veryfast or ultrafast). Later on, it'll be useful to
be able to set up pre-configured presets for hardware encoders once
more are implemented and tested.
- I decided to allow the use of x264 at both veryfast or ultrafast
presets. The reasoning is two-fold:
1.) ultrafast is perfectly viable even for near indistinguishable
quality as long as it has the appropriate CRF value. It's nice if you
want to record but would like to or need to reduce the impact of
encoding on the CPU. It will automatically compensate for the preset at
the cost of larger file size.
2.) It was suggested to just always use ultrafast, but ultrafast
requires 2-4x as much disk space for the same CRF (most likely due to
x264 compensating for the preset). Providing veryfast is important if
you really want to reduce file size and/or reduce blocking at lower
quality levels.
- When a recording preset is used, a secondary audio encoder is also
spawned at 192 bitrate to ensure high quality audio. I chose 192
because that's the limit of the media foundation aac encoder on
windows, which I want to make sure is used if available due to its
high performance.
- The CRF calculation is based upon resolution, quality, and whether
it's set to ultrafast. First, quality sets the base CRF, 23 for
"good" quality, 16 for "very high" quality. If set to ultrafast,
it'll subtract 2 points from the CRF value to help compensate. Lower
resolutions will also lower the CRF value to help improve higher
details with a smaller pixel ratio.
Selecting any supported FFmpeg format where ff_format_desc_extensions
returns NULL would crash the std::string constructor, so we pass an
empty extension instead (rtsp is one candidate format that triggers
the crash, on my machine at least)
CBR is now always on by default for streaming, so there's no reason to
have a setting for this in particular. Still available in advanced
output settings of course, but simple output mode really should be kept
as simple as possible.
This is mostly just to remove the unnecessary clutter from the output
sections. The reconnect settings are generally rarely modified by users
as it is.
I added stream delay options to advanced settings not just because I
feel it's an advanced option, but also to reduce clutter in the outputs
section and its sub-sections, which already have far too many options as
it is.
Replaces all the json/config loading/saving functions with safe
variants to reduce the chance of potential file corruption as much as
possible.
Also does a minor refactor of json writing by using
obs_data_save_json_safe for writing json files instead of manually using
obs_data_get_json and os_quick_write_utf8 each time.
This changes the way the advanced output section's FFmpeg output
settings work by allowing the user to select whether they want to output
to a file or output to a URL, and makes it so file names are
automatically generated like other recording outputs.
If they choose to output to a file, it'll only require an output
directory similarly to how other recording outputs work. They can
select a directory to output to rather than being required to type in a
full path and filename; the filename is automatically generated. The
extension is also automatically retrieved from libff depending on the
format selected.
Otherwise if they have Output to URL selected, it'll show a simple edit
box where they can type in the target URL.
When settings the base resolution, try to find the closest output
resolution to the old output resolution, and use that for its value. If
the aspect ratio is about the same, then don't modify the value. If the
aspect ratio is significantly different, then find the closest
approximation while keeping with the new aspect ratio.
This particular issue has been an annoyance for quite some time.
Adds setting profiles to the basic user interface. For each profile, a
subdirectory for the profile will be created in
[config_dir]/obs-studio/basic/profiles which will contain the settings
data for each profile.
Right now, information about global audio sources is stored in both
scene files and in the config. These must be separated; there's no need
to store them in both when they can just be stored in the scenes file.
Certain output formats don't support multiple tracks, so it's important
to warn the user if they select multiple tracks.
Also warn the user if they select no tracks.
On windows vista/7, you cannot really use display capture efficiently
without disabling aero, so this will add an option to settings to allow
it to be disabled and cause it to be disabled on startup.
Reduces required scrolling when lots of new audio sources are added
(e.g. aux devices being enabled in the same dialog) when the dialog
was opened with just a few audio sources being present. Unfortunately,
the "restart required" warning is pushed all the way to the bottom
even if the source list is empty
NoFocusFrameStyle was meant to disable focus frames around the scenes/sources
list (on OSX); unfortunately it also removed focus frames from controls that
should have focus frames like input boxes in the settings window
Non-NV12 video formats are primarily intended for recording. For
streaming, if the libobs color format is not set to NV12, it's likely
that the video frames will have to be converted to NV12, which will use
extra CPU usage. Due to that fact, it's important to warn the user of
that potential extra increased CPU usage that may be required when
streaming.
OBS will offer the user a list of themes which are .qss files inside
data/obs-studio/themes. If no theme is found in the configuration, it
loads the default theme for the system.
Currently, this allows the setting of values such as:
- Audio buffering time
- Color format (still somewhat unsupported)
- YUV color space (if a YUV format)
= YUV color range (if a YUV format)
More color formats will be added in the future, such as RGB and YUV
4:2:2 formats.
Add a checkbox named "Enforce streaming service encoder settings"
checkbox to advanced output. Disabling this checkbox allows the user to
optionally disable the enforcement of streaming service encoder
settings. I had a user complain that they didn't want to always have
the service's preferred encoder settings forced on them.
For the 'output resolution' setting in video settings, do not show
values in the list that it does not support (width must be aligned to a
128bit boundry, and height must be divisible by two)
For the 'rescale' settings in advanced outputs, the scales must all be
divisible by two.
The 'rescale' values in advanced output section are supposed to be based
upon the output resolution of the program. Meaning they should not be
used for scaling up, because the resolution downloaded from the graphics
processor is the output resolution set in video settings; thus any
resolution you set for the 'rescale' values is scaled from that.
To prevent from causing confusion/issues for our most awesome and
respected locale editors, only localize and translate the relevant text
rather than the extensions of the filter.
Before: After:
obs_service_gettype obs_service_get_type
It seems there was an API function that was missed when we were doing
our big API consistency update. Unsquishes obs_service_gettype to
obs_service_get_type.
Currently service settings are updated in real time via the properties
view, which means that OK/Cancel/Apply have no effect. This fixes that
by using the new type of properties view that operates only on settings
data, not an object.
Adds an 'advanced' mode to the output settings to allow more powerful
and complex streaming and recording options:
- Optionally use a different encoder for recording than for streaming to
allow the recording to use a different encoder or encoder settings if
desired (though at the cost if increased CPU usage depending on the
encoders being used)
- Use encoders other than x264
- Rescale the recording or streaming encoders in case the user wishes to
stream and record at different resolutions
- Select the specific mixer to use for recording and for streaming,
allowing the stream and recording to use separate mixers (to for
example allow a user to stream the game/mic audio but only record the
game audio)
- Use FFmpeg output for the recording button instead of only recording
h264/aac to FLV, allowing the user to output to various different
types of file formats or remote URLs, as well as allowing the user to
select and use different encoders and encoder settings that are
available in the FFmpeg library
- Optionally allow the use of multiple audio tracks in a single output
if the file formats or stream services support it
