For some reason librtmp treats a close message as a request to reconnect.
It does this syncronously, and uses the same event processing so that
another close request will continue the cycle until the stack is exhausted
and the application crashes.
Fixes https://obsproject.com/mantis/view.php?id=634
Twitch no longer recommend Main profile on their "Broadcast requirements"
page. The number of devices that can't decode High profile is very small
and limited to old, legacy devices such as the 1st generation iPad (six
years old). Additionally, these devices have limited resolution / FPS
support, so streams are often encoded at levels beyond what they could
decode even disregarding the H264 profile requirement. It's not worth
degrading the quality of streams for modern devices just to support these
legacy decoders. Twitch still provides low-complexity transcodes for
devices which cannot handle Source quality streams.
When a pulseaudio source is created from existing settings and the
device is not found, then the private data will return NULL, making it
impossible to change the source to use a different device. NULL should
never be returned when possible, otherwise settings cannot be changed
and properties cannot be displayed.
Closesjp9000/obs-studio#604
The maximum shutdown timeout value was added as a setting, but never
actually fully implemented. This implements it properly, and sets its
default timeout value to 30 seconds.
This commit fixes the issue outlined in the following thread:
https://obsproject.com/forum/threads/50045/
When the image source file was replaced by an outside process, it would
only reload when the file's new timestamp was newer than the file's
previous timestamp. This fixes that behavior. Now an image source
will reload any time the file's new timestamp is different than its
previous timestamp.
libvlc_media_new_path implies a file. To get media based upon URLs, use
libvlc_media_new_location. Additionally, if using a URL, it's best to
give it some playback caching/buffering, at least 100 milliseconds.
Closesjp9000/obs-studio#590
Previously, for an unknown reason p-frames were marked as highest
priority along with i-frames (keyframes), which means they could not be
dropped. This would cause a problem where if for whatever reason
there's too much congestion, data would continually buffer. This fixes
the issue by dropping p-frames at a separate (higher) threshold than
b-frames.
Measures packet data rate and sleeps to prevent data from going above
the set data rate. Uncomment the TEST_FRAMEDROPS macro, then set
DROPTEST_MAX_KBPS to the desired kb/s (for example 3000 for 3000
kilobits per second), and then it will limit the data rate to that
specified amount, forcing the thread to sleep to ensure it can only
output the desired data rate.
Causes issues with certain servers that don't parse it properly and only
expect a specific string.
Use the RTMP onMetaData to get the OBS version instead when possible.
Even if the hotkey is not enabled, always allow configuration of the
hotkey. Fixes a bug where the hotkey configuration settings would not
save if the settings were changed.
Annoyingly this means that the hotkey will still be shown to the user,
possibly confusing the user as to whether they can use it, but for the
time being it's better than having their hotkey configuration removed
each time they change the mode.
It's supposed to look for patch segments in ascending order, from the
smallest offset to the largest offset. Patch type/comparison is
identical to the one it's being swapped with, so only the offsets need
to be swapped.
(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.
Prevents the common problem of injecting in to certain processes and
getting the hook DLL "stuck":
- windows explorer
- steam
- battle.net
- gog galaxy
- skype
- uplay
- origin
- microsoft visual studio
- task manager
- league of legends lobby window
- windows 10 system settings window
Changed the first property of game capture to be a "mode" list (with
"any fullscreen window", "specific window", and "hotkey").
When hotkey mode is set, it'll add a hotkey pair to hotkey settings to
activate/deactivate game capture. When the hotkey to activate is
pressed, it'll treat the current foreground window as the target window
similar to "selected window" mode; it'll keep trying to capture the same
window even if the window or its application closes/reopens, and will
continue to do so until deactivated via the deactivate hotkey, or until
a new window is set via the activate hotkey.
Doing something like '#if TRUE' when 'TRUE' isn't defined is the
equivalent to '#if 0' because the preprocessor evaluates an undefined
macro as 0/false.
These comments have been added to clean up the code and make it more
clear of what the code is doing. The code felt a bit messy, and this
should help prevent the original author of the noise suppression filter
from being lost in case he decides to modify/improve the filter.
When buffering audio data, we don't want to buffer audio data that may
be old. If the audio timing jumps significant and old audio data is
buffered, clear that old data.
The noise suppression filter mistakenly operated on the assumption that
input audio data would always be in 10ms segments, and would crash if
audio data was larger than that size.
Because speexdsp operates on fixed audio frame sizes only, we must
buffer audio data to fit that frame processing size. This creates a
troublesome situation where you must buffer around that specified frame
size.
The new steps for processing are:
1. Push audio data to input circular buffer.
2. Push number of audio frames and timestamp for that audio packet to an
'info' circular buffer.
3. Check size of input circular buffer, and while it's equal to or above
the speexdsp frame size (10ms for minimum latency), pop from the
input buffer to a temporary buffer (10ms frames) and process it, then
push that temporary buffer to the output circular buffer.
4. Peek at the front of the 'info' circular buffer.
5. If the output circular buffer frame size is equal or larger than next
expected number of frames, pop both the info and output buffer, and
return the audio data with the expected audio frames/timestamp.
libVLC doesn't seem to provide full speaker configuration info, so when
downmixing audio from a file that had more than 2 channels, the audio
would sound wrong.
This change makes it so that libVLC does the downmixing to stereo rather
than libobs, just due to that lack of speaker configuration info.
It seems that it is not possible to create multiple OpenGL textures
for the same window. When sources are duplicated in studio mode this
causes window capture sources to become black. The
OBS_SOURCE_DO_NOT_DUPLICATE flag prevents that this happens.
Update Restream.io ingests list and set recommended keyframe interval to
2.
(Commit edited and formatted by Jim: Fixed commit message)
Closesjp9000/obs-studio#565
Michael Fabian Dirks
Jim
Richard Stanway
CoDEmanX
Gol-D-Ace
Robin Hielscher
Christoph Hohmann
Michael Hoang
hwdro
Andrei Nistor
derrod
Ryan Foster
rpslack
Radzaquiel
Cephas Reis
Andreas Reischuck
Jamy Timmermans
John Bradley
Will Jamieson
pipll
Simon
Andrew Surzhynskyi