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.
Use the config button on volume controls to allow the ability for
filters/properties to be accessed via the mixer. Particularly useful
for the purpose of accessing filters/properties of global audio outputs
that are added via audio settings.
This simply saves/loads the actual list widget item order.
The reason why this is done is because internally, libobs doesn't have a
list of scenes, it only has a list of sources. The list of scenes is
actually something artificially implemented by the basic window user
interface.
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.
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.
The size parameter is the size of the elements, not the size of the
data. The size parameter should be 1, and the elements should be the
number of bytes.
The reason why I'm making this change is because the fread/fwrite would
fail when the parameters were swapped.
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)
In the (unlikely) event of multiple concurrent calls to
input_method_changed it was possible that the log messages would appear
out of order with respect to which layout would actually be active
after the last log message
jp9000
Palana
Night
Gol-D-Ace
John Bradley