Adds Microsoft Media Foundation AAC Encoder that supports
96k to 192k bitrates. This plugin is only enabled on Microsoft
Windows 8+ due to performance issues found on Windows 7.
Authentication code has been updated as per the changes to support
multiple streams.
Authentication is now also enabled by default, and should be a no-op
if the server does not request authentication or username and password
details are not provided.
I've come to realize that it's probably not wise to deviate from the
original version's functionality due to the fact that the original
version works without issues. I'm wondering if some of the capture
problems have been due to the fact that the direct hook method (via
CreateRemoteThread) was removed, so I put it back in, made it default,
and added an option to use anti-cheat compatibility just like in the
original version.
This particularly affected audio encoding, audio encoding previously
would count samples and use it to create an encoding timestamp, but
because I was using a standard integer (which is 32bit by default on
x86), it would max out at about 0x7FFFFFFF samples, which is about 12
hours of samples at 48000 sample rate. After that, it would start going
into negative territory (overflowing). By changing it to int64_t, it
will make it so that audio at 48000 samples per second would only be
able to overflow after about.. 6.09 million years. In other words,
this should fix the issue for good.
Livecoding.tv (coding), gaminglive.tv (gaming), and beam.pro
(gaming/music)
I really don't see any problems with adding these particular services to
the local list while the actual remote ingest lookup code has yet to be
even started yet (as of this writing). They seem to be harmless
services that are dedicated to specific types of content (stated above).
When hooking 64bit functions, sometimes the offset between the function
being hooked and the hook itself can be large enough to where it
requires a 64bit offset to be used. However, because a 64bit jump
requires overwriting so many code instructions in the function, it can
sometimes overwrite code in to an adjacent function, thereby causing a
crash.
The 64bit hook bounce (created by R1CH) is designed to prevent using
very long jumps in the target by creating executable memory within a
32bit offset of that target, and then writing it with the 64bit long
jump instruction instead. Then in the target function, it will jump to
that memory instead, thus forcing the actual hooked function to use a
32bit hook instead of a 64bit hook, and using at most 5 bytes for the
actual hook, preventing any likelihood of it overwriting an adjacent
function.
System timestamps were being used instead of timestamps from the
audio/video input. This would cause potential desync as well as
incremental buffering when using devices with the blackmagic video
source. Using the timestamps direct from the SDK itself fixes those
issues, and causes audio/video to play back properly and in sync.
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.
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.
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)
This allows the ability to output the audio of the device as desktop
audio (via the WaveOut or DirectSound audio renderers) instead of
capturing the audio only.
In the future, we'll implement audio monitoring which will make this
feature obsolete, but for the time being I decided to add this option as
a temporary measure to allow users to play the audio from their devices
via the DirectShow output.
Found via UBSan, actual (sample) error:
"plugins/text-freetype2/text-functionality.c:284:26: runtime error: left
shift of 194 by 24 places cannot be represented in type 'int'"
