mbedtls_x509_crt_parse_path returns a positive number if it partially
succeeds and a negative number on complete failure. This changes the
positive result to no longer error and prevent TLS connections (OBS
verifies all endpoints so having no CA chain prevents TLS).
mbedtls_md5_starts(), mbedtls_md5_update() and mbedtls_md5_finish()
have been marked deprecated since mbedtls version 2.7 and may go
away in the future.
These function have been superseded by versions with a return
value. They are suffixed with "_ret". We do not actually check
return values.
Also the header "mbedtls/net.h" has been superseded by
"mbedtls/net_sockets.h".
The dynamic bitrate operates based upon estimating the current bitrate
output, and then adjusting the bitrate on the fly as necessary when
congestion is detected as a replacement for dropping frames.
This may still need adjustment, as it is difficult to accurately emulate
real-world frame drop scenarios. This does not currently drop frames at
all, and because of that, very high congestion may cause additional
stream delay to viewers (because data will be buffered), but from
limited testing, most congestion will not cause that and it can safely
recover pretty quickly without adding significant delay.
When doing the bitrate limit test, it can be useful to have the ability
to change the current maximum bitrate limit. This adds the ability to
press keys on windows (numpad 0-6) to change between bitrates. Numpad 0
being no limit, 1 being 1000, 2 being 2000, etc.
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.
(This commit also modifies the UI, obs-ffmpeg, and obs-output modules)
Fixes a long-time regression where the program would lock up if an
encode call fails. Shuts down all outputs associated with the failing
encoder and displays an error message to the user.
Ideally, it would be best if a more detailed error could be displayed to
the user about the nature of the error, though the primary problem is
the encoder errors are typically not something the user would be able to
understand. The current message is a bit of a generic error message;
improvement is welcome.
Another suggestion is to try to have the encoder restart seamlessly,
though it would take a significant amount of work to be able to make it
do something like that properly, and it sort of assumes that encoder
failures are sporadic, which may not necessarily be the case with some
hardware encoders on some systems. It may be better just to use another
encoder in that case. For now, seamless restart is ruled out.
Some security layer libraries code path used by the rtmp output had a
not used variable and not used param on HMAC_finish macro that was
triggering warnings during compilation.
On 64bit systems, this check will always evaluate to false due to
SIZE_MAX type and triggers a compiler warning.
This both makes it clearer that its only needed on 32bit system and
clear the compiler warning.
This diff adds mbedTLS support to the obs-outputs plugin. PolarSSL and
mbedTLS have grown so different between 2015-or-so when libRTMP was
written, and now it's no longer feasible to just use the USE_POLARSSL
flag.
This commit adds a WITH_RTMPS tri-state CMake variable (auto/on/off),
set to "Auto" by default. "Auto" will use RTMPS if mbedTLS is found,
otherwise will disable RTMPS. "On" will make it require mbedTLS,
otherwise fails configuration, and "Off" disables RTMPS support
altogether.
Closesobsproject/obs-studio#1360
(also obs, deps/media-playback, libobs/audio-monitoring, decklink,
linux-alsa, linux-pulseaudio, mac-capture, obs-ffmpeg, win-dshow,
win-wasapi)
Default channel layout for 4 channels is 4.0 in FFmpeg.
Replacing quad with 4.0 will improve compatibility since FFmpeg has
better support of its default channel layouts.
(also modifies obs-ffmpeg, audio-monitoring, win-wasapi, decklink,
obs-outputs)
Removes speaker layouts which are not exposed in UI. The speaker
layouts selectable by users in the UI are the most common ones. It is
not necessary to keep other layouts. (This basically removes
5POINT1_SURROUND, 7POINT1_SURROUND, SURROUND =3.0).
(This commit also modifies the following modules: UI,
deps/media-playback, coreaudio-encoder, decklink, linux-alsa,
linux-pulseaudio, mac-capture, obs-ffmpeg, obs-filters, obs-libfdk,
obs-outputs, win-dshow, and win-wasapi)
Adds surround sound audio support to the core, core plugins, and user
interface.
Compatible streaming services: Twitch, FB 360 live
Compatible protocols: rtmp / mpeg-ts tcp udp
Compatible file formats: mkv mp4 ts (others untested)
Compatible codecs: ffmpeg aac, fdk_aac, CoreAudio aac,
opus, vorbis, pcm (others untested).
Tested streaming servers: wowza, nginx
HLS, mpeg-dash : surround passthrough
Html5 players tested with live surround:
videojs, mediaelement, viblast (hls+dash), hls.js
Decklink: on win32, swap channels order for 5.1 7.1
(due to different channel mapping on wav, mpeg, ffmpeg)
Audio filters: surround working.
Monitoring: surround working (win macOs linux (pulse-audio)).
VST: stereo plugins keep in general only the first two channels.
surround plugins should work (e.g. mcfx does).
OS: win, macOs, linux (alsa, pulse-audio).
Misc: larger audio bitrates unlocked to accommodate more channels
NB: mf-aac only supports mono and stereo + 5.1 on win 10
(not implemented due to lack of usefulness)
Closesjp9000/obs-studio#968
The new code in 3032535f56 would signal that the output has stopped to
the back-end and front-end, but the event used in the outputs themselves
to shut down the send thread would still be signaled, causing the next
connection to immediately stop as soon as it had started. This fixes it
so that the event does not get signaled unless the thread is active.
The end of an FLV tag would contain the size of the tag, but the code
was erroneously including the end size value in addition, which it's not
supposed to do normally.
(This commit also modifies the obs-outputs module)
The first video packet video offset (the value used to set the starting
point of video data) would be set to the DTS value of the first video
packet. However, when b-frames are used, the first DTS value will be
negative. This was originally done because FLV muxing requires that the
first packet's DTS start from 0. Unfortunately, this would also
effectively cause the first packet's PTS/DTS value to be shifted forward
by the negative amount, which would cause video sync to be off by a
video frame or two.
This fixes it to start at the PTS value instead and preserve any
negative offsets. Additionally, the FLV muxing code has been fixed to
ensure that it adjusts the starting video DTS to 0, and now correctly
adjusts the first audio packet's timestamp according to that DTS as well
(which it didn't do before).