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).
Some encoders such as the AMD AMF encoder don't set their bitrate
property, so it gets returned as 0. This causes the new network code to
allocate a tiny buffer, resulting in output starvation and throttling
to the point of uselessness if low latency mode is also enabled.
The ftl output gets very spammy (3-6 lines every 5 seconds),
this commit changes the loglevel of those messages to debug
so they do not clutter up the logs of users.
In case these messages are needed they can be re-enabled using
the '--verbose' flag.
(also fixes minor formatting issues)
It's erroneously including an entire directory, including a file that no
longer exists, and the obs-outputs_HEADERS variable had a typo when used
in the add_library call.
x264 on ultrafast won't drop any b-frames, so it might buffer packets a
bit too much in that case.
The frame drop duration calculation now searches for the first valid
video frame within the packets buffer. Additionally, it no longer
prevents frame dropping for a specific time after dropping frames,
because the first case eliminates the need for that.
Clears/reinitializes RTMP structure to defaults, preventing data from
potentially persisting between different connections.
Closesjp9000/obs-studio#878
Jim
Kurt Kartaltepe
Shaolin
Richard Stanway
Gol-D-Ace
Michael Goulet
Quinn Damerell
pkviet
Matt Morrissette
Quinn Damerell
derrod
Ryan Foster
SoraYuki