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
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.
(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.
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
There are two variables used for 'priority', the general marked frame
priority, and the drop priority. They can sometimes be different
because some encoders don't always use the correct frame priority.
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.
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.
(Note: This commit also modifies obs-ffmpeg and obs-outputs)
API Changed:
obs_output_info::void (*stop)(void *data);
To:
obs_output_info::void (*stop)(void *data, uint64_t ts);
This fixes the long-time design flaw where obs_output_stop and the
output 'stop' callback would just shut down the output without
considering the timing of when obs_output_stop was used, discarding any
possible buffering and causing the output to get cut off at an
unexpected timing.
The 'stop' callback of obs_output_info now takes a timestamp with the
expectation that the output will use that timestamp to stop output data
in accordance to that timing. obs_output_stop now records the timestamp
at the time that the function is called and calls the 'stop' callback
with that timestamp. If needed, obs_output_force_stop will still stop
the output immediately without buffering.
Because output stop timing has been fixed, there is no need to send the
remaining packets in the queue because it now just waits for the stop
timing anyway.
There was no error checking when sending headers/metadata, so what would
happen is that if a header/metadata send failed (meaning the socket was
disconnected), it would continue to act as if it was still connected,
and it would block and lock up on the next send/recv call.
Not calling recv when data is received will accumulate data in the
internal recveive buffer until it's full, in which case is will stop
acknowledging. This can lead to unjustified disconnections.
If there was an attempt to destroy the rtmp-stream output while it was
already connecting and stopping at the same time, it would try to join
with the stop thread rather than with the connect thread. The connect
thread would then continue past destruction.
