Removes flag guard to enable HDR streaming for HEVC over Enhanced RTMP.
This functionality is currently only supported by YouTube.
See github.com/veovera/enhanced-rtmp for the Enhanced RTMP spec.
Effectively reverting parts of d314d47, this commit removes the new
functions that got added to remove the flags parameter. Instead, it just
marks the parameter as unused and documents this. Having what is
effectively an API break just to remove a parameter is a bit overkill.
The other parts of d314d47 which cleaned up the usage of the flags
parameter are untouched here.
Meta have filed a patent application on the Go Away feature. This goes
against the spirit of free software and puts the use of this feature in
questionable legal status, so let's remove this code until the patent
situation is resolved.
This reverts commits a593fe6755 and
dada82fec1.
This deprecates the following functions, replacing them with new
versions:
- `obs_output_can_begin_data_capture()` - now `*capture2()`
- `obs_output_initialize_encoders()` - now `*encoders2()`
- `obs_output_begin_data_capture()` - now `*capture2()`
The flags parameter was initially designed to support audio-only or
video-only operation of an output which had the `OBS_OUTPUT_AV` flag,
however, full support for that was never implemented, and there are
likely fundamental issues with an implementation, mainly that most
outputs are programmed assuming that there will always be at least one
audio and one video track. This requires new flags specifying support
for optional audio/video, among other things.
An implementation to allow audio/video to be optional is best done
using the flag technique above, with audio/video enablement specified
by whether media (raw, `video_t/audio_t`) or encoder (`obs_encoder_t`)
objects are specified.
Since every implementation I could find always specifies `flags` as 0,
I was able to safely conclude that immediately removing the parameter's
functionality is safe to do.
Sequence start and end packets never have a valid DTS, but with a
non-zero offset the timestamp in the packet would end up also being
non-zero, which leads to librtmp attempting to calculate a timestamp
delta and underflowing.
As a fix, simply remove the ability to even set a DTS offset for
start/end packets and always keep it at zero.
This logs that dynamic bitrate is disabled when a codec which does not
support bitrate reconfiguration is used, such as aom, svt-av1 ...
Signed-off-by: pkv <pkv@obsproject.com>
"New Socket Loop" and "Low Latency Mode" RTMP options are only available
on Windows.
Those options should be ignored and forced-disabled on non-Windows
builds.
Now uses GetIfEntry2 which supports 64-bit values for reporting speed, so
10+ gbps adapters are now reported correctly in the log. Also added an
additional log line if the interface error counters are non-zero to possibly
help identify physical faults. Finally the transmit and receive speeds are
logged independently so that asynchronous mediums such as Wi-Fi that might
have good RX but poor TX can be better diagnosed.
The dynamic increment timeout does not need to be 30 seconds. Change it
to 4 seconds instead to make dynamic bitrate something that people
actually want to use.
This commit fixes what is arguably a long-winded series of previous
commits that have possibly caused just as many problems as they have
fixed. I'll spare the details, but basically, there's no reason that
any of the RTMP object should ever be used across socket sessions.
This provides a slight enhancement by removing the `RTMP_Init` call
in `rtmp_stream_create()`, since it effectively just initializes TLS
just for `try_connect` to deinitialize it before it is even used.
This also fixes the current `SO_RCVTIMEO` timeout functionality by
making sure that `RTMP_Reset` is called last.
A race condition can occur in obs-outputs where the send_thread is in
the process of (re)connecting but the encoder was shut down in the
meantime. This causes the expected header data to be garbage, resulting
in a crash.
The silent reconnect (GOAWAY) is supposed to be used for switching the
connection to a different server. As such, cleanly shutting down the
RTMP connection can destroy state that needs to be preserved in order
for the GOAWAY-enabled server to properly resume the same stream on
reconnect.
This commit closes the TLS/TCP connection before calling RTMP_Close,
causing librtmp to skip the FCUnpublish and deleteStream messages.
Sometimes when reconnecting the internal RTMP data is not cleared
(particularly the TLS data). This can cause TLS data to carry over from
one connection to another, causing issues with the secondary connection.
From Windows 7 onwards, dynamic send buffering is enabled. By setting
SO_SNDBUF explicitly, we actually disabled the dynamic send buffering
feature which results in reduced throughput. Thankfully this did not
affect the majority of users since the default send buffer is usually
already 64k.
This commit replaces the setting of SO_SNDBUF with log output showing
the current value of SO_SNDBUF at stream start and end. This will aid in
debugging throughput issues caused by a buffer that isn't big enough,
perhaps as a result of the user disabling dynamic send buffering
system-wide.
Since this is activated after starting the output, it assumes there is
no need to read anything from the connection as RTMP is send-only from
that point on. However with TLS, reading protocol messages is required
for proper operation, causing it to immediately fail on RTMPS
connections. This is a complex fix due to the way it interacts with
librtmp and mbedTLS so let's just disable it for now to avoid breaking
things for users.
Bug is caused by the internal connection variables not being reset on
reconnect, leading OBS to both be unable to parse valid packets from and
send valid packets to the remote end. This commit splits RTMP_Init off
into a new RTMP_Reset function, which resets these internal variables
without re-initing the rest of the library. The original RTMP_Init
calls the new function, perfectly preserving the old behaviour while
adding a new reset function to address the issue with.
Fixesobsproject/obs-studio#2865
RTMP authentication requires the curStreamIdx and nStreams be set
between the disconnect / reconnect that occurs in PublisherAuth. Now
that there's no good place to clean them up, zero the whole rtmp->Link
before starting a new stream.
Every addStream call would increment this counter. After merging the
mbedTLS fixes, we no longer have extraneous RTMP_Init calls which were
masking the prescence of this bug. This caused every stream after the
first stream to have the wrong channel index, and eventually OBS would
crash due to an out of bounds write if the counter reached
RTMP_MAX_STREAMS.
Though this should now be very rare, it's more helpful than "Failed to
connect to server". Other TLS error codes are now also stored for future
use instead of copying them on a case by case basis.
Per mbedtls documentation, "If you share a context between threads, you
need to call these functions only from the main thread, at the beginning
and end of the context's lifetime.". OBS violated this since librtmp
uses a global context and it was allocated and freed in different
threads such as the auto config test.
This commit attaches the mbedtls context to an RTMP structure so there
is no more global state. It also fixes a rare double-free crash that
could occur if RTMP_TLS_Free was called twice (this happened in rare
situations such as the auto config running followed by a mode change
from Advanced to Simple).
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.
