Unlike get_properties, there is not reason to not call get_defaults if it is
given in addition to get_defaults2. Additonally this fixes the bug with
'init_encoder' which would only ever call get_defaults, resulting in broken
encoders if those used get_defaults2.
This implements pausing of outputs. To accomplish this, raw audio/video
data is halted to the encoders or raw output. Pausing is as precisely
timed as possible according to the timing of the obs_output_pause call,
and audio data will be spliced down to the exact audio sample in
accordance to that timing at the start/end marks.
Outputs that support this (outputs used for recording) can set the
OBS_OUTPUT_CAN_PAUSE capability flag.
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.
If the remove_connection call of obs_encoder_stop_internal took too
long, obs_encoder_destroy could get called before that function
completed, causing a race condition.
Allows the ability to encode by passing NV12 textures. This uses a
separate thread for texture-based encoders with a small queue of
textures. An output texture with a keyed mutex shared texture is locked
between OBS and each encoder. A new encoder callback and capability
flag is used to encode with textures.
This splits the "do_encode" function in to "do_encode" and
"send_off_encoder_packet", the latter of which allows the ability for
texture-based encoders to manage their own encoding and just simply send
off a packet to the outputs.
Allows the ability for one encoder to defer to another in case of
failure or unsupported feature. Okay, fine, it's mostly a hack so the
new NVENC encoder can fall back to the FFmpeg encoder if NV12 textures
aren't in use, that way it does not have to implement raw fallback
support itself. The settings and properties are pretty much the same,
so there's no reason not to utilize it in order to save time that could
otherwise be spent more productively.
Reduces GPU usage when encoding is not active. Does not perform color
conversion, frame staging, or frame downloading unless encoding is
explicitly active.
On audio encoder startup, audio encoders paired with a video encoder
would unintentionally discard a single audio data segment, causing it to
be 1024 audio frames out of sync.
(Note: This commit also modifies coreaudio-encoder, win-capture, and
win-mf modules)
This reduces logging to the user's log file. Most of the things
specified are not useful for examining log files, and make reading log
files more painful.
The things that are useful to log should be up to the front-end to
implement. The core and core plugins should have minimal mandatory
logging.
With the new audio subsystem, audio buffering is minimal at all times.
However, when the audio buffering is too small or non-existent, it would
cause the audio encoders to start with a timestamp that was actually
higher than the first video frame timestamp. Video would have some
inherent buffering/delay, but then audio could return and encode almost
immediately. This created a possible window of empty time between the
first encoded video packet and the first encoded audio packet, where as
audio buffering would cause the first audio packet's timestamp to always
be way before the first video packet's timestamp. It would then
incorrectly assume the two starting points were in sync.
So instead of assuming the audio data is always first, this patch makes
video wait for audio data comes in, and conversely buffers audio data
until video comes in, and tries to find a starting point within that
video data instead, ensuring a synced starting point whether audio
buffering is active or not.
Ensures that the packet dts_usec vals which are generated for
syncing/interleaving use the proper offset relative to where they're
supposed to be starting from. The negative DTS of a first video packet
could potentially have been applied twice due to this.
API changed from:
obs_source_info::get_name(void)
obs_output_info::get_name(void)
obs_encoder_info::get_name(void)
obs_service_info::get_name(void)
API changed to:
obs_source_info::get_name(void *type_data)
obs_output_info::get_name(void *type_data)
obs_encoder_info::get_name(void *type_data)
obs_service_info::get_name(void *type_data)
This allows the type data to be used when getting the name of the
object (useful for plugin wrappers primarily).
NOTE: Though a parameter was added, this is backward-compatible with
older plugins due to calling convention. The new parameter will simply
be ignored by older plugins, and the stack (if used) will be cleaned up
by the caller.
This prevents encoders (hardware encoders in particular) from being
continually active when all outputs disconnect from an encoder. This is
mostly just a temporary measure; the encoding interface may need a bit
of a redesign. It will also definitely needs to be able to flush at
some point. Currently when an output is stopped, the pending data is
discarded, which needs to be fixed.
Allows objects to be created regardless of whether the actual id exists
or not. This is a precaution that preserves objects/settings if for
some reason the id was removed for whatever reason (plugin removed, or
hardware encoder that disappeared). This was already added for sources,
but really needs to be added for other libobs objects as well: outputs,
encoders, services.
In case the encoder has to use a different sample rate (due to the
sample rate being unsupported), we need an API function for the encoder
to get the sample rate that the encoder is actually running at.
I realized that the get_video_info and get_audio_info encoder callbacks
always have to manually query the libobs audio/video information.
This fixes that problem by passing the libobs video/audio information in
the structures passed to those callbacks so they don't have to query it
each time, reducing needless boilerplate code for encoders.
Allows the ability to hint at encoders what format should be used.
This is particularly useful if libobs is currently operating in planar
4:4:4, but you want to force an encoder used for streaming to convert to
NV12 to prevent streaming issues.
When using multiple video encoders together with a single audio encoder,
the audio wouldn't be in sync.
The reason why this occurred is because the dts_usec variable of the
encoder packet (which is based on system time) would always be reset to
a value based upon the dts (which is not guaranteed to be based on
system time) in the apply_interleaved_packet_offset function. This
would then in turn cause it to miscalculate the starting audio/video
offsets, which are required to calculate sync.
So instead of calling that function unnecessarily, separate the check
for whether audio/video has been received in to a new function, and only
start applying the interleaved offsets after audio and video have
actually started up and the starting offsets have been calculated.
API changed:
--------------------------
void obs_output_set_audio_encoder(
obs_output_t *output,
obs_encoder_t *encoder);
obs_encoder_t *obs_output_get_audio_encoder(
const obs_output_t *output);
obs_encoder_t *obs_audio_encoder_create(
const char *id,
const char *name,
obs_data_t *settings);
Changed to:
--------------------------
/* 'idx' specifies the track index of the output */
void obs_output_set_audio_encoder(
obs_output_t *output,
obs_encoder_t *encoder,
size_t idx);
/* 'idx' specifies the track index of the output */
obs_encoder_t *obs_output_get_audio_encoder(
const obs_output_t *output,
size_t idx);
/* 'mixer_idx' specifies the mixer index to capture audio from */
obs_encoder_t *obs_audio_encoder_create(
const char *id,
const char *name,
obs_data_t *settings,
size_t mixer_idx);
Overview
--------------------------
This feature allows multiple audio mixers to be used at a time. This
capability was able to be added with surprisingly very little extra
overhead. Audio will not be mixed unless it's assigned to a specific
mixer, and mixers will not mix unless they have an active mix
connection.
Mostly this will be useful for being able to separate out specific audio
for recording versus streaming, but will also be useful for certain
streaming services that support multiple audio streams via RTMP.
I didn't want to use a variable amount of mixers due to the desire to
reduce heap allocations, so currently I set the limit to 4 simultaneous
mixers; this number can be increased later if needed, but honestly I
feel like it's just the right number to use.
Sources:
Sources can now specify which audio mixers their audio is mixed to; this
can be a single mixer or multiple mixers at a time. The
obs_source_set_audio_mixers function sets the audio mixer which an audio
source applies to. For example, 0xF would mean that the source applies
to all four mixers.
Audio Encoders:
Audio encoders now must specify which specific audio mixer they use when
they encode audio data.
Outputs:
Outputs that use encoders can now support multiple audio tracks at once
if they have the OBS_OUTPUT_MULTI_TRACK capability flag set. This is
mostly only useful for certain types of RTMP transmissions, though may
be useful for file formats that support multiple audio tracks as well
later on.
When the encoder is set to scale to a different resolution than the obs
output resolution, make sure it uses the current video colorspace and
range by default.
If an encoder did not possess any SEI data, it would never send data at
all because the sent_first_packet wasn't set despite the first packet
being sent.
