This fixes an design flaw where a delayed output would schedule a
stop even while in the process of reconnecting instead of just shutting
down right away.
When obs_output_actual_stop is called on shutdown, it should wait for
the output to fully stop before doing anything, and then it should wait
for the data capture to end. The service should not be removed until
after the output has stopped, otherwise it could result in a possible
memory leak on stop. Packets should be freed last.
(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.
On outputs that use already-active video/audio encoder, the audio
pruning to sync up audio packets with video packets doesn't always get
called (for example if the video pruning function was called). Always
prune excess starting audio packets.
If audio buffering is very high, the audio packets built up in the
interleaved buffer would be significantly before the first video packet,
causing the offset between the starting video/audio packet pairs to be
significantly off, leading to desync.
This issue was not spotted until recently because it only happens when
streaming/recording with same encoders while audio buffering is very
high.
When starting a multi-track output, attempt to pair the video encoder
with one of the audio encoders to ensure that the video and audio
encoders start as close together in time as possible. This ensures the
best possible audio/video syncing point when using multi-track audio
output.
When using multi-track audio, encoders cannot be paired like they can
when only using a single audio track with video, so it has to choose the
best point in the interleaved buffer as the "starting point", and if the
encoders start up at different times, it has to prune that data and wait
to start the output on the next video keyframe. When the audio encoders
started up, there was the case where the encoders would take some time
to load, and it would cause the pruning code to wait for the next
keyframe to ensure startup syncing.
Starting the audio encoders before starting the video encoder should
reduce the possibility of that happening in a multi-track scenario.
In a multi-track scenario it was not taking in to consideration the
possibility of secondary audio tracks, which could have caused desync on
some of the audio tracks.
Problem:
When an output is started with encoders that have already been started
by another output, and it starts in between the window in between where
the first audio packets start coming in and where the first video packet
comes in, the output would get audio packets with timestamps potentially
much later than the first video frame when the first video frame finally
comes in. The audio/video encoders will almost always have a differing
delay.
Solution:
Detect that starting window, and if within that starting window, wait
for a new keyframe from video instead of trying to sync up video.
Additional Notes:
In these cases where an output starts with already-active encoders, this
patch also reduces the potential sync offset between the first video
frame and the first audio frame. Before, it would sync the first video
frame with the first audio frames right after that, but now it syncs
with the closest audio frame in the interleaved array, which halves the
potential sync difference between the first video frame and the first
audio frame of already-active encoders. (So now the potential sync
difference is roughly 11.6 milliseconds at 44.1khz audio, where it was
23.2 before)
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.
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.
This fixes the issue when an output cancels reconnecting, reconnect is
left at true, causing obs_output_active to always return true even
though reconnecting has actually been canceled.
This feature allows a user to delay an output (as long as the output
itself supports it). Needless to say this intended for live streams,
where users may want to delay their streams to prevent stream sniping,
cheating, and other such things.
The design this time was a bit more elaborate, but still simple in
design: the user can now schedule stops/starts without having to wait
for the stream itself to stop before being able to take any action.
Optionally, they can also forcibly stop stream (and delay) in case
something happens which they might not want to be streamed.
Additionally, a new option was added to preserve stream cutoff point on
disconnections/reconnections, so that if you get disconnected while
streaming, when it reconnects, it will reconnect right at the point
where it left off. This will probably be quite useful for a number of
applications in addition to regular delay, such as setting the delay to
1 second and then using this feature to minimize, for example, a
critical stream such as a tournament stream from getting any of its
stream data cut off. However, using this feature will of course cause
the stream data to buffer and increase delay (and memory usage) while
it's in the process of reconnecting.
Implements exponential backoff for consecutive reconnects, which is
useful to prevent too many connections from trying to reconnect back to
a service at once over a short period of time in the case of potential
service downtime. Exponential backoff causes each subsequent reconnect
attempt to double its timeout duration.
This optionally allows the front-end to know what the current timeout
value in seconds is set to for the reconnection without having to call
an extra API function to find that out.
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.
In certain circumstances where the output was stopping, and where data
took a long enough time to send (such as when using an encoding preset
that causes high CPU usage), the output would sometimes still send data
even after it was stopped, typically causing the output to crash.
Apparently the audio isn't guaranteed to start up past the first video
frame, so it would trigger that assert (which I'm glad I put in). I
didn't originally have this happen when I was testing because my audio
buffering was not at the default value and didn't trigger it to occur.
A blunder on my part, and once again a fine example of how you should
never make assumptions about possible code path.
This bug would happen if audio packets started being received before
video packets. It would erroneously cause audio packets to be
completely thrown away, and in certain cases would cause audio and video
to start way out of sync.
My original intention was "don't accept audio until video has started",
but instead mistakenly had the effect of "don't start audio until a
video packet has been received". This was originally was intended as a
way to handle outputs hooking in to active encoders and compensating
their existing timestamp information.
However, this made me realize that there was a major flaw in the design
for handling this, so I basically rewrote the entire thing.
Now, it does the following steps when inserting packets:
- Insert packets in to the interleaved packet array
- When both audio/video packets are received, prune packets up until the
point in which both audio/video start at the same time
- Resort the interleaved packet array
I have tested this code extensively and it appears to be working well,
regardless of whether or not the encoders were already active with
another output.
Apparently I unintentionally typed received_video = false twice instead
of one for video and one for audio.
This fixes a bug where audio would not start up again on an output that
had recently started and then stopped.
When the output sets a new audio/video encoder, it was not properly
removing itself from the previous audio/video encoders it was associated
with. It was erroneously removing itself from the encoder parameter
instead.
This Fixes a minor flaw with the API where data had to always be mutable
to be usable by the API.
Functions that do not modify the fundamental underlying data of a
structure should be marked as constant, both for safety and to signify
that the parameter is input only and will not be modified by the
function using it.
Typedef pointers are unsafe. If you do:
typedef struct bla *bla_t;
then you cannot use it as a constant, such as: const bla_t, because
that constant will be to the pointer itself rather than to the
underlying data. I admit this was a fundamental mistake that must
be corrected.
All typedefs that were pointer types will now have their pointers
removed from the type itself, and the pointers will be used when they
are actually used as variables/parameters/returns instead.
This does not break ABI though, which is pretty nice.
This makes it easier to do two things:
1.) Get the skipped frames count relative to each specific output
2.) Make it so that getting the 'current' log will always contain
information about skipped frames. Before, you'd have to force the
user to restart the program and get the last log, which was really
annoying when you just wanted to see how the encoders were
performing.
API functions added:
-----------------------------------------------
obs_output_set_preferred_size
obs_output_get_width
obs_output_get_height
obs_encoder_set_scaled_size
obs_encoder_get_width
obs_encoder_get_height
These functions allow for easier means of setting a custom resolution on
an output or encoder.
If an output uses an encoder and you set the preferred width/height
using the output, then the output will attempt to set the scaled
width/height for the encoder it's currently using.
Outputs and encoders now should use these functions to determine the
width/height of the raw frame data instead of using the video-io
functions.
Instead of having functions like obs_signal_handler() that can fail to
properly specify their actual intent in the name (does it signal a
handler, or does it return a signal handler?), always prefix functions
that are meant to get information with 'get' to make its functionality
more explicit.
Previous names: New names:
-----------------------------------------------------------
obs_audio obs_get_audio
obs_video obs_get_video
obs_signalhandler obs_get_signal_handler
obs_prochandler obs_get_proc_handler
obs_source_signalhandler obs_source_get_signal_handler
obs_source_prochandler obs_source_get_proc_handler
obs_output_signalhandler obs_output_get_signal_handler
obs_output_prochandler obs_output_get_proc_handler
obs_service_signalhandler obs_service_get_signal_handler
obs_service_prochandler obs_service_get_proc_handler
With the recent change to module handling by BtbN, I felt that having
this information might be useful in case someone is actually using make
install to set up their libraries.
Total bytes, total frames, and frames dropped. Total frames is
generated automatically, but total bytes and total dropped frames are
returned via callbacks.
Before it would assign the encoder/media callbacks directly to the
output's callbacks, so instead of doing that, it now goes through
intermediary functions for the sake of counting the frames.
Usually if you are reconnecting after network outage, it will give a
different code (such as OBS_OUTPUT_CONNECT_FAILED). So, if already
reconnecting, ignore the code unless it's OBS_OUTPUT_SUCCESS.
The core itself now provides reconnection options (enabled by default, 2
second timeout between reconnects, 20 retries max until actual
disconnection occurs). This will make things easier for both module
developers and UI developers.
Reconnecting treats the stream as though it were still active, and
signals are sent when reconnecting and upon successful reconnection.
Need to implement user interface information for reconnections.
The locale parameter was a mistake, because it puts extra needless
burden upon the module developer to have to handle this variable for
each and every single callback function. The parameter is being removed
in favor of a single centralized module callback function that
specifically updates locale information for a module only when needed.
The 'initialize' callback is used before the encoders/output start up so
it can adjust encoder settings to required values if needed.
Also added the function 'obs_encoder_active' that returns true or false
depending on whether that encoder is active or not.
The 'wait' constant was a terrible means of trying to ensure that the
packets were interleaved. Instead, calculate the current highest
timestamps of each encoder that's present in the interleaved buffer, and
use that as a means of detecting whether the current packet should be
sent off. This will guarantee sorting without relying on some arbirary
constant that 'assumes' that it'll be interleaved. It also reduces
buffering any more than what is needed to interleave.
- Updated the services API so that it links up with an output and
the output gets data from that service rather than via settings.
This allows the service context to have control over how an output is
used, and makes it so that the URL/key/etc isn't necessarily some
static setting.
Also, if the service is attached to an output, it will stick around
until the output is destroyed.
- The settings interface has been updated so that it can allow the
usage of service plugins. What this means is that now you can create
a service plugin that can control aspects of the stream, and it
allows each service to create their own user interface if they create
a service plugin module.
- Testing out saving of current service information. Saves/loads from
JSON in to obs_data_t, seems to be working quite nicely, and the
service object information is saved/preserved on exit, and loaded
again on startup.
- I agonized over the settings user interface for days, and eventually
I just decided that the only way that users weren't going to be
fumbling over options was to split up the settings in to simple/basic
output, pre-configured, and then advanced for advanced use (such as
multiple outputs or services, which I'll implement later).
This was particularly painful to really design right, I wanted more
features and wanted to include everything in one interface but
ultimately just realized from experience that users are just not
technically knowledgable about it and will end up fumbling with the
settings rather than getting things done.
Basically, what this means is that casual users only have to enter in
about 3 things to configure their stream: Stream key, audio bitrate,
and video bitrate. I am really happy with this interface for those
types of users, but it definitely won't be sufficient for advanced
usage or for custom outputs, so that stuff will have to be separated.
- Improved the JSON usage for the 'common streaming services' context,
I realized that JSON arrays are there to ensure sorting, while
forgetting that general items are optimized for hashing. So
basically I'm just using arrays now to sort items in it.
Add API for streaming services. The services API simplifies the
creation of custom service features and user interface.
Custom streaming services later on will be able to do things such as:
- Be able to use service-specific APIs via modules, allowing a more
direct means of communicating with the service and requesting or
setting service-specific information
- Get URL/stream key via other means of authentication such as OAuth,
or be able to build custom URLs for services that require that sort
of thing.
- Query information (such as viewer count, chat, follower
notifications, and other information)
- Set channel information (such as current game, current channel title,
activating commercials)
Also, I reduce some repeated code that was used for all libobs objects.
This includes the name of the object, the private data, settings, as
well as the signal and procedure handlers.
I also switched to using linked lists for the global object lists,
rather than using an array of pointers (you could say it was..
pointless.) ..Anyway, the linked list info is also stored in the shared
context data structure.
- Fix an issue that could occur when using more than one video encoder.
Audio/video would not sync up correctly because they were expected to
be paired with a particular encoder. This simply adds a little
helper variable to encoder packets that specifies the system time in
microseconds. We then use that system time to sync
- Fix an issue with x264 with fractional FPS rates (29.97 and 59.94
particularly) where it would create ridiculously large stream
outputs. The problem was that you shouldn't set the timebase_*
variables in the x264 params manually, let x264 handle the default
values for it and leave them at 0.
- Make x264 use CFR output, because there's no reason to ever use VFR
in this case.
- Implement the RTMP output module. This time around, we just use a
simple FLV muxer, then just write to the stream with RTMP_Write.
Easy and effective.
- Fix the FLV muxer, the muxer now outputs proper FLV packets.
- Output API:
* When using encoders, automatically interleave encoded packets
before sending it to the output.
* Pair encoders and have them automatically wait for the other to
start to ensure sync.
* Change 'obs_output_signal_start_fail' to 'obs_output_signal_stop'
because it was a bit confusing, and doing this makes a lot more
sense for outputs that need to stop suddenly (disconnections/etc).
- Encoder API:
* Remove some unnecessary encoder functions from the actual API and
make them internal. Most of the encoder functions are handled
automatically by outputs anyway, so there's no real need to expose
them and end up inadvertently confusing plugin writers.
* Have audio encoders wait for the video encoder to get a frame, then
start at the exact data point that the first video frame starts to
ensure the most accrate sync of video/audio possible.
* Add a required 'frame_size' callback for audio encoders that
returns the expected number of frames desired to encode with. This
way, the libobs encoder API can handle the circular buffering
internally automatically for the encoder modules, so encoder
writers don't have to do it themselves.
- Fix a few bugs in the serializer interface. It was passing the wrong
variable for the data in a few cases.
- If a source has video, make obs_source_update defer the actual update
callback until the tick function is called to prevent threading
issues.
- Add interleaving of video/audio packets for outputs that are encoded
and expect both video and audio data, sorting the packets and sending
them to the output when both video and audio is received.
- Combine create and initialize callbacks for the encoder API callback
interface.
Improve the properties API so that it can actually respond somewhat to
user input. Maybe later this might be further improved or replaced with
something script-based.
When creating a property, you can now add a callback to that property
that notifies when the property has been changed in the user interface.
Return true if you want the properties to be refreshed, or false if not.
Though now that I think about it I doubt there would ever be a case
where you would have this callback and *not* refresh the properties.
Regardless, this allows functions to change the values of properties or
settings, or enable/disable/hide other property controls from view
dynamically.
- Add start/stop code to obs-output module
- Use a circular buffer for the buffered encoder packets instead of a
dynamic array
- Add pthreads.lib as a dependency to obs-output module on windows in
visual studio project files
- Fix an windows export bug for avc parsing functions on windows.
Also, rename those functions to be more consistent with each other.
- Make outputs use a single function for encoded data rather than
multiple functions
- Add the ability to make 'text' properties be passworded
- obs-outputs module: Add preliminary code to send out data, and add
an FLV muxer. This time we don't really need to build the packets
ourselves, we can just use the FLV muxer and send it directly to
RTMP_Write and it should automatically parse the entire stream for us
without us having to do much manual code at all. We'll see how it
goes.
- libobs: Add AVC NAL packet parsing code
- libobs/media-io: Add quick helper functions for audio/video to get
the width/height/fps/samplerate/etc rather than having to query the
info structures each time.
- libobs (obs-output.c): Change 'connect' signal to 'start' and 'stop'
signals. 'start' now specifies an error code rather than whether it
simply failed, that way the client can actually know *why* a failure
occurred. Added those error codes to obs-defs.h.
- libobs: Add a few functions to duplicate/free encoder packets
- Make it so that encoders can be assigned to outputs. If an encoder
is destroyed, it will automatically remove itself from that output.
I specifically didn't want to do reference counting because it leaves
too much potential for unchecked references and it just felt like it
would be more trouble than it's worth.
- Add a 'flags' value to the output definition structure. This lets
the output specify if it uses video/audio, and whether the output is
meant to be used with OBS encoders or not.
- Remove boilerplate code for outputs. This makes it easier to program
outputs. The boilerplate code involved before was mostly just
involving connecting to the audio/video data streams directly in each
output plugin.
Instead of doing that, simply add plugin callback functions for
receiving video/audio (either encoded or non-encoded, whichever it's
set to use), and then call obs_output_begin_data_capture and
obs_output_end_data_capture to automatically handle setting up
connections to raw or encoded video/audio streams for the plugin.
- Remove 'active' function from output callbacks, as it's no longer
really needed now that the libobs output context automatically knows
when the output is active or not.
- Make it so that an encoder cannot be destroyed until all data
connections to the encoder have been removed.
- Change the 'start' and 'stop' functions in the encoder interface to
just an 'initialize' callback, which initializes the encoder.
- Make it so that the encoder must be initialized first before the data
stream can be started. The reason why initialization was separated
from starting the encoder stream was because we need to be able to
check that the settings used with the encoder *can* be used first.
This problem was especially annoying if you had both video/audio
encoding. Before, you'd have to check the return value from
obs_encoder_start, and if that second encoder fails, then you
basically had to stop the first encoder again, making for
unnecessary boilerplate code whenever starting up two encoders.
- Add a properties window for sources so that you can now actually edit
the settings for sources. Also, display the source by itself in the
window (Note: not working on mac, and possibly not working on linux).
When changing the settings for a source, it will call
obs_source_update on that source when you have modified any values
automatically.
- Add a properties 'widget', eventually I want to turn this in to a
regular nice properties view like you'd see in the designer, but
right now it just uses a form layout in a QScrollArea with regular
controls to display the properties. It's clunky but works for the
time being.
- Make it so that swap chains and the main graphics subsystem will
automatically use at least one backbuffer if none was specified
- Fix bug where displays weren't added to the main display array
- Make it so that you can get the properties of a source via the actual
pointer of a source/encoder/output in addition to being able to look
up properties via identifier.
- When registering source types, check for required functions (wasn't
doing it before). getheight/getwidth should not be optional if it's
a video source as well.
- Add an RAII OBSObj wrapper to obs.hpp for non-reference-counted
libobs pointers
- Add an RAII OBSSignal wrapper to obs.hpp for libobs signals to
automatically disconnect them on destruction
- Move the "scale and center" calculation in window-basic-main.cpp to
its own function and in its own source file
- Add an 'update' callback to WASAPI audio sources
- Implement OBS encoder interface. It was previously incomplete, but
now is reaching some level of completion, though probably should
still be considered preliminary.
I had originally implemented it so that encoders only have a 'reset'
function to reset their parameters, but I felt that having both a
'start' and 'stop' function would be useful.
Encoders are now assigned to a specific video/audio media output each
rather than implicitely assigned to the main obs video/audio
contexts. This allows separate encoder contexts that aren't
necessarily assigned to the main video/audio context (which is useful
for things such as recording specific sources). Will probably have
to do this for regular obs outputs as well.
When creating an encoder, you must now explicitely state whether that
encoder is an audio or video encoder.
Audio and video can optionally be automatically converted depending
on what the encoder specifies.
When something 'attaches' to an encoder, the first attachment starts
the encoder, and the encoder automatically attaches to the media
output context associated with it. Subsequent attachments won't have
the same effect, they will just start receiving the same encoder data
when the next keyframe plays (along with SEI if any). When detaching
from the encoder, the last detachment will fully stop the encoder and
detach the encoder from the media output context associated with the
encoder.
SEI must actually be exported separately; because new encoder
attachments may not always be at the beginning of the stream, the
first keyframe they get must have that SEI data in it. If the
encoder has SEI data, it needs only add one small function to simply
query that SEI data, and then that data will be handled automatically
by libobs for all subsequent encoder attachments.
- Implement x264 encoder plugin, move x264 files to separate plugin to
separate necessary dependencies.
- Change video/audio frame output structures to not use const
qualifiers to prevent issues with non-const function usage elsewhere.
This was an issue when writing the x264 encoder, as the x264 encoder
expects non-const frame data.
Change stagesurf_map to return a non-const data type to prevent this
as well.
- Change full range parameter of video scaler to be an enum rather than
boolean
- Add some temporary streaming code using FFmpeg. FFmpeg itself is not
very ideal for streaming; lack of direct control of the sockets and
no framedrop handling means that FFmpeg is definitely not something
you want to use without wrapper code. I'd prefer writing my own
network framework in this particular case just because you give away
so much control of the network interface. Wasted an entire day
trying to go through FFmpeg issues.
There's just no way FFmpeg should be used for real streaming (at
least without being patched or submitting some sort of patch, but I'm
sort of feeling "meh" on that idea)
I had to end up writing multiple threads just to handle both
connecting and writing, because av_interleaved_write_frame blocks
every call, stalling the main encoder thread, and thus also stalling
draw signals.
- Add some temporary user interface for streaming settings. This is
just temporary for the time being. It's in the outputs section of
the basic-mode settings
- Make it so that dynamic arrays do not free all their data when the
size just happens to be reduced to 0. This prevents constant
reallocation when an array keeps going from 1 item to 0 items. Also,
it was bad to become dependent upon that functionality. You must now
always explicitly call "free" on it to ensure the data is free, and
that's how it should be. Implicit functionality can lead to
confusion and maintainability issues.
- Implement a means of obtaining default settings for an
input/output/encoder. obs_source_defaults for example will return
the default settings for a particular source type.
- Because C++ doesn't have designated initializers, use functions in
the WASAPI plugin to register the sources instead.
LOG_ERROR should be used in places where though recoverable (or at least
something that can be handled safely), was unexpected, and may affect
the user/application.
LOG_WARNING should be used in places where it's not entirely unexpected,
is recoverable, and doesn't really affect the user/application.
Make it so obs_data settings input in to *_update are applied to the
existing settings rather than fully replace the existing settings. That
way you can update with only certain specific settings, leaving other
settings untouched. Of course if you're already using the original
settings pointer in the first place then you've already done that, so
it'll just ignore it because you've already applied them.
The API used to be designed in such a way to where it would expect
exports for each individual source/output/encoder/etc. You would export
functions for each and it would automatically load those functions based
on a specific naming scheme from the module.
The idea behind this was that I wanted to limit the usage of structures
in the API so only functions could be used. It was an interesting idea
in theory, but this idea turned out to be flawed in a number of ways:
1.) Requiring exports to create sources/outputs/encoders/etc meant that
you could not create them by any other means, which meant that
things like faruton's .net plugin would become difficult.
2.) Export function declarations could not be checked, therefore if you
created a function with the wrong parameters and parameter types,
the compiler wouldn't know how to check for that.
3.) Required overly complex load functions in libobs just to handle it.
It makes much more sense to just have a load function that you call
manually. Complexity is the bane of all good programs.
4.) It required that you have functions of specific names, which looked
and felt somewhat unsightly.
So, to fix these issues, I replaced it with a more commonly used API
scheme, seen commonly in places like kernels and typical C libraries
with abstraction. You simply create a structure that contains the
callback definitions, and you pass it to a function to register that
definition (such as obs_register_source), which you call in the
obs_module_load of the module.
It will also automatically check the structure size and ensure that it
only loads the required values if the structure happened to add new
values in an API change.
The "main" source file for each module must include obs-module.h, and
must use OBS_DECLARE_MODULE() within that source file.
Also, started writing some doxygen documentation in to the main library
headers. Will add more detailed documentation as I go.
- Add planar audio support. FFmpeg and libav use planar audio for many
encoders, so it was somewhat necessary to add support in libobs
itself.
- Improve/adjust FFmpeg test output plugin. The exports were somewhat
messed up (making me rethink how exports should be done). Not yet
functional; it handles video properly, but it still does not handle
audio properly.
- Improve planar video code. The planar video code was not properly
accounting for row sizes for each plane. Specifying row sizes for
each plane has now been added. This will also make it more compatible
with FFmpeg/libav.
- Fixed a bug where callbacks wouldn't create properly in audio-io and
video-io code.
- Implement 'blogva' function to allow for va_list usage with libobs
logging.
- Add property list callbacks to sources/outputs/encoders so that if
necessary user interface can be automatically generated or perhaps a
property list widget can be used for them.
- Change some of the property API names. obs_property_list_t felt a bit
awkward when actually using it, so I just renamed it to
obs_properties_t.
- Removed the getdata/setdata nad getparam/setparam functions from
sources/services, they will be superseded by the dynamic procedure
call API.
- Add 'set_default' functions to obs-data.*. These functions ensure
that a paramter exists and that the parameter is of a specific type.
If not, it will create or overwrite the value with the default setting
instead.
These functions are meant to be explicitly called before using any of
the 'get' functions. The reason why it was designed this way is to
encourage defaults to be set in a single place/function.
For example, ideal usage is to create one function for your data,
"set_my_defaults(obs_data_t data)", set all the default values within
that function, and then call that function on create/update, that way
all defaults are centralized to a single place.
- Ensure that data passed to sources/encoders/outputs/etc is always
valid, and not a null value.
- While I'm remembering, fix a few defaults of the main program config
file data.
Add a fairly easy to use settings interface that can be passed to
plugins, and replaced the old character string system that was being
used before. The new data interface allows for an easier method of
getting/altering settings for plugins, and is built to be serializable
to/from JSON.
Also, removed another wxFormBuilder file that was no longer in use.
Just a minor fix mostly because I noticed that I kept accidentally
forgetting to add checks to the code properly. This is one of those
cases where macros come in useful, as macros can automate the process
and help prevent these mistakes from happening by accident.
Changed the comments to properly reflect the new callbacks, as I had
forgotten to update the comments for them both.
Also, changed "setbitrate" and "request_keyframe" return values to be
boolean.
- First, I redid the output interface for libobs. I feel like it's
going in a pretty good direction in terms of design.
Right now, the design is so that outputs and encoders are separate.
One or more outputs can connect to a specific encoder to receive its
data, or the output can connect directly to raw data from libobs
output itself, if the output doesn't want to use a designated encoder.
Data is received via callbacks set when you connect to the encoder or
raw output. Multiple outputs can receive the data from a single
encoder context if need be (such as for streaming to multiple channels
at once, and/or recording with the same data).
When an encoder is first connected to, it will connect to raw output,
and start encoding. Additional connections will receive that same
data being encoded as well after that. When the last encoder has
disconnected, it will stop encoding. If for some reason the encoder
needs to stop, it will use the callback with NULL to signal that
encoding has stopped. Some of these things may be subject to change
in the future, though it feels pretty good with this design so far.
Will have to see how well it works out in practice versus theory.
- Second, Started adding preliminary RTMP/x264 output plugin code.
To speed things up, I might just make a direct raw->FFmpeg output to
create a quick output plugin that we can start using for testing all
the subsystems.
