If FFmpeg's experimental aac encoder is used, this changes the cutoff
frequency to better values in order to try to help make up for the
inherent lack of encoder quality a bit. If FFmpeg is compiled to use
another encoder by default, these settings will not be applied.
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 functionality can now be handled automatically because locale can
now be freed seaparately from obs_module_unload with
obs_module_free_locale, which is called automatically when the module is
being freed.
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.
Having the value stored here is somewhat pointless, so this is one step
in fixing the locale handling. Locale should be handled by the modules
themselves with their own loaded locale lookup information.
- 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.
- 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.
- 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
Make sure it locks the write mutex before freeing the packets, and put
the detach code in the main thread loop rather than off in a separate
function for clarity
...The reason why audio didn't work was because I overwrote the bitrate
values.
As for semaphores, mac doesn't support unnamed semaphores without using
mach semaphores. So, I just implemented a semaphore wrapper for each
OS.
- 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.
If the default device changes, set the reconnect interval to 200
milliseconds so it pretty much immediately tries to reinitialize the
audio with the newly selected default device. Otherwise, use 2000
millisecond intervals, and assume disconnection.
Also, reduced FFmpeg logging to just regular FFmpeg information rather
than everything FFmpeg logs.
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.
I can't believe I wasn't doing this. This is why file output was
getting corrupted. Audio and video send in data from separate threads.
I should be embarassed for not having considered that.
Key lesson: Increase threading paranoia levels. Apparently my
threading paranoid levels are lackluster.
FFmpeg test output wasn't make any attempt to sync data before. Should
be much more accurate now.
Also, added a restart message to audio settings if base audio settings
are changed.
Implement a few audio options in to the user interface as well as a few
inline audio functions in audio-io.h.
Make it so ffmpeg plugin automatically converts to the desired format.
Use regular interleaved float internally for audio instead of planar
float.
Add a scaler interface (defaults to swscale), and if a separate output
wants to use a different scale or format than the default output format,
allow a scaler instance to be created automatically for that output,
which will then receive the new scaled output.
jp9000
dodgepong
Ben Torell
Palana
BtbN