The frame is definitely meant to be modifiable if needed, so it
shouldn't be const. I originally meant for these frames to be
duplicated, but with the source video cache that's both unnecessary and
would reduce performance.
Fixes a bug where get_format was overloaded by our own version
when forcing the codec to load with a HW decoder and not
reset to the original get_format if it failed to load.
The code specific to Windows: helps convert `BSTR` instances to
`std::string`s; provides a Windows COM-specific implementation of
`CreateDeckLinkDiscoveryInstance`; aliases CFUUIDGetUUIDBytes,
CFUUIDBytes, and IUnknownUUID (the Linux SDK does this, but for some
reason the Windows SDK does not).
Some changes were made to the stock DeckLink SDK: removed all references
to legacy API headers in DeckLinkAPI.idl; removed all instances of
`importlib("stdole2.tlb");`.
In the same manner that PNG doesn't appear to work properly
with multiple threads, TIFF, JPEG2000 and WEBP also appears
to not render correctly (use of FFmpeg's ff_thread_* routines)
if decode is called before the automatic thread detection
has returned a suggested thread count for the decoder.
If this is omitted and you use an input that requires the network
you get a warning message about future versions not automatically
doing this for you.
This attaches clocks to packets and frames and defers
the start time until that particular frame is presented.
Any packets/frames in the future with the same clock
will reference that start time.
This fixes issues when there are multiple start times
in a large buffer (looped video/images/audio) and different
frames need different reference clocks to present correctly.
Enables clocks to wait if the main sync clock has not been started yet. An example of this is a video stream (video/audio) being synced to the video clock. If an audio frame gets produced before the video clock has started it will wait.
Add referencing counting to determine when to release a clock Due to no fixed ownership of clocks (packets, frames, decoders and refresh thread all may own a clock at some point).
The bug was undetected because it accidentally fell into an error case that slept the correct amount of time. pthread_cond_timedwait takes an absolute time in the future to wait until. The value we were passing was always in the past so it was immediately failing with a TIMEDOUT error code.
This lets the decoder make decisions based on whether it is a hardware decoder or not. Specifically, hardware decoders are more strict as to which frames can be dropped in an h264 stream.
This fixes an issue where the borders for certain types of windows would
not match the intended border style/color. It was supposed to be 1
through 6 for frameShape, but I ended up putting 5 twice.
If the audio data had the same format/samplerate as the obs audio
subsystem, it would fail to simply destroy the resampler and set it to
NULL, and then any audio data going through would use the resampler that
was being used before that, causing audio to become garbage.
This bug only started appearing when I recently changed the libobs
internal audio subsystem format to non-interleaved floating point, which
is a common format, and thus caused this bug to actually occur more
often.
I when a source has both async audio/video capability, it would ignore
audio until the video has started. There's really no need to do this,
when the video starts it'll just fix up the timing automatically.
This should fix the case where things like the media source would not be
able to play audio-only files.
Core API functions changed:
-----------------------------
EXPORT bool obs_reset_audio(struct audio_output_info *aoi);
EXPORT bool obs_get_audio_info(struct audio_output_info *aoi);
To:
-----------------------------
EXPORT bool obs_reset_audio(const struct obs_audio_info *oai);
EXPORT bool obs_get_audio_info(struct obs_audio_info *oai);
Core structure added:
-----------------------------
struct obs_audio_info {
uint32_t samples_per_sec;
enum speaker_layout speakers;
uint64_t buffer_ms;
};
Non-interleaved (planar) floating point output is standard with audio
filtering, so to prevent audio filters from having to worry about
different audio format implementations and for the sake consistency
between user interfaces, make it so that audio is always set to
non-interleaved floating point output.
jp9000
kc5nra
Skyler Lipthay
Radzaquiel
Gol-D-Ace