Due to crashes being caused by hardware acceleration in the media source
on mac, disable hardware acceleration of the media source being on by
default for the time being.
The encoder name was changed from "nvenc_h264" to "h264_nvenc", and will
throw a warning in the log file if you use the former, so try the latter
first, then the former.
This reverts commit bd70e73c25.
Turns out the commit was due to a miscommunication -- the commit it was
fixing actually worked fine, and this fix was unnecessary.
(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.
For some reason in the FFmpeg output, this AVCodecContext variable is
being set to 1 by FFmpeg itself somewhere, and it's causing a massive
slowdown when encoding with FFmpeg directly. This should be set to 0 to
specify to use as many threads as necessary.
On windows, if you were saving a file name or directory with characters
that are not of the current windows character set, it could cause the
file saving process to fail. This fixes it so that on windows it uses
wmain and converts the unicode command line to a UTF-8 command line,
which works with FFmpeg.
Instead of using an option that turns CBR on/off, adds rate control
methods: VBR, CBR, CQP, Lossless.
This moves lossless from being a preset to being a rate control method.
(Also modifies obs-ffmpeg to handle empty frames on EOF)
Previously the demuxer could hit EOF before the decoder threads are
finished, resulting in truncated output. In the worse case scenario the
demuxer could read small files before ff_decoder_refresh even has a chance
to start the clocks, resulting in no output at all.
The if statement erroneously ended with a ';', which means that the code
is always executed, but there's no reason to even have these if checks
in the first place as the functions themselves return safely with null
pointers.
(Note: Also modified the obs-ffmpeg plugin module)
Allows the ability for frame data to pass 8-bit grayscale images (Y800
color format).
Closesjp9000/obs-studio#515
If the media source is set to restart on activation, it also shuts down
when not active. However, it would *always* start regardless of
active/inactive when the source is first created. It shouldn't do that,
it should start up only when it becomes active.
Certain types of sources (display captures, game captures, audio
device captures, video device captures) should not be duplicated. This
capability flag hints that the source prefers references over full
duplication.
Adds the option of making the media file restart when the source becomes
active (such as switching to a scene with it).
Due to lack of libff features to start/stop/pause/seek media files,
currently this just destroys the demuxer and recreates it. Ideally,
libff should have some functions to allow a more optimal means of doing
those things.
Reactors a bit of code related to starting up FFmpeg and makes it so the
initial view for the media source's properties displays the most
commonly desired settings.
Instead of the media source properties showing the URL mode by default
along with a whole bunch of properties that are confusing to most users,
starts on file mode and changes defaults to be a bit more sensible
related to file input.
Also, as a temporary measure for fixing color format issues (some video
files would display their color information incorrectly), forced format
conversion is now enabled by default, and has been moved to advanced
settings. Ideally, the actual bug causing color format issues in either
media-io or libff should be fixed at some point.
When browsing for a file, it would also just use *.* for the file
filter, which is a pain to use. This has been changed to use a
reasonable file filter related to common video/audio files so you don't
have to wade through non-media files just to select a media file. A
filter to show all files is still available as well.
Another thread could be manipulating the active_log_contexts array while the current thread is trying to read it, resulting in an uninitialized memory crash as the da_push_back call was not protected by the mutex.
This also adds the ability to detect whether it stopped due to lack of
space or not -- particularly useful for the FFmpeg output due to
lossless file format support.
For the FFmpeg output, the encoder ids are sort of superfluous. They
really should be optional. If they're not set, it should use the
encoder name string instead to determine the ids automatically.
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 is used by some muxers that set AVFMT_NOFILE and doesn't seem to
hurt muxers that don't set it; notable this makes the hls muxer output
its m3u8 playlist with the proper filename in the proper directory
This particularly affected audio encoding, audio encoding previously
would count samples and use it to create an encoding timestamp, but
because I was using a standard integer (which is 32bit by default on
x86), it would max out at about 0x7FFFFFFF samples, which is about 12
hours of samples at 48000 sample rate. After that, it would start going
into negative territory (overflowing). By changing it to int64_t, it
will make it so that audio at 48000 samples per second would only be
able to overflow after about.. 6.09 million years. In other words,
this should fix the issue for good.
In addition to the flv file format, this allows the ability to save to
container formats such as mp4, ts, mkv, and any other containers that
support the current codecs being used.
It pipes the encoded data to the ffmpeg-mux process, which then safely
muxes the file from the encoded data. If the main program unexpectedly
terminates, the ffmpeg-mux piped program will safely close the file and
write trailer data, preventing file corruption.
Instead of using system timestamps for playback, use the timestamps
directly from the video/audio data to ensure all the data is synced up
using the obs_source back-end.
I think the original misconception when this was written was that OBS
would not handle timestamp resets/loops, which isn't the case; it will
actually handle all timestamp resets and abnormalities. It's always
best to use the obs_source back-end for all playback and syncing.
In the settings if you select default container then the
format becomes null. If null then audio/video codec ids should
not be set on the output format as they will both be
AV_CODEC_ID_NONE causing a context with no streams specified
to be created (error).
API Changed (in struct obs_encoder_info):
----------------------------------------
bool (*get_audio_info)(void *data, struct audio_convert_info *info);
bool (*get_video_info)(void *data, struct video_scale_info *info);
To:
----------------------------------------
void (*get_audio_info)(void *data, struct audio_convert_info *info);
void (*get_video_info)(void *data, struct video_scale_info *info);
The encoder video/audio information callbacks no longer need to manually
query the libobs video/audio information, that information is now passed
via the parameter, which the callbacks can modify.
The refactor that reduces boilerplate in the encoder video/audio
information callbacks also removes the need for their return values, so
change the return types to void.
Check the actual name of the codec before applying an x264-specific
preset so we don't encounter an "Invalid argument" error when using
other h264 encoders in FFmpeg (such as NVEnc).
Closesjp9000/obs-studio#412
Some formats (like WMV) would send out audio packets that
had channels set but did not specify a channel layout.
Solution is to no longer rely on channel layout to get the
channels and just get the channel count directly off the
FFmpeg audio frame.
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.
This makes FFmpeg usable as an output, and removes or changes most of
the code that was originally intended for testing purposes.
Changes the settings for the FFmpeg output to the following:
* url: Sets the output URL or file path
* video_bitrate: Sets the video bitrate
* audio_bitrate: Sets the audio bitrate
* video_encoder: Sets the video encoder (by name, blank for default)
* audio_encoder: Sets the audio encoder (by name, blank for default)
* video_settings: Sets custom video encoder FFmpeg settings
* audio_settings: Sets custom audio encoder FFmpeg settings
* scale_width: Image scale width (0 if none)
* scale_height: Image scale height (0 if none)
The reason why scale_width and scale_height are provided is because it
may internally convert formats, and it may be a bit more optimal to use
that scaler instead of the pre-output scaler just in case it already has
to convert formats internally anyway (though you can do it either way
you wish).
Video format handling has also changed; it will now attempt to use the
closest format to the current format if available for a given video
codec.
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.