If the transition point was above or equal to 1.0, it would cause a
divide by 0 error a few lines down. This could cause audio data to
become corrupted with NAN audio data when mixing, which can cause
certain audio encoders (namely the FFmpeg AAC encoder) to fail.
It was possible for the transition point to be above or equal to 1.0 if
the stinger media file was no longer loadable for whatever reason.
Back in an older commit, the default YUV colorspace was changed to 601
in order to ensure correct playback in video players that ignore the
contents of the H264 header.
x264's "undef" is unfortunately not what players that don't ignore the
header expect for BT.601, resulting in incorrect colors when played
back. Setting it to "bt470bg", similar to what is specified in ffmpeg
outputs, remedies this issue.
We're expecting a variable with double precision. Since we don't read
the value of these doubles with a particular precision, it can often
lead to unpredictable results where the value set isn't the one
intended due to the loss of precision from float->double conversion.
When drawing cursor to window capture area - use actual resource width
and height instead of system metric values for icons. Fixes an issue
where under rare circumstances, certain cursors would not draw at the
correct size.
Closesobsproject/obs-studio#1284
Allows the ability to (optionally) synchronously create/update a
directshow device source rather than always asynchronously update the
device. This is useful if creating/destroying scenes/sources very
quickly, and helps minimize the risk of creating new directshow sources
that use the same device, yet may not activate because an existing
source may already exist. To use, set "synchronous_activate" to true in
its settings when updating or creating. Note that this setting will be
erased after it's used, and will not be saved to user settings, so it
must be set each time in order to be used.
Closesobsproject/obs-studio#1228
After you call av_frame_alloc(), ffmpeg expects you to fill in certain
fields on the frame, depending on whether it's an audio or video frame.
obs-ffmpeg did this in the two places where it allocates video frames,
but not where it allocates audio frames. On my system, using trunk
ffmpeg and the Opus codec, this causes OBS to crash while calling
avcodec_send_frame, ultimately because av_frame_copy fails due to
'dst->format < 0' (as 'format' stays at the default of -1), causing a
null pointer to be added to a buffer queue, which later gets
dereferenced.
Oddly, the fields in question can just be copied directly from
corresponding fields in the AVCodecContext, but I don't see any ffmpeg
API to automatically copy all relevant fields, and all the examples I've
seen do it by hand. So this patch does the same.
If the user selected the "default" device for audio capture on mac, it
would no longer be able to be switched to another device due to the
"default_device" variable being set.
In the vt_h264_video_info function, the format of the video to be encoded is
always being set to VIDEO_FORMAT_NV12, dispite the presence of code to set
the video format to VIDEO_FORMAT_I420 or VIDEO_FORMAT_I444. This commit fixes
that function to respect the video format choice of the user.
In addition, whilst testing this fix initially, I also discovered that the
4:4:4 colour format is not supported by the VideoToolbox H264 encoder.
Looking at the VideoToolbox code in ffmpeg as a reference, the ffmpeg code
errors out if a color format other than NV12 or I420 is set. Therefore, this
commit also logs a warning about I444 not being supported, and uses the NV12
default.
The fullrange variable is used to set appropriate video format information in
vt_h264_video_info. Set fullrange first so the video format data is correct in
all cases.
The cutoff hack was added many, many years ago as recommended by
Konverter. Since then, there has been much work on the AAC encoder, so
this hack should no longer be necessary.
The windows media foundation H264 encoders have been deprecated for over
a year, and microsoft's media foundation AAC encoder has had a continued
issue with occasional random audio glitches. The FFmpeg AAC encoder has
had recent development, and is more than sufficient to be able to handle
the task of encoding in terms of both quality and performance, so it's
better just to use the FFmpeg encoder from here on out.
As this plugin is no longer needed, for the next year or two it'll still
be compiled and included, but as a blank plugin that does nothing. The
reason why it's still being included as a blank no-operation plugin is
to overwrite older versions of the plugin. That way if a user installs
a newer OBS version over an older one, it won't load up the older win-mf
plugin where the encoders still were enabled.
This also fixes some rarely reported media foundation crashes that can
happen on startup.
Fixes an issue where align_pos could be smaller than
sizeof(struct shmem_data), potentially overwriting memory of the header.
References jp9000/obs-studio#1202
