Implement automatic audio device selection for devices that use two
separate DirectShow filters for audio and video instead of having a
single filter with audio and video output pins.
Please note that this fix is currently only active for Elgato USB and
PCIe devices (e.g. Cam Link, HD60 S, HD60 Pro, 4K60 Pro) to avoid
unintentionally changing the behavior for any other devices (e.g.
webcams).
(Jim edit: This fixes an issue with newer Elgato devices where the
devices would not automatically have their audio coupled with the video;
users would have to manually select the audio device in order to get
audio functioning.)
Closesjp9000/obs-studio#1081
(This commit also modifies the following modules: UI,
deps/media-playback, coreaudio-encoder, decklink, linux-alsa,
linux-pulseaudio, mac-capture, obs-ffmpeg, obs-filters, obs-libfdk,
obs-outputs, win-dshow, and win-wasapi)
Adds surround sound audio support to the core, core plugins, and user
interface.
Compatible streaming services: Twitch, FB 360 live
Compatible protocols: rtmp / mpeg-ts tcp udp
Compatible file formats: mkv mp4 ts (others untested)
Compatible codecs: ffmpeg aac, fdk_aac, CoreAudio aac,
opus, vorbis, pcm (others untested).
Tested streaming servers: wowza, nginx
HLS, mpeg-dash : surround passthrough
Html5 players tested with live surround:
videojs, mediaelement, viblast (hls+dash), hls.js
Decklink: on win32, swap channels order for 5.1 7.1
(due to different channel mapping on wav, mpeg, ffmpeg)
Audio filters: surround working.
Monitoring: surround working (win macOs linux (pulse-audio)).
VST: stereo plugins keep in general only the first two channels.
surround plugins should work (e.g. mcfx does).
OS: win, macOs, linux (alsa, pulse-audio).
Misc: larger audio bitrates unlocked to accommodate more channels
NB: mf-aac only supports mono and stereo + 5.1 on win 10
(not implemented due to lack of usefulness)
Closesjp9000/obs-studio#968
(This commit also modifies the deps/media-playback, obs-ffmpeg, and
win-dshow modules)
More fixes due to ffmpeg renaming some constants and deprecating
AVFMT_RAWPICTURE and AV_PIX_FMT_VDA_VLD.
Latter replaced by AV_PIX_FMT_VIDEOTOOLBOX per ffmpeg dev advice.
Closesjp9000/obs-studio#1061
Video playback doesn't work if the default format is MJPEG and there are
other formats to use; this is because the useDefaultConfig variable is
still set to true, which overrides the format value that would normally
tell it to convert to RGB.
(This commit also modifies the decklink, linux-v4l2, mac-avcapture,
obs-ffmpeg, and win-dshow modules)
Originally, async buffering for sources was supposed to be a
user-controllable flag. However, that turned out to be less than ideal
because sources (such as the win-dshow plugin) were programmed with
automatic control over their buffering (such as automatically detecting
USB 2.0 capture devices and then enabling in those cases).
The fact that it was a flag caused a design flaw to where buffering
values would be overwritten when a source is loaded from save data.
Because of that, this flag is being deprecated and replaced with a
specific function to enable unbuffered mode instead.
When the windows video device source source is set to only activate when
showing, it would still activate on first startup of the program even if
it was in another scene and not showing anywhere to the user. This
fixes that issue.
The LGP issue is caused by the device drivers returning two or more
packets in a single segment of audio data. This fixes it by detecting
that and decoding subsequent packets.
When the FFmpeg audio decoder returns, it returns how many bytes of data
was decoded. To have it decode multiple packets in a single segment,
just subtract the return value from the expected size, and if that size
is still larger than zero, then there are more packets in the segment to
decode. Otherwise, stop.
LGP devices are devices that induce anger in any sane developer because
they're prone to bad audio timestamps when using their decoded data
directly. For that reason, add a hack that smooths the timestamps
within a large threshold to prevent audio skipping.
Useful for two purposes:
1.) When many devices are hooked up to the system and used in separate
scenes, but only one device active at once is desired
2.) Allows users who are dependent on outputting audio to desktop to
disable that audio (via disabling that device) when the device isn't
being displayed
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.
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 allows the ability to output the audio of the device as desktop
audio (via the WaveOut or DirectSound audio renderers) instead of
capturing the audio only.
In the future, we'll implement audio monitoring which will make this
feature obsolete, but for the time being I decided to add this option as
a temporary measure to allow users to play the audio from their devices
via the DirectShow output.
Thomas Schnitzler
pkviet
jp9000
Gol-D-Ace
mntone
Kazuki Oishi
jpk
dodgepong