The uninit function prematurely released the memory allocated for the
channel map setting, which exists in the scope of source life cycle
(compared to channel names which are valid during the life cycle of a
configured device).
Splitting up the clean-up for both (memory for channel names is
released when the device is uninitialized, memory for channel map
setting is released when the source is removed) ensures that the memory
is released but pointers don't become unexpectedly invalid.
OBS handles up to 8 channels of audio, which requires the CoreAudio
input callback to omit/ignore any audio data from channels above that
count and also report a maximum amount of 8 channels to match the
speaker setup given to swresample by libobs.
The timestamp returned by mHostTime in the AudioTimeStamp structure
is the current timestamp equivalent to mach_absolute_time(), which
is relative to the machine's time base. In order to convert this
to milliseconds, it's necessary to get the host's timebase with
mach_timebase_info() and scale the timestamp accordingly, since
the rest of the timestamp synchronization code expects the timestamp
to be in milliseconds.
This is effectively equivalent to the code which was previously in
libobs/util/platform-coca.m, but must be applied here instead.
Code submissions have continually suffered from formatting
inconsistencies that constantly have to be addressed. Using
clang-format simplifies this by making code formatting more consistent,
and allows automation of the code formatting so that maintainers can
focus more on the code itself instead of code formatting.
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.
Some audio devices do not have a fixed number of channels. For example,
Soundflower. This was previously fixed by defaulting the speaker layout
to stereo. With surround sound support, the default has been changed to
the output speaker layout as set in Settings > Audio.
Closesjp9000/obs-studio#1110
(also obs, deps/media-playback, libobs/audio-monitoring, decklink,
linux-alsa, linux-pulseaudio, mac-capture, obs-ffmpeg, win-dshow,
win-wasapi)
Default channel layout for 4 channels is 4.0 in FFmpeg.
Replacing quad with 4.0 will improve compatibility since FFmpeg has
better support of its default channel layouts.
(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
(Note: This commits also modifies the linux-pulseaudio, mac-capture, and
win-wasapi plugins)
Do not prevent the targeted output device from being monitored if the
selected monitor output device is a different one.
Closesjp9000/obs-studio#872
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.
Certain devices (particularly certain mixers and soundflower 64ch) would
have an arbitrary number of channels, and wouldn't really be mappable to
a specific speaker layout supported by libobs.
So to fix this issue, if the channel count is above 8, force the data to
stereo to ensure playback can still occur, rather than cause it to just
fail.
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.
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.
Apparently, despite the fact that Apple added
kAudioHardwarePropertyTranslateUIDToDevice in 10.8, it's not actually
usable in 10.8, only 10.9. So, instead of being able to use it like a
normal, sane person, we have to enumerate all devices manually and find
the AudioDeviceID ourselves. A slight annoyance and a mark against
apple's competence, but audio devices should now be working again on
10.8 at least, so whatever.
Improve the properties API so that it can actually respond somewhat to
user input. Maybe later this might be further improved or replaced with
something script-based.
When creating a property, you can now add a callback to that property
that notifies when the property has been changed in the user interface.
Return true if you want the properties to be refreshed, or false if not.
Though now that I think about it I doubt there would ever be a case
where you would have this callback and *not* refresh the properties.
Regardless, this allows functions to change the values of properties or
settings, or enable/disable/hide other property controls from view
dynamically.
- Implement a means of obtaining default settings for an
input/output/encoder. obs_source_defaults for example will return
the default settings for a particular source type.
- Because C++ doesn't have designated initializers, use functions in
the WASAPI plugin to register the sources instead.
- Split input and output audio captures so that they're different
sources. This allows easier handling and enumeration of audio
devices without having to do some sort of string processing.
This way the user interface code can handle this a bit more easily,
and so that it doesn't confuse users either. This should be done for
all audio capture sources for all operating systems. You don't have
to duplicate any code, you just need to create input/output wrapper
functions to designate the audio as input or output before creation.
- Make it detect soundflower and wavtap devices as mac "output" devices
(even though they're actually input) for the mac output capture, and
make it so that users can select a default output capture and
automatically use soundflower or wavtap.
I'm not entirely happy about having to do this, but because mac is
designed this way, this is really the only way to handle it that
makes it easier for users and UI code to deal with.
Note that soundflower and wavtap are still also designated as input
devices, so will still show up in input device enumeration.
- Remove pragma messages because they were kind polluting the other
compiler messages and just getting in the way. In the future we can
just do a grep for TODO to find them.
- Redo list property again, this time using a safer internal array,
rather than requiring sketchy array inputs. Having functions handle
everything behind the scenes is much safer.
- Remove the reference counter debug log code, as it was included
unintentionally in a commit.
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.
