Currently, ifdefs are used to determine if monitoring is supported.
This is difficult to maintain and restricts plugins from knowing if
monitoring is supported by OBS. This adds a runtime function to fix
that issue.
Previously we would wait for pulse to attempt to read from the monitor
source and obs buffered at least 5ms of audio data before we tried to
fill the buffer. In some cases this resulted in consistently triggering
underruns in pulse.
Instead we try to fill the buffer immediately as obs outputs audio data
and while the pa buffer is not full. We also stop trying to grow the
buffer to prevent underruns after we reach 1s of latency.
When signed 32-bit audio arrived to pulseaudio-output and volume was
lowered, audio data was broken. In the function `process_volume`, the
type of the data is switched by `bytes_per_channel`. However the size of
signed 32-bit integer and the size of float are same so that the signed
32-bit integer is processed as float.
This commit changes these items.
- Use `format` instead of `bytes_per_channel` so that all the sample
types can be differentiated.
- Change `short` to `int16_t` and renames existing function
`process_short` to `process_s16` to clarify the function is
processing signed 16-bit.
PulseAudio code needs to be called with the PA lock held. This chiefly
fixes multiple races during stream shutdown:
* If the functions are called without the lock held, deferred event
handling races end up with PA infinite looping on the mainloop, or
asserting, or other badness, as the reentrant calls cause data
structure corruption on the PA side.
* If we don't reset our callbacks, PA might call us even after we
request stream disconnection (since the stream actually getting fully
shut down is asynchronous), and then we dereference NULL pointers from
our userdata etc. PA will keep its data structures alive until necessary
via reference counting, but not ours.
The lock around pa_stream_begin_write doesn't result from any issues I
experienced, but it looks correct; PA doesn't say anywhere that that
function is thread-safe.
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.
When using more than two channels, the channel map of pulse-audio is incorrect.
Add an API for getting a speaker map based on OBS speaker layout. Then use the
speaker map when connecting to a pulse-audio device, for both source and
monitor output.
This pull request changes the fallback sample format for pulse-audio
to from PA_SAMPLE_S16LE to PA_SAMPLE_FLOAT32LE.
The pulseaudio plugin can handle the following sample format:
* PA_SAMPlE_U8
* PA_SAMPLE_S16LE
* PA_SAMPLE_S32LE
* PA_SAMPLE_FLOAT32LE
When an audio device advertises itself as another format, the pulseaudio-plugin
will ask pulse audio to convert to the fallback sample format.
The fallback PA_SAMPLE_S16LE is not ideal when your audio interface advertises
as PA_SAMPLE_S24LE since the conversion will lose precision.
With PA_SAMPLE_FLOAT32LE there is no precision loss and it is also equals OBS's
internal format.
(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
