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
(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
When a pulseaudio source is created from existing settings and the
device is not found, then the private data will return NULL, making it
impossible to change the source to use a different device. NULL should
never be returned when possible, otherwise settings cannot be changed
and properties cannot be displayed.
Closesjp9000/obs-studio#604
Add checks to make sure the pulseaudio operation objects are actually
valid and the request did not fail right away. Other pulseaudio
functions dealing with the objects expect them not to be NULL,
otherwise they will kill the application with a failing assert.
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.
Add the include directories found by cmake to the pulseaudio plugin.
This allows for the plugin to compile when the pulseaudio headers were
found in a directory that is not normally in the search path of the
compiler (e.g. /usr/local/include)
This adds a check to change the capture settings to use 2 channels when
a channel number is encountered that would otherwise be interpreted as
SPEAKERS_UNKNOWN.
If the sample format used by PulseAudio can not be converted into an
OBS audio format it will be handled as AUDIO_FORMAT_UNKNOWN which will
not result in a proper audio recording. So instead we request a format
that OBS supports from PulseAudio and let it do the format conversion.
The format PA_SAMPLE_S24_32LE is a 24 bit audio format in 32 bit integers
and not a 32 bit audio format and so it should no be mapped to
AUDIO_FORMAT_32BIT.
Before it was giving timestamps based upon system time for each new
segment of audio data. Also, it was subtracting pulse latency from the
audio timestamp, which seems like it was really meant for use with the
pulse audio time rather than system time.
Now, it just uses system time for timestamps. Still might not be
totally perfect, but seems to be much better than it was.
This also removes the latency calculation. Latency is no longer used
because we're not using pulseaudio timing.
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.
When setting up the capture, the plugin will now query pulse for the default
format of the specific source instead of the server.
This is useful if a source has different settings than what the defaults are
for the server, e.g. when the source is an output with 5.1 surround sound
and the microphone input is mono while the server defaults to stereo sound.
This fixes a bug where the pulseaudio plugin always reported
a speaker layout of stereo to obs, regardless of how many channels
pulseaudio actually recorded.
If the default number of channels was different to 2 this would
cause audio distortion.
