This reverts commit 94b5982216.
Reverting this commit because it had some negative side effects, such as
adding 500 milliseconds to the startup time. NVENC detection should
really be done through its proper API, and not via creating an encoder
on startup.
There were cases where the channel format could be set to 7, which used
to be a valid format but now no longer is. If that format is set, just
use SPEAKERS_7POINT1 instead.
Makes it a bit more clear this option shouldn't be used unless you're on
SLI/crossfire.
In the future, something should be put in to the program that detects
laptops and warns on how to set up their adapter for efficient capture.
Closesjp9000/obs-studio#1138
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.
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
The list of channel layouts available for decklink input is missing 2.1
& 4.1 layouts. The commit adds them. This aligns the decklink input
with the speaker layouts available at outputs. Having different layouts
as input and output invokes FFmpeg resampler, which remixes the channels
in non trivial way except when downmixing to stereo. This patch allows
to avoid such uncontrolled remix of channels with decklink input.
The core audio aac encoder has bitrates maps specific to speaker
layouts. Previously, the bitrate map maxed at 320 kbs and was the map
for stereo. The bitrate map is now tailored to the speaker layout. In
practice this unlocks higher bitrates. For instance up to 960 kbs for
7.1. Additionally the commit fixes a bug with 2.1 with channels not
ordered correctly.
(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.
(also modifies obs-ffmpeg, audio-monitoring, win-wasapi, decklink,
obs-outputs)
Removes speaker layouts which are not exposed in UI. The speaker
layouts selectable by users in the UI are the most common ones. It is
not necessary to keep other layouts. (This basically removes
5POINT1_SURROUND, 7POINT1_SURROUND, SURROUND =3.0).
Noise Suppression: Clamp sample values before converting to integer.
This fixes an issue where samples exceeding full scale would overflow,
resulting in heavy distortion.
Closesjp9000/obs-studio#1113
Fixes ticket 1070.
See also
https://obsproject.com/forum/threads/ffmpeg-recording.77378/#post-330473
(related bugs).
The ffmpeg constant AVFMT_RAWPICTURE was deprecated in october 2015
and marked for removal at avformat major bump to version 58
(ffmpeg commit 34ed5c2 , oct 12, 2015).
The bump occured with commit 69b5ce6 (oct21, 2017).
The constant was subsequently removed (commit 693a11b, oct 26 2017).
It was removed from obs-studio with commit d670d7b (from me).
But the code block which was executed with this constant was not
removed, causing issues with ffmpeg output.
The commit fixes the issue for old ffmpeg builds as well as new ones.
The constant is reintegrated for avformat major version < 58 and removed
for version >= 58 (along with its accompanying code).
Thanks to J Lowe for help in solving the bug.
(tested on win 10, macos 10.13, ubuntu 17.10 with ffmpeg head & ffmpeg
3.4.1)
If the target process re-creates its D3D context, the game capture tick
can trigger before the capture is setup, in which case OBS gets a
CAPTURE_RETRY message. However with the memory capture method, it
continues to try and copy from the shared memory pointer which is no
longer valid, resulting in a crash. The fix uses the old texture until
the next tick at which point the new capture should be ready for use.
Windows SDK 10.0.16299.0 defines these structures as part of winternl.h
but using different types and names. Unfortunately there's no macro to
detect the SDK version, so to avoid conflicting with newer / older SDKs
the OBS structs have been renamed.
When this was being fixed up, the incorrect function name was used --
however it still compiled because the author was using the newer FFmpeg
version at the time.
Common multi-channel setup is 5.1 and 7.1 with rear speakers.
Thus only setups that include SPEAKER_SIDE_LEFT and SPEAKER_SIDE_RIGHT
needs the marking as not common (or "side" use), while it stays the
true Side setup (with side speakers) by its internal meaning.
This "side" is named "surround" by Microsoft. To not confuse users and
translators, it is wise to use "Side" mark next to format name.
Certain NVIDIA GPUs don't support NVENC, but ship with the NVENC
library, causing OBS to mistakenly think that NVENC is available when it
actually isn't.
Closesjp9000/obs-studio#1087
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
When device timing is used, it shouldn't be modifying the timestamp.
Fixes an issue where certain devices with large audio segments would
seem a bit out of sync.
Lucian Poston
jp9000
Michel
Joel Bethke
Tjienta Vara
derrod
Matt Morrissette
pkviet
J.D. Purcell
Richard Stanway
SuslikV
James Hurley
Thomas Schnitzler
Ryan Foster
Alcaros
Fazrad Karkhani