The cutoff hack was added many, many years ago as recommended by
Konverter. Since then, there has been much work on the AAC encoder, so
this hack should no longer be necessary.
The windows media foundation H264 encoders have been deprecated for over
a year, and microsoft's media foundation AAC encoder has had a continued
issue with occasional random audio glitches. The FFmpeg AAC encoder has
had recent development, and is more than sufficient to be able to handle
the task of encoding in terms of both quality and performance, so it's
better just to use the FFmpeg encoder from here on out.
As this plugin is no longer needed, for the next year or two it'll still
be compiled and included, but as a blank plugin that does nothing. The
reason why it's still being included as a blank no-operation plugin is
to overwrite older versions of the plugin. That way if a user installs
a newer OBS version over an older one, it won't load up the older win-mf
plugin where the encoders still were enabled.
This also fixes some rarely reported media foundation crashes that can
happen on startup.
Fixes an issue where align_pos could be smaller than
sizeof(struct shmem_data), potentially overwriting memory of the header.
References jp9000/obs-studio#1202
(This commit also modifies the obs-ffmpeg module)
The default channel layouts from aac spec are implemented in FFmpeg
native aac encoder as follows:
AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
AV_CH_LAYOUT_SURROUND,
AV_CH_LAYOUT_4POINT0,
AV_CH_LAYOUT_5POINT0_BACK,
AV_CH_LAYOUT_5POINT1_BACK,
AV_CH_LAYOUT_7POINT1,
The correspondence of speaker layouts to AV_CH_LAYOUT from FFmpeg is
changed to reflect the previous table.
Although FFmpeg native aac encoder can now encode all the layouts listed
in avutil channel_layout.h (on master), there might be issues with older
FFmpeg binaries.
Note that 2.1 speaker layout will be encoded as AV_CH_LAYOUT_SURROUND
(FL FR FC) because it is not listed as the default layout for three
channels.
This just means some optimizations for LFE channel will not be used by
the encoder which will treat it as an SCE (single channel element).
Closesjp9000/obs-studio#1182
Adjusts the enc-amf submodule remote to the jp9000 fork, which tests AMF
support in a separate process before attempting to initialize AMF
in-process. Reduces the possibility of driver crashes caused by AMF
initialization.
The reason this code is being reverted/removed is because this code is a
risk to startup stability. This check will be added again in the
future, however this code needs to be executed from a secondary piped
process instead of directly in the process to reduce risk of
driver/hardware issues impacting program startup. The same needs to be
accomplished for the AMF plugin as well.
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 reverts commit 94b5982216f253f4f4d355109a6dcb81f3a3b980.
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).
