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#1 2019-09-19 13:31:26

nahush
Member
Registered: 2019-01-18
Posts: 157

Not able to record Sound in OBS Studio's video recording?

As I was trying to make tutorial for my self of the ardour 5.12 , wanted to record the whole thing with obs studio but the output of video does not have any audio whatsoever.In my earphone i am able to hear the sound properly,jack input sink in mixer of obs also shows movement.

I did tried all the methods such as :

1. Trying to connect the jack input sink from ardour audio connection settings.(As per this https://discourse.ardour.org/t/how-to-s … our/101667)

2.Using cadence and catia for the connection through jack.

3.Using the graph option given in qjackctl control of jack.

4. changing the tracks from the advance sound setting option from mixer cog in obs.

Things  already done by me to solve the problem.

**Output of the obs studio at start and stop recording.(obs version =24.0.0.r3.geee40ec6d at AUR and also was having problem with the stable version of obs 23.0)

[narch-music@archmusic ~]$ obs

(process:20517): Gtk-WARNING **: 13:03:51.823: Locale not supported by C library.
	Using the fallback 'C' locale.
Attempted path: share/obs/obs-studio/locale/en-US.ini
Attempted path: /usr/share/obs/obs-studio/locale/en-US.ini
Attempted path: share/obs/obs-studio/themes/System.qss
Attempted path: /usr/share/obs/obs-studio/themes/System.qss
info: CPU Name: Intel(R) Core(TM) i3-2365M CPU @ 1.40GHz
info: CPU Speed: 1033.355MHz
info: Physical Cores: 2, Logical Cores: 4
info: Physical Memory: 3840MB Total, 755MB Free
info: Kernel Version: Linux 5.2.14-arch2-1-ARCH
info: Distribution: "Arch Linux" Unknown
info: Window System: X11.0, Vendor: The X.Org Foundation, Version: 1.20.5
info: Portable mode: false
Attempted path: share/obs/obs-studio/themes/Dark/no_sources.svg
Attempted path: /usr/share/obs/obs-studio/themes/Dark/no_sources.svg
QMetaObject::connectSlotsByName: No matching signal for on_advAudioProps_clicked()
QMetaObject::connectSlotsByName: No matching signal for on_advAudioProps_destroyed()
QMetaObject::connectSlotsByName: No matching signal for on_program_customContextMenuRequested(QPoint)
info: OBS 24.0.0.r3.geee40ec6d (linux)
info: ---------------------------------
info: ---------------------------------
info: audio settings reset:
	samples per sec: 44100
	speakers:        2
info: ---------------------------------
info: Initializing OpenGL...
info: Loading up OpenGL on adapter Intel Open Source Technology Center Mesa DRI Intel(R) Sandybridge Mobile 
info: OpenGL loaded successfully, version 3.3 (Core Profile) Mesa 19.1.7, shading language 3.30
info: ---------------------------------
info: video settings reset:
	base resolution:   1280x720
	output resolution: 1280x720
	downscale filter:  Bicubic
	fps:               30/1
	format:            NV12
	YUV mode:          601/Partial
info: NV12 texture support not available
info: Audio monitoring device:
	name: Default
	id: default
info: ---------------------------------
warning: Failed to load 'en-US' text for module: 'decklink-ouput-ui.so'
libDeckLinkAPI.so: cannot open shared object file: No such file or directory
info: No blackmagic support
error: os_dlopen(libnvidia-encode.so.1->libnvidia-encode.so.1): libnvidia-encode.so.1: cannot open shared object file: No such file or directory

info: FFMPEG VAAPI supported
info: VLC found, VLC video source enabled
info: ---------------------------------
info:   Loaded Modules:
info:     vlc-video.so
info:     text-freetype2.so
info:     rtmp-services.so
info:     obs-x264.so
info:     obs-transitions.so
info:     obs-outputs.so
info:     obs-libfdk.so
info:     obs-filters.so
info:     obs-ffmpeg.so
info:     linux-v4l2.so
info:     linux-pulseaudio.so
info:     linux-jack.so
info:     linux-decklink.so
info:     linux-capture.so
info:     linux-alsa.so
info:     image-source.so
info:     frontend-tools.so
info:     decklink-ouput-ui.so
info: ---------------------------------
info: ==== Startup complete ===============================================
error: Service '' not found
info: All scene data cleared
info: ------------------------------------------------
info: pulse-input: Server name: 'pulseaudio 13.0-dirty'
info: pulse-input: Audio format: s32le, 44100 Hz, 2 channels
info: pulse-input: Started recording from 'alsa_output.pci-0000_00_1b.0.analog-stereo.monitor'
info: pulse-input: Server name: 'pulseaudio 13.0-dirty'
info: pulse-input: Audio format: s32le, 44100 Hz, 2 channels
info: pulse-input: Started recording from 'alsa_output.pci-0000_00_1b.0.analog-stereo.monitor'
info: pulse-input: Server name: 'pulseaudio 13.0-dirty'
info: pulse-input: Audio format: s32le, 44100 Hz, 2 channels
info: pulse-input: Started recording from 'alsa_input.pci-0000_00_1b.0.analog-stereo'
Jack: JackClient::SetupDriverSync driver sem in flush mode
Jack: JackLinuxFutex::Connect name = jack_sem.1002_default_JACK Input Client
Jack: Clock source : system clock via clock_gettime
Jack: JackLibClient::Open name = JACK Input Client refnum = 2
Jack: JackClient::PortRegister ref = 2 name = JACK Input Client:in_1 type = 32 bit float mono audio port_index = 5
Jack: JackClient::PortRegister ref = 2 name = JACK Input Client:in_2 type = 32 bit float mono audio port_index = 6
Jack: JackClient::Activate
Jack: JackPosixThread::StartImp : create non RT thread
Jack: JackPosixThread::ThreadHandler : start
Jack: JackClient::kBufferSizeCallback buffer_size = 1024
Jack: JackClient::Init : period = 23219 computation = 100 constraint = 23219
Jack: JackPosixThread::AcquireRealTimeImp priority = 5
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 2
Jack: JackClient::kActivateClient name = JACK Input Client ref = 2 
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
info: xshm-input: Geometry 1280x720 @ 0,0
info: Switched to scene 'Ardou3'
info: ------------------------------------------------
info: Loaded scenes:
info: - scene 'Ardour':
info: - scene 'Ardou3':
info:     - source: 'Screen Capture (XSHM)' (xshm_input)
info:     - source: 'JACK Input Client' (jack_output_capture)
info: ------------------------------------------------
Attempted path: share/obs/obs-studio/images/overflow.png
Attempted path: /usr/share/obs/obs-studio/images/overflow.png
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
Jack: JackClient::ClientNotify ref = 2 name = JACK Input Client notify = 18
info: ---------------------------------
info: [x264 encoder: 'streaming_h264'] preset: veryfast
info: [x264 encoder: 'streaming_h264'] settings:
	rate_control: CBR
	bitrate:      2500
	buffer size:  2500
	crf:          0
	fps_num:      30
	fps_den:      1
	width:        1280
	height:       720
	keyint:       250

info: libfdk_aac encoder created
info: libfdk_aac bitrate: 160, channels: 2
info: libfdk_aac encoder created
info: libfdk_aac bitrate: 160, channels: 2
info: libfdk_aac encoder created
info: libfdk_aac bitrate: 160, channels: 2
info: libfdk_aac encoder created
info: libfdk_aac bitrate: 160, channels: 2
info: ==== Recording Start ===============================================
info: [ffmpeg muxer: 'adv_file_output'] Writing file '/home/narch-music/OBS-Studio/2019-09-19_13-04-38.mkv'...
[matroska @ 0x555bb1144740] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[matroska @ 0x555bb1144740] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[matroska @ 0x555bb1144740] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[matroska @ 0x555bb1144740] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
[matroska @ 0x555bb1144740] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
info: [ffmpeg muxer: 'adv_file_output'] Output of file '/home/narch-music/OBS-Studio/2019-09-19_13-04-38.mkv' stopped
info: Output 'adv_file_output': stopping
info: Output 'adv_file_output': Total frames output: 802
info: Output 'adv_file_output': Total drawn frames: 822 (824 attempted)
info: Output 'adv_file_output': Number of lagged frames due to rendering lag/stalls: 2 (0.2%)
info: ==== Recording Stop ================================================
info: libfdk_aac encoder destroyed
info: libfdk_aac encoder destroyed
info: libfdk_aac encoder destroyed
info: libfdk_aac encoder destroyed

**Output of the commands mentioned on this arch forum  https://bbs.archlinux.org/viewtopic.php?id=247499  while recording video with obs.as this frum dosnt have the solution mentioned but the output was important in terms of checking the problem with pusleaudio side of things.

[narch-music@archmusic ~]$ pacmd list-sinks                      
1 sink(s) available.
  * index: 0
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY FLAT_VOLUME DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: APPLICATION
	priority: 9039
	volume: front-left: 68813 / 105% / 1.27 dB,   front-right: 68813 / 105% / 1.27 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max request: 0 KiB
	max rewind: 0 KiB
	monitor source: 0
	sample spec: s32le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 0
	linked by: 2
	configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
	card: 0 <alsa_card.pci-0000_00_1b.0>
	module: 6
	properties:
		alsa.resolution_bits = "32"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC269VC Analog"
		alsa.id = "ALC269VC Analog"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "0"
		alsa.card_name = "HDA Intel PCH"
		alsa.long_card_name = "HDA Intel PCH at 0xd0610000 irq 31"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "705600"
		device.buffering.fragment_size = "352800"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Analog Stereo"
		device.description = "Built-in Audio Analog Stereo"
		alsa.mixer_name = "Realtek ALC269VC"
		alsa.components = "HDA:10ec0269,17aac02a,00100202 HDA:80862806,80860101,00100000"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
	ports:
		analog-output-speaker: Speakers (priority 10000, latency offset 0 usec, available: no)
			properties:
				device.icon_name = "audio-speakers"
		analog-output-headphones: Headphones (priority 9900, latency offset 0 usec, available: yes)
			properties:
				device.icon_name = "audio-headphones"
	active port: <analog-output-headphones>

[narch-music@archmusic ~]$ pacmd list-sink-inputs
0 sink input(s) available.

[narch-music@archmusic ~]$ pacmd list-sources
2 source(s) available.
    index: 0
	name: <alsa_output.pci-0000_00_1b.0.analog-stereo.monitor>
	driver: <module-alsa-card.c>
	flags: DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: APPLICATION
	priority: 1030
	volume: front-left: 65536 / 100% / 0.00 dB,   front-right: 65536 / 100% / 0.00 dB
	        balance 0.00
	base volume: 65536 / 100% / 0.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s32le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 2
	linked by: 2
	configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
	monitor_of: 0
	card: 0 <alsa_card.pci-0000_00_1b.0>
	module: 6
	properties:
		device.description = "Monitor of Built-in Audio Analog Stereo"
		device.class = "monitor"
		alsa.card = "0"
		alsa.card_name = "HDA Intel PCH"
		alsa.long_card_name = "HDA Intel PCH at 0xd0610000 irq 31"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller"
		device.form_factor = "internal"
		device.string = "0"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
  * index: 1
	name: <alsa_input.pci-0000_00_1b.0.analog-stereo>
	driver: <module-alsa-card.c>
	flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
	state: SUSPENDED
	suspend cause: APPLICATION
	priority: 9039
	volume: front-left: 65540 / 100% / 0.00 dB,   front-right: 65540 / 100% / 0.00 dB
	        balance 0.00
	base volume: 6554 /  10% / -60.00 dB
	volume steps: 65537
	muted: no
	current latency: 0.00 ms
	max rewind: 0 KiB
	sample spec: s32le 2ch 44100Hz
	channel map: front-left,front-right
	             Stereo
	used by: 1
	linked by: 1
	configured latency: 0.00 ms; range is 0.50 .. 2000.00 ms
	card: 0 <alsa_card.pci-0000_00_1b.0>
	module: 6
	properties:
		alsa.resolution_bits = "32"
		device.api = "alsa"
		device.class = "sound"
		alsa.class = "generic"
		alsa.subclass = "generic-mix"
		alsa.name = "ALC269VC Analog"
		alsa.id = "ALC269VC Analog"
		alsa.subdevice = "0"
		alsa.subdevice_name = "subdevice #0"
		alsa.device = "0"
		alsa.card = "0"
		alsa.card_name = "HDA Intel PCH"
		alsa.long_card_name = "HDA Intel PCH at 0xd0610000 irq 31"
		alsa.driver_name = "snd_hda_intel"
		device.bus_path = "pci-0000:00:1b.0"
		sysfs.path = "/devices/pci0000:00/0000:00:1b.0/sound/card0"
		device.bus = "pci"
		device.vendor.id = "8086"
		device.vendor.name = "Intel Corporation"
		device.product.id = "1e20"
		device.product.name = "7 Series/C216 Chipset Family High Definition Audio Controller"
		device.form_factor = "internal"
		device.string = "front:0"
		device.buffering.buffer_size = "705600"
		device.buffering.fragment_size = "352800"
		device.access_mode = "mmap+timer"
		device.profile.name = "analog-stereo"
		device.profile.description = "Analog Stereo"
		device.description = "Built-in Audio Analog Stereo"
		alsa.mixer_name = "Realtek ALC269VC"
		alsa.components = "HDA:10ec0269,17aac02a,00100202 HDA:80862806,80860101,00100000"
		module-udev-detect.discovered = "1"
		device.icon_name = "audio-card-pci"
	ports:
		analog-input-internal-mic: Internal Microphone (priority 8900, latency offset 0 usec, available: no)
			properties:
				device.icon_name = "audio-input-microphone"
		analog-input-mic: Microphone (priority 8700, latency offset 0 usec, available: yes)
			properties:
				device.icon_name = "audio-input-microphone"
	active port: <analog-input-mic>
[narch-music@archmusic ~]$

**Change the default sample rate in /etc/pulse/deamon.conf to 48000 ,avoid -resampling=yes (not commented out as they have changed.) as told in this help page of pulseaudio  https://wiki.archlinux.org/index.php/Pu … PulseAudio

  GNU nano 4.4                                        /etc/pulse/daemon.conf                                        Modified  
# This file is part of PulseAudio.
#
# PulseAudio is free software; you can redistribute it and/or modify
# it under the terms of the GNU Lesser General Public License as published by
# the Free Software Foundation; either version 2 of the License, or
# (at your option) any later version.
#
# PulseAudio is distributed in the hope that it will be useful, but
# WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
# General Public License for more details.
#
# You should have received a copy of the GNU Lesser General Public License
# along with PulseAudio; if not, see <http://www.gnu.org/licenses/>.

## Configuration file for the PulseAudio daemon. See pulse-daemon.conf(5) for
## more information. Default values are commented out.  Use either ; or # for
## commenting.

; daemonize = no
; fail = yes
; allow-module-loading = yes
; allow-exit = yes
; use-pid-file = yes
; system-instance = no
; local-server-type = user
; enable-shm = yes
; enable-memfd = yes
; shm-size-bytes = 0 # setting this 0 will use the system-default, usually 64 MiB
; lock-memory = no
; cpu-limit = no

; high-priority = yes
; nice-level = -11

; realtime-scheduling = yes
; realtime-priority = 5

; exit-idle-time = 20
; scache-idle-time = 20

; dl-search-path = (depends on architecture)

; load-default-script-file = yes
; default-script-file = /etc/pulse/default.pa

; log-target = auto
; log-level = notice
; log-meta = no
; log-time = no
; log-backtrace = 0

; resample-method = speex-float-1
 avoid-resampling = yes
; enable-remixing = yes
; remixing-use-all-sink-channels = yes
; enable-lfe-remixing = no
; lfe-crossover-freq = 0

flat-volumes = no
; flat-volumes = yes

; rlimit-fsize = -1
; rlimit-data = -1
; rlimit-stack = -1
; rlimit-core = -1
; rlimit-as = -1
; rlimit-rss = -1
; rlimit-nproc = -1
; rlimit-nofile = 256
; rlimit-memlock = -1
; rlimit-locks = -1
; rlimit-sigpending = -1
; rlimit-msgqueue = -1
; rlimit-nice = 31
; rlimit-rtprio = 9
; rlimit-rttime = 200000

; default-sample-format = s16le
 default-sample-rate = 48000
; alternate-sample-rate = 48000
; default-sample-channels = 2
; default-channel-map = front-left,front-right

; default-fragments = 4
; default-fragment-size-msec = 25

; enable-deferred-volume = yes
; deferred-volume-safety-margin-usec = 8000
; deferred-volume-extra-delay-usec = 0

need anything else from my part do let me know.

Offline

#2 2019-09-19 20:37:07

Maniaxx
Member
Registered: 2014-05-14
Posts: 738

Re: Not able to record Sound in OBS Studio's video recording?

As a workaround try a loopback device (virtual sound card), add a connection in Ardour and record from that device in obs or ffmpeg.

modprobe snd-aloop

https://alsa-project.org/wiki/Matrix:Module-aloop

Example:
- Load the module:

modprobe snd-aloop

- Create some ALSA devices in ~/.asoundrc:

pcm.ploop {
  type plug
  slave.pcm "hw:Loopback,0,0"
}

pcm.cloop {
  type plug
  slave.pcm "hw:Loopback,1,0"
}

- Create an ALSA/JACK bridge (needs to be re-created on reboot):

$ setsid alsa_out -j ploop -d ploop &

- Close/re-open any audio apps to register the changes above
- Connect main channel to 'ploop' in Ardour (see video).
- Record 'cloop' device outside of Ardour (e.g. with ffmpeg or obs)

The video was recorded with ffmpeg (intermediate file/lossless):

$ ffmpeg -y -thread_queue_size 512 -f alsa -i cloop -video_size 1600x900 -show_region 1 -framerate 60 -f x11grab -i :0.0+0,150 -draw_mouse 1 -c:v libx264rgb -preset veryfast -crf 0 -pix_fmt rgb24 -c:a pcm_s16le /tmp/capture.mkv

Edit: Actually audio recording in this example is 16bit only (not lossless as Jack is 32bit). You can raise ffmpeg audio recording precision to 32bit as well if you like (for the price of a slightly larger intermediate file) with 'pcm_s16le' -> 'pcm_s32le' for true lossless recording. Opus might benefit from that as well.

Final compression (x264-rgb24 lossy, Opus audio, trimmed 00:04-01:46):

ffmpeg -y -i /tmp/capture.mkv -ss 00:04 -to 01:46 -c:v libx264rgb -preset veryslow -crf 24 -tune film -level 4.2 -c:a libopus -b:a 128k /tmp/capture-rgb24-crf24.mkv

Beware that libx264rgb is a somewhat exotic codec. Maybe not all devices might be able to play this files. But for screen recordings its a much better choice than the usual x264 (YUV4:2:0) compression that would involve color space conversion and degraded color precision by factor 4.

Last edited by Maniaxx (2019-09-19 23:18:42)

sys2064

Offline

#3 2019-09-21 18:49:43

nahush
Member
Registered: 2019-01-18
Posts: 157

Re: Not able to record Sound in OBS Studio's video recording?

Maniaxx wrote:

As a workaround try a loopback device (virtual sound card), add a connection in Ardour and record from that device in obs or ffmpeg.

modprobe snd-aloop

https://alsa-project.org/wiki/Matrix:Module-aloop

Example:
- Load the module:

modprobe snd-aloop

- Create some ALSA devices in ~/.asoundrc:

pcm.ploop {
  type plug
  slave.pcm "hw:Loopback,0,0"
}

pcm.cloop {
  type plug
  slave.pcm "hw:Loopback,1,0"
}

- Create an ALSA/JACK bridge (needs to be re-created on reboot):

$ setsid alsa_out -j ploop -d ploop &

- Close/re-open any audio apps to register the changes above
- Connect main channel to 'ploop' in Ardour (see video).
- Record 'cloop' device outside of Ardour (e.g. with ffmpeg or obs)

The video was recorded with ffmpeg (intermediate file/lossless):

$ ffmpeg -y -thread_queue_size 512 -f alsa -i cloop -video_size 1600x900 -show_region 1 -framerate 60 -f x11grab -i :0.0+0,150 -draw_mouse 1 -c:v libx264rgb -preset veryfast -crf 0 -pix_fmt rgb24 -c:a pcm_s16le /tmp/capture.mkv

Edit: Actually audio recording in this example is 16bit only (not lossless as Jack is 32bit). You can raise ffmpeg audio recording precision to 32bit as well if you like (for the price of a slightly larger intermediate file) with 'pcm_s16le' -> 'pcm_s32le' for true lossless recording. Opus might benefit from that as well.

Final compression (x264-rgb24 lossy, Opus audio, trimmed 00:04-01:46):

ffmpeg -y -i /tmp/capture.mkv -ss 00:04 -to 01:46 -c:v libx264rgb -preset veryslow -crf 24 -tune film -level 4.2 -c:a libopus -b:a 128k /tmp/capture-rgb24-crf24.mkv

Beware that libx264rgb is a somewhat exotic codec. Maybe not all devices might be able to play this files. But for screen recordings its a much better choice than the usual x264 (YUV4:2:0) compression that would involve color space conversion and degraded color precision by factor 4.


i will try it and confirm.thanks for reply.

Offline

#4 2020-08-03 14:42:34

lockywolf
Member
Registered: 2020-08-03
Posts: 1

Re: Not able to record Sound in OBS Studio's video recording?

Did you get it sorted out?

UPD: I actually did it myself, had to rebuild ffmpeg with --enable-libfdk-aac and --enable-nonfree

Last edited by lockywolf (2020-08-04 01:57:49)

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