You are not logged in.
the index should not change should have added check with
pacmd stat
But it may well have been using that before and I made a bad assumption based on the list order from pacmd list-sinks
Sorry I am out of useful suggestions
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Well, i followed some instructions and edited /etc/pulse/default.pa (only this line)
load-module module-stream-restore restore_device=false
so after logging out pulseaudio was not starting after reboot too. So, when i play some tune with audacious and alsa as output it is working.
Does this lead us to the conclusion that it's pulseaudio related or not?
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http://www.freedesktop.org/wiki/Softwar … ultDevice/
As you changed only pulse's configuration and now it works not sure how you could conclude other than it was a pulse configuration issue.
You could try the alternative methods mentioned in the link if you want to revert that change so as to keep an unmodified /etc/pulse/default.pa
Glad you have working audio
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http://www.freedesktop.org/wiki/Softwar … ultDevice/
As you changed only pulse's configuration and now it works not sure how you could conclude other than it was a pulse configuration issue.
You could try the alternative methods mentioned in the link if you want to revert that change so as to keep an unmodified /etc/pulse/default.pa
Glad you have working audio
I have working audio without pulseaudio. Pulseaudio does not run at all , and this is the reason I Have working audio through alsa. When pulseaudio was running i could not have audio work with alsa. As i Mentioned above, i want to configure alsa to pulseaudio and after that to jack., the way it was before the upgrade. I have seen that pulseaudio indeed upgraded to 7.0.0. So, now i have working audio WITHOUT pulseaudio, so the problem is not solved.
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What output do you get if you try to start pulseaudio from a terminal?
There is nothing to go on at the moment other than that config change stopped pulse audio starting?
Also does reverting the change allow pulse to start?
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What output do you get if you try to start pulseaudio from a terminal?
There is nothing to go on at the moment other than that config change stopped pulse audio starting?
Also does reverting the change allow pulse to start?
zab ~ $ pulseaudio --start
E: [pulseaudio] main.c: Daemon startup failed.
No, reverting the change does not allow pulse to start. Strange ah?
Last edited by zabrielza (2015-10-24 19:13:31)
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I had found this
Running
Since version 7.0, PulseAudio on Arch uses socket activation. By default, pulseaudio.socket is enabled for the systemd/User instance.
Note:To disable pulseaudio.socket, make sure that $XDG_CONFIG_HOME/systemd/user/ exists and run systemctl --user mask pulseaudio.socket.
Many desktop environments autostart programs based on desktop files in the /etc/xdg/autostart/ directory. In this case, PulseAudio will be launched automatically regardless of the socket activation status.
in the pulseaudio archwiki.
When i tried to mask pulseaudio /systemd/user did not exist.
zab ~ $ sudo systemctl --user mask pulseaudio
[sudo] password for zab:
Failed to connect to bus: No such file or directory
Also, I have to mention that I had followed priot to the upgrade, this , similar to the pulseaudio-alsa package etc
Last edited by zabrielza (2015-10-24 19:25:55)
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Actually, reverting changes started pulseaudio. My mistake, I apologize. But, it blocks alsa again. Same error.
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I had found this
Running
Since version 7.0, PulseAudio on Arch uses socket activation. By default, pulseaudio.socket is enabled for the systemd/User instance.
Note:To disable pulseaudio.socket, make sure that $XDG_CONFIG_HOME/systemd/user/ exists and run systemctl --user mask pulseaudio.socket.
Many desktop environments autostart programs based on desktop files in the /etc/xdg/autostart/ directory. In this case, PulseAudio will be launched automatically regardless of the socket activation status.in the pulseaudio archwiki.
When i tried to mask pulseaudio /systemd/user did not exist.
zab ~ $ sudo systemctl --user mask pulseaudio [sudo] password for zab: Failed to connect to bus: No such file or directory
Also, I have to mention that I had followed priot to the upgrade, this , similar to the pulseaudio-alsa package etc
Almost certainly the sudo is stopping that command from working.
Why is pulse blocking applications from using the alsa user space interface an issue as the audio streams would be routed through pulse to the device correct?
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Well, I am afraid it is not sudo related.
zab ~ $ systemctl --user mask pulseaudio
Failed to execute operation: No such file or directory
Yes, audio should be routed from alsa to pulseaudio. I think that It is related to masking pulseaudio, so maybe if we solve this the sound configuration will be solved too.
make sure that $XDG_CONFIG_HOME/systemd/user/ exists
How can I check if this exists? And if not HOw can I create one?
Last edited by zabrielza (2015-10-24 20:12:01)
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systemctl --user mask pulseaudio.socket
Note this will only stop socket based startup not start if its spawned by kde etc.
Have you tried using any other audio player than audacious to try and send sound to pulse and see if it produces output from the soundcard?
Its also strange about pulse failing at startup with the config change might try adding -vvv to see if more verbose output is helpful
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zab ~ $ systemctl --user mask pulseaudio.socket
Failed to execute operation: No such file or directory
zab ~ $ sudo systemctl --user mask pulseaudio.socket
[sudo] password for zab:
Failed to connect to bus: No such file or directory
Same thing with vlc.
Also, now with restore_device=false pulseaudio starts but same problems with sound. I am convinced that if we fix maskin pulseaudio the problem will be there
Last edited by zabrielza (2015-10-24 20:33:49)
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systemctl mask pulseaudio.socket
sudo maybe needed if you do not have a policy kit agent
checked that command definitely works for me without --user and errors with --user
edit:
missing and
Last edited by loqs (2015-10-24 21:38:43)
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systemctl mask pulseaudio.socket
sudo maybe needed if you do not have a policy kit agent
checked that command definitely works for me without --user and errors with --user
edit:
missing and
well, masking pulseaudio succeeded. But, unfortunatly the error is still there.
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pacmd info
with all music players stopped there should be
.... sink(s) available.
index: .....
....
state: SUSPENDED
....
0 sink input(s) available.
.... signifies skipped output
pacmd info
with a music player playing
.... sink(s) available.
index: X
....
state: RUNNING
....
1 sink input(s) available.
index: ....
driver: ....
flags: ....
state: RUNNING
sink: X
Might be simpler to get the same info from pavucontrol
This should show if pulse is receiving the stream and where it is routing it to.
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Yes, it's as you said it should be
4 sink(s) available.
index: 0
name: <alsa_output.pci-0000_01_05.1.hdmi-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9050
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 request: 0 KiB
max rewind: 0 KiB
monitor source: 0
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
card: 0 <alsa_card.pci-0000_01_05.1>
module: 6
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "HDMI 0"
alsa.id = "HDMI 0"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "3"
alsa.card = "2"
alsa.card_name = "HDA ATI HDMI"
alsa.long_card_name = "HDA ATI HDMI at 0xfdafc000 irq 19"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:01:05.1"
sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:05.1/sound/card2"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "970f"
device.product.name = "RS880 HDMI Audio [Radeon HD 4200 Series]"
device.string = "hdmi:2"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
device.access_mode = "mmap+timer"
device.profile.name = "hdmi-stereo"
device.profile.description = "Digital Stereo (HDMI)"
device.description = "RS880 HDMI Audio [Radeon HD 4200 Series] Digital Stereo (HDMI)"
alsa.mixer_name = "ATI RS690/780 HDMI"
alsa.components = "HDA:1002791a,00791a00,00100000"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
active port: <hdmi-output-0>
index: 1
name: <alsa_output.pci-0000_02_00.1.hdmi-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9050
volume: front-left: 65534 / 100% / -0.00 dB, front-right: 65534 / 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 request: 0 KiB
max rewind: 0 KiB
monitor source: 1
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
card: 1 <alsa_card.pci-0000_02_00.1>
module: 7
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "HDMI 0"
alsa.id = "HDMI 0"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "3"
alsa.card = "3"
alsa.card_name = "HDA ATI HDMI"
alsa.long_card_name = "HDA ATI HDMI at 0xfdffc000 irq 27"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:02:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:02.0/0000:02:00.1/sound/card3"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "aa30"
device.product.name = "RV770 HDMI Audio [Radeon HD 4850/4870] (Radeon HD 4850 512MB GDDR3 PCI-E Dual Slot Fansink)"
device.string = "hdmi:3"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
device.access_mode = "mmap+timer"
device.profile.name = "hdmi-stereo"
device.profile.description = "Digital Stereo (HDMI)"
device.description = "RV770 HDMI Audio [Radeon HD 4850/4870] (Radeon HD 4850 512MB GDDR3 PCI-E Dual Slot Fansink) Digital Stereo (HDMI)"
alsa.mixer_name = "ATI R6xx HDMI"
alsa.components = "HDA:1002aa01,00aa0100,00100100"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
active port: <hdmi-output-0>
* index: 2
name: <alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9049
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 request: 0 KiB
max rewind: 0 KiB
monitor source: 2
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
fixed latency: 99.95 ms
card: 2 <alsa_card.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00>
module: 8
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "USB Audio CODEC"
alsa.long_card_name = "Burr-Brown from TI USB Audio CODEC at usb-0000:00:12.1-2, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:12.1-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:12.1/usb4/4-2/4-2:1.0/sound/card0"
udev.id = "usb-Burr-Brown_from_TI_USB_Audio_CODEC-00"
device.bus = "usb"
device.vendor.id = "08bb"
device.vendor.name = "Texas Instruments"
device.product.id = "2902"
device.product.name = "PCM2902 Audio Codec"
device.serial = "Burr-Brown_from_TI_USB_Audio_CODEC"
device.string = "front:0"
device.buffering.buffer_size = "17632"
device.buffering.fragment_size = "4408"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "PCM2902 Audio Codec Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USB08bb:2902"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
ports:
analog-output: Analog Output (priority 9900, latency offset 0 usec, available: unknown)
properties:
active port: <analog-output>
index: 3
name: <alsa_output.pci-0000_00_14.2.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9959
volume: front-left: 30419 / 46% / -20.00 dB, front-right: 30419 / 46% / -20.00 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: 4
sample spec: s16le 2ch 48000Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1837.33 ms
card: 3 <alsa_card.pci-0000_00_14.2>
module: 9
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC889A Analog"
alsa.id = "ALC889A Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "HDA ATI SB"
alsa.long_card_name = "HDA ATI SB at 0xfe024000 irq 16"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:14.2"
sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card1"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "4383"
device.product.name = "SBx00 Azalia (Intel HDA)"
device.form_factor = "internal"
device.string = "front:1"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
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 ALC889A"
alsa.components = "HDA:10ec0885,1458a102,00100101"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-output-lineout: Line Out (priority 9900, latency offset 0 usec, available: no)
properties:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: no)
properties:
device.icon_name = "audio-headphones"
active port: <analog-output-lineout>
4 sink(s) available.
index: 0
name: <alsa_output.pci-0000_01_05.1.hdmi-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9050
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 request: 0 KiB
max rewind: 0 KiB
monitor source: 0
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
card: 0 <alsa_card.pci-0000_01_05.1>
module: 6
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "HDMI 0"
alsa.id = "HDMI 0"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "3"
alsa.card = "2"
alsa.card_name = "HDA ATI HDMI"
alsa.long_card_name = "HDA ATI HDMI at 0xfdafc000 irq 19"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:01:05.1"
sysfs.path = "/devices/pci0000:00/0000:00:01.0/0000:01:05.1/sound/card2"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "970f"
device.product.name = "RS880 HDMI Audio [Radeon HD 4200 Series]"
device.string = "hdmi:2"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
device.access_mode = "mmap+timer"
device.profile.name = "hdmi-stereo"
device.profile.description = "Digital Stereo (HDMI)"
device.description = "RS880 HDMI Audio [Radeon HD 4200 Series] Digital Stereo (HDMI)"
alsa.mixer_name = "ATI RS690/780 HDMI"
alsa.components = "HDA:1002791a,00791a00,00100000"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
active port: <hdmi-output-0>
index: 1
name: <alsa_output.pci-0000_02_00.1.hdmi-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9050
volume: front-left: 65534 / 100% / -0.00 dB, front-right: 65534 / 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 request: 0 KiB
max rewind: 0 KiB
monitor source: 1
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
configured latency: 0.00 ms; range is 0.50 .. 1999.82 ms
card: 1 <alsa_card.pci-0000_02_00.1>
module: 7
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "HDMI 0"
alsa.id = "HDMI 0"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "3"
alsa.card = "3"
alsa.card_name = "HDA ATI HDMI"
alsa.long_card_name = "HDA ATI HDMI at 0xfdffc000 irq 27"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:02:00.1"
sysfs.path = "/devices/pci0000:00/0000:00:02.0/0000:02:00.1/sound/card3"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "aa30"
device.product.name = "RV770 HDMI Audio [Radeon HD 4850/4870] (Radeon HD 4850 512MB GDDR3 PCI-E Dual Slot Fansink)"
device.string = "hdmi:3"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
device.access_mode = "mmap+timer"
device.profile.name = "hdmi-stereo"
device.profile.description = "Digital Stereo (HDMI)"
device.description = "RV770 HDMI Audio [Radeon HD 4850/4870] (Radeon HD 4850 512MB GDDR3 PCI-E Dual Slot Fansink) Digital Stereo (HDMI)"
alsa.mixer_name = "ATI R6xx HDMI"
alsa.components = "HDA:1002aa01,00aa0100,00100100"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
hdmi-output-0: HDMI / DisplayPort (priority 5900, latency offset 0 usec, available: no)
properties:
device.icon_name = "video-display"
active port: <hdmi-output-0>
* index: 2
name: <alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY
state: SUSPENDED
suspend cause: IDLE
priority: 9049
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 request: 0 KiB
max rewind: 0 KiB
monitor source: 2
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 0
linked by: 0
fixed latency: 99.95 ms
card: 2 <alsa_card.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00>
module: 8
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "USB Audio"
alsa.id = "USB Audio"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "0"
alsa.card_name = "USB Audio CODEC"
alsa.long_card_name = "Burr-Brown from TI USB Audio CODEC at usb-0000:00:12.1-2, full speed"
alsa.driver_name = "snd_usb_audio"
device.bus_path = "pci-0000:00:12.1-usb-0:2:1.0"
sysfs.path = "/devices/pci0000:00/0000:00:12.1/usb4/4-2/4-2:1.0/sound/card0"
udev.id = "usb-Burr-Brown_from_TI_USB_Audio_CODEC-00"
device.bus = "usb"
device.vendor.id = "08bb"
device.vendor.name = "Texas Instruments"
device.product.id = "2902"
device.product.name = "PCM2902 Audio Codec"
device.serial = "Burr-Brown_from_TI_USB_Audio_CODEC"
device.string = "front:0"
device.buffering.buffer_size = "17632"
device.buffering.fragment_size = "4408"
device.access_mode = "mmap"
device.profile.name = "analog-stereo"
device.profile.description = "Analog Stereo"
device.description = "PCM2902 Audio Codec Analog Stereo"
alsa.mixer_name = "USB Mixer"
alsa.components = "USB08bb:2902"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-usb"
ports:
analog-output: Analog Output (priority 9900, latency offset 0 usec, available: unknown)
properties:
active port: <analog-output>
index: 3
name: <alsa_output.pci-0000_00_14.2.analog-stereo>
driver: <module-alsa-card.c>
flags: HARDWARE HW_MUTE_CTRL HW_VOLUME_CTRL DECIBEL_VOLUME LATENCY DYNAMIC_LATENCY
state: RUNNING
suspend cause:
priority: 9959
volume: front-left: 30419 / 46% / -20.00 dB, front-right: 30419 / 46% / -20.00 dB
balance 0.00
base volume: 65536 / 100% / 0.00 dB
volume steps: 65537
muted: no
current latency: 458.47 ms
max request: 79 KiB
max rewind: 344 KiB
monitor source: 4
sample spec: s16le 2ch 44100Hz
channel map: front-left,front-right
Stereo
used by: 1
linked by: 1
configured latency: 458.87 ms; range is 0.50 .. 1999.82 ms
card: 3 <alsa_card.pci-0000_00_14.2>
module: 9
properties:
alsa.resolution_bits = "16"
device.api = "alsa"
device.class = "sound"
alsa.class = "generic"
alsa.subclass = "generic-mix"
alsa.name = "ALC889A Analog"
alsa.id = "ALC889A Analog"
alsa.subdevice = "0"
alsa.subdevice_name = "subdevice #0"
alsa.device = "0"
alsa.card = "1"
alsa.card_name = "HDA ATI SB"
alsa.long_card_name = "HDA ATI SB at 0xfe024000 irq 16"
alsa.driver_name = "snd_hda_intel"
device.bus_path = "pci-0000:00:14.2"
sysfs.path = "/devices/pci0000:00/0000:00:14.2/sound/card1"
device.bus = "pci"
device.vendor.id = "1002"
device.vendor.name = "Advanced Micro Devices, Inc. [AMD/ATI]"
device.product.id = "4383"
device.product.name = "SBx00 Azalia (Intel HDA)"
device.form_factor = "internal"
device.string = "front:1"
device.buffering.buffer_size = "352768"
device.buffering.fragment_size = "176384"
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 ALC889A"
alsa.components = "HDA:10ec0885,1458a102,00100101"
module-udev-detect.discovered = "1"
device.icon_name = "audio-card-pci"
ports:
analog-output-lineout: Line Out (priority 9900, latency offset 0 usec, available: no)
properties:
analog-output-headphones: Headphones (priority 9000, latency offset 0 usec, available: no)
properties:
device.icon_name = "audio-headphones"
active port: <analog-output-lineout>
So, pulseaudio outputs from index 3, right?
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So, pulseaudio outputs from index 3, right?
Yes and as I understand it you want it to output from index 2.
Can you change that using pavucontrol?
It should also be possible from the console using
pacmd move-sink-input index, sink
but I am unsure of what values you should substitute for index and sink my guess would be 2 for the index and the index value of source sink for sink.
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Some strange thing is happening here. In the arch wiki says
Run $ pacmd list-sinks|egrep -i 'index:|name:' to list available sinks. The present default sink is marked with an asterisk.
so
zab ~ $ pacmd list-sinks|egrep -i 'index:|name:'
index: 0
name: <alsa_output.pci-0000_01_05.1.hdmi-stereo>
index: 1
name: <alsa_output.pci-0000_02_00.1.hdmi-stereo>
* index: 2
name: <alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00.analog-stereo>
index: 3
name: <alsa_output.pci-0000_00_14.2.analog-stereo>
So, the default sink is the index i want. But nothing in the output. Instead it uses index 3.
In the pavucontrol I have index 2 as default.
Last edited by zabrielza (2015-10-25 16:47:59)
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http://www.freedesktop.org/wiki/Softwar … ultDevice/
Standard PulseAudio setup doesn't really have a concept of a "default" device. Whenever a new stream appears, PulseAudio's stream-restore module checks if the same stream has been seen before, and if it is, then the stream is connected to the same sink or source where it was previously. There is a concept of a "fallback" device, which is used if the stream has not been seen before.
Any success changing a stream from sink 3 to sink 2 using using pavucontrol?
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I apologize for my late response, I am searching in the internet.
zab ~ $ pacmd list-sink-inputs
1 sink input(s) available.
index: 84
driver: <protocol-native.c>
flags:
state: RUNNING
sink: 3 <alsa_output.pci-0000_00_14.2.analog-stereo>
volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB
balance 0.00
muted: no
current latency: 477.60 ms
requested latency: 458.87 ms
sample spec: s24-32le 2ch 44100Hz
channel map: front-left,front-right
Stereo
resample method: copy
module: 11
client: 49 <Audacious>
properties:
media.name = "Audacious"
application.name = "Audacious"
native-protocol.peer = "UNIX socket client"
native-protocol.version = "30"
application.process.id = "4221"
application.process.user = "zab"
application.process.host = "zabArch"
application.process.binary = "audacious"
application.language = "en_US.UTF-8"
window.x11.display = ":0"
application.process.machine_id = "77b6e5a013fd4259b1743d7521751435"
application.process.session_id = "c2"
application.icon_name = "audacious"
module-stream-restore.id = "sink-input-by-application-name:Audacious"
so i ran
zab ~ $ pacmd move-sink-input 2 84
No sink input found with this index.
zab ~ $ pacmd move-sink-input 2,84
Failed to parse index.
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I apologize for my late response, I am searching in the internet.
zab ~ $ pacmd list-sink-inputs 1 sink input(s) available. index: 84 driver: <protocol-native.c> flags: state: RUNNING sink: 3 <alsa_output.pci-0000_00_14.2.analog-stereo> volume: front-left: 65536 / 100% / 0.00 dB, front-right: 65536 / 100% / 0.00 dB balance 0.00 muted: no current latency: 477.60 ms requested latency: 458.87 ms sample spec: s24-32le 2ch 44100Hz channel map: front-left,front-right Stereo resample method: copy module: 11 client: 49 <Audacious> properties: media.name = "Audacious" application.name = "Audacious" native-protocol.peer = "UNIX socket client" native-protocol.version = "30" application.process.id = "4221" application.process.user = "zab" application.process.host = "zabArch" application.process.binary = "audacious" application.language = "en_US.UTF-8" window.x11.display = ":0" application.process.machine_id = "77b6e5a013fd4259b1743d7521751435" application.process.session_id = "c2" application.icon_name = "audacious" module-stream-restore.id = "sink-input-by-application-name:Audacious"
so i ran
zab ~ $ pacmd move-sink-input 2 84 No sink input found with this index. zab ~ $ pacmd move-sink-input 2,84 Failed to parse index.
Seems the first form without the comma is the expected one so try reversing the position of the parameters as in
pacmd move-sink-input 84 2
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zab ~ $ pacmd move-sink-input 84 2
No sink input found with this index.
Does not work either..
But,
zab ~ $ pacmd move-sink-input 87 alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00.analog-stereo
returns nothing. What does this mean?
Well, now audacious outputs from the device index 2 . Will this output active after reboot?
Alsa output fails again though.
Also, trying to play some tune with vlc sends sound to index 3.
Last edited by zabrielza (2015-10-25 19:37:32)
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zab ~ $ pacmd move-sink-input 84 2 No sink input found with this index.
Does not work either..
I expected you to change 84 to the value of the source sink if it changed ( eg you stopped and started output ) it appears to have become 87
But,
zab ~ $ pacmd move-sink-input 87 alsa_output.usb-Burr-Brown_from_TI_USB_Audio_CODEC-00.analog-stereo
returns nothing. What does this mean?
Well, now audacious outputs from the device index 2 . Will this output active after reboot?
That should be covered in post 44
Does that mean you now have sound output from audacious through pulse?
Alsa output fails again though.
Is that not expected with pulse controlling the device?
Also, trying to play some tune with vlc sends sound to index 3.
Again that should be covered by post 44
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Do I have to configure sound in every application once and after that pulseaudio will output to index 2? I understand the post 44 you mentioned before, but why , why before the 7.0.0 version of pulseaudio, I did not have to do all this ? I had alsa routed to pulseaudio, with an .asound.conf file and after that pulse routed to jack. I did not have to worry about which application plays some sound. Also, as far i remember when I had alsa output in an application then sound was routed automatically to pulse device and then pulse to jack. No error at all. I have not rebooted yet, but I suppose from the post 44 that when I reboot and play a tune with audacious it will output to index 2. But do I really need to configure every app?
Can I set alsa as sink-input? So, that every app is with alsa output and then this output is an input for pulseaudio? I think that this is what pulseaudio-alsa package does, it creates this file
zab ~ $ cat /etc/asound.conf
# Use PulseAudio by default
pcm.!default {
type pulse
fallback "sysdefault"
hint {
show on
description "Default ALSA Output (currently PulseAudio Sound Server)"
}
}
ctl.!default {
type pulse
fallback "sysdefault"
}
# vim:set ft=alsaconf:
Am I right?
Last edited by zabrielza (2015-10-25 20:48:50)
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Do I have to configure sound in every application once and after that pulseaudio will output to index 2? I understand the post 44 you mentioned before, but why , why before the 7.0.0 version of pulseaudio, I did not have to do all this ? I had alsa routed to pulseaudio, with an .asound.conf file and after that pulse routed to jack. I did not have to worry about which application plays some sound. Also, as far i remember when I had alsa output in an application then sound was routed automatically to pulse device and then pulse to jack. No error at all. I have not rebooted yet, but I suppose from the post 44 that when I reboot and play a tune with audacious it will output to index 2. But do I really need to configure every app?
The default sink will be applied only to applications that pulse has not got a stored sink for already. So you need to change each the output for each app that pulse has an incorrect output set for once.
For applications you have never run before it should now work as the default / fallback is set to index2.
I do not think that it was pulse 7 that is the main cause here but
options snd-usb-audio index=0 nrpacks=1
became invalid due to support for the option nrpacks being removed. So the module was never loaded. Now this would mean that sink 2 was no longer available so when you tried to output audio pulse routed it to an available sink, sink 3.
Pulse then recorded that sink 3 was the sink it used with for instance audacious. You removed the now unsupported option and allowed sink 2 to return. Then adjusted the default sink back to sink 2 but pulse still had records of sources linked to sink 3.
Can I set alsa as sink-input? So, that every app is with alsa output and then this output is an input for pulseaudio? I think that this is what pulseaudio-alsa package does, it creates this file
zab ~ $ cat /etc/asound.conf # Use PulseAudio by default pcm.!default { type pulse fallback "sysdefault" hint { show on description "Default ALSA Output (currently PulseAudio Sound Server)" } } ctl.!default { type pulse fallback "sysdefault" } # vim:set ft=alsaconf:
Am I right?
That in theory should work but then applications are connecting via the alsa emulation of pulse rather than natively and it assumes every application you use support alsa as well as pulse.
This assumes that doing that bypasses pulse's memory of source sink connections when alsa emulation is used and I have no idea if that is true or not.
I have no knowledge of jack so can not help with that.
edit:
struck through text and addendum below
Last edited by loqs (2015-10-25 21:25:12)
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