>
> So every time I reboot, pulse doesn't 'see' my mic anymore. I reboot and
pavucontrol shows this: https://i.imgur.com/ivFBCII.png As you can see, no
mic under input devices, the whole device is missing that would normally
show a drop-down menu to select front, rear or line-in. Then I simply pull
>>
>>
>> >
>> >> do loopback module assume sink is not running when it start source
capture?
>> >>
>> > Sink and source are started independently. Source data is discarded
until the sink-input has
>> > called pop() the first time.
>>
>> But you cannot skip more than 9ms source data since you need 1
>
>> do loopback module assume sink is not running when it start source
capture?
>>
> Sink and source are started independently. Source data is discarded until
the sink-input has
> called pop() the first time.
But you cannot skip more than 9ms source data since you need 1ms( or 25ms
fragment time
>
> Sorry for the late reply, had a busy week.
>
> > The easy way is to use hdajackretask your grey line out as internal
> > speaker, there will be headphone, 5.1 line out jacks and stereo speaker
> I tried that (removing the remap config), but for some reason the only
thing pavucontrol now sees is
than the reported
>>>> latency at later times.
>>>>
>>>> I hope this also clarifies why I don't buy your argument that the time
>>>> stamp difference is somehow related to the unreported latency.
>>>
>>> No, in fact it doesn't
>> The capture device may already started by other application (e.g. mic
peak of pavucontrol), there is some audio already captured by driver but
not read by server
>>
>> At low latency, usb pointer incremented by number of frames in urb
packet but hda intel increment by frames in dma brust
>>
>> D
>>>
>>> 5) The pulseaudio sink code takes the first 10ms of audio out of the
>>> loopback buffer,
>>> writes it to the alsa buffer and calls snd_pcm_start().
>>
>> If the sink takes something from the loopback buffer, this means that
>> the first pop() call has been made
> I am trying to get simultaneous output on the Line Out and Headphones
> outputs of my Intel HD Audio (AsRock Z97 motherboard) chipset and PC
> case. With the default pulseaudio configuration in pavucontrol I can
> choose Analog Stereo Duplex and then select either the "Line Ou
> > Hello everyone,
> > I am trying to get simultaneous output on the Line Out and Headphones
> > outputs of my Intel HD Audio (AsRock Z97 motherboard) chipset and PC
> > case. With the default pulseaudio configuration in pavucontrol I can
> > choose Analog Stereo Duplex and then select either the
>>
> 5) The pulseaudio sink code takes the first 10ms of audio out of the
> loopback buffer,
> writes it to the alsa buffer and calls snd_pcm_start().
If the sink takes something from the loopback buffer, this means that
the first pop() call has been made. Assuming no tim
> >
> > I'm using an ALC899 codec with 6x3.5mm jacks and so far, I've been
> > using sink-remap [1], which worked fine, 5.1 sound being remixed to
> > 2.0 as expected.
> >
> > However, there are two problems with this:
> > - even though I select the Stereo sink as output in pavucontrol, the
> > sou
>>
>>> > I don't want to shorten the latency. I only want the latency reported
correctly. To me it still
>>> > looks like the real latency of the driver is not what it reports,
because the time that the
>>> > audio spends in the URB's is not taken into account. What I am seeing
is, that the real
>>
>>
>> The USB driver will submit N silence URBs on startup in the prepare and
you will have to wait for those URBs to retire before the samples are
queued. There is very little 'USB processing'. If you want to reduce this
delay you have to use smaller periods, it'll decrease the size of the URBs.
I
>
> when a sink is started, there is some delay before the first sample is
really played.
> This delay is a constant part of the sink latency that will be always
present, so the
> minimum sink latency cannot go below that start delay.
> Would it be acceptable to adjust the latency range for the dev
2016-03-09 4:52 GMT+08:00 Tanu Kaskinen :
> On Wed, 2016-03-09 at 00:28 +0800, Raymond Yau wrote:
> > > > Alsa mixer api just combine volume control and mute switch with same
> name
> > > > to an element,
> > > >
> > > > Is it possible that
> > :
> > > > The volume_use is set to ignore, but we continue the volume parsing
> > > > code, potentially referencing somewhere outside the array (which has
> > > > max two channels).
> >
> > Alsa mixer api just combine volume control and mute switch with same
name
> > to an element,
> >
> > Is i
:
> > The volume_use is set to ignore, but we continue the volume parsing
> > code, potentially referencing somewhere outside the array (which has
> > max two channels).
Alsa mixer api just combine volume control and mute switch with same name
to an element,
Is it possible that these two control
2016-03-01 18:41 GMT+08:00 David Henningsson <
david.hennings...@canonical.com>:
> Looking at errors.ubuntu.com, a new crasher has climbed the charts a bit
> (at least
> compared to other ones), starting with PA 7.0. The assertion failure
> occurs when
> trying to set ALSA source volume. (
> https
2016-2-18 下午5:38於 "Bob Ham" 寫道：
>
> Make ALSA profiles stick around and update their availability if the
> HDMI ELD changes. This alters the presence of card profiles quite
> dramatically. Specifically, the existing autogeneration of profiles
> generates a large set of input/output mapping combin
>
> I'm a bit stuck. On my system (chip AD1888 in a VIA 8237R) everything
> seems to be detected fine, but I only get front-right and front-left
> instead of 5.1. I configured "daemon.conf" to use a correct default
> mapping and later also "default.pa" to create a new sink with right
> mapping. How
> > If you have headphones plugged in and plug in HDMI; you want sound
> > to stay on headphones.
> > If you have HDMI plugged in and you plug in headphones; you want sound
> > to switch to headphones.
> >
> > Hence we need to take priority into account as well when determining
> > whether to switc
>
> I have a problem of hearing white-noise on my headphones when I plug them
in my laptop. I know the work around:
>
> $ amixer -c 0 cset 'numid=10' 1
> numid=10,iface=MIXER,name='Headphone Mic Boost Volume'
> ; type=INTEGER,access=rw---R--,values=2,min=0,max=3,step=0
> : values=1,1
> | dBsc
>
> Hello to all on the list,
> I submitted this bug report on the Debian BTS:
> https://bugs.debian.org/cgi-bin/bugreport.cgi?bug=808170
> regarding pulseaudio >= 5.x and I was instructed to ask
> the same question here.
>
> The bug report contains all the logs with pulseaudio 4.x
> (the last vers
>
> I've been looking at those lines. Do you think it should be done in
USB-Audio, or should it be added into separate config file specific for
this sound card?
http://git.alsa-project.org/?p=alsa-lib.git;a=blob;f=src/conf/pcm/surround21.conf;hb=HEAD
The route of surround21 is hardcoded to use su
>
> Ok. Let's hope that in the future somebody actually fixes the driver so
that 2.1 works and we can add it back, but for now, I've pushed your patch.
type route
ttable.0.0 1
ttable.1.1 1
ttable.2.4 1
ttable.3.5 1
ttable.4.2 1
ttable.5.3 1
type route
ttable.0.FL 1 ttable.1.FR 1 ttable.2.LFE 1
Y
>>
>> > Do you mean 2.1 mode won't work with all usb audio since surround 5.1
of usb audio is already using route plugin
>> What do you mean by "all usb audio"? In 4.1/5.1 mode LFE and all other
outputs work properly, it is just for 2.1, which works as stereo.
>>
>> > If you don't have SPDIF output
>
> http://pastebin.com/1fNAViKp
>
> i switched to surround 4.0 output and i can change volume but its just a
> workaround because i have 2.0 speakers
>
>
https://bugs.freedesktop.org/show_bug.cgi?id=84983
The xonar D1 in the bug report does not have have headphone jack kctl but
AV200 have headph
>
> I was responsible for plugin the cables from the front panel to the
motherboard, I don't remember where exactly on the MOBO I plugged the front
panel audio cables but I followed my MOBO manual (Z7-DS3H). And on Windows,
this problem doesn't ocurr.
Do you mean Z77-DS3H ?
http://www.gigabyte.co
>
> I'm trying to get the following to work: There's a SoundBlaster Live
> (emu10k1) soundcard in my PC and the bass is set to 90%. This causes a
> lot of audio distortion on the audio output because Pulseaudio always
> sets the front volume to 100 % (and keeps resetting it to that value).
> If I s
>
>>
>>
>> I'm not sure that's relevant for dealing with the "Front" element.
>> Maybe you mean that the kernel doesn't know what paths "Front" affects?
>> In that case, the kernel could name the control "Gigabyte Front" so
>> that the userspace knows too that it's not known what the control does.
> > http://pastebin.com/1fNAViKp
>
> PulseAudio finds only the headphone path based on the information that
> the kernel gives. The card doesn't seem to have a separate headphone
> jack itself[1], so I suppose the "Headphone" mixer element controls the
> front panel output volume, if you have such
> If you change that to
>[Element Front]
>switch = off
>volume = off
>>does that fix the problem for you?
>Indeed this fixes my problem, thanks!
>
> Thanks for this. Although I think I'm going to stick with the first
solution provided, where would this piece of configuration go?
Why do pulseaud
>
> I'm not sure I understood 30% of what you said but basically, this is a
firmware problem that the kernel driver is trying to wrap its code around?
https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/pci/hda?id=9155f82a6a26da4a5b8d2d29f1d31836906b4712
You can specify
opt
> >
> > > I suspect that the "Front" volume and mute elements can be used to
> > > control the line out without affecting the headphones. The problem is
> > > that PulseAudio's alsa configuration assumes that the "Front" element
> > > affects both line out and headphones (there's a comment saying t
>
>
> >
> > I was responsible for plugin the cables from the front panel to the
motherboard, I don't remember where exactly on the MOBO I plugged the front
panel audio cables but I followed my MOBO manual (Z77-DS3H). And on
Windows, this problem doesn't occur.
>
> The driver disable jack detection
>
> I was responsible for plugin the cables from the front panel to the
motherboard, I don't remember where exactly on the MOBO I plugged the front
panel audio cables but I followed my MOBO manual (Z77-DS3H). And on
Windows, this problem doesn't occur.
The driver disable jack detection of headphon
>
> http://pastebin.com/1fNAViKp
>
> i switched to surround 4.0 output and i can change volume but its just a
> workaround because i have 2.0 speakers
>
Simple mixer control 'Master',0
Capabilities: pvolume pswitch pswitch-joined
Playback channels: Front Left - Front Right - Rear Left - Rear R
>
> After switching to my headphones, I activate the surround sink. This is
intentional as I want surround and HRTF with my headphones. Still, the
problem occurs even if this module isn't loaded.
Your alc887vd codec have enough volume controls for 5.1 and headphone
Only Headphone and Green Line O
> >
> > > Desktop
> > > Arch Linux
> > > PulseAudio 7.1
> >
> > Using motherboard embedded sound system.
> >
> > _Rear:_ 3 regular jacks: red, green, blue
> > -> Red jack: Mic connected
> > -> Green Jack: Main speakers (Stereo)
> >
> > _Front:_ 2 regular jacks: Mic and Headphones
> >
> > Also, m
>
> The gnome/unity-control-center UIs have a master volume slider, and
> three sub-sliders: balance, fade, and subwoofer. Balance and fade
> use PA's set_balance and set_fade APIs accordingly, but the subwoofer
> slider sometimes does unintuitive things.
>
> In order to make that slider behave bet
>>
>> We currently only support one and two channels for volumes, and
>> bail out otherwise. This makes Xonar users unhappy because they
>> have a volume with eight channels, and bailing out means they
>> don't have a path/port at all.
>>
>> This way they will at least have a port, which will in tu
>
> In 2.1 mode LFE is not actually working at all, so it is removed.
Do you mean 2.1 mode won't work with all usb audio since surround 5.1 of
usb audio is already using route plugin
USB-Audio.pcm.surround51.0 {
@args [ CARD ]
@args.CARD { type string }
@func refer
name {
@func concat
strings [
"
2015-9-18 下午8:48於 "David Henningsson" 寫道：
>
> The gnome/unity-control-center UIs have a master volume slider, and
> three sub-sliders: balance, fade, and subwoofer. Balance and fade
> use PA's set_balance and set_fade APIs accordingly, but the subwoofer
> slider sometimes does unintuitive things.
>
>
> I am having issue with pulseaudiosink,
> I used following command to play a wav file
> gst-launch filesrc location= ! wavparse ! pulseaudiosink
>
> if tsched is enabled, then play of wav file via pulseaudiosink gives me
lots of Underrun, and the sound is distorted, following is the pulseaudio
l
>>
>> >
>> > I tried a lot already. No success to get the sound working on my old
maxdata laptop.
>> > This is what I got:
>> >
>> > sl@max:~$ lspci -nnk | grep -iA2 audio
>> > 00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW
(ICH6 Family) High Definition Audio Controller [8086:
>
> I tried a lot already. No success to get the sound working on my old
maxdata laptop.
> This is what I got:
>
> sl@max:~$ lspci -nnk | grep -iA2 audio
> 00:1b.0 Audio device [0403]: Intel Corporation 82801FB/FBM/FR/FW/FRW
(ICH6 Family) High Definition Audio Controller [8086:2668] (rev 04)
>
maining issues with this card, for instance, microphone
works only with 2.0 speakers mode, 2.1 and others are not compatible with
it currently (when selecting mic speakers resets to stereo).
> There was few threads with Raymond Yau, but we didn't figure out how to
fix it completely, please,
>
> Yesterday I got an interesting email related to dcaenc support.
>
> The person has an "XFX nForce 780i SLI" MCP board:
http://www.nvidia.com/object/product_nforce_780i_sli_us.html ,
http://www.evga.com/support/manuals/files/132-CK-NF78.pdf
>
> As you see on page 18 of the manual, this board has
>
> Alsa-info patched kernel:
http://www.alsa-project.org/db/?f=cd23a3f0bb72e38d73224620e3e8ab57a73aa441
>
> Chmap results:
>
> Type = FIXED, Channels = 2
> FL FR
> Type = FIXED, Channels = 4
> FL FR NA LFE
>
> When unplugging the subwoofer, then chmap gives this and stays like this.
>
> Cannot
> >
> > Are you sure that it is alsa driver bug since pulseaudio seem not using
> > line out jack to mute internal speakers ?
>
> No, I'm not. I set all my jacks to line out and it works now.
>
Do pulseaudio 2.1 profile select the internal subwoofer or your 5.1
external speaker ?
diff --git a/sou
> Yesterday I got an interesting email related to dcaenc support.
>
> The person has an "XFX nForce 780i SLI" MCP board:
> http://www.nvidia.com/object/product_nforce_780i_sli_us.html ,
> http://www.evga.com/support/manuals/files/132-CK-NF78.pdf
>
> As you see on page 18 of the manual, this board h
>
> First I think I am going to split this mail into two topics, starting
with the easiest one, the External Base speaker.
>
> - The Base speaker does not switch automatically to any Surround when
plugged in.
>
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/profile-
>
> I can confirm that there is no such setting in the BIOS.
>
Node 0x11 [Pin Complex] wcaps 0x400300: Mono Digital Control: name="HDMI
Phantom Jack", index=0, device=0 Pincap 0x0010: OUT Pin Default
0x18567530: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Yellow
DefAssociation = 0x3
>>
>> >
>> > Yesterday I got an interesting email related to dcaenc support.
>> >
>> > The person has an "XFX nForce 780i SLI" MCP board:
>> http://www.nvidia.com/object/product_nforce_780i_sli_us.html ,
>> http://www.evga.com/support/manuals/files/132-CK-NF78.pdf
>> >
>> > As you see on page
>
> Yesterday I got an interesting email related to dcaenc support.
>
> The person has an "XFX nForce 780i SLI" MCP board:
http://www.nvidia.com/object/product_nforce_780i_sli_us.html ,
http://www.evga.com/support/manuals/files/132-CK-NF78.pdf
>
> As you see on page 18 of the manual, this board has
>>
>> It looks like in the future the ALSA drivers for some Intel hardware
>> will dynamically create a new PCM device when a DisplayPort monitor is
>> plugged in. This is being discussed in this thread (part of the thread
>> is also cross-posted to pulseaudio-discuss):
>>
http://thread.gmane.org/g
>> >
>> > I'd prefer a multi-purpose configuration with multiple
>> sinks/sources:
>> >
>> > - Source 'DAW': aux1,aux2,aux3,aux4,aux5,aux6
>> > - Source 'Mic/Instrument':channel 0/1 = left/right
>> > - Source 'Line': channel 2/3 = left/right
>> > -
2015-6-24 下午7:44於 "Rene Bartsch" 寫道：
>
> I'd prefer a multi-purpose configuration with multiple sinks/sources:
>
> - Source 'DAW': aux1,aux2,aux3,aux4,aux5,aux6
> - Source 'Mic/Instrument':channel 0/1 = left/right
> - Source 'Line': channel 2/3 = left/righ
>>
>> Do you mean mic capture volume always reset to maximum by driver or
pulseaudio ?
>
> No
>>
>> pactl list sources
>
> http://pastebin.com/Dd0XLamN
Seem usb audio is not your default source
Refer to pact list sources in
https://bugs.freedesktop.org/show_bug.cgi?id=90781#c32
Порти:
>>
>> pactl list sources
>
> http://pastebin.com/Dd0XLamN
> I'm using 4.1 mode, since 2.1 is not working properly yet
http://git.alsa-project.org/?p=alsa-lib.git;a=commit;h=1af088e39b75a0a0897c7036487b143e983cd423
Your pcm playback volume control is stereo , do pulseaudio allow you to
change the
>>
>> Do you mean mic capture volume always reset to maximum by driver or
pulseaudio ?
>
> No
>>
>> pactl list sources
>
> http://pastebin.com/Dd0XLamN
Your output did not include source usb audio input
Do you mean your usb audio are not default source ?
What is the default source ? alc892 analo
>
> ( 0.094| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Setting
analog-input-microphone (analog-input-microphone) priority=0
> ( 0.094| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_switch(): e->alsa_name Line
> ( 0.094| 0.000) D: [lt-pulseaudio] alsa-mixer.c:
element_set_switch(): e
>> >
>> >> Do you mean you still cannot force pulseaudio to use 24 bits for
>> your creative usb audio ?
>> >>
>> >> Seem try auto format did not include 24bits
>> >
>> > I have 24bit explicitly in pulseaudio config currently:
>> >>>
>> >>> default-sample-format = s24le
>> >
>> > But it s
>>
>> ( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Activating path
analog-input-mic
>> ( 0.090| 0.000) D: [lt-pulseaudio] alsa-mixer.c: Path
analog-input-mic (Microphone), direction=2, priority=87, probed=yes,
supported=yes, has_mute=yes, has_volume=yes, has_dB=yes, min_volume=0,
max_vo
>
>> Do you mean you still cannot force pulseaudio to use 24 bits for your
creative usb audio ?
>>
>> Seem try auto format did not include 24bits
>
> I have 24bit explicitly in pulseaudio config currently:
>>>
>>> default-sample-format = s24le
>
> But it still uses 16bit, didn't managed how to forc
>
> Looks good, but analog-output-speaker-always.conf should be patched too.
>
> Sjoerd, this does not disturb anything on your end, right?
>
http://cgit.freedesktop.org/pulseaudio/pulseaudio/commit/src/modules/alsa/mixer/paths/analog-output-headphones.conf?id=594da41d07edcebc5fd319388852a66cc3f12
.>
> Here is fresh verbose log with new debug messages:
http://pastebin.com/bfbrdfiv
> My debug lines start with element_set_option() and element_set_switch(),
also element_set_volume() was not called on start in my case, so it is not
present.
> I've tried to reflect in debug messages what exactly
>
> Looks good, but analog-output-speaker-always.conf should be patched too.
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/profile-sets/90-pulseaudio.rules
It seem that those stac9200 in 90-pulseaudio.rules are affected by this
patch
https://git.kernel.org/cgit/l
> > >> > >> >
> > >> > >> > below is what the terminate shows when running pcm_avail.c
> > >> > >> >
> > >> > >> > uid=0 gid=1007@nutshell:/ # alsactl_test
> > >> > >> > min_period_size: 8 frames, dir: 0
> > >> > >> > Playback hwparams: FIFO size is 8
> > >> > >> > Hardware PCM card 0 'r
>
> Some machines provide "Dock Line Out Jack" control that should be
> handled like a normal line out.
Do it mean the name of the playback switch of dock line out jack will not
change from line out playback switch to dock line out playback switch ?
___
>> >
>> > >>
>> > >>
>> > >> >
>> > >> > below is what the terminate shows when running pcm_avail.c
>> > >> >
>> > >> > uid=0 gid=1007@nutshell:/ # alsactl_test
>> > >> > min_period_size: 8 frames, dir: 0
>> > >> > Playback hwparams: FIFO size is 8
>> > >> > Hardware PCM card 0 'rsnd-dai
>>
>> During Activating path analog-input-mic,
>> You may need add pa_log_info() to element_set_volume() ,
element_set_switch() and element_set_option() to log alsa mixer element
and values (mic capture volume, mic capture switch and pcm capture source)
set by pulseaudio
>
> Do I need to put it in
> >> >
> >> > >>
> >> > >>
> >> > >> >
> >> > >> > below is what the terminate shows when running pcm_avail.c
> >> > >> >
> >> > >> > uid=0 gid=1007@nutshell:/ # alsactl_test
> >> > >> > min_period_size: 8 frames, dir: 0
> >> > >> > Playback hwparams: FIFO size is 8
> >> > >> > Hardware
>
> I tried using a smaller period_size (about 1/16 of hwbuf_size),
snd_pcm_avail returned a better value.
Period bytes = 2Kbytes which is still larger than the default
rewind_safeguard 512 bytes
> But cpu usage moved from 1% to 4%, since we are using it in a
automotive, we don`t care ab
> >
> > >>
> > >>
> > >> >
> > >> > below is what the terminate shows when running pcm_avail.c
> > >> >
> > >> > uid=0 gid=1007@nutshell:/ # alsactl_test
> > >> > min_period_size: 8 frames, dir: 0
> > >> > Playback hwparams: FIFO size is 8
> > >> > Hardware PCM card 0 'rsnd-dai.0-dirana3
>>
>>
>> >
>> > below is what the terminate shows when running pcm_avail.c
>> >
>> > uid=0 gid=1007@nutshell:/ # alsactl_test
>> > min_period_size: 8 frames, dir: 0
>> > Playback hwparams: FIFO size is 8
>> > Hardware PCM card 0 'rsnd-dai.0-dirana3.0' device 0 subdevice 0
>> > Its setup
>>
>> >
>> > below is what the terminate shows when running pcm_avail.c
>> >
>> > uid=0 gid=1007@nutshell:/ # alsactl_test
>> > min_period_size: 8 frames, dir: 0
>> > Playback hwparams: FIFO size is 8
>> > Hardware PCM card 0 'rsnd-dai.0-dirana3.0' device 0 subdevice 0
>> > Its setup is
>
> below is what the terminate shows when running pcm_avail.c
>
> uid=0 gid=1007@nutshell:/ #
alsactl_test
> min_period_size: 8 frames, dir: 0
> Playback hwparams: FIFO size is 8
> Hardware PCM card 0 'rsnd-dai.0-dirana3.0' device 0 subdevice 0
> Its setup is:
> stream : PLAYBACK
> a
>
>I found some audio noise problem when I trying to set the sink latency
to a lower value.
>
>here is the alsa dump:
>
> D/NMAudio ( 1959): Its setup is:
> D/NMAudio ( 1959): stream : PLAYBACK
> D/NMAudio ( 1959): access : MMAP_INTERLEAVED
> D/NMAudio ( 1959
> > > No change.
> > >
http://www.alsa-project.org/db/?f=30b3f0087374b914a20dbe20a618fb892a5d6fd5
> >
> > Are there any offical specificaion ?
> >
> > So far most review only mention stereo speakers and subwoofer, the
service
> > guide only show how to replace two internal speakers
>
> Service man
>
> My problem is that when I set channel output to "Analog output surround
5.1"
> (or 2.1 or 4.1), I hear no sound in front speakers, but in case of 5.1 or
4.1
> I can hear sound in rear speakers. I use kernel 3.19.5 and it didn't
happen in
> earlier kernels that is 3.18 and in 3.17 for sure.
>
>
>>
>> Do capture work as there is no output of snd_pcm_dump of the
snd-usb-audio 's hw params ?
>>
> Well, after system restart Mic is not working, while it is available and
selected in UI.
> After single switch to Line and back to Mic it starts working fine.
> I wasn't able to reproduce it twice d
>>
>> Do capture work as there is no output of snd_pcm_dump of the
snd-usb-audio 's hw params ?
>>
> Well, after system restart Mic is not working, while it is available and
selected in UI.
> After single switch to Line and back to Mic it starts working fine.
> I wasn't able to reproduce it twice d
>
> I've removed "default-sample-format = s24le", now it uses S16_LE
everywhere, fresh log: http://pastebin.com/aaLvtEUQ
Some error messages can be reproduced by set default-sample-format to s24le
But cannot reproduce SNDRV_PCM_IOCTL_HW_PARAMS
Any error meesage in system log , it seem snd_usb_p
>>
>> Have you tested surround21?
>
> Yes, front speakers work fine, however LFE doesn't seem to work. Instead
when I try to test LFE (from sound UI) I have weird noise in one of rear
speakers. There are no such problems with any other mode though (I see that
officially only 5.1 mode is supported,
>>
>> Do you mean try_auto does not work as expected ?
>>
>> [PA_SAMPLE_S24LE] = SND_PCM_FORMAT_S24_3LE,
>>
>> static const pa_sample_format_t try_order[] = {
>> PA_SAMPLE_FLOAT32NE,
>> PA_SAMPLE_FLOAT32RE,
>> PA_SAMPLE_S32NE,
>> PA_SAMPLE_S32RE,
>> PA_SAMP
>
> According to specification sound card is capable of 24bit 96kHz max, so I
have added:
>>
>> default-sample-format = s24le
>
> to my /etc/pulse/daemon.conf
Do you mean try_auto does not work as expected ?
[PA_SAMPLE_S24LE] = SND_PCM_FORMAT_S24_3LE,
static const pa_sample_format_t try_order[
>> pactl list sinks
>
> Приймач даних №1
> Стан: RUNNING
> Назва:
alsa_output.usb-Creative_Technology_Ltd_SB_Omni_Surround_5.1_00Q6-00-S51.analog-stereo
> Опис: SB Omni Surround 5.1 Аналогове стерео
> Драйвер: module-alsa-card.c
> Частотна специфікація: s24le 2кан. 48000Гц
>
0.098| 0.000) D: [pulseaudio] alsa-util.c:
snd_pcm_hw_params_set_channels(4) failed: Invalid argument
0.098| 0.000) D: [pulseaudio] alsa-mixer.c: Skipping profile
output:analog-surround-40+input:analog-stereo-input - will not be able to
open output:analog-surround-40
Seem not support 4 cha
>>
>> Have you tested surround21?
>
> Yes, front speakers work fine, however LFE doesn't seem to work. Instead
when I try to test LFE (from sound UI) I have weird noise in one of rear
speakers. There are no such problems with any other mode though (I see that
officially only 5.1 mode is supported,
>
>> Do surround51 works since you have a hardware volume knob but PCM
playback volume is mono
>
> If I understood you correctly - I've took mono sound (wav from ALSA
package) and played it through VLC, all channels work as expected.
Card hw:2 'S51'/'Creative Technology Ltd SB Omni Surround 5.1 at
>>
>> Why pulseaudio still try to open front device for recording ?
>> Should pulseaudio just start from here ?
>
> I have no answers on those questions.
Seem missing direction = output in analog-stereo
___
pulseaudio-discuss mailing list
pulseaudio-disc
>
> ---
> .../alsa/mixer/paths/analog-input-linein.conf | 7 ++
> src/modules/alsa/mixer/paths/analog-input-mic.conf | 7 ++
> .../alsa/mixer/profile-sets/90-pulseaudio.rules| 1 +
> .../mixer/profile-sets/sb-omni-surround-5.1.conf | 78
++
> 4 files changed, 93 i
>
> Who/where is sensitivity settings for the microphone kept?
http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixer/paths/analog-input-mic.conf
>
> I try to use SFLPhone on a OpenSUSE KDE installation. SFLphone
> recognizes pulse and most seems to work. However when I rec
> ---
> src/modules/alsa/mixer/paths/analog-output-headphones-2.conf | 4
> src/modules/alsa/mixer/paths/analog-output-headphones.conf | 4
> src/modules/alsa/mixer/paths/analog-output-lineout.conf| 6 ++
> src/modules/alsa/mixer/paths/analog-output-speaker-always.conf
2015-05-05 15:45 GMT+08:00 David Henningsson <
david.hennings...@canonical.com>:
> This case was apparently overlooked.
>
> Signed-off-by: David Henningsson
> ---
> src/modules/alsa/mixer/paths/analog-output-lineout.conf | 4
> 1 file changed, 4 insertions(+)
>
> diff --git a/src/modules/al
>> Since many years ago we have an API for setting default source/sink.
>>>
>>> Our default routing today is more port centric. So our UIs (at least the
>>> unity/gnome one) has developed ways around this, so that when a port is
>>> selected it first selects the right profile if needed.
>>>
>>> The
> > Only tested with snd-dummy with model=ice1712
> >
> > I expect pulseaudio only use those multi channel profiles only when
driver
> > 's channel_min > 2 and channel_min equal to channel_max
> >
> > So pulseaudio just playback and capture the hardware supported channels
> > instead of the first 4
>
> >
> http://cgit.freedesktop.org/pulseaudio/pulseaudio/tree/src/modules/alsa/mixe
> > > > r/paths/analog-output-headphones.conf
> > > >
> > > > pulseaudio mute external center/lfe controls when headphone jack is
> >
> > plugged
>
> That would be unwanted.
>
This is common to all notebook using
> > > My laptop has built-in front left, front right, rear left, rear right
and
> > > subwoofer and something is strange.
> > > When I set channel output to 5.1 or 4.1 and external speakers are
> > > disconnected, then front left and front right work as expected but
> >
> > built-in
> >
> > > subwo
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