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[alsa-devel] Fwd: multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Sean McNamara 22 Oct '08

22 Oct '08

Oops! Forgot to include the ML on this reply. Maybe someone else will find this info useful, or be able to add constructive comments to what has been said :) Sean ---------- Forwarded message ---------- From: Sean McNamara <smcnam(a)gmail.com> Date: Wed, Oct 22, 2008 at 4:17 PM Subject: Re: [alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters) To: Jelle de Jong <jelledejong(a)powercraft.nl> On Wed, Oct 22, 2008 at 1:48 PM, Jelle de Jong <jelledejong(a)powercraft.nl> wrote: > Hello everybody, > > I have a four to eighth muliseat debian linux system here and it is > missing sound for the users. I was hoping the alsa team can help me with > this issue. All sound issues on Linux are extremely application-specific. Getting sound to work without any specific application in mind is asking for trouble, because there is no single configuration of the sound stack (which, if you expand into historical sound APIs and servers, contains about 20 different ways to play sound) that will satisfy any application without further configuration. Since this is not a really well-formed question, I will proceed with the assumption that you want to be able to run, for example, aplay /usr/share/sounds/pop.wav in the console, and have it produce sound in the "appropriate" headset. Here's how you proceed: 1. Each person on the multiseat box must have their own UNIX user account. 2. Each user account must have an ~/.asoundrc file. 3. Each ~/.asoundrc file must redefine the "default" device. How you redefine it depends on a lot of things: (a) Do you want _direct_ hardware access to the sound card, i.e. one application at a time? (b) Do you want to use ALSA's built-in software mixing, i.e. dmix? (c) How many channels? (d) Do you want to use PulseAudio? (e) Do you want to use JACK? (f) For each USB headset, what is its corresponding "raw device string"? This will be something like hw:n, where n is a number starting from 0, depending on the order in which the USB drivers initialize each USB controller. All of the above questions, and others, will determine exactly how each user configures their ~/.asoundrc file. The *extremely naive* (not recommended) example of an ~/.asoundrc is like: pcm.!default { card 3 } to have ALSA clients play sound, by default, to the fourth sound device that happened to be initialized. > > I have usb headsets and usb audio devices and all users have there own > usb hub. There are no internal audio cards in the multiseat server. > > How can we configure an system that the user can listen to audio only to > the devices connected to his usb hub. Well, this is guaranteed by design, isn't it? Without physical access to someone else's headset, how would I get access to the audio streams being played to that headset? I don't think this really expresses what your problem is. If we flip your question around, it might be a little more interesting: "How can we configure a system so that each user's applications will only play audio to that user's corresponding headset?" You can interpret this question in two ways: 1. "How can we _prevent_ users from playing sound to each other's sound cards?" [mutually untrust[ed/ing] users] or 2. "How can we configure the default settings so that, if untampered with, a user's applications will play sound to that user's own headset and not anyone else's?" [mutually trust[ed/ing] users] The lack of hardware mixing actually comes back to being a security benefit to user space control in this kind of situation. Recall that if I run aplay --device=hw:0 /usr/share/sounds/pop.wav then as long as the aplay program is playing sound to hw:0, it is *impossible* (short of some really devious hacking in alsa-lib or the kernel) for another application -- from this user, or another -- to also play sound to hw:0 at the same time. I'll call this claim "No Native SW Mixing". Now, let's look at what is required to satisfy the question in each of the two above cases. Trusted: 1. We know that users aren't going to maliciously try and hurt one another's ears by setting someone else's mixer really loud and playing static to them. 2. We know that users won't try to tamper with eachother's processes or configuration files; once things are configured correctly, it'll stay that way. 3. By 'No Native SW Mixing', we know that users can *theoretically* hog another user's sound device whenever that user has zero applications currently using it. 4. Since users trust each other, they won't do this. 5. We can use something like dmix, or just use hw:0 if users don't need simultaneous app output, and things will be fine. 6. dmix is a viable option to implement software mixing. 7. PulseAudio is also a viable option to implement software mixing. Untrusted: 1. Users may want to maliciously try and hurt one another's ears by setting someone else's mixer really loud and playing static to them. 2. Users may try to tamper with eachother's processes or configuration files; once things are configured correctly, it could get messed up again. [There's not much you can do to prevent this ... at least not based on existing open source tools.] 3. By 'No Native SW Mixing', we know that users can *theoretically* hog another user's sound device whenever that user has zero applications currently using it. 4. Since users do not trust each other, there is potential for 3 to happen. 5. We must use 'No Native SW Mixing' to our advantage, to create a walled garden around each user's sound experience. We need a process that, once started by a user, will permanently "hog" the sound card that user wants to use. This process should, ideally, allow other processes run by that user (and by that user only) to access their USB headset for output or input. We need a "gatekeeper" that is sensitive to the context in which an app is being run. 6. dmix is a NOT viable option to implement software mixing, because: (a) Once all dmix applications close their streams, the audio device hw:n is available, and some malicious user can decide to acquire it, abuse it, and prevent its rightful user from accessing it with his own applications. (b) dmix does not restrict users from outputting sound based on their UID or session. It is NOT a gatekeeper, just an open gate. 7. PulseAudio is [perhaps the best] viable solution to implement software mixing for each respective user. It is also a viable solution to preventing users from taking control of one another's sound devices. Configuring pulseaudio for a multi-soundcard multi-user system is out of the scope of this mailing list, but suffice it to say that it can be done. You will want to run one PulseAudio daemon per user, and each daemon will hold exclusive control over exactly one USB headset. Each daemon will only accept clients' requests for audio I/O from the user who started the daemon. Do make sure that you disable module-suspend-on-idle, or PulseAudio will actually give up the sound device after a period of inactivity, making it insecure in an untrusted environment. BTW, if this project is for a commercial interest, you usually pay someone for this kind of in-depth analysis. If you're that someone, you might want to read up on some of the underlying technologies so that you can develop this sort of reasoning on your own. I can envision a very real situation for both the trusted and the untrusted environment, so there is definitely a demand for this kind of specialist. Trusted environments are popular at workplaces, and untrusted multi-seat boxes are popular at computer labs, public libraries, etc. > > Any help is really appreciated. Please let me know if this information was helpful -- it will help me gauge, in turn, whether I should spend my time writing up these sorts of answers on the ML. FWIW, I searched around for existing articles touching this subject, but I couldn't find anyone who addresses this specific issue accurately or in enough detail. Sean P.S. - I once spoke to a user on IRC who was having intermittent problems with multiple USB headsets. It turns out that the USB controllers in the computer didn't have enough bandwidth, or power (or both) to allow all of the headsets to play at once. If, after configuring, you run into issues where only so many users can play sound at once, then you should invest in additional PCI or (preferably) PCI-Express USB controllers. Note that adding hubs upstream of a USB controller does absolutely nothing to lessen the load on the USB controllers that are integrated into your motherboard. > > Thank in advance, > > Jelle > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] multiseat and alsa audio systems (usb headsets and usb audio adapters)
by Jelle de Jong 22 Oct '08

22 Oct '08

Hello everybody, I have a four to eighth muliseat debian linux system here and it is missing sound for the users. I was hoping the alsa team can help me with this issue. I have usb headsets and usb audio devices and all users have there own usb hub. There are no internal audio cards in the multiseat server. How can we configure an system that the user can listen to audio only to the devices connected to his usb hub. Any help is really appreciated. Thank in advance, Jelle

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Re: [alsa-devel] Nvidia hdmi problems
by Takashi Iwai 22 Oct '08

22 Oct '08

At Tue, 21 Oct 2008 14:46:19 +0200, Nir Tzachar wrote: > > >> The bad news are that mplayer misbehaves and will get stuck every time > >> I try to change the volume. However, this does not happen in vlc. Any > >> ideas? > > > > Well, we need to sort out where the problem comes from. > > Try the vanilla Linus tree without sound-2.6 whether the problem > > persists. > > I'll try to remove the sound-2.6 tree and report. Nir, do you have any results? I tried to reproduce the problem by myself, but I couldn't. So I wonder whether it's a codec specific or I missed something... thanks, Takashi

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[alsa-devel] [PATCH] ASoC: OMAP: Continue fixing DSP DAI format in McBSP DAI driver
by Jarkko Nikula 22 Oct '08

22 Oct '08

Fix "ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver" was not correct due misunderstanding of DSP_A format and similar error in TLV320AIC33 codec which was used to test the original fix. This patch corrects now DSP_A format in OMAP McBSP DAI driver and is verified with TLV320AIC23 codec that's implementing DSP_A correctly. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Arun KS <arunks(a)mistralsolutions.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 7 ++----- 1 files changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 853b33a..8485a8a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -265,7 +265,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, break; case SND_SOC_DAIFMT_DSP_A: regs->srgr2 |= FPER(wlen * 2 - 1); - regs->srgr1 |= FWID(0); + regs->srgr1 |= FWID(wlen * 2 - 2); break; } @@ -284,7 +284,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; - unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; @@ -307,8 +306,6 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); - /* Invert bit clock and FS polarity configuration for DSP_A */ - temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -332,7 +329,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- 1.5.6.5

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[alsa-devel] [PATCH 2.6.28] ALSA: Ensure PXA runtime data is initialised
by Mark Brown 22 Oct '08

22 Oct '08

The rest of the code relies on the runtime data being zero initialised so we need to use kzalloc() to allocate it. Reported-by: Oliver Ford <ipaqlinux(a)oliford.co.uk> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/arm/pxa2xx-pcm-lib.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/arm/pxa2xx-pcm-lib.c b/sound/arm/pxa2xx-pcm-lib.c index 1c93eb7..75a0d74 100644 --- a/sound/arm/pxa2xx-pcm-lib.c +++ b/sound/arm/pxa2xx-pcm-lib.c @@ -194,7 +194,7 @@ int __pxa2xx_pcm_open(struct snd_pcm_substream *substream) goto out; ret = -ENOMEM; - rtd = kmalloc(sizeof(*rtd), GFP_KERNEL); + rtd = kzalloc(sizeof(*rtd), GFP_KERNEL); if (!rtd) goto out; rtd->dma_desc_array = -- 1.5.6.5

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Re: [alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Tony Vroon 22 Oct '08

22 Oct '08

On a related note, it looks to me as if the leading two characters (the manufacturing location) have been erroneously included for E-MU products. You'll see this MA code on some sound cards as well (MASBxxxx) Regards, Tony V.

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[alsa-devel] [PATCH] ASoC: OMAP: Fix DSP DAI format in McBSP DAI driver
by Jarkko Nikula 22 Oct '08

22 Oct '08

Fix word clock length which must equal to one bit clock cycle in DSP mode. Surprisingly McBSP is able synchronize into wrong length when it's slave but e.g. TLV320AIC33 codec in slave configuration is outputting some amount of noise if word clock length is longer than one bit clock cycle. Fix also bit clock and frame sync polarities in DSP mode since they are opposite from I2S. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> Cc: Arun KS <arunks(a)mistralsolutions.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)nokia.com> --- sound/soc/omap/omap-mcbsp.c | 24 ++++++++++++++++++++---- 1 files changed, 20 insertions(+), 4 deletions(-) diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c index 0a063a9..853b33a 100644 --- a/sound/soc/omap/omap-mcbsp.c +++ b/sound/soc/omap/omap-mcbsp.c @@ -43,6 +43,7 @@ struct omap_mcbsp_data { unsigned int bus_id; struct omap_mcbsp_reg_cfg regs; + unsigned int fmt; /* * Flags indicating is the bus already activated and configured by * another substream @@ -200,6 +201,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id; + int wlen; unsigned long port; if (cpu_class_is_omap1()) { @@ -244,19 +246,29 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: /* Set word lengths */ + wlen = 16; regs->rcr2 |= RWDLEN2(OMAP_MCBSP_WORD_16); regs->rcr1 |= RWDLEN1(OMAP_MCBSP_WORD_16); regs->xcr2 |= XWDLEN2(OMAP_MCBSP_WORD_16); regs->xcr1 |= XWDLEN1(OMAP_MCBSP_WORD_16); - /* Set FS period and length in terms of bit clock periods */ - regs->srgr2 |= FPER(16 * 2 - 1); - regs->srgr1 |= FWID(16 - 1); break; default: /* Unsupported PCM format */ return -EINVAL; } + /* Set FS period and length in terms of bit clock periods */ + switch (mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(wlen - 1); + break; + case SND_SOC_DAIFMT_DSP_A: + regs->srgr2 |= FPER(wlen * 2 - 1); + regs->srgr1 |= FWID(0); + break; + } + omap_mcbsp_config(bus_id, &mcbsp_data->regs); mcbsp_data->configured = 1; @@ -272,10 +284,12 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, { struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data); struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs; + unsigned int temp_fmt = fmt; if (mcbsp_data->configured) return 0; + mcbsp_data->fmt = fmt; memset(regs, 0, sizeof(*regs)); /* Generic McBSP register settings */ regs->spcr2 |= XINTM(3) | FREE; @@ -293,6 +307,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, /* 0-bit data delay */ regs->rcr2 |= RDATDLY(0); regs->xcr2 |= XDATDLY(0); + /* Invert bit clock and FS polarity configuration for DSP_A */ + temp_fmt ^= SND_SOC_DAIFMT_IB_IF; break; default: /* Unsupported data format */ @@ -316,7 +332,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai, } /* Set bit clock (CLKX/CLKR) and FS polarities */ - switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) { case SND_SOC_DAIFMT_NB_NF: /* * Normal BCLK + FS. -- 1.5.6.5

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[alsa-devel] [PATCH] ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style
by Vedran Miletić 22 Oct '08

22 Oct '08

ALSA: emu10k1: fix device names for Live!/Audigy1/2/4/E-mu and coding style Changelog: * added missing SBxxxx, CTxxxx, PCxxx and MAEMxxxx where they were missing, and fixed some of them which were wrong (according to kx.inf, which is pretty accurate compared to anything out there) * fixed device names to make them more consistent across various cards * fixed lots of coding style errors and some warnings, now it is down to: checkpatch.pl-0.24 --no-tree --file --strict --terse emu10k1_main.c total: 0 errors, 62 warnings, 7 checks, 2074 lines checked Signed-off-by: Vedran Miletic <rivanvx(a)gmail.com> -- Vedran Miletić

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[alsa-devel] help test sbxfi
by William Zumwalt 22 Oct '08

22 Oct '08

I've been very interested in a working driver for the X-Fi for a long time, saw work was finally started, and would like to help test things out. Can someone list where the source is and the steps needed to get this working on linux? Kernel version, which X-Fi card, etc... I'd like to help and follow along if possible.

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[alsa-devel] Intel dg45id motherboard - ICH10 HD audio issues
by Anton Blanchard 22 Oct '08

22 Oct '08

Hi, I was unable to get sound to work (analog or SPDIF) on my dg45id using 2.6.27. After looking at the pinout information I made a quick hack to get SPDIF out going (it's a mythtv box so thats all I need): Index: linux-source-2.6.27/sound/pci/hda/patch_sigmatel.c =================================================================== --- linux-source-2.6.27.orig/sound/pci/hda/patch_sigmatel.c 2008-08-31 11:06:37.000000000 +1000 +++ linux-source-2.6.27/sound/pci/hda/patch_sigmatel.c 2008-08-31 11:07:14.000000000 +1000 @@ -3591,7 +3591,7 @@ spec->num_pwrs = ARRAY_SIZE(stac92hd73xx_pwr_nids); spec->pwr_nids = stac92hd73xx_pwr_nids; - err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + err = stac92xx_parse_auto_config(codec, 0x26, 0x24); if (!err) { if (spec->board_config < 0) { And pointed mythtv at ALSA:plughw:0,1. Quick and dirty I know :) Details of the chipset are below. Anton -- # lspci -vvnn 00:1b.0 Audio device [0403]: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller [8086:3a3e] Subsystem: Intel Corporation Device [8086:5002] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fe820000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> Kernel driver in use: HDA Intel Kernel modules: snd-hda-intel # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 # aplay -L default:CARD=Intel HDA Intel, STAC92xx Analog Default Audio Device front:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog Front speakers surround40:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Intel,DEV=0 HDA Intel, STAC92xx Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers hdmi:CARD=Intel,DEV=0 HDA Intel HDMI Audio Output null Discard all samples (playback) or generate zero samples (capture) # cat /proc/asound/card0/codec#2 Codec: IDT 92HD73E1X5 Address: 2 Vendor Id: 0x111d7676 Subsystem Id: 0x80865002 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0 Node 0x0a [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0xc0: OUT HP VREF_HIZ Unsolicited: tag=30, enabled=1 Connection: 6 0x15 0x16* 0x17 0x18 0x19 0x1e Node 0x0b [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x08173f: IN OUT HP Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x4f00000f: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0x0, Sequence = 0xf Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0c [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0d [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x083f: IN OUT HP Detect Trigger ImpSense Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0e [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x081737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x03a11020: [Jack] Mic at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=30, enabled=1 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x0f [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x0321101f: [Jack] HP Out at Ext Left Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x40: OUT Unsolicited: tag=35, enabled=1 Connection: 6 0x15 0x16 0x17* 0x18 0x19 0x1e Node 0x10 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x11 [Pin Complex] wcaps 0x400183: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0837: IN OUT Detect Trigger ImpSense Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 6 0x15* 0x16 0x17 0x18 0x19 0x1e Node 0x12 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Node 0x13 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0820: IN Pin Default 0x90a60160: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x6, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x400003: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Pincap 0x0820: IN Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Node 0x15 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x40 0x40] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x16 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xd9 0xd9] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x17 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xd9 0xd9] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x18 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x19 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x19 0x19] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x1a [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x20 Processing caps: benign=0, ncoeff=0 Node 0x1b [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D0, actual=D0 Delay: 13 samples Connection: 1 0x21 Processing caps: benign=0, ncoeff=0 Node 0x1c [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x1d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 5 0x28 0x29 0x2a 0x2b 0x12 Node 0x1e [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x1d Node 0x1f [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=1, val=108 Connection: 5 0x15* 0x16 0x17 0x18 0x19 Node 0x20 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d* Node 0x21 [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 12 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14 0x1d* Node 0x22 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x4f0000f0: [N/A] Line Out at Ext UNKNOWN Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Connection: 3 0x25* 0x20 0x21 Node 0x23 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x01451140: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 3 0x26* 0x20 0x21 Node 0x24 [Pin Complex] wcaps 0x400681: Stereo Digital Pincap 0x0810024: IN EAPD Detect EAPD 0x0: Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Power: setting=D0, actual=D0 Node 0x25 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x26 [Audio Output] wcaps 0x4021d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x27 [Audio Input] wcaps 0x14031b: Stereo Digital Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Connection: 1 0x24 Node 0x28 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a 0x0b* 0x0d 0x0f Node 0x29 [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0a 0x0e* 0x10 0x11 Node 0x2a [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x0b 0x0c* 0x10 0x11 Node 0x2b [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x15* 0x16 0x17 0x18

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