September 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [ASoc] TDM over one SSC channel on AT91 possible?
by Friedrich Lobenstock 24 Jun '11

24 Jun '11

Hello! The background: I'm just designing an ATMEL AT91RM9200 based board which got two TLV320AIC33 on board. It is either possible to hook up the two TLV320AIC33 to a separate SSC (synchronous serial control) channel each or use one SSC channel and configure it for TDM (time division multiplex) mode. There is already a driver for TLV320AIC14k/TLV320AIC2k in the development tree, so adding support for the TLV320AIC33 should be feasible. What I'm not sure about is the fact if ASOC is prepared for using channels, especially SSC, in TDM mode. Is the core ASOC structure prepared to handle that? Would we be better off instantiate one drive twice each time with different parameters for the SSC channel? Thanks for any feedback. -- MfG / Regards Friedrich Lobenstock _________________________________________________________ SCOTTY Group Austria GmbH a SCOTTY Group plc company, http://www.scottygroup.com/ _________________________________________________________

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Re: [alsa-devel] Patchmix-style application for EMU cards
by Camilo Polymeris 30 Aug '10

30 Aug '10

Hello > That is more or less what I had figured out, but it how are playback_1 to > > _16 channels routed to DSP0-15? It seems that it depends on the setting > of > > Front, Surround, LFE, etc. mixer controls. If they are turned up, I > don't > > understand what happens, sound to playback channels appears on various > DSP > > buses, at different levels. > > Really? Well, I didn't much experiment with surround mode. In source code, > I > can see that there are up to playback 8 channels at hw:0,0 with can be > controlled by faders in alsamixer. I tried surround40 and surroun51 on my > emu1616m and it worked as expected. The channles were mapped to first > couple > of DSP ports in a correct order. In fact this should work the same as > audigy > does, except the result is passed to DSP ports instead of physical > outputs. > On my card, playback channels 1 & 2 (hw:0,3) appear on all DSP busses associated with surround. Say you turn up the "Side" fader in alsamixer, channels 1 & 2 will appear at the relevant DSPs (6-7, I think) Whatever, for now I am just keeping all surround controls down (that is, Front, Surround, Center, Side & LFE). > Anyway, I think if there is another layer of routing done in audigy DSP, > it > shouldn't be used in EMU specific application. Or at least not for now. > The > most important thing is to provide all the switches and FPGA routing > scheme > for each EMU model. Or, to be precise, three routing schemes for each EMU > model, because for "double speed" (e.g. 96kHz) and "quadro speed" (e.g. > 192kHz) modes of operation the numebr of both physical and virtual > channels > changes. > I think the switches and enums for the card models are rather trivial, one can handle it with a customized version of qamix (done with a simple XML file), for example. A separate application would not be necessary for only that. > > If I turn the Front/Surround/LFE/etc. faders down, I can mix playback > > channels to DSP using the "Multichannel PCM Send/Routing/Volume" faders, > in > > a more or less intuitive way, besides that routing values do not map > > directly to DSP buses (i.e. 14-15 map to DSP0-1, but 2-3 map to DSP2-3). > Handling of this controls motivated me to write the application in the first place. No sound at all will appear if I don't set the appropiate controls for each playback channel: Multichannel PCM Volume (full), Multichannel PCM Send (an index of 8 controls, the first 4 select the target DSP) & Multichannel PCM Send Volume (again faders 0-3 of 8) By the way, only DSP0-7 work in this way. I have no idea how to make DSP8-31 work. On which card you did tested it? The driver was primary written for the > 1820M/1212M V1, so some things may not work well on other models and need > to > be fixed. I have a 1212m, don't know which version, but I think it's V1-- it does have firewire. I am using driver version 1.0.16. Thanks, Camilo.

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[alsa-devel] Audio buffer mmap-ing on sh arch vs. cache aliasing problem
by Pawel MOLL 16 Jan '10

16 Jan '10

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_area = vmalloc(size); [...] } ... and vmalloc will return a page(4k)-aligned pointer, possibly not 16k-aligned one. This is the source of all evil ;-) When using RW transfers everything is fine, as data is memcopied between two independent buffers. In a case of MMAP mode we will end up with such a situation: 1. The driver will memcpy data from an URB to the buffer. It will populate several cache lines. Let's call it a "kernel cache". 2. As the library has mapped the non-16k-aligned address via different cache line ("user cache"), it will read some rubbish from the physical memory, populating the "user cache" by the way. 3. Some time later the "kernel cache" is flushed, writing the data to the memory. 4. Some new data from URB is entering "kernel cache". 5. The library will access mmap-ed area again, going via the "user cache", which may or may not reflect the correct data (depending on the fact was the "user cache" flushed or not) etc. Of course this cycle is completely not deterministic, so some of the "kernel cache" lines will be flushed before being accessed from user space, other not... The final effect is... hmm... bizarre :-) First, of course, you will get a (probably loud) glitch (rubbish from buffer's underlying memory, before the first valid data is written back), and then something that could be described as an "echo" ;-) I mean - you are capturing "1 2 3 4 5 6 7..." and the result is (G stands for Glitch ;-) "G 1 23 3 4 4 6 67..." As a quick-and-dirty work-around I have modified snd_pcm_alloc_vmalloc_buffer() to allocate always 12kB more and then use 16k-aligned pointer: static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { struct snd_pcm_runtime *runtime = subs->runtime; if (runtime->dma_addr) { if (runtime->dma_bytes >= size) return 0; /* already large enough */ vfree((void *)runtime->dma_addr); } runtime->dma_addr = (unsigned long)vmalloc(size + 12 * 1024); if (!runtime->dma_addr) return -ENOMEM; runtime->dma_area = (void *)ALIGN(runtime->dma_addr, 16 * 1024); runtime->dma_bytes = size; return 0; } Of course it cannot be regarded as anything more as a hack. And definitely not as a solution... So my question is: Any idea how the "proper solution" should look like? I see no obvious method to get an arch-independent "mmap-compliant" buffer. This problem was found on a sh arch, using an USB Audio Class device, however out there some other architectures are suffering from this MMU "feature" as well (ie. see http://docs.hp.com/en/B3906-90006/ch07s09.html) and possibly other drivers could behave "wrongly" (quoted as the driver is actually innocent...) To be honest I just have absolutely no idea what to do with this all! :-O I hope I was clear enough in the description... Any feedback, advise, idea etc. will be more than appreciated. Cheers Paweł

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[alsa-devel] Roland/Edirol M-16DX
by Lasse Kärkkäinen 01 Jun '09

01 Jun '09

This device doesn't seem to be supported yet. Does Roland make the specifications available, etc? The device is not compatible with USB Audio, but rather uses Vendor Specific Class. lsusb -vvv is attached below and if you need more information, I can debug the device on Linux and on Vista. Bus 007 Device 025: ID 0582:00c4 Roland Corp. Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 255 Vendor Specific Class bDeviceSubClass 0 bDeviceProtocol 255 bMaxPacketSize0 64 idVendor 0x0582 Roland Corp. idProduct 0x00c4 bcdDevice 0.00 iManufacturer 1 EDIROL iProduct 2 M-16DX iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 167 bNumInterfaces 3 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xc0 Self Powered MaxPower 0mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 06 24 f1 01 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 2 iInterface 0 ** UNRECOGNIZED: 07 24 01 01 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 02 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0038 1x 56 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 2 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 2 bInterfaceProtocol 1 iInterface 0 ** UNRECOGNIZED: 07 24 01 07 00 01 00 ** UNRECOGNIZED: 0b 24 02 01 12 04 18 01 80 bb 00 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 37 Transfer Type Isochronous Synch Type Asynchronous Usage Type Implicit feedback Data wMaxPacketSize 0x01f8 1x 504 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 ** UNRECOGNIZED: 06 24 f1 02 01 01 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 2 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 1 bInterfaceProtocol 3 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1

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[alsa-devel] Issue with Alsa and PulseAudio
by Ash 11 Feb '09

11 Feb '09

Hello, I have a media application that uses alsa as its sound renderer. It has worked on numerous systems without any problems until I came across a mandriva distro that bundles with PulseAudio. The alsa pulseaudio plugin is installed correctly, however I am having weird problems. When I am setting up my hw params, and I call snd_pcm_hw_params_any(), I get an "Operation not permitted" error. I am not sure why this is happening, anyone else see this before? I am using pulseaudio version 0.9.9 and alsa 1.0.16. Thanks in advance! _________________________________________________________________ Be the filmmaker you always wanted to be—learn how to burn a DVD with Windows®. http://clk.atdmt.com/MRT/go/108588797/direct/01/

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[alsa-devel] snd_pcm_set_params() bug?
by Christian Morales Vega 26 Jan '09

26 Jan '09

This code: #include <alsa/asoundlib.h> int main(){ snd_pcm_t *handle; if(snd_pcm_open(&handle, "default:1", SND_PCM_STREAM_PLAYBACK, 0) < 0) { exit(1); } if(snd_pcm_set_params(handle, SND_PCM_FORMAT_S16_LE, SND_PCM_ACCESS_RW_INTERLEAVED, 2, 24000, 1, 100000) < 0) { exit(2); } } fails with error "ALSA lib pcm.c:7193:(snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Disable resampling and you get "ALSA lib pcm.c:7142:(snd_pcm_set_params) Rate doesn't match (requested 24000Hz, get 0Hz)" (normal, I suppose). With other samplerates works fine. I'm using alsa-driver-kmp-default-1.0.16.20080602 and alsa-1.0.16.hg20080510 RPM packages from multimedia:audio openSUSE Build Service repository (from Takashi Iwai) and an AD1988 (hda-intel) codec. A XUbuntu user also has the same problem with an hda-intel codec (that I suppose uses stable 1.0.16). With a SB Live! in the same computer there are no problems. This was discovered with bsnes ( http://byuu.cinnamonpirate.com/bsnes/ - http://board.zsnes.com/phpBB2/viewtopic.php?t=11584 ) that can use openal, libao or alsa directly. The problem just happens when using 24KHz with alsa. So I suppose the problem is in the snd_pcm_set_params() itself, since openal and libao have not the problem but they set parameters "manually", not with snd_pcm_set_params().

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[alsa-devel] snd_pcm_avail_update() needed before snd_pcm_delay() with hda-intel
by Christian Morales Vega 24 Jan '09

24 Jan '09

snd_pcm_avail_update() documentation says: "Using of this function is useless for the standard read/write operations. Use it only for mmap access. See to snd_pcm_delay." snd_pcm_delay() documentation says: "Note this function does not update the actual r/w pointer for applications. The function snd_pcm_avail_update() have to be called before any begin+commit operation." My english is far from perfect, and so the problem can be that I missunderstood the docs but: 1- "Using of this function"? Should not be "Use of this function"? 2- Is snd_pcm_avail_update() really useless when not using mmap? I have a SB Live! where snd_pcm_delay() always returns the correct value, without any need for snd_pcm_avail_update(). But the same code in the same machine with an hda-intel AD1988 needs a call to snd_pcm_avail_update() before snd_pcm_delay() to give the correct result. Without it snd_pcm_delay() always returns the value that *was* correct in the last snd_pcm_writei() call. So should the "application frame position" be always updated, or only in read/write operations? In the first case AD1988 fails, in the second case perhaps SB Live! is making too mucho work ;-)

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[alsa-devel] Driver: EMU USB 0404 driver development
by Klaus Schulz 03 Nov '08

03 Nov '08

Hi folks. I learned that EMU handed over specs related to the EMU 0202/0404 USB to you guys some time ago. http://ubuntuforums.org/showpost.php?p=4892734&postcount=157 Is anybody assigned to the task? Cheers

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[alsa-devel] ALSA 1.0.18rc3 packages
by Jaroslav Kysela 25 Oct '08

25 Oct '08

Hello all, ALSA 1.0.18rc3 packages are available for download on http://www.alsa-project.org. I expect to release 1.0.18 final in two weeks. Changes from 1.0.18rc2: http://www.alsa-project.org/main/index.php/Changes_v1.0.18rc2_v1.0.18rc3 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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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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