June 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] View alsa-lib plugins setup for any app?
by Eliot Blennerhassett 05 Dec '25

05 Dec '25

Greetings, running aplay -v will show the plugin chain setup. /proc/asound/card/pcm/* shows hw and sw params for the card driver when in use by any app Is there any way to show the alsa-lib plugin settings a-la aplay for an arbitrary application? thanks -- Eliot

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[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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[alsa-devel] idt STAC927x adat supported (sigmatel HDA codec) ?
by Eric 03 Mar '11

03 Mar '11

Is the ADAT output of the IDT (sigmatel) STAC927x HDA codec supported in alsa ? If not could someone please try to add this incredibly nice feature in the existing alsa driver ? Thank you very much

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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] Intel G45 8-Channel LPCM over HDMI
by Alexander Werner 11 Dec '09

11 Dec '09

Hi, I'm planning to build a linux-based, small sized PC for hearing aid acousticians. It would be highly unfavorable, if the connection between Receiver and PC was analog, because of the lower quality and the high number of cables. I searched for a appropriate mainboard, and found the DG45FC, which fits my needs perfectly, apart from the missing audio over HDMI support. Is there any plan to support multichannel LPCM output over HDMI on the Intel G45 platform? If not, is there anyone willing to implement HDMI support for the G45 chipset if I donate the needed hardware (permanently)? But if there's no chance to implement it in the near future, has anyone experience with an onboard chipset with multichannel LPCM over HDMI and ALSA support? Thanks, Alexander Werner

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[alsa-devel] Trying to get unsupported CODEC to work
by Alex Austin 05 Nov '09

05 Nov '09

HelloI have an MSI Media Live DIVA motherboard. It has an AMD SB600 HD Audio controller, and two CODECs on the HDA bus. First is an ALC888 Second is an Intersil DAE-3. Two alsa-info outputs in different circumstances http://www.alsa-project.org/db/?f=20c86e755fdbbdbc8761c8c2f4de76bce2ea0c11 http://www.alsa-project.org/db/?f=15c5e5eab0104622a84b2dd0df03f26c17c7242d Output of codecgraph with "modprobe hda-intel probe_mask=2" http://pastebin.com/f54b6a9b8 What's the next step to get this supported? -- - Alex Austin (651) 238-9273 "...and then I visited Wikipedia ...and the next 8 hours are a blur."

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Re: [alsa-devel] Acer Aspire 7730G support
by Denis Kup 24 Aug '09

24 Aug '09

2009/6/24 Denis Kup <dener.kup(a)gmail.com>: > 2009/6/24 Takashi Iwai <tiwai(a)suse.de>: >> At Wed, 24 Jun 2009 16:09:58 +0400, >> Denis Kup wrote: >>> >>> Hey, when i first installed alsa to my laptop, there no sound from >>> subwoofer, jacks were map very strange (2 surrounds for ex.). I try >>> all models from the list, but all don't support my subwoofer. After a >>> weak of hard work i made a patch that enables ACER_ASPIRE_7730G model >>> for my laptop. It have ALC888S, 3 jacks, 6 channels, stereo internal >>> frontal speakers and internal subwoofer.You can see acer.com for more >>> info about hardware and jacks. >>> >>> It's the changes i made to /alsa-kernel/pci/hda/patch_realtek.c: >> >> Can you try Emilio's patch recently posted (and pass >> model=acer-aspire-6530g) with the very latest alsa-driver-snapshot? >> This essentially includes your changes. It's in: >> From: Emilio López <buhitoescolar(a)gmail.com> >> Subject: Re: [alsa-devel] Regarding Sound on Acer Aspire 6935G >> Date: Tue, 23 Jun 2009 22:32:45 -0300 >> Message-ID: <d54969c50906231832w48ce7315uc5cf8b3f81534c76(a)mail.gmail.com> >> >>> After changing file i install driver, and it automatically detect its >>> model (maybe there are old settings from previous install, so i dont >>> 100% sure that it was detected). With this settings subwoofer works >>> perfect, internal stereo and subwoofer mute when headphone are plugged >>> all works, except 1 thing: subwoofer uses center channel instead LFE. >>> To fix it i use such asound.conf file: >> ... >>> Is there any chance to swap Center and LFE in patch_realtek.c? >> >> Unlikely as ALC888* doesn't support the channel-swap feature (although >> some other ALC codecs do). >> >> >> thanks, >> >> Takashi >> > > You are talking about that > http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html > ?? Its right link to drivers http://www.alsa-project.org/snapshot/ ?? > And there is easier way to apply patches than manually adding strings > to files? > I manage with installing. I use alsa-driver-snapshot.tar.gz http://ftp.kernel.org/pub/linux/kernel/people/tiwai/alsa/alsa-driver/ and this Amilo changes http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018815.html . But this patch has smaller functionality against i have. There is advantages and disadvantages of both patches: My patch: + 1) when black jack plugged, internal speakers and subwoofer mute + 2) there is 3 maps for jacks: (black/pink/blue) 2ch: front/ext mic/ (something i dont know, it should be line in), 4ch: front/ ext mic/ side 6ch: front/ CLFE / side; (its changes in channel mode option in mixer) + 3) internal mic works + 4) there are separate boost mic channels for internal and external mics - 5) internal subwoofer controls with Center channel in mixer ? 6) i dont know how line in works (i will check it when reinstall to my patch) ? 7) i dont know how spdif works, because i have not any spdif speakers in house Amilo patch: - 1) when black jack pluged, only internal speaker mutes, subwoofer continue to work (it has no meaning when i listen to music in headphones) - 2) there is only 1 map for jacks: front/ext mic/ (something i dont know, it should be line in) - 3) there no side and center channels - 3) internal mic dont work - 4) there no line in in input source menu in mixer + 5) Subwoofer controls with LFE channel in mixer ? 7) i dont know how spdif works, because i have not any spdif speakers in house

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[alsa-devel] load_for_all_cards bug
by Kevin Goodsell 10 Aug '09

10 Aug '09

(I originally tried to send this via the moderator without subscribing to the list, but it never went through and I never received a moderator response. Apparently this list is set up as moderated, but nobody is actually moderating it.) Summary: load_for_all_cards fails with existing configuration elements In snd_config_hook_load_for_all_cards, the first call to snd_config_search attempts to locate an existing configuration node with the name of the driver. Typically none is found, and everything is good. However, if such a node is located, the next line assumes it is a leaf node with type 'string' and calls snd_config_get_string to fetch the string value. If this fails, the entire hook is abandoned. Because of this, setting something like the following in asoundrc: cards.<driver name>.foo 0 is sufficient to disable the entire card-specific configuration. As a concrete example, I have a HDA-Intel sound card. dmix.conf includes a way to set period_size, period_time, and periods by using configuration elements of the form cards.<driver name>.pcm.dmix.<var>. In ~/.asoundrc I add cards.HDA-Intel.pcm.dmix.period_size 1024 This will cause HDA-Intel.conf to fail to load, and the pcm defined in default.conf will fail to find the device-specific pcm cards.HDA-Intel.pcm.default, and fall back on the default pcm using plughw. By attempting to configure dmix, I have disabled it. Fixing this may just be a matter of changing the line in snd_config_hook_load_for_all_cards from: if (snd_config_search(root, fdriver, &n) >= 0) { to: if (snd_config_search(root, fdriver, &n) >= 0 && n->type != SND_CONFIG_TYPE_COMPOUND) { I have not tested this. -Kevin

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[alsa-devel] [PATCH 1/3] ASoC: change set_tdm_slot api to allow slot_width override.
by Daniel Ribeiro 06 Aug '09

06 Aug '09

Extend set_tdm_slot to allow the user to arbitrarily set the frame width and active TX/RX slots. Updates magician.c and wm9081.c for the new set_tdm_slot(). wm9081.c still doesn't handle the slot_width override. While being there, correct an incorrect use of SlotsPerFrm(7) use in bitmask on pxa-ssp.c (SSCR0_SlotsPerFrm(x) is (((x) - 1) << 24)) ). (this series is meant for Mark's for-2.6.32 branch) Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- include/sound/soc-dai.h | 5 +++-- sound/soc/codecs/wm9081.c | 5 +++-- sound/soc/pxa/magician.c | 2 +- sound/soc/pxa/pxa-ssp.c | 27 ++++++++++++++++++++------- sound/soc/soc-core.c | 9 ++++++--- 5 files changed, 33 insertions(+), 15 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 352d7ee..6692410 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -106,7 +106,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); @@ -140,7 +140,8 @@ struct snd_soc_dai_ops { */ int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt); int (*set_tdm_slot)(struct snd_soc_dai *dai, - unsigned int mask, int slots); + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 86fc57e..c47d573 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1206,8 +1206,9 @@ static int wm9081_set_sysclk(struct snd_soc_dai *codec_dai, return 0; } +/* FIXME: Needs to handle slot_width */ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct snd_soc_codec *codec = dai->codec; unsigned int aif1 = wm9081_read(codec, WM9081_AUDIO_INTERFACE_1); @@ -1219,7 +1220,7 @@ static int wm9081_set_tdm_slot(struct snd_soc_dai *dai, aif1 |= (slots - 1) << WM9081_AIFDAC_TDM_MODE_SHIFT; - switch (mask) { + switch (rx_mask) { case 1: break; case 2: diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c index c89a3cd..34437d8 100644 --- a/sound/soc/pxa/magician.c +++ b/sound/soc/pxa/magician.c @@ -188,7 +188,7 @@ static int magician_playback_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 1); + ret = snd_soc_dai_set_tdm_slot(cpu_dai, 1, 0, 1, width); if (ret < 0) return ret; diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index e22c5ce..5b9ed64 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -375,21 +375,34 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, * Set the active slots in TDM/Network mode */ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { struct ssp_priv *priv = cpu_dai->private_data; struct ssp_device *ssp = priv->dev.ssp; u32 sscr0; - sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); + sscr0 = ssp_read_reg(ssp, SSCR0); + sscr0 &= ~(SSCR0_MOD | SSCR0_SlotsPerFrm(8) | SSCR0_EDSS | SSCR0_DSS); + + /* set slot width */ + if (slot_width > 16) + sscr0 |= SSCR0_EDSS | SSCR0_DataSize(slot_width - 16); + else + sscr0 |= SSCR0_DataSize(slot_width); - /* set number of active slots */ - sscr0 |= SSCR0_SlotsPerFrm(slots); + if (slots > 1) { + /* enable network mode */ + sscr0 |= SSCR0_MOD; + + /* set number of active slots */ + sscr0 |= SSCR0_SlotsPerFrm(slots); + + /* set active slot mask */ + ssp_write_reg(ssp, SSTSA, tx_mask); + ssp_write_reg(ssp, SSRSA, rx_mask); + } ssp_write_reg(ssp, SSCR0, sscr0); - /* set active slot mask */ - ssp_write_reg(ssp, SSTSA, mask); - ssp_write_reg(ssp, SSRSA, mask); return 0; } diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1a920c..c72a8e8 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2133,17 +2133,20 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_set_fmt); /** * snd_soc_dai_set_tdm_slot - configure DAI TDM. * @dai: DAI - * @mask: DAI specific mask representing used slots. + * @tx_mask: bitmask representing active TX slots. + * @rx_mask: bitmask representing active RX slots. * @slots: Number of slots in use. + * @slot_width: Width in bits for each slot. * * Configures a DAI for TDM operation. Both mask and slots are codec and DAI * specific. */ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, - unsigned int mask, int slots) + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { if (dai->ops && dai->ops->set_tdm_slot) - return dai->ops->set_tdm_slot(dai, mask, slots); + return dai->ops->set_tdm_slot(dai, tx_mask, rx_mask, + slots, slot_width); else return -EINVAL; } -- tg: (c787980..) asoc/set_tdm_slot (depends on: asoc/remove-code-duplication) -- Daniel Ribeiro

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[alsa-devel] my udev rules are breaking my dmixer setup why?
by Jelle de Jong 29 Jul '09

29 Jul '09

Hello everybody, I am trying to use udev to get fixed device names depending on the location of the device on the usb bus. The udev rules seems to be working fine, however i cant get dmix to work with the udev rules using only hw dies work. If i remove the udev rules the dmixer works fine. Can somebody look at the attachment, and maybe do some testing, i would highly appreciate it. I am also on chat.freenode.org and #alsa as tuxcrafter Thanks in advance, Jelle ls -hal /dev/snd/ total 0 drwxr-xr-x 2 root root 320 2008-11-09 15:44 . drwxr-xr-x 12 root root 4.8K 2008-11-09 15:50 .. crw-rw---- 1 root user0 116, 0 2008-11-09 15:44 controlC0 crw-rw---- 1 root user1 116, 32 2008-11-09 15:44 controlC1 crw-rw---- 1 root user2 116, 64 2008-11-09 15:44 controlC2 crw-rw---- 1 root user3 116, 96 2008-11-09 15:44 controlC3 crw-rw---- 1 root user0 116, 24 2008-11-09 15:44 pcmC0D0c crw-rw---- 1 root user0 116, 16 2008-11-09 15:49 pcmC0D0p crw-rw---- 1 root user1 116, 56 2008-11-09 15:44 pcmC1D0c crw-rw---- 1 root user1 116, 48 2008-11-09 15:44 pcmC1D0p crw-rw---- 1 root user2 116, 88 2008-11-09 15:44 pcmC2D0c crw-rw---- 1 root user2 116, 80 2008-11-09 15:44 pcmC2D0p crw-rw---- 1 root user3 116, 120 2008-11-09 15:44 pcmC3D0c crw-rw---- 1 root user3 116, 112 2008-11-09 15:44 pcmC3D0p crw-rw---- 1 root audio 116, 1 2008-11-09 15:44 seq crw-rw---- 1 root audio 116, 33 2008-11-09 15:44 timer ------------------------------------------------------------------------ udevinfo info --attribute-walk --name=/dev/snd/controlC0 ------------------------------------------------------------------------ echo 'SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="controlC[0-9]*", NAME="snd/controlC4" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="controlC[0-9]", NAME="snd/controlC5" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="controlC[0-9]", NAME="snd/controlC6" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="controlC[0-9]", NAME="snd/controlC7" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="hwC[0-9]D0c", NAME="snd/hwC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC4D0c" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC5D0c" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC6D0c" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0c", NAME="snd/pcmC7D0c" SUBSYSTEM=="sound", KERNELS=="3-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC4D0p" SUBSYSTEM=="sound", KERNELS=="3-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC5D0p" SUBSYSTEM=="sound", KERNELS=="4-1", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC6D0p" SUBSYSTEM=="sound", KERNELS=="4-2", KERNEL=="pcmC[0-9]D0p", NAME="snd/pcmC7D0p"' | tee /etc/udev/rules.d/10-persistent-sound.rules chmod 600 /etc/udev/rules.d/10-persistent-sound.rules ------------------------------------------------------------------------ # without dmixer # this works .... echo 'pcm.softvol { type softvol slave.pcm hw:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.softvol { type softvol slave.pcm dmix:1 control.name Master control.card 1 } pcm.!default { type plug slave.pcm softvol } ctl.!default { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------ # with dmixer # this does not work echo 'pcm.!default { type plug slave.pcm dmixer } pcm.dmixer { type dmix ipc_key 2048 slave.pcm hw:1 } ctl.dmixer { type hw card 1 } pcm.dsp { type plug slave.pcm dmixer } ctl.mixer { type hw card 1 }' | tee /home/user1/.asoundrc chmod 600 /home/user1/.asoundrc chown user1:user1 /home/user1/.asoundrc ------------------------------------------------------------------------

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[alsa-devel] Testers wanted: New SB X-Fi driver
by Matthias Diehl 21 Jul '09

21 Jul '09

> At Tue, 30 Jun 2009 16:10:45 +0200, > Matthias Diehl wrote: > > > > hi, > > > > i guess you dont speak german, do you? well, i'll try it in > english ;-) > > > > i tried to install your newest x-fi "hack" in ubuntu 9.04 64bit. > it's an x-fi extreme music. i downloaded this snapshot: >> > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-uns… > > Use alsa-driver-snapshot.tar.gz instead of the unstable one. > > > while the installation process there are some warnings. after > replacing the demon.conf and reboot there was no sound. > > > > then i noticed the sentence "Note that the ctxfi driver might > conflict with the HD-audio driver" in the description. i think my > problem is the hdmi soundchip from my graphics adapter (sapphire > radeon hd 4870) which was automatically installed at the > ubuntu-installation > ("hda intel sound"). > > You are likely looking at irrelevant things. > > There are several variants of X-Fi boards. Many of them are equipped > with emu20k1 or emu20k2 chip. Some of these models can behave as > HD-audio compatible, too. These are supported by snd-ctxfi driver. > > In addition, there is X-Fi models that are without emu20k1/emu20k2 > chips. Some of them (CA0110-IBG) behave only as a HD-audio device, > and some others are with CA0106 chip. The former is supported by > snd-hda-intel driver and the latter by snd-ca0106 driver. > > So, they cannot be deactivated. The only question is whether you have > emu20k1/emu20k2 chip on your board. And this can be checked when you > load snd-ctxfi driver. If the driver is properly loaded and the > card instance is created (see /proc/asound/cards), then it's good. > Otherwise, your board is no such one. You need other drivers. > > Please ask further questions on alsa-devel ML (you can add me to Cc, > though) so that other people with your board can advice better. > > > HTH, > > Takashi hi there, i now tried to install the stable version, but i have still no sound. here's what lspci | grep audio says: 01:00.1 Audio device: ATI Technologies Inc HD48x0 audio 05:00.0 Multimedia audio controller: Creative Labs SB X-Fi lsmod | grep snd says: snd_hda_intel 557492 1 snd_pcm_oss 52352 0 snd_mixer_oss 24960 2 snd_pcm_oss snd_pcm 99336 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 11524 0 snd_seq_oss 41984 0 snd_seq_midi 15744 0 snd_rawmidi 33920 1 snd_seq_midi snd_seq_midi_event 16512 2 snd_seq_oss,snd_seq_midi snd_seq 66272 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 34064 2 snd_pcm,snd_seq snd_seq_device 16276 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 78792 9 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 16800 2 snd snd_page_alloc 18704 2 snd_hda_intel,snd_pcm cat /proc/asound/cards says: 0 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xf7dfc000 irq 17 i hope, someone can help me ... greetings from germany, matthias

4 3

[alsa-devel] Testers wanted: New SB X-Fi driver
by Christian Esken 12 Jul '09

12 Jul '09

Hello, unfortunately I cannot playback anything (PCM) with the new SB X-Fi driver. Driver: ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa- driver-snapshot.tar.bz2 Distribution: openSuSE 11.1 Kernel: 2.6.27.23-0.1-xen My card is a "X-Fi Titanium" (PCI device code: 1102:000b). The card is recognized by the driver, and any mixer (alsamixer, kmix) also recognizes the card. But playing back isn't possible. I tried speaker-test, audacity, amarok and aplay. Most applications hang, some also produce an error in the /var/log/messages (see below). I then used "./configure --with-debug=full --with-cards=ctxfi" due to the possible hd-audio conflict, but it didn't make a difference. Is there anything else I could try? Christian --- Console output from speaker-test --- bash # LANG=en speaker-test speaker-test 1.0.18 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 512 to 65536 Period size range from 32 to 32768 Using max buffer size 65536 Periods = 4 was set period_size = 16384 was set buffer_size = 65536 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Operation not permitted ------------------------------------------------------------------------ --- output in /var/log/messages when running speaker-test --- Jun 27 12:52:53 firefly kernel: ALSA /home/chris/Desktop/alsa- driver/acore/../alsa-kernel/core/pcm_lib.c:1693: playback write error (DMA or IRQ trouble?) ------------------------------------------------------------------------ ---output in /var/log/messages when running amarok with pulseudio --- 20 11:11:00 firefly pulseaudio[5649]: module-alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Most likely this is an ALSA driver bug. ------------------------------------------------------------------------ --- Hardware information --- firefly:~ # lspci -nn | grep -i audio 00:07.0 Audio device [0403]: nVidia Corporation MCP65 High Definition Audio [10de:044a] (rev a1) 01:07.0 Multimedia audio controller [0401]: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] [1013:6003] (rev 01) 05:00.0 Audio device [0403]: Creative Labs Device [1102:000b] (rev 03) --- Hardware information from lshal --- udi = '/org/freedesktop/Hal/devices/pci_1102_b_sound_card_0' info.capabilities = {'sound'} (string list) info.category = 'sound' (string) info.parent = '/org/freedesktop/Hal/devices/pci_1102_b' (string) info.product = 'Creative X-Fi 20K2 SB0880 Sound Card' (string) info.subsystem = 'sound' (string) info.udi = '/org/freedesktop/Hal/devices/pci_1102_b_sound_card_0' (string) linux.hotplug_type = 2 (0x2) (int) linux.subsystem = 'sound' (string) linux.sysfs_path = '/sys/devices/pci0000:00/0000:00:0e.0/0000:05:00.0/sound/card0' (string) sound.card = 0 (0x0) (int) sound.card_id = 'Creative X-Fi 20K2 SB0880' (string) sound.originating_device = '/org/freedesktop/Hal/devices/pci_1102_b' (string)

2 5

[alsa-devel] Problems with snd-hda-intel and Asus M2NPV-VM. Please help!
by VDR User 08 Jul '09

08 Jul '09

I posted this to the alsa-user mailing list but since then I've read that the problem may actually be a bug in alsa so I've decided to forward the posting here as well. Thanks for your consideration. Best regards, Derek ---------- Forwarded message ---------- From: VDR User <user.vdr(a)gmail.com> Date: Wed, Jun 24, 2009 at 11:08 AM Subject: Problems with snd-hda-intel and Asus M2NPV-VM. Please help! To: "mailing list: alsa" <alsa-user(a)lists.sourceforge.net> Greetings. I've been trying for several days now to get the internal spdif working and am posting here as a last resort in hopes someone will know what's wrong. I'll post system details at the end. My mainboard has 3 1/8" plugs on the back and an internal spdif_out connector. If I plug say headphones into the green connector on the back, I hear sound in them. I don't actually want to use analog sound at all. Instead I've plugged the internal spdif_out+ground into my video cards spdif input. Then I have the dvi output from the video card going to an hdmi input on my tv. Note, I have the same setup (except for different mainboard) in another box and have used it to test the video card, which worked fine. The problem is spdif doesn't seem to be working at all. I've eliminated everything else down to either a problem in alsa somewhere, or possibly my xine settings (though I doubt it since I used the same settings from my working box). For alsa I've tried loading the snd_hda_intel driver with "model=3stack", "model=3stack position_fix=1", and "model=3stack position_fix=1 enable_msi=1". None of these worked, I still get nothing on spdif out. I've also tried (un)muting tons of combinations in alsamixer. Could the problem be my kernel version? Alsa version? libasound? Any ideas at all guys?? Any and all help is GREATLY appreciated! Best regards, Derek System details: Please note there is no desktop/windows manager. There is also no keyboard/mouse plugged in. This box is a dedicated htpc running VDR and connected directly to a tv. I'm using PuTTy SSH from my Windows desktop for maintenance. debian testing kernel 2.6.30 asus m2npv-vm mainboard (tried both firmware 0901 and 1401) - uses mcp51 chipset asus EN8400GS Silent 512MB pci-e video card (with spdif input connector) vdr-1.7.8 xine-0.9.3 xine-vdpau r273 ii alsa-base 1.0.20+dfsg-1 ALSA driver configuration files ii alsa-utils 1.0.20-1 ALSA utilities ii libesd-alsa0 0.2.41-4 Enlightened Sound Daemon (ALSA) - Shared lib ii libsdl1.2debian-alsa 1.2.13-4+b1 Simple DirectMedia Layer (with X11 and ALSA htpc:~$ sudo alsactl init Unknown hardware: "HDA-Intel" "Analog Devices AD1986A" "HDA:11d41986,104381b3,00100500" "0x1043" "0x81cb" Hardware is initialized using a guess method htpc:~$ cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe024000 irq 22 htpc:~$ cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer alsamixer shows: Card: HDA NVidia Chip: Analog Devices AD1986A Master Headphon PCM Front Surround Center LFE Line CD Mic Mic Boos IEC958 IEC958 D Aux Mono Beep Channel Stereo D (everything is unmuted except stereo downmix) htpc:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: AD198x Analog [AD198x Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: AD198x Digital [AD198x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 htpc:~$ aplay -L default:CARD=NVidia HDA NVidia, AD198x Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, AD198x Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=NVidia,DEV=0 HDA NVidia, AD198x Digital IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) htpc:~$ cat /proc/asound/card0/codec#0 Codec: Analog Devices AD1986A Address: 0 Function Id: 0x1 Vendor Id: 0x11d41986 Subsystem Id: 0x104381b3 Revision Id: 0x100500 No Modem Function Group found Default PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 GPIO: io=0, o=1, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital Converter: stream=5, channel=0 Digital: Enabled GenLevel Digital category: 0x2 PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 Delay: 3 samples Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Processing caps: benign=1, ncoeff=70 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=0, channel=0 Power: setting=D0, actual=D0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Converter: stream=12, channel=0 Power: setting=D0, actual=D0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: setting=D0, actual=D0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f 0x20 0x1d 0x1d 0x27* 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20 0x1c* 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo R/L Connection: 2 0x0f* 0x2b Processing caps: benign=1, ncoeff=0 Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x08 0x08] Connection: 8 0x11* 0x22 0x00 0x21 0x10 0x07 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x17 0x17] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x80] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x0000001f: OUT HP Detect Trigger ImpSense Pin Default 0x02214021: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x2, Sequence = 0x1 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x0001001f: OUT HP EAPD Detect Trigger ImpSense EAPD 0x2: EAPD Pin Default 0x01014011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x91 0x91] Pincap 0x00000037: IN OUT Detect Trigger ImpSense Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out R/L Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Pincap 0x00001737: IN OUT Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x01019015: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x1, Sequence = 0x5 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x03] Pincap 0x00000010: OUT Pin Default 0x501700f0: [N/A] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x02a190f0: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001727: IN Detect Trigger ImpSense Vref caps: HIZ 50 GRD 80 Pin Default 0x018130f0: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00000027: IN Detect Trigger ImpSense Pin Default 0x509700f0: [N/A] Aux at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x993310f0: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x50b700f0: [N/A] Telephony at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x90f700f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x014510f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: setting=D0, actual=D0 Connection: 8 0x07 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f

3 9

[alsa-devel] Third time: Unlink PCM resource ?
by Werner Van Belle 07 Jul '09

07 Jul '09

Hello, This is the third time I tried to get information on the same topic but there has been no response. Since it is crucial for me to get this to work, I really would appreciate your input. If necssary I can provide further documentation. The problem is as follows: if a process which opened a DSP device forks of a new process, then this child will have the same handle towards the DSP. What I now want to do is to release the handle in the child without closing the device for the parent as well. So essentially the sequence below illustrates the problem if one is unable to 'unlink' a dsp device. a- parent process creates and open dsp device b- parent process forks of a child 1. Child 1 has access to the same dsp device c- parent process closes the DSP device d- client process can no longer write to the DSP device (these are already odd semantics, but okay, can live with it) e- parent process forks of a second child. Child 2 has no dsp device f- Child 2 tries to open the DSP device. This fails because the device, although closed, is not released yet by Child 1. Is this normal behavior ? Is there some rigid documentation on the closing and sharing semantics of pcm devices ? Is there a way to unlink a device without closing it ? If this is the wrong mailinglist, then who should I contact regarding these erratic semantics ? Wkr, -- Dr. Werner Van Belle http://werner.yellowcouch.org/

3 18

[alsa-devel] [RFC] cmi8330: Add basic CMI8329 support
by Ondrej Zary 05 Jul '09

05 Jul '09

Add basic support for CMI8329 cards. It seems to be very similar to CMI8330, but the PnP devices are different: - AD1848 (WSS) and SB16 device IDs are swapped - the OPL3 port is contained in SB16 device, not AD1848 (WSS) - there is A@@0001 device (looks like some control interface - it has only one 8-bit I/O address) Is my detection code correct, shouldn't the struct pnp_dev returned pnp_request_card_device() be released somehow? Also the mixer seems to be a bit different - master volume does not work currently. My card includes wavetable MIDI - it works in Windows when MPU-401 is selected. It does not work with this driver - pmidi plays but no sound comes out. OPL3 works. My card has also an IDE interface but it's not visible as PnP device and it does not work even in Windows (and I'm unable to find any driver for it for any OS). Here's a scan of the card: http://www.rainbow-software.org/images/hardware/cmi8329a_.jpg The main chip is labelled AD Chips CMI8329A. --- linux-test/sound/isa/cmi8330.c 2009-06-27 16:32:13.000000000 +0200 +++ linux-2.6.30-pentium/sound/isa/cmi8330.c 2009-06-27 16:33:25.000000000 +0200 @@ -156,6 +156,11 @@ typedef int (*snd_pcm_open_callback_t)(struct snd_pcm_substream *); +enum card_type { + CMI8329, + CMI8330 +}; + struct snd_cmi8330 { #ifdef CONFIG_PNP struct pnp_dev *cap; @@ -172,6 +177,8 @@ snd_pcm_open_callback_t open; void *private_data; /* sb or wss */ } streams[2]; + + enum card_type type; }; #ifdef CONFIG_PNP @@ -328,12 +335,26 @@ { struct pnp_dev *pdev; int err; + const char *sb16_id, *wss_id; + + /* CMI8329 has a device with ID A@@0001, CMI8330 does not */ + pdev = pnp_request_card_device(card, "A@@0001", NULL); + if (pdev) { + acard->type = CMI8329; + pnp_release_card_device(pdev); + sb16_id = id->devs[0].id; + wss_id = id->devs[1].id; + } else { + acard->type = CMI8330; + sb16_id = id->devs[1].id; + wss_id = id->devs[0].id; + } - acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL); + acard->cap = pnp_request_card_device(card, wss_id, NULL); if (acard->cap == NULL) return -EBUSY; - acard->play = pnp_request_card_device(card, id->devs[1].id, NULL); + acard->play = pnp_request_card_device(card, sb16_id, NULL); if (acard->play == NULL) return -EBUSY; @@ -351,7 +372,8 @@ wssport[dev] = pnp_port_start(pdev, 0); wssdma[dev] = pnp_dma(pdev, 0); wssirq[dev] = pnp_irq(pdev, 0); - fmport[dev] = pnp_port_start(pdev, 1); + if (acard->type == CMI8330) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate SB16 resources */ pdev = acard->play; @@ -365,6 +387,8 @@ sbdma8[dev] = pnp_dma(pdev, 0); sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); + if (acard->type == CMI8329) + fmport[dev] = pnp_port_start(pdev, 1); /* allocate MPU-401 resources */ pdev = acard->mpu; -- Ondrej Zary

2 3

[alsa-devel] ESI Maya44 Driver
by Dany Svirsky 04 Jul '09

04 Jul '09

Hi All, This subject was disscused before, but it has been very long time without any updates. It been more then half a year since the patch is out. Was someone able to contact Rainer? If not what can be done to finally merge the patch? Is it possible for someone to take over the development of the driver? Thanks, Dany

4 5

[alsa-devel] Conexant CX20582 HDA codec support
by Daniel Drake 03 Jul '09

03 Jul '09

Hi, I'm working on some sound issues with the new OLPC XO-1.5 alpha board. This laptop is now based on a VIA VX855 chipset and our board includes a CX20582-10Q HDA codec. We are running kernel v2.6.30. We have a few issues, which are: 1. Volume range is ridiculous, you can go all the way down to -74dB gain. Beyond -42dB you can barely hear anything from the speakers, and I note that all my other systems only go down as far as -46.5dB. 2. No microphone automute. The internal microphone continues recording sound while there is an external one plugged in. 3. No speaker automute. The internal speakers continue playing sound while there are headphones plugged in. 4. Mixer arrangement is odd. There are 3 microphones: Mic, Mic 1, and Mic 2. Mic (the default) records nothing. Mic 1 and Mic 2 both seem to behave the same, and record from both internal and external microphones. I looked into adding some quirks into patch_conexant.c to hopefully solve some of the above. However I note that there is no support for the CX20582, hence the generic HDA parser is being used. How has support for the other CX chips been developed -- are datasheets always required? (I'll try and get hold of one anyway) Here is some more info. codec dump http://dev.laptop.org/~dsd/20090629/codec0.txt codec graph http://dev.laptop.org/~dsd/20090629/codecgraph.svg lspci entry http://dev.laptop.org/~dsd/20090629/lspci-entry.txt verbose dmesg from probing http://dev.laptop.org/~dsd/20090629/dmesg-probe.txt Any thoughts/input appreciated! Thanks, Daniel

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Re: [alsa-devel] MCP7A HDMI passthrough audio Linux/ALSA
by Stephane BERTHELOT 03 Jul '09

03 Jul '09

Takashi Iwai a écrit : > At Tue, 30 Jun 2009 15:04:05 +0200, > Stéphane BERTHELOT wrote: > >> Hi Takashi, >> >> Thanks for you answer. >> I have the exact same problem and configuration than reported here >> https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4406 >> apart : >> - I'm using a ION platform (Acer Revo R3600 but still MCP7A : 0x10de0007) >> - I never get the Dolby Digital or DTS indicator illuminated on my Amp. >> - My TV is a Sharp LC-52X20E >> >> I thought this was a known problem and than someone at Nvidia could >> provide the missing HDA Verbs, if you prefer I can post to the >> alsa-devel mailing list to investigate further. >> I can get a alsa-info output tonight, though it's very very similar to >> the one in the bug report. (and Raymond was saying that we should get >> "PCM AC3" on digital outputs) >> > > Try ac3dec for a primary testing instead of mplayer. It's included in > alsa-tools. > > % ac3dec -C -H foo.ac3 > > Use the latest one so that it accespt -H option. > > > Takashi > > > Thank you Takashi for taking time to review my problem. I cc'ed alsa-devel since I have more info now and maybe some other people have the same problem or already a workaround. I'm attaching the output of alsa-info.sh with my mixer settings and after having run ac3dec -H -C file.ac3 It works with -R instead of -C but does not work with -C or with -P (consumer or pro) I also tried modifying settings using iecset (audio off / audio on and consumer/pro) and no combination seems to work. When I play stereo LPCM my TV set plays sound and so does my Amp in PCM mode. When I try to play in AC3 mode (-C) my TV set says "incompatible format received" and both the TV and Amp stay silent. I may try again on XP/Vista tomorrow to be really sure it works on them. When changing play mode AES0 changes from 0x00 to 0x02 and AES1 to AES3 stay the same (0x82,0x00,0x02 if I remember well) Is there any other test you want me to do, I'll have time tomorrow evening (CEST). Thanks a lot, Stephane. (http://www.alsa-project.org/db/?f=cf3bc4202b4fcbf93ba68db55e492d51a3e46177) !!################################ !!ALSA Information Script v 0.4.56 !!################################ !!Script ran on: Tue Jun 30 20:22:23 UTC 2009 !!Linux Distribution !!------------------ Slackware 13.0.0.0.0 !!Kernel Information !!------------------ Kernel release: 2.6.29.5 Operating System: GNU/Linux Architecture: x86_64 Processor: Intel(R) Atom(TM) CPU 230 @ 1.60GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.18 Utilities version: 1.0.18 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfae78000 irq 22 !!PCI Soundcards installed in the system !!-------------------------------------- 00:08.0 Audio device: nVidia Corporation MCP79 High Definition Audio (rev b1) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 1025:0222 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC662 rev1 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0662 Subsystem Id: 0x10250222 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Converter: stream=0, channel=0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x89 0x89] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x09, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 9 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 Node 0x0c [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x04 0x0b Node 0x0f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0001003c: IN OUT HP EAPD Detect EAPD 0x0: Pin Default 0x01214010: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0xc0: OUT HP Unsolicited: tag=00, enabled=0 Connection: 1 0x0c Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x00010034: IN OUT EAPD Detect EAPD 0x0: Pin Default 0x99130120: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00001734: IN OUT Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 1 0x0e Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1a [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x00000034: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 1 0x0d Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000173c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 2 0x0c* 0x0e Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x40020601: [N/A] Line Out at Ext N/A Conn = 1/4, Color = Unknown DefAssociation = 0x0, Sequence = 0x1 Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=12 Processing Coefficient: 0x00 Coefficient Index: 0x06 Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x26 [Vendor Defined Widget] wcaps 0xf00000: Mono Codec: Nvidia MCP7A HDMI Address: 3 Function Id: 0x1 Vendor Id: 0x10de0007 Subsystem Id: 0x10de0101 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x0]: bits [0x0]: formats [0x0]: Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x04 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Non-Audio Pro Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x05 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x18560110: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x04 Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x07 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560121: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x1 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x06 Node 0x08 [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x09 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560122: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x08 Node 0x0a [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0b [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560123: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x3 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0a Node 0x0c [Audio Output] wcaps 0x211: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0xc0]: 48000 88200 bits [0xf]: 8 16 20 24 formats [0x1]: PCM Node 0x0d [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000014: OUT Detect Pin Default 0x58560124: [N/A] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x2, Sequence = 0x4 Misc = NO_PRESENCE Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 1 0x0c --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jul 1 2009 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jul 1 2009 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 7 Jul 1 2009 /dev/snd/hwC0D3 crw-rw---- 1 root audio 116, 24 Jul 1 2009 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jun 30 22:11 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 19 Jun 30 22:21 /dev/snd/pcmC0D3p crw-rw---- 1 root audio 116, 1 Jun 30 22:10 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jul 1 2009 /dev/snd/timer !!ALSA configuration files !!------------------------ !!System wide config file (/etc/asound.conf) #pcm.!default { # type plug # slave.pcm "hdmi" #} pcm.hdmiconv { type plug slave { pcm "hw:0,3" rate 48000 } } pcm.headset { type bluetooth } ctl.headset { type bluetooth } !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 Analog [ALC662 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 3: NVIDIA HDMI [NVIDIA HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC662 Analog [ALC662 Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xfae78000 irq 22' Mixer name : 'Nvidia MCP7A HDMI' Components : 'HDA:10ec0662,10250222,00100101 HDA:10de0007,10de0101,00100100' Controls : 18 Simple ctrls : 9 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 52 [81%] [-12.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] [on] Front Right: Playback 0 [0%] [-64.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 213 [84%] [-8.40dB] Front Right: Playback 213 [84%] [-8.40dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 52 [81%] [-12.00dB] [on] Front Right: Playback 52 [81%] [-12.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-13.50dB] [on] Front Right: Capture 0 [0%] [-13.50dB] [on] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 52 value.1 52 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Headphone Playback Volume' value.0 0 value.1 0 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mic Boost' value.0 0 value.1 0 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -1350 comment.dbmax 3300 iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 true value.1 true } control.12 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 52 } control.13 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.14 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.15 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.16 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0782000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.18 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 213 value.1 213 } } --endcollapse-- !!All Loaded Modules !!------------------ Module snd_seq_dummy snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_pcm_oss snd_mixer_oss ipv6 ppdev lp parport_pc parport fuse snd_hda_codec_nvhdmi nvidia snd_hda_codec_realtek ath5k mac80211 rfkill led_class snd_hda_intel cfg80211 snd_hda_codec psmouse serio_raw shpchp snd_hwdep snd_pcm snd_timer sg snd soundcore forcedeth snd_page_alloc rtc_cmos rtc_core thermal wmi rtc_lib processor button thermal_sys hwmon evdev

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Re: [alsa-devel] No sound with nVidia Corporation MCP79 High Definition Audio (rev b1) (Macbook Pro 5, 5)
by Sean Burke 03 Jul '09

03 Jul '09

A quick google shows that the vendor ID is Cirrus Logic. The Boot Camp driver set does include installers for Cirrus audio cards. I don't have Windows installed on here, so I can't check to see if it is indeed the Cirrus drivers that get installed. What can be done to facilitate the implementation of this codec? Sean

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[alsa-devel] RFC: Jack-detection control elements on HD-audio
by Takashi Iwai 03 Jul '09

03 Jul '09

Hi, yet another thing I've worked recently on is a bit more generalized framework for jack-sense reporting for HD-audio. This supersedes the existing jack-sense reporting via the input layer. In addition, it gives the corresponding control elements. For example, you'll have "Jack HP-Out" (or a bit more verbose like "Jack HP-Out at Ext Rear") control via control API as well as other mixer controls. This can notify when a jack is plugged/unplugged. But, another big win is that you can read the current pin sense just by reading this element. I think this is a sort of feature Lennart wanted for PA. Lennart, does it make sense? The test codes are already in topic/hda-jack branch on sound-unstable GIT tree: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git topic/hda-jack Any comments appreciated. This is also intended for 2.6.32, not for 2.6.31 merge, of course :) thanks, Takashi

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[alsa-devel] Problems with snd-hda-intel on hp 6930p (2.6.30)
by Rafael J. Wysocki 02 Jul '09

02 Jul '09

Hi Takashi, I have a problem with snd-hda-intel from 2.6.30 on an hp EliteBook 6930p, which is that the sound is not audible, although everything appears to work just fine. Except maybe that when the box is rebooted and KDE (3.5) is started, the master control is "muted" and the led indicator next to the keyboard shows "no sound". Then, changing the "muted" status of the master control doesn't change the state of the led indicator and vice versa (the two things appear to be totally independent of each other). Can you please tell me what I can do to debug this issue? Best, Rafael

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[alsa-devel] [PATCH] Line In for Acer Inspire 6530G model
by Tony Vroon 01 Jul '09

01 Jul '09

The Line In connector is set up as PIN_IN by default, using VREF_HIZ. It is connected to both ADCs, so add it to both input selectors. Also add the ability to use the input mix (on a SoundBlaster one would call this "What You Hear"). Signed-off-by: Tony Vroon <tony(a)linx.net> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 3345331..67dd883 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1560,18 +1560,22 @@ static struct hda_input_mux alc888_2_capture_sources[2] = { static struct hda_input_mux alc888_acer_aspire_6530_sources[2] = { /* Interal mic only available on one ADC */ { - .num_items = 3, + .num_items = 5, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, { "Int Mic", 0xb }, }, }, { - .num_items = 2, + .num_items = 4, .items = { { "Ext Mic", 0x0 }, + { "Line In", 0x2 }, { "CD", 0x4 }, + { "Input Mix", 0xa }, }, } }; @@ -8189,6 +8193,8 @@ static struct snd_kcontrol_new alc888_acer_aspire_6530_mixer[] = { HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0f, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("LFE Playback Switch", 0x0f, 2, HDA_INPUT), + HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), + HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT),

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[alsa-devel] [PATCH] alsa/soc: add locking to mpc5200-psc-ac97 driver
by Grant Likely 01 Jul '09

01 Jul '09

From: Grant Likely <grant.likely(a)secretlab.ca> AC97 bus register read/write hooks need to provide locking, but the mpc5200-psc-ac97 driver does not. This patch adds a mutex around the register access routines. Signed-off-by: Grant Likely <grant.likely(a)secretlab.ca> --- sound/soc/fsl/mpc5200_dma.c | 1 + sound/soc/fsl/mpc5200_dma.h | 1 + sound/soc/fsl/mpc5200_psc_ac97.c | 17 ++++++++++++++++- 3 files changed, 18 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index efec33a..f0a2d40 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -456,6 +456,7 @@ int mpc5200_audio_dma_create(struct of_device *op) return -ENODEV; spin_lock_init(&psc_dma->lock); + mutex_init(&psc_dma->mutex); psc_dma->id = be32_to_cpu(*prop); psc_dma->irq = irq; psc_dma->psc_regs = regs; diff --git a/sound/soc/fsl/mpc5200_dma.h b/sound/soc/fsl/mpc5200_dma.h index 2000803..8d396bb 100644 --- a/sound/soc/fsl/mpc5200_dma.h +++ b/sound/soc/fsl/mpc5200_dma.h @@ -55,6 +55,7 @@ struct psc_dma { unsigned int irq; struct device *dev; spinlock_t lock; + struct mutex mutex; u32 sicr; uint sysclk; int imr; diff --git a/sound/soc/fsl/mpc5200_psc_ac97.c b/sound/soc/fsl/mpc5200_psc_ac97.c index 794a247..7eb5499 100644 --- a/sound/soc/fsl/mpc5200_psc_ac97.c +++ b/sound/soc/fsl/mpc5200_psc_ac97.c @@ -34,13 +34,20 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) int status; unsigned int val; + mutex_lock(&psc_dma->mutex); + /* Wait for command send status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (rdy)\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } + + /* Force clear the data valid bit */ + in_be32(&psc_dma->psc_regs->ac97_data); + /* Send the read */ out_be32(&psc_dma->psc_regs->ac97_cmd, (1<<31) | ((reg & 0x7f) << 24)); @@ -50,16 +57,19 @@ static unsigned short psc_ac97_read(struct snd_ac97 *ac97, unsigned short reg) if (status == 0) { pr_err("timeout on ac97 read (val) %x\n", in_be16(&psc_dma->psc_regs->sr_csr.status)); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } /* Get the data */ val = in_be32(&psc_dma->psc_regs->ac97_data); if (((val >> 24) & 0x7f) != reg) { pr_err("reg echo error on ac97 read\n"); + mutex_unlock(&psc_dma->mutex); return -ENODEV; } val = (val >> 8) & 0xffff; + mutex_unlock(&psc_dma->mutex); return (unsigned short) val; } @@ -68,16 +78,21 @@ static void psc_ac97_write(struct snd_ac97 *ac97, { int status; + mutex_lock(&psc_dma->mutex); + /* Wait for command status zero = ready */ status = spin_event_timeout(!(in_be16(&psc_dma->psc_regs->sr_csr.status) & MPC52xx_PSC_SR_CMDSEND), 100, 0); if (status == 0) { pr_err("timeout on ac97 bus (write)\n"); - return; + goto out; } /* Write data */ out_be32(&psc_dma->psc_regs->ac97_cmd, ((reg & 0x7f) << 24) | (val << 8)); + + out: + mutex_unlock(&psc_dma->mutex); } static void psc_ac97_warm_reset(struct snd_ac97 *ac97)

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[alsa-devel] [PATCH 1/4] ASoC: pxa-ssp: enhance I2S and add Left_J support
by Eric Miao 01 Jul '09

01 Jul '09

Make the pxa I2S configuration generic, add support for Left_J, add support for variable frame width like 32fs, 48fs, 64fs and 96fs Signed-off-by: Paul Shen <bshen9(a)marvell.com> Signed-off-by: Eric Miao <eric.miao(a)marvell.com> Cc: Daniel Mack <daniel(a)caiaq.de> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 +++--- sound/soc/pxa/pxa-ssp.c | 62 ++++++++++++++-------------- sound/soc/pxa/pxa-ssp.h | 9 ++++ 3 files changed, 47 insertions(+), 38 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 6a2ed35..27f0cd4 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -108,21 +108,21 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ -#if defined(CONFIG_PXA3xx) -#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ -#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ -#endif - #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ -#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ #define SSPSP_SFRMDLY(x) ((x) << 9) /* Serial Frame Delay */ -#define SSPSP_DMYSTRT(x) ((x) << 7) /* Dummy Start */ #define SSPSP_STRTDLY(x) ((x) << 4) /* Start Delay */ #define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ #define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ #define SSPSP_SCMODE(x) ((x) << 0) /* Serial Bit Rate Clock Mode */ +/* NOTE: PXA3xx extends the bit number of dummy start and stop, the macros + * below are compatible with PXA25x/27x as long as the parameter is within + * the correct limits, driver code has to take care of this. + */ +#define SSPSP_DMYSTRT(x) ((((x) & 3) << 7) | ((((x) >> 2) & 3) << 26)) +#define SSPSP_DMYSTOP(x) ((((x) & 3) << 23) | ((((x) >> 2) & 7) << 28)) + #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 6fc7876..2831c16 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -463,7 +463,8 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; + case SND_SOC_DAIFMT_LEFT_J: + sscr0 |= SSCR0_PSP | SSCR0_MOD; sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; /* See hw_params() */ @@ -541,6 +542,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, int chn = params_channels(params); u32 sscr0; u32 sspsp; + int frame_width; int width = snd_pcm_format_physical_width(params_format(params)); int ttsa = ssp_read_reg(ssp, SSTSA) & 0xf; @@ -585,40 +587,38 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); - - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ + sspsp = ssp_read_reg(ssp, SSPSP); + frame_width = PXA_SSP_FRM_WIDTH(priv->dai_fmt); -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else + if (frame_width < width * 2) return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. + + if (frame_width == width * 2) + /* frame width is exactly double of data sample width, + * use FSRT instead */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); + sspsp |= SSPSP_FSRT | SSPSP_SFRMWDTH(width); + else { sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP((frame_width / 2 - width - 1)); + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + } + + ssp_write_reg(ssp, SSPSP, sspsp); + break; + + case SND_SOC_DAIFMT_LEFT_J: + sspsp = ssp_read_reg(ssp, SSPSP); + frame_width = PXA_SSP_FRM_WIDTH(priv->dai_fmt); + + if (frame_width < width * 2) + return -EINVAL; + + if (frame_width == width * 2) + sspsp |= SSPSP_SFRMWDTH(width); + else { + sspsp |= SSPSP_DMYSTOP((frame_width / 2 - width)); + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); } ssp_write_reg(ssp, SSPSP, sspsp); diff --git a/sound/soc/pxa/pxa-ssp.h b/sound/soc/pxa/pxa-ssp.h index 91deadd..fda51d0 100644 --- a/sound/soc/pxa/pxa-ssp.h +++ b/sound/soc/pxa/pxa-ssp.h @@ -40,6 +40,15 @@ #define PXA_SSP_CLK_SCDB_1 1 #define PXA_SSP_CLK_SCDB_8 2 +/* frame size definitions for I2S and Left_J formats - default is + * 32fs, other possibilities are 48fs, 64fs and 96fs + */ +#define PXA_SSP_FRM_32FS (0 << 16) +#define PXA_SSP_FRM_48FS (1 << 16) +#define PXA_SSP_FRM_64FS (2 << 16) +#define PXA_SSP_FRM_96FS (3 << 16) +#define PXA_SSP_FRM_WIDTH(x) (((((x) >> 16) & 0x3) + 2) << 4) + #define PXA_SSP_PLL_OUT 0 extern struct snd_soc_dai pxa_ssp_dai[4]; -- 1.6.0.4

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