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[alsa-devel] alsa-lib module path question
by Ben C 16 Nov '07

16 Nov '07

Hello, I've written an ALSA external PCM plugin for a client's embedded device. During the client's compilation of alsa-lib (which I have no control over), they did not specify where the module directory should be at. In fact, they're using 1.0.13 where there isn't a configure option to specify the module directory (this was added 1.0.14?). Since this is an embedded project, due to cross compilation, alsa-lib is currently looking at some host directory that does not exist on the target to search for the libasound_module_pcm_xxx.so files. What's worse is that seems like it is dependent on where the client decides to build the RFS (say from /home/joe/project/xxx, but then later /home/bill/project/yyy) -- the alsa-lib module directory could change. For now I've manually created the non-existent host directory on the target and dropped a symbolic link there. But since that could change should they decide to rebuild the RFS, that's not even a viable bad hack solution. Is there any environment variable, configuration option or some other methods where this could be overwritten at runtime? Thank you, Ben ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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Re: [alsa-devel] [BUG] New Kernel Bugs
by Neil Brown 16 Nov '07

16 Nov '07

On Wednesday November 14, bfields(a)fieldses.org wrote: > On Wed, Nov 14, 2007 at 09:38:20AM -0800, Randy Dunlap wrote: > > On Wed, 14 Nov 2007 15:08:47 +0100 Ingo Molnar wrote: > > > so please stop this "too busy and too noisy" nonsense already. It was > > > nonsense 10 years ago and it's nonsense today. In 10 years the kernel > > > grew from a 1 million lines codebase to an 8 million lines codebase, so > > > what? Deal with it and be intelligent about filtering your information > > > influx instead of imposing a hard pre-filtering criteria that restricts > > > intelligent processing of information. > > > > So you have a preferred method of handling email. Please don't > > force it on the rest of us. > > I'd be curious for any pointers on tools, actually. I "read" (ok, skim) > lkml but still overlook relevant bug reports occasionally. > (Fortunately, between Trond and Andrew and others forwarding things it's > not actually a problem, but I'm still curious). Virtual Folders. I use VM mode in EMACS, but I believe some other mail readers have the same functionality. I have a virtual folder called "nfs" which shows me all mail in my inbox which has the string 'nfs' or 'lockd' in a To, Cc, or Subject field. When I visit that folder, I see all mail about nfs, whether it was sent to me personally, or to a relevant list, or to lkml. Admittedly if someone doesn't bother to choose a meaningful Subject, then I might miss that. I think this mostly happens when Andrew sends a "-mm" announcement, asked people to change the subject line when following up, and someone follows up without changing the subject line and say "NFS doesn't work any more". I have another virtual folder which matches "md" and "raid" and "mdadm" in any header (so when the people from coraid.com talk about ATA over Ethernet, that gets badly filed, but it is a small cost). Then I have the "bkernel" (boring kernel) folder for all mail from lkml that doesn't mention nfs or raid or md, and isn't from or to me. That folder I skim every week or so and just read the juicy debates and look for interesting tidbits from interesting people - then delete the whole folder, mostly unread. I don't think I could cope with mail without virtual folders. NeilBrown

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[alsa-devel] Mitac 8252D hda_intel
by glv 15 Nov '07

15 Nov '07

(Second attempt, hope this works) Hello, I have a Mitac 8252D computer with hda_intel, it has 3 sockets in front: microphone, audio(headphones)and S/PDIF. When using 5.1 the manual says: Microphone: rear speakers Audio: front speakers S/PDIF: center and subwoofer Modprobing without any options gives me sound to the headphone and digital socket and Modprobing with model=6stack-hp gives me sound to the speakers to but they stay on when headph. are pluged in Output from alsa-info.sh: http://pastebin.ca/772406 Output from lspci -vv: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Mitac Unknown device 8253 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fc200000 (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 Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) thank you.

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[alsa-devel] Removal of indirect access in control API
by Takashi Iwai 15 Nov '07

15 Nov '07

Hi, ALSA control elements have "indirect" flag, which is supposed to allow the indirect access via pointer. But, as far as I know, this feature has never been used nor implemented, and unsupported via 32/64bit compat layer. So, I'd like to get rid of this now. The clean-up patch is below. If you have objections, please let me know. thanks, Takashi --- diff -r 34e3c66d7a5c core/control.c --- a/core/control.c Mon Nov 12 18:04:54 2007 +0100 +++ b/core/control.c Mon Nov 12 18:05:16 2007 +0100 @@ -232,8 +232,6 @@ struct snd_kcontrol *snd_ctl_new1(const access = ncontrol->access == 0 ? SNDRV_CTL_ELEM_ACCESS_READWRITE : (ncontrol->access & (SNDRV_CTL_ELEM_ACCESS_READWRITE| SNDRV_CTL_ELEM_ACCESS_INACTIVE| - SNDRV_CTL_ELEM_ACCESS_DINDIRECT| - SNDRV_CTL_ELEM_ACCESS_INDIRECT| SNDRV_CTL_ELEM_ACCESS_TLV_READWRITE| SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK)); kctl.info = ncontrol->info; @@ -692,7 +690,7 @@ int snd_ctl_elem_read(struct snd_card *c struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; unsigned int index_offset; - int result, indirect; + int result; down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); @@ -701,17 +699,12 @@ int snd_ctl_elem_read(struct snd_card *c } else { index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - indirect = vd->access & SNDRV_CTL_ELEM_ACCESS_INDIRECT ? 1 : 0; - if (control->indirect != indirect) { - result = -EACCES; - } else { - if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && kctl->get != NULL) { - snd_ctl_build_ioff(&control->id, kctl, index_offset); - result = kctl->get(kctl, control); - } else { - result = -EPERM; - } - } + if ((vd->access & SNDRV_CTL_ELEM_ACCESS_READ) && + kctl->get != NULL) { + snd_ctl_build_ioff(&control->id, kctl, index_offset); + result = kctl->get(kctl, control); + } else + result = -EPERM; } up_read(&card->controls_rwsem); return result; @@ -748,7 +741,7 @@ int snd_ctl_elem_write(struct snd_card * struct snd_kcontrol *kctl; struct snd_kcontrol_volatile *vd; unsigned int index_offset; - int result, indirect; + int result; down_read(&card->controls_rwsem); kctl = snd_ctl_find_id(card, &control->id); @@ -757,23 +750,19 @@ int snd_ctl_elem_write(struct snd_card * } else { index_offset = snd_ctl_get_ioff(kctl, &control->id); vd = &kctl->vd[index_offset]; - indirect = vd->access & SNDRV_CTL_ELEM_ACCESS_INDIRECT ? 1 : 0; - if (control->indirect != indirect) { - result = -EACCES; + if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) || + kctl->put == NULL || + (file && vd->owner && vd->owner != file)) { + result = -EPERM; } else { - if (!(vd->access & SNDRV_CTL_ELEM_ACCESS_WRITE) || - kctl->put == NULL || - (file && vd->owner != NULL && vd->owner != file)) { - result = -EPERM; - } else { - snd_ctl_build_ioff(&control->id, kctl, index_offset); - result = kctl->put(kctl, control); - } - if (result > 0) { - up_read(&card->controls_rwsem); - snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &control->id); - return 0; - } + snd_ctl_build_ioff(&control->id, kctl, index_offset); + result = kctl->put(kctl, control); + } + if (result > 0) { + up_read(&card->controls_rwsem); + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, + &control->id); + return 0; } } up_read(&card->controls_rwsem); diff -r 34e3c66d7a5c include/asound.h --- a/include/asound.h Mon Nov 12 18:04:54 2007 +0100 +++ b/include/asound.h Mon Nov 12 18:05:16 2007 +0100 @@ -738,8 +738,7 @@ typedef int __bitwise snd_ctl_elem_iface #define SNDRV_CTL_ELEM_ACCESS_OWNER (1<<10) /* write lock owner */ #define SNDRV_CTL_ELEM_ACCESS_TLV_CALLBACK (1<<28) /* kernel use a TLV callback */ #define SNDRV_CTL_ELEM_ACCESS_USER (1<<29) /* user space element */ -#define SNDRV_CTL_ELEM_ACCESS_DINDIRECT (1<<30) /* indirect access for matrix dimensions in the info structure */ -#define SNDRV_CTL_ELEM_ACCESS_INDIRECT (1<<31) /* indirect access for element value in the value structure */ +/* bits 30 and 31 are obsoleted (for indirect access) */ /* for further details see the ACPI and PCI power management specification */ #define SNDRV_CTL_POWER_D0 0x0000 /* full On */ @@ -793,30 +792,25 @@ struct snd_ctl_elem_info { } value; union { unsigned short d[4]; /* dimensions */ - unsigned short *d_ptr; /* indirect */ } dimen; unsigned char reserved[64-4*sizeof(unsigned short)]; }; struct snd_ctl_elem_value { struct snd_ctl_elem_id id; /* W: element ID */ - unsigned int indirect: 1; /* W: use indirect pointer (xxx_ptr member) */ + unsigned int indirect: 1; /* W: indirect access - obsoleted */ union { union { long value[128]; - long *value_ptr; } integer; union { long long value[64]; - long long *value_ptr; } integer64; union { unsigned int item[128]; - unsigned int *item_ptr; } enumerated; union { unsigned char data[512]; - unsigned char *data_ptr; } bytes; struct snd_aes_iec958 iec958; } value; /* RO */

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[alsa-devel] interrupting snd_pcm_writei or snd_pcm_drain?
by Mike Gorse 15 Nov '07

15 Nov '07

I've noticed that it is unsafe to call snd_pcm_writei or snd_pcm_drain in a pthread with its cancelation type set to asynchronous. snd_pcm_dmix_sync_area can lower a semaphore, and, if the thread is canceled while the semaphore is down, then a subsequent call to the sound device (calling snd_pcm_close, for instance) will also lower the semaphore and cause deadlock. Is this a known issue, and is there a way around it other than to always use deferred cancelation when calling these functions? Thanks, -Mike G-

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Re: [alsa-devel] [BUG] New Kernel Bugs
by Neil Brown 15 Nov '07

15 Nov '07

On Tuesday November 13, vda.linux(a)googlemail.com wrote: > On Tuesday 13 November 2007 07:08, Mark Lord wrote: > > Ingo Molnar wrote: > > .. > > > > > This is all QA-101 that _cannot be argued against on a rational basis_, > > > it's just that these sorts of things have been largely ignored for > > > years, in favor of the all-too-easy "open source means many eyeballs and > > > that is our QA" answer, which is a _good_ answer but by far not the most > > > intelligent answer! Today "many eyeballs" is simply not good enough and > > > nature (and other OS projects) will route us around if we dont change. > > > > .. > > > > QA-101 and "many eyeballs" are not at all in opposition. > > The latter is how we find out about bugs on uncommon hardware, > > and the former is what we need to track them and overall quality. > > > > A HUGE problem I have with current "efforts", is that once someone > > reports a bug, the onus seems to be 99% on the *reporter* to find > > the exact line of code or commit. Ghad what a repressive method. > > This is the only method that scales. That sounds overly hash, and the rest of you mail sounds much more moderate and sensible - I can only assume you were using hyperbole?? Putting the "onus on the reporter" is simply not going to work unless you have a business relationship. In the community, we are all volunteering our time (well, maybe my employer is volunteering my time to do community support, but the effect is the same). I would hope that the focus of developers is to empower bug reporters to provide further information (and as has been said, "git bisect" is a great empowerer). Some people will be incredibly help, especially if you ask politely and say thankyou. Others won't for any of a number of reasons - and maybe that means their bug won't get fixed. To my eyes, the "only method that scales" is investing effort in encouraging and training bug reporters. Some of that effort might not produce results, but when others among those you have encouraged start answering the newbee questions on the list and save you the time, you get a distinct feeling that it was all worth while. I think we are in agreement - I just wanted to take issue with that one sentence :-) The rest is great. NeilBrown > > Developer has only 24 hours in each day, and sometimes he needs to eat, > sleep, and maybe even pay attention to e.g. his kids. > > But bug reporters are much more numerous and they have more > hours in one day combined. > > BUT - it means that developers should try to increase user base, > not scare users away. > > > And if the "developer" who broke the damn thing, or who at least > > "claims" to be supporting that code, cannot "reproduce" the bug, > > they drop it completely. > > Developer should let reporter know that reporter needs to help > a bit here. Sometimes a bit of hand holding is needed, but it > pays off because you breed more qualified testers/bug reporters. > > > Contrast that flawed approach with how Linus does things.. > > he thinks through the symptoms, matches them to the code, > > and figures out what the few possibilities might be, > > and feeds back some trial balloon patches for the bug reporter to try. > > > > MUCH better. > > > > And remember, *I'm* an old-time Linux kernel developer.. just think about > > the people reporting bugs who haven't been around here since 1992.. > > Yes. Developers should not grow more and more unhelpful > and arrogant towards their users just because inexperienced > users send incomplete/poorly written bug reports. > They need to provide help, not humiliate/ignore. > > I think we agree here. > -- > vda > - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/

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[alsa-devel] params_buffer_size() and params_period_bytes() are returning the same value
by Timur Tabi 14 Nov '07

14 Nov '07

I have an ASOC v1 driver that I'm wrapping up. I've noticed something weird in my hw_params() function. I have this code: size_t buffer_size = params_buffer_size(hw_params); size_t period_size = params_period_bytes(hw_params); printk(KERN_INFO "%s: buffer_size=%x period_size=%x\n", __FUNCTION__, buffer_size, period_size); printk(KERN_INFO "%s: params_periods=%u\n", __FUNCTION__, params_periods(hw_params)); And it produces this output: fsl_dma_hw_params: buffer_size=5dc0 period_size=5dc0 fsl_dma_hw_params: params_periods=4 Why am I getting the same value for buffer_size and period_size? buffer_size should be four times period_size. -- Timur Tabi Linux kernel developer at Freescale

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Re: [alsa-devel] [PATCH] ASoC TLV320AIC3X codec driver
by Vladimir A. Barinov 14 Nov '07

14 Nov '07

Hello Takashi, Thank you for the review. Takashi Iwai wrote: > At Tue, 13 Nov 2007 22:41:25 +0300, > Vladimir Barinov wrote: > >> --- /dev/null >> +++ linux-2.6.24.rc2.alsa/sound/soc/codecs/tlv320aic3x.c >> > (snip) > >> + >> +struct snd_soc_codec_device soc_codec_dev_aic3x; >> > > Do you need this here? It's defined in the later position. > You are right. Thanks for pointing to this. That is an artefact of the driver that I based on. > >> +/* >> + * All input lines are connected when !0xf and disconnected with 0xf bit field, >> > > Please keep the line within 80 chars. Try to run checkpatch.pl in > linuxkernel/scripts for checking such minor coding-style issues. > But this line is less then 80 chars :) I've applied checkpatch.pl before sending the patch. The driver really has 3 lines with 80 chars warning, but they are less then 82 char. It's in the codec interconnection map for DAPM. If that is a strong demand with 80 chars _warning_ then I'll try to rename the mixers/path/endpoint names to match, if not then the code will be more understandable. That lines are: + {"Right HPCOM Mux", "differential of HPROUT", "Right Line2 Bypass Mixer"}, + {"Right HPCOM Mux", "differential of HPLCOM", "Right Line2 Bypass Mixer"}, If I'll split it with 2 lines each then will it be good to watch in accordance with DAPM usage strategy? Destination Widget <=== Path Name <=== Source Widget > >> +static const char *aic3x_left_dac_mux[] = { "DAC_L1", "DAC_L3", "DAC_L2" }; >> +static const char *aic3x_right_dac_mux[] = { "DAC_R1", "DAC_R3", "DAC_R2" }; >> +static const char *aic3x_left_hpcom_mux[] = >> + { "differential of HPLOUT", "constant VCM", "single-ended" }; >> +static const char *aic3x_right_hpcom_mux[] = >> + { "differential of HPROUT", "constant VCM", "single-ended", >> + "differential of HPLCOM", "external feedback" }; >> +static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; >> > > Ditto. > Ditto :) These lines are less then 80 chars. > >> +#define AIC3X_RATES SNDRV_PCM_RATE_8000_96000 >> +#define AIC3X_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ >> + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) >> > > Are you sure that it's FMTBIT_S24_LE? It's not packed in 3 bytes but > uses lower 3 bytes of 4 bytes frame. > > Aic3x supports 16/20/24/32 bits data word length and in accordance with aic33 documentation the number of clocks per half-frame for each channel are equal to exact number of bits of the word. Unfortunately, now I have no h/w to test 24bits mode. > >> Index: linux-2.6.24.rc2.alsa/sound/soc/codecs/Kconfig >> =================================================================== >> --- linux-2.6.24.rc2.alsa.orig/sound/soc/codecs/Kconfig >> +++ linux-2.6.24.rc2.alsa/sound/soc/codecs/Kconfig >> @@ -37,3 +37,6 @@ config SND_SOC_CS4270_VD33_ERRATA >> bool >> depends on SND_SOC_CS4270 >> >> +config SND_SOC_TLV320AIC3X >> + tristate >> + depends on SND_SOC >> > > Currently, there is a known problem with dependency on I2C over soc > codecs drivers. If CONFIG_I2C=m and CONFIG_SND_SOC_*=y, you'll get > unresolved symbols. > > I don't know how to resolve this issue right now. But, the drivers > that support *only* I2C right now can add "depends on I2C" to its > Kconfig. That doesn't work for CS4270 which may work without I2C as > standalone mode, though. > Ok, will add I2C dependency. Regards, Vladimir

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[alsa-devel] Naming multiple USB devices (solved)
by Giovanni Maruzzelli 14 Nov '07

14 Nov '07

Hi list, I've added the following script to http://alsa.opensrc.org/index.php/Udev (it's not the most elegant solution, but it works for me): BEWARE of the linebreaks, they are mangled by the mail client ================================= #!/usr/bin/perl # fixed persistent naming for multiple usb soundcards # gmaruzz (at) celliax.org # This is to be executed by udev with the following rules: # KERNEL=="controlC[0-9]*", DRIVERS=="usb",PROGRAM="/usr/bin/alsa_name.pl %k", NAME="snd/%c{1}" # KERNEL=="hwC[D0-9]*", DRIVERS=="usb", PROGRAM="/usr/bin/alsa_name.pl %k", NAME="snd/%c{1}" # KERNEL=="midiC[D0-9]*", DRIVERS=="usb",PROGRAM="/usr/bin/alsa_name.pl %k", NAME="snd/%c{1}" # KERNEL=="pcmC[D0-9cp]*", DRIVERS=="usb",PROGRAM="/usr/bin/alsa_name.pl %k", NAME="snd/%c{1}" use strict; use warnings; my $alsaname = $ARGV[0]; #udev called us with this argument (%k) my $physdevpath = $ENV{PHYSDEVPATH}; #udev put this in our environment my $alsanum = "cucu"; #you can find the physdevpath of a device with "udevinfo -a -p $(udevinfo -q path -n /dev/snd/pcmC0D0c)" $physdevpath =~ s/.*\/([^\/]*)/$1/; #eliminate until last slash $physdevpath =~ s/([^:]*):.*/$1/; #eliminate from twopoint (:) to end_of_line if($physdevpath eq "1-5.2") # you can find this value with "dmesg" { $alsanum="11"; #start from "10" (easier for debugging), "0" is for motherboard soundcard, max is "31" } if($physdevpath eq "1-5.3") # you can find this value with "dmesg" { $alsanum="12"; #start from "10" (easier for debugging), "0" is for motherboard soundcard, max is "31" } if($physdevpath eq "1-5.4") # you can find this value with "dmesg" { $alsanum="13"; #start from "10" (easier for debugging), "0" is for motherboard soundcard, max is "31" } if($physdevpath eq "3-2") # you can find this value with "dmesg" { $alsanum="14"; #start from "10" (easier for debugging), "0" is for motherboard soundcard, max is "31" } # other bus positions.... if($alsanum ne "cucu") { $alsaname=~ s/(.*)C([0-9]+)(.*)/$1C$alsanum$3/; } print $alsaname; exit 0; ======================= Ciao, Giovanni On Feb 10, 2007 8:31 PM, Jean-Michel Pouré <jmpoure(a)free.fr> wrote: > Le samedi 10 février 2007 à 18:53 +0100, Ingo Müller a écrit : > > I'm going to play with udev now, too, so if I find something, I'll > > tell > > you. In return, it would be awesome if you could write down - even it > > is > > only short - a howto about how to manage two identical devices, once > > we > > got it working. > > > > Fantastic. > > Thank you very much for your help. > I will add a page on the wiki about udev rules. > > Kind regards, > Jean-Michel > > > > ------------------------------------------------------------------------- > Using Tomcat but need to do more? Need to support web services, security? > Get stuff done quickly with pre-integrated technology to make your job easier. > Download IBM WebSphere Application Server v.1.0.1 based on Apache Geronimo > http://sel.as-us.falkag.net/sel?cmd=lnk&kid=120709&bid=263057&dat=121642 > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-devel >

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[alsa-devel] Query regarding snd_pcm_lib_preallocate_pages_for_all()
by Pharaoh . 14 Nov '07

14 Nov '07

Hi, I am trying to use snd_pcm_lib_preallocate_pages_for_all() on an arm based platform for preallocating the DMA buffer. I am using it as snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, NULL, 64 * 1024, 64 * 1024); since I dont know what should be passed as the 3'rd param. I tried passing my device pointer i.e. card->dev after correctly populating it but I get an error as 'DMA coherent mask in not set' and buffer is not allocated. If I pass NULL as shown above, which I think is appropriate, the allocation is done well and I can play some samples too using aplay, but when I do a rmmod all hell breaks free. I am thinking this is corrupting memory some where, since when I preallocate using SNDRV_DMA_TYPE_CONTINUOUS, everything works well but I since I need to mmap the buffer I dont use it. I get following: c001f7e0 c001f7e0 c001f814 c03b3040 0000000b c34c1c98 c34c1c88 1c80: c003642c c0071978 c34c0000 c34c1cb8 c34c1c9c c003b780 c00363fc c34c0000 1ca0: c34c0000 c03b3040 00000001 c34c1cd4 c34c1cbc c003bd94 c003b690 c34c1ce8 1cc0: 60000093 00000001 c34c1ce8 c34c1cd8 c002908c c003bbe8 bf000000 c34c1d08 1ce0: c34c1cec c002b2ac c0028d84 ffffffec c03b3040 c34733c8 c001f7e0 c34c1d40 1d00: c34c1d0c c002b588 c002b250 c03a7060 c34c1d94 00000017 ffffffff c026dd40 1d20: 00000017 c34c1dec 00000004 00000093 40130000 c34c1de8 c34c1d44 c002b688 1d40: c002b3a0 0000007e 00000000 00000000 00000000 00000000 00000000 00000000 1d60: 00000000 c03a7060 c34c1e14 c03b3040 c022eef4 c34c1f14 00000000 00000a00 1d80: c34c1e80 c34c1d90 c019ee84 c01bb390 00000000 7a035438 00000005 00000000 1da0: 00000a00 c03a7060 c0271cdc c34c1d50 00000000 c34c1e08 c34c1dc4 c0063a94 1dc0: c0063984 00000044 ffffffff c34c1e20 00000000 c02eabe0 c347317c c34c1e44 1de0: c34c1dec c00249a0 c002b65c c02eabe0 00000000 00000013 00000093 c34c0000 1e00: c0306200 00000000 c02eabe0 c347317c c001f7e0 40130000 c34c1e44 c02eabe0 1e20: c34c1e34 c02c0bfc c0067478 00000093 ffffffff 10d5f0ff c34c1e7c c34c1e48 1e40: c006d658 c0067444 c3541cc0 00000000 00000002 11b100df 000004c0 c347317c 1e60: c34c0000 00000001 c001f7e0 40130000 c34c1ecc c34c1e80 c006e61c c006d1b8 1e80: c3545000 c001f820 11b100df 0000007e 00000000 00000010 c3545000 00000800 1ea0: c34c0000 ffffffeb c03b3040 c347317c c001f7e0 c001f814 c34c1fb0 40130958 1ec0: c34c1f04 c34c1ed0 c002b470 c006dedc c004822c c03b307c 0000081f ffffffff 1ee0: c026ddc0 0000081f c34c1fb0 40130958 00000055 4012f000 c34c1fac c34c1f08 1f00: c002b688 c002b3a0 c34c1f14 c0222148 c0221a50 c34c1e94 0000000c c34c1e84 1f20: 00000001 00000000 00000000 c1b01040 00000000 00000009 0001d2b4 00000066 1f40: c0024fe4 00000001 c34c1f6c c34c1f58 c01a02c0 c01a01c0 00000000 00000000 1f60: c34c1fa4 c34c1f70 c01a09d0 c01a02b0 c34c1e70 c34c1e74 00000004 0001d2b4 1f80: 00000a00 00000000 0000008e ffffffff 00000115 0001d2dd 00000000 00000000 1fa0: 00000000 c34c1fb0 c0024de8 c002b65c 00000000 0000b05c 54434100 00000001 1fc0: 40130958 00000115 0001d2dd 00000000 00000000 00000055 4012f000 00000007 1fe0: 0001d150 be8ba5ec 0000b05c 40082fbc 60000010 ffffffff 00000000 00000000 Backtrace: [<c0073880>] (page_remove_rmap+0x0/0x118) from [<c006c628>] (unmap_vmas+0x3c4/0x60c) r4 = C02EABE0 [<c006c264>] (unmap_vmas+0x0/0x60c) from [<c00719e8>] (exit_mmap+0x80/0x150) [<c0071968>] (exit_mmap+0x0/0x150) from [<c003642c>] (mmput+0x40/0xe4) r8 = 0000000B r7 = C03B3040 r6 = C001F814 r5 = C001F7E0 r4 = C001F7E0 [<c00363ec>] (mmput+0x0/0xe4) from [<c003b780>] (exit_mm+0x100/0x10c) r4 = C34C0000 [<c003b680>] (exit_mm+0x0/0x10c) from [<c003bd94>] (do_exit+0x1bc/0x9c4) r7 = 00000001 r6 = C03B3040 r5 = C34C0000 r4 = C34C0000 [<c003bbd8>] (do_exit+0x0/0x9c4) from [<c002908c>] (die+0x318/0x36c) [<c0028d74>] (die+0x0/0x36c) from [<c002b2ac>] (__do_kernel_fault+0x6c/0x7c) [<c002b240>] (__do_kernel_fault+0x0/0x7c) from [<c002b588>] (do_page_fault+0x1f8/0x210) r7 = C001F7E0 r6 = C34733C8 r5 = C03B3040 r4 = FFFFFFEC [<c002b390>] (do_page_fault+0x0/0x210) from [<c002b688>] (do_DataAbort+0x3c/0xa0) [<c002b64c>] (do_DataAbort+0x0/0xa0) from [<c00249a0>] (__dabt_svc+0x40/0x60) r8 = C347317C r7 = C02EABE0 r6 = 00000000 r5 = C34C1E20 r4 = FFFFFFFF [<c0067434>] (lru_cache_add_active+0x0/0xa0) from [<c006d658>] (do_wp_page+0x4b0/0x574) r4 = 10D5F0FF [<c006d1a8>] (do_wp_page+0x0/0x574) from [<c006e61c>] (__handle_mm_fault+0x750/0x824) [<c006decc>] (__handle_mm_fault+0x0/0x824) from [<c002b470>] (do_page_fault+0xe0/0x210) [<c002b390>] (do_page_fault+0x0/0x210) from [<c002b688>] (do_DataAbort+0x3c/0xa0) [<c002b64c>] (do_DataAbort+0x0/0xa0) from [<c0024de8>] (ret_from_exception+0x0/0x10) r8 = 00000000 r7 = 00000000 r6 = 0001D2DD r5 = 00000115 r4 = FFFFFFFF Code: e3130901 e1a03004 1594300c e59f008c (e5931004) <1>Fixing recursive fault but reboot is needed! BUG: scheduling while atomic: udevd/0x00000004/245 Bad page state in process 'exe' page:c02eaa60 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaa60 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaa80 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaa80 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaaa0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaaa0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaac0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaac0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaae0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaae0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab00 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab00 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab20 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab20 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab40 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab40 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab60 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab60 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab80 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eab80 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaba0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eaba0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eabc0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: Bad page state in process 'exe' page:c02eabc0 flags:0x00004000 mapping:00000000 mapcount:0 count:0 Trying to fix it up, but a reboot is needed Backtrace: so on

3 4

Re: [alsa-devel] [Pkg-alsa-devel] Bug#450805: alsa-tools-gui: hdspmixer initializes only first Hammerfall DSP card
by Elimar Riesebieter 14 Nov '07

14 Nov '07

forwarded 450805 alsa-devel(a)alsa-project.org tags 450805 pending patch Thanks Hi ALSA developers, does it make sense to apply the patch to HEAD? I don't have a Hammerfall handy and can not test it. Thanks for cooperation. Elimar On Sun, 11 Nov 2007 the mental interface of Frank Heckenbach told: Package: alsa-tools-gui Version: 1.0.13-1 Severity: normal Tags: patch We are using Hammerfall DSP cards. After booting, their audio output remains silent until hdspmixer is started. No interaction in the GUI of hdspmixer is necessary to unmute the first HDSP card; however, further cards are only unmuted when activating the respective GUI page ("2", "3"). Apparently, hdspmixer does some automatic initialization of the card when activating the page. Since we'd like to have a fully automatic startup, the following patch activates the page for each existing card on startup, thereby initializing them. There are surely more elegant solutions, but this patch is tested and solves the problem for us. --- alsa-tools-1.0.13/hdspmixer/src/HDSPMixerCardSelector.h.orig 2007-04-16 19:49:22.000000000 +0200 +++ alsa-tools-1.0.13/hdspmixer/src/HDSPMixerCardSelector.h 2007-04-16 19:50:51.000000000 +0200 @@ -37,6 +37,7 @@ int card; HDSPMixerWindow *basew; HDSPMixerCardSelector(int x, int y, int w, int h, int card); + void ActivateCard (int i); void draw(); int handle(int e); }; --- alsa-tools-1.0.13/hdspmixer/src/HDSPMixerCardSelector.cxx.orig 2007-04-16 19:49:23.000000000 +0200 +++ alsa-tools-1.0.13/hdspmixer/src/HDSPMixerCardSelector.cxx 2007-04-16 19:53:45.000000000 +0200 @@ -45,27 +45,25 @@ } } +void HDSPMixerCardSelector::ActivateCard (int i) +{ + card = i + 1; + basew->current_card = i; + basew->cards[i]->setMode (basew->cards[i]->getSpeed ()); + redraw (); +} + int HDSPMixerCardSelector::handle(int e) { int xpos = Fl::event_x()-x(); switch (e) { case FL_PUSH: - if (xpos < 13 && card != 1) { - card = 1; - basew->current_card = 0; - basew->cards[0]->setMode(basew->cards[0]->getSpeed()); - redraw(); - } else if (xpos >= 24 && xpos < 37 && card != 2 && basew->cards[1] != NULL) { - card = 2; - basew->current_card = 1; - basew->cards[1]->setMode(basew->cards[1]->getSpeed()); - redraw(); - } else if (xpos >= 48 && card != 3 && basew->cards[2] != NULL) { - card = 3; - basew->current_card = 2; - basew->cards[2]->setMode(basew->cards[2]->getSpeed()); - redraw(); - } + if (xpos < 13 && card != 1) + ActivateCard (0); + else if (xpos >= 24 && xpos < 37 && card != 2 && basew->cards[1] != NULL) + ActivateCard (1); + else if (xpos >= 48 && card != 3 && basew->cards[2] != NULL) + ActivateCard (2); return 1; default: return Fl_Widget::handle(e); --- alsa-tools-1.0.13/hdspmixer/src/HDSPMixerWindow.cxx.orig 2007-04-16 19:41:21.000000000 +0200 +++ alsa-tools-1.0.13/hdspmixer/src/HDSPMixerWindow.cxx 2007-04-16 19:55:13.000000000 +0200 @@ -700,6 +701,9 @@ Fl::atclose = atclose_cb; Fl::add_handler(handler_cb); Fl::add_timeout(0.030, readregisters_cb, this); + i = 0; + while (cards[i] != NULL) + inputs->buttons->cardselector->ActivateCard (i++); } int HDSPMixerWindow::handle(int e) -- Experience is something you don't get until just after you need it!

2 1

[alsa-devel] [ALSA] fix private data pointer calculation in CS4270 driver
by Timur Tabi 14 Nov '07

14 Nov '07

Fix the calculation of the private_data pointer in the CS4270 driver. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 5d601ad..abac628 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -725,7 +725,8 @@ static int cs4270_probe(struct platform_device *pdev) codec->owner = THIS_MODULE; codec->dai = &cs4270_dai; codec->num_dai = 1; - codec->private_data = codec + ALIGN(sizeof(struct snd_soc_codec), 4); + codec->private_data = (void *) codec + + ALIGN(sizeof(struct snd_soc_codec), 4); socdev->codec = codec; -- 1.5.2.4

2 1

[alsa-devel] mixer API documentation
by Richard Kettlewell 13 Nov '07

13 Nov '07

Hello, Where is the ALSA mixer API documentation? I looked at http://www.alsa-project.org/alsa-doc/alsa-lib/mixer.html, but that only says "Mixer interface is designed to access the abstracted mixer controls. This is an abstraction layer over the hcontrol layer". ttfn/rjk

2 2

[alsa-devel] [PATCH] ASoC TLV320AIC3X codec driver
by Vladimir Barinov 13 Nov '07

13 Nov '07

Hi Liam, This patch adds ALSA SoC support for TI TLV320AIC3X audio codecs. The features that are supported: o Capture/Playback/Bypass. o 16/20/24/32 bit audio. o 8k - 96k sample rates. o codec master only mode o DAPM. Vladimir

1 0

[alsa-devel] Intel HD Audio
by Chris Pemberton 13 Nov '07

13 Nov '07

I've been trying to fix audio on my Lenovo 3000 N100 laptop for quite some time; on this list and via bug #0003468. I finally got tired of the xorg graphics woes and hacking for sound, so I am now "temporarily" running Vista. I'd like to get this audio fixed permanently, so if you need any details from the working Vista drivers, let me know before I get sick of Windows and switch back to Linux. Chris

2 3

[alsa-devel] New developer question
by Pete Fawcett 13 Nov '07

13 Nov '07

Dear Alsa Developers I am trying to find out how to submit some fixes to ALSA drivers. All I could find on the web site was to subscribe to this mailing list so I hope this is the correct place to raise this. I have made some changes to the driver for AD1988 device. This is an HDA device and handled by the snd_hda_intel module. The specific hardware is handled by patch_analog.c I have made some changes so that 5.1 surround sound is now output on my motherboard. Previously stereo sound was output to all all three pairs of speakers. That's all fine for me, but in the spirit of open source I thought I should submit these changes for wider use. Please advise me how I proceed from here. Thanks Pete

2 1

[alsa-devel] issues with pulse plugin
by Mike Gorse 13 Nov '07

13 Nov '07

I have run into what is basically another instance of bug 3470 (pulse_hw_params asserting because the stream is already initialized). The application that I am working with is calling snd_pcm_hw_params each time it plays a sample. The following patch makes things work for me (actually the app still doesn't work because it uses poll() in a way that is incompatible with the pulse plugin, but that is a separate issue, and I don't know if it should be considered a bug in the plugin.) --- pulse/pcm_pulse.c.orig 2007-11-08 15:13:12.000000000 -0500 +++ pulse/pcm_pulse.c 2007-11-08 15:13:12.000000000 -0500 @@ -497,7 +497,7 @@ assert(pcm->p); //Resolving bugtrack ID 0003470 - if(!(base && snd_pcm_state(base) == SND_PCM_STATE_PREPARED)) + if(!(base && (snd_pcm_state(base) == SND_PCM_STATE_PREPARED || snd_pcm_state(base) == SND_PCM_STATE_SETUP))) assert(!pcm->stream); pa_threaded_mainloop_lock(pcm->p->mainloop); -- I couldn't add this to the bug because it was marked as resolved. Any comments? Or is the app behaving in a way that is counter to the specification? Also, I'm still encountering bug 2601 (a separate assert failure that happens if period_size == 0) even with the patch to pcm_ioplug.c. I think that PERIOD_TIME and PERIODS need to be constrained; I added a couple of comments to the bug. Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

2 3

[alsa-devel] HDA Intel SigmaTel STAC9228 in Dell Vostro 1400 defines different pin nids
by Ivan N. Zlatev 13 Nov '07

13 Nov '07

Hello, I've bought a new laptop Dell Vostro 1400 with the following audio card: 00:1b.0 Audio device [Class 0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 02) and with the SigmaTel Codec: Codec: SigmaTel STAC9228 Address: 0 Vendor Id: 0x83847616 Subsystem Id: 0x10280227 Revision Id: 0x100201 I was able to get sound to come out of the internal speakers by explicitly specifying either the 3stack or the dell-3stack models with alsa 1.0.14 . Unfortunately with neither of those does the internal microphone work. Now I am running ALSA HG and have decided to try and fix my problem by adding the proper pinconfig for my card (this is what I've done before for my Mac and it worked). I downloaded the win32 drivers which include all pinconfigs for all Dell machines with intel hda cards. The problem I am facing is that the win32 driver .ini provides a configuration for 13 pins which do not match the ones in patch_sigmatel.c . The ones in ALSA are: static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, }; Where as the ones defined by the driver are [1]: 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x12, 0x13, 0x14, 0x21, 0x22 So I am wondering how should I approach this? I have attached my codec info and mixer output without the model to be specified. [1] win32 driver pinconfigs [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x02211210 AltCfg = hex: 20,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\Pin\0B] CfgDflt = dword: 0x400003FA [HKR\Settings\Pin\0C] CfgDflt = dword: 0x400003FB [HKR\Settings\Pin\0D] CfgDflt = dword: 0x90170310 ExtAmpCtrl = hex: 02,05,02 [HKR\Settings\Pin\0E] CfgDflt = dword: 0x0281103E Bias = hex: FF,04,02 AltCfg = hex: 31,10,A1,02,22,10,11,02 Action = hex: 0A,0E [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0221122F AltCfg = hex: 21,12,11,02 Action = hex: 80,00,87,00,81,0D [HKR\Settings\Pin\10] CfgDflt = dword: 0x400003FC [HKR\Settings\Pin\11] CfgDflt = dword: 0x400003FD [HKR\Settings\Pin\12] CfgDflt = dword: 0x400003FE [HKR\Settings\Pin\13] CfgDflt = dword: 0x90A60330 [HKR\Settings\Pin\14] CfgDflt = dword: 0x400003FF [HKR\Settings\Pin\21] CfgDflt = dword: 0x04441340 [HKR\Settings\Pin\22] CfgDflt = dword: 0x40C003F1 [HKR\Settings\GPIO] 02 = hex: 0x82 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

2 1

[alsa-devel] aplay: xrun:1104: read/write error ???
by 13 Nov '07

13 Nov '07

Hi All: im using aplay from ALSA-1.0.15 to play music through bluetooth a2dp. im using montavista 2.6.18 on ARM9 platform. I got an "aplay: xrun:1104: read/write error, state = RUNNING" error after the music has been played for one second or two. Could someone tell me how this happened? Here's the command that i use : aplay -v -D bluetooth /1.wav And the message log is as follows: /$ aplay -v -D bluetooth /1.wav audio[842]: Accepted new client connection on unix socket (fd=8) audio[842]: Package PKT_TYPE_CFG_REQ:0 audio[842]: avdtp_ref(0x3c990): ref=2 audio[842]: a2dp_source_request_stream: selected SEP 0x3c218 audio[842]: avdtp_ref(0x3c990): ref=3 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) audio[842]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[842]: session_cb audio[842]: DISCOVER request succeeded audio[842]: seid 1 type 1 media 0 in use 0 audio[842]: seid 2 type 1 media 0 in use 0 audio[842]: session_cb audio[842]: GET_CAPABILITIES request succeeded audio[842]: seid 1 type 1 media 0 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) audio[842]: session_cb audio[842]: GET_CAPABILITIES request succeeded audio[842]: seid 2 type 1 media 0 audio[842]: Discovery complete audio[842]: avdtp_set_configuration(0x3c990): int_seid=1, acp_seid=1 audio[842]: session_cb audio[842]: SET_CONFIGURATION request succeeded audio[842]: SBC Source: Set_Configuration_Cfm audio[842]: avdtp_ref(0x3c990): ref=4 audio[842]: stream state changed: IDLE -> CONFIGURED audio[842]: session_cb audio[842]: OPEN request succeeded audio[842]: stream state changed: CONFIGURED -> OPEN audio[842]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[842]: SBC Source: Open_Cfm audio[842]: session_cb audio[842]: START request succeeded audio[842]: SBC Source: Start_Cfm audio[842]: SBC Source SEP 0x3c218 locked audio[842]: fd=10, fd_opt=1, pkt_len=672, sample_size=2, rate=44100 audio[842]: 34 bytes sent audio[842]: 1 bytes sent audio[842]: avdtp_unref(0x3c990): ref=3 audio[842]: stream state changed: OPEN -> STREAMING Playing WAVE '/1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Bluetooth Audio Device Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 6400 period_size : 128 period_time : 2902 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 128 xfer_align : 128 start_threshold : 6400 stop_threshold : 6400 silence_threshold: 0 silence_size : 0 boundary : 1677721600 hcid[828]: link_key_request (sba=00:80:E1:00:AB:BA, dba=00:0D:3C:EB:53:F0) Status(R/W): state : RUNNING trigger_time: 24.586883000 tstamp : 0.000000 delay : 0 avail : 6912 avail_max : 6912 aplay: xrun:1104: read/write error, state = RUNNING ---------------->>>>> this is where this error happens audio[842]: Unix client disconnected (fd=8) audio[842]: SBC Source SEP 0x3c218 unlocked audio[842]: avdtp_unref(0x3c990): ref=2 thanks semiyd _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[alsa-devel] [PATCH] cmipci - allow capture of raw spdif subframes
by Timofei V. Bondarenko 13 Nov '07

13 Nov '07

Hi. Enable capturing of raw 32bit IEC958_SUBFRAME. The 24-bits PCM data can be obtained using iec958 plugin. Known problem: captured stream may begin with either left or right subframe. Since the iec958 plugin doesn't decode preamble it may swap the channels sometime. Regards, Tim.

2 4

[alsa-devel] patch for Dell 1210 which using stac9221 codec
by zhejiang 13 Nov '07

13 Nov '07

Hi, Dell 1210 have three jacks on the front. Two of them serve as output jack and one serve as microphone. The microphone can be switched to output model too. There are some problems with current dell 1210 driver: 1.Unable to use the front microphone 2.Unable to switch the microphone 3.If plug the left hp,another hp will be muted,so it's impossible to work as 6-channel According to the issue 0003427 on alsa-bugzilla, I wrote a patch and the reporter helped me to test it on his machine. It can support the front microphone. When switch the microphone to output model, it can mute/unmute the internal speaker with jack-sensing. diff -Nur a/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c b/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c --- a/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-17 08:00:08.000000000 +0800 +++ b/alsa-driver-hg20071017/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-17 17:07:27.000000000 +0800 @@ -171,6 +171,10 @@ 0x02, }; +static hda_nid_t dell_m82_1210_dac_nids[3] = { + 0x02, 0x05, 0x03, +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -874,10 +878,10 @@ 102801D7 (Dell XPS M1210) */ static unsigned int dell_922x_m82_pin_configs[10] = { - 0x0221121f, 0x408103ff, 0x02111212, 0x90100310, - 0x408003f1, 0x02111211, 0x03451340, 0x40c003f2, + 0x02211211, 0x408103ff, 0x02a1123e, 0x90100310, + 0x408003f1, 0x0221121f, 0x03451340, 0x40c003f2, 0x508003f3, 0x405003f4, -}; +}; static unsigned int d945gtp3_pin_configs[10] = { 0x0221401f, 0x01a19022, 0x01813021, 0x01014010, @@ -1512,6 +1516,15 @@ AC_VERB_SET_PIN_WIDGET_CONTROL, pin_type); } +static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, + unsigned int event) +{ + if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) + snd_hda_codec_write_cache(codec, nid, 0, + AC_VERB_SET_UNSOLICITED_ENABLE, + (AC_USRSP_EN | event)); +} + #define stac92xx_io_switch_info snd_ctl_boolean_mono_info static int stac92xx_io_switch_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1542,6 +1555,18 @@ pinctl |= stac92xx_get_vref(codec, nid); stac92xx_auto_set_pinctl(codec, nid, pinctl); } + + if (spec->board_config == STAC_922X_DELL_M82) { + if (val) { + /*If front mic serves as output,let it mute the internal speaker*/ + spec->autocfg.hp_outs = 3; + spec->autocfg.hp_pins[2] = 0x0c; + enable_pin_detect(codec, 0x0c, STAC_HP_EVENT); + } else { + spec->autocfg.hp_outs = 2; + } + codec->patch_ops.unsol_event(codec, STAC_HP_EVENT << 26); + } return 1; } @@ -1811,9 +1836,15 @@ struct sigmatel_spec *spec = codec->spec; unsigned int wid_caps; + int max_controls; + if (spec->board_config == STAC_922X_DELL_M82) { + max_controls = spec->multiout.num_dacs; + } else { + max_controls = cfg->line_outs; + } - for (i = 0; i < cfg->line_outs; i++) { + for (i = 0; i < max_controls; i++) { if (!spec->multiout.dac_nids[i]) continue; @@ -2268,15 +2299,6 @@ AC_VERB_SET_GPIO_DATA, gpiostate); } -static void enable_pin_detect(struct hda_codec *codec, hda_nid_t nid, - unsigned int event) -{ - if (get_wcaps(codec, nid) & AC_WCAP_UNSOL_CAP) - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_UNSOLICITED_ENABLE, - (AC_USRSP_EN | event)); -} - static int stac92xx_init(struct hda_codec *codec) { struct sigmatel_spec *spec = codec->spec; @@ -2677,6 +2699,11 @@ spec->multiout.dac_nids = spec->dac_nids; + if (spec->board_config == STAC_922X_DELL_M82) { + spec->multiout.num_dacs = 3; + spec->multiout.dac_nids = dell_m82_1210_dac_nids; + } + err = stac92xx_parse_auto_config(codec, 0x08, 0x09); if (!err) { if (spec->board_config < 0) { @@ -2687,6 +2714,14 @@ } err = -EINVAL; } + + if ((err >= 0) && (spec->board_config == STAC_922X_DELL_M82)) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_IO_SWITCH, "Front Mic as Output Switch", (spec->autocfg.input_pins[AUTO_PIN_FRONT_MIC] << 8) | 1); + /*In case that add_controls realloc the spec->kctl_alloc*/ + if ((err >= 0) && (spec->kctl_alloc != spec->mixers[spec->num_mixers - 1])) + spec->mixers[spec->num_mixers - 1] = spec->kctl_alloc; + } + if (err < 0) { stac92xx_free(codec); return err;

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[alsa-devel] hardware-dependent features
by Daniel Mack 13 Nov '07

13 Nov '07

Hi, some of the devices the caiaq driver supports have special hardware features such as switchable ground lift, input impedance selection or other, hardware-specific features. On both Windows and MacOSX there are special tools to set those features but the ALSA driver still lacks support for them which I want to change soon. However, I couldn't find a generic way to do that. I was thinking about a sysfs entry which is attached to the sound card instance but didn't find a driver which did similar things. Also, I'd like to support the LEDs and displays on the devices and was wondering which interface to choose for that. I've got a patch ready which uses the generic led subsystem, but that approach seems to be a bit ugly as I end up having one entry per led on each device in one single folder. Any idea of a better way? Thanks, Daniel

4 5

Re: [alsa-devel] No sound from usb on Motorola C350 cellphone
by Giovanni Maruzzelli 13 Nov '07

13 Nov '07

With an old copy of motorola driver, the usb sound device was recognized by windows XP, with mixer elements only for the playback. I was not able to record from it, playing give no errors, but no sound either With the last copy of driver from motorola developer devsite, on XP a whole new "Motorola Device" is created, with the audio interface listed on it in the usb control panel, but no audio device is available for normal usage. Needs more info, tests? On Nov 12, 2007 2:50 PM, Alexander E. Patrakov <patrakov(a)gmail.com> wrote: > Giovanni Maruzzelli wrote: > > Maybe there is a way to not using soundcards, that would be *so* much > > better: I found that Motorola c350 phones after receiving an AT+MODE=8 > > command via USB, make available an usb sound device. > > Confirmed with my Motorola C350 cellphone. It appears to accept S16_LE, MU_LAW > and A_LAW formats, 8 kHz sample rate, but actually records no data and plays no > sounds (even if a cal is in progress). > > One question: what playback and recording functionality is available on this > cellphone under Windows? > > -- > Alexander E. Patrakov >

4 4

[alsa-devel] No sound from usb on Motorola C350 cellphone
by Giovanni Maruzzelli 12 Nov '07

12 Nov '07

Hi list participants, I am using cellphones connected to soundcards as cheap gsm gateways (more complex than that, actually. You can see the whole GPL project at www.celliax.org). Maybe there is a way to not using soundcards, that would be *so* much better: I found that Motorola c350 phones after receiving an AT+MODE=8 command via USB, make available an usb sound device. That would be wonderful (less cables, more robustness, etc). Problem is: the usb sound device do not plays or captures samples. I suspect that the usb audio has to be activated, maybe by an AT or a P2K command. Or that a quirk has to be added to the alsa driver (snd_usb_audio). Do you know anything about (maybe the quirk to be added to ALSA driver...)? Do you have any hint? I'm available to do all the research-testing-debugging that may be useful. What happens is as follow: 1) attach the c350 to usb 2) is recognized as acm0 3) send at+mode=8 to c350 (via minicom, for eg) 4) it changes its usbid, disconnect from usb, reconnect, is recognized as acm1, makes available a sound device 5) at commands work on the new acm1 6) trying to use the audio device directly (eg aplay -Dhw:1) do not give errors on play (but no sound is head on the phone), gives error pcm_read 1349 on record (eg arecord -Dhw:1) 7) trying to use the audiodevice through alsalib does not gives errors at all (eg -Dplughw:1) but does not caputeres samples, and no sound is heard on the phone I'm attaching: lusb_without -> result of lusb -vv with no device attached lsusb_with_ACM0 -> result of lusb -vv with c350 just attached, recognized as acm modem lsusb_with_ACM1 -> result of lusb -vv with c350 after sending at+mode=8 to him, he has disconnected and was recognized again, this time as ACM1, with a lot of additional usb interfaces dmesg-acm0 -> result of dmesg after first c350 attachment dmesg-acm1 -> result of dmesg after at+mode=8 was sent to the c350 aplay-l -> result of aplay -l showing the c350 as an alsa device stream0_pstop_cstop -> result of cat /proc/asound/C350/stream0 while not active stream0_pstop_crun -> result of cat /proc/asound/C350/stream0 while arecord -vv -Dplughw:1 -fS16_LE ciapa.raw (if using -Dhw:1 it fails with error: "pcm_read:1349: read error: Input/output error"). arecord does not gives errors, but do not write samples stream0_prun_crun -> result of cat /proc/asound/C350/stream0 while aplay and arecord -vv -Dplughw:1 -fS16_LE ciapa.raw (if using -Dhw:1 arecord fails with error: "pcm_read:1349: read error: Input/output error", aplay works but no audio) arecord does not gives errors, but do not write samples aplay_play -> results of aplay arecord_rec -> results of arecord Thank you all in advance, Giovanni Maruzzelli

2 3

[alsa-devel] Non standard sound card
by Ning Zhang 12 Nov '07

12 Nov '07

Hi Dear, I am a fresh man in this community. Now I plan to develope a sound driver for my embedded platform. The problem is that my audio chip is a DSP ,not a standard codec chip. The MCU communicates with this DSP via a shared ram. So at first , I have to make the DSP as a platform_device and use the platform_driver_register() to register the DSP , right? Then I can use alsa driver API to develop the sound card. Is my understanding right? Thanks. Where can I find some information about ALSA architecture? David.

2 1

[alsa-devel] [PATCH] use convenient treble scale on WM8750
by Stanislav Brabec 12 Nov '07

12 Nov '07

On Zaurus SL-C3200 (terrier/spitz) based on WM8750, treble scale is inconveniently reverted (increase level = decrease treble), in opposite to bass scale, which uses convenient scale. Fix ALSA WM8750 mixer treble to use convenient treble scale (increase = increase treble level): --- linux-2.6.23/sound/soc/codecs/wm8750.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23/sound/soc/codecs/wm8750.c 2007-11-02 16:47:35.000000000 +0100 @@ -189,7 +189,7 @@ SOC_ENUM("Bass Filter", wm8750_enum[1]), SOC_SINGLE("Bass Volume", WM8750_BASS, 0, 15, 1), -SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 0), +SOC_SINGLE("Treble Volume", WM8750_TREBLE, 0, 15, 1), SOC_ENUM("Treble Cut-off", wm8750_enum[2]), SOC_SINGLE("3D Switch", WM8750_3D, 0, 1, 0), ________________________________________________________________________ Stanislav Brabec http://www.penguin.cz/~utx

2 1

[alsa-devel] I could maybe help for Creative Labs SB X-Fi integration
by Frederic Bouy 10 Nov '07

10 Nov '07

Hello everybody, I have just installed a distribution of Ubuntu (7.10 Gutsy Gibbon) on my computer. The issue is that my sound card is not working. According to your website, this is because my SB X-Fi is not yet supported by ALSA : "Card delivered to developers. Completely new architecture. Creative actively preventing support due to no datasheets being released to ALSA developers. Reverse engineering work not started due to lack of time." I would like to know if there is a way for me to help you integrating this sound card. What can I do to help? Thanks. Frédéric

2 1

[alsa-devel] Getting XRUN after writei in RUNNING state?
by Piers O'Hanlon 10 Nov '07

10 Nov '07

Hi folks, I've trying to improve the ALSA driver for the Robust Audio Tool (RAT) tool: http://frostie.cs.ucl.ac.uk/nets/mmedia/wiki/RatWiki Specifically the code I'm talking about is: http://frostie.cs.ucl.ac.uk/nets/mmedia/browser/rat/trunk/auddev_alsa.c I've added some status dumps in the code that I'm running which aren't in the repository version but basically the code is doing the same thing. However I'm seeing weird XRUNs (under runs on the playback side) - There's as couple of instances I don't understand: (The pcm device setup is shown at the bottom); 1) RAT is happily writing audio to the playback device for while then the device just seems to go in XRUN state after being in RUNNING state and doing a writei()??: [+945 auddev_alsa.c] Time since last write: 42734us, Trying to write 640 frames state : RUNNING trigger_time: 1194622844.333930000 tstamp : 0.000000 delay : 121 avail : 1927 avail_max : 2043 [pid/9901 +954 auddev_alsa.c] Wrote 1280 bytes, frames: 640 state : XRUN trigger_time: 1194622844.333930000 tstamp : 0.000000 delay : 758 avail : 1290 avail_max : 2046 2) RAT is writing audio to the playback device for while then the device just seems to go in XRUN state when there's still positive delay in the buffer. Here's last write that went ok - showing the snd_pcm_status_dump before doing the write and afterwards: [+945 auddev_alsa.c] Time since last write: 42635us, Trying to write 640 frames state : RUNNING trigger_time: 1194622843.886113000 tstamp : 0.000000 delay : 275 avail : 1773 avail_max : 2041 [+954 auddev_alsa.c] Wrote 1280 bytes, frames: 640 state : RUNNING trigger_time: 1194622843.886113000 tstamp : 0.000000 delay : 907 avail : 1141 avail_max : 2043 The next time the audio write is called the state is checked and found to be in XRUN with 225 delay in the buffer:??? (And yet I've seen other writes find the device still running with alot less delay in the buffer??): [ +936 auddev_alsa.c] In SND_PCM_STATE_XRUN - preparing audio [pid/9901 +945 auddev_alsa.c] Frames avail to be written=2048, time diff=43182, Trying to write 640 frames, Curr delay 0 state : XRUN trigger_time: 1194622843.886113000 tstamp : 0.000000 delay : 225 avail : 1823 avail_max : 2049 [pid/9901 +954 auddev_alsa.c] Wrote 1280 bytes, frames: 640 state : RUNNING trigger_time: 1194622844.142516000 tstamp : 0.000000 delay : 637 avail : 1411 avail_max : 1708 Currently I'm stumped at to solve these problems.... And basically ALSA sound quality as worse than the OSS driver for RAT as a result. Any suggestions are to how improve or sort these problems is welcome. Thanks, Piers O'Hanlon. University College London ===snd_pcm_dump_setup:=============== stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 16000 exact rate : 16000 (16000/1) msbits : 16 buffer_size : 2048 period_size : 341 period_time : 21333 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 341 xfer_align : 1 start_threshold : 341 stop_threshold : 2048 silence_threshold: 0 silence_size : 0 boundary : 536870912

2 3

Re: [alsa-devel] [linux-usb-devel] [PATCH] ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also
by Karsten Wiese 09 Nov '07

09 Nov '07

Am Donnerstag, 8. November 2007 schrieb David Brownell: > > Looks interesting, but could you make this apply to *BOTH* types of > ISO transfer descriptor? You did it for ITDs, but not Split-ITDs. Done. Split-ITD changes untested. > > Plus there are two non-technical problems with this patch. First, > it doesn't apply against 2.6.24-rc2-git; second, it triggers lots > of messages from scripts/checkpatch.pl ... both issues need to get > fixed before I can properly review this. Fixed. Karsten -------------------------------------> From: Karsten Wiese <fzu(a)wemgehoertderstaat.de> ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also Upstream ehci-hcd only completes iso urbs, if the last frame (= 8 uframes) they overlapped with has elapsed. That can be as late as when the following urb emits its interrupt on completion. soundcard drivers tend to work around by only transfering iso urbs with number_of_packets = (n * 8). Patch lets iso urbs complete asap by scanning ITDs always up to the elapsed uframe. An ITD's last scanned uframe is stored in the new struct ehci_itd member uframe_scanned. ITDs stay kept in the hcd's schedule until the complete frame they cover has elapsed, like without patch. To make this possible, ehci->periodic_sched is changed based on the number of active itds instead of the number of active iso urbs. SITD handling is unchanged, only sitd_complete() is renamed to sitd_scan() and the active check is moved to sitd_scan() to make the "case Q_TYPE_ITD:" and "case Q_TYPE_SITD:" sections under scan_periodic() look alike. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- drivers/usb/host/ehci-sched.c | 156 ++++++++++++++++++++--------------------- drivers/usb/host/ehci.h | 3 +- 2 files changed, 76 insertions(+), 83 deletions(-) diff --git a/drivers/usb/host/ehci-sched.c b/drivers/usb/host/ehci-sched.c index 80d99bc..e647842 100644 --- a/drivers/usb/host/ehci-sched.c +++ b/drivers/usb/host/ehci-sched.c @@ -1486,6 +1486,8 @@ itd_link (struct ehci_hcd *ehci, unsigned frame, struct ehci_itd *itd) itd->frame = frame; wmb (); ehci->periodic[frame] = cpu_to_hc32(ehci, itd->itd_dma | Q_TYPE_ITD); + if (unlikely(!ehci->periodic_sched++)) + enable_periodic(ehci); } /* fit urb's itds into the selected schedule slot; activate as needed */ @@ -1558,30 +1560,28 @@ itd_link_urb ( urb->hcpriv = NULL; timer_action (ehci, TIMER_IO_WATCHDOG); - if (unlikely (!ehci->periodic_sched++)) - return enable_periodic (ehci); return 0; } #define ISO_ERRS (EHCI_ISOC_BUF_ERR | EHCI_ISOC_BABBLE | EHCI_ISOC_XACTERR) -static unsigned -itd_complete ( - struct ehci_hcd *ehci, - struct ehci_itd *itd -) { +static bool +itd_scan(struct ehci_hcd *ehci, struct ehci_itd *itd, unsigned uframe_after) +{ struct urb *urb = itd->urb; struct usb_iso_packet_descriptor *desc; u32 t; unsigned uframe; int urb_index = -1; struct ehci_iso_stream *stream = itd->stream; - struct usb_device *dev; /* for each uframe with a packet */ - for (uframe = 0; uframe < 8; uframe++) { + for (uframe = itd->uframe_scanned; uframe < uframe_after; uframe++) { + struct usb_device *dev; + if (likely (itd->index[uframe] == -1)) continue; + urb_index = itd->index[uframe]; desc = &urb->iso_frame_desc [urb_index]; @@ -1608,45 +1608,54 @@ itd_complete ( desc->status = 0; desc->actual_length = EHCI_ITD_LENGTH (t); } - } - usb_put_urb (urb); - itd->urb = NULL; - itd->stream = NULL; - list_move (&itd->itd_list, &stream->free_list); - iso_stream_put (ehci, stream); - - /* handle completion now? */ - if (likely ((urb_index + 1) != urb->number_of_packets)) - return 0; + /* handle completion now? */ + if (urb_index + 1 != urb->number_of_packets) + continue; - /* ASSERT: it's really the last itd for this urb - list_for_each_entry (itd, &stream->td_list, itd_list) - BUG_ON (itd->urb == urb); - */ + usb_put_urb(urb); + /* ASSERT: it's really the last itd for this urb + struct ehci_itd *stritd; + list_for_each_entry(stritd, &stream->td_list, itd_list) + BUG_ON(stritd != itd && stritd->urb == urb); + */ + itd->urb = NULL; - /* give urb back to the driver ... can be out-of-order */ - dev = urb->dev; - ehci_urb_done(ehci, urb, 0); - urb = NULL; + /* give urb back to the driver ... can be out-of-order */ + dev = urb->dev; + ehci_urb_done(ehci, urb, 0); + urb = NULL; - /* defer stopping schedule; completion can submit */ - ehci->periodic_sched--; - if (unlikely (!ehci->periodic_sched)) - (void) disable_periodic (ehci); - ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; + ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; - if (unlikely (list_empty (&stream->td_list))) { - ehci_to_hcd(ehci)->self.bandwidth_allocated + if (unlikely(list_empty(&stream->td_list))) { + ehci_to_hcd(ehci)->self.bandwidth_allocated -= stream->bandwidth; - ehci_vdbg (ehci, - "deschedule devp %s ep%d%s-iso\n", - dev->devpath, stream->bEndpointAddress & 0x0f, - (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); + ehci_vdbg(ehci, + "deschedule devp %s ep%d%s-iso\n", + dev->devpath, + stream->bEndpointAddress & 0x0f, + (stream->bEndpointAddress & USB_DIR_IN) ? + "in" : "out"); + } + iso_stream_put(ehci, stream); } + + itd->uframe_scanned = uframe_after; + if (uframe_after < 8) + return false; + + itd->stream = NULL; + itd->uframe_scanned = 0; + list_move(&itd->itd_list, &stream->free_list); iso_stream_put (ehci, stream); - return 1; + /* defer stopping schedule; completion can submit */ + ehci->periodic_sched--; + if (unlikely(!ehci->periodic_sched)) + disable_periodic(ehci); + + return true; } /*-------------------------------------------------------------------------*/ @@ -1950,11 +1959,9 @@ sitd_link_urb ( #define SITD_ERRS (SITD_STS_ERR | SITD_STS_DBE | SITD_STS_BABBLE \ | SITD_STS_XACT | SITD_STS_MMF) -static unsigned -sitd_complete ( - struct ehci_hcd *ehci, - struct ehci_sitd *sitd -) { +static bool +sitd_scan(struct ehci_hcd *ehci, struct ehci_sitd *sitd, int live) +{ struct urb *urb = sitd->urb; struct usb_iso_packet_descriptor *desc; u32 t; @@ -1962,9 +1969,12 @@ sitd_complete ( struct ehci_iso_stream *stream = sitd->stream; struct usb_device *dev; + t = hc32_to_cpup(ehci, &sitd->hw_results); + if (t & SITD_STS_ACTIVE && live) + return false; + urb_index = sitd->index; desc = &urb->iso_frame_desc [urb_index]; - t = hc32_to_cpup(ehci, &sitd->hw_results); /* report transfer status */ if (t & SITD_ERRS) { @@ -1991,7 +2001,7 @@ sitd_complete ( /* handle completion now? */ if ((urb_index + 1) != urb->number_of_packets) - return 0; + return true; /* ASSERT: it's really the last sitd for this urb list_for_each_entry (sitd, &stream->td_list, sitd_list) @@ -2019,7 +2029,7 @@ sitd_complete ( } iso_stream_put (ehci, stream); - return 1; + return true; } @@ -2091,13 +2101,11 @@ sitd_submit (struct ehci_hcd *ehci, struct urb *urb, gfp_t mem_flags) return -ENOSYS; } -static inline unsigned -sitd_complete ( - struct ehci_hcd *ehci, - struct ehci_sitd *sitd -) { - ehci_err (ehci, "sitd_complete %p?\n", sitd); - return 0; +static inline bool +sitd_scan(struct ehci_hcd *ehci, struct ehci_sitd *sitd) +{ + ehci_err(ehci, "%s %p?\n", __FUNCTION__, sitd); + return false; } #endif /* USB_EHCI_SPLIT_ISO */ @@ -2175,47 +2183,33 @@ restart: q = q.fstn->fstn_next; break; case Q_TYPE_ITD: - /* skip itds for later in the frame */ rmb (); - for (uf = live ? uframes : 8; uf < 8; uf++) { - if (0 == (q.itd->hw_transaction [uf] - & ITD_ACTIVE(ehci))) - continue; + modified = itd_scan(ehci, q.itd, + uf = live ? uframes : 8); + if (!modified) { q_p = &q.itd->itd_next; hw_p = &q.itd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.itd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.itd->itd_next; + *hw_p = q.itd->hw_next; } - if (uf != 8) - break; - - /* this one's ready ... HC won't cache the - * pointer for much longer, if at all. - */ - *q_p = q.itd->itd_next; - *hw_p = q.itd->hw_next; type = Q_NEXT_TYPE(ehci, q.itd->hw_next); + wmb(); - modified = itd_complete (ehci, q.itd); q = *q_p; break; case Q_TYPE_SITD: - if ((q.sitd->hw_results & SITD_ACTIVE(ehci)) - && live) { + modified = sitd_scan(ehci, q.sitd, live); + if (!modified) { q_p = &q.sitd->sitd_next; hw_p = &q.sitd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.sitd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.sitd->sitd_next; + *hw_p = q.sitd->hw_next; } - *q_p = q.sitd->sitd_next; - *hw_p = q.sitd->hw_next; type = Q_NEXT_TYPE(ehci, q.sitd->hw_next); + wmb(); - modified = sitd_complete (ehci, q.sitd); q = *q_p; break; default: diff --git a/drivers/usb/host/ehci.h b/drivers/usb/host/ehci.h index 951d69f..156c02f 100644 --- a/drivers/usb/host/ehci.h +++ b/drivers/usb/host/ehci.h @@ -581,6 +581,7 @@ struct ehci_itd { struct urb *urb; struct ehci_iso_stream *stream; /* endpoint's queue */ struct list_head itd_list; /* list of stream's itds */ + unsigned uframe_scanned; /* any/all hw_transactions here may be used by that urb */ unsigned frame; /* where scheduled */ @@ -615,8 +616,6 @@ struct ehci_sitd { #define SITD_STS_MMF (1 << 2) /* incomplete split transaction */ #define SITD_STS_STS (1 << 1) /* split transaction state */ -#define SITD_ACTIVE(ehci) cpu_to_hc32(ehci, SITD_STS_ACTIVE) - __hc32 hw_buf [2]; /* EHCI table 3-12 */ __hc32 hw_backpointer; /* EHCI table 3-13 */ __hc32 hw_buf_hi [2]; /* Appendix B */ -- 1.5.3.3

1 0

[alsa-devel] [PATCH] sound - remove dead config symbol from sound code
by Jiri Olsa 09 Nov '07

09 Nov '07

remove dead config symbols from sound code Signed-off-by: Jiri Olsa <olsajiri(a)gmail.com> --- include/sound/cs46xx.h | 3 - include/sound/driver.h | 4 -- sound/pci/cs46xx/cs46xx_lib.c | 99 ----------------------------------------- 3 files changed, 0 insertions(+), 106 deletions(-) diff --git a/include/sound/cs46xx.h b/include/sound/cs46xx.h index 6b40ee6..e3005a6 100644 --- a/include/sound/cs46xx.h +++ b/include/sound/cs46xx.h @@ -1708,9 +1708,6 @@ struct snd_cs46xx { struct gameport *gameport; -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO - int current_gpio; -#endif #ifdef CONFIG_SND_CS46XX_NEW_DSP struct mutex spos_mutex; diff --git a/include/sound/driver.h b/include/sound/driver.h index 5ccb6c5..1889929 100644 --- a/include/sound/driver.h +++ b/include/sound/driver.h @@ -38,10 +38,6 @@ #define CONFIG_SND_MAJOR 116 #endif -#ifndef CONFIG_SND_DEBUG -#undef CONFIG_SND_DEBUG_MEMORY -#endif - #ifdef ALSA_BUILD #include "adriver.h" #endif diff --git a/sound/pci/cs46xx/cs46xx_lib.c b/sound/pci/cs46xx/cs46xx_lib.c index 2c7bfc9..8c44fef 100644 --- a/sound/pci/cs46xx/cs46xx_lib.c +++ b/sound/pci/cs46xx/cs46xx_lib.c @@ -2084,71 +2084,6 @@ static int snd_cs46xx_spdif_stream_put(struct snd_kcontrol *kcontrol, #endif /* CONFIG_SND_CS46XX_NEW_DSP */ -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO -static int snd_cs46xx_egpio_select_info(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_info *uinfo) -{ - uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = 8; - return 0; -} - -static int snd_cs46xx_egpio_select_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - ucontrol->value.integer.value[0] = chip->current_gpio; - - return 0; -} - -static int snd_cs46xx_egpio_select_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int change = (chip->current_gpio != ucontrol->value.integer.value[0]); - chip->current_gpio = ucontrol->value.integer.value[0]; - - return change; -} - - -static int snd_cs46xx_egpio_get(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value; - - snd_printdd ("put: reg = %04x, gpio %02x\n",reg,chip->current_gpio); - ucontrol->value.integer.value[0] = - (snd_cs46xx_peekBA0(chip, reg) & (1 << chip->current_gpio)) ? 1 : 0; - - return 0; -} - -static int snd_cs46xx_egpio_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_cs46xx *chip = snd_kcontrol_chip(kcontrol); - int reg = kcontrol->private_value; - int val = snd_cs46xx_peekBA0(chip, reg); - int oldval = val; - snd_printdd ("put: reg = %04x, gpio %02x\n",reg,chip->current_gpio); - - if (ucontrol->value.integer.value[0]) - val |= (1 << chip->current_gpio); - else - val &= ~(1 << chip->current_gpio); - - snd_cs46xx_pokeBA0(chip, reg,val); - snd_printdd ("put: val %08x oldval %08x\n",val,oldval); - - return (oldval != val); -} -#endif /* CONFIG_SND_CS46XX_DEBUG_GPIO */ - static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -2241,40 +2176,6 @@ static struct snd_kcontrol_new snd_cs46xx_controls[] __devinitdata = { }, #endif -#ifdef CONFIG_SND_CS46XX_DEBUG_GPIO -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO select", - .info = snd_cs46xx_egpio_select_info, - .get = snd_cs46xx_egpio_select_get, - .put = snd_cs46xx_egpio_select_put, - .private_value = 0, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO Input/Output", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIODR, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO CMOS/Open drain", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIOPTR, -}, -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "EGPIO On/Off", - .info = snd_mixer_boolean_info, - .get = snd_cs46xx_egpio_get, - .put = snd_cs46xx_egpio_put, - .private_value = BA0_EGPIOSR, -}, -#endif }; #ifdef CONFIG_SND_CS46XX_NEW_DSP

1 0

[alsa-devel] When command SNDRV_PCM_TRIGGER_STOP is asked in snd_pcm_ops trigger(), which buffer in the buffer list should be the last one to output by the DMA?
by Sonic Zhang 09 Nov '07

09 Nov '07

In our blackfin ad1981 linux driver, there is an output ring with "periods" buffers. Each buffer is of the size of "period_size". After ALSA starts playing, several buffers ahead of current one running by the DMA controller are usually filled in advance by the ALSA layer. When command SNDRV_PCM_TRIGGER_STOP is asked in snd_pcm_ops trigger(), which buffer should be the last one to output by the DMA? 1. Current buffer, which is running by the DMA controller 2. The last buffer, which the DMA controller just finished outputting and snd_pcm_ops pointer() is notifiyed to ALSA layer. Does this mean current buffer in DMA should be abandoned in the middle? 3. The last buffer, which the ALSA layer has just filled in snd_pcm_ops copy()? Thanks

2 1

[alsa-devel] User-space OSS emulation?
by Timur Tabi 09 Nov '07

09 Nov '07

Can someone tell me more about "user-space OSS emulation"? I've read a few references to it in the mailing list archives, but a google search doesn't turn up anything useful. -- Timur Tabi Linux kernel developer at Freescale

3 4

[alsa-devel] How can OSS applications support multiple streams?
by Timur Tabi 09 Nov '07

09 Nov '07

I've written an ALSA driver that supports only one playback stream, because that's what our hardware supports. I have a customer that has an OSS application that tries to open 32 simultaneous playback streams, which obviously doesn't work. What do I (or the customer) need to do to get 32 streams to work? Is this what a sound server is for? If so, which one should I use? -- Timur Tabi Linux kernel developer at Freescale

1 0

[alsa-devel] Any OSS changes from kernel 2.6.21 to 2.6.23? Something broke.
by Timur Tabi 08 Nov '07

08 Nov '07

I have a customer who says that mplayer (using OSS) does not work when he tries a kernel based on 2.6.23-rc4, but it works when he uses a kernel based on 2.6.21. Everything else is the same (including the version of alsa-lib), so I presume something broke in the OSS emulation in the kernel. Unfortunately, our git repositories for these trees are not compatible, so I can't use git-bisect to narrow the problem down. Is it possible to back-level the ALSA support in the kernel itself? Can I just download ftp://ftp.alsa-project.org/pub/driver/alsa-driver-1.0.XX.tar.bz2 and follow the instructions in the INSTALL file? If you update the kernel, are you supposed to use the latest alsa-lib as well? For a given kernel, how do I know what the right version of alsa-lib is? -- Timur Tabi Linux kernel developer at Freescale

4 9

[alsa-devel] Fwd: [PATCH] ehci-hcd: complete iso urbs ASAP for number_of_packets != (n * 8) also
by Karsten Wiese 08 Nov '07

08 Nov '07

Hi I just sent attached patch to usb-devel. It makes special workaround for iso usb communication using ehci-hcd highspeed needless. Comments, Bug/success reports welcome! kind regards Karsten Upstream (as of 2.6.23) ehci-hcd only completes iso urbs, if the last frame (= 8 uframes) they overlapped with has elapsed. That can be as late as when the following urb emits its interrupt on completion. soundcard drivers tend to work around by only transfering iso urbs with number_of_packets = (n * 8). Patch lets iso urbs complete asap by scanning itd always up to the elapsed uframe. An itd's last scanned uframe is stored in the new struct ehci_itd member uframe_scanned. Itds stay kept in the hcd's schedule until the complete frame they cover has elapsed, like without patch. To make this possible, ehci->periodic_sched is changed based on the number of active itds instead of the number of active iso urbs. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- drivers/usb/host/ehci-sched.c | 108 ++++++++++++++++++++--------------------- drivers/usb/host/ehci.h | 1 + 2 files changed, 54 insertions(+), 55 deletions(-) diff --git a/drivers/usb/host/ehci-sched.c b/drivers/usb/host/ehci-sched.c index e682f23..c456b16 100644 --- a/drivers/usb/host/ehci-sched.c +++ b/drivers/usb/host/ehci-sched.c @@ -1481,6 +1481,8 @@ itd_link (struct ehci_hcd *ehci, unsigned frame, struct ehci_itd *itd) itd->frame = frame; wmb (); ehci->periodic[frame] = cpu_to_hc32(ehci, itd->itd_dma | Q_TYPE_ITD); + if (unlikely (!ehci->periodic_sched++)) + enable_periodic (ehci); } /* fit urb's itds into the selected schedule slot; activate as needed */ @@ -1553,30 +1555,30 @@ itd_link_urb ( urb->hcpriv = NULL; timer_action (ehci, TIMER_IO_WATCHDOG); - if (unlikely (!ehci->periodic_sched++)) - return enable_periodic (ehci); return 0; } #define ISO_ERRS (EHCI_ISOC_BUF_ERR | EHCI_ISOC_BABBLE | EHCI_ISOC_XACTERR) -static unsigned -itd_complete ( - struct ehci_hcd *ehci, - struct ehci_itd *itd -) { +static bool +itd_scan ( + struct ehci_hcd *ehci, + struct ehci_itd *itd, + unsigned uframe_after +) +{ struct urb *urb = itd->urb; struct usb_iso_packet_descriptor *desc; u32 t; unsigned uframe; int urb_index = -1; struct ehci_iso_stream *stream = itd->stream; - struct usb_device *dev; /* for each uframe with a packet */ - for (uframe = 0; uframe < 8; uframe++) { + for (uframe = itd->uframe_scanned; uframe < uframe_after; uframe++) { if (likely (itd->index[uframe] == -1)) continue; + urb_index = itd->index[uframe]; desc = &urb->iso_frame_desc [urb_index]; @@ -1603,45 +1605,53 @@ itd_complete ( desc->status = 0; desc->actual_length = EHCI_ITD_LENGTH (t); } + + /* handle completion now? */ + if (urb_index + 1 == urb->number_of_packets) { + struct usb_device *dev; + + usb_put_urb (urb); + /* ASSERT: it's really the last itd for this urb + struct ehci_itd *stritd; + list_for_each_entry (stritd, &stream->td_list, itd_list) + BUG_ON (stritd != itd && stritd->urb == urb); + */ + itd->urb = NULL; + + /* give urb back to the driver ... can be out-of-order */ + dev = urb->dev; + ehci_urb_done (ehci, urb); + urb = NULL; + + ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; + + if (unlikely (list_empty (&stream->td_list))) { + ehci_to_hcd(ehci)->self.bandwidth_allocated + -= stream->bandwidth; + ehci_vdbg (ehci, + "deschedule devp %s ep%d%s-iso\n", + dev->devpath, stream->bEndpointAddress & 0x0f, + (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); + } + iso_stream_put (ehci, stream); + } } - usb_put_urb (urb); - itd->urb = NULL; + itd->uframe_scanned = uframe_after; + if (uframe_after < 8) + return false; + itd->stream = NULL; + itd->uframe_scanned = 0; list_move (&itd->itd_list, &stream->free_list); iso_stream_put (ehci, stream); - /* handle completion now? */ - if (likely ((urb_index + 1) != urb->number_of_packets)) - return 0; - - /* ASSERT: it's really the last itd for this urb - list_for_each_entry (itd, &stream->td_list, itd_list) - BUG_ON (itd->urb == urb); - */ - - /* give urb back to the driver ... can be out-of-order */ - dev = urb->dev; - ehci_urb_done (ehci, urb); - urb = NULL; - /* defer stopping schedule; completion can submit */ ehci->periodic_sched--; if (unlikely (!ehci->periodic_sched)) (void) disable_periodic (ehci); - ehci_to_hcd(ehci)->self.bandwidth_isoc_reqs--; - if (unlikely (list_empty (&stream->td_list))) { - ehci_to_hcd(ehci)->self.bandwidth_allocated - -= stream->bandwidth; - ehci_vdbg (ehci, - "deschedule devp %s ep%d%s-iso\n", - dev->devpath, stream->bEndpointAddress & 0x0f, - (stream->bEndpointAddress & USB_DIR_IN) ? "in" : "out"); - } - iso_stream_put (ehci, stream); - - return 1; + return true; } /*-------------------------------------------------------------------------*/ @@ -2156,30 +2166,18 @@ restart: q = q.fstn->fstn_next; break; case Q_TYPE_ITD: - /* skip itds for later in the frame */ rmb (); - for (uf = live ? uframes : 8; uf < 8; uf++) { - if (0 == (q.itd->hw_transaction [uf] - & ITD_ACTIVE(ehci))) - continue; + modified = itd_scan(ehci, q.itd, uf = live ? uframes : 8); + if (!modified) { q_p = &q.itd->itd_next; hw_p = &q.itd->hw_next; - type = Q_NEXT_TYPE(ehci, - q.itd->hw_next); - q = *q_p; - break; + } else { + *q_p = q.itd->itd_next; + *hw_p = q.itd->hw_next; } - if (uf != 8) - break; - - /* this one's ready ... HC won't cache the - * pointer for much longer, if at all. - */ - *q_p = q.itd->itd_next; - *hw_p = q.itd->hw_next; type = Q_NEXT_TYPE(ehci, q.itd->hw_next); + wmb(); - modified = itd_complete (ehci, q.itd); q = *q_p; break; case Q_TYPE_SITD: diff --git a/drivers/usb/host/ehci.h b/drivers/usb/host/ehci.h index 951d69f..df35964 100644 --- a/drivers/usb/host/ehci.h +++ b/drivers/usb/host/ehci.h @@ -581,6 +581,7 @@ struct ehci_itd { struct urb *urb; struct ehci_iso_stream *stream; /* endpoint's queue */ struct list_head itd_list; /* list of stream's itds */ + unsigned uframe_scanned; /* any/all hw_transactions here may be used by that urb */ unsigned frame; /* where scheduled */ -- 1.5.3.3

2 6

[alsa-devel] Audio quality issues with dmix plugin
by Pharaoh . 08 Nov '07

08 Nov '07

Hi, When I play a 44Khz file without using any ALSA plugin like #aplay -Dhw:0,0 44k.wav the sound quality is awesome but when I play the same file using dmix, clicks are introduced. #aplay -Dplug:dmix 44k.wav changing period size and buffer size doesn't help. I am testing my ALSA driver on an OMAP850 based board. Regards, Pharaoh.

2 4

Re: [alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 08 Nov '07

08 Nov '07

Takashi, You were right, it was a linking issue. I'm doing cross-compiling for a target of different processor type. I just had to make sure I'm pointing to the right location for the includes and libs. Thanks, Ben ----- Original Message ---- From: Takashi Iwai <tiwai(a)suse.de> To: Ben C <bchang_2(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Sent: Wednesday, November 7, 2007 4:24:04 AM Subject: Re: [alsa-devel] snd_pcm_hw_params_get_xxx question At Tue, 6 Nov 2007 18:21:01 -0800 (PST), Ben C wrote: > > Hello, > > I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. > > For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? > > I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Yes, it sounds like you're accessing the old API. Do you use libasound from a module? If it's used from a module, the module itself should be built once with link to libasound. For example, $(CC) -shared -o MODULE.so -lasound In such a way, the properly versioned symbol is chosen. The weak link results in the old API. This is irrelveant whether it's ALSA 1.0.13 or 1.0.15. Takashi __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[alsa-devel] [HDA Intel/STAC92xx] Alsa 1.0.14
by Greg Jizmagian 07 Nov '07

07 Nov '07

I am trying to fix the sound on the m1330 in OS X. It currently only works out of one of the headphone jacks. I stumbled across the patch you wrote for ubuntu and hoped that I could adapt it for OS X. Or is there a way to see a post patch linux dump that would fix the problem? Any suggestions/help would be greatly appreciated. Thanks. Greg

1 0

[alsa-devel] snd_pcm_hw_params_get_xxx question
by Ben C 07 Nov '07

07 Nov '07

Hello, I'm trying to develop an external PCM plugin. I'm calling snd_pcm_hw_params_get_xxx functions to retrieve some values. I got everything to work as expected on alsa-lib 1.0.15. However the client's setup is 1.0.13, and I don't think they want to upgrade. For 1.0.13, everything compiled and linked fine after some minor changes. But when I called say snd_pcm_hw_params_get_rate(hw, &val, &dir), I was expecting the return value to be a status, and val to contain "44100" which was the case for 1.0.15. However for 1.0.13, val is 0, but returned status = 44100? I looked in the header files, and 1.0.13 seems to be the same as 1.0.15 in regards to this function. Am I accidentally using some sort of old API that was intended for backward compatibility? If so, what should I do to use the "new" API's? Thanks, Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[alsa-devel] [PATCH] hda: STAC9228 Subsystem update
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added more laptops subsystem id's that have STAC9228 DMIC support. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 43708ece3f6f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Nov 06 11:56:17 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Nov 06 14:01:53 2007 -0500 @@ -2921,8 +2921,10 @@ static int patch_stac927x(struct hda_cod } switch (codec->subsystem_id) { - case 0x1028020A: /* STAC 9228 */ - case 0x10280209: /* STAC 9228 */ + case 0x10280242: /* STAC 9228 */ + case 0x102801f3: + case 0x1028020A: + case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b;

2 1

[alsa-devel] [PATCH] cmipci - utilize ADC48K44K bit
by Timofei V. Bondarenko 07 Nov '07

07 Nov '07

Hi. Setting the ADC48K44K greatly improves capture quality at 48k sampling rate. With this bit clear ADC does ZOH interpolation of every 22th sample at 48k. At frequencies higher than 48k there ADC performs a little better with ADC48K44K bit set. At 44.1k ADC performs a little better with this bit clear. At frequencies below 44.1k there is no difference. Regards, Tim.

2 1

[alsa-devel] [PATCH] hda: STAC92HD71 codec mixer
by Matthew Ranostay 07 Nov '07

07 Nov '07

Added analog loopback support and missing ADC capture mixer for the STAC92HD71 codec family. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 43708ece3f6f pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Nov 06 11:56:17 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Nov 06 21:05:54 2007 -0500 @@ -428,6 +428,11 @@ static struct hda_verb stac92hd71bxx_cor { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* connect ports 0d and 0f to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + /* unmute dac0 input in audio mixer */ + { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, @@ -512,8 +517,7 @@ static struct hda_verb stac9205_core_ini .put = stac92xx_volknob_put, \ .private_value = 127 | (knob_nid << 16), \ } - - + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -530,12 +534,19 @@ static struct snd_kcontrol_new stac92hd7 STAC_VOLKNOB(0x28), /* hardware gain controls */ - HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), - - HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), - HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ };

2 1

[alsa-devel] Mr. Takashi Iwai Help-me please
by Eduardo 07 Nov '07

07 Nov '07

Mr. Takashi Escuse-me for my question, but i want a developer a driver for MCM590 i have a dump for USB and the manuals to develop audio driver for linux, but nothing for a manufacter documentation. This is a Philips Microsystem with a USB port This USB periferic need a special boot sequence, i debug this sequence in the linux kernel and the default boot for "usbaudio.c" don't solve the problem. Do you can help-me? Eduardo Please answer this mail with no or yes tks!

2 1

[alsa-devel] [PATCH] soc/sh: help GCC improve generated code
by Manuel Lauss 06 Nov '07

06 Nov '07

2 1

[alsa-devel] [PATCH] hda: Added new IDT codec family
by Matthew Ranostay 06 Nov '07

06 Nov '07

Added initial support for the STAC92HD71BXX family of codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 3e1a17f8baca pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Wed Oct 31 17:36:20 2007 +0100 +++ b/pci/hda/hda_codec.h Sat Nov 03 22:55:14 2007 -0500 @@ -92,6 +92,8 @@ enum { */ #define AC_VERB_SET_STREAM_FORMAT 0x200 #define AC_VERB_SET_AMP_GAIN_MUTE 0x300 +#define AC_VERB_SET_RIGHT_AMP_IN_MUTE 0x350 +#define AC_VERB_SET_LEFT_AMP_IN_MUTE 0x360 #define AC_VERB_SET_PROC_COEF 0x400 #define AC_VERB_SET_COEF_INDEX 0x500 #define AC_VERB_SET_CONNECT_SEL 0x701 diff -r 3e1a17f8baca pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Oct 31 17:36:20 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Nov 05 09:34:05 2007 -0500 @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD71BXX_REF, + STAC_92HD71BXX_MODELS +}; + +enum { STAC_925x_REF, STAC_M2_2, STAC_MA6, @@ -172,6 +177,23 @@ static hda_nid_t stac9200_dac_nids[1] = 0x02, }; +static hda_nid_t stac92hd71bxx_adc_nids[2] = { + 0x12, 0x13, +}; + +static hda_nid_t stac92hd71bxx_mux_nids[2] = { + 0x1a, 0x1b +}; + +static hda_nid_t stac92hd71bxx_dac_nids[2] = { + 0x10, /*0x11, */ +}; + +#define STAC92HD71BXX_NUM_DMICS 2 +static hda_nid_t stac92hd71bxx_dmic_nids[STAC92HD71BXX_NUM_DMICS + 1] = { + 0x18, 0x19, 0 +}; + static hda_nid_t stac925x_adc_nids[1] = { 0x03, }; @@ -236,6 +258,11 @@ static hda_nid_t stac922x_pin_nids[10] = static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, +}; + +static hda_nid_t stac92hd71bxx_pin_nids[10] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x14, 0x18, 0x19, 0x1e, }; static hda_nid_t stac927x_pin_nids[14] = { @@ -393,6 +420,25 @@ static struct hda_verb stac9200_eapd_ini /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + +static struct hda_verb stac92hd71bxx_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* unmute right and left channels for node 0x0a */ + { 0x0a, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, + { 0x0a, AC_VERB_SET_RIGHT_AMP_IN_MUTE, 0x0}, + /* connect headphone jack to dac1 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + /* unmute right and left channels for node 0x0d */ + { 0x0d, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, + { 0x0d, AC_VERB_SET_RIGHT_AMP_IN_MUTE, 0x0}, + /* unmute right and left channels for node 0x0f */ + { 0x0f, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, + { 0x0f, AC_VERB_SET_RIGHT_AMP_IN_MUTE, 0x0}, + /* unmute mono out node */ + { 0x14, AC_VERB_SET_LEFT_AMP_IN_MUTE, 0x0}, {} }; @@ -480,6 +526,21 @@ static struct snd_kcontrol_new stac9200_ HDA_CODEC_VOLUME("Capture Volume", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x0a, 0, HDA_OUTPUT), HDA_CODEC_VOLUME("Capture Mux Volume", 0x0c, 0, HDA_OUTPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), + STAC_INPUT_SOURCE(2), + STAC_VOLKNOB(0x28), + + /* hardware gain controls */ + HDA_CODEC_VOLUME("Digital Mic 1 Volume", 0x18, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Digital Mic 2 Volume", 0x19, 0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Capture Volume", 0x1c, 0, HDA_OUTPUT), + HDA_CODEC_MUTE("Capture Switch", 0x1c, 0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Capture Mux Volume", 0x1a, 0, HDA_OUTPUT), { } /* end */ }; @@ -835,6 +896,27 @@ static struct snd_pci_quirk stac925x_cfg SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6), SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6), SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2), + {} /* terminator */ +}; + +static unsigned int ref92hd71bxx_pin_configs[10] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, + 0x0181302e, 0x01114010, 0x01a19020, 0x90a000f0, + 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_REF] = ref92hd71bxx_pin_configs, +}; + +static const char *stac92hd71bxx_models[STAC_92HD71BXX_MODELS] = { + [STAC_92HD71BXX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD71BXX_REF), {} /* terminator */ }; @@ -2614,6 +2696,77 @@ static int patch_stac925x(struct hda_cod return 0; } + +static int patch_stac92hd71bxx(struct hda_codec *codec) { + struct sigmatel_spec *spec; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->pin_nids = stac92hd71bxx_pin_nids; + + spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, + stac92hd71bxx_models, + stac92hd71bxx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC92HD71BXX, " + "using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } + else { + spec->pin_configs = stac92hd71bxx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + stac92xx_enable_gpio_mask(codec); + + spec->init = stac92hd71bxx_core_init; + spec->mixer = stac92hd71bxx_mixer; + + spec->mux_nids = stac92hd71bxx_mux_nids; + spec->adc_nids = stac92hd71bxx_adc_nids; + spec->dmic_nids = stac92hd71bxx_dmic_nids; + spec->dmux_nid = 0x1c; + + spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); + spec->num_dmics = STAC92HD71BXX_NUM_DMICS; + + spec->multiout.num_dacs = 2; + spec->multiout.hp_nid = 0x11; + spec->multiout.dac_nids = stac92hd71bxx_dac_nids; + + err = stac92xx_parse_auto_config(codec, 0x21, 0x23); + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD71BXX_REF; + goto again; + } + err = -EINVAL; + } + + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +}; static int patch_stac922x(struct hda_codec *codec) { @@ -3176,5 +3329,7 @@ struct hda_codec_preset snd_hda_preset_s { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d76b0, .name = "STAC92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ }; +

3 4

[alsa-devel] Wyse Winterm CS5535 Driver
by Darryl Bond 05 Nov '07

05 Nov '07

I have a Wyse S50 terminal that I am booting diskless. I have been trying to get sound working. The box has a cs5535 chipset. After much fiddling in the kernel source (2.6.22.5 <http://2.6.22.5> ) I have been able to make it work. I'm not sure if it is a cs5535 problem as the code that fixes it is in ac97_pcm.c. It would appear that the card is getting passed an unexpected rate. The function sound/pci/ac97/ac97_pcm.c snd_ac97_set_rate() has a section: case AC97_PCM_FRONT_DAC_RATE: case AC97_PCM_LR_ADC_RATE: if ((ac97->regs[AC97_EXTENDED_STATUS] & AC97_EA_VRA) == 0) /* VRA */ if (rate != 48000 && rate != 96000) return -EINVAL; break; If I use 'aplay wavfile' the rate passed to this function is 44100. I I changethe code to: if (rate != 48000 && rate != 96000 && rate != 44100) it plays but at a slightly incorrect rate. If I use the Citrix ICA client (which is what I really want to work) the rate passed in is 8000. If I add 8000 to the test, it works, but is far too fast. Any hints as to what I should try? Regards Darryl Bond __________________________________________________________________________________ DISCLAIMER The contents of this electronic message and any attachments are intended only for the addressee and may contain legally privileged, personal, sensitive or confidential information. If you are not the intended addressee, and have received this email, any transmission, distribution, downloading, printing or photocopying of the contents of this message or attachments is strictly prohibited. Any legal privilege or confidentiality attached to this message and attachments is not waived, lost or destroyed by reason of delivery to any person other than intended addressee. If you have received this message and are not the intended addressee you should notify the sender by return email and destroy all copies of the message and any attachments. Unless expressly attributed, the views expressed in this email do not necessarily represent the views of the company.

2 4

[PATCH] ASoC: sh: improve generated code for HAC module (AC97)
by Manuel Lauss 05 Nov '07

05 Nov '07

Change loops in ac97_read/write functions to count down to zero rather than up. Gcc will then use the 'dt' (decrement-and-test) op instead of an increment/compare op-pair. Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> --- sound/soc/sh/hac.c | 13 +++++-------- 1 files changed, 5 insertions(+), 8 deletions(-) diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c index 8e3f039..34b77b9 100644 --- a/sound/soc/sh/hac.c +++ b/sound/soc/sh/hac.c @@ -105,7 +105,7 @@ static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, unsigned int to1, to2, i; unsigned short adr; - for (i = 0; i < AC97_READ_RETRY; ++i) { + for (i = AC97_READ_RETRY; i; i--) { *v = 0; /* wait for HAC to receive something from the codec */ for (to1 = TMO_E4; @@ -132,7 +132,7 @@ static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, udelay(21); } HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); - return (i < AC97_READ_RETRY); + return i; } static unsigned short hac_read_codec_aux(struct hac_priv *hac, @@ -141,7 +141,7 @@ static unsigned short hac_read_codec_aux(struct hac_priv *hac, unsigned short val; unsigned int i, to; - for (i = 0; i < AC97_READ_RETRY; i++) { + for (i = AC97_READ_RETRY; i; i--) { /* send_read_request */ local_irq_disable(); HACREG(HACTSR) &= ~(TSR_CMDAMT); @@ -159,10 +159,7 @@ static unsigned short hac_read_codec_aux(struct hac_priv *hac, break; } - if (i == AC97_READ_RETRY) - return ~0; - - return val; + return i ? val : ~0; } static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, @@ -172,7 +169,7 @@ static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, struct hac_priv *hac = &hac_cpu_data[unit_id]; unsigned int i, to; /* write_codec_aux */ - for (i = 0; i < AC97_WRITE_RETRY; i++) { + for (i = AC97_WRITE_RETRY; i; i--) { /* send_write_request */ local_irq_disable(); HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); -- 1.5.3.4

1 0

[alsa-devel] [PATCH] [ML403-AC97CR] Fix capture/periodic overrun bug.
by Joachim Foerster 05 Nov '07

05 Nov '07

Hi all, this is a bug fix for my ALSA driver I posted in August. Since then I hadn't the time to look at it, but some days ago I took some time and found the reason why there are so many overruns while capturing. Furthermore, this patch removes the one "heavy" printk for capture debugging. From: Joachim Foerster <JOFT(a)gmx.de> We have to do fairly accurate counting of the minimal periods, instead of being lazy and just setting the number to zero as soon as one period elapses. Signed-off-by: Joachim Foerster <JOFT(a)gmx.de> --- sound/drivers/ml403-ac97cr.c | 6 ++---- sound/drivers/pcm-indirect2.c | 20 ++------------------ 2 files changed, 4 insertions(+), 22 deletions(-) diff --git a/sound/drivers/ml403-ac97cr.c b/sound/drivers/ml403-ac97cr.c index 2636249..1b16a21 100644 --- a/sound/drivers/ml403-ac97cr.c +++ b/sound/drivers/ml403-ac97cr.c @@ -28,11 +28,9 @@ * accesses to a minimum, because after a variable amount of accesses, the AC97 * controller doesn't raise the register access finished bit anymore ... * - * - Capture support works - basically, but after ~30s (with rates > ~20kHz) - * ALSA stops reading captured samples from the intermediate buffer and - * therefore a overrun happens - ATM I don't know what's wrong. - * * - Playback support seems to be pretty stable - no issues here. + * - Capture support "works" now, too. Overruns don't happen any longer so often. + * But there might still be some ... */ #include <sound/driver.h> diff --git a/sound/drivers/pcm-indirect2.c b/sound/drivers/pcm-indirect2.c index a36215a..ef74a38 100644 --- a/sound/drivers/pcm-indirect2.c +++ b/sound/drivers/pcm-indirect2.c @@ -403,7 +403,7 @@ snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream, rec->min_multiple); rec->mul_elapsed++; #endif - rec->min_periods = 0; + rec->min_periods = (rec->min_periods % rec->min_multiple); snd_pcm_period_elapsed(substream); } } @@ -568,24 +568,8 @@ snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream, rec->mul_elapsed_real += (rec->min_periods / rec->min_multiple); rec->mul_elapsed++; - - if (!(rec->mul_elapsed % 4)) { - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int appl_ptr = - frames_to_bytes(runtime, - (unsigned int)runtime->control-> - appl_ptr) % rec->sw_buffer_size; - int diff = rec->sw_data - appl_ptr; - if (diff < 0) - diff += rec->sw_buffer_size; - snd_printk(KERN_DEBUG - "STAT: mul_elapsed: %d, sw_data: %u, " - "appl_ptr (bytes): %u, diff: %d\n", - rec->mul_elapsed, rec->sw_data, appl_ptr, - diff); - } #endif - rec->min_periods = 0; + rec->min_periods = (rec->min_periods % rec->min_multiple); snd_pcm_period_elapsed(substream); } } -- 1.5.2.5

3 3

[alsa-devel] [PATCH] hda: STAC9228 DMIC
by Matthew Ranostay 05 Nov '07

05 Nov '07

Added support for the dmics and enabled EAPD for several laptops with STAC9228 cards. --- Signed-off-by: Matthew Ranostay <mranostay(a)gmail.com> diff -r 808153682813 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Oct 26 15:10:15 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Nov 01 15:46:17 2007 -0500 @@ -2778,14 +2778,21 @@ static int patch_stac927x(struct hda_cod spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; spec->dmux_nid = 0x1b; + + /* Enable DMIC0 */ + stac92xx_set_config_reg(codec, 0x13, 0x90a60040); + + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; break; default: - spec->num_dmics = 0; + spec->num_dmics = 0; + + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; } spec->multiout.dac_nids = spec->dac_nids; - /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20);

3 2

[alsa-devel] HDA: Add Asus VX1 support.
by Tobin Davis 05 Nov '07

05 Nov '07

Simple patch to add the Asus VX1 laptop to the Analog Devices pci quirk list. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] Upgrading alsa drivers. current and hg from suse site
by Nigel Henry 03 Nov '07

03 Nov '07

I try to help folks now and then with sound problems. I don't have an hda intel card, but try to point folks in the right direction to get help. Question: I've sorted out the upgrading of the alsa driver on Fedora, Archlinux, and Debian, even though I don't need too, as my audigy2 soundblaster, and on the other machine, an ensoniq (ens1371) are fine. this was from the latest 1.0.15 release at http://alsa-project.org. My question is to do with the nightly hg releases obtainable from. ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver/ Will these install using the same ./configure, make, (su to root), then make install, as with upgrading to the current 1.0.15 version from. http://alsa-project.org. Sorry if this is a trivial question, but I don't want to give bad information to folks looking for help. I've only recently subscribed to the alsa-devel list, and hadn't realised how much effort was being put in to resolve soundcard problems, particularly hda intel ones. Great work folks. I'm not clued up enough to help out, but do appreciate your hard work. Nigel.

2 1

[alsa-devel] hdmi sound output on abit an-m2hd mainboard (ALC888)
by Gernot Pansy 02 Nov '07

02 Nov '07

hy, can some one explain me what the developers need to get hdmi sound working on this mainboard? it's a nvidia nforce mcp67 with an alc888 sound chip onboard. should i open a bug? thx, gernot notz@hollywood:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 notz@hollywood:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 notz@hollywood:~$ aplay -L default:CARD=NVidia HDA NVidia, ALC883 Analog Default Audio Device front:CARD=NVidia,DEV=0 HDA NVidia, ALC883 Analog Front speakers surround40:CARD=NVidia,DEV=0 HDA NVidia, ALC883 Analog 4.0 Surround output to Front and Rear speakers surround41:CARD=NVidia,DEV=0 HDA NVidia, ALC883 Analog 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=NVidia,DEV=0 HDA NVidia, ALC883 Analog 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=NVidia,DEV=0 HDA NVidia, ALC883 Analog 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=NVidia,DEV=0 HDA NVidia, ALC883 Analog 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers null Discard all samples (playback) or generate zero samples (capture) notz@hollywood:~$ cat /proc/asound/card0/codec#0 Codec: Realtek ALC888 Address: 0 Vendor Id: 0x10ec0888 Subsystem Id: 0x147ba601 Revision Id: 0x100001 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] 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=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Connection: 1 0x1f 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] [0x00 0x00] [0x1a 0x1a] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x05 0x0b 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 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0836: IN OUT Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0836: IN OUT Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Pin Default 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Pin Default 0x02a1983f: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Pin Default 0x01813031: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 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] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08373e: IN OUT HP Detect Pin Default 0x02214c1f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330132: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x99830133: [Fixed] Line In at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x01451120: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono 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] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

2 2

[alsa-devel] STAC9228
by Matthew Meyer 02 Nov '07

02 Nov '07

Hi, I'm using a Fujitsu Stylistic ST5112 tablet PC with Ubuntu 7.10. It has the Sigma Tel STAC9228 sound card. I am having tons of trouble getting the sound to work; I had this problem in 7.04 too. I used both alsa 1.0.14 and 1.0.15RC3, and no luck. I read through ALSA-Configuration.txt and found the model at line 924, and I tried using model=ref, model=3stack and model=5stack, but none of them worked. The only time I've been able to get any sound at all was when I had a completely wrong bit of text entered for model=xxx, and in that case the system beep worked, but no sound cards were recognized or loaded at all. I've checked for the card being muted, and it's not... so I don't know what could be wrong. Maybe you have some insight, otherwise, I guess this could be a bug? Anyway, I hope this helps. Matt lspci -nv output: 00:00.0 0600: 8086:27a0 (rev 03) Subsystem: 10cf:1378 Flags: bus master, fast devsel, latency 0 Capabilities: <access denied> 00:02.0 0300: 8086:27a2 (rev 03) (prog-if 00 [VGA]) Subsystem: 10cf:137e Flags: bus master, fast devsel, latency 0, IRQ 19 Memory at f0300000 (32-bit, non-prefetchable) [size=512K] I/O ports at 1800 [size=8] Memory at e0000000 (32-bit, prefetchable) [size=256M] Memory at f0400000 (32-bit, non-prefetchable) [size=256K] Capabilities: <access denied> 00:02.1 0380: 8086:27a6 (rev 03) Subsystem: 10cf:137e Flags: bus master, fast devsel, latency 0 Memory at f0380000 (32-bit, non-prefetchable) [size=512K] Capabilities: <access denied> 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:13b4 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> 00:1c.0 0604: 8086:27d0 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f0100000-f01fffff Prefetchable memory behind bridge: 0000000090000000-00000000900fffff Capabilities: <access denied> 00:1c.1 0604: 8086:27d2 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=04, sec-latency=0 Capabilities: <access denied> 00:1c.2 0604: 8086:27d4 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=05, sec-latency=0 Memory behind bridge: 90100000-901fffff Capabilities: <access denied> 00:1d.0 0c03: 8086:27c8 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 1820 [size=32] 00:1d.1 0c03: 8086:27c9 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1840 [size=32] 00:1d.2 0c03: 8086:27ca (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 1860 [size=32] 00:1d.3 0c03: 8086:27cb (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1880 [size=32] 00:1d.7 0c03: 8086:27cc (rev 02) (prog-if 20 [EHCI]) Subsystem: 10cf:138a Flags: bus master, medium devsel, latency 0, IRQ 20 Memory at f0644000 (32-bit, non-prefetchable) [size=1K] Capabilities: <access denied> 00:1e.0 0604: 8086:2448 (rev e2) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=08, subordinate=10, sec-latency=32 I/O behind bridge: 00003000-00003fff Memory behind bridge: f0200000-f02fffff Prefetchable memory behind bridge: 0000000088000000-000000008fffffff Capabilities: <access denied> 00:1f.0 0601: 8086:27b9 (rev 02) Subsystem: 10cf:1384 Flags: bus master, medium devsel, latency 0 Capabilities: <access denied> 00:1f.1 0101: 8086:27df (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: 10cf:1385 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] 00:1f.2 0106: 8086:27c5 (rev 02) (prog-if 01 [AHCI 1.0]) Subsystem: 10cf:1387 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 22 I/O ports at 18d0 [size=8] I/O ports at 18c4 [size=4] I/O ports at 18c8 [size=8] I/O ports at 18c0 [size=4] I/O ports at 18b0 [size=16] Memory at f0644400 (32-bit, non-prefetchable) [size=1K] Capabilities: <access denied> 00:1f.3 0c05: 8086:27da (rev 02) Subsystem: 10cf:1388 Flags: medium devsel, IRQ 11 I/O ports at 18e0 [size=32] 02:00.0 0200: 11ab:4363 (rev 12) Subsystem: 10cf:139a Flags: bus master, fast devsel, latency 0, IRQ 19 Memory at f0100000 (64-bit, non-prefetchable) [size=16K] I/O ports at 2000 [size=256] [virtual] Expansion ROM at 90000000 [disabled] [size=128K] Capabilities: <access denied> 05:00.0 0200: 168c:001c (rev 01) Subsystem: 10cf:139c Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 90100000 (64-bit, non-prefetchable) [size=64K] Capabilities: <access denied> 08:03.0 0607: 1217:7134 (rev 21) Subsystem: 10cf:131e Flags: bus master, stepping, slow devsel, latency 168, IRQ 19 Memory at f0203000 (32-bit, non-prefetchable) [size=4K] Bus: primary=08, secondary=09, subordinate=0c, sec-latency=176 Memory window 0: 88000000-8bfff000 (prefetchable) Memory window 1: 94000000-97fff000 I/O window 0: 00003000-000030ff I/O window 1: 00003400-000034ff 16-bit legacy interface ports at 0001 08:03.1 0607: 1217:7134 (rev 21) Subsystem: 10cf:131e Flags: bus master, stepping, slow devsel, latency 168, IRQ 19 Memory at f0204000 (32-bit, non-prefetchable) [size=4K] Bus: primary=08, secondary=0d, subordinate=10, sec-latency=176 Memory window 0: 8c000000-8ffff000 (prefetchable) Memory window 1: 98000000-9bfff000 I/O window 0: 00003800-000038ff I/O window 1: 00003c00-00003cff 16-bit legacy interface ports at 0001 08:03.2 0805: 1217:7120 (rev 01) Subsystem: 10cf:131e Flags: slow devsel, IRQ 19 Memory at f0202800 (32-bit, non-prefetchable) [size=256] Capabilities: <access denied> 08:03.3 0680: 1217:7130 (rev 01) Subsystem: 10cf:131e Flags: slow devsel, IRQ 11 Memory at f0200000 (32-bit, non-prefetchable) [size=4K] Capabilities: <access denied> 08:03.4 0c00: 1217:00f7 (rev 02) (prog-if 10 [OHCI]) Subsystem: 1217:00f7 Flags: bus master, medium devsel, latency 32, IRQ 19 Memory at f0201000 (32-bit, non-prefetchable) [size=4K] Memory at f0202000 (32-bit, non-prefetchable) [size=2K] Capabilities: <access denied>

2 1

[alsa-devel] Output levels issue on Plantronics .Audio 510 USB Headset
by David Hilvert 01 Nov '07

01 Nov '07

The Plantronics .Audio 510 USB headset (Vendor=047f ProdID=c001) seems to have certain output levels issues with drivers in 2.6.23-rc2-mm2 (at least). In particular, setting amixer -c 1 sset PCM 3 results in no output, while amixer -c 1 sset PCM 4 results in levels that are somewhat loud for applications tested. Volume appears to be roughly linear in the range above PCM=4, if loud for applications tested. The USB ID information (Vendor=047f ProdID=c001) doesn't seem to be noted as exceptional by any driver code in the 2.6.23-rc2-mm2 kernel, and web searches for 'plantronics' or 'audio 510' and linux or alsa seem to return little pertinent information. What would be most useful to try next? Various outputs follow: $ amixer -c 1 Simple mixer control 'PCM',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 151 Mono: Front Left: Playback 6 [4%] [-27.29dB] [on] Front Right: Playback 6 [4%] [-27.29dB] [on] Simple mixer control 'Mic',0 Capabilities: cvolume cswitch cswitch-joined Capture channels: Mono Limits: Capture 0 - 16 Mono: Capture 14 [88%] [21.00dB] [on] Simple mixer control 'Auto Gain Control',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] $ cat /proc/bus/usb/devices T: Bus=05 Lev=01 Prnt=01 Port=01 Cnt=01 Dev#= 6 Spd=12 MxCh= 0 D: Ver= 1.10 Cls=00(>ifc ) Sub=00 Prot=00 MxPS=64 #Cfgs= 1 P: Vendor=047f ProdID=c001 Rev= 1.00 C:* #Ifs= 4 Cfg#= 1 Atr=a0 MxPwr=100mA I:* If#= 0 Alt= 0 #EPs= 0 Cls=01(audio) Sub=01 Prot=00 Driver=snd-usb-audio I:* If#= 1 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio I: If#= 1 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio E: Ad=01(O) Atr=09(Isoc) MxPS= 200 Ivl=1ms I:* If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio E: Ad=82(I) Atr=05(Isoc) MxPS= 100 Ivl=1ms I:* If#= 3 Alt= 0 #EPs= 1 Cls=03(HID ) Sub=00 Prot=00 Driver=usbhid E: Ad=83(I) Atr=03(Int.) MxPS= 4 Ivl=32ms

1 0

[alsa-devel] Philips USB sound MCM590
by Eduardo 01 Nov '07

01 Nov '07

Hi peoples, excuse my poor English I'm newbe in this community, by i have a problem. I've a Philips mcm590 Microsystem, this microsystem have a usb connection but only windows driver. I try a dump my linux kernel using messages for dmesg and i conclude, this sound need a special boot sequence in a usbaudio.c I install a winxp in my Fedora in a VirtualBox and make a dump for USB connection using a USBNOOP also i print the "PROGRAMMING GUIDE FOR LINUX USB DEVICE" and "Universal Serial Bus device class defintion for audio devices" but i'm a newbe in a kernel devel and C devel I'm java developer i need your help Eduardo

1 0

[alsa-devel] Info + Osprey 440 Audio help
by Norbert Naveen 01 Nov '07

01 Nov '07

1 0

[alsa-devel] [PATCH] cmipci at 96kHz
by Timofei V. Bondarenko 31 Oct '07

31 Oct '07

Hi. This patch adds support for 88.2k, 96k, and 128k samplerates on cmi8738-55 chip. Analog playback works fine on all channels. Analog capture works well too, though the extra samples seems interpolated by hardware. spdif playback and capture works fine. Regards, Tim.

3 6

[alsa-devel] via82xx DC offset at -30dB
by plessas＠mur.at 31 Oct '07

31 Oct '07

Dear List, having a Compaq Presario notebook in front of me, running Debian testing and a 2.6.18-4-686 kernel, i have a strange DC offset when trying to record audio via the mic/line input. As the audacity level meters suggest, it resides at -30dBFS. Audio comes through and gets recorded though, but the offset severely limits the headroom of course. The sensitivity of the line/mic input seems to be quite high, although i have the mic boost setting turned off of course. Setting the module option ac97_quirk=yes did not help either. Can it be that a soundcard of this strange behavior is manufactured, or is it an alsa/driver issue? I really don't want to install another M$ operating system to check if it works there, so any help is deeply appreciated. lspci lists the device as: 00:07.5 Multimedia audio controller: VIA Technologies, Inc. VT82C686 AC97 Audio Controller (rev 20) regards, PP PS: please CC: me, as i am not subscribed to alsa-devel (yet)

3 2

[alsa-devel] Testers Wanted: OPL3 hwdep loader interface
by Takashi Iwai 31 Oct '07

31 Oct '07

Hi, one of big pending patches on my local tree is the removal of seq-instr interface. The only real user of instrument layer is OPL3 patch loader, and I'd like to move this to hwdep interface. The patch below is to move the OPL3 patch loading to hwdep device. Two more patches will follow in other mails for alsa-lib addition and sbiload change. Can someone test whether these patches work? Unfortunately, I have no hardware with OPL3 at this moment around here. Thanks, Takashi diff -r 88d273f360dd drivers/opl3/opl3_lib.c --- a/drivers/opl3/opl3_lib.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_lib.c Fri Oct 26 16:09:56 2007 +0200 @@ -327,6 +327,7 @@ static int snd_opl3_free(struct snd_opl3 snd_assert(opl3 != NULL, return -ENXIO); if (opl3->private_free) opl3->private_free(opl3); + snd_opl3_clear_patches(opl3); release_and_free_resource(opl3->res_l_port); release_and_free_resource(opl3->res_r_port); kfree(opl3); @@ -521,6 +522,7 @@ int snd_opl3_hwdep_new(struct snd_opl3 * /* operators - only ioctl */ hw->ops.open = snd_opl3_open; hw->ops.ioctl = snd_opl3_ioctl; + hw->ops.write = snd_opl3_write; hw->ops.release = snd_opl3_release; opl3->seq_dev_num = seq_device; diff -r 88d273f360dd drivers/opl3/opl3_midi.c --- a/drivers/opl3/opl3_midi.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_midi.c Fri Oct 26 16:09:56 2007 +0200 @@ -289,8 +289,6 @@ void snd_opl3_note_on(void *p, int note, void snd_opl3_note_on(void *p, int note, int vel, struct snd_midi_channel *chan) { struct snd_opl3 *opl3; - struct snd_seq_instr wanted; - struct snd_seq_kinstr *kinstr; int instr_4op; int voice; @@ -306,11 +304,13 @@ void snd_opl3_note_on(void *p, int note, unsigned char voice_offset; unsigned short opl3_reg; unsigned char reg_val; + unsigned char prg, bank; int key = note; unsigned char fnum, blocknum; int i; + struct fm_patch *patch; struct fm_instrument *fm; unsigned long flags; @@ -320,19 +320,17 @@ void snd_opl3_note_on(void *p, int note, snd_printk("Note on, ch %i, inst %i, note %i, vel %i\n", chan->number, chan->midi_program, note, vel); #endif - wanted.cluster = 0; - wanted.std = SNDRV_SEQ_INSTR_TYPE2_OPL2_3; /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, drum voice numbers are notes on drum channel */ if (opl3->synth_mode == SNDRV_OPL3_MODE_SEQ) { if (chan->drum_channel) { /* percussion instruments are located in bank 128 */ - wanted.bank = 128; - wanted.prg = note; + bank = 128; + prg = note; } else { - wanted.bank = chan->gm_bank_select; - wanted.prg = chan->midi_program; + bank = chan->gm_bank_select; + prg = chan->midi_program; } } else { /* Prepare for OSS mode */ @@ -340,8 +338,8 @@ void snd_opl3_note_on(void *p, int note, return; /* OSS instruments are located in bank 127 */ - wanted.bank = 127; - wanted.prg = chan->midi_program; + bank = 127; + prg = chan->midi_program; } spin_lock_irqsave(&opl3->voice_lock, flags); @@ -353,15 +351,14 @@ void snd_opl3_note_on(void *p, int note, } __extra_prg: - kinstr = snd_seq_instr_find(opl3->ilist, &wanted, 1, 0); - if (kinstr == NULL) { + patch = snd_opl3_find_patch(opl3, prg, bank, 0); + if (!patch) { spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } - fm = KINSTR_DATA(kinstr); - - switch (fm->type) { + fm = &patch->inst; + switch (patch->type) { case FM_PATCH_OPL2: instr_4op = 0; break; @@ -371,14 +368,12 @@ void snd_opl3_note_on(void *p, int note, break; } default: - snd_seq_instr_free_use(opl3->ilist, kinstr); spin_unlock_irqrestore(&opl3->voice_lock, flags); return; } - #ifdef DEBUG_MIDI snd_printk(" --> OPL%i instrument: %s\n", - instr_4op ? 3 : 2, kinstr->name); + instr_4op ? 3 : 2, patch->name); #endif /* in SYNTH mode, application takes care of voices */ /* in SEQ mode, allocate voice on free OPL3 channel */ @@ -569,8 +564,6 @@ void snd_opl3_note_on(void *p, int note, /* get extra pgm, but avoid possible loops */ extra_prg = (extra_prg) ? 0 : fm->modes; - snd_seq_instr_free_use(opl3->ilist, kinstr); - /* do the bookkeeping */ vp->time = opl3->use_time++; vp->note = key; @@ -601,12 +594,12 @@ void snd_opl3_note_on(void *p, int note, /* allocate extra program if specified in patch library */ if (extra_prg) { if (extra_prg > 128) { - wanted.bank = 128; + bank = 128; /* percussions start at 35 */ - wanted.prg = extra_prg - 128 + 35 - 1; + prg = extra_prg - 128 + 35 - 1; } else { - wanted.bank = 0; - wanted.prg = extra_prg - 1; + bank = 0; + prg = extra_prg - 1; } #ifdef DEBUG_MIDI snd_printk(" *** allocating extra program\n"); diff -r 88d273f360dd drivers/opl3/opl3_oss.c --- a/drivers/opl3/opl3_oss.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_oss.c Fri Oct 26 16:09:56 2007 +0200 @@ -195,17 +195,6 @@ static int snd_opl3_close_seq_oss(struct /* load patch */ -/* offsets for SBI params */ -#define AM_VIB 0 -#define KSL_LEVEL 2 -#define ATTACK_DECAY 4 -#define SUSTAIN_RELEASE 6 -#define WAVE_SELECT 8 - -/* offset for SBI instrument */ -#define CONNECTION 10 -#define OFFSET_4OP 11 - /* from sound_config.h */ #define SBFM_MAXINSTR 256 @@ -213,112 +202,42 @@ static int snd_opl3_load_patch_seq_oss(s const char __user *buf, int offs, int count) { struct snd_opl3 *opl3; - int err = -EINVAL; + struct sbi_instrument sbi; + char name[32]; + int err, type; snd_assert(arg != NULL, return -ENXIO); opl3 = arg->private_data; - if ((format == FM_PATCH) || (format == OPL3_PATCH)) { - struct sbi_instrument sbi; + if (format == FM_PATCH) + type = FM_PATCH_OPL2; + else if (format == OPL3_PATCH) + type = FM_PATCH_OPL3; + else + return -EINVAL; - size_t size; - struct snd_seq_instr_header *put; - struct snd_seq_instr_data *data; - struct fm_xinstrument *xinstr; + if (count < (int)sizeof(sbi)) { + snd_printk("FM Error: Patch record too short\n"); + return -EINVAL; + } + if (copy_from_user(&sbi, buf, sizeof(sbi))) + return -EFAULT; - struct snd_seq_event ev; - int i; + if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { + snd_printk("FM Error: Invalid instrument number %d\n", + sbi.channel); + return -EINVAL; + } - mm_segment_t fs; + memset(name, 0, sizeof(name)); + sprintf(name, "Chan%d", sbi.channel); - if (count < (int)sizeof(sbi)) { - snd_printk("FM Error: Patch record too short\n"); - return -EINVAL; - } - if (copy_from_user(&sbi, buf, sizeof(sbi))) - return -EFAULT; + err = snd_opl3_load_patch(opl3, sbi.channel, 127, type, name, NULL, + sbi.operators); + if (err < 0) + return err; - if (sbi.channel < 0 || sbi.channel >= SBFM_MAXINSTR) { - snd_printk("FM Error: Invalid instrument number %d\n", sbi.channel); - return -EINVAL; - } - - size = sizeof(*put) + sizeof(struct fm_xinstrument); - put = kzalloc(size, GFP_KERNEL); - if (put == NULL) - return -ENOMEM; - /* build header */ - data = &put->data; - data->type = SNDRV_SEQ_INSTR_ATYPE_DATA; - strcpy(data->data.format, SNDRV_SEQ_INSTR_ID_OPL2_3); - /* build data section */ - xinstr = (struct fm_xinstrument *)(data + 1); - xinstr->stype = FM_STRU_INSTR; - - for (i = 0; i < 2; i++) { - xinstr->op[i].am_vib = sbi.operators[AM_VIB + i]; - xinstr->op[i].ksl_level = sbi.operators[KSL_LEVEL + i]; - xinstr->op[i].attack_decay = sbi.operators[ATTACK_DECAY + i]; - xinstr->op[i].sustain_release = sbi.operators[SUSTAIN_RELEASE + i]; - xinstr->op[i].wave_select = sbi.operators[WAVE_SELECT + i]; - } - xinstr->feedback_connection[0] = sbi.operators[CONNECTION]; - - if (format == OPL3_PATCH) { - xinstr->type = FM_PATCH_OPL3; - for (i = 0; i < 2; i++) { - xinstr->op[i+2].am_vib = sbi.operators[OFFSET_4OP + AM_VIB + i]; - xinstr->op[i+2].ksl_level = sbi.operators[OFFSET_4OP + KSL_LEVEL + i]; - xinstr->op[i+2].attack_decay = sbi.operators[OFFSET_4OP + ATTACK_DECAY + i]; - xinstr->op[i+2].sustain_release = sbi.operators[OFFSET_4OP + SUSTAIN_RELEASE + i]; - xinstr->op[i+2].wave_select = sbi.operators[OFFSET_4OP + WAVE_SELECT + i]; - } - xinstr->feedback_connection[1] = sbi.operators[OFFSET_4OP + CONNECTION]; - } else { - xinstr->type = FM_PATCH_OPL2; - } - - put->id.instr.std = SNDRV_SEQ_INSTR_TYPE2_OPL2_3; - put->id.instr.bank = 127; - put->id.instr.prg = sbi.channel; - put->cmd = SNDRV_SEQ_INSTR_PUT_CMD_CREATE; - - memset (&ev, 0, sizeof(ev)); - ev.source.client = SNDRV_SEQ_CLIENT_OSS; - ev.dest = arg->addr; - - ev.flags = SNDRV_SEQ_EVENT_LENGTH_VARUSR; - ev.queue = SNDRV_SEQ_QUEUE_DIRECT; - - fs = snd_enter_user(); - __again: - ev.type = SNDRV_SEQ_EVENT_INSTR_PUT; - ev.data.ext.len = size; - ev.data.ext.ptr = put; - - err = snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, &ev, - opl3->seq_client, 0, 0); - if (err == -EBUSY) { - struct snd_seq_instr_header remove; - - memset (&remove, 0, sizeof(remove)); - remove.cmd = SNDRV_SEQ_INSTR_FREE_CMD_SINGLE; - remove.id.instr = put->id.instr; - - /* remove instrument */ - ev.type = SNDRV_SEQ_EVENT_INSTR_FREE; - ev.data.ext.len = sizeof(remove); - ev.data.ext.ptr = &remove; - - snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, &ev, - opl3->seq_client, 0, 0); - goto __again; - } - snd_leave_user(fs); - - kfree(put); - } - return err; + return sizeof(sbi); } /* ioctl */ diff -r 88d273f360dd drivers/opl3/opl3_seq.c --- a/drivers/opl3/opl3_seq.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_seq.c Fri Oct 26 16:09:56 2007 +0200 @@ -152,15 +152,7 @@ static int snd_opl3_synth_event_input(st { struct snd_opl3 *opl3 = private_data; - if (ev->type >= SNDRV_SEQ_EVENT_INSTR_BEGIN && - ev->type <= SNDRV_SEQ_EVENT_INSTR_CHANGE) { - if (direct) { - snd_seq_instr_event(&opl3->fm_ops, opl3->ilist, ev, - opl3->seq_client, atomic, hop); - } - } else { - snd_midi_process_event(&opl3_ops, ev, opl3->chset); - } + snd_midi_process_event(&opl3_ops, ev, opl3->chset); return 0; } @@ -249,16 +241,6 @@ static int snd_opl3_seq_new_device(struc return err; } - /* initialize instrument list */ - opl3->ilist = snd_seq_instr_list_new(); - if (opl3->ilist == NULL) { - snd_seq_delete_kernel_client(client); - opl3->seq_client = -1; - return -ENOMEM; - } - opl3->ilist->flags = SNDRV_SEQ_INSTR_FLG_DIRECT; - snd_seq_fm_init(&opl3->fm_ops, NULL); - /* setup system timer */ init_timer(&opl3->tlist); opl3->tlist.function = snd_opl3_timer_func; @@ -287,8 +269,6 @@ static int snd_opl3_seq_delete_device(st snd_seq_delete_kernel_client(opl3->seq_client); opl3->seq_client = -1; } - if (opl3->ilist) - snd_seq_instr_list_free(&opl3->ilist); return 0; } diff -r 88d273f360dd drivers/opl3/opl3_synth.c --- a/drivers/opl3/opl3_synth.c Fri Oct 26 15:48:22 2007 +0200 +++ b/drivers/opl3/opl3_synth.c Fri Oct 26 16:09:56 2007 +0200 @@ -186,6 +186,170 @@ int snd_opl3_release(struct snd_hwdep * mutex_unlock(&opl3->access_mutex); return 0; +} + +/* + * write the device - load patches + */ +long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, + loff_t *offset) +{ + struct snd_opl3 *opl3 = hw->private_data; + long result = 0; + int err = 0; + struct sbi_patch inst; + + while (count >= sizeof(inst)) { + unsigned char type; + if (copy_from_user(&inst, buf, sizeof(inst))) + return -EFAULT; + if (!memcmp(inst.key, FM_KEY_SBI, 4) || + !memcmp(inst.key, FM_KEY_2OP, 4)) + type = FM_PATCH_OPL2; + else if (!memcmp(inst.key, FM_KEY_4OP, 4)) + type = FM_PATCH_OPL3; + else /* invalid type */ + break; + err = snd_opl3_load_patch(opl3, inst.prog, inst.bank, type, + inst.name, inst.extension, + inst.data); + if (err < 0) + break; + result += sizeof(inst); + count -= sizeof(inst); + } + return result > 0 ? result : err; +} + + +/* + * Patch management + */ + +/* offsets for SBI params */ +#define AM_VIB 0 +#define KSL_LEVEL 2 +#define ATTACK_DECAY 4 +#define SUSTAIN_RELEASE 6 +#define WAVE_SELECT 8 + +/* offset for SBI instrument */ +#define CONNECTION 10 +#define OFFSET_4OP 11 + +/* + * load a patch, obviously. + * + * loaded on the given program and bank numbers with the given type + * (FM_PATCH_OPLx). + * data is the pointer of SBI record _without_ header (key and name). + * name is the name string of the patch. + * ext is the extension data of 7 bytes long (stored in name of SBI + * data up to offset 25), or NULL to skip. + * return 0 if successful or a negative error code. + */ +int snd_opl3_load_patch(struct snd_opl3 *opl3, + int prog, int bank, int type, + const char *name, + const unsigned char *ext, + const unsigned char *data) +{ + struct fm_patch *patch; + int i; + + patch = snd_opl3_find_patch(opl3, prog, bank, 1); + if (!patch) + return -ENOMEM; + + patch->type = type; + + for (i = 0; i < 2; i++) { + patch->inst.op[i].am_vib = data[AM_VIB + i]; + patch->inst.op[i].ksl_level = data[KSL_LEVEL + i]; + patch->inst.op[i].attack_decay = data[ATTACK_DECAY + i]; + patch->inst.op[i].sustain_release = data[SUSTAIN_RELEASE + i]; + patch->inst.op[i].wave_select = data[WAVE_SELECT + i]; + } + patch->inst.feedback_connection[0] = data[CONNECTION]; + + if (type == FM_PATCH_OPL3) { + for (i = 0; i < 2; i++) { + patch->inst.op[i+2].am_vib = + data[OFFSET_4OP + AM_VIB + i]; + patch->inst.op[i+2].ksl_level = + data[OFFSET_4OP + KSL_LEVEL + i]; + patch->inst.op[i+2].attack_decay = + data[OFFSET_4OP + ATTACK_DECAY + i]; + patch->inst.op[i+2].sustain_release = + data[OFFSET_4OP + SUSTAIN_RELEASE + i]; + patch->inst.op[i+2].wave_select = + data[OFFSET_4OP + WAVE_SELECT + i]; + } + patch->inst.feedback_connection[1] = + data[OFFSET_4OP + CONNECTION]; + } + + if (ext) { + patch->inst.echo_delay = ext[0]; + patch->inst.echo_atten = ext[1]; + patch->inst.chorus_spread = ext[2]; + patch->inst.trnsps = ext[3]; + patch->inst.fix_dur = ext[4]; + patch->inst.modes = ext[5]; + patch->inst.fix_key = ext[6]; + } + + if (name) + strlcpy(patch->name, name, sizeof(patch->name)); + + return 0; +} +EXPORT_SYMBOL(snd_opl3_load_patch); + +/* + * find a patch with the given program and bank numbers, returns its pointer + * if no matching patch is found and create_patch is set, it creates a + * new patch object. + */ +struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, + int create_patch) +{ + /* pretty dumb hash key */ + unsigned int key = (prog + bank) % OPL3_PATCH_HASH_SIZE; + struct fm_patch *patch; + + for (patch = opl3->patch_table[key]; patch; patch = patch->next) { + if (patch->prog == prog && patch->bank == bank) + return patch; + } + if (!create_patch) + return NULL; + + patch = kzalloc(sizeof(*patch), GFP_KERNEL); + if (!patch) + return NULL; + patch->prog = prog; + patch->bank = bank; + patch->next = opl3->patch_table[key]; + opl3->patch_table[key] = patch; + return patch; +} +EXPORT_SYMBOL(snd_opl3_find_patch); + +/* + * Clear all patches of the given OPL3 instance + */ +void snd_opl3_clear_patches(struct snd_opl3 *opl3) +{ + int i; + for (i = 0; i < OPL3_PATCH_HASH_SIZE; i++) { + struct fm_patch *patch, *next; + for (patch = opl3->patch_table[i]; patch; patch = next) { + next = patch->next; + kfree(patch); + } + } + memset(opl3->patch_table, 0, sizeof(opl3->patch_table)); } /* ------------------------------ */ diff -r 88d273f360dd include/asound_fm.h --- a/include/asound_fm.h Fri Oct 26 15:48:22 2007 +0200 +++ b/include/asound_fm.h Fri Oct 26 16:09:56 2007 +0200 @@ -112,4 +112,21 @@ struct snd_dm_fm_params { #define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24 #define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25 +/* + * Patch Record - fixed size for write + */ + +#define FM_KEY_SBI "SBI\032" +#define FM_KEY_2OP "2OP\032" +#define FM_KEY_4OP "4OP\032" + +struct sbi_patch { + unsigned char prog; + unsigned char bank; + char key[4]; + char name[25]; + char extension[7]; + unsigned char data[32]; +}; + #endif /* __SOUND_ASOUND_FM_H */ diff -r 88d273f360dd include/opl3.h --- a/include/opl3.h Fri Oct 26 15:48:22 2007 +0200 +++ b/include/opl3.h Fri Oct 26 16:09:56 2007 +0200 @@ -63,7 +63,7 @@ #include "seq_oss_legacy.h" #endif #include "seq_device.h" -#include "ainstr_fm.h" +#include "asound_fm.h" /* * Register numbers for the global registers @@ -240,6 +240,47 @@ struct snd_opl3; struct snd_opl3; /* + * Instrument record, aka "Patch" + */ + +/* FM operator */ +struct fm_operator { + unsigned char am_vib; + unsigned char ksl_level; + unsigned char attack_decay; + unsigned char sustain_release; + unsigned char wave_select; +} __attribute__((packed)); + +/* Instrument data */ +struct fm_instrument { + struct fm_operator op[4]; + unsigned char feedback_connection[2]; + unsigned char echo_delay; + unsigned char echo_atten; + unsigned char chorus_spread; + unsigned char trnsps; + unsigned char fix_dur; + unsigned char modes; + unsigned char fix_key; +}; + +/* type */ +#define FM_PATCH_OPL2 0x01 /* OPL2 2 operators FM instrument */ +#define FM_PATCH_OPL3 0x02 /* OPL3 4 operators FM instrument */ + +/* Instrument record */ +struct fm_patch { + unsigned char prog; + unsigned char bank; + unsigned char type; + struct fm_instrument inst; + char name[24]; + struct fm_patch *next; +}; + + +/* * A structure to keep track of each hardware voice */ struct snd_opl3_voice { @@ -297,8 +338,8 @@ struct snd_opl3 { struct snd_midi_channel_set * oss_chset; #endif - struct snd_seq_kinstr_ops fm_ops; - struct snd_seq_kinstr_list *ilist; +#define OPL3_PATCH_HASH_SIZE 32 + struct fm_patch *patch_table[OPL3_PATCH_HASH_SIZE]; struct snd_opl3_voice voices[MAX_OPL3_VOICES]; /* Voices (OPL3 'channel') */ int use_time; /* allocation counter */ @@ -333,8 +374,19 @@ int snd_opl3_open(struct snd_hwdep * hw, int snd_opl3_open(struct snd_hwdep * hw, struct file *file); int snd_opl3_ioctl(struct snd_hwdep * hw, struct file *file, unsigned int cmd, unsigned long arg); +long snd_opl3_write(struct snd_hwdep *hw, const char __user *buf, long count, + loff_t *offset); int snd_opl3_release(struct snd_hwdep * hw, struct file *file); void snd_opl3_reset(struct snd_opl3 * opl3); +int snd_opl3_load_patch(struct snd_opl3 *opl3, + int prog, int bank, int type, + const char *name, + const unsigned char *ext, + const unsigned char *data); +struct fm_patch *snd_opl3_find_patch(struct snd_opl3 *opl3, int prog, int bank, + int create_patch); +void snd_opl3_clear_patches(struct snd_opl3 *opl3); + #endif /* __SOUND_OPL3_H */

3 11

[alsa-devel] Sample format coversion bug?
by Alexander E. Patrakov 31 Oct '07

31 Oct '07

Hello, I am writing an application that uses ALSA, and I don't want to write code to support all possible sample formats. On IRC, I asked whether it is OK to support only S24 format internally and rely on the "plug" to do the required conversion. I got a positive answer, but, when testing my application against my SAA7134-based tuner as a capture device, found that it doesn't work. The problem is also reproducible with "arecord". Given that plughw:1 refers to a SAA7134-based tuner (which obviously doesn't support S24_LE natively), running the following command arecord -vvv -f S24_LE -c 2 -r 32000 -D plughw:1 /dev/null produces the following output: ALSA lib pcm_mmap.c:369:(snd_pcm_mmap) mmap failed: Invalid argument arecord: set_params:961: Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S24_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: 64 CHANNELS: 2 RATE: 32000 PERIOD_TIME: 125000 PERIOD_SIZE: 4000 PERIOD_BYTES: 32000 PERIODS: 4 BUFFER_TIME: 500000 BUFFER_SIZE: 16000 BUFFER_BYTES: 128000 TICK_TIME: 4000 This is on Debian, the kernel is 2.6.22-2-amd64, libasound2 is 1.0.14a-2. Same for S32_LE. -- Alexander E. Patrakov

3 8

[alsa-devel] [PATCH] Free after memory allocation failure in emu10k1
by Roel Kluin 31 Oct '07

31 Oct '07

I can confirm that it builds. -- Free after memory allocation failure Signed-off-by: Roel Kluin <12o3l(a)tiscali.nl> --- diff --git a/sound/pci/emu10k1/emu10k1_main.c b/sound/pci/emu10k1/emu10k1_main.c index 97c41d7..88f8fb3 100644 --- a/sound/pci/emu10k1/emu10k1_main.c +++ b/sound/pci/emu10k1/emu10k1_main.c @@ -1894,25 +1894,31 @@ static unsigned char saved_regs_audigy[] = { static int __devinit alloc_pm_buffer(struct snd_emu10k1 *emu) { int size; size = ARRAY_SIZE(saved_regs); if (emu->audigy) size += ARRAY_SIZE(saved_regs_audigy); emu->saved_ptr = vmalloc(4 * NUM_G * size); if (! emu->saved_ptr) return -ENOMEM; - if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) + + if (snd_emu10k1_efx_alloc_pm_buffer(emu) < 0) { + vfree(emu->saved_ptr); return -ENOMEM; + } + if (emu->card_capabilities->ca0151_chip && - snd_p16v_alloc_pm_buffer(emu) < 0) + snd_p16v_alloc_pm_buffer(emu) < 0) { + vfree(emu->saved_ptr); return -ENOMEM; + } return 0; } static void free_pm_buffer(struct snd_emu10k1 *emu) { vfree(emu->saved_ptr); snd_emu10k1_efx_free_pm_buffer(emu); if (emu->card_capabilities->ca0151_chip) snd_p16v_free_pm_buffer(emu); } diff --git a/sound/pci/emu10k1/emufx.c b/sound/pci/emu10k1/emufx.c index 9bf1cd5..a31ad02 100644 --- a/sound/pci/emu10k1/emufx.c +++ b/sound/pci/emu10k1/emufx.c @@ -2634,30 +2634,43 @@ int __devinit snd_emu10k1_fx8010_new(struct snd_emu10k1 *emu, int device, struct #ifdef CONFIG_PM int __devinit snd_emu10k1_efx_alloc_pm_buffer(struct snd_emu10k1 *emu) { int len; len = emu->audigy ? 0x200 : 0x100; emu->saved_gpr = kmalloc(len * 4, GFP_KERNEL); if (! emu->saved_gpr) return -ENOMEM; + len = emu->audigy ? 0x100 : 0xa0; emu->tram_val_saved = kmalloc(len * 4, GFP_KERNEL); + if (! emu->tram_val_saved) + goto free_gpr; + emu->tram_addr_saved = kmalloc(len * 4, GFP_KERNEL); - if (! emu->tram_val_saved || ! emu->tram_addr_saved) - return -ENOMEM; + if (! emu->tram_addr_saved) + goto free_tram_val; + len = emu->audigy ? 2 * 1024 : 2 * 512; emu->saved_icode = vmalloc(len * 4); if (! emu->saved_icode) - return -ENOMEM; + goto free_tram_addr; + return 0; +free_gpr: + kfree(emu->saved_gpr); +free_tram_val: + kfree(emu->tram_val_saved); +free_tram_addr: + kfree(emu->tram_addr_saved); + return -ENOMEM; } void snd_emu10k1_efx_free_pm_buffer(struct snd_emu10k1 *emu) { kfree(emu->saved_gpr); kfree(emu->tram_val_saved); kfree(emu->tram_addr_saved); vfree(emu->saved_icode); }

2 1

[alsa-devel] how to handle bt848 audio driver
by Trent Piepho 30 Oct '07

30 Oct '07

I've made some more patches for the ALSA bt87x audio driver to support more features of the Osprey 440 card. In order to support certain features, the ALSA driver needs access to the bt878's GPIO lines and I2C bus. These are controlled by PCI function 0, which is driven by the V4L driver. The bt878 DVB driver (which drives PCI function 1 just like the ALSA driver) needs to do the same thing and there is already some code to handle it. So, the ALSA driver (snd-bt87x) needs to use functions from the V4L driver (bttv). As it is now, the two drivers are completely separate and don't share any code and can be loaded individually. The DVB driver in the same tree as bttv, so this isn't a problem. But the snd-bt87x is in another tree, so it's somewhat harder, and I'm not sure what to do about it. The driver will need some header files from the bttv directory. I guess this can be handled by adding a -I flag in the Makefile, though that seems a little ugly. The bigger problem is that changes to snd-bt87x will depend on changes to bttv. How can the ALSA build system handle this? For instance, my patch to add a digital volume control to snd-bt87x requires a new function to be exported from bttv. How will ALSA build snd-bt87x if the kernel source doesn't yet have this function?

3 3

[alsa-devel] External Sound Card
by John Dey 30 Oct '07

30 Oct '07

I am looking for a external sound card that has spdif (7.1) functioning. The driver Soundblaster nx usb seems like it is under active development. The alsa documentation page says that spdif is not working. Is the documentation page out of date or is someone working on that capability? If spdif capability is not in the near future, can someone suggest a sound device (spdif capable) that I could look at. Thanks. John

2 3

Re: [alsa-devel] sysfs: WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() - with ALSA
by Kamalesh Babulal 30 Oct '07

30 Oct '07

On Fri, Oct 26, 2007 at 12:44:38PM +0530, Amit Shah wrote: > On 2.6.24-rc1, I get this for all my eth interfaces: > > [ 150.389821] sysfs: duplicate filename 'eth1_rename' can not be created > [ 150.389824] WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() > [ 150.389826] > [ 150.389826] Call Trace: > [ 150.389840] [<ffffffff802ee9cf>] sysfs_add_one+0xaf/0xf0 > [ 150.389844] [<ffffffff802ef9fe>] sysfs_create_link+0x9e/0x150 > [ 150.389849] [<ffffffff8038a418>] device_rename+0x1d8/0x230 > [ 150.389854] [<ffffffff803c2626>] dev_change_name+0xb6/0x230 > [ 150.389857] [<ffffffff803c2d5b>] dev_ioctl+0x2bb/0x5a0 > [ 150.389862] [<ffffffff80437436>] _spin_unlock_irqrestore+0x16/0x40 > [ 150.389864] [<ffffffff8043947a>] do_page_fault+0x24a/0x930 > [ 150.389869] [<ffffffff803b341d>] sock_ioctl+0x7d/0x250 > [ 150.389872] [<ffffffff802aafef>] do_ioctl+0x2f/0xa0 > [ 150.389875] [<ffffffff802ab0d4>] vfs_ioctl+0x74/0x2d0 > [ 150.389878] [<ffffffff802ab3c1>] sys_ioctl+0x91/0xb0 > [ 150.389881] [<ffffffff8020c32e>] system_call+0x7e/0x83 > [ 150.389884] > [ 150.389886] net eth1_rename: device_rename: sysfs_create_symlink failed (-17) > [ 151.169923] sysfs: duplicate filename 'eth2' can not be created > [ 151.169927] WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() > [ 151.169929] > [ 151.169930] Call Trace: > [ 151.169946] [<ffffffff802ee9cf>] sysfs_add_one+0xaf/0xf0 > [ 151.169949] [<ffffffff802ef9fe>] sysfs_create_link+0x9e/0x150 > [ 151.169955] [<ffffffff8038a418>] device_rename+0x1d8/0x230 > [ 151.169960] [<ffffffff803c2626>] dev_change_name+0xb6/0x230 > [ 151.169963] [<ffffffff803c2d5b>] dev_ioctl+0x2bb/0x5a0 > [ 151.169967] [<ffffffff80437436>] _spin_unlock_irqrestore+0x16/0x40 > [ 151.169970] [<ffffffff8043947a>] do_page_fault+0x24a/0x930 > [ 151.169974] [<ffffffff803b341d>] sock_ioctl+0x7d/0x250 > [ 151.169978] [<ffffffff802aafef>] do_ioctl+0x2f/0xa0 > [ 151.169980] [<ffffffff802ab0d4>] vfs_ioctl+0x74/0x2d0 > [ 151.169983] [<ffffffff802ab3c1>] sys_ioctl+0x91/0xb0 > [ 151.169987] [<ffffffff8020c32e>] system_call+0x7e/0x83 > [ 151.169990] > [ 151.169992] net eth2: device_rename: sysfs_create_symlink failed (-17) > [ 151.176762] e1000: eth2: e1000_watchdog: NIC Link is Up 100 Mbps > Full Duplex, Flow Control: RX/TX > [ 151.209396] sysfs: duplicate filename 'eth0' can not be created > Hi, I get similar warning, while loading the alsa driver [ 191.933548] Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Oct 23 06:09:18 2007 UTC). [ 191.996323] ALSA sound/core/seq/oss/seq_oss.c:232: can't register device seq [ 192.046004] sysfs: duplicate filename 'audio' can not be created [ 192.086048] WARNING: at fs/sysfs/dir.c:424 sysfs_add_one() [ 192.122882] [ 192.122883] Call Trace: [ 192.154111] [<ffffffff802fbe74>] sysfs_add_one+0x5c/0xc9 [ 192.190144] [<ffffffff802fc498>] create_dir+0x58/0x93 [ 192.224571] [<ffffffff802fc50b>] sysfs_create_dir+0x38/0x4f [ 192.262068] [<ffffffff813a622b>] _spin_unlock+0x26/0x2a [ 192.297448] [<ffffffff8048eb92>] kobject_add+0x13e/0x1f9 [ 192.333292] [<ffffffff8065c746>] device_add+0xac/0x610 [ 192.367959] [<ffffffff8065ccc3>] device_register+0x19/0x1d [ 192.404621] [<ffffffff8065cda6>] device_create+0xdf/0x110 [ 192.440760] [<ffffffff8049301c>] sprintf+0x68/0x6a [ 192.473288] [<ffffffff80ff9b4a>] sound_insert_unit+0xcf/0x172 [ 192.511443] [<ffffffff80ff9ba4>] sound_insert_unit+0x129/0x172 [ 192.549996] [<ffffffff80ff9deb>] register_sound_special_device+0x14a/0x159 [ 192.594734] [<ffffffff8101ed06>] snd_register_oss_device+0x11d/0x18f [ 192.636281] [<ffffffff81034470>] register_oss_dsp+0x54/0x8a [ 192.673031] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 192.709275] [<ffffffff813a50f1>] mutex_lock_nested+0x2a8/0x2c3 [ 192.747533] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 192.783587] [<ffffffff813a4e32>] __mutex_unlock_slowpath+0x12e/0x13a [ 192.824764] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 192.862739] [<ffffffff810378d4>] snd_pcm_oss_register_minor+0x35/0x16b [ 192.904816] [<ffffffff813a4e47>] mutex_unlock+0x9/0xb [ 192.938016] [<ffffffff810208de>] snd_timer_dev_register+0x125/0x12c [ 192.978469] [<ffffffff8101e27c>] snd_device_register+0x73/0xe0 [ 193.016307] [<ffffffff8102e03d>] snd_pcm_timer_init+0xba/0xf5 [ 193.053660] [<ffffffff810245bd>] snd_pcm_dev_register+0x1d7/0x211 [ 193.093096] [<ffffffff8101e1e1>] snd_device_register_all+0x5d/0x85 [ 193.133041] [<ffffffff810197ac>] snd_card_register+0x4e/0x2ea [ 193.170427] [<ffffffff8104f66d>] snd_dummy_probe+0x201/0x21a [ 193.207240] [<ffffffff806601b5>] platform_drv_probe+0x12/0x14 [ 193.244615] [<ffffffff8065ea16>] driver_probe_device+0x132/0x1d2 [ 193.283580] [<ffffffff8065eab6>] __device_attach+0x0/0xb [ 193.318382] [<ffffffff8065eabf>] __device_attach+0x9/0xb [ 193.353173] [<ffffffff8065df69>] bus_for_each_drv+0x47/0x7d [ 193.389534] [<ffffffff8065eb5a>] device_attach+0x75/0x8d [ 193.424319] [<ffffffff8065dd78>] bus_attach_device+0x2d/0x79 [ 193.461202] [<ffffffff8065ca2d>] device_add+0x393/0x610 [ 193.495910] [<ffffffff806605a5>] platform_device_add+0x133/0x16a [ 193.534870] [<ffffffff806607a8>] platform_device_register_simple+0x46/0x64 [ 193.579013] [<ffffffff8208d029>] alsa_card_dummy_init+0x38/0x7e [ 193.617437] [<ffffffff82028678>] kernel_init+0x176/0x2ed [ 193.652173] [<ffffffff813a5d27>] trace_hardirqs_on_thunk+0x35/0x3a [ 193.692147] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 193.730030] [<ffffffff8020cef8>] child_rip+0xa/0x12 [ 193.762145] [<ffffffff8020c60f>] restore_args+0x0/0x30 [ 193.795821] [<ffffffff8023fc7f>] release_console_sem+0x4d/0x1f5 [ 193.834171] [<ffffffff82028502>] kernel_init+0x0/0x2ed [ 193.867870] [<ffffffff8020ceee>] child_rip+0x0/0x12 [ 193.899987] [ 193.911270] kobject_add failed for audio with -EEXIST, don't try to register things with the same name in the same directory. [ 193.983989] [ 193.983990] Call Trace: [ 194.012668] [<ffffffff8048e440>] kobject_put+0x19/0x1b [ 194.046571] [<ffffffff8048ec17>] kobject_add+0x1c3/0x1f9 [ 194.081445] [<ffffffff8065c746>] device_add+0xac/0x610 [ 194.115311] [<ffffffff8065ccc3>] device_register+0x19/0x1d [ 194.151267] [<ffffffff8065cda6>] device_create+0xdf/0x110 [ 194.186663] [<ffffffff8049301c>] sprintf+0x68/0x6a [ 194.218463] [<ffffffff80ff9b4a>] sound_insert_unit+0xcf/0x172 [ 194.255953] [<ffffffff80ff9ba4>] sound_insert_unit+0x129/0x172 [ 194.293932] [<ffffffff80ff9deb>] register_sound_special_device+0x14a/0x159 [ 194.338208] [<ffffffff8101ed06>] snd_register_oss_device+0x11d/0x18f [ 194.379363] [<ffffffff81034470>] register_oss_dsp+0x54/0x8a [ 194.415853] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 194.451820] [<ffffffff813a50f1>] mutex_lock_nested+0x2a8/0x2c3 [ 194.489889] [<ffffffff802604e6>] mark_held_locks+0x4a/0x6a [ 194.525880] [<ffffffff813a4e32>] __mutex_unlock_slowpath+0x12e/0x13a [ 194.567098] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 194.605551] [<ffffffff810378d4>] snd_pcm_oss_register_minor+0x35/0x16b [ 194.647848] [<ffffffff813a4e47>] mutex_unlock+0x9/0xb [ 194.681271] [<ffffffff810208de>] snd_timer_dev_register+0x125/0x12c [ 194.721968] [<ffffffff8101e27c>] snd_device_register+0x73/0xe0 [ 194.760041] [<ffffffff8102e03d>] snd_pcm_timer_init+0xba/0xf5 [ 194.797490] [<ffffffff810245bd>] snd_pcm_dev_register+0x1d7/0x211 [ 194.836921] [<ffffffff8101e1e1>] snd_device_register_all+0x5d/0x85 [ 194.876879] [<ffffffff810197ac>] snd_card_register+0x4e/0x2ea [ 194.914262] [<ffffffff8104f66d>] snd_dummy_probe+0x201/0x21a [ 194.951096] [<ffffffff806601b5>] platform_drv_probe+0x12/0x14 [ 194.988480] [<ffffffff8065ea16>] driver_probe_device+0x132/0x1d2 [ 195.027449] [<ffffffff8065eab6>] __device_attach+0x0/0xb [ 195.062267] [<ffffffff8065eabf>] __device_attach+0x9/0xb [ 195.097053] [<ffffffff8065df69>] bus_for_each_drv+0x47/0x7d [ 195.133408] [<ffffffff8065eb5a>] device_attach+0x75/0x8d [ 195.168203] [<ffffffff8065dd78>] bus_attach_device+0x2d/0x79 [ 195.205133] [<ffffffff8065ca2d>] device_add+0x393/0x610 [ 195.239455] [<ffffffff806605a5>] platform_device_add+0x133/0x16a [ 195.278432] [<ffffffff806607a8>] platform_device_register_simple+0x46/0x64 [ 195.322586] [<ffffffff8208d029>] alsa_card_dummy_init+0x38/0x7e [ 195.361032] [<ffffffff82028678>] kernel_init+0x176/0x2ed [ 195.395798] [<ffffffff813a5d27>] trace_hardirqs_on_thunk+0x35/0x3a [ 195.435764] [<ffffffff802606d0>] trace_hardirqs_on+0x11c/0x147 [ 195.473657] [<ffffffff8020cef8>] child_rip+0xa/0x12 [ 195.505771] [<ffffffff8020c60f>] restore_args+0x0/0x30 [ 195.539459] [<ffffffff8023fc7f>] release_console_sem+0x4d/0x1f5 [ 195.577825] [<ffffffff82028502>] kernel_init+0x0/0x2ed [ 195.611533] [<ffffffff8020ceee>] child_rip+0x0/0x12 [ 195.643690] [ 195.659326] ALSA sound/core/rawmidi.c:1542: unable to register OSS rawmidi device 1:0 [ 195.712879] no UART detected at 0x1 [ 195.739880] Motu MidiTimePiece on parallel port irq: 7 ioport: 0x378 [ 195.804370] usbcore: registered new interface driver snd-usb-audio [ 195.844370] usbcore: registered new interface driver snd-usb-usx2y [ 195.884286] usbcore: registered new interface driver snd-usb-caiaq [ 195.924293] ASoC version 0.13.1 [ 195.946107] ALSA device list: [ 195.966550] #0: Dummy 1 [ 195.984913] #1: Virtual MIDI Card 1 [ 196.009489] #2: MTPAV on parallel port at 0x378 -- Thanks & Regards, Kamalesh Babulal, Linux Technology Center, IBM, ISTL.

5 9

[alsa-devel] [PATCH] SB Live24-External better handling
by Timofei V. Bondarenko 30 Oct '07

30 Oct '07

Hi. This patch improves support for 'SB Live 24-bit Extarnal' USB card. 1) This card can go into muted state when a headphones connected or disconnected. So notify mixer about changes in headphone jack. 2) Add LED controls and procfs support just as in similar Audigy 2 NX card. 3) Rename 'PCM Capture' conrol to 'Mic Capture' to reflect reality: the card may adjust microphone input level only. Regards. Tim.

2 3

[alsa-devel] Removal of sequencer instrument layer
by Takashi Iwai 30 Oct '07

30 Oct '07

Hi, JFYI, together with OPL3 hwdep loader, $SUBJECT is done as I mentioned earlier. This resulted in removal of 5600 LOC in alsa-kernel tree. Yeah, a good number. Takashi

1 0

[alsa-devel] cmipci CM_FLINKOFF disables multinchannel playback
by Timofei V. Bondarenko 30 Oct '07

30 Oct '07

Hi. I found the latest cmipci driver cannot playback in multichannel mode anymore. My hardware is 'CMI8738 model 55' attempting to play 2 or 4 channels through the 2nd DAC i get no sound at all. 6 ch produces sound only at center/bass while front and rear still mute. I found the root of problem in redefinition of some bits of the CM_REG_MISC_CTRL (0x18) register. Was: CM_FLINKON=0x40, CM_FLINKOFF=0x20 Now it is: CM_FLINKON=0x80 CM_FLINKOFF=0x40 Note the CM_FLINKOFF does collide with the CM_MUTECH1=0x40 bit. This bit disables playback on 2nd DAC. The CM_FLINKOFF/CM_FLINKON bits used in the 'Modem' mixer control only. During startup the mixer turns the modem off somewhere and this disables 2nd DAC as well. Complete removal of this 'Modem' control fixes the problem. Is it an acceptable solution? I guess the snd_cmipci_control_switches could be removed too. Regards. Tim. --- alsa-driver-hg20071027/alsa-kernel/pci/cmipci.c 2007-09-18 04:00:10.000000000 +0400 +++ alsa-driver/alsa-kernel/pci/cmipci.c 2007-10-29 17:09:31.000000000 +0300 @@ -2389,7 +2389,6 @@ DEFINE_BIT_SWITCH_ARG(fourch, CM_REG_MIS // DEFINE_BIT_SWITCH_ARG(line_rear, CM_REG_MIXER1, CM_REAR2LIN, 1, 0); // DEFINE_BIT_SWITCH_ARG(line_bass, CM_REG_LEGACY_CTRL, CM_CENTR2LIN|CM_BASE2LIN, 0, 0); // DEFINE_BIT_SWITCH_ARG(joystick, CM_REG_FUNCTRL1, CM_JYSTK_EN, 0, 0); /* now module option */ -DEFINE_SWITCH_ARG(modem, CM_REG_MISC_CTRL, CM_FLINKON|CM_FLINKOFF, CM_FLINKON, 0, 0); #define DEFINE_SWITCH(sname, stype, sarg) \ { .name = sname, \ @@ -2597,7 +2596,7 @@ static struct snd_kcontrol_new snd_cmipc /* card control switches */ static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ - DEFINE_CARD_SWITCH("Modem", modem), +// DEFINE_CARD_SWITCH("Modem", modem), };

2 3

[alsa-devel] PATCH: ln2440sbc ac97 support
by Ivan Kuten 29 Oct '07

29 Oct '07

Hello, This patch adds ac97 support for ln2440sbc board from LittleChips. This board is based on s3c2440 SoC + AC97 Realtek ALC650 codec. Existing s3c2443 implementation is slightly modified because s3c2440 and s3c2443 have different AC97 interrupts. Regards, Ivan Signed-off-by: Ivan Kuten <ivan.kuten(a)promwad.com> diff -urN linux-2.6.23/sound/soc/s3c24xx/Kconfig linux-2.6.23-2440/sound/soc/s3c24xx/Kconfig --- linux-2.6.23/sound/soc/s3c24xx/Kconfig 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/Kconfig 2007-10-28 01:18:23.000000000 +0200 @@ -34,4 +34,12 @@ Say Y if you want to add support for SoC audio on smdk2443 with the WM9710. +config SND_S3C24XX_SOC_LN2440SBC_ALC650 + tristate "SoC AC97 Audio support for LN2440SBC - ALC650" + depends on SND_S3C24XX_SOC + select SND_S3C2443_SOC_AC97 + select SND_SOC_AC97_CODEC + help + Say Y if you want to add support for SoC audio on ln2440sbc + with the ALC650. diff -urN linux-2.6.23/sound/soc/s3c24xx/Makefile linux-2.6.23-2440/sound/soc/s3c24xx/Makefile --- linux-2.6.23/sound/soc/s3c24xx/Makefile 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/Makefile 2007-10-25 01:28:18.000000000 +0200 @@ -10,6 +10,8 @@ # S3C24XX Machine Support snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o snd-soc-smdk2443-wm9710-objs := smdk2443_wm9710.o +snd-soc-ln2440sbc-alc650-objs := ln2440sbc_alc650.o obj-$(CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753) += snd-soc-neo1973-wm8753.o obj-$(CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710) += snd-soc-smdk2443-wm9710.o +obj-$(CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650) += snd-soc-ln2440sbc-alc650.o diff -urN linux-2.6.23/sound/soc/s3c24xx/ln2440sbc_alc650.c linux-2.6.23-2440/sound/soc/s3c24xx/ln2440sbc_alc650.c --- linux-2.6.23/sound/soc/s3c24xx/ln2440sbc_alc650.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.23-2440/sound/soc/s3c24xx/ln2440sbc_alc650.c 2007-10-28 00:27:41.000000000 +0200 @@ -0,0 +1,86 @@ +/* + * SoC audio for ln2440sbc + * + * Copyright 2007 KonekTel, a.s. + * Author: Ivan Kuten + * ivan.kuten(a)promwad.com + * + * Heavily based on smdk2443_wm9710.c + * Copyright 2007 Wolfson Microelectronics PLC. + * Author: Graeme Gregory + * graeme.gregory(a)wolfsonmicro.com or linux(a)wolfsonmicro.com + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + */ + +#include <linux/module.h> +#include <linux/device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include "../codecs/ac97.h" +#include "s3c24xx-pcm.h" +#include "s3c24xx-ac97.h" + +static struct snd_soc_machine ln2440sbc; + +static struct snd_soc_dai_link ln2440sbc_dai[] = { +{ + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &s3c2443_ac97_dai[0], + .codec_dai = &ac97_dai, +}, +}; + +static struct snd_soc_machine ln2440sbc = { + .name = "LN2440SBC", + .dai_link = ln2440sbc_dai, + .num_links = ARRAY_SIZE(ln2440sbc_dai), +}; + +static struct snd_soc_device ln2440sbc_snd_ac97_devdata = { + .machine = &ln2440sbc, + .platform = &s3c24xx_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *ln2440sbc_snd_ac97_device; + +static int __init ln2440sbc_init(void) +{ + int ret; + + ln2440sbc_snd_ac97_device = platform_device_alloc("soc-audio", -1); + if (!ln2440sbc_snd_ac97_device) + return -ENOMEM; + + platform_set_drvdata(ln2440sbc_snd_ac97_device, + &ln2440sbc_snd_ac97_devdata); + ln2440sbc_snd_ac97_devdata.dev = &ln2440sbc_snd_ac97_device->dev; + ret = platform_device_add(ln2440sbc_snd_ac97_device); + + if (ret) + platform_device_put(ln2440sbc_snd_ac97_device); + + return ret; +} + +static void __exit ln2440sbc_exit(void) +{ + platform_device_unregister(ln2440sbc_snd_ac97_device); +} + +module_init(ln2440sbc_init); +module_exit(ln2440sbc_exit); + +/* Module information */ +MODULE_AUTHOR("Ivan Kuten"); +MODULE_DESCRIPTION("ALSA SoC ALC650 LN2440SBC"); +MODULE_LICENSE("GPL"); diff -urN linux-2.6.23/sound/soc/s3c24xx/s3c2443-ac97.c linux-2.6.23-2440/sound/soc/s3c24xx/s3c2443-ac97.c --- linux-2.6.23/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-28 00:55:57.000000000 +0200 @@ -32,7 +32,7 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-ac97.h> +#include <asm/plat-s3c/regs-ac97.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> @@ -253,7 +253,7 @@ ac_glbctrl |= S3C_AC97_GLBCTRL_TRANSFERDATAENABLE; writel(ac_glbctrl, s3c24xx_ac97.regs + S3C_AC97_GLBCTRL); - ret = request_irq(IRQ_S3C2443_AC97, s3c2443_ac97_irq, + ret = request_irq(IRQ_S3C244x_AC97, s3c2443_ac97_irq, IRQF_DISABLED, "AC97", NULL); if (ret < 0) { printk(KERN_ERR "s3c24xx-ac97: interrupt request failed.\n"); @@ -266,7 +266,7 @@ static void s3c2443_ac97_remove(struct platform_device *pdev) { - free_irq(IRQ_S3C2443_AC97, NULL); + free_irq(IRQ_S3C244x_AC97, NULL); clk_disable(s3c24xx_ac97.ac97_clk); clk_put(s3c24xx_ac97.ac97_clk); iounmap(s3c24xx_ac97.regs); diff -urN linux-2.6.23/sound/soc/s3c24xx/s3c24xx-ac97.h linux-2.6.23-2440/sound/soc/s3c24xx/s3c24xx-ac97.h --- linux-2.6.23/sound/soc/s3c24xx/s3c24xx-ac97.h 2007-10-09 22:31:38.000000000 +0200 +++ linux-2.6.23-2440/sound/soc/s3c24xx/s3c24xx-ac97.h 2007-10-28 00:32:04.000000000 +0200 @@ -20,6 +20,12 @@ #define AC_CMD_ADDR(x) (x << 16) #define AC_CMD_DATA(x) (x & 0xffff) +#ifdef CONFIG_CPU_S3C2440 +#define IRQ_S3C244x_AC97 IRQ_S3C2440_AC97 +#else +#define IRQ_S3C244x_AC97 IRQ_S3C2443_AC97 +#endif + extern struct snd_soc_cpu_dai s3c2443_ac97_dai[]; #endif /*S3C24XXAC97_H_*/

3 3

[alsa-devel] [PATCH] ac97_patch: compilation warning fix
by Krzysztof Helt 29 Oct '07

29 Oct '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch kills these two compilation warnings if power management is disabled: sound/pci/ac97/ac97_patch.h:86: warning: 'snd_ac97_restore_status' declared 'static' but never defined sound/pci/ac97/ac97_patch.h:87: warning: 'snd_ac97_restore_iec958' declared 'static' but never defined Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The patch was done against kernel 2.6.24-rc1-git4. diff -urp linux-ref/sound/pci/ac97/ac97_patch.h linux-new/sound/pci/ac97/ac97_patch.h --- linux-ref/sound/pci/ac97/ac97_patch.h 2007-10-28 11:11:20.000000000 +0100 +++ linux-new/sound/pci/ac97/ac97_patch.h 2007-10-28 12:46:34.000000000 +0100 @@ -83,8 +83,10 @@ static int snd_ac97_swap_ctl(struct snd_ const char *s2, const char *suffix); static void snd_ac97_rename_vol_ctl(struct snd_ac97 *ac97, const char *src, const char *dst); +#ifdef CONFIG_PM static void snd_ac97_restore_status(struct snd_ac97 *ac97); static void snd_ac97_restore_iec958(struct snd_ac97 *ac97); +#endif static int snd_ac97_info_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo); static int snd_ac97_get_enum_double(struct snd_kcontrol *kcontrol,

2 1

[alsa-devel] [PATCH] Dreamcast AICA sound - Get rid of annoying compiler warning
by Adrian McMenamin 29 Oct '07

29 Oct '07

This patch supresses an annoying compiler warning that the variable err may be used uninitialised. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 88dc840..8861d2f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -237,6 +237,7 @@ static int aica_dma_transfer(int channels, int buffer_size, struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; unsigned long flags; + err = 0; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels);

2 1

[alsa-devel] [PATCH] Fix for assertion in pulse plugin when calling hw_params on a prepared stream
by Sean McNamara 29 Oct '07

29 Oct '07

Hello, I have opened a bug and recently submitted a patch for it on bugtrack: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3470 All the details are there. But here is a 5-minute summary for you: --The Problem-- (1) All plugins implementing the PCM interface, that I've tested so far, allow calling snd_*_hw_params on a stream when snd_*_state returns SND_PCM_STATE_PREPARED. (2) Applications have used this fact to their advantage, and sometimes employ lazy initialization of the hw params (initialize it once, get to another path in code, oh no wait! we need to initialize them differently...) (3) Per the spec, calling snd_*_hw_params also makes the stream PREPARED, so calling snd_*_hw_params on a stream that has _either_ been explicitly prepared, _or_ had its hw_params previously set, will trigger this problem. (4) The ALSA<->PulseAudio plugin is an exception to (1) above; i.e., it triggers an assertion when exercising this functionality, as if to say, "I can't do that!" (5) Indeed, the PulseAudio client API forbids re-initialization of the parameters in question on a PulseAudio stream, after instantiation. --The Solution-- Only perform the assertion if snd_pcm_state(io->pcm) != SND_PCM_STATE_PREPARED. The assertion is still useful for cases where the stream is mysteriously non-NULL and the stream has not already been prepared; I can't think of a valid case where: (1) The state is not SND_PCM_STATE_PREPARED, and (2) The underlying PulseAudio stream (pcm->stream) is non-NULL, and (3) snd_pcm_hw_params would be expected to do anything meaningful or correct. For this reason, I kept the assertion in, rather than just removing it (which by itself "fixes" the problem with alsaplayer and my test case, which is submitted with the patch.) --Why It Works-- (1) This assertion might be a throwback from before ioplug was so smart; but apparently, ioplug will now call pulse_prepare, which reinitializes the underlying pulseaudio stream, after pulse_hw_params. (2) There is nothing in pulse_hw_params that actually requires any particular state out of pcm->stream. The underlying pulse stream is simply not used in this call. (3) ioplug catches the fact that the hw_params have been changed, and calls back to pulse_prepare to handle the underlying stream teardown and reinitialization (which pulse_prepare does correctly). --Impact-- (1) At least one open source application, alsaplayer, can now play through the ALSA<->PulseAudio plugin, where it could never do so before. (2) The ALSA<->PulseAudio plugin is now more tightly compatible with the other alsa-lib and alsa-plugin backends (dmix, hw, asym, oss, jack, ...) (3) Older versions of sox might have also suffered from this bug, according to information here: http://www.pulseaudio.org/ticket/120 (although the Ubuntu 7.10 version of sox does not exhibit this behavior). --Notes-- (1) I submitted a first-attempt patch that fixed the problem, but it was inefficient: it needlessly called pulse_prepare _prior_ to the main pulse_hw_params function body. This was a result of my lack of understanding of how ioplug was actually saving me from disaster in the background. Be sure to grab 2_pcm_pulse.c.patch rather than pcm_pulse.c.patch. (2) I tested both positive and negative cases; the ALSA<->PulseAudio plugin works with applications that hit the modified code, as well as applications that don't. I've tested both branches of the conditional I added. Thanks, Sean

2 1

[alsa-devel] snd-hdsp in 2.6.22 crashes badly
by ilya .d 29 Oct '07

29 Oct '07

Hello. i'm using RME Hammerfall HDSP9632.. it worked fine with 2.6.17 and 2.6.20 (-gentoo), i don't think gentoo team modifies alsa tree .. as soon as i upgraded to 2.6.22[-gentoo-r8], it booted fine, but when i started gnome-settigs-daemon, kernell generated OOPS, and i couldn't inload the module with modprobe. i disabled gnome-settings-daemon, and reboot. i managed to use hdspconf , hdspmixer + a couple of applications (puredata and snd editor), it was all good. as soon as i ran alsamixer (well , normaly you don't need it for snd-hdsp, but i wanted to configure my second soundcard, which is SB_live, but forgot to give it '-c 1') - so here the same crash heppends. i tried 2.6.20 back - and all absolutely fine with it. looking at hdsp.c in both kernels i spoted quite a few changes , however i don't know much about it ;[ so i attached the details below. -- cheers, ilya .d

2 2

[alsa-devel] ALSA dma ring buffer issue on 2.6.14
by Filipe Braz 27 Oct '07

27 Oct '07

Hi, I am trying to use ALSA with the OMAP2430 embedded platform, using kernel 2.6.14 for low latency voip applications. The problem seems to be that the driver ring buffer is not filled by the alsa kernel/core on time to be sent to the hw codec. So, If I use the following large buffer size / period-size it works fine: aplay --period-size=1024 --buffer-size=8192 -f S16_LE -c 2 -r 8000 pattern.pcm But if I use smaller buffer / period sizes such as the following it does not work: aplay --period-size=32 --buffer-size=128 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=32 --buffer-size=256 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=64 --buffer-size=512 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=128 --buffer-size=512 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=256 --buffer-size=1024 -f S16_LE -c 2 -r 8000 pattern.pcm aplay --period-size=512 --buffer-size=2048 -f S16_LE -c 2 -r 8000 pattern.pcm I am playing a special pattern sound file that I verify before each period is transfered to the codec via dma (on the driver). On smaller buffer sizes. the 1st period AFTER the ringbuffer rotates to the beginning, appears to not contain the pattern data (instead is filled with zeros). This does not happens with larger buffer / periods. On all cases the pattern file is large enough so that the dma ring buffer rotates several times until transfer is complete. I have tested this on the pc and it works just fine. Also if I use the OSS driver for the OMAP2430 instead of the ALSA it works fine. ALSA with OSS emulation works significantly better but there is still some distortion. Any ideas why this happens? Is this a known issue or is there a solution for this problem? Thanks in advance, Filipe

1 0

[alsa-devel] Confused about 'default' device
by Timur Tabi 26 Oct '07

26 Oct '07

I'm having a hard time understand when ALSA does sample rate and format conversion, and when it doesn't. When I issue the following command: mplayer -ao alsa filename.mov -v I get this output: Building audio filter chain for 48000Hz/2ch/s24le -> 0Hz/0ch/s8... [libaf] Adding filter dummy [dummy] Was reinitialized: 48000Hz/2ch/s24le [dummy] Was reinitialized: 48000Hz/2ch/s24le alsa-init: requested format: 48000 Hz, 2 channels, 11 alsa-init: using ALSA 1.0.8 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opend in blocking mode alsa-init: format s24le are not supported by hardware, trying default alsa-lib: pcm_plug.c:729:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' alsa-init: unable to set format: Invalid argument Sure enough, my device can't handle s24le. It can only handle s16be. I was under the impression that ALSA can do format and sample rate conversion if necessary, but I guess that's not always the case. I know ALSA has a bunch of plug-ins for that, but I guess they're not used by default. What do I need to do to tell ALSA to convert s24le to s16be?

4 14

[alsa-devel] New alsa lowlevel driver for the SE-200PCI and SE-90PCI
by sh_okada＠d4.dion.ne.jp 26 Oct '07

26 Oct '07

HI. I mail from Japan. I wrote a driver for the SE-200PCI and SE-90PCI what are salling in Japan. And I want to merge it to current source tree. How should I do? Thanks.

4 16

[alsa-devel] [PATCH] make sbiload to use hwdep interface
by Takashi Iwai 26 Oct '07

26 Oct '07

This patch makes sbiload using the hwdep interface instead of seq instr layer. This simplifies the logic a lot. Takashi diff -r 9b306261208b seq/sbiload/README --- a/seq/sbiload/README Mon Oct 22 15:16:35 2007 +0200 +++ b/seq/sbiload/README Fri Oct 26 16:10:50 2007 +0200 @@ -1,4 +1,4 @@ This is sbiload - an OPL2/3 FM instrumen -This is sbiload - an OPL2/3 FM instrument loader for ALSA sequencer +This is sbiload - an OPL2/3 FM instrument loader for ALSA hwdep Written by Uros Bizjak <uros(a)kss-loka.si> Web: http://www.kss-loka.si/~uros/sbiload.html @@ -14,62 +14,42 @@ By default it will install into /usr/loc By default it will install into /usr/local/bin, change INSTDIR in make.conf to change this. -Before use ----------- - -Please check that the correct sequencer modules are loaded for your -sound card. You need to load snd-synth-opl3 module. - Usage ----- - loadsbi [-p client:port] [-4] [-l] [-P path] [-v level] instfile drumfile + loadsbi [-D device] [-4] [-P path] [-v level] instfile drumfile - -p, --port=client:port - An ALSA client and port number to use + -D, --device=name - An ALSA hwdep name to use -4 --opl3 - Four operators (OPL3) file type - -l, --list - List possible output ports -P, --path=path - Specify the patch path -v, --verbose=level - Verbose level -Because you may have more than one sound card and each sound -card may have several MIDI connections, you have to tell sbiload -which one to use. +When you have multiple sound cards and/or there may be multiple OPL2/3 +synth devices, you have to tell sbiload which one to use, and pass its +hwdep name via -D option. Otherwise, sbiload will load the data onto +the first found device. -First find out what the possibilities are for your system: +To find out the hwdep name of the OPL devices, check +/proc/asound/hwdep file first. It has the list of hwdep devices +currently available on your system. -If you run: - - sbiload -l + % cat /proc/asound/hwdep + 00-00: Emux WaveTable + 00-01: OPL3 FM + 01-00: OPL3 FM -it will give you a list of the devices. On my system I get: - - Port Client name Port name - 64:0 0: MIDI Synth MIDI 0-0 - 65:0 OPL3 FM Synth OPL3 FM Synth port +In the example above, you have two OPL3 devices. To the first one +(00-01), which means the first card (00-) and the second device (01), +pass "-D hw:0,1". For the second OPL3 device (01-00), pass "-D +hw:1,0" instead. - -Only OPLx FM Synth port can be used with sbiload. In the example above -the first one is the external Midi port, and the second is the internal -four-operator FM synthesizer (OPL3). - -If you don't see anything listed then check that the correct ALSA -modules are loaded as in the section "Before use". Running sbiload --------------- -You specify the port to use with the -p option, or by setting the -environment variable ALSA_OUT_PORT. +Pass the instrument and drum file data as arguments. -For example to use port 0 on client 65 to load four-operator FM -instrument - - sbiload -p 65:0 --opl3 std.o3 drums.o3 - -or - - export ALSA_OUT_PORT=65:0 - sbiload --opl3 std.o3 drums.o3 + % sbiload std.o3 drums.o3 Acknowledgements diff -r 9b306261208b seq/sbiload/sbiload.c --- a/seq/sbiload/sbiload.c Mon Oct 22 15:16:35 2007 +0200 +++ b/seq/sbiload/sbiload.c Fri Oct 26 16:10:50 2007 +0200 @@ -1,5 +1,5 @@ /* - * ALSA sequencer SBI FM instrument loader + * ALSA hwdep SBI FM instrument loader * Copyright (c) 2000 Uros Bizjak <uros(a)kss-loka.si> * * This program is free software; you can redistribute it and/or modify @@ -16,11 +16,13 @@ * along with this program; if not, write to the Free Software * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * + * + * Oct. 2007 - Takashi Iwai <tiwai(a)suse.de> + * Changed to use hwdep instead of obsoleted seq-instr interface */ #include <errno.h> #include <getopt.h> -#include <alsa/sound/ainstr_fm.h> #include <stdio.h> #include <stdlib.h> #include <string.h> @@ -29,24 +31,12 @@ #include <fcntl.h> #include <errno.h> #include <alsa/asoundlib.h> +#include <alsa/sound/asound_fm.h> #include <sys/time.h> #include <unistd.h> -typedef struct sbi_header -{ - char key[4]; - char name[32]; -} -sbi_header_t; - -typedef struct sbi_inst -{ - sbi_header_t header; #define DATA_LEN_2OP 16 #define DATA_LEN_4OP 24 - char data[DATA_LEN_4OP]; -} -sbi_inst_t; /* offsets for SBI params */ #define AM_VIB 0 @@ -59,21 +49,20 @@ sbi_inst_t; #define CONNECTION 10 #define OFFSET_4OP 11 -/* offsets in sbi_header.name for SBI extensions */ -#define ECHO_DELAY 25 -#define ECHO_ATTEN 26 -#define CHORUS_SPREAD 27 -#define TRNSPS 28 -#define FIX_DUR 29 -#define MODES 30 -#define FIX_KEY 31 +/* offsets for SBI extensions */ +#define ECHO_DELAY 0 +#define ECHO_ATTEN 1 +#define CHORUS_SPREAD 2 +#define TRNSPS 3 +#define FIX_DUR 4 +#define MODES 5 +#define FIX_KEY 6 /* Options for the command */ #define HAS_ARG 1 static struct option long_opts[] = { - {"port", HAS_ARG, NULL, 'p'}, + {"device", HAS_ARG, NULL, 'D'}, {"opl3", 0, NULL, '4'}, - {"list", 0, NULL, 'l'}, {"path", HAS_ARG, NULL, 'P'}, {"verbose", HAS_ARG, NULL, 'v'}, {"version", 0, NULL, 'V'}, @@ -83,87 +72,35 @@ static struct option long_opts[] = { /* Number of elements in an array */ #define NELEM(a) ( sizeof(a)/sizeof((a)[0]) ) -#define ADDR_PARTS 4 /* Number of part in a port description addr 1:2:3:4 */ -#define SEP ", \t" /* Separators for port description */ - -#define SBI_FILE_TYPE_2OP 0 -#define SBI_FILE_TYPE_4OP 1 +enum { + FM_PATCH_UNKNOWN, + FM_PATCH_OPL2, + FM_PATCH_OPL3 +}; /* Default file type */ -int file_type = SBI_FILE_TYPE_2OP; +static int file_type = FM_PATCH_OPL2; /* Default verbose level */ -int verbose = 0; +static int verbose = 0; /* Global declarations */ -snd_seq_t *seq_handle; - -int seq_client; -int seq_port; -int seq_dest_client; -int seq_dest_port; +static snd_hwdep_t *handle; #ifndef PATCHDIR #define PATCHDIR "/usr/share/sounds/opl3" #endif -char *patchdir = PATCHDIR; +static char *patchdir = PATCHDIR; /* Function prototypes */ -static void show_list (); -static void show_usage (); -static void show_op (fm_instrument_t * fm_instr); +static void show_usage (void); +static void show_op (struct sbi_patch * instr, int type); -static int load_patch (fm_instrument_t * fm_instr, int bank, int prg, char *name); +static int load_patch (struct sbi_patch * instr); static int load_file (int bank, char *filename); -static int parse_portdesc (char *portdesc); -static int init_client (); - -static void -ignore_errors (const char *file, int line, const char *function, int err, const char *fmt, ...) -{ - /* ignore */ -} - -/* - * Show a list of possible output ports that midi could be sent to. - */ -static void -show_list () { - snd_seq_client_info_t *cinfo; - snd_seq_port_info_t *pinfo; - - int client, err; - - snd_lib_error_set_handler (ignore_errors); - if ((err = snd_seq_open (&seq_handle, "hw", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { - fprintf (stderr, "Could not open sequencer: %s\n", snd_strerror (err)); - return; - } - - printf (" Port %-30.30s %s\n", "Client name", "Port name"); - snd_seq_client_info_alloca(&cinfo); - snd_seq_client_info_set_client(cinfo, -1); - while (snd_seq_query_next_client(seq_handle, cinfo) >= 0) { - client = snd_seq_client_info_get_client(cinfo); - snd_seq_port_info_alloca(&pinfo); - snd_seq_port_info_set_client(pinfo, client); - snd_seq_port_info_set_port(pinfo, -1); - while (snd_seq_query_next_port(seq_handle, pinfo) >= 0) { - unsigned int cap; - - cap = (SND_SEQ_PORT_CAP_SUBS_WRITE | SND_SEQ_PORT_CAP_WRITE); - if ((snd_seq_port_info_get_capability(pinfo) & cap) == cap) { - printf ("%3d:%-3d %-30.30s %s\n", - client, snd_seq_port_info_get_port(pinfo), - snd_seq_client_info_get_name(cinfo), - snd_seq_port_info_get_name(pinfo)); - } - } - } - snd_seq_close (seq_handle); -} +static int init_hwdep (const char *name); /* * Show usage message @@ -172,11 +109,10 @@ show_usage () { show_usage () { char **cpp; static char *msg[] = { - "Usage: sbiload [-p client:port] [-4] [-l] [-P path] [-v level] instfile drumfile", + "Usage: sbiload [-D device] [-4] [-P path] [-v level] instfile drumfile", "", - " -p client:port - A alsa client and port number to send midi to", + " -D device - hwdep device string", " -4 - four operators file type (default = two ops)", - " -l - List possible output ports that could be used", " -P path - Specify the patch path", " -v level - Verbose level (default = 0)", " -V - Show version", @@ -199,169 +135,86 @@ show_version () { * Show instrument FM operators */ static void -show_op (fm_instrument_t * fm_instr) { +show_op (struct sbi_patch * inst, int type) { int i = 0; + int ofs = 0; do { - printf (" OP%i: flags: %s %s %s %s\011OP%i: flags: %s %s %s %s\n", - i, - fm_instr->op[i].am_vib & (1 << 7) ? "AM" : " ", - fm_instr->op[i].am_vib & (1 << 6) ? "VIB" : " ", - fm_instr->op[i].am_vib & (1 << 5) ? "EGT" : " ", - fm_instr->op[i].am_vib & (1 << 4) ? "KSR" : " ", - i + 1, - fm_instr->op[i + 1].am_vib & (1 << 7) ? "AM" : " ", - fm_instr->op[i + 1].am_vib & (1 << 6) ? "VIB" : " ", - fm_instr->op[i + 1].am_vib & (1 << 5) ? "EGT" : " ", - fm_instr->op[i + 1].am_vib & (1 << 4) ? "KSR" : ""); - printf (" OP%i: MULT = 0x%x" "\011\011OP%i: MULT = 0x%x\n", - i, fm_instr->op[i].am_vib & 0x0f, - i + 1, fm_instr->op[i + 1].am_vib & 0x0f); - printf (" OP%i: KSL = 0x%x TL = 0x%.2x\011OP%i: KSL = 0x%x TL = 0x%.2x\n", - i, (fm_instr->op[i].ksl_level >> 6) & 0x03, fm_instr->op[i].ksl_level & 0x3f, - i + 1, (fm_instr->op[i + 1].ksl_level >> 6) & 0x03, fm_instr->op[i + 1].ksl_level & 0x3f); - printf (" OP%i: AR = 0x%x DL = 0x%x\011OP%i: AR = 0x%x DL = 0x%x\n", - i, (fm_instr->op[i].attack_decay >> 4) & 0x0f, fm_instr->op[i].attack_decay & 0x0f, - i + 1, (fm_instr->op[i + 1].attack_decay >> 4) & 0x0f, fm_instr->op[i + 1].attack_decay & 0x0f); - printf (" OP%i: SL = 0x%x RR = 0x%x\011OP%i: SL = 0x%x RR = 0x%x\n", - i, (fm_instr->op[i].sustain_release >> 4) & 0x0f, fm_instr->op[i].sustain_release & 0x0f, - i + 1, (fm_instr->op[i + 1].sustain_release >> 4) & 0x0f, fm_instr->op[i + 1].sustain_release & 0x0f); - printf (" OP%i: WS = 0x%x\011\011OP%i: WS = 0x%x\n", - i, fm_instr->op[i].wave_select & 0x07, - i + 1, fm_instr->op[i + 1].wave_select & 0x07); - printf (" FB = 0x%x, %s\n", - (fm_instr->feedback_connection[i / 2] >> 1) & 0x07, - fm_instr->feedback_connection[i / 2] & (1 << 0) ? "parallel" : "serial"); - i += 2; + unsigned char val; + val = inst->data[AM_VIB + ofs]; + printf (" OP%i: flags: %s %s %s %s", i, + val & (1 << 7) ? "AM" : " ", + val & (1 << 6) ? "VIB" : " ", + val & (1 << 5) ? "EGT" : " ", + val & (1 << 4) ? "KSR" : " "); + val = inst->data[AM_VIB + ofs + 1]; + printf ("\011OP%i: flags: %s %s %s %s\n", i + 1, + val & (1 << 7) ? "AM" : " ", + val & (1 << 6) ? "VIB" : " ", + val & (1 << 5) ? "EGT" : " ", + val & (1 << 4) ? "KSR" : " "); + val = inst->data[AM_VIB + ofs]; + printf (" OP%i: MULT = 0x%x", i, val & 0x0f); + val = inst->data[AM_VIB + ofs + 1]; + printf ("\011\011OP%i: MULT = 0x%x\n", i + 1, val & 0x0f); + + val = inst->data[KSL_LEVEL + ofs]; + printf (" OP%i: KSL = 0x%x TL = 0x%.2x", i, + (val >> 6) & 0x03, val & 0x3f); + val = inst->data[KSL_LEVEL + ofs + 1]; + printf ("\011OP%i: KSL = 0x%x TL = 0x%.2x\n", i + 1, + (val >> 6) & 0x03, val & 0x3f); + val = inst->data[ATTACK_DECAY + ofs]; + printf (" OP%i: AR = 0x%x DL = 0x%x", i, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[ATTACK_DECAY + ofs + 1]; + printf ("\011OP%i: AR = 0x%x DL = 0x%x\n", i + 1, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[SUSTAIN_RELEASE + ofs]; + printf (" OP%i: SL = 0x%x RR = 0x%x", i, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[SUSTAIN_RELEASE + ofs + 1]; + printf ("\011OP%i: SL = 0x%x RR = 0x%x\n", i + 1, + (val >> 4) & 0x0f, val & 0x0f); + val = inst->data[WAVE_SELECT + ofs]; + printf (" OP%i: WS = 0x%x", i, val & 0x07); + val = inst->data[WAVE_SELECT + ofs + 1]; + printf ("\011\011OP%i: WS = 0x%x\n", i + 1, val & 0x07); + val = inst->data[CONNECTION + ofs]; + printf (" FB = 0x%x, %s\n", (val >> 1) & 0x07, + val & (1 << 0) ? "parallel" : "serial"); + i += 2; + ofs += OFFSET_4OP; } - while (i == (fm_instr->type == FM_PATCH_OPL3) << 1); + while (i == (type == FM_PATCH_OPL3) << 1); printf (" Extended data:\n" " ED = %.3i EA = %.3i CS = %.3i TR = %.3i\n" " FD = %.3i MO = %.3i FK = %.3i\n", - fm_instr->echo_delay, fm_instr->echo_atten, fm_instr->chorus_spread, fm_instr->trnsps, - fm_instr->fix_dur, fm_instr->modes, fm_instr->fix_key); -} - -/* - * Check the result of the previous instr event - */ -static int -check_result (int evtype) { - snd_seq_event_t *pev; - int err; - - for (;;) { - if ((err = snd_seq_event_input (seq_handle, &pev)) < 0) { - fprintf (stderr, "Unable to read event input: %s\n", - snd_strerror (err)); - return -ENXIO; - } - if (pev->type == SND_SEQ_EVENT_RESULT && pev->data.result.event == evtype) - break; - snd_seq_free_event (pev); - } - err = pev->data.result.result; - snd_seq_free_event (pev); - - return err; + inst->extension[ECHO_DELAY], inst->extension[ECHO_ATTEN], + inst->extension[CHORUS_SPREAD], inst->extension[TRNSPS], + inst->extension[FIX_DUR], inst->extension[MODES], + inst->extension[FIX_KEY]); } /* * Send patch to destination port */ static int -load_patch (fm_instrument_t * fm_instr, int bank, int prg, char *name) { - snd_instr_header_t *put; - snd_seq_instr_t id; - snd_seq_event_t ev; +load_patch (struct sbi_patch * inst) { - size_t size; - int err; - - if (verbose > 1) { - printf ("%.3i: [OPL%i] %s\n", prg, fm_instr->type == FM_PATCH_OPL3 ? 3 : 2, name); - show_op (fm_instr); - } - - if ((err = snd_instr_fm_convert_to_stream (fm_instr, name, &put, &size)) < 0) { - fprintf (stderr, "Unable to convert instrument %.3i to stream: %s\n", - prg, snd_strerror (err)); - return -1; - } - memset(&id, 0, sizeof(id)); - id.std = SND_SEQ_INSTR_TYPE2_OPL2_3; - id.prg = prg; - id.bank = bank; - snd_instr_header_set_id(put, &id); - - /* build event */ - memset (&ev, 0, sizeof (ev)); - ev.source.client = seq_client; - ev.source.port = seq_port; - ev.dest.client = seq_dest_client; - ev.dest.port = seq_dest_port; - - ev.flags = SND_SEQ_EVENT_LENGTH_VARUSR; - ev.queue = SND_SEQ_QUEUE_DIRECT; - -__again: - ev.type = SND_SEQ_EVENT_INSTR_PUT; - ev.data.ext.len = size; - ev.data.ext.ptr = put; - - if ((err = snd_seq_event_output (seq_handle, &ev)) < 0) { + ssize_t ret; + ret = snd_hwdep_write(handle, inst, sizeof(*inst)); + if (ret != sizeof(*inst)) { fprintf (stderr, "Unable to write an instrument %.3i put event: %s\n", - prg, snd_strerror (err)); + inst->prog, snd_strerror (ret)); return -1; } - if ((err = snd_seq_drain_output (seq_handle)) < 0) { - fprintf (stderr, "Unable to write instrument %.3i data: %s\n", prg, - snd_strerror (err)); - return -1; - } - - err = check_result (SND_SEQ_EVENT_INSTR_PUT); - if (err >= 0) { - if (verbose) - printf ("Loaded instrument %.3i, bank %.3i: %s\n", prg, bank, name); - return 0; - } else if (err == -EBUSY) { - snd_instr_header_t *remove; - - snd_instr_header_alloca(&remove); - snd_instr_header_set_cmd(remove, SND_SEQ_INSTR_FREE_CMD_SINGLE); - snd_instr_header_set_id(remove, snd_instr_header_get_id(put)); - - /* remove instrument */ - ev.type = SND_SEQ_EVENT_INSTR_FREE; - ev.data.ext.len = snd_instr_header_sizeof(); - ev.data.ext.ptr = remove; - - if ((err = snd_seq_event_output (seq_handle, &ev)) < 0) { - fprintf (stderr, "Unable to write an instrument %.3i free event: %s\n", - prg, snd_strerror (err)); - return -1; - } - - if ((err = snd_seq_drain_output (seq_handle)) < 0) { - fprintf (stderr, "Unable to write instrument %.3i data: %s\n", prg, - snd_strerror (err)); - return -1; - } - - if ((err = check_result (SND_SEQ_EVENT_INSTR_FREE)) < 0) { - fprintf (stderr, "Instrument %.3i, bank %.3i - free error: %s\n", - prg, bank, snd_strerror (err)); - return -1; - } - goto __again; - } - - fprintf (stderr, "Instrument %.3i, bank %.3i - put error: %s\n", - prg, bank, snd_strerror (err)); - return -1; + if (verbose) + printf ("Loaded instrument %.3i, bank %.3i: %s\n", + inst->prog, inst->bank, inst->name); + return 0; } /* @@ -369,66 +222,45 @@ __again: */ static void load_sb (int bank, int fd) { - int len, i; + int len; int prg; - - sbi_inst_t sbi_instr; - fm_instrument_t fm_instr; + struct sbi_patch inst; int fm_instr_type; - len = (file_type == SBI_FILE_TYPE_4OP) ? DATA_LEN_4OP : DATA_LEN_2OP; + len = (file_type == FM_PATCH_OPL2) ? DATA_LEN_4OP : DATA_LEN_2OP; for (prg = 0;; prg++) { - if (read (fd, &sbi_instr.header, sizeof (sbi_header_t)) < (ssize_t)sizeof (sbi_header_t)) + inst.prog = prg; + inst.bank = bank; + + if (read (fd, inst.key, 4) != 4) break; - if (!strncmp (sbi_instr.header.key, "SBI\032", 4) || !strncmp (sbi_instr.header.key, "2OP\032", 4)) { + if (!memcmp (inst.key, "SBI\032", 4) || !memcmp (inst.key, "2OP\032", 4)) { + len = DATA_LEN_2OP; fm_instr_type = FM_PATCH_OPL2; - } else if (!strncmp (sbi_instr.header.key, "4OP\032", 4)) { + } else if (!strncmp (inst.key, "4OP\032", 4)) { + len = DATA_LEN_4OP; fm_instr_type = FM_PATCH_OPL3; } else { - fm_instr_type = 0; if (verbose) printf ("%.3i: wrong instrument key!\n", prg); + fm_instr_type = FM_PATCH_UNKNOWN; } - if (read (fd, &sbi_instr.data, len) < len) + if (read (fd, &inst.name, sizeof(inst.name)) != sizeof(inst.name) || + read (fd, &inst.extension, sizeof(inst.extension)) != sizeof(inst.extension) || + read (fd, &inst.data, len) != len) break; - if (fm_instr_type == 0) + if (fm_instr_type == FM_PATCH_UNKNOWN) continue; - memset (&fm_instr, 0, sizeof (fm_instr)); - fm_instr.type = fm_instr_type; - - for (i = 0; i < 2; i++) { - fm_instr.op[i].am_vib = sbi_instr.data[AM_VIB + i]; - fm_instr.op[i].ksl_level = sbi_instr.data[KSL_LEVEL + i]; - fm_instr.op[i].attack_decay = sbi_instr.data[ATTACK_DECAY + i]; - fm_instr.op[i].sustain_release = sbi_instr.data[SUSTAIN_RELEASE + i]; - fm_instr.op[i].wave_select = sbi_instr.data[WAVE_SELECT + i]; - } - fm_instr.feedback_connection[0] = sbi_instr.data[CONNECTION]; - - if (fm_instr_type == FM_PATCH_OPL3) { - for (i = 0; i < 2; i++) { - fm_instr.op[i + 2].am_vib = sbi_instr.data[OFFSET_4OP + AM_VIB + i]; - fm_instr.op[i + 2].ksl_level = sbi_instr.data[OFFSET_4OP + KSL_LEVEL + i]; - fm_instr.op[i + 2].attack_decay = sbi_instr.data[OFFSET_4OP + ATTACK_DECAY + i]; - fm_instr.op[i + 2].sustain_release = sbi_instr.data[OFFSET_4OP + SUSTAIN_RELEASE + i]; - fm_instr.op[i + 2].wave_select = sbi_instr.data[OFFSET_4OP + WAVE_SELECT + i]; - } - fm_instr.feedback_connection[1] = sbi_instr.data[OFFSET_4OP + CONNECTION]; + if (verbose > 1) { + printf ("%.3i: %s\n", inst.prog, inst.name); + show_op (&inst, fm_instr_type); } - fm_instr.echo_delay = sbi_instr.header.name[ECHO_DELAY]; - fm_instr.echo_atten = sbi_instr.header.name[ECHO_ATTEN]; - fm_instr.chorus_spread = sbi_instr.header.name[CHORUS_SPREAD]; - fm_instr.trnsps = sbi_instr.header.name[TRNSPS]; - fm_instr.fix_dur = sbi_instr.header.name[FIX_DUR]; - fm_instr.modes = sbi_instr.header.name[MODES]; - fm_instr.fix_key = sbi_instr.header.name[FIX_KEY]; - - if (load_patch (&fm_instr, bank, prg, sbi_instr.header.name) < 0) + if (load_patch (&inst) < 0) break; } return; @@ -461,92 +293,57 @@ load_file (int bank, char *filename) { } /* - * Parse port description + * Open a hwdep device */ -static int -parse_portdesc (char *portdesc) { - char *astr; - char *cp; - int a[ADDR_PARTS]; - int count; +static int open_hwdep (const char *name) +{ + int err; + snd_hwdep_info_t *info; - if (portdesc == NULL) - return -1; + if ((err = snd_hwdep_open (&handle, name, SND_HWDEP_OPEN_WRITE)) < 0) + return err; - for (astr = strtok (portdesc, SEP); astr; astr = strtok (NULL, SEP)) { - for (cp = astr, count = 0; cp && *cp; cp++) { - if (count < ADDR_PARTS) - a[count++] = atoi (cp); - cp = strchr (cp, ':'); - if (cp == NULL) - break; - } - - if (count == 2) { - seq_dest_client = a[0]; - seq_dest_port = a[1]; - } else { - printf ("Addresses in %d parts not supported yet\n", count); - return -1; - } + snd_hwdep_info_alloca(&info); + if (!snd_hwdep_info (handle, info)) { + snd_hwdep_iface_t iface = snd_hwdep_info_get_iface(info); + if (iface == SND_HWDEP_IFACE_OPL2 || iface == SND_HWDEP_IFACE_OPL3) + return 0; } - return 0; + snd_hwdep_close(handle); + handle = NULL; + return -EINVAL; } -/* - * Open sequencer, create client port and - * subscribe client to destination port - */ static int -init_client () { - char name[64]; - snd_seq_port_subscribe_t *sub; - snd_seq_addr_t addr; +init_hwdep (const char *name) { + int err; + char tmpname[16]; - if ((err = snd_seq_open (&seq_handle, "hw", SND_SEQ_OPEN_DUPLEX, 0)) < 0) { - fprintf (stderr, "Could not open sequencer: %s\n", snd_strerror (err)); + if (!name || !*name) { + /* auto probe */ + int card = -1; + snd_ctl_t *ctl; + + while (!snd_card_next(&card) && card >= 0) { + int dev; + sprintf(tmpname, "hw:%d", card); + if (snd_ctl_open(&ctl, tmpname, 0) < 0) + continue; + dev = -1; + while (!snd_ctl_hwdep_next_device(ctl, &dev) && dev >= 0) { + sprintf(tmpname, "hw:%d,%d", card, dev); + if (!open_hwdep(tmpname)) + return 0; + } + snd_ctl_close(ctl); + } + fprintf (stderr, "Can't find any OPL3 hwdep device\n"); return -1; } - seq_client = snd_seq_client_id (seq_handle); - if (seq_client < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to determine my client number: %s\n", - snd_strerror (err)); - return -1; - } - - sprintf (name, "sbiload - %i", getpid ()); - if ((err = snd_seq_set_client_name (seq_handle, name)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to set client info: %s\n", - snd_strerror (err)); - return -1; - } - - if ((seq_port = snd_seq_create_simple_port (seq_handle, "Output", - SND_SEQ_PORT_CAP_READ | SND_SEQ_PORT_CAP_WRITE, - SND_SEQ_PORT_TYPE_SPECIFIC)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to create a client port: %s\n", - snd_strerror (seq_port)); - return -1; - } - - snd_seq_port_subscribe_alloca(&sub); - addr.client = seq_client; - addr.port = seq_port; - snd_seq_port_subscribe_set_sender(sub, &addr); - addr.client = seq_dest_client; - addr.port = seq_dest_port; - snd_seq_port_subscribe_set_dest(sub, &addr); - snd_seq_port_subscribe_set_exclusive(sub, 1); - - if ((err = snd_seq_subscribe_port (seq_handle, sub)) < 0) { - snd_seq_close (seq_handle); - fprintf (stderr, "Unable to subscribe destination port: %s\n", - snd_strerror (errno)); + if ((err = open_hwdep (name)) < 0) { + fprintf (stderr, "Could not open hwdep %s: %s\n", name, snd_strerror (err)); return -1; } @@ -558,40 +355,25 @@ init_client () { * and close sequencer */ static void -finish_client () +finish_hwdep () { - snd_seq_port_subscribe_t *sub; - snd_seq_addr_t addr; - int err; - - snd_seq_port_subscribe_alloca(&sub); - addr.client = seq_client; - addr.port = seq_port; - snd_seq_port_subscribe_set_sender(sub, &addr); - addr.client = seq_dest_client; - addr.port = seq_dest_port; - snd_seq_port_subscribe_set_dest(sub, &addr); - if ((err = snd_seq_unsubscribe_port (seq_handle, sub)) < 0) { - fprintf (stderr, "Unable to unsubscribe destination port: %s\n", - snd_strerror (errno)); - } - snd_seq_close (seq_handle); + snd_hwdep_close(handle); + handle = NULL; } /* * Load a .SBI FM instrument patch * sbiload [-p client:port] [-l] [-P path] [-v level] instfile drumfile * - * -p, --port=client:port - An ALSA client and port number to use + * -D, --device=name - An ALSA hwdep name to use * -4 --opl3 - four operators file type - * -l, --list - List possible output ports that could be used * -P, --path=path - Specify the patch path * -v, --verbose=level - Verbose level */ int main (int argc, char **argv) { char opts[NELEM (long_opts) * 2 + 1]; - char *portdesc; + char *name; char *cp; int c; struct option *op; @@ -604,7 +386,7 @@ main (int argc, char **argv) { *cp++ = ':'; } - portdesc = NULL; + name = NULL; /* Deal with the options */ for (;;) { @@ -613,11 +395,11 @@ main (int argc, char **argv) { break; switch (c) { - case 'p': - portdesc = optarg; + case 'D': + name = optarg; break; case '4': - file_type = SBI_FILE_TYPE_4OP; + file_type = FM_PATCH_OPL3; break; case 'v': verbose = atoi (optarg); @@ -625,9 +407,6 @@ main (int argc, char **argv) { case 'V': show_version(); exit (1); - case 'l': - show_list (); - exit (0); case 'P': patchdir = optarg; break; @@ -637,30 +416,14 @@ main (int argc, char **argv) { } } - if (portdesc == NULL) { - portdesc = getenv ("ALSA_OUT_PORT"); - if (portdesc == NULL) { - fprintf (stderr, "No client/port specified.\n" - "You must supply one with the -p option or with the\n" - "environment variable ALSA_OUT_PORT\n"); - exit (1); - } - } - - /* Parse port description to dest_client and dest_port */ - if (parse_portdesc (portdesc) < 0) { - return 1; - } - - /* Initialize client and subscribe to destination port */ - if (init_client () < 0) { + if (init_hwdep (name) < 0) { return 1; } /* Process instrument and drum file */ if (optind < argc) { if (load_file (0, argv[optind++]) < 0) { - finish_client(); + finish_hwdep(); return 1; } } else { @@ -668,15 +431,14 @@ main (int argc, char **argv) { } if (optind < argc) { if (load_file (128, argv[optind]) < 0) { - finish_client(); + finish_hwdep(); return 1; } } else { fprintf(stderr, "Warning: drum file was not specified\n"); } - /* Unsubscribe destination port and close client */ - finish_client(); + finish_hwdep(); return 0; }

1 0

[alsa-devel] [PATCH] Change asound_fm.h in alsa-lib for OPL3 hwdep loader
by Takashi Iwai 26 Oct '07

26 Oct '07

The change for alsa-lib to make sbiload compile. Takashi diff -r f6057aeb5008 include/sound/asound_fm.h --- a/include/sound/asound_fm.h Thu Oct 25 15:36:03 2007 +0200 +++ b/include/sound/asound_fm.h Fri Oct 26 13:09:29 2007 +0200 @@ -29,16 +29,16 @@ #define SNDRV_DM_FM_MODE_OPL2 0x00 #define SNDRV_DM_FM_MODE_OPL3 0x01 -typedef struct snd_dm_fm_info { +struct snd_dm_fm_info { unsigned char fm_mode; /* OPL mode, see SNDRV_DM_FM_MODE_XXX */ unsigned char rhythm; /* percussion mode flag */ -} snd_dm_fm_info_t; +}; /* * Data structure composing an FM "note" or sound event. */ -typedef struct snd_dm_fm_voice { +struct snd_dm_fm_voice { unsigned char op; /* operator cell (0 or 1) */ unsigned char voice; /* FM voice (0 to 17) */ @@ -60,25 +60,25 @@ typedef struct snd_dm_fm_voice { unsigned char left; /* stereo left */ unsigned char right; /* stereo right */ unsigned char waveform; /* 3 bits: waveform shape */ -} snd_dm_fm_voice_t; +}; /* * This describes an FM note by its voice, octave, frequency number (10bit) * and key on/off. */ -typedef struct snd_dm_fm_note { +struct snd_dm_fm_note { unsigned char voice; /* 0-17 voice channel */ unsigned char octave; /* 3 bits: what octave to play */ unsigned int fnum; /* 10 bits: frequency number */ unsigned char key_on; /* set for active, clear for silent */ -} snd_dm_fm_note_t; +}; /* * FM parameters that apply globally to all voices, and thus are not "notes" */ -typedef struct snd_dm_fm_params { +struct snd_dm_fm_params { unsigned char am_depth; /* amplitude modulation depth (1=hi) */ unsigned char vib_depth; /* vibrato depth (1=hi) */ unsigned char kbd_split; /* keyboard split */ @@ -90,17 +90,17 @@ typedef struct snd_dm_fm_params { unsigned char tomtom; unsigned char cymbal; unsigned char hihat; -} snd_dm_fm_params_t; +}; /* * FM mode ioctl settings */ -#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, snd_dm_fm_info_t) +#define SNDRV_DM_FM_IOCTL_INFO _IOR('H', 0x20, struct snd_dm_fm_info) #define SNDRV_DM_FM_IOCTL_RESET _IO ('H', 0x21) -#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, snd_dm_fm_note_t) -#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, snd_dm_fm_voice_t) -#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, snd_dm_fm_params_t) +#define SNDRV_DM_FM_IOCTL_PLAY_NOTE _IOW('H', 0x22, struct snd_dm_fm_note) +#define SNDRV_DM_FM_IOCTL_SET_VOICE _IOW('H', 0x23, struct snd_dm_fm_voice) +#define SNDRV_DM_FM_IOCTL_SET_PARAMS _IOW('H', 0x24, struct snd_dm_fm_params) #define SNDRV_DM_FM_IOCTL_SET_MODE _IOW('H', 0x25, int) /* for OPL3 only */ #define SNDRV_DM_FM_IOCTL_SET_CONNECTION _IOW('H', 0x26, int) @@ -112,4 +112,21 @@ typedef struct snd_dm_fm_params { #define SNDRV_DM_FM_OSS_IOCTL_SET_MODE 0x24 #define SNDRV_DM_FM_OSS_IOCTL_SET_OPL 0x25 +/* + * Patch Record - fixed size for write + */ + +#define FM_KEY_SBI "SBI\032" +#define FM_KEY_2OP "2OP\032" +#define FM_KEY_4OP "4OP\032" + +struct sbi_patch { + unsigned char prog; + unsigned char bank; + char key[4]; + char name[25]; + char extension[7]; + unsigned char data[32]; +}; + #endif /* __SOUND_ASOUND_FM_H */

1 0

[alsa-devel] HDA: Add support for Samsung Q1 Ultra Vista edition.
by Tobin Davis 26 Oct '07

26 Oct '07

This patch adds full record and playback support for the Samsung Q1 Ultra - Vista model (different codec than the earlier Q1 Ultra models). Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] [PATCH] AD1986A 6-channel bugs
by zhejiang 26 Oct '07

26 Oct '07

Hi, I have a ICH8M + AD1986A machine. In 6 channel modes, if adjust the volumes when playing 2-channel or 4-channel music, codec will be muted and all widget's Amp-Out vals will be reset. If you play 2-channel files in 6 channel modes,there is noise in the jacks. By doing experiments, I found that the 6-channel bug was caused by sharing channels. If play 2-channel audio file,the rest channels will share channels with the first 2-channel. The HDA spec doesn’t define the exact behavior under such circumstances. Majority of the codec chips support sharing channel, but seem that AD1986A doesn’t support. I wrote a patch and tested it in my hardware,now it works well. diff -Nur a/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c b/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c --- a/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c 2007-10-25 10:03:29.000000000 -0400 +++ b/alsa-driver-1.0.15/alsa-kernel/pci/hda/hda_codec.c 2007-10-25 10:02:46.000000000 -0400 @@ -2506,7 +2506,8 @@ if (chs >= (i + 1) * 2) /* independent out */ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, i * 2, format); - else /* copy front */ + else if (codec->vendor_id != 0x11d41986) + /* copy front if it is NOT AD1986A*/ snd_hda_codec_setup_stream(codec, nids[i], stream_tag, 0, format); } In alsa bugzilla and maillist, I found a lot of issues about AD1986A 6-channel .Anybody know the detail about AD1986A codec chip and give any comments? Here is a old description of the issue. > > Hi, > > I have a ICH8M + AD1986A machine and run alsa-driver-1.0.15rc. > > I found that : > In 6 channel mode,if adjust the front or pcm volume when playing music, > all channel will be muted,but if adjust the volume alone(don't play > music at the same time),every thing is ok. > I use printk to display the verb in the azx_send_cmd(), > when adjusting the volume in either case, there are just two verb > (0x300) sent to the nid(0x1b) [correct action]. > But in the first case, the dac widget amp-out vals are all affected. > > > Some other interesting thing, > If i adjust the Sur or CLFE volume when playing music, the printk show > that correct verb is sent, but the codec status don't change. > If I adjust them alone,every thing is ok.

2 3

[alsa-devel] aplay appication failed to open the alsa device
by Tharmarajan G 24 Oct '07

24 Oct '07

Hi, I am working on a Development Board which has the PXA270 processor and TSC2301 Audio Codec Chip. The TSC2301 Chip is interfaced with the processor by SPI interface for reading/writing the TSC2301's registers. The TSC2301 Chip is interfaced with the processor by I2S interface for sending/receiving Audio Data to/from the TSC2301 Audio Codec Chip. I have downloaded the pxa-uda1380.c alsa driver source for uda1380 audio codec chip and modified it for tsc2301 . I am using the kernel version 2.6.22 and cross-compiler version arm-linux-gcc 3.4.1. I have built the kernel image with ALSA audio support and the applications aplay and arecord for playing and recording audio data using ALSA api. I have got the alsa related entries in the /proc filesystem . But when playing the aplay, it failed to open the device. [root@SIRIUS /flash2]#./aplay /usr/local/Songs/AjRaj-Anthimanthirai-Theme4.mp3 ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM default aplay: main:545: audio open error: No such file or directory [root@SIRIUS /flash2]# /aplay -D /dev/snd/pcmC0D0p /usr/local/Songs/AjRaj-Anthimanthirai-Theme4.mp3 ALSA lib pcm.c:2144:(snd_pcm_open_noupdate) Unknown PCM /dev/snd/pcmC0D0p aplay: main:545: audio open error: No such file or directory What could be the issue ? Am I missing anyting in the driver for alsa ? Regards, Tharmarajan G

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[alsa-devel] New hardware report
by Alain DANGER 24 Oct '07

24 Oct '07

Hi. I managed to get the sound work with ALSA on the following MB : Shuttle AK32VN. OS : Kubuntu 7.10. Regards. Alain Danger

2 1

[alsa-devel] mplayer plays video twice as fast with Alsa than with OSS
by Timur Tabi 24 Oct '07

24 Oct '07

Hi, I wrote an Alsa device driver that supposedly works with the OSS emulation, but one of our customers is reporting that mplayer with the "-ao oss" option plays video twice at double speed. If they select "-ao alsa" instead, then it plays normally. Is this a known problem?

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Re: [alsa-devel] [Jackit-devel] RME HDSPe floating point question
by Didier Villevalois 24 Oct '07

24 Oct '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, (Cross-posting my report - last time i do this) Great news (almost for me in anyways). RME HDSPe PCI *works as is* with the current driver. Firmware uploads as well. I had nothing to do to make it work! (I think the ExpressCard should work as well...) Maybe the alsa-project wiki page can be updated. Little by little in the next weeks, I will delve into the driver's code to identify parts that needs to be enhanced in order to take profit of the 32bit floating point format that the HDSPe card can provide. When ready, I'll contact RME to request them for informations about the activation of this mode. Didier. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHHcG6VE1mALQL3JsRAjeZAKCJqicClHgy1UDc1YvAywuw1FokyQCeJO2d BhWpkxL8xLGxKB3zmbWFQpk= =wpbm -----END PGP SIGNATURE-----

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[alsa-devel] [RFC] ALSA scenario API
by Liam Girdwood 23 Oct '07

23 Oct '07

A little late with this but "better late than never".....I did promise a few people that I would post what was imho needed for ALSA scenarios. Also, apologies for cross posting, but it some of this probably applies to both lists... ALSA probably needs functionality to define and set audio usage scenarios. A scenario is simply a list of sound card mixer settings for a given use case. This is quite important for mobile devices (e.g. Smartphones, GPS, PDA, etc) with complex audio routing and power requirements. Some example scenarios could be :- "HiFi Headphone Playback" "HiFi Speaker Playback" "Handset Phone Call" "Headset Phone Call" etc... Scenarios are changed atm by either manually setting each control (e.g. using alsamixer or alsa-lib in apps) or by using alsactl to restore a scenario. The above two methods are not very portable between devices and applications. We would ideally like to just say "set Phone Call Handset" for a regular handset phone call or "set HiFi Headphones" for MP3 playback. ALSA also needs to have functionality to alias mixer controls. This would allow a single master volume/mute control to be exposed to user space and have it aliased to different hardware controls based upon the current scenario. e.g. Many mobile devices have different physical volume controls for speaker and headphones. This makes it somewhat difficult for user space apps to change the correct volume (without being bound to the hardware). We would like to be able to just say "set volume to 0dB" and have the alias take care of the correct physical CODEC mixer for the current scenario. Proposal: ======== 1. Add some new scenario functions to (s)alsa-lib. e.g. snd_scenario_set(); snd_scenario_get(); snd_scenario_list(); _set() would set the scenario e.g. snd_scenario_set(card, SND_VOICE_CALL_HANDSET); We would need to have a list of standard scenarios so that applications could be more portable. Vendor scenario's should also be supported. _get() would return the current scenario. _list() would return a list of supported scenarios for the given device. 2. Add some new mixer alias functions to set/get the current "master" volume/mute. e.g. snd_kcontrol *snd_scenario_volume_set(card, type, volume); snd_kcontrol *snd_scenario_switch_set(card, type, switch); where type is playback, capture, sidetone, bypass, etc This would set the volume for the current scenario and would guarantee always to get/set the _correct_ volume/switch. 3. Add a new function to return the active PCM for a given scenario. e.g. snd_scenario_get_pcm(card, scenario); Would return the given PCM for a particular scenario. i.e. the WM9713 driver has PCM 0.0 for high quality (higher power) audio (MP3), PCM 0.1 for lower power voice and PCM 0.2 for lowest power and quality system sounds. Note: scenarios may need to be more tightly bound to PCM's rather than cards. 4. Define a file format to represent each scenario. Probably base this upon alsactl's format as alsamixer could be used to quickly generate good settings and then save them with alsactl. Alias, scenario names and PCM would need to be added later somehow. 5. (Optional) Allow scenarios to define active kcontrols. This would allow tools like alsamixer to only display mixer controls for the given scenario. Controls that don't effect the scenario would not be rendered. (similar to alsamixers Playback|Capture|All tab) 6. (OHM list) Add input driver (or similar) for sound card user events. We need to be able to send events like "Headphone Jack Insertion", "Mic Jack Insertion", etc to user space. This would allow changing of scenario and power configuration depending on policy and configuration. Just my 2p Unfortunately I'm not in a position to implement any of this functionality. However, it's probably useful to folks working on mobile devices. Maybe a good project for Google SoC. HTH Liam

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Re: [alsa-devel] [PATCH] asihpi checkpatch clean plus control name refactor
by Takashi Iwai 23 Oct '07

23 Oct '07

At Tue, 23 Oct 2007 16:48:20 +1300, Eliot Blennerhassett wrote: > > checkpatch.pl --no-tree now reports no errors on asihpi.c > Redo control names to be strings only (not using control index) > > Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> Thanks, applied to HG tree now. (BTW, I couldn't access the ftp. Anon-ftp isn't allowed?) Takashi

1 0

[alsa-devel] [PATCH] asihpi checkpatch clean plus control name refactor
by Eliot Blennerhassett 23 Oct '07

23 Oct '07

checkpatch.pl --no-tree now reports no errors on asihpi.c Redo control names to be strings only (not using control index) Signed-off-by: Eliot Blennerhassett <linux(a)audioscience.com> Patch here; (70k) ftp://ftp.audioscience.com/internet/download/beta/asihpi_checkpatch_clean.d… regards Eliot

1 0

[alsa-devel] USB device gives wrong data?
by william estrada 23 Oct '07

23 Oct '07

Hi group, I think I have found a problem with the USB sound driver? It appears that the current drive returns a x'81' for silence but it should be a x'7f'. When I use my laptop's sound device and record without a mic, I get a file full of x'7f's. If I do the same with my cheap-o USB device I get a file full of x'81's. The sound quality from the laptop is unacceptable, so that is way I want to use the USB device. I have written a recording program that I have a 'silence' filter that filters out segments of dead air. It works with the laptop's sound device but not the USB device. The filter works by remove x'7f's after allowing a selectable amount. I get the results using arecord which does not have the filtering. The USB device is a C -Media Electronics, Inc. I am running FC6. > [root@laptop ~]# lsmod |grep snd > snd_usb_audio 81985 0 > snd_usb_lib 21057 1 snd_usb_audio > snd_rawmidi 28481 1 snd_usb_lib > snd_hwdep 13637 1 snd_usb_audio > snd_ali5451 27853 1 > snd_ac97_codec 99813 1 snd_ali5451 > snd_ac97_bus 6593 1 snd_ac97_codec > snd_seq_dummy 8133 0 > snd_seq_oss 37057 0 > snd_seq_midi_event 11841 1 snd_seq_oss > snd_seq 57009 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event > snd_seq_device 12365 4 snd_rawmidi,snd_seq_dummy,snd_seq_oss,snd_seq > snd_pcm_oss 46433 0 > snd_mixer_oss 20545 1 snd_pcm_oss > snd_pcm 80965 4 snd_usb_audio,snd_ali5451,snd_ac97_codec,snd_pcm_oss > snd_timer 26693 2 snd_seq,snd_pcm > snd 58181 140 snd_usb_audio,snd_rawmidi,snd_hwdep,snd_ali5451,snd_ac97_codec,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer > soundcore 12193 1 snd > snd_page_alloc 14281 1 snd_pcm Where do I report this 'bug' or how do I fix it? Thanks for your time. Sorry for the multiple posting, not sure where is should have gone. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

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Re: [alsa-devel] WM9712 monoout
by Mike Rapoport 22 Oct '07

22 Oct '07

Liam Girdwood wrote: > On Sun, 2007-10-21 at 09:35 +0200, Mike Rapoport wrote: >> Hello, >> >> I'm trying to enable phone on a board with WM9715 codec. >> I have not found a control responsible for mute/unmute the MONOUT. >> My question is whether I should add this control as platform specific control, >> or it would be correct to patch sound/soc/codecs/wm9712.c? > > It should probably go in the codec driver. It looks like an omission. > >> Also, "Mono Playback Volume" control is not inverted, which seems to be a bug. >> > > Could you send a patch ? Sure. A also found that treble and bass need to be inverted. > Many Thanks > > Liam > Signed-off-by: Mike Rapoport <mike(a)compulab.co.il> sound/soc/codecs/wm9712.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 986b5d5..427cb61 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -102,7 +102,8 @@ SOC_SINGLE("Speaker Playback ZC Switch", AC97_MASTER, 7, 1, 0), SOC_SINGLE("Speaker Playback Invert Switch", AC97_MASTER, 6, 1, 0), SOC_SINGLE("Headphone Playback ZC Switch", AC97_HEADPHONE, 7, 1, 0), SOC_SINGLE("Mono Playback ZC Switch", AC97_MASTER_MONO, 7, 1, 0), -SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 0), +SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), +SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("ALC Target Volume", AC97_CODEC_CLASS_REV, 12, 15, 0), SOC_SINGLE("ALC Hold Time", AC97_CODEC_CLASS_REV, 8, 15, 0), @@ -145,8 +146,8 @@ SOC_ENUM("Bass Control", wm9712_enum[5]), SOC_SINGLE("Bass Cut-off Switch", AC97_MASTER_TONE, 12, 1, 1), SOC_SINGLE("Tone Cut-off Switch", AC97_MASTER_TONE, 4, 1, 1), SOC_SINGLE("Playback Attenuate (-6dB) Switch", AC97_MASTER_TONE, 6, 1, 0), -SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 0), -SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 0), +SOC_SINGLE("Bass Volume", AC97_MASTER_TONE, 8, 15, 1), +SOC_SINGLE("Treble Volume", AC97_MASTER_TONE, 0, 15, 1), SOC_SINGLE("Capture ADC Switch", AC97_REC_GAIN, 15, 1, 1), SOC_ENUM("Capture Volume Steps", wm9712_enum[6]),

2 3

[alsa-devel] snd-pcsp doesn't accept the index parameter
by Alexander E. Patrakov 22 Oct '07

22 Oct '07

It is impossible to put pcsp at a stable index without the trivial patch below. Signed-Off-By: Alexander E. Patrakov <patrakov(a)ums.usu.ru> --- pcsp.c +++ pcsp.c @@ -27,6 +27,8 @@ static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ static int enable = SNDRV_DEFAULT_ENABLE1; /* Enable this card */ +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for pcsp soundcard."); module_param(id, charp, 0444); MODULE_PARM_DESC(id, "ID string for pcsp soundcard."); module_param(enable, bool, 0444); -- Alexander E. Patrakov

2 1

[alsa-devel] [PATCH] hda: STAC9228 updated DMUX nid.
by Matthew Ranostay 22 Oct '07

22 Oct '07

Changed the dmux for STAC9228 from ADC1MUX to ADC0MUX to avoid confusion. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 7cf5e23f804e pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Oct 19 08:23:00 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Fri Oct 19 13:47:35 2007 -0400 @@ -2777,7 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1c; + spec->dmux_nid = 0x1b; break; default: spec->num_dmics = 0;

2 1

[alsa-devel] SALSA Lib - TLV Deprecated warnings
by J. Scott Merritt 22 Oct '07

22 Oct '07

It appears that if SALSA Lib 0.0.12 is built -without- TLV enabled in the configuration options, then simply -including- asoundlib.h from an application program will trigger deprecated warnings on some of the TLV read/write function definitions (even though they are not called from the application). Is there something we can do to the header files to eliminate these warnings ? Thanks, Scott.

2 3

[alsa-devel] alsa-lib not installing plugins
by Larry D. Pyeatt 22 Oct '07

22 Oct '07

Hi, I wrote a new plugin for alsa-lib. After running make, there are .o and .lo files in src/pcm. However, when I run make install, nothing gets put into /usr/lib/alsa-lib. I first tried installing alsa-lib-1.0.15rc3. Then I tried alsa-lib-1.0.13rc3 and alsa-lib-1.0.14rc4. All of them did the same thing. Am I missing something obvious? Since they are all doing the same thing, it must be either that I just don't know what I'm doing, or something else in my system is not working. I would appreciate any suggestions on what could cause this behavior. Thanks, -- Larry D. Pyeatt, PhD. pyeatt(a)cs.ttu.edu http://www.cs.ttu.edu/~pyeatt

2 1

Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 21 Oct '07

21 Oct '07

Hi, I understand why this happens. Your sound card has a single "headphone" output, and since it is single, it is called "Master" the stac you have also has a real master volume control, that controls all the DACs, not just the headphones. I added support for it, thus two controls collided. The following patch will rename the "Master" output to "Headphone" Note that on the stac 9205/9204, the only one that has no master volume (the volume knob) but has a "Master Volume" amplifier widget, this will probably rename the "Master Volume" to Headphones. This chip is quite different, thus I think this should be handled in its patch function. I see what I can do later. Meanwhile the rename from "Master" to Headphones" is the only regression. Best regards, Maxim Levitsky PS: I reply to this mail with the patch.

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Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Jan-Simon Möller 21 Oct '07

21 Oct '07

I've also no sound. Had it working using the alsa.git tree _before_ the last merge (>48h). Needed to use model=asus for this fujitsu v5505 here. Now it doesn't work, even with other model-options (fujitsu, laptop). legolas:/proc/asound # cat cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc400000 irq 22 legolas:/proc/asound # cat pcm 00-01: Conexant Digital : Conexant Digital : playback 1 00-00: CONEXANT Analog : CONEXANT Analog : playback 1 : capture 1 legolas:/proc/asound # cat hwdep 00-00: HDA Codec 0 legolas:/proc/asound # cat devices 2: : timer 3: [ 0- 1]: digital audio playback 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0- 0]: hardware dependent 7: [ 0] : control 8: : sequencer legolas:/proc/asound # cat card0/codec#0 Codec: Conexant CX20549 (Venice) Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x1734110e Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x140]: 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19 0x17* Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19 0x17* Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x2b 0x2b] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19 0x17* Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x40]: 48000 bits [0x6]: 16 20 formats [0x5]: PCM AC3 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates [0x540]: 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14* 0x12 0x11 0x15 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Best regards Jan-Simon

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Re: [alsa-devel] [Jackit-devel] RME HDSP driver enhancements for HDSPe cards
by John Rigg 20 Oct '07

20 Oct '07

On Fri, Oct 19, 2007 at 08:58:00AM +0100, Ed W wrote: > RME are also helpful to the extent that you ask them specific questions > and in the name of writing drivers. They should give you hardware > details, etc > > > As an aside, apart from CPU load I will be suprised if turning floats > into 32bit ints makes any difference to the audio, but I agree that it > would be a nice feature to have! IIRC the new RME cards convert 24 bit integers to 32 bit floats in hardware (and vice versa), thus saving apps like jackd (and the CPU) the trouble. jackd doesn't support that yet, but AFAIK it shouldn't be too much effort to change that. John

1 0

[alsa-devel] [PATCH] hda: Add dmux to STAC 9228
by Matthew Ranostay 19 Oct '07

19 Oct '07

Added a dmux to the STAC9228 cards with DMIC support. And added a STAC_DIGITAL_INPUT_SOURCE macro for repeating mixer code. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 821a4558dd47 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Oct 18 17:38:17 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Oct 18 17:35:56 2007 -0400 @@ -430,6 +430,16 @@ static struct hda_verb stac9205_core_ini {} }; +#define STAC_DIGITAL_INPUT_SOURCE(cnt) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Digital Input Source", \ + .count = cnt, \ + .info = stac92xx_dmux_enum_info, \ + .get = stac92xx_dmux_enum_get, \ + .put = stac92xx_dmux_enum_put,\ + } + #define STAC_INPUT_SOURCE(cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ @@ -482,14 +492,7 @@ static struct snd_kcontrol_new stac925x_ }; static struct snd_kcontrol_new stac9205_mixer[] = { - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Digital Input Source", - .count = 1, - .info = stac92xx_dmux_enum_info, - .get = stac92xx_dmux_enum_get, - .put = stac92xx_dmux_enum_put, - }, + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), STAC_VOLKNOB(0x24), @@ -521,6 +524,7 @@ static struct snd_kcontrol_new stac922x_ static struct snd_kcontrol_new stac927x_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), @@ -2773,6 +2777,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: /* STAC 9228 */ spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; + spec->dmux_nid = 0x1c; break; default: spec->num_dmics = 0;

3 5

[alsa-devel] ALSA Bluetooth connection issue
by 19 Oct '07

19 Oct '07

Hi All: Im using ALSA 1.0.14 and bluez 3.20 and linux 2.6.18.The linux 2.6.18 (montavista) is running on an embeded development board(using S3C2410). When i tried to use 'aplay' to connect my bluetooth headset, i got the following error from ALSA: $ aplay -D bluetooth /1.wav audio[326]: Accepted new client connection on unix socket (fd=8) audio[326]: Package PKT_TYPE_CFG_REQ:0 hcid[318]: /org/bluez: org.bluez.Manager.FindAdapter() audio[326]: Got path /org/bluez/hci0 for adapter with address 00:80:E1:00:00:00 hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: no matching session found. creating a new one hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceHandles() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache hcid[318]: /org/bluez/hci0: org.bluez.Adapter.GetRemoteServiceRecord() hcid[318]: found matching session, removing from list hcid[318]: sdp session added to cache audio[326]: Audio service discovery completed with success hcid[318]: /org/bluez/hci0: org.bluez.Adapter.FinishRemoteServiceTransaction() hcid[318]: found matching session, removing from list audio[326]: Found Hansfree record audio[326]: Discovered Headset service on RFCOMM channel 1 audio[326]: Found Headset record audio[326]: Found Audio Sink audio[326]: Selecting default device audio[326]: avdtp_ref(0x3b018): ref=2 audio[326]: a2dp_source_request_stream: selected SEP 0x37250 audio[326]: avdtp_ref(0x3b018): ref=3 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: AVDTP: connected signaling channel to 00:0D:3C:EB:53:F0 audio[326]: session_cb audio[326]: DISCOVER request succeeded audio[326]: seid 1 type 1 media 0 in use 0 audio[326]: seid 2 type 1 media 0 in use 0 audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 1 type 1 media 0 hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) hcid[318]: link_key_request (sba=00:80:E1:00:00:00, dba=00:0D:3C:EB:53:F0) audio[326]: session_cb audio[326]: GET_CAPABILITIES request succeeded audio[326]: seid 2 type 1 media 0 audio[326]: Discovery complete audio[326]: avdtp_set_configuration(0x3b018): int_seid=1, acp_seid=1 audio[326]: session_cb audio[326]: SET_CONFIGURATION request succeeded audio[326]: SBC Source: Set_Configuration_Cfm audio[326]: avdtp_ref(0x3b018): ref=4 audio[326]: stream state changed: IDLE -> CONFIGURED audio[326]: session_cb audio[326]: OPEN request succeeded audio[326]: stream state changed: CONFIGURED -> OPEN audio[326]: AVDTP: connected transport channel to 00:0D:3C:EB:53:F0 audio[326]: SBC Source: Open_Cfm audio[326]: session_cb audio[326]: START request succeeded audio[326]: SBC Source: Start_CAlignment trap: bluetoothd-serv (326) PC=0x0001be44 Instr=0xe1c320b0 Address=0xbec85c2d FSR 0x813 fm audio[326]: SBC Source SEP 0x37250 locked hcid[318]: Service owner exited: :1.2 aplay: main:550: audio open error: Input/output error hcid[318]: Audio service (audio) was killed by signal 7 ALSA's aplay reports an "Input/output error". Later i digged into alsa-lib-1.0.14/src/pcm/pcm.c and found that snd_config_search_definition() returns an error. It seems ALSA didn't find the proper conf definition or something like that. My ".asoundrc" is as followed: pcm.a2dpd { type a2dpd } pcm.bluetooth { type bluetooth device 00:0D:3C:EB:53:F0 } pcm.!default { type hw card 0 device 0 } pcm.card0 { type hw card 0 } ctl.card0 { type hw card 0 } Does someone have any idea about what's going wrong in this? Thanks in advance! semiyd _________________________________________________________________ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vista&mkt=en-US&form=QBRE

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[alsa-devel] ALSA-for custom Operating System????
by Anoop Thomas 19 Oct '07

19 Oct '07

First of all don't make fun of me if i am asking a foolishness.Could you plz tell me can i use alsa for my custom OS which concentrates more on audio side.It doesn't uses linux kernel.Does it make a difference???? (ie. is it enough to give sufficient data to ALSA to help me to make the audio out of the system).If I can plz give me a brief description.... Help me plz................. --------------------------------- Forgot the famous last words? Access your message archive online. Click here.

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[alsa-devel] [PATCH] s3c2443-ac97: compilation fix
by Krzysztof Helt 18 Oct '07

18 Oct '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The Samsung S3C24xx uses new architecture file layout in the post 2.6.23 kernel. This patch fixes include path for the s3c2443-ac97.c. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The patch was done against the 2.6.23-git13 kernel. diff -urp linux-ref/sound/soc/s3c24xx/s3c2443-ac97.c linux-new/sound/soc/s3c24xx/s3c2443-ac97.c --- linux-ref/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-09 22:31:38.000000000 +0200 +++ linux-new/sound/soc/s3c24xx/s3c2443-ac97.c 2007-10-18 22:27:43.547470033 +0200 @@ -32,7 +32,7 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-ac97.h> +#include <asm/plat-s3c/regs-ac97.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> ---------------------------------------------------------------------- Jaros�aw Drob, Przyjdziemy i Spie***ymy - poznaj kandydata! Kliknij >>> http://link.interia.pl/f1bef

1 0

[alsa-devel] Kernel hierarchy
by Adrian McMenamin 18 Oct '07

18 Oct '07

Why are sound drivers in /sound and not /drivers/sound? Ok, not the world's most pressing problem, but enquiring minds want to know! Adrian

2 1

[alsa-devel] [PATCH] hda: STAC9228 DMIC Support
by Matthew Ranostay 18 Oct '07

18 Oct '07

Added DMIC support for STAC9228 cards, as well as adding terminators to other hda_nid DMIC arrays, and added a new Vendor ID tag. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r e91f57f8127a pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Oct 09 11:58:41 2007 +0200 +++ b/pci/hda/hda_codec.c Wed Oct 17 11:59:47 2007 -0400 @@ -55,6 +55,7 @@ static struct hda_vendor_id hda_vendor_i { 0x10ec, "Realtek" }, { 0x1057, "Motorola" }, { 0x1106, "VIA" }, + { 0x111d, "IDT" }, { 0x11d4, "Analog Devices" }, { 0x13f6, "C-Media" }, { 0x14f1, "Conexant" }, diff -r e91f57f8127a pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Oct 09 11:58:41 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Wed Oct 17 13:06:58 2007 -0400 @@ -182,8 +182,9 @@ static hda_nid_t stac925x_dac_nids[1] = 0x02, }; -static hda_nid_t stac925x_dmic_nids[1] = { - 0x15, +static hda_nid_t stac925x_dmic_nids[2] = { + 0x15, + 0x00, /* terminator */ }; static hda_nid_t stac922x_adc_nids[2] = { @@ -202,6 +203,11 @@ static hda_nid_t stac927x_mux_nids[3] = 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmic_nids[3] = { + 0x13, 0x14, + 0x00, /* terminator */ +}; + static hda_nid_t stac9205_adc_nids[2] = { 0x12, 0x13 }; @@ -210,8 +216,9 @@ static hda_nid_t stac9205_mux_nids[2] = 0x19, 0x1a }; -static hda_nid_t stac9205_dmic_nids[2] = { +static hda_nid_t stac9205_dmic_nids[3] = { 0x17, 0x18, + 0x00, /* terminator */ }; static hda_nid_t stac9200_pin_nids[8] = { @@ -2716,7 +2723,6 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2725,7 +2731,6 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; @@ -2734,9 +2739,18 @@ static int patch_stac927x(struct hda_cod spec->mux_nids = stac927x_mux_nids; spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); - spec->num_dmics = 0; spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; + } + + switch (codec->subsystem_id) { + case 0x1028020A: /* STAC 9228 */ + case 0x10280209: /* STAC 9228 */ + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = 2; + break; + default: + spec->num_dmics = 0; } spec->multiout.dac_nids = spec->dac_nids; @@ -2798,7 +2812,7 @@ static int patch_stac9205(struct hda_cod spec->mux_nids = stac9205_mux_nids; spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; - spec->num_dmics = ARRAY_SIZE(stac9205_dmic_nids); + spec->num_dmics = 2; spec->dmux_nid = 0x1d; spec->init = stac9205_core_init;

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[alsa-devel] alsa-driver-1.0.15rc2 build issues on RHEL4/CentOS4
by James Pearson 18 Oct '07

18 Oct '07

alsa-driver-1.0.15rc2 won't build cleanly on RHEL4/CentOS4 (2.6.9 based kernel). I've included the relevant errors below after running: configure --with-redhat make -i The problem with misc/ac97_bus.c, I've already reported at: <http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003119.…> James Pearson Build errors: CC [M] /tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.o In file included from /tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.c:3: /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c: In function `ac97_bus_suspend': /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c:35: error: too few arguments to function /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c: In function `ac97_bus_resume': /tmp/alsa-driver-1.0.15rc2/misc/../alsa-kernel/ac97_bus.c:45: error: too few arguments to function make[4]: [/tmp/alsa-driver-1.0.15rc2/misc/ac97_bus.o] Error 1 (ignored) CC [M] /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o In file included from /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.c:2: /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c: In function `snd_cs5535audio_suspend': /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c:77: error: void value not ignored as it ought to be /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c: In function `snd_cs5535audio_resume': /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/../../alsa-kernel/pci/cs5535audio/cs5535audio_pm.c:95: error: void value not ignored as it ought to be make[5]: [/tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o] Error 1 (ignored) LD [M] /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/snd-cs5535audio.o ld: /tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/cs5535audio_pm.o: No such file: No such file or directory make[5]: [/tmp/alsa-driver-1.0.15rc2/pci/cs5535audio/snd-cs5535audio.o] Error 1 (ignored) Building modules, stage 2. MODPOST *** Warning: "class_device_destroy" [/tmp/alsa-driver-1.0.15rc2/pci/asihpi/snd-asihpi.ko] undefined! *** Warning: "ac97_bus_type" [/tmp/alsa-driver-1.0.15rc2/pci/ac97/snd-ac97-codec.ko] undefined! *** Warning: "platform_device_alloc" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_add" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_put" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-portman2x4.ko] undefined! *** Warning: "platform_device_alloc" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined! *** Warning: "platform_device_add" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined! *** Warning: "platform_device_put" [/tmp/alsa-driver-1.0.15rc2/drivers/snd-mts64.ko] undefined!

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[alsa-devel] [patch] snd_pcm_dmix_close: raise semaphore if unable to discard
by Mike Gorse 18 Oct '07

18 Oct '07

This patch causes snd_pcm_dmix_close() to up a semaphore after downing it if it is unable to discard it. It prevents some deadlock that I am getting when a couple of applications interact and one of them closes the device and later re-opens it. -- --- src/pcm/pcm_dmix.c.orig 2007-10-17 20:19:47.000000000 -0400 +++ src/pcm/pcm_dmix.c 2007-10-17 20:50:46.000000000 -0400 @@ -652,8 +652,11 @@ if (dmix->client) snd_pcm_direct_client_discard(dmix); shm_sum_discard(dmix); - if (snd_pcm_direct_shm_discard(dmix)) - snd_pcm_direct_semaphore_discard(dmix); + if (snd_pcm_direct_shm_discard(dmix)) { + if (snd_pcm_direct_semaphore_discard(dmix)) { + snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); + } + } else snd_pcm_direct_semaphore_up(dmix, DIRECT_IPC_SEM_CLIENT); free(dmix->bindings);

2 1

[alsa-devel] [patch] Fix bug with bt87x driver's load_all option
by Trent Piepho 18 Oct '07

18 Oct '07

snd-bt87x: Make the load_all option work correctly If the load_all option was turned on all cards would be treated as unknown, even those which are in the database. Of course, if the card is in the database there is no reason to use the load_all option. It's there to force loading when the card isn't in the database. But there are out of date wikis that say to do this and some distros might turn this option on by default. So, we keep the load_all option from turning known cards into unknown cards. Signed-off-by: Trent Piepho <xyzzy(a)speakeasy.org> diff --git a/pci/bt87x.c b/pci/bt87x.c --- a/pci/bt87x.c +++ b/pci/bt87x.c @@ -1021,7 +1021,7 @@ static int __devinit snd_bt87x_detect_ca int i; const struct pci_device_id *supported; - supported = pci_match_device(&driver, pci); + supported = pci_match_id(snd_bt87x_ids, pci); if (supported && supported->driver_data > 0) return supported->driver_data;

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[alsa-devel] [PATCH] ALSA - protect Dreamcast PCM driver (AICA) from G2 bus effects
by Adrian McMenamin 18 Oct '07

18 Oct '07

The G2 bus on the SEGA Dreamcast connects both the maple peripheral bus and the AICA sound memory. DMA requests on one can cause the other to timeout on memory operations. This patch prevents maple interrupts from causing hiccoughs in the AICA sound (maple bus code will land in 2.6.24). There are other cleanups for this (AICA) code - but this is in effect a regression fix rather than a cleanup. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 7397865..407981d 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -106,11 +106,14 @@ static void spu_write_wait(void) static void spu_memset(u32 toi, u32 what, int length) { int i; + unsigned long flags; snd_assert(length % 4 == 0, return); for (i = 0; i < length; i++) { if (!(i % 8)) spu_write_wait(); + local_irq_save(flags); writel(what, toi + SPU_MEMORY_BASE); + local_irq_restore(flags); toi++; } } @@ -118,6 +121,7 @@ static void spu_memset(u32 toi, u32 what, int length) /* spu_memload - write to SPU address space */ static void spu_memload(u32 toi, void *from, int length) { + unsigned long flags; u32 *froml = from; u32 __iomem *to = (u32 __iomem *) (SPU_MEMORY_BASE + toi); int i; @@ -128,7 +132,9 @@ static void spu_memload(u32 toi, void *from, int length) if (!(i % 8)) spu_write_wait(); val = *froml; + local_irq_save(flags); writel(val, to); + local_irq_restore(flags); froml++; to++; } @@ -138,28 +144,36 @@ static void spu_memload(u32 toi, void *from, int length) static void spu_disable(void) { int i; + unsigned long flags; u32 regval; spu_write_wait(); regval = readl(ARM_RESET_REGISTER); regval |= 1; spu_write_wait(); + local_irq_save(flags); writel(regval, ARM_RESET_REGISTER); + local_irq_restore(flags); for (i = 0; i < 64; i++) { spu_write_wait(); regval = readl(SPU_REGISTER_BASE + (i * 0x80)); regval = (regval & ~0x4000) | 0x8000; spu_write_wait(); + local_irq_save(flags); writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + local_irq_restore(flags); } } /* spu_enable - set spu registers to enable sound output */ static void spu_enable(void) { + unsigned long flags; u32 regval = readl(ARM_RESET_REGISTER); regval &= ~1; spu_write_wait(); + local_irq_save(flags); writel(regval, ARM_RESET_REGISTER); + local_irq_restore(flags); } /* @@ -168,25 +182,34 @@ static void spu_enable(void) */ static void spu_reset(void) { + unsigned long flags; spu_disable(); spu_memset(0, 0, 0x200000 / 4); /* Put ARM7 in endless loop */ + local_irq_save(flags); ctrl_outl(0xea000002, SPU_MEMORY_BASE); + local_irq_restore(flags); spu_enable(); } /* aica_chn_start - write to spu to start playback */ static void aica_chn_start(void) { + unsigned long flags; spu_write_wait(); + local_irq_save(flags); writel(AICA_CMD_KICK | AICA_CMD_START, (u32 *) AICA_CONTROL_POINT); + local_irq_restore(flags); } /* aica_chn_halt - write to spu to halt playback */ static void aica_chn_halt(void) { + unsigned long flags; spu_write_wait(); + local_irq_save(flags); writel(AICA_CMD_KICK | AICA_CMD_STOP, (u32 *) AICA_CONTROL_POINT); + local_irq_restore(flags); } /* ALSA code below */ @@ -213,12 +236,13 @@ static int aica_dma_transfer(int channels, int buffer_size, int q, err, period_offset; struct snd_card_aica *dreamcastcard; struct snd_pcm_runtime *runtime; - err = 0; + unsigned long flags; dreamcastcard = substream->pcm->private_data; period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels); runtime = substream->runtime; for (q = 0; q < channels; q++) { + local_irq_save(flags); err = dma_xfer(AICA_DMA_CHANNEL, (unsigned long) (runtime->dma_area + (AICA_BUFFER_SIZE * q) / @@ -228,9 +252,12 @@ static int aica_dma_transfer(int channels, int buffer_size, AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + AICA_PERIOD_SIZE * period_offset, buffer_size / channels, AICA_DMA_MODE); - if (unlikely(err < 0)) + if (unlikely(err < 0)) { + local_irq_restore(flags); break; + } dma_wait_for_completion(AICA_DMA_CHANNEL); + local_irq_restore(flags); } return err; }

2 1

[alsa-devel] RME HDSP driver enhancements for HDSPe cards
by Ptitjes 18 Oct '07

18 Oct '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, First, sorry for the cross-post, but as i did not know where to start, i wanted to ask to both lists. I bought a RME HDSPe PCI card. I should receive it in the following days. My plan is to: 1) Determine whether the current driver works as is with the HDSPe cards and make my report to both lists. 2) Enhance the current driver in order to take profit of the new feature (32bit float format) However, i am new to Alsa, Jack and HDSP driver internal code structure. So i wanted to collect informations and hints as where to start from and be the most efficient in starting the work as i can. Could you please give me links as to: - - General Alsa driver structure and data flow - - Work done by Alsa HDSP driver and work done by Jack (I ever thought most of the work is done by Jack as my current HDSP PCCard do not work with Alsa only) - - Parts of the Alsa HDSP driver that would a priori need changes. - - Parts of Jack that would a priori need changes. Thank you very much for your help. I hope i did not bother everyone with my newbie questions. Best regards, Didier. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHFdRLVE1mALQL3JsRAvp1AKCV7L7DdX1doenajmXFX8vnYkbbqQCgkbuS GZgsAKdtGyiwfP4vNYYdZb4= =vvgU -----END PGP SIGNATURE-----

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[alsa-devel] 1.0.15rc3 patch_analog.s bug + fix
by Peter Skensved 17 Oct '07

17 Oct '07

Hi, I've come across a problem with the 1.0.15rc3 alsa driver on my Lenovo 3000 N100 laptop using the auto mute feature - it appears to work backwards ! Plugging in the headphones unmutes the external speakers and unplugging the heaphones mutes them ... Changing the if statement in line ~600 in patch_analog.c in ad1986a_update_hp to if (!spec->jack_preset) makes it work the intended way. ( spec->jack_present changes state _before_ the call to ad1986a_hp_automute ) peter ---- Peter Skensved Email : peter(a)SNO.Phy.QueensU.CA Dept. of Physics, Queen's University, Kingston, Ontario, Canada

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[alsa-devel] Recent ALSA version needs special configuration for Terratec Phase 26 USB
by Janne Himanka 17 Oct '07

17 Oct '07

Hello, I run Debian unstable on my Athlon XP1800 machine, and updating libasound2 package from version 1.0.13-2 to 1.0.14 (and later) made my Terratec Phase 26 USB soundcard useless. Whenever I tried to play something, the program would crash with a segmentation fault. When I downgraded to 1.0.13, everything would work again. It took a lot of head scratching and experimentation to find out that adding cards.USB-Audio.pcm.use_dmix."USB2448" no to my .asoundrc (it is otherwise empty) is needed for it to work. After adding that, it seems to work as it did before. The string in hyphens varies according to the mode one selects from the Terratec front panel. There are four modes with different word length and bitrates. The Phase 26 is adaptive; it doesn't need converting for a specific format. Bitrates from 32 KHz to 96 KHz and word lengths from 16 to 32 bits just work. I'm writing you this because a comment in USB-Audio.conf says: # If your device requires such a definition to work correctly, please report it # to <alsa-devel(a)alsa-project.org>. If you need any additional info, I'll be glad to help. Regards, Janne Himanka

2 1

[alsa-devel] Asus Eeepc missing in alsa models list?
by Herton Ronaldo Krzesinski 17 Oct '07

17 Oct '07

Looking at the change 291702f017efdfe556cb87b8530eb7d1ff08cbae, may be the addition of the Asus Eeepc model was missed? The following patch fixes it if it was not intended originally: diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 4b48c2e..e2a1985 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -843,6 +843,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. 3stack-6ch-dig 3-stack (6-channel) with SPDIF 6stack-dig 6-stack with SPDIF lenovo-101e Lenovo laptop + eeepc-p701 ASUS Eeepc auto auto-config reading BIOS (default) ALC882/885 diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 53b0428..2ceaf4c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -12058,6 +12058,7 @@ static const char *alc662_models[ALC662_MODEL_LAST] = { [ALC662_3ST_6ch] = "3stack-6ch", [ALC662_5ST_DIG] = "6stack-dig", [ALC662_LENOVO_101E] = "lenovo-101e", + [ALC662_ASUS_EEEPC_P701] = "eeepc-p701", [ALC662_AUTO] = "auto", }; -- []'s Herton

2 1

Re: [alsa-devel] hda-intel: no soundcard with current linus' git tree
by Maxim Levitsky 17 Oct '07

17 Oct '07

On Wednesday 17 October 2007 00:18:42 Thomas Meyer wrote: > $ dmesg > > [schnipp] > > ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 21 > PCI: Enabling bus mastering for device 0000:00:1b.0 > PCI: Setting latency timer of device 0000:00:1b.0 to 64 > hda_codec: STAC922x, Apple subsys_id=106b0200 > ACPI: PCI interrupt for device 0000:00:1b.0 disabled > HDA Intel: probe of 0000:00:1b.0 failed with error -16 > > [schnipp] > > > - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ > Please bisect this (the alsa merge happened just short time ago) Which kernel did work? Best regards, Maxim Levitsky

1 0

[alsa-devel] About the buffer_time and period_time
by Fengwei Yin 17 Oct '07

17 Oct '07

Hi all, I met a problem about period_time/buffer_time on PXA with ASOC enabled. The kernel version is 2.6.21. The Audio stream is 32k which is not supported by PXA ASOC driver. Please refer following RATES definition for PXA2xx. #define PXA2XX_AC97_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_44100 | \ SNDRV_PCM_RATE_48000) So I set plughw as default ALSA device enable auto samplerate conversion. The code to set the period_time/buffer_time is as following: err = snd_pcm_hw_params_set_rate_near(hPCM, params, &rate, 0); if (err < 0) { error("Rate not available: %d\n", rate); return err; } err = snd_pcm_hw_params_get_buffer_time_max(params, &buffer_time, 0); if (buffer_time > 500000) buffer_time = 500000; buffer_time = 400000; period_time = buffer_time/4; err = snd_pcm_hw_params_set_buffer_time_near(hPCM, params, &buffer_time, 0); if (err < 0) { error("Buffer time not available: %d, error number: %d\n", buffer_time, err); return err; } err = snd_pcm_hw_params_set_period_time_near(hPCM, params, &period_time, 0); if (err < 0) { error("Period time not available: %d\n", period_time); return err; } If I set buffer_time to 400000, I will get: " Buffer time not available: 400000, error number: -22 Cannot init audio driver. Floating point exception " If I set buffer time to 500000, buffer_time can be set correctly. But audio playback failed. If I set period_time before setting buffer_time, Everything is OK. Any thought about this problem. Thanks. Regards Yin, Fengwei

2 2

[alsa-devel] Intel HDA digital out distortion
by pr0t0cod7 16 Oct '07

16 Oct '07

Hi, On Monday 23 July 2007 23:09:24 Ingmar Stieger wrote: > a couple of weeks ago, I bought an ASUS mainboard with integrated > Intel > HDA and an AD1988B codec. Unfortunately, the digital / SPDIF output > gets distorted with stock 1.0.14 drivers. It does not matter what the > volume levels of the various mixers are set to. I have the same issues. Any sound is distorted through spdif output (seems more pronounced with low frequencies.) Sound is not distorted: - using passthough (playing AC3) - when volume is set to the very minimal (no perceptible distortion) - with analog output My motherboard: ASUS P5B deluxe with integrated soundchip Kernel: 2.6.22 Alsa: Intel HD audio kernel built module asound.conf: http://pastebin.com/f2c54b47b > I think I might have found the source for this problem. Maybe it's > only a workaround, but it solved the issue and digital out now works > fine for me, so I thought I'd share. The patch worked for me, no more distortion :) Thanks for sharing, Alex.

2 2

[alsa-devel] fsc esprimo v5505 (snd_intel_hda) - output in /var/log/messages
by Jan-Simon Möller 16 Oct '07

16 Oct '07

Hi! What do these messages mean ? Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_intel.c:1185: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x5, channel=0, format=0x11 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x19, stream=0x0, channel=0, format=0x0 Oct 15 09:26:22 legolas kernel: ALSA sound/pci/hda/hda_codec.c:623: hda_codec_setup_stream: NID=0x13, stream=0x0, channel=0, format=0x0 Thanks ! Jan

2 1

Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 16 Oct '07

16 Oct '07

On 10/07/2007 07:04 PM, Peteris Krisjanis wrote: >> "Current sample rate" only has meaning with the device open , and if I >> understand you right, a "cat >> /proc/asound/card0/pcm0{p,c}/sub0/hw_params will then get you what you >> want as to the "get" part (substitute for 0 in the above as required). > > I actually meant what kind of sample rate is set in sound card. For > example, my onboard sound card has fixed Hz. I would like to know such > information _before_ I touch sound card :) Okay, you want the information that the driver author supplied in the struct snd_pcm_hardware for the stream. I've in fact also wanted that information on a number of occasions and just looked at the driver source then. > So you suggest to open device via any means and then read that file? I > did this and it works. > I guess I can even try to implement it in app and check for sample > rate everytime before I play. > > But is this only solution? It's rather hack, and I would like to have > more reliable method. You might consider this less of of a hack -- it uses the ALSA API to query the information from the PCM handle directly. Please see the library docoumentation for other information you can extract from a hw_params struct. (I don't suppose you can ever get anything but dir == 0 for "hw" devices). Introducing a /proc/asound/card0/pcm0{p,c}/hw_limits or similar might not be a bad idea though? Rene /* gcc -W -Wall -o snd_rate snd_rate.c -l asound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(void) { snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; int err; snd_pcm_hw_params_alloca(&params); err = snd_pcm_open(&handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return -1; } err = snd_pcm_hw_params_get_rate_min(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return -1; } printf("min: (%c) %u\n", "<=>"[dir + 1], val); err = snd_pcm_hw_params_get_rate_max(params, &val, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return -1; } printf("max: (%c) %u\n", "<=>"[dir + 1], val); snd_pcm_close(handle); return 0; }

4 7

[alsa-devel] intel-hda patch for Dell Latitude D531
by glogow＠fbihome.de 16 Oct '07

16 Oct '07

Hi I have attached three files to this mail: 1. patch_sigmatel.c.diff I have a Latitude D531 which has a ATI SB600 with a STAC 9205. The chip is detected, but no config was read with the latest drivers. The problems are the dmic_nids arrays, which are ignored for auto config lookup. This calles is_in_nid_list, which expects a 0 terminated list, but these arrays are not. Additionally I updated the dell_9205_m43_pin_configs with values from the latest win drivers. 2. create_dell_tables.pl When looking into the ini files and comparing the nids and config values of the various tables, I got really confused, so I wrote a litte script to extract the values and build the table. This is the extended version, after I got my sound working. It extracts all pin infos from all ini files, generates the apropriate tables and prepends the Version.txt from the driver, resultiung in a kind of C header. Maybe it's usefull for you. 3. patch_sigmatel_dell.h Output from running create_dell_tables.pl on the latest drivers for my D531 from Dell. HTH Jan-Marek Glogowski ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program.

3 3

[alsa-devel] ALSA 1.0.15 final
by Jaroslav Kysela 16 Oct '07

16 Oct '07

Hello all, delayed but ... all ALSA packages 1.0.15 final are available for download on http://www.alsa-project.org . The changelog is available on web for 1.0.14 -> 1.0.15. I will make also available 1.0.15rc3 -> 1.0.15 changelog in next few hours. Have fun, Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

1 0

[alsa-devel] How to detect 'dead air' while recording?
by william estrada 16 Oct '07

16 Oct '07

Hi group, I want to filter out 'dead air' while recording. What is the best way to do this? I don't want to use JACK, just ALSA. Any URLs? -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

1 0

[alsa-devel] LGPL Licensing
by WOODEN John 15 Oct '07

15 Oct '07

I wish to use either the ALSA or SALSA API within my application. I also intend to commercially distribute my application and sell it. What affect does ALSA's and/or SALSA's Lesser GPL license have upon applications that utilize their API's? Do I have to make the source code of my application freely available if I choose to utilize the ALSA or SALSA API? Thanks, John. IMPORTANT NOTICE: This e-mail and any attachment to it are intended only to be read or used by the named addressee. It is confidential and may contain legally privileged information. No confidentiality or privilege is waived or lost by any mistaken transmission to you. The RTA is not responsible for any unauthorised alterations to this e-mail or attachment to it. Views expressed in this message are those of the individual sender, and are not necessarily the views of the RTA. If you receive this e-mail in error, please immediately delete it from your system and notify the sender. You must not disclose, copy or use any part of this e-mail if you are not the intended recipient.

2 1

Re: [alsa-devel] [Pkg-alsa-devel] Bug#429326: Bug#437827: libasound2: Sound creaks badly when playing DVDs or OGG files
by Elimar Riesebieter 15 Oct '07

15 Oct '07

forwarded 429326 alsa-devel(a)alsa-project.org thanks On Sun, 14 Oct 2007 the mental interface of Loïc Minier told: > On Sun, Oct 14, 2007, Elimar Riesebieter wrote: > > From my point of view it is not reasonable to patch Debian packages, > > which of course will break 32bit only sound cards. So we've to find > > a way to force a solution directly on alsa-devel. Who will forward > > the _Debian_bug to alsa-devel(a)alsa-project.org ? > > I don't understand your remark about patching the Debian package? Why > would reverting the bit shift macros break anything? > > The attached patch for alsa-lib should be enough to fix this, upstream > should revert the broken change in the bit shifts. Yeah, we can provide it. But shouldn't it be a solution for all ALSA users? This is Debian #429326. As well reported as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3360. We have to look for a general solution as the distributors don't need to patch sources. > Alsa #3360; GNOME #436192; LP #116990; Mandriva #33908; Revert HG r2264 > (changeset 23c4c0f5de40) as the bit shifts overflow when the volume is high > for 32 bits to 16 bits resampling > > diff -r 23c4c0f5de40 -r 883ae69471d2 src/pcm/plugin_ops.h > --- a/src/pcm/plugin_ops.h Fri Mar 09 17:06:23 2007 +0100 > +++ b/src/pcm/plugin_ops.h Fri Mar 09 15:30:38 2007 +0100 > @@ -85,8 +85,6 @@ static inline u_int32_t sx24s(u_int32_t > #define _put_triple(ptr,val) _put_triple_be(ptr,val) > #define _put_triple_s(ptr,val) _put_triple_le(ptr,val) > #endif > - > -#define shift_down(val, bits) (((val) + (1 << ((bits) - 1))) >> (bits)) > > #ifdef COPY_LABELS > static void *copy_labels[5] = { > @@ -258,14 +256,14 @@ conv_xxx1_009x: as_u32(dst) = sx24s((u_i > conv_xxx1_009x: as_u32(dst) = sx24s((u_int32_t)(as_u8c(src) ^ 0x80) << 8); goto CONV_END; > conv_xxx1_9000: as_u32(dst) = (u_int32_t)(as_u8c(src) ^ 0x80) << 24; goto CONV_END; > conv_xxx1_0009: as_u32(dst) = (u_int32_t)(as_u8c(src) ^ 0x80); goto CONV_END; > -conv_xx12_xxx1: as_u8(dst) = shift_down(as_u16c(src), 8); goto CONV_END; > +conv_xx12_xxx1: as_u8(dst) = as_u16c(src) >> 8; goto CONV_END; > conv_xx12_xx12: as_u16(dst) = as_u16c(src); goto CONV_END; > conv_xx12_xx21: as_u16(dst) = bswap_16(as_u16c(src)); goto CONV_END; > conv_xx12_x120: as_u32(dst) = sx24((u_int32_t)as_u16c(src) << 8); goto CONV_END; > conv_xx12_021x: as_u32(dst) = sx24s((u_int32_t)bswap_16(as_u16c(src)) << 8); goto CONV_END; > conv_xx12_1200: as_u32(dst) = (u_int32_t)as_u16c(src) << 16; goto CONV_END; > conv_xx12_0021: as_u32(dst) = (u_int32_t)bswap_16(as_u16c(src)); goto CONV_END; > -conv_xx12_xxx9: as_u8(dst) = shift_down(as_u16c(src), 8) ^ 0x80; goto CONV_END; > +conv_xx12_xxx9: as_u8(dst) = (as_u16c(src) >> 8) ^ 0x80; goto CONV_END; > conv_xx12_xx92: as_u16(dst) = as_u16c(src) ^ 0x8000; goto CONV_END; > conv_xx12_xx29: as_u16(dst) = bswap_16(as_u16c(src)) ^ 0x80; goto CONV_END; > conv_xx12_x920: as_u32(dst) = sx24((u_int32_t)(as_u16c(src) ^ 0x8000) << 8); goto CONV_END; > @@ -284,57 +282,57 @@ conv_xx12_01Ax: as_u32(dst) = sx24s((u_i > conv_xx12_01Ax: as_u32(dst) = sx24s((u_int32_t)(as_u16c(src) ^ 0x80) << 8); goto CONV_END; > conv_xx12_A100: as_u32(dst) = (u_int32_t)bswap_16(as_u16c(src) ^ 0x80) << 16; goto CONV_END; > conv_xx12_001A: as_u32(dst) = (u_int32_t)(as_u16c(src) ^ 0x80); goto CONV_END; > -conv_x123_xxx1: as_u8(dst) = shift_down(as_u32c(src), 16); goto CONV_END; > -conv_x123_xx12: as_u16(dst) = shift_down(as_u32c(src), 8); goto CONV_END; > -conv_x123_xx21: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)); goto CONV_END; > +conv_x123_xxx1: as_u8(dst) = as_u32c(src) >> 16; goto CONV_END; > +conv_x123_xx12: as_u16(dst) = as_u32c(src) >> 8; goto CONV_END; > +conv_x123_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 8); goto CONV_END; > conv_x123_x123: as_u32(dst) = sx24(as_u32c(src)); goto CONV_END; > conv_x123_321x: as_u32(dst) = sx24s(bswap_32(as_u32c(src))); goto CONV_END; > conv_x123_1230: as_u32(dst) = as_u32c(src) << 8; goto CONV_END; > -conv_x123_0321: as_u32(dst) = shift_down(bswap_32(as_u32c(src)), 8); goto CONV_END; > - conv_x123_xxx9: as_u8(dst) = shift_down(as_u32c(src), 16) ^ 0x80; goto CONV_END; > -conv_x123_xx92: as_u16(dst) = shift_down(as_u32c(src), 8) ^ 0x8000; goto CONV_END; > -conv_x123_xx29: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)) ^ 0x80; goto CONV_END; > +conv_x123_0321: as_u32(dst) = bswap_32(as_u32c(src)) >> 8; goto CONV_END; > +conv_x123_xxx9: as_u8(dst) = (as_u32c(src) >> 16) ^ 0x80; goto CONV_END; > +conv_x123_xx92: as_u16(dst) = (as_u32c(src) >> 8) ^ 0x8000; goto CONV_END; > +conv_x123_xx29: as_u16(dst) = bswap_16(as_u32c(src) >> 8) ^ 0x80; goto CONV_END; > conv_x123_x923: as_u32(dst) = sx24(as_u32c(src) ^ 0x800000); goto CONV_END; > conv_x123_329x: as_u32(dst) = sx24s(bswap_32(as_u32c(src)) ^ 0x8000); goto CONV_END; > conv_x123_9230: as_u32(dst) = (as_u32c(src) ^ 0x800000) << 8; goto CONV_END; > -conv_x123_0329: as_u32(dst) = shift_down(bswap_32(as_u32c(src)), 8) ^ 0x80; goto CONV_END; > -conv_123x_xxx3: as_u8(dst) = shift_down(as_u32c(src), 8) & 0xff; goto CONV_END; > -conv_123x_xx32: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8)); goto CONV_END; > -conv_123x_xx23: as_u16(dst) = shift_down(as_u32c(src), 8) & 0xffff; goto CONV_END; > +conv_x123_0329: as_u32(dst) = (bswap_32(as_u32c(src)) >> 8) ^ 0x80; goto CONV_END; > +conv_123x_xxx3: as_u8(dst) = (as_u32c(src) >> 8) & 0xff; goto CONV_END; > +conv_123x_xx32: as_u16(dst) = bswap_16(as_u32c(src) >> 8); goto CONV_END; > +conv_123x_xx23: as_u16(dst) = (as_u32c(src) >> 8) & 0xffff; goto CONV_END; > conv_123x_x321: as_u32(dst) = sx24(bswap_32(as_u32c(src))); goto CONV_END; > conv_123x_123x: as_u32(dst) = sx24s(as_u32c(src)); goto CONV_END; > conv_123x_3210: as_u32(dst) = bswap_32(as_u32c(src)) << 8; goto CONV_END; > -conv_123x_0123: as_u32(dst) = shift_down(as_u32c(src), 8); goto CONV_END; > -conv_123x_xxxB: as_u8(dst) = (shift_down(as_u32c(src), 8) & 0xff) ^ 0x80; goto CONV_END; > -conv_123x_xxB2: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 8) ^ 0x80); goto CONV_END; > -conv_123x_xx2B: as_u16(dst) = (shift_down(as_u32c(src), 8) & 0xffff) ^ 0x80; goto CONV_END; > +conv_123x_0123: as_u32(dst) = as_u32c(src) >> 8; goto CONV_END; > +conv_123x_xxxB: as_u8(dst) = ((as_u32c(src) >> 8) & 0xff) ^ 0x80; goto CONV_END; > +conv_123x_xxB2: as_u16(dst) = bswap_16((as_u32c(src) >> 8) ^ 0x80); goto CONV_END; > +conv_123x_xx2B: as_u16(dst) = ((as_u32c(src) >> 8) & 0xffff) ^ 0x80; goto CONV_END; > conv_123x_xB21: as_u32(dst) = sx24(bswap_32(as_u32c(src)) ^ 0x800000); goto CONV_END; > conv_123x_12Bx: as_u32(dst) = sx24s(as_u32c(src) ^ 0x8000); goto CONV_END; > conv_123x_B210: as_u32(dst) = bswap_32(as_u32c(src) ^ 0x8000) << 8; goto CONV_END; > -conv_123x_012B: as_u32(dst) = shift_down(as_u32c(src), 8) ^ 0x80; goto CONV_END; > -conv_1234_xxx1: as_u8(dst) = shift_down(as_u32c(src), 24); goto CONV_END; > -conv_1234_xx12: as_u16(dst) = shift_down(as_u32c(src), 16); goto CONV_END; > -conv_1234_xx21: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 16)); goto CONV_END; > -conv_1234_x123: as_u32(dst) = sx24(shift_down(as_u32c(src), 8)); goto CONV_END; > +conv_123x_012B: as_u32(dst) = (as_u32c(src) >> 8) ^ 0x80; goto CONV_END; > +conv_1234_xxx1: as_u8(dst) = as_u32c(src) >> 24; goto CONV_END; > +conv_1234_xx12: as_u16(dst) = as_u32c(src) >> 16; goto CONV_END; > +conv_1234_xx21: as_u16(dst) = bswap_16(as_u32c(src) >> 16); goto CONV_END; > +conv_1234_x123: as_u32(dst) = sx24(as_u32c(src) >> 8); goto CONV_END; > conv_1234_321x: as_u32(dst) = sx24s(bswap_32(as_u32c(src)) << 8); goto CONV_END; > conv_1234_1234: as_u32(dst) = as_u32c(src); goto CONV_END; > conv_1234_4321: as_u32(dst) = bswap_32(as_u32c(src)); goto CONV_END; > -conv_1234_xxx9: as_u8(dst) = shift_down(as_u32c(src), 24) ^ 0x80; goto CONV_END; > -conv_1234_xx92: as_u16(dst) = shift_down(as_u32c(src), 16) ^ 0x8000; goto CONV_END; > -conv_1234_xx29: as_u16(dst) = bswap_16(shift_down(as_u32c(src), 16)) ^ 0x80; goto CONV_END; > -conv_1234_x923: as_u32(dst) = sx24(shift_down(as_u32c(src), 8) ^ 0x800000); goto CONV_END; > +conv_1234_xxx9: as_u8(dst) = (as_u32c(src) >> 24) ^ 0x80; goto CONV_END; > +conv_1234_xx92: as_u16(dst) = (as_u32c(src) >> 16) ^ 0x8000; goto CONV_END; > +conv_1234_xx29: as_u16(dst) = bswap_16(as_u32c(src) >> 16) ^ 0x80; goto CONV_END; > +conv_1234_x923: as_u32(dst) = sx24((as_u32c(src) >> 8) ^ 0x800000); goto CONV_END; > conv_1234_329x: as_u32(dst) = sx24s((bswap_32(as_u32c(src)) ^ 0x80) << 8); goto CONV_END; > conv_1234_9234: as_u32(dst) = as_u32c(src) ^ 0x80000000; goto CONV_END; > conv_1234_4329: as_u32(dst) = bswap_32(as_u32c(src)) ^ 0x80; goto CONV_END; > conv_1234_xxx4: as_u8(dst) = as_u32c(src) & 0xff; goto CONV_END; > conv_1234_xx43: as_u16(dst) = bswap_16(as_u32c(src)); goto CONV_END; > conv_1234_xx34: as_u16(dst) = as_u32c(src) & 0xffff; goto CONV_END; > -conv_1234_x432: as_u32(dst) = sx24(shift_down(bswap_32(as_u32c(src)), 8)); goto CONV_END; > +conv_1234_x432: as_u32(dst) = sx24(bswap_32(as_u32c(src)) >> 8); goto CONV_END; > conv_1234_234x: as_u32(dst) = sx24s(as_u32c(src) << 8); goto CONV_END; > conv_1234_xxxC: as_u8(dst) = (as_u32c(src) & 0xff) ^ 0x80; goto CONV_END; > conv_1234_xxC3: as_u16(dst) = bswap_16(as_u32c(src) ^ 0x80); goto CONV_END; > conv_1234_xx3C: as_u16(dst) = (as_u32c(src) & 0xffff) ^ 0x80; goto CONV_END; > -conv_1234_xC32: as_u32(dst) = sx24(shift_down(bswap_32(as_u32c(src)), 8) ^ 0x800000); goto CONV_END; > +conv_1234_xC32: as_u32(dst) = sx24((bswap_32(as_u32c(src)) >> 8) ^ 0x800000); goto CONV_END; > conv_1234_23Cx: as_u32(dst) = sx24s((as_u32c(src) ^ 0x80) << 8); goto CONV_END; > conv_1234_C321: as_u32(dst) = bswap_32(as_u32c(src) ^ 0x80); goto CONV_END; > conv_1234_123C: as_u32(dst) = as_u32c(src) ^ 0x80; goto CONV_END; > @@ -384,26 +382,26 @@ get16_12_92: sample = as_u16c(src) ^ 0x8 > get16_12_92: sample = as_u16c(src) ^ 0x8000; goto GET16_END; > get16_12_21: sample = bswap_16(as_u16c(src)); goto GET16_END; > get16_12_A1: sample = bswap_16(as_u16c(src) ^ 0x80); goto GET16_END; > -get16_0123_12: sample = shift_down(as_u32c(src), 8); goto GET16_END; > -get16_0123_92: sample = shift_down(as_u32c(src), 8) ^ 0x8000; goto GET16_END; > -get16_1230_32: sample = bswap_16(shift_down(as_u32c(src), 8)); goto GET16_END; > -get16_1230_B2: sample = bswap_16(shift_down(as_u32c(src), 8) ^ 0x80); goto GET16_END; > -get16_1234_12: sample = shift_down(as_u32c(src), 16); goto GET16_END; > -get16_1234_92: sample = shift_down(as_u32c(src), 16) ^ 0x8000; goto GET16_END; > +get16_0123_12: sample = as_u32c(src) >> 8; goto GET16_END; > +get16_0123_92: sample = (as_u32c(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_1230_32: sample = bswap_16(as_u32c(src) >> 8); goto GET16_END; > +get16_1230_B2: sample = bswap_16((as_u32c(src) >> 8) ^ 0x80); goto GET16_END; > +get16_1234_12: sample = as_u32c(src) >> 16; goto GET16_END; > +get16_1234_92: sample = (as_u32c(src) >> 16) ^ 0x8000; goto GET16_END; > get16_1234_43: sample = bswap_16(as_u32c(src)); goto GET16_END; > get16_1234_C3: sample = bswap_16(as_u32c(src) ^ 0x80); goto GET16_END; > -get16_123_12: sample = shift_down(_get_triple(src), 8); goto GET16_END; > -get16_123_92: sample = shift_down(_get_triple(src), 8) ^ 0x8000; goto GET16_END; > -get16_123_32: sample = shift_down(_get_triple_s(src), 8); goto GET16_END; > -get16_123_B2: sample = shift_down(_get_triple_s(src), 8) ^ 0x8000; goto GET16_END; > -get16_123_12_20: sample = shift_down(_get_triple(src), 4); goto GET16_END; > -get16_123_92_20: sample = shift_down(_get_triple(src), 4) ^ 0x8000; goto GET16_END; > -get16_123_32_20: sample = shift_down(_get_triple_s(src), 4); goto GET16_END; > -get16_123_B2_20: sample = shift_down(_get_triple_s(src), 4) ^ 0x8000; goto GET16_END; > -get16_123_12_18: sample = shift_down(_get_triple(src), 2); goto GET16_END; > -get16_123_92_18: sample = shift_down(_get_triple(src), 2) ^ 0x8000; goto GET16_END; > -get16_123_32_18: sample = shift_down(_get_triple_s(src), 2); goto GET16_END; > -get16_123_B2_18: sample = shift_down(_get_triple_s(src), 2) ^ 0x8000; goto GET16_END; > +get16_123_12: sample = _get_triple(src) >> 8; goto GET16_END; > +get16_123_92: sample = (_get_triple(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_123_32: sample = _get_triple_s(src) >> 8; goto GET16_END; > +get16_123_B2: sample = (_get_triple_s(src) >> 8) ^ 0x8000; goto GET16_END; > +get16_123_12_20: sample = _get_triple(src) >> 4; goto GET16_END; > +get16_123_92_20: sample = (_get_triple(src) >> 4) ^ 0x8000; goto GET16_END; > +get16_123_32_20: sample = _get_triple_s(src) >> 4; goto GET16_END; > +get16_123_B2_20: sample = (_get_triple_s(src) >> 4) ^ 0x8000; goto GET16_END; > +get16_123_12_18: sample = _get_triple(src) >> 2; goto GET16_END; > +get16_123_92_18: sample = (_get_triple(src) >> 2) ^ 0x8000; goto GET16_END; > +get16_123_32_18: sample = _get_triple_s(src) >> 2; goto GET16_END; > +get16_123_B2_18: sample = (_get_triple_s(src) >> 2) ^ 0x8000; goto GET16_END; > } > #endif > > @@ -431,8 +429,8 @@ static void *put16_labels[4 * 2 * 2] = { > > #ifdef PUT16_END > while (0) { > -put16_12_1: as_u8(dst) = shift_down(sample, 8); goto PUT16_END; > -put16_12_9: as_u8(dst) = shift_down(sample, 8) ^ 0x80; goto PUT16_END; > +put16_12_1: as_u8(dst) = sample >> 8; goto PUT16_END; > +put16_12_9: as_u8(dst) = (sample >> 8) ^ 0x80; goto PUT16_END; > put16_12_12: as_u16(dst) = sample; goto PUT16_END; > put16_12_92: as_u16(dst) = sample ^ 0x8000; goto PUT16_END; > put16_12_21: as_u16(dst) = bswap_16(sample); goto PUT16_END; > @@ -502,8 +500,8 @@ get32_12_A100: sample = (u_int32_t)bswap > get32_12_A100: sample = (u_int32_t)bswap_16(as_u16c(src) ^ 0x80) << 16; goto GET32_END; > get32_0123_1230: sample = as_u32c(src) << 8; goto GET32_END; > get32_0123_9230: sample = (as_u32c(src) << 8) ^ 0x80000000; goto GET32_END; > -get32_1230_3210: sample = bswap_32(shift_down(as_u32c(src), 8)); goto GET32_END; > -get32_1230_B210: sample = bswap_32(shift_down(as_u32c(src), 8) ^ 0x80); goto GET32_END; > +get32_1230_3210: sample = bswap_32(as_u32c(src) >> 8); goto GET32_END; > +get32_1230_B210: sample = bswap_32((as_u32c(src) >> 8) ^ 0x80); goto GET32_END; > get32_1234_1234: sample = as_u32c(src); goto GET32_END; > get32_1234_9234: sample = as_u32c(src) ^ 0x80000000; goto GET32_END; > get32_1234_4321: sample = bswap_32(as_u32c(src)); goto GET32_END; > @@ -570,32 +568,32 @@ static void *put32_labels[4 * 2 * 2 + 4 > > #ifdef PUT32_END > while (0) { > -put32_1234_1: as_u8(dst) = shift_down(sample, 24); goto PUT32_END; > -put32_1234_9: as_u8(dst) = shift_down(sample, 24) ^ 0x80; goto PUT32_END; > -put32_1234_12: as_u16(dst) = shift_down(sample, 16); goto PUT32_END; > -put32_1234_92: as_u16(dst) = shift_down(sample, 16) ^ 0x8000; goto PUT32_END; > -put32_1234_21: as_u16(dst) = bswap_16(shift_down(sample, 16)); goto PUT32_END; > -put32_1234_29: as_u16(dst) = bswap_16(shift_down(sample, 16)) ^ 0x80; goto PUT32_END; > -put32_1234_0123: as_u32(dst) = sx24(shift_down(sample, 8)); goto PUT32_END; > -put32_1234_0923: as_u32(dst) = sx24(shift_down(sample, 8) ^ 0x800000); goto PUT32_END; > +put32_1234_1: as_u8(dst) = sample >> 24; goto PUT32_END; > +put32_1234_9: as_u8(dst) = (sample >> 24) ^ 0x80; goto PUT32_END; > +put32_1234_12: as_u16(dst) = sample >> 16; goto PUT32_END; > +put32_1234_92: as_u16(dst) = (sample >> 16) ^ 0x8000; goto PUT32_END; > +put32_1234_21: as_u16(dst) = bswap_16(sample >> 16); goto PUT32_END; > +put32_1234_29: as_u16(dst) = bswap_16(sample >> 16) ^ 0x80; goto PUT32_END; > +put32_1234_0123: as_u32(dst) = sx24(sample >> 8); goto PUT32_END; > +put32_1234_0923: as_u32(dst) = sx24((sample >> 8) ^ 0x800000); goto PUT32_END; > put32_1234_3210: as_u32(dst) = sx24s(bswap_32(sample) << 8); goto PUT32_END; > put32_1234_3290: as_u32(dst) = sx24s((bswap_32(sample) ^ 0x80) << 8); goto PUT32_END; > put32_1234_1234: as_u32(dst) = sample; goto PUT32_END; > put32_1234_9234: as_u32(dst) = sample ^ 0x80000000; goto PUT32_END; > put32_1234_4321: as_u32(dst) = bswap_32(sample); goto PUT32_END; > put32_1234_4329: as_u32(dst) = bswap_32(sample) ^ 0x80; goto PUT32_END; > -put32_1234_123: _put_triple(dst, shift_down(sample, 8)); goto PUT32_END; > -put32_1234_923: _put_triple(dst, shift_down(sample ^ 0x80000000, 8)); goto PUT32_END; > -put32_1234_321: _put_triple_s(dst, shift_down(sample, 8)); goto PUT32_END; > -put32_1234_329: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 8)); goto PUT32_END; > -put32_1234_123_20: _put_triple(dst, shift_down(sample, 12)); goto PUT32_END; > -put32_1234_923_20: _put_triple(dst, shift_down(sample ^ 0x80000000, 12)); goto PUT32_END; > -put32_1234_321_20: _put_triple_s(dst, shift_down(sample, 12)); goto PUT32_END; > -put32_1234_329_20: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 12)); goto PUT32_END; > -put32_1234_123_18: _put_triple(dst, shift_down(sample, 14)); goto PUT32_END; > -put32_1234_923_18: _put_triple(dst, shift_down(sample ^ 0x80000000, 14)); goto PUT32_END; > -put32_1234_321_18: _put_triple_s(dst, shift_down(sample, 14)); goto PUT32_END; > -put32_1234_329_18: _put_triple_s(dst, shift_down(sample ^ 0x80000000, 14)); goto PUT32_END; > +put32_1234_123: _put_triple(dst, sample >> 8); goto PUT32_END; > +put32_1234_923: _put_triple(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; > +put32_1234_321: _put_triple_s(dst, sample >> 8); goto PUT32_END; > +put32_1234_329: _put_triple_s(dst, (sample ^ 0x80000000) >> 8); goto PUT32_END; > +put32_1234_123_20: _put_triple(dst, sample >> 12); goto PUT32_END; > +put32_1234_923_20: _put_triple(dst, (sample ^ 0x80000000) >> 12); goto PUT32_END; > +put32_1234_321_20: _put_triple_s(dst, sample >> 12); goto PUT32_END; > +put32_1234_329_20: _put_triple_s(dst, (sample ^ 0x80000000) >> 12); goto PUT32_END; > +put32_1234_123_18: _put_triple(dst, sample >> 14); goto PUT32_END; > +put32_1234_923_18: _put_triple(dst, (sample ^ 0x80000000) >> 14); goto PUT32_END; > +put32_1234_321_18: _put_triple_s(dst, sample >> 14); goto PUT32_END; > +put32_1234_329_18: _put_triple_s(dst, (sample ^ 0x80000000) >> 14); goto PUT32_END; > } > #endif > > @@ -1081,4 +1079,3 @@ norms_32_s32s: _norms(src, dst, 32, 1, > #undef _put_triple_le > #undef _put_triple_be > > -#undef shift_down -- Excellent day for drinking heavily. Spike the office water cooler;-)

2 1

[alsa-devel] intel-hda patch for sigmatel codec volume knob
by zhejiang 15 Oct '07

15 Oct '07

Hi, I found that STAC_VOLKNOB hardwired the KNOB nid to 0x24. It is okay for stac9205 and stac927x. But the VolumeKnob nid of stac9220-9221 is 0x16. diff -Nur a/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c b/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c --- a/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-21 16:41:28.000000000 +0800 +++ b/alsa-driver-1.0.15rc3/alsa-kernel/pci/hda/patch_sigmatel.c 2007-10-15 15:13:14.000000000 +0800 @@ -351,7 +351,7 @@ static int stac92xx_volknob_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - ucontrol->value.integer.value[0] = kcontrol->private_value; + ucontrol->value.integer.value[0] = kcontrol->private_value & 0xFFFF; return 0; } @@ -360,14 +360,15 @@ { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - if (kcontrol->private_value == ucontrol->value.integer.value[0]) + if ((kcontrol->private_value & 0xFFFF) == ucontrol->value.integer.value[0]) return 0; - kcontrol->private_value = ucontrol->value.integer.value[0]; + kcontrol->private_value &= 0xFFFF0000; + kcontrol->private_value += ucontrol->value.integer.value[0]; - snd_hda_codec_write_cache(codec, 0x24, 0, + snd_hda_codec_write_cache(codec, kcontrol->private_value >> 16, 0, AC_VERB_SET_VOLUME_KNOB_CONTROL, - kcontrol->private_value | 0x80); + (kcontrol->private_value & 0xFFFF) | 0x80); return 1; } @@ -433,7 +434,7 @@ .private_value = verb_read | (verb_write << 16), \ } -#define STAC_VOLKNOB \ +#define STAC_VOLKNOB(knob_nid) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Master Playback Volume", \ @@ -441,7 +442,7 @@ .info = stac92xx_volknob_info, \ .get = stac92xx_volknob_get, \ .put = stac92xx_volknob_put, \ - .private_value = 127, \ + .private_value = 127 | (knob_nid << 16), \ } @@ -474,7 +475,7 @@ }, STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), - STAC_VOLKNOB, + STAC_VOLKNOB(0x24), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -490,7 +491,7 @@ /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac922x_mixer[] = { STAC_INPUT_SOURCE(2), - STAC_VOLKNOB, + STAC_VOLKNOB(0x16), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x17, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Mux Capture Volume", 0x0, 0x12, 0x0, HDA_OUTPUT), @@ -504,7 +505,7 @@ static struct snd_kcontrol_new stac927x_mixer[] = { STAC_INPUT_SOURCE(3), - STAC_VOLKNOB, + STAC_VOLKNOB(0x24), STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT),

3 2

[alsa-devel] Fwd: FSC Esprimo Mobile V5505 - info, partly working
by Jan-Simon Möller 13 Oct '07

13 Oct '07

Hi ! I got a new FSC Esprimo Mobile V5505 (full system info attached). When i use "options snd-hda-intel model=fujitsu-c1410 enable=1 index=0" sound works, but it sounds not good and smooth. Chip is a conexant afaik. Attachements: * Full lspci -vv -nn Whats needed to improve sound and make the mixer available ? If any other data is needed -> please ping ! Sincerely yours Jan-Simon Möller dl9pf(a)gmx.de

2 3

[alsa-devel] Patch for Intel-HDA ALC262 on FSC Lifebook C1410
by Veli-Matti Lintu 12 Oct '07

12 Oct '07

Hello, After error and trial I got sound working on Fujitsu-Siemens Computers Lifebook C1410. Attached is a patch that works for me, but probably needs some cleaning. The laptop uses an ALC262 chip that is also used on other FSC models, but the needed settings seem to be somewhat different from the existing Fujitsu-settings in patch_realtek.c. If I understand correctly, the chips use same subsystem ids, so the patch does not automatically detect the correct chipset. Maybe someone more knowledgeable can help with this? The patch names the internal mic as Int Mic and external mic that can be plugged in as Ext Mic. There's some more background information in Ubuntu's Launchpad: https://bugs.launchpad.net/ubuntu/+source/alsa-driver/+bug/125790 The patch is tested with kernel 2.6.22.9. I am not really familiar with alsa development, so I cannot say which version of the alsa drivers that kernel version uses. The environment where the testing was done is running Ubuntu 7.04 with alsa version 1.0.13. Only updated component is the kernel. The patch needs model=fujitsu-c1410 parameter to be passed to the module: /etc/modprobe.d/alsa-base: options snd-hda-intel model=fujitsu-c1410 The subsystem 10cf:1397 is mapped in patch_realtek.c alc262_cfg_tbl to ALC262_FUJITSU and ALC262_FUJITSU_C1410 would need to be mapped to same ids: ------------------------------------------------------------------- sound/pci/hda/patch_realtek.c: SND_PCI_QUIRK(0x10cf, 0x1397, "Fujitsu", ALC262_FUJITSU), ------------------------------------------------------------------- lspci -nv (on C1410) 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:1397 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) ------------------------------------------------------------------- I hope someone with more knowledge about alsa drivers could have a look at this. The patch has been in internal use for a few weeks and people seem to be happy with it, so I hope also others can get their sound working. The patch and hw information follow: --------------------------------------------------------------------------------------------- --- linux-2.6.22.9/sound/pci/hda/patch_realtek.c.orig 2007-09-26 21:03:01.000000000 +0300 +++ linux-2.6.22.9/sound/pci/hda/patch_realtek.c 2007-10-07 13:28:12.000000000 +0300 @@ -89,6 +89,7 @@ ALC262_HIPPO, ALC262_HIPPO_1, ALC262_FUJITSU, + ALC262_FUJITSU_C1410, ALC262_HP_BPC, ALC262_HP_BPC_D7000_WL, ALC262_HP_BPC_D7000_WF, @@ -7068,6 +7069,15 @@ }, }; +static struct hda_input_mux alc262_fujitsu_c1410_capture_source = { + .num_items = 3, + .items = { + { "Ext Mic", 0x0 }, + { "Int Mic", 0x1 }, + { "CD", 0x4 }, + }, +}; + static struct hda_input_mux alc262_HP_capture_source = { .num_items = 5, .items = { @@ -7182,6 +7192,43 @@ { } /* end */ }; +static struct snd_kcontrol_new alc262_fujitsu_c1410_mixer[] = { + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .info = snd_hda_mixer_amp_volume_info, + .get = snd_hda_mixer_amp_volume_get, + .put = alc262_fujitsu_master_vol_put, + .tlv = { .c = snd_hda_mixer_amp_tlv }, + .private_value = HDA_COMPOSE_AMP_VAL(0x0c, 3, 0, HDA_OUTPUT), + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Switch", + .info = snd_hda_mixer_amp_switch_info, + .get = snd_hda_mixer_amp_switch_get, + .put = alc262_fujitsu_master_sw_put, + .private_value = HDA_COMPOSE_AMP_VAL(0x0d, 3, 0, HDA_OUTPUT), + }, + + HDA_CODEC_VOLUME("Headphones Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Headphones Playback Switch", 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Speaker Playback Volume", 0x0d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE("Speaker Playback Switch", 0x15, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Ext Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Boost", 0x19, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Ext Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Ext Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Int Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Int Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Playback Volume", 0x0b, 0x04, HDA_INPUT), + HDA_CODEC_MUTE("CD Playback Switch", 0x0b, 0x04, HDA_INPUT), + { } /* end */ +}; + /* additional init verbs for Benq laptops */ static struct hda_verb alc262_EAPD_verbs[] = { {0x20, AC_VERB_SET_COEF_INDEX, 0x07}, @@ -7581,6 +7628,7 @@ [ALC262_HIPPO] = "hippo", [ALC262_HIPPO_1] = "hippo_1", [ALC262_FUJITSU] = "fujitsu", + [ALC262_FUJITSU_C1410] = "fujitsu-c1410", [ALC262_HP_BPC] = "hp-bpc", [ALC262_HP_BPC_D7000_WL]= "hp-bpc-d7000", [ALC262_BENQ_ED8] = "benq", @@ -7659,6 +7707,18 @@ .input_mux = &alc262_fujitsu_capture_source, .unsol_event = alc262_fujitsu_unsol_event, }, + [ALC262_FUJITSU_C1410] = { + .mixers = { alc262_fujitsu_c1410_mixer }, + .init_verbs = { alc262_init_verbs, alc262_EAPD_verbs, alc262_fujitsu_unsol_verbs }, + .num_dacs = ARRAY_SIZE(alc262_dac_nids), + .dac_nids = alc262_dac_nids, + .hp_nid = 0x03, + .dig_out_nid = ALC262_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc262_modes), + .channel_mode = alc262_modes, + .input_mux = &alc262_fujitsu_c1410_capture_source, + .unsol_event = alc262_fujitsu_unsol_event, + }, [ALC262_HP_BPC] = { .mixers = { alc262_HP_BPC_mixer }, .init_verbs = { alc262_HP_BPC_init_verbs }, --------------------------------------------------------------------------------------------- ALSA Audio Debug v0.1.0 - su 7.10.2007 15:13:09 +0300 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux oppilas 2.6.22.9-c1410 #2 SMP Sun Oct 7 11:57:15 EEST 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 257560 1 snd_pcm_oss 44928 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80772 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33408 0 snd_seq_midi 9600 0 snd_rawmidi 25856 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53360 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24196 2 snd_pcm,snd_seq snd_seq_device 9356 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54916 11 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf0640000 irq 22 2: : timer 3: : sequencer 4: [ 0- 6]: digital audio playback 5: [ 0- 6]: digital audio capture 6: [ 0- 2]: digital audio capture 7: [ 0- 1]: digital audio playback 8: [ 0- 0]: digital audio playback 9: [ 0- 0]: digital audio capture 10: [ 0] : control 00-06: Si3054 Modem : Si3054 Modem : playback 1 : capture 1 00-02: ALC262 Analog : ALC262 Analog : capture 2 00-01: ALC262 Digital : ALC262 Digital : playback 1 00-00: ALC262 Analog : ALC262 Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0p pcmC0D2c pcmC0D6p timer pcmC0D0c pcmC0D1p pcmC0D6c seq CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 CPU T5600 @ 1.83GHz cpu MHz : 1000.000 model name : Intel(R) Core(TM)2 CPU T5600 @ 1.83GHz cpu MHz : 1000.000 RAM ------------------------------------------------------- MemTotal: 1026252 kB SwapTotal: 2441872 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS/940GML and 945GT Express Memory Controller Hub (rev 03) 00:02.0 VGA compatible controller: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03) 00:02.1 Display controller: Intel Corporation Mobile 945GM/GMS/940GML Express Integrated Graphics Controller (rev 03) 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 02) 00:1c.1 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 2 (rev 02) 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #3 (rev 02) 00:1d.3 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI #4 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801G (ICH7 Family) USB2 EHCI Controller (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev e2) 00:1f.0 ISA bridge: Intel Corporation 82801GBM (ICH7-M) LPC Interface Bridge (rev 02) 00:1f.1 IDE interface: Intel Corporation 82801G (ICH7 Family) IDE Controller (rev 02) 00:1f.2 SATA controller: Intel Corporation 82801GBM/GHM (ICH7 Family) Serial ATA Storage Controller AHCI (rev 02) 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 02) 02:00.0 Ethernet controller: Marvell Technology Group Ltd. 88E8055 PCI-E Gigabit Ethernet Controller (rev 12) 05:00.0 Network controller: Intel Corporation PRO/Wireless 3945ABG Network Connection (rev 02) 08:03.0 CardBus bridge: Ricoh Co Ltd RL5c476 II (rev b3) 08:03.1 FireWire (IEEE 1394): Ricoh Co Ltd R5C552 IEEE 1394 Controller (rev 08) 08:03.2 Generic system peripheral [0805]: Ricoh Co Ltd R5C822 SD/SDIO/MMC/MS/MSPro Host Adapter (rev 17) 00:00.0 0600: 8086:27a0 (rev 03) Subsystem: 10cf:1378 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:02.0 0300: 8086:27a2 (rev 03) (prog-if 00 [VGA]) Subsystem: 10cf:1381 Flags: bus master, fast devsel, latency 0, IRQ 19 Memory at f0300000 (32-bit, non-prefetchable) [size=512K] I/O ports at 1800 [size=8] Memory at e0000000 (32-bit, prefetchable) [size=256M] Memory at f0400000 (32-bit, non-prefetchable) [size=256K] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [d0] Power Management version 2 00:02.1 0380: 8086:27a6 (rev 03) Subsystem: 10cf:1381 Flags: bus master, fast devsel, latency 0 Memory at f0380000 (32-bit, non-prefetchable) [size=512K] Capabilities: [d0] Power Management version 2 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 10cf:1397 Flags: bus master, fast devsel, latency 0, IRQ 22 Memory at f0640000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) 00:1c.0 0604: 8086:27d0 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f0000000-f00fffff Prefetchable memory behind bridge: 0000000054000000-00000000540fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.1 0604: 8086:27d2 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=04, sec-latency=0 Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.2 0604: 8086:27d4 (rev 02) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=05, subordinate=07, sec-latency=0 Memory behind bridge: 54100000-541fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 10cf:138b Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1d.0 0c03: 8086:27c8 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 1820 [size=32] 00:1d.1 0c03: 8086:27c9 (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1840 [size=32] 00:1d.2 0c03: 8086:27ca (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 1860 [size=32] 00:1d.3 0c03: 8086:27cb (rev 02) (prog-if 00 [UHCI]) Subsystem: 10cf:1389 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1880 [size=32] 00:1d.7 0c03: 8086:27cc (rev 02) (prog-if 20 [EHCI]) Subsystem: 10cf:138a Flags: bus master, medium devsel, latency 0, IRQ 20 Memory at f0644000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:2448 (rev e2) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=08, subordinate=0c, sec-latency=32 I/O behind bridge: 00003000-00003fff Memory behind bridge: f0200000-f02fffff Prefetchable memory behind bridge: 0000000050000000-0000000053ffffff Capabilities: [50] Subsystem: 10cf:1383 00:1f.0 0601: 8086:27b9 (rev 02) Subsystem: 10cf:1384 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:1f.1 0101: 8086:27df (rev 02) (prog-if 8a [Master SecP PriP]) Subsystem: 10cf:1385 Flags: bus master, medium devsel, latency 0, IRQ 21 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] 00:1f.2 0106: 8086:27c5 (rev 02) (prog-if 01 [AHCI 1.0]) Subsystem: 10cf:1387 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 220 I/O ports at 18d0 [size=8] I/O ports at 18c4 [size=4] I/O ports at 18c8 [size=8] I/O ports at 18c0 [size=4] I/O ports at 18b0 [size=16] Memory at f0644400 (32-bit, non-prefetchable) [size=1K] Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [70] Power Management version 2 00:1f.3 0c05: 8086:27da (rev 02) Subsystem: 10cf:1388 Flags: medium devsel, IRQ 11 I/O ports at 18e0 [size=32] 02:00.0 0200: 11ab:4363 (rev 12) Subsystem: 10cf:139a Flags: bus master, fast devsel, latency 0, IRQ 219 Memory at f0000000 (64-bit, non-prefetchable) [size=16K] I/O ports at 2000 [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [48] Power Management version 3 Capabilities: [50] Vital Product Data Capabilities: [5c] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable+ Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting 05:00.0 0280: 8086:4222 (rev 02) Subsystem: 8086:1001 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 54100000 (32-bit, non-prefetchable) [size=4K] Capabilities: [c8] Power Management version 2 Capabilities: [d0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting Capabilities: [140] Device Serial Number b5-ec-71-ff-ff-77-1b-00 08:03.0 0607: 1180:0476 (rev b3) Subsystem: 10cf:1254 Flags: bus master, medium devsel, latency 168, IRQ 19 Memory at f0201000 (32-bit, non-prefetchable) [size=4K] Bus: primary=08, secondary=09, subordinate=0c, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00003000-000030ff I/O window 1: 00003400-000034ff 16-bit legacy interface ports at 0001 08:03.1 0c00: 1180:0552 (rev 08) (prog-if 10 [OHCI]) Subsystem: 10cf:1255 Flags: bus master, medium devsel, latency 32, IRQ 22 Memory at f0200000 (32-bit, non-prefetchable) [size=2K] Capabilities: [dc] Power Management version 2 08:03.2 0805: 1180:0822 (rev 17) Subsystem: 10cf:1323 Flags: bus master, medium devsel, latency 32, IRQ 19 Memory at f0200800 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 --------------------------------------------------------------------------------------------- Veli-Matti Lintu

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[alsa-devel] snd-hda-intel: no sound on ASUS F3Sa
by Luca Tettamanti 11 Oct '07

11 Oct '07

Hi, I can't enable sound on a (new) ASUS F3Sa laptop with this sound device: 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: ASUSTeK Computer Inc. Unknown device 1339 Control: I/O- Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Interrupt: pin A routed to IRQ 22 Region 0: Memory at febf8000 (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 Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Codec: kronos@nb:~$ cat /proc/asound/card0/codec#0 | head -n1 Codec: Realtek ALC660-VD (full output attached) without arguments I get: ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 codec_mask = 0x3 hda_codec: model '3stack-660' is selected for config 1043:1339 (Asus G1) while with model=auto: ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 codec_mask = 0x3 hda_codec: model 'auto' is selected autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hp_outs=1 (0x1b/0x0/0x0/0x0/0x0) inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 I also tried all other possible models without success; I can't get any output on the built-in speakers nor on the headphones. I tried both 2.6.23-git (from a couple of days ago) and ALSA 1.0.15rc3. Luca (CC me, I'm not subscribed) -- "Sei l'unica donna della mia vita". (Adamo)

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[alsa-devel] How to enforce 32bit access?
by Joachim Foerster 11 Oct '07

11 Oct '07

Hi ALSA devs, (based on a Xilinx ML403, PPC405, OPB bus) I have a custom sound controller, which allows 32bit write accesses to its IO memory, only. The controller supports S16_LE, S16_BE, U16_LE, U16_BE with min. and max. 2 channels (stereo), only! => So a frame has to be 32bits wide and one frame has to be written at one point of time (= one 32bit bus access). Now, playing a 16kHz, S16_LE, Mono .wav-file with aplay invokes the conversion framework of the alsa-lib. The following line of code (of pcm_plugin.h) is executed: conv_xx12_xx12: as_u16(dst) = as_u16c(src); This is a 16bit access and thus fails with a Bus error/Kernel Oops. Therefore my question is: In my driver for the controller, how do I tell ALSA, that it is only allowed to write in 32bit quantities? Is this possible at all? Or do I have to use a special .asoundrc file? Thanks, Joachim PS: 16bit, signed/unsigned, LE/BE, stereo (!) .wav files can be played without any problem.

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[alsa-devel] alsa 1.0.15
by paul blakeley 11 Oct '07

11 Oct '07

Can someone please tell me when version 1.0.15 will be released and then become the current stable package?

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[alsa-devel] dont know if bug in /tmp/217usbkeyboardbug.log
by linuxcbon 11 Oct '07

11 Oct '07

hi, I have no idea if this is a bug : tv card is hauppauge wintv model 44346, tuner model is Philips FI1216MF MK2 I found in /tmp/217usbkeyboardbug.log Last 20 lines of dmesg: tveeprom 2-0050: has radio bttv0: Hauppauge eeprom indicates model#44346 bttv0: using tuner=3 bttv0: i2c: checking for MSP34xx @ 0x80... found msp3400 2-0040: MSP3415G-B8 found @ 0x80 (bt878 #0 [sw]) msp3400 2-0040: MSP3415G-B8 supports nicam and radio, mode is autodetect and autoselect bttv0: i2c: checking for TDA9875 @ 0xb0... not found bttv0: i2c: checking for TDA7432 @ 0x8a... not found bttv0: i2c: checking for TDA9887 @ 0x86... not found tuner 2-0061: chip found @ 0xc2 (bt878 #0 [sw]) tuner 2-0061: type set to 3 (Philips (SECAM+PAL_BG) (FI1216MF, FM1216MF, FR1216MF)) tuner 2-0061: type set to 3 (Philips (SECAM+PAL_BG) (FI1216MF, FM1216MF, FR1216MF)) bttv0: registered device video0 bttv0: registered device vbi0 bttv0: registered device radio0 bttv0: PLL: 28636363 => 35468950 .. ok ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:872: unknown card 0x878-0x70:0x13eb ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:875: please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> ACPI: PCI Interrupt 0000:01:04.1[A] -> Link [LNK1] -> GSI 5 (level, low) -> IRQ 5 ALSA /mnt/hda3/sources/sources116/alsa-driver-1.0.15rc2/pci/bt87x.c:943: bt87x0: Using board 0, analog, digital (rate 32000 Hz) I hope this can help. regards _____________________________________________________________________________ Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail

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[alsa-devel] vaio_mux's num_items is not equal to the actual items
by zhejiang 11 Oct '07

11 Oct '07

In patch_sigmatel.c, static struct hda_input_mux vaio_mux = { .num_items = 2, .items = { /* { "HP", 0x0 }, */ { "Mic Jack", 0x1 }, { "Internal Mic", 0x2 }, { "PCM", 0x3 }, } }; Why is the num_items not 3 ? The "PCM" won't be selected with num_times = 2.

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[alsa-devel] Creative Labs XFi drivers.
by Tobin Davis 11 Oct '07

11 Oct '07

Creative Labs has released a proprietary driver base for their XFi cards. I downloaded it because it claims to have compatibility with alsa. In fact, it requires alsa include files to be able to compile. I haven't looked at the source yet (this information is all in the Readme.txt in the main download) and probably won't (especially since I don't own a Creative Labs XFi card). What I don't understand, is are they allowed to use our code to build their drivers? They use alsa core modules to interoperate (from what I can tell in the Readme.txt). Also, this line in the License.txt is especially troubling: > In the event that the Software includes Linux code or drivers, then, unless expressly stated to the contrary, such Linux > code and drivers are proprietary to Creative and all rights therein are reserved. I'm deleting all but the big tarball I downloaded. I'll delete it later. -- Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Alsa 1.0.15rc2 and oops with Dell M1210
by zhejiang 10 Oct '07

10 Oct '07

Seems that the spec->input_mux contains no item at all. So system maybe crash in the snd_hda_input_mux_info() when try to copy the items[].label,which is null pointer. May you replace the function stac92xx_mux_enum_info() in alsa-driver-1.0.15rc2/alsa-kernel/pci/hda/patch_sigmatel.c ? static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; printk("spec->input_mux->num_items is %d \n",spec->input_mux->num_items); if(!spec->input_mux->num_items) return 0; return snd_hda_input_mux_info(spec->input_mux, uinfo); } Does it still crash? What's the number of num_items? The pin_config for M1210 didn't include any input pin. I wonder why.

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Re: [alsa-devel] chmod with USB sound device?
by william estrada 10 Oct '07

10 Oct '07

Harald, I revisited /etc/security/console.perms.d/50-default.perms and got it to work. For the list, I changed: In /etc/security/console.perms.d/50-default.perms > # permission definitions . > <console> 0666 <sound> 0666 root And In /etc/udev/rules.d/50-udev.rules > KERNEL=="snd/*", MODE="0666", OWNER="wre00:root" Then I did the udevtrigger. Then i did the udevtrigger, plugged in my USB sound device and got this: > [root@laptop wre00]# ls -lrt /dev/snd > total 0 > crw-rw-rw- 1 wre00 root 116, 2 Oct 10 09:45 timer > crw-rw-rw- 1 wre00 root 116, 3 Oct 10 09:45 seq > crw-rw-rw- 1 wre00 root 116, 4 Oct 10 09:45 pcmC0D0p > crw-rw-rw- 1 wre00 root 116, 5 Oct 10 09:45 pcmC0D0c > crw-rw-rw- 1 wre00 root 116, 6 Oct 10 09:45 controlC0 > crw-rw-rw- 1 wre00 root 116, 7 Oct 10 10:28 pcmC1D0p > crw-rw-rw- 1 wre00 root 116, 8 Oct 10 10:28 pcmC1D0c > crw-rw-rw- 1 wre00 root 116, 9 Oct 10 10:28 controlC1 I guess both changes are required but it now works. Thanks for your time. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Harald Hoyer wrote: > william estrada wrote: >> Hi guys, >> >> I'm trying to run 'chmod' on a USB sound card using 'Hot Plug' udev >> support. >> From reading, I think I need to update my >> '/etc/udev/rules.d/50-udev.rules'? >> I have made these changes: >>> # audio devices . >>> KERNEL=="snd/*", MODE="0666" > > You may have a look in pam_console... > /etc/security/console.perms > /etc/security/console.perms.d/* >

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[alsa-devel] snd-hdsp & digiface
by Fernando Lopez-Lezcano 10 Oct '07

10 Oct '07

Hi... We are using both digiface and multiface interfaces here at ccrma and we have a (small) problem of consistency in the channel numbering because of the way they are connected. Both cards are connected digitally to digital mixers, but in the case of the multiface the digital channels (which would be the ones users use) are 9-16 while on the digiface the first eight are 1-8[*]. Is there a way to change the ordering in the multiface so that the digital interface is presented as ports 1-8 instead of 9-16? (even a hack to the driver would be welcome) - and preferably one that does not need the default interface to be changed from hw:0 in jack... -- Fernando [*] accounts are shared so any configuration you do on one machine for 1-8 will be wrong for the other where the real outputs are 9-16.

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Re: [alsa-devel] different user causes 'no such device'
by william estrada 10 Oct '07

10 Oct '07

Thanks, I did a 'chmod -R 666 /dev/snd/' and it worked. Now how do I make the hot plugged USB sound device use these setting?? Thanks for your time. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Takashi Iwai wrote: > At Wed, 03 Oct 2007 12:55:27 -0700, > william estrada wrote: >> Hi guys, >> >> I have what seams like a interesting problem. I have written an ALSA app >> to record wav files and it runs while I am logged in an terminal as the >> logged in Gnome user or root. But when I su to another user, it can't >> find the default sound card. >> >> I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did >> not work. >> >> Anyone know of this problem and a fix for it. >> >> Thanks for your time. >> >> -- >> William Estrada >> MrUmunhum(a)popdial.com >> Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) >> Ymessenger: MrUmunhum >> >>> MSGrec - Version 1.1, Mt-Umunhum-Wireless.net >>> WavMSG: Paused: enter 'q' to stop, <ENTER> to start. >>> ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device >>> ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device >>> ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device >>> ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device >>> ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default > > ALSA-native apps access only ALSA device files, usually /dev/snd/*, > not /dev/dsp*, those for OSS. In this case, it should be > /dev/snd/controlC* files. Check the premission of these files. > > > Takashi >

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[alsa-devel] Add quirk support for Dell XPS1330 and Inspiron 1420.
by Tim Gardner 10 Oct '07

10 Oct '07

-- Tim Gardner tim.gardner(a)ubuntu.com

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[alsa-devel] [PATCH] stac9200 eapd fixes
by Brian Hinz 10 Oct '07

10 Oct '07

Hi, The following patch addresses alsa bug #0002948 and it's (many) duplicates. Please consider it for inclusion in the next release. Thanks, -brian --- alsa-kernel/pci/hda/hda_codec.c 2007-09-24 21:46:11.000000000 -0400 +++ alsa-kernel/pci/hda/hda_codec.c 2007-09-24 22:00:15.000000000 -0400 @@ -1629,11 +1629,30 @@ power_state); nid = codec->start_nid; + unsigned int pincap; for (i = 0; i < codec->num_nodes; i++, nid++) { - if (get_wcaps(codec, nid) & AC_WCAP_POWER) - snd_hda_codec_write(codec, nid, 0, + if (get_wcaps(codec, nid) & AC_WCAP_POWER) { + pincap = snd_hda_param_read(codec, nid, + AC_PAR_PIN_CAP); + /* + * don't power down the widget if it controls eapd + * and EAPD_BTLENABLE is set. + */ + if (pincap & AC_PINCAP_EAPD) { + int eapd = snd_hda_codec_read(codec, nid, + 0, AC_VERB_GET_EAPD_BTLENABLE, 0); + eapd &= 0x02; + if (power_state != AC_PWRST_D3 || !eapd) { + snd_hda_codec_write(codec, nid, 0, + AC_VERB_SET_POWER_STATE, + power_state); + } + } else { + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, power_state); + } + } } if (power_state == AC_PWRST_D0) { --- alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-24 21:46:12.000000000 -0400 +++ alsa-kernel/pci/hda/patch_sigmatel.c 2007-09-26 20:57:55.000000000 -0400 @@ -378,6 +378,13 @@ {} }; +static struct hda_verb stac9200_eapd_init[] = { + /* set dac0mux for dac converter */ + { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + static struct hda_verb stac925x_core_init[] = { /* set dac0mux for dac converter */ { 0x06, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -2492,7 +2499,23 @@ spec->num_dmics = 0; spec->num_adcs = 1; - spec->init = stac9200_core_init; + /* + * EAPD needs to be set on resume + */ + switch (codec->subsystem_id) { + case 0x107b0205: /* Gateway S-7110M */ + spec->init = stac9200_eapd_init; + break; + case 0x107b0317: /* Gateway MT3423, MX3412, MX3410, MX3414 */ + spec->init = stac9200_eapd_init; + break; + case 0x107b0318: /* Gateway ML3109, MT3707 */ + spec->init = stac9200_eapd_init; + break; + default: + spec->init = stac9200_core_init; + break; + } spec->mixer = stac9200_mixer; err = stac9200_parse_auto_config(codec);

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[PATCH] UBUNTU: Re-add quirk support for Dell XPS 1330 and Inspiron 1420.
by Tim Gardner 09 Oct '07

09 Oct '07

Signed-off-by: Tim Gardner <tim.gardner(a)canonical.com> --- .../alsa-kernel/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c b/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c index 27360d2..f714de1 100644 --- a/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c +++ b/updates/alsa/alsa-driver/alsa-kernel/pci/hda/patch_sigmatel.c @@ -1097,11 +1097,13 @@ static struct snd_pci_quirk stac927x_cfg_tbl[] = { SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), /* 965 based 5 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), -- 1.5.2.5 --------------080009030006050203010104--

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[alsa-devel] ALSA Support for Maya44 PCI?
by Piotr Makowski 09 Oct '07

09 Oct '07

There is already a programmer working on the driver, but thanks for your interest! You can expect the driver for ESI MAYA44 PCI card soon. There will be also an official announcement on alsa-devel when the programmer will start his work. Regards, Piotr Makowski 2007/10/9, Nicolas Rodriguez <nicolas.rod(a)gmail.com>: > I bought a Maya44 PCI 4-channel soundcard and I'll like to use it in Linux. > I couldn't get it to work with Ubuntu (I don't remember the ALSA version, > sorry). Do you support this card? If you don't, I can create the driver for > it. I've read somewhere that ESI-PRO can give the card specifications to > developers interested in build drivers for their cards. I've sent them an > email asking for the card specifications for I've got no response. What do > you recommend? > > Thanks in advance, > Nico. > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] [RFC] sis7019: Initial support for the SiS 7019 Audio Accelerator
by Dave Dillow 09 Oct '07

09 Oct '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth or power management support at the moment, but the audio playback has been tested with several channel and rate combinations, as well as various period and buffer configurations. Signed-off-by: David Dillow <dave(a)thedillows.org> --- include/sound/sis7019.h | 342 ++++++++++++++ sound/pci/Kconfig | 10 + sound/pci/Makefile | 2 + sound/pci/sis7019.c | 1197 +++++++++++++++++++++++++++++++++++++++++++++++ 4 files changed, 1551 insertions(+), 0 deletions(-) diff --git a/include/sound/sis7019.h b/include/sound/sis7019.h new file mode 100644 index 0000000..d395abb --- /dev/null +++ b/include/sound/sis7019.h @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_LEFT_NO_ATTEN 0xff000000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */ diff --git a/sound/pci/Kconfig b/sound/pci/Kconfig index c6b4410..801bb48 100644 --- a/sound/pci/Kconfig +++ b/sound/pci/Kconfig @@ -705,6 +705,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff --git a/sound/pci/Makefile b/sound/pci/Makefile index cd76e02..1904a11 100644 --- a/sound/pci/Makefile +++ b/sound/pci/Makefile @@ -23,6 +23,7 @@ snd-intel8x0m-objs := intel8x0m.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_INTEL8X0M) += snd-intel8x0m.o obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff --git a/sound/pci/sis7019.c b/sound/pci/sis7019.c new file mode 100644 index 0000000..3a74f64 --- /dev/null +++ b/sound/pci/sis7019.c @@ -0,0 +1,1197 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/sis7019.h> + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS 7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the End-Loop Interrupt + * to clock out the periods, and adjust the Loop End Offset of the timing + * channel to maintain synchronization. This algorithm came from the + * Trident driver. + * + * FIXME: The timing channel is configured to give quite a bit of + * attenuation (~96dB), but it may be possible to gain a bit more, or + * even better, play from a page of silence. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u32 sso; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +struct sis7019 { + long unsigned ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* hw_lock protects access to the registers and wave engine parameters + * voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t hw_lock; + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3ffdc, + .period_bytes_min = 8, + .period_bytes_max = 0x3ffdc, + .periods_min = 1, + .periods_max = 0x1ffe, +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = 0x3ffdc, + .period_bytes_min = 8, + .period_bytes_max = 0x3ffdc, + .periods_min = 1, + .periods_max = 0x1ffe, +}; + +static void sis_update_end_offsets(struct voice *voice) +{ + void __iomem *base = voice->ctrl_base; + + if (voice->flags & VOICE_SSO_TIMING) { + /* Calculate the new SSO */ + voice->sso += voice->period_size; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); + } else if (voice->flags & VOICE_SYNC_TIMING) { + u16 cso = readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + int target = voice->sync_cso; + int sync; + u16 leo; + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = cso; + sync -= target; + if (sync > (voice->buffer_size / 2)) + sync -= voice->buffer_size; + + /* If sync is negative, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + * + * FIXME: the first interrupt seems to come when we're at + * offset 0, so we have a huge early entry. Not a huge issue, + * but it would be nice if it didn't happen. + */ + if (sync < 0) { + leo = -sync; + if (leo < 16) + leo = 16; + writew(leo, base + SIS_PLAY_DMA_CONTROL); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * minimum wait time, which could cause issues. + * + * But we don't want to be too late, as we don't want to + * cut into the user's time to refresh the buffers too + * much, especially if we're using small periods. + */ + if (sync < 12) + leo = voice->period_size + 2; + else + leo = voice->period_size - 7; + + sync = voice->sync_cso + voice->period_size; + if (sync >= voice->buffer_size) + sync -= voice->buffer_size; + + voice->sync_cso = sync; + writew(leo, base + SIS_PLAY_DMA_CONTROL); + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status, orig; + int bit; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + spin_lock(&sis->hw_lock); + do { + status = inl(io + SIS_PISR_A); + if (status) { + orig = status; + voice = sis->voices; + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + snd_pcm_period_elapsed(voice->substream); + sis_update_end_offsets(voice); + voice++; + } + outl(orig, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + orig = status; + voice = &sis->voices[32]; + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + snd_pcm_period_elapsed(voice->substream); + sis_update_end_offsets(voice); + voice++; + } + outl(orig, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + spin_unlock(&sis->hw_lock); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + voice->timing->flags = ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + int i; + + spin_lock_irqsave(&sis->voice_lock, flags); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + int needed; + + /* If the period size is exactly one half of the buffer, we don't + * need a timing voice, as we can use the middle loop interrupt + * to clock out the periods. + */ + needed = (params_period_size(hw_params) != + (params_buffer_size(hw_params) / 2)); + + if (needed && !voice->timing) { + voice->timing = sis_alloc_playback_voice(sis); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* Prepare the parameters for this channel before we take the lock. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + if (runtime->period_size == (runtime->buffer_size / 2)) { + control = leo | SIS_PLAY_DMA_LOOP; + control |= SIS_PLAY_DMA_INTR_AT_LEO | SIS_PLAY_DMA_INTR_AT_MLP; + sso_eso = leo; + } else { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control = leo | SIS_PLAY_DMA_LOOP; + control |= SIS_PLAY_DMA_INTR_AT_SSO | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo | (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + + /* Ok, we're ready to go, so take the lock and set up the channel. + */ + spin_lock_irq(&sis->hw_lock); + + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + spin_unlock_irq(&sis->hw_lock); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + spin_lock(&sis->hw_lock); + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + spin_unlock(&sis->hw_lock); + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + struct voice *timing = voice->timing; + void __iomem *rec_base = voice->ctrl_base; + void __iomem *play_base = NULL; + void __iomem *wave_base = NULL; + u32 format, dma_addr, rec_ctrl; + u16 leo; + + /* The following variables are only used if there is a timing channel. + */ + u32 uninitialized_var(timing_ctrl); + u32 uninitialized_var(sso_eso); + u32 uninitialized_var(delta); + + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + rec_ctrl = leo | SIS_CAPTURE_DMA_LOOP; + if (timing) { + /* We're using a timing voice, so the capture voice will + * not need to generate interrupts. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_cso = runtime->period_size; + timing->period_size = runtime->period_size; + timing->buffer_size = runtime->buffer_size; + timing->sync_base = voice->ctrl_base; + + timing_ctrl = leo + 16; + timing_ctrl |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + delta = sis_rate_to_delta(runtime->rate); + + play_base = timing->ctrl_base; + wave_base = timing->wave_base; + } else { + /* We've just got two periods, so we can use the capture + * voice's interrupt generation. + */ + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_MLP; + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_LEO; + } + + /* We're ready to program the channel(s), so take the lock and go. + */ + spin_lock_irq(&sis->hw_lock); + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(rec_ctrl, rec_base + SIS_CAPTURE_DMA_CONTROL); + + if (play_base) { + u32 reg; + + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(timing_ctrl, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + /* We don't want the timing channel to generate any sound, + * so set it to maximum attenuation (31.75 dB), and run it + * through the music amplifier, which has been set to add + * an additional 64 dB of attenuation. That will be 95.76 dB + * total. + * + * FIXME: While this works well in practice, purists may + * wish to investigate using the mixer lists to further + * attenuate the timing channel -- or set aside a page + * of zeros for silence. This may also free up the music + * amplifier for other purposes. + */ + writel(SIS_WAVE_GENERAL_MUSIC_VOLUME | (0x7f << 24), + wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + } + + readl(rec_base + 0); + + spin_unlock_irq(&sis->hw_lock); + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + rc = snd_pcm_new(sis->card, "SiS 7019", 0, 40, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS 7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned long flags; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + spin_lock_irqsave(&sis->hw_lock, flags); + + /* Get the AC97 semaphore ... + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + spin_unlock_irqrestore(&sis->hw_lock, flags); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * Do maximum attenuation on audio set to use the music amplifier, + * as we'll set any timing channels to use that. + */ + outl(0xffff0000, SIS_WEVCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + /* FIXME: Not sure of our DMA limit, use 32 bit mask for now. + */ + if (pci_set_dma_mask(pci, DMA_32BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "32-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + spin_lock_init(&sis->hw_lock); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS 7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + "SiS 7019", sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + static int dev; + int rc; + + rc = -ENODEV; + if (dev >= SNDRV_CARDS) + goto error_out; + + rc = -ENOENT; + if (!enable[dev]) { + dev++; + goto error_out; + } + + rc = -ENOMEM; + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + strcpy(card->driver, "SiS 7019"); + strcpy(card->shortname, "SiS 7019"); + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + dev++; + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + + /* FIXME: need to support power management */ +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); -- 1.5.0.6

3 5

Re: [alsa-devel] different user causes 'no such device'
by william estrada 09 Oct '07

09 Oct '07

Thanks, I will try this. William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum Takashi Iwai wrote: > At Wed, 03 Oct 2007 12:55:27 -0700, > william estrada wrote: >> Hi guys, >> >> I have what seams like a interesting problem. I have written an ALSA app >> to record wav files and it runs while I am logged in an terminal as the >> logged in Gnome user or root. But when I su to another user, it can't >> find the default sound card. >> >> I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did >> not work. >> >> Anyone know of this problem and a fix for it. >> >> Thanks for your time. >> >> -- >> William Estrada >> MrUmunhum(a)popdial.com >> Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) >> Ymessenger: MrUmunhum >> >>> MSGrec - Version 1.1, Mt-Umunhum-Wireless.net >>> WavMSG: Paused: enter 'q' to stop, <ENTER> to start. >>> ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device >>> ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device >>> ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name >>> ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device >>> ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device >>> ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default > > ALSA-native apps access only ALSA device files, usually /dev/snd/*, > not /dev/dsp*, those for OSS. In this case, it should be > /dev/snd/controlC* files. Check the premission of these files. > > > Takashi >

1 0

[alsa-devel] ALSA Support for Maya44 PCI?
by Nicolas Rodriguez 09 Oct '07

09 Oct '07

I bought a Maya44 PCI 4-channel soundcard and I'll like to use it in Linux. I couldn't get it to work with Ubuntu (I don't remember the ALSA version, sorry). Do you support this card? If you don't, I can create the driver for it. I've read somewhere that ESI-PRO can give the card specifications to developers interested in build drivers for their cards. I've sent them an email asking for the card specifications for I've got no response. What do you recommend? Thanks in advance, Nico.

1 0

[alsa-devel] Envy24control s/pdif input bugfix
by Stephan Zaubzer 09 Oct '07

09 Oct '07

Hi! This patch against the stable version of envy24control fixes a bug where the S/Pdif input switch doesn't work on Terratec DMX 6fire cards. This bug first appeared in version 1.0.11rc3. diff -ubr alsa-tools-1.0.14patch/envy24control/hardware.c alsa-tools-1.0.14/envy24control/hardware.c --- alsa-tools-1.0.14patch/envy24control/hardware.c 2007-09-23 15:58:53.000000000 +0200 +++ alsa-tools-1.0.14/envy24control/hardware.c 2007-06-04 13:27:25.000000000 +0200 @@ -775,7 +775,7 @@ snd_ctl_elem_value_set_interface(volume_rate, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_value_set_name(volume_rate, "Multi Track Volume Rate"); - if (card_is_dmx6fire) { + if (! card_is_dmx6fire) { snd_ctl_elem_value_set_interface(spdif_input, SND_CTL_ELEM_IFACE_MIXER); snd_ctl_elem_value_set_name(spdif_input, "Optical Digital Input Switch"); } else { Regards Stephan

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Re: [alsa-devel] [Alsa-user] Asus M2NPV-VM snd_hda_intel problem - also with Alsa 1.0.15rc3
by Ma Begaj 09 Oct '07

09 Oct '07

> Firstly, I don't have the same board. Similar, but different. > > I have an Asus A8N-VM CSM board, but it has the same AD1986A chip on it. > I didn't have the same issue with sound muting, but I did have issues > with surround (it just didn't work). I don't remember the details of how > > > --- patch_analog.c.backup 2007-04-14 11:01:17.209146249 +0100 > +++ patch_analog.c 2007-04-14 13:37:47.005206476 +0100 > @@ -895,14 +895,14 @@ > case AD1986A_3STACK: > spec->num_mixers = 2; > spec->mixers[1] = ad1986a_3st_mixers; > - spec->num_init_verbs = 3; > +/* spec->num_init_verbs = 3; > spec->init_verbs[1] = ad1986a_3st_init_verbs; > - spec->init_verbs[2] = ad1986a_ch2_init; > + spec->init_verbs[2] = ad1986a_ch2_init; */ > spec->channel_mode = ad1986a_modes; > spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); > spec->need_dac_fix = 1; > - spec->multiout.max_channels = 2; > - spec->multiout.num_dacs = 1; > + spec->multiout.max_channels = 6; > + spec->multiout.num_dacs = 3; > break; > case AD1986A_LAPTOP: > spec->mixers[0] = ad1986a_laptop_mixers; > thanks for your try. it helped partly ... I tried 3 patches (incl. yours) and only yours helped partly at least. I patched Alsa 1.0.15rc3. Sound mutes as soon as I change volume, and I can only rmmod/modprobe snd-hda-intel to get it back. But Surround is working with your patch as I can see: "speaker-test -Dplug:surround51 -c6 -t waw" recognizes correctly all speakears. "speaker-test -c6 -t waw" does not recognize the speakers correctly, but it probably should not, or ?! MythTv is playing stereo files only on the right speakers. I do not now is this a MythTv or Alsa problem. probably MythTv and I should create a .asoundrc which should correct this somehow. I did not try hg, because I thought that Alsa 1.0.15rc3 is very new. Thanks. M.

1 1

[alsa-devel] Capturing of Samsung Q1 Ultra (ALC262) does not work
by Pengcheng Zou 09 Oct '07

09 Oct '07

Hi, capture function of the Samsung Q1 Ultra (using ALC262 codec) does not work. I tried the most recent ALSA build (alsa-driver-1.0.15rc3), the playback works well, however the recording does not work. Samsung Q1 (using AD1986A codec) has been supported already, wonder if there is any patches for Q1 Ultra? please note Samsung released 2 types of UPMC, Q1 and Q1 Ultra. the Q1 ultra is the one I have: http://computers.pricegrabber.com/tablet-pcs/m/37616779/details/ TIA, -- Pengcheng

3 2

[alsa-devel] snd_pcm_drain and lock-ups
by Mike Gorse 08 Oct '07

08 Oct '07

Hi all, I've followed Arve Knudsen over here. I am getting a lock-up when calling snd_pcm_drain(). This happens when using espeak, which is using portaudio. Portaudio calls snd_pcm_drain while trying to recover from a XRUN. snd_pcm_drain is calling snd_pcm_rate_drain in my case. snd_pcm_rate_drain then calls snd_pcm_wait. snd_pcm_wait first checks if snd_pcm_mmap_avail(pcm) >= pcm->avail_min. snd_pcm_map_avail is returning a number less than 300, often between 135 and 140 but not always, where pcm->avail_min == 1024. So it does not check for the XRUN state since it is conditional on snd_pcm_mmap_avail returning >= 1024. This is on my laptop, which is running Linux 2.6.22 and alsa-lib 1.0.15-rc1 and uses the snd-intel8x0 driver. Does this give people enough information to tell what might be going on? I have a small test program that causes the lock-up on my laptop where the rate plugin gets used (it sometimes locks up immediately, sometimes after a few seconds), but it needs espeak and portaudio to be installed. Thanks, -- Mike Gorse / AIM:linvortex / http://mgorse.freeshell.org --

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Re: [alsa-devel] [Alsa-user] How to get set/fixed sample rate of ALSA device?
by Rene Herman 08 Oct '07

08 Oct '07

On 10/08/2007 12:41 AM, Mark Constable wrote: > Any chance that your program could take an arg to investigate > other cards (or perhaps plugins) and also return an errno so > it could be used in scripts ? Only if you promise to never drop CCs again (although I don't know about plugins). > # gcc -W -Wall -o snd_rate snd_rate.c -l asound > snd_rate.c: In function 'main': > snd_rate.c:14: warning: the address of ‘params’ will always evaluate as 'true' That's the ALSA snd_pcm_hw_params_alloca(). No such warning here, but I've made it a regular malloc. > # snd_rate > snd_pcm_open: Device or resource busy > > In fact, why isn't there a single simple "alsainfo" type > program that tells a user some (or a lot) of info about > their hardware and alsa-lib install ? I guess because noone wrote one :-) It seems by the way that dir != 0 _can_ be returned for hw devices, and this just ignores it. Just some rounding issue I guess. Rene. /* gcc -W -Wall -o alsa-rate alsa-rate.c -lasound */ #include <stdio.h> #include <alsa/asoundlib.h> int main(int argc, char *argv[]) { char *device; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int min; unsigned int max; int dir; int err; device = argc > 1 ? argv[1] : "hw:0,0"; err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0); if (err < 0) { fprintf(stderr, "snd_pcm_open: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_malloc(&params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_alloc: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_any(handle, params); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_any: %s\n", snd_strerror(err)); return err; } err = snd_pcm_close(handle); if (err < 0) { fprintf(stderr, "snd_pcm_close: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_rate_min(params, &min, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_min: %s\n", snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_rate_max(params, &max, &dir); if (err < 0) { fprintf(stderr, "snd_pcm_hw_params_get_rate_max: %s\n", snd_strerror(err)); return err; } snd_pcm_hw_params_free(params); printf("%u-%u\n", min, max); return 0; }

1 0

[alsa-devel] Use-before-initialization in alsa-oss
by Alexander E. Patrakov 06 Oct '07

06 Oct '07

dsp->oss_format was used before it was set correctly spotted by running "aoss flite_time 10:00" Signed-Off-By: Alexander E. Patrakov --- alsa-oss-1.0.14.orig/alsa/pcm.c 2007-09-29 16:12:44.000000000 +0600 +++ alsa-oss-1.0.14/alsa/pcm.c 2007-09-29 16:13:06.000000000 +0600 @@ -215,10 +215,10 @@ unsigned int rate, periods_min; if (!pcm) continue; + dsp->format = oss_format_to_alsa(dsp->oss_format); str->frame_bytes = snd_pcm_format_physical_width(dsp->format) * dsp->channels / 8; snd_pcm_hw_params_alloca(&hw); snd_pcm_hw_params_any(pcm, hw); - dsp->format = oss_format_to_alsa(dsp->oss_format); err = snd_pcm_hw_params_set_format(pcm, hw, dsp->format); if (err < 0) -- Alexander E. Patrakov

2 1

[alsa-devel] ASoC v2 on PowerPC
by Jon Smirl 05 Oct '07

05 Oct '07

I'm looking into write a ASoC v2 driver for the Efika. Efika uses a Freescale MPC5200B and a STAC9766. This is a practice driver before writing one for our target hardware (which doesn't work yet, same CPU but different codec). I've brought the changes from wolfson/asoc-v2-dev into my powerpc tree and got a couple of config errors with ARCH=powerpc. sound/soc/pxa/Kconfig:23:warning: 'select' used by config symbol 'SND_PXA2XX_SOC_SSP' refers to undefined symbol 'PXA_SSP' sound/soc/au1x/Kconfig:35:warning: 'select' used by config symbol 'SND_SOC_EXM1200_SSI' refers to undefined symbol 'SND_SOC_CS4251X' Suggestions on the best way to start? Target hardware also has a WM8804 on it. -- Jon Smirl jonsmirl(a)gmail.com

3 5

[alsa-devel] Why was the built-in software volume control removed?
by kyle cronan 04 Oct '07

04 Oct '07

I think I may have posted this to the wrong list. There wasn't any response yet on alsa-user, so I'm reposting here. Summary: I'm seeing a lot of glitches ever since the built-in software volume was removed. What was the reason for this change? Softvol doesn't work nearly as well for me. ________________ Hi everyone, I recently updated my media center which is running Fedora devel. It pulled down a new kernel with updated ALSA drivers for my HDA audio chip (Intel 82801H + ALC883). I use IEC958 output to a pair of studio monitors (good sound with minimal fuss). The first thing I noticed about the update was that my software volume control was no longer saving state. It was resetting to something around 80% every time an app opened the audio device, a level which, in my setup, is quite loud! Then I saw that in alsamixer I had no master volume control, not even a PCM control! So I started experimented with my settings, and I found that after I selected "PCM" under "Default Mixer Tracks" in gnome-sound-properties I had a PCM volume control again (still no master), and apps using alsa output would once again remember the system volume. Don't know what that gnome setting actually did, but I've not been able to get it back to the original state, even when I ctrl-click to deselect "PCM" in the list. However, apps that use legacy OSS output still blast me with nearly full volume every time. I've also noticed another glitch: using the volume control applet to adjust my PCM volume, the audio sometimes skips. If I quickly drag the slider back and forth, it always does. I'm pretty sure I would have noticed this if it occurred with my previous kernel (haven't had a chance to test that yet). Anyway, looking into this problem, I found the following note in the changes document at http://www.alsa-project.org/main/index.php/Changes_v1.0.15rc2_v1.0.15rc3 Sound Core ... - Remove the built-in software volume control in favour of softvol. So, my (main) question is: is there a good reason that the built-in software volume was removed? I didn't find anything about it in the archives for this list or -devel. I'm really not in favor of this change, because It used to work flawlessly for me, whereas now I'm running into all these strange problems with my IEC958 output. Thanks, Kyle Cronan <kyle(a)pbx.org>

2 1

[alsa-devel] Tascam US122 on Arm
by ST 04 Oct '07

04 Oct '07

Hi I am a little irritated that there has been no replies to my questions to this list (http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003387.…, http://mailman.alsa-project.org/pipermail/alsa-devel/2007-September/003083.…). Am i on the wrong list? Please tell me if so. Probably its also that i didn't make clear in my last post that the tascam driver works on arm with my US122 sound card, for ca. 51 seconds. After that i get the following error: Sequence Error!(hcd_frame=6 ep=8in;wait=65538,frame=2). Most propably some urb of usb-frame 65538 is still missing. Cause could be too long delays in usb-hcd interrupt handling. I also tried insmod ./2.6.18/kernel/sound/usb/usx2y/snd-usb-usx2y.ko nrpacks=X with X=[1,2,4,8]. I thought that if it is really an CPU performance problem than probably a bigger buffer would eliminate the problem. However the error persisted. Even commenting out /* #define USX2Y_NRPACKS_VARIABLE y */ and setting #define USX2Y_NRPACKS 1 didn't help. So my question is now, is the 250Mhz ARM really to slow or is there any chance that there is an error in either the usb stack or the driver exposed only on the ARM architecture? Things speaking for the latter are: *Error allways in the 65536-65540 value. *Error happens most of the times after about 50secs. Since i do not know much about USB any hint is apreciated. ST

2 2

[alsa-devel] about sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?)
by Akio 04 Oct '07

04 Oct '07

Dear ALL: after long time amr-wb codec loopback test. (above 90 hours) I got the following kernel message. it comes from snd_pcm_lib_read1() function. ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) ALSA sound/core/pcm_lib.c:1997: capture read error (DMA or IRQ trouble?) What should I do? My platform is ARM926, with Kernel 2.6.19.2 ALSA version is 1.0.13. Best Regards, Akio

2 9

[alsa-devel] different user causes 'no such device'
by william estrada 03 Oct '07

03 Oct '07

Hi guys, I have what seams like a interesting problem. I have written an ALSA app to record wav files and it runs while I am logged in an terminal as the logged in Gnome user or root. But when I su to another user, it can't find the default sound card. I am running FC6. I tried changing own of /dev/dsp to 'nobody', that did not work. Anyone know of this problem and a fix for it. Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum > MSGrec - Version 1.1, Mt-Umunhum-Wireless.net > WavMSG: Paused: enter 'q' to stop, <ENTER> to start. > ALSA lib confmisc.c:848:(snd_func_card_driver) cannot find card '0' > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_card_driver returned error: No such device > ALSA lib confmisc.c:397:(snd_func_concat) error evaluating strings > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_concat returned error: No such device > ALSA lib confmisc.c:1248:(snd_func_refer) error evaluating name > ALSA lib conf.c:3500:(_snd_config_evaluate) function snd_func_refer returned error: No such device > ALSA lib conf.c:3972:(snd_config_expand) Evaluate error: No such device > ALSA lib pcm.c:2145:(snd_pcm_open_noupdate) Unknown PCM default

1 0

[alsa-devel] Snd problem ... ( ACCESS_TYPE )
by errordeveloper＠gmail.com 03 Oct '07

03 Oct '07

Hi list. I' m using alsa-lib v1.0.14rc2 on gentoo-linux (v2.6.20-gentoo-r8) my soundcard is RME Hammerfall HDSP9632 (snd-hdsp driver) i have been using it for a while - all fine i only run a few applications for my sound experiments. one of those is Snd (from CCRMA), it was okay until i have done a few updates lately - kernel from 2.6.17 to 2.6.20 and alsa-lib as well. i'm not sure about either of these but i check Snd src-tree and the mailing list - nothing was changed .. the error msg that i get is: :(play) ;status of hw:0 hw_params status of hw:0 ACCESS: MMAP_NONINTERLEAVED RW_NONINTERLEAVED FORMAT: S32_LE SUBFORMAT: STD SAMPLE_BITS: 32 FRAME_BITS: [128 384] CHANNELS: [4 12] RATE: [32000 192000] PERIOD_TIME: (333 256000] PERIOD_SIZE: [64 8192] PERIOD_BYTES: [2560 393216] PERIODS: 2 BUFFER_TIME: (666 512000] BUFFER_SIZE: [128 16384] BUFFER_BYTES: [2560 786432] TICK_TIME: 1000 sw_params status of hw:0 start_mode: DATA xrun_mode: STOP tstamp_mode: NONE period_step: 0 sleep_min: 0 avail_min: 0 xfer_align: 0 silence_threshold: 0 silence_size: 0 boundary: 0 ;Invalid argument: hw:0: access type RW_INTERLEAVED not available can't play : Invalid argument: hw:0: access type RW_INTERLEAVED not available #f : i had tried to modify audio.c in Snd's tree to use both ACCESS types given above , but it didn't help - showed device is busy error instead. i'm only studying C, so can't be sure .. anyway here is a problem which i could not solve yet and haven't seen anyone helpful around #alsa, neither i could google anything out. for a ref.: Cmdist mailing list Cmdist(a)ccrma.stanford.edu http://ccrma-mail.stanford.edu/mailman/listinfo/cmdist and our discussion is here: http://ccrma-mail.stanford.edu/pipermail/cmdist/2007-July/004221.html could anyone have a look at this at somepoint, please . -- thank you, Ilya. D errordeveloper|*@*|gmail|*dot*|com

1 0

[alsa-devel] ASOC: Best way to get private data from machine driver to the other drivers?
by Timur Tabi 02 Oct '07

02 Oct '07

I'm writing an ASOC driver. When my machine driver loads, it determines some hardware information for the DMA controllers, some other hardware information for the I2S controller, and even more information for the codec. I want to be able to pass all this information to each of the three other drivers. Unfortunately, it doesn't appear that I have enough "private_data" pointers available to me. The current ASOC drivers all use statically defined global structures and indexes into arrays of these structures to identify individual devices, such associating I2S controller #2 with DMA channels 3 and 4. On PowerPC, this doesn't work. The actual hardware configuration is determined at runtime (see arch/powerpc/boot/dts/), and so the machine driver needs to collect all the information and make all the device-to-device associations. For example, the PCM driver has an open function: static struct snd_pcm_ops fsl_dma_ops = { .open = fsl_dma_open, ... static int fsl_dma_open(struct snd_pcm_substream *substream) { ... What I would like is for fsl_dma_open() to be able to obtain pointers to the DMA controller's registers (there are two controllers with 4 channels each) by looking inside 'substream'. How can I do that? -- Timur Tabi Linux Kernel Developer @ Freescale

2 8

[alsa-devel] some problem about alsa lib usage on arm926
by Akio 02 Oct '07

02 Oct '07

Dear All: right now, I'm using alsa lib interface to work with voip application. Our hardware platform is an ARM926 266MHz box with Linux kernel 2.6.19.2. We got some problems. would you like to do us favors? 1. what is the different between "plughw:0,0", "defalut", "hw:0.0"? after reading the document in http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html about "PCM naming conventions", we still not very sure the relationship between those device conventions. is there any more detail documents talk about on this? right now, we are using alsa lib 1.0.13 and the device is "plughw:0,0". 2. Is there any documents detail discuss the setting about sample rate, buffer size, period size. because, I have two different voice codec to handle, one is narrow band(8k Hz), the other is wide band(16K Hz). right now, I using the following functions and setting to make alsa work. snd_pcm_hw_params_set_rate_near sets sample rate, 8k or 16k. snd_pcm_hw_params_set_period_size_near sets period size, current is 128. snd_pcm_hw_params_set_buffer_size_near sets buffer size, 8k is 2048, 16k is 4096. the unit seems be samples not byte. 3. the code was based on the following URL. http://www.suse.de/~mana/alsa090_howto.html http://www.equalarea.com/paul/alsa-audio.html open capture and playback at one function, and provide one read and one write interface to up layer. here comes three problems: a. the channel opened is just only one, but If I set the snd_pcm_hw_params_set_access as SND_PCM_ACCESS_RW_NONINTERLEAVED. using snd_pcm_readn and snd_pcm_writen the code will not work. what is the wrong? but if set snd_pcm_hw_params_set_access as SND_PCM_ACCESS_RW_INTERLEAVED and using snd_pcm_readi and snd_pcm_writei. the code work very well. one channel also have data interleave problem? I don't understand why? b. in snd_pcm_open the block flag is set as 0. what is the mode of the operation? blocking or non-blocking? c. call the xrun_recovery() timing. currently the xrun_recovery only do on EPIPE. the following is the handling core. err = snd_pcm_prepare(handle); if(err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) { nanosleep(&retryTime, &tmpTime); LOG("audio device resume fail, %s\n", snd_strerror(err)); } if(err < 0) { err = snd_pcm_prepare(handle); if(err < 0) LOG("audio data IO fail, %s\n", snd_strerror(err)); } } else { if(err < 0) LOG("audio device prepare fail, %s\n", snd_strerror(err)); } Does the code have any missing point to handle? If any missing, what should I to add? 4. BTW, I call xrun_recovery both in capture and playback direction. After long time test. the test procedure is as the following steps: 1. capture voice data as PCM data. 2. encoded as amr-wb bit stream. 3. decoded as PCM data from encoded bit stream. 4. playback PCM data. after 16 hours continue tests, there are some delay, about 800 mili seconds. is there any way to reduce the delay after long time test. we want make the delay about < 400 mili seconds after one week continue test. BTW, is there any way to prevent xrun_recovery calling? is there any index to monitor ALSA device buffer information? 5. If we want using salsa lib to replace the usage of alsa lib, what do we have to take care? how about the latency and performance between salsa and alsa? how about the stability of salsa? There are so many problems after many tests. If you found any documents we need to study more detail. please tell us. Thanks in advance. Thank you. Best Regards, Akio

2 3

[alsa-devel] issue with ADI A1986A
by l1983＠bankerinter.net 01 Oct '07

01 Oct '07

Hello, I need some help from you guys. I,m a new linux user and totally unfamiliar with all this coding... I have an issue with my motherboard "ASSUS M2N-MX" and it's onboard sound chip :"ADI AD1986A" witch supports 6 channel CODEC , Jack-Sensing & Enumeration technology , and with S/PDIF OUT interface. Can you please somehow, (or somebody you know) , make for me some installable .deb archive that will help me to set up my sound, since i can't do it on my own. :( Or help me some how else, (either sending me some address's or by giving me some other advice. HELP ME PLEASE!!!!!! l1983(a)bankerinter.net

1 0

[alsa-devel] short writes?
by william estrada 30 Sep '07

30 Sep '07

Hi group, I have written my own ALSA recorder and player. Along with that I have several bugs. The bug I am working on right now is 'short writes' when I use the pipe command into my player. My player runs and asks for the file to play, so that I may play incoming files as they are sent in. I have a program that uses 'inotify' to watch for incoming file and then pipe the file name into my player. Now with all of that, is there a API call that will put the writei into blocking mode? Or an API call that will flush the output from writei? Or any other suggestions? Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

2 1

[alsa-devel] nforce ac97 plays too fast
by Stas Sergeev 30 Sep '07

30 Sep '07

Hello. Trying to get a software amplification to work with NForce CK804 AC'97, I've found out that whatever the asound.conf I write, the sound starts playing much faster than it should. After some debugging, I've found out the following: - The card (or the driver) only supports 48000KHz. - The program (any, because libao and artsd suffer from that) is trying to set some rate with snd_pcm_hw_params_set_rate_near(), and that succeeds, but the returned rate is 48000, while the requested is, say, 22050. No wonder it then plays with the wrong speed. Of course the program can check the returned rate, but it sets the "direction" parameter to 0, which should mean "take exactly that value". I've looked into an alsa-tutorial, and the examples there do not check the returned rate too - dir==0 should be enough. Looking into the sources, it seems the culprit is the snd_interval_refine_min/max(), which do not care about the (i->min > min && !openmin) case at all. The question is: is setting dir=0 (or even NULL, since it is a pointer) enough to set the exact rate? And if so - how snd_interval_refine_min() should handle the (i->min > min && !openmin) case? Or if not - what does the dir==0 mean and why it is not adjusted to the proper value on return, when the rate is changed? (i.e. the rate parameter changes from 22050 to 48000, but the dir parameter remains 0, doesn't change) With the default configuration the rate is proper, but I find it annoying that altering asound.conf in pretty much any way gets the wrong results.

2 2

[alsa-devel] Asus M2NPV-VM snd_hda_intel problem - also with Alsa 1.0.15rc3
by Ma Begaj 29 Sep '07

29 Sep '07

Hi, for the sweet love of ... does anyone have an idea how to make this sound chip working... I have Asus M2NPV-VM with AD1986A sound chip. I tried "model=3stack position_fix=1", but it does not help. surround does not work and sound mutes whenever I change volume. when I use model=ultra or eapd-laptop, stereo is at stable and I can change volume. but surround I tried it with vanilla kernel 2.6.21.5/26.22.6 and snd_hda_intel from the kernel without any success. I even compiled Alsa 1.0.15rc3, but situation did no change. anybody ... with any idea ... I would appreciate any help. any help at all. thanks. M.

1 1

[alsa-devel] Alsa 1.0.15rc2 and oops with Dell M1210
by Herton Ronaldo Krzesinski 29 Sep '07

29 Sep '07

Hi, I have some reports of problems with Alsa 1.0.15rc2 and Dell M1210 (dell-m82 model). The following oops happens: Sep 27 19:28:34 dru kernel: Unable to handle kernel paging request at ffff81107b70f3a0 RIP: Sep 27 19:28:34 dru kernel: [<ffffffff88108751>] :snd_hda_intel:snd_hda_input_mux_info+0x31/0x50 Sep 27 19:28:34 dru kernel: PGD 8063 PUD 0 Sep 27 19:28:34 dru kernel: Oops: 0000 [1] SMP Sep 27 19:28:34 dru kernel: CPU 0 Sep 27 19:28:34 dru kernel: Modules linked in: ipv6 video thermal sbs fan container snd_seq_dummy button snd_seq_oss snd_seq_midi_event snd_seq do ck battery ac ide_cd binfmt_misc loop ntfs nls_utf8 nls_cp437 vfat fat dm_mirror dm_mod usb_storage ide_core hidp l2cap usbmouse usbhid ff_memless usbkbd hci_usb bluetooth cpufreq_ondemand cpufreq_conservative cpufreq_powersave acpi_cpufreq freq_table processor nvram pcspkr snd_usb_lib snd_r awmidi snd_seq_device b44 snd_pcm_oss snd_mixer_oss ssb pcmcia pcmcia_core uhci_hcd ehci_hcd usbcore mii ohci1394 ieee1394 i2c_i801 i2c_core evdev snd_hda_intel snd_pcm snd_timer snd_page_alloc snd_hwdep joydev tsdev mmc_block sdhci mmc_core snd soundcore iTCO_wdt iTCO_vendor_support sg sr_m od ext3 jbd scsi_wait_scan sd_mod ahci ata_piix libata scsi_mod Sep 27 19:28:34 dru kernel: Pid: 3466, comm: alsactl Not tainted 2.6.22.9-laptop-1mdv #1 Sep 27 19:28:34 dru kernel: RIP: 0010:[<ffffffff88108751>] [<ffffffff88108751>] :snd_hda_intel:snd_hda_input_mux_info+0x31/0x50 Sep 27 19:28:34 dru kernel: RSP: 0018:ffff81007aa65d48 EFLAGS: 00010206 Sep 27 19:28:34 dru kernel: RAX: 0000000ffffffff0 RBX: ffff81007b734d80 RCX: 00000000ffffffff Sep 27 19:28:34 dru kernel: RDX: 0000000000000000 RSI: ffff81007aa65d88 RDI: ffff81007aa65de0 Sep 27 19:28:34 dru kernel: RBP: ffff81007aa65d88 R08: ffff81007b70f3a8 R09: 000000000000000a Sep 27 19:28:34 dru kernel: R10: 0000000000000000 R11: 0000000000000246 R12: ffff81007c44a968 Sep 27 19:28:34 dru kernel: R13: ffff81007c44a800 R14: ffff81007adbaf40 R15: 00007fff95ccbdb0 Sep 27 19:28:34 dru kernel: FS: 00002ad615cc6e50(0000) GS:ffffffff8053e000(0000) knlGS:0000000000000000 Sep 27 19:28:34 dru kernel: CS: 0010 DS: 0000 ES: 0000 CR0: 000000008005003b Sep 27 19:28:34 dru kernel: CR2: ffff81107b70f3a0 CR3: 000000007a926000 CR4: 00000000000006e0 Sep 27 19:28:34 dru kernel: Process alsactl (pid: 3466, threadinfo ffff81007aa64000, task ffff81007c659860) Sep 27 19:28:34 dru kernel: Stack: 00007fff95ccbdb0 ffffffff880b3835 0000000000000000 00007fff95ccbdb0 Sep 27 19:28:34 dru kernel: 00000000c1105511 ffff81007c44a800 ffff81007adbaf40 ffffffff880b4d88 Sep 27 19:28:34 dru kernel: 0000000000000001 0000000000000000 0000000000000000 0000000000000000 Sep 27 19:28:34 dru kernel: Call Trace: Sep 27 19:28:34 dru kernel: [<ffffffff880b3835>] :snd:snd_ctl_elem_info+0x55/0x150 Sep 27 19:28:34 dru kernel: [<ffffffff880b4d88>] :snd:snd_ctl_ioctl+0x648/0x8b0 Sep 27 19:28:34 dru kernel: [<ffffffff802aa045>] do_ioctl+0x35/0xe0 Sep 27 19:28:34 dru kernel: [<ffffffff802aa310>] vfs_ioctl+0x220/0x2c0 Sep 27 19:28:34 dru kernel: [<ffffffff802aa445>] sys_ioctl+0x95/0xb0 Sep 27 19:28:34 dru kernel: [<ffffffff8020a28e>] system_call+0x7e/0x83 Sep 27 19:28:34 dru kernel: Sep 27 19:28:34 dru kernel: Sep 27 19:28:34 dru kernel: Code: 49 8b 74 00 08 e8 95 bd 21 f8 31 c0 48 83 c4 08 c3 0f 1f 80 Sep 27 19:28:34 dru kernel: RIP [<ffffffff88108751>] :snd_hda_intel:snd_hda_input_mux_info+0x31/0x50 Sep 27 19:28:34 dru kernel: RSP <ffff81007aa65d48> Sep 27 19:28:34 dru kernel: CR2: ffff81107b70f3a0 I have a better report on a i386 kernel, unfortunately it's tainted: BUG: unable to handle kernel NULL pointer dereference at virtual address 00000000 printing eip: f8f89ed8 *pde = 00000000 Oops: 0000 [#1] SMP Modules linked in: snd_usb_audio snd_usb_lib snd_rawmidi snd_seq_device iwl3945 mac80211 cfg80211 b44 mmc_block nvidia(P) snd_hda_intel snd_pcm ohci1394 snd_timer ssb pcmcia sdhci pcmcia_core mii ieee1394 mmc_core joydev ipw3945 ieee80211 ehci_hcd snd_page_alloc snd_hwdep uhci_hcd ieee80211_crypt snd intel_agp i2c_i801 tsdev usbcore soundcore sg agpgart evdev i2c_core iTCO_wdt iTCO_vendor_support sr_mod reiserfs scsi_wait_scan sd_mod ahci ata_piix libata scsi_mod CPU: 0 EIP: 0060:[<f8f89ed8>] Tainted: P VLI EFLAGS: 00010246 (2.6.22.7-laptop-1mdv #1) EIP is at snd_hda_input_mux_info+0x38/0x50 [snd_hda_intel] eax: f6764a4c ebx: 00000000 ecx: ffffffff edx: f6269de4 esi: 00000000 edi: f6269e3c ebp: f6269db0 esp: f6269da4 ds: 007b es: 007b fs: 00d8 gs: 0033 ss: 0068 Process alsactl (pid: 2318, ti=f6268000 task=c237a550 task.ti=f6268000) Stack: f66da5c0 00000000 f6269de4 f6269db8 f8f995e4 f6269f48 f8edd1eb 00000000 bf979c70 f64fcac0 f66b6c00 f66b6d4c f8cf0000 f66b6d4c c0199456 00000000 00000001 00000002 00000000 00000000 75706e49 6f532074 65637275 00000000 Call Trace: [<c010529a>] show_trace_log_lvl+0x1a/0x30 [<c010535b>] show_stack_log_lvl+0xab/0xd0 [<c0105551>] show_registers+0x1d1/0x2d0 [<c0105768>] die+0x118/0x240 [<c0121399>] do_page_fault+0x199/0x630 [<c0313e6a>] error_code+0x72/0x78 [<f8f995e4>] stac92xx_mux_enum_info+0x14/0x20 [snd_hda_intel] [<f8edd1eb>] snd_ctl_ioctl+0x80b/0xc30 [snd] [<c018f308>] do_ioctl+0x28/0xb0 [<c018f5af>] vfs_ioctl+0x21f/0x2a0 [<c018f689>] sys_ioctl+0x59/0x70 [<c01041ba>] sysenter_past_esp+0x6b/0xa1 ======================= Code: 08 8d 7a 58 89 1c 24 c7 42 40 03 00 00 00 c7 42 48 01 00 00 00 8b 08 89 4a 50 8b 4a 54 8b 18 8d 73 ff 39 d9 0f 43 ce 8b 74 c8 04 <ac> aa 84 c0 75 fa 8b 1c 24 31 c0 8b 74 24 04 8b 7c 24 08 89 ec EIP: [<f8f89ed8>] snd_hda_input_mux_info+0x38/0x50 [snd_hda_intel] SS:ESP 0068:f6269da4 Accordingly to one of the reporters, when changing the model to 5stack the oops doesn't happens anymore and everything works flawlessly. At first I thought that the problem could be the different pin cfgs of dell-m82 compared to the old and replaced dell model. I was in contact with one of the reporters, but unfortunately reverting only the pin cfg didn't work (if the report was correct the module now didn't break with an oops but sound didn't work either). So I'm out of ideas, failed to see why stac92xx_mux_enum_info received uinfo == NULL (where uinfo comes from?) and passed it to snd_hda_input_mux_info. -- []'s Herton

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[alsa-devel] using ALSA OSS Emulation interface in long time will make voice delay
by Akio 28 Sep '07

28 Sep '07

Dear All: Is there any futher action to this topic "BF537Stamp Linphone audio falling behind". the URL is http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/? action=ForumBrowse&forum_id=39&_forum_action=ForumMessageBrowse&thread_id=22524 we have the same problem, but if we switch to alsa lib interface. the problem seems gone. Thanks in advance. Best Regards, Akio

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[alsa-devel] Asus M2NPV-VM (AD1986A) - Surround does work, snd-hda-intel dies on volume change
by Ma Begaj 28 Sep '07

28 Sep '07

Hi, Sorry if this email is not for devel mailing list, but this sounds to me like a bug or development problem. If I am wrong, I will post it on the user list. And for the long email ... I just wanted to describe everything I tried. I found a lot of messages about problems with sound chip AD1986A on motherboard Asus M2NPV-VM. It is almost always the same problem, surround does not work and sound mutes (module dies) when the user change volume. That is my problem too. I found a lot of solutions by using "model=3stack", sometimes with "position_fix=1", but that does not work for all the users. I checked the list, I read dozens of emails, but nothing helped. I found a few patches on the list, but all of them were for kernels < 2.6.22. I used Slackware 12.0 and model=3stack caused a terrible "weeping" sound on the rear left speaker, but the stereo worked. I read a lot of success stories with this motherboard and Ubuntu and after playing whole week long on Slackware (kernel 2.6.21.5 and 2.6.22.6), I decided to try it with Ubuntu Feisty (2.6.20 and 2.6.22.9), although I did not believe that Ubuntu should help. It actually helped partly because the weeping sound disappeared, but I had only stereo. When I changed the "Channel Mode" jack in alsamixer from 2ch to 6ch, sound muted and module died. The only way to get the sound back was to stop playing file, and rmmod/modprobe snd_hda_intel again. Using model=6stack gives me surround sound by using "speaker-test -c6 ..." (nice female voice guessed right every speaker) but module was still dying as soon as I changed volume. Does someone have an idea what to do with this chip ... I am really desperate and beginning to hate this great motherboard ... Thanks a lot. M.

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[alsa-devel] Userspace drivers
by Josh Litherland 27 Sep '07

27 Sep '07

Is there any facility for developing an ALSA device driver in userspace? I'm particularly interested in mixer devices; my goal is to expose the Thinkpad hardware volume buttons as an ALSA mixer. Thanks for any pointers! -- Josh Litherland (josh(a)temp123.org)

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[alsa-devel] 1.0.15rc3
by Jaroslav Kysela 26 Sep '07

26 Sep '07

Hello all, I put 1.0.15rc3 packages (driver, library) to the ALSA FTP server. Please, report any critical bugs to release 1.0.15 final. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] tiwai downtime again
by Takashi Iwai 25 Sep '07

25 Sep '07

Hi, I'll be on vacation for two weeks from today (mostly because of movement to a new apartment, no beach with a cocktail in shiny south islands), so likely will be unresponsive to e-mails during that. If your patches weren't be reviewed/applied, don't hesitate to repost later. Thanks, Takashi

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[alsa-devel] snd_hda_intel bug
by Pablo Yaggi 25 Sep '07

25 Sep '07

Please let me know if a bug report must be placed somewhere else because i get no reply about this issue and i don't know if the snd_hda_intel driver maintainer received it. The bug repost: i bought a sony vaio fz19vn, i try both modes for the snd_hda_intel, vaio and vaio-ar, the same problem happen with both of them, when i plugin the headset internal speakers are not muted, so here is the card info: 00:00.0 0600: 8086:2a00 (rev 0c) Subsystem: 104d:9005 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:02.0 0300: 8086:2a02 (rev 0c) (prog-if 00 [VGA]) Subsystem: 104d:9005 Flags: bus master, fast devsel, latency 0, IRQ 5 Memory at fc000000 (64-bit, non-prefetchable) [size=1M] Memory at d0000000 (64-bit, prefetchable) [size=256M] I/O ports at 1800 [size=8] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [d0] Power Management version 3 00:02.1 0380: 8086:2a03 (rev 0c) Subsystem: 104d:9005 Flags: bus master, fast devsel, latency 0 Memory at fc100000 (64-bit, non-prefetchable) [size=1M] Capabilities: [d0] Power Management version 3 00:1a.0 0c03: 8086:2834 (rev 03) (prog-if 00 [UHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 1820 [size=32] 00:1a.1 0c03: 8086:2835 (rev 03) (prog-if 00 [UHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 1840 [size=32] 00:1a.7 0c03: 8086:283a (rev 03) (prog-if 20 [EHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 18 Memory at fc604800 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 104d:9005 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at fc400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) 00:1c.0 0604: 8086:283f (rev 03) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=03, sec-latency=0 I/O behind bridge: 00002000-00002fff Memory behind bridge: f6000000-f7ffffff Prefetchable memory behind bridge: 00000000f0000000-00000000f1ffffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 104d:9005 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.1 0604: 8086:2841 (rev 03) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=05, sec-latency=0 I/O behind bridge: 00003000-00003fff Memory behind bridge: f8000000-f9ffffff Prefetchable memory behind bridge: 00000000f2000000-00000000f3ffffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 104d:9005 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.2 0604: 8086:2843 (rev 03) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=06, subordinate=07, sec-latency=0 I/O behind bridge: 00004000-00004fff Memory behind bridge: fa000000-fbffffff Prefetchable memory behind bridge: 00000000f4000000-00000000f5ffffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 104d:9005 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1c.4 0604: 8086:2847 (rev 03) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=08, subordinate=08, sec-latency=0 I/O behind bridge: 00005000-00005fff Memory behind bridge: fc200000-fc2fffff Capabilities: [40] Express Root Port (Slot+) IRQ 0 Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable+ Capabilities: [90] Subsystem: 104d:9005 Capabilities: [a0] Power Management version 2 Capabilities: [100] Virtual Channel Capabilities: [180] Unknown (5) 00:1d.0 0c03: 8086:2830 (rev 03) (prog-if 00 [UHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 22 I/O ports at 1860 [size=32] 00:1d.1 0c03: 8086:2831 (rev 03) (prog-if 00 [UHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 1880 [size=32] 00:1d.2 0c03: 8086:2832 (rev 03) (prog-if 00 [UHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 18a0 [size=32] 00:1d.7 0c03: 8086:2836 (rev 03) (prog-if 20 [EHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 22 Memory at fc604c00 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:2448 (rev f3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=09, subordinate=0d, sec-latency=56 I/O behind bridge: 00006000-00006fff Memory behind bridge: fc300000-fc3fffff Prefetchable memory behind bridge: 0000000088000000-000000008bffffff Capabilities: [50] Subsystem: 104d:9005 00:1f.0 0601: 8086:2815 (rev 03) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:1f.1 0101: 8086:2850 (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 1810 [size=16] 00:1f.2 0106: 8086:2829 (rev 03) (prog-if 01 [AHCI 1.0]) Subsystem: 104d:9005 Flags: bus master, 66MHz, medium devsel, latency 0, IRQ 219 I/O ports at 1c00 [size=8] I/O ports at 18d4 [size=4] I/O ports at 18d8 [size=8] I/O ports at 18d0 [size=4] I/O ports at 18e0 [size=32] Memory at fc604000 (32-bit, non-prefetchable) [size=2K] Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/2 Enable+ Capabilities: [70] Power Management version 3 Capabilities: [a8] #12 [0010] 00:1f.3 0c05: 8086:283e (rev 03) Subsystem: 104d:9005 Flags: medium devsel, IRQ 19 Memory at 8c000000 (32-bit, non-prefetchable) [size=256] I/O ports at 1c20 [size=32] 06:00.0 0280: 8086:4222 (rev 02) Subsystem: 8086:1051 Flags: bus master, fast devsel, latency 0, IRQ 10 Memory at fa000000 (32-bit, non-prefetchable) [size=4K] Capabilities: [c8] Power Management version 2 Capabilities: [d0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting Capabilities: [140] Device Serial Number ba-8c-73-ff-ff-77-1b-00 08:00.0 0200: 11ab:4351 (rev 16) Subsystem: 104d:9005 Flags: bus master, fast devsel, latency 0, IRQ 218 Memory at fc200000 (64-bit, non-prefetchable) [size=16K] I/O ports at 5000 [size=256] Capabilities: [48] Power Management version 2 Capabilities: [50] Vital Product Data Capabilities: [5c] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable+ Capabilities: [e0] Express Legacy Endpoint IRQ 0 Capabilities: [100] Advanced Error Reporting 09:03.0 0607: 104c:8039 Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at fc300000 (32-bit, non-prefetchable) [size=4K] Bus: primary=09, secondary=0a, subordinate=0d, sec-latency=176 Memory window 0: 88000000-8bfff000 (prefetchable) Memory window 1: 90000000-93fff000 I/O window 0: 00006000-000060ff I/O window 1: 00006400-000064ff 16-bit legacy interface ports at 0001 09:03.1 0c00: 104c:803a (prog-if 10 [OHCI]) Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 32, IRQ 16 Memory at fc302000 (32-bit, non-prefetchable) [size=2K] Memory at fc304000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 09:03.2 0180: 104c:803b Subsystem: 104d:9005 Flags: bus master, medium devsel, latency 57, IRQ 18 Memory at fc301000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 *************************** /proc/asound/card0/codec#1: *************************** Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x104d1700 Revision Id: 0x100000 Modem Function Group: 0x2 *************************** /proc/asound/card0/codec#0 : *************************** Codec: SigmaTel STAC9872AK Address: 0 Vendor Id: 0x83847662 Subsystem Id: 0x104d2300 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Power: 0x0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x07 Node 0x07 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x0e Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x09 Node 0x09 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x0c 0x0c] Connection: 1 0x15 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173c: IN OUT HP Detect Pin Default 0x04214020: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0814: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0814: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173c: IN OUT HP Detect Pin Default 0x04a15030: [Jack] Mic at Ext Right Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0814: OUT Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x05 Node 0x10 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x3e0]: 44100 48000 88200 96000 176400 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x11 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 2 0x10* 0x09 Node 0x12 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x13 Node 0x13 [Pin Complex] wcaps 0x440381: Stereo Digital Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x90a7013e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=1 Amp-Out vals: [0x00 0x00] Connection: 4 0x0a 0x0d* 0x14 0x02 Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x17 [Volume Knob Widget] wcaps 0x600000: Mono Node 0x18 [Audio Output] wcaps 0x40201: Stereo Digital _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] New Aztech Sound Galaxy driver
by Rene Herman 25 Sep '07

25 Sep '07

Hi Takashi. This is a new driver for Aztech Sound Galaxy ISA soundcards based on the AZT1605 and AZT2316 chipsets. It's constructed as two seperate drivers for either chipset generated from the same source file, with (very) minimal ifdeffery. The drivers check the SB DSP version to decide if they are being loaded for the right chip. AZT1605 returns 2.1 by default, and AZT2316 3.1. This isn't full-proof as the DSP version can actually be set through software but it's close enough -- as far as I've been able to see, the DSP version can not be stored in the EEPROM and the cards will therefore startup with the defaults. This distinction could (with the same success rate) also be used to decide which chip we're looking at at runtime meaning a single, merged driver is also an option but I feel it's actually nicer this way. A merged driver would have to postpone translating the passed in resource values to the card configuration until it knew which one it was looking at and would need to postpone erring out on mpu_irq=10 for azt1605 and mpu_irq=3 for azt2316. If you differ, I will merge them, but I'd rather keep them seperate. The drivers have been tested on various cards. For snd-azt1605: FCC-ID I38-MMSN811: Aztech Sound Galaxy Nova 16 Extra FCC-ID I38-MMSN822: Aztech Sound Galaxy Pro 16 II and for snd-azt2316: FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB FCC-ID I38-MMSN826: Trust Sound Expert DeLuxe Wave 32 (05201) FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ (05202) FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D (06301) FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D I38-MMSN826 is also known as "Aztech Sound Galaxy Waverider 32+". All cards work fully -- full-duplex PCM, MIDI and FM. Full-duplex is a little flaky on some. I38-MSN811 tends to not work in full duplex, but sometimes does with the highest success rate being achieved when you first start the capture and then a playback instead of the other way around (it's a CS4231-KL codec). The cards with an AD1845XP codec (my I38-MMSN826 and one of my I38-MMSN830s) are also somewhat duplex-challenged. Sometimes full-duplex works, sometimes not and this varies from try to try. This seems likely to be a timing problem somewhere inside cs4231-lib. I38-MMSN826 has an additional "ICS2115 WaveFront" wavetable synth onboard that isn't supported yet. The wavetable synths on I38-MMSN847 and I38-MMSN852 are wired directly to the standard MPU-401 UART and the AUX1 input on the codec and work without problem. CD-ROM audio on the cards is routed to the codec "Line" input, Line-In to its Aux input, and FM/Wavetable to its AUX1 input. I did not rename the controls due to the capture source enumeration: I see that capture-source overrides are hardcoded in cs4231-lib and this is just too ugly to live. Versus the old snd-sgalaxy driver, these drivers add support for the models without a configuration EEPROM (common), full-duplex, MPU-401 UART and OPL3. In the future they should grow support for that ICS2115 WaveFront synth on 826 and an hwdep interface to write to the EEPROM on the models that have one. lib.c was split off in preparation of having snd-azt2320 use it as well and generally containing non soundgalaxy specific code that might at some point make it into an isa/lib.c or something. It's tiny, so it's not constructed as a helper module but simply compiled into both drivers. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> [ all of the above is intended as a changelog ] Given that they are new drivers with new names, they shouldn't disturb anything and can as far as I'm concerned make 1.0.15 but if you feel it's too late (and/or don't want the two seperate drivers), so be it. I can also submit a next patch removing the old sgalaxy driver now or just wait with that one... Patch is against current alsa-kernel hg. Rene. diff -r 0687faf0e009 Documentation/ALSA-Configuration.txt --- a/Documentation/ALSA-Configuration.txt Wed Sep 19 21:20:17 2007 +0200 +++ b/Documentation/ALSA-Configuration.txt Thu Sep 20 14:15:22 2007 +0200 @@ -296,6 +296,74 @@ Prior to version 0.9.0rc4 options had a Note: Some ALSA mixer applications don't handle the SPDIF sample rate control correctly. If you have problems regarding this, try another ALSA compliant mixer (alsamixer works). + + Module snd-azt1605 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + + Module snd-azt2316 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. Module snd-azt2320 ------------------ diff -r 0687faf0e009 isa/Kconfig --- a/isa/Kconfig Wed Sep 19 21:20:17 2007 +0200 +++ b/isa/Kconfig Thu Sep 20 14:15:22 2007 +0200 @@ -76,6 +76,32 @@ config SND_ALS100 To compile this driver as a module, choose M here: the module will be called snd-als100. +config SND_AZT1605 + tristate "Aztech AZT1605 Driver" + depends on SND + select SND_CS4231_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + help + Say Y here to include support for Aztech Sound Galaxy cards + based on the AZT1605 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-azt1605. + +config SND_AZT2316 + tristate "Aztech AZT2316 Driver" + depends on SND + select SND_CS4231_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + help + Say Y here to include support for Aztech Sound Galaxy cards + based on the AZT2316 chipset. + + To compile this driver as a module, choose M here: the module + will be called snd-azt2316. + config SND_AZT2320 tristate "Aztech Systems AZT2320" depends on SND && PNP && ISA diff -r 0687faf0e009 isa/Makefile --- a/isa/Makefile Wed Sep 19 21:20:17 2007 +0200 +++ b/isa/Makefile Thu Sep 20 14:15:22 2007 +0200 @@ -26,5 +26,5 @@ obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o -obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ +obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ galaxy/ gus/ opti9xx/ \ sb/ wavefront/ diff -r 0687faf0e009 isa/galaxy/Makefile --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/isa/galaxy/Makefile Thu Sep 20 14:15:22 2007 +0200 @@ -0,0 +1,10 @@ +# +# Makefile for ALSA +# Copyright (c) 2001 by Jaroslav Kysela <perex(a)suse.cz> +# + +snd-azt1605-objs := azt1605.o lib.o +snd-azt2316-objs := azt2316.o lib.o + +obj-$(CONFIG_SND_AZT1605) += snd-azt1605.o +obj-$(CONFIG_SND_AZT2316) += snd-azt2316.o diff -r 0687faf0e009 isa/galaxy/azt1605.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/isa/galaxy/azt1605.c Thu Sep 20 14:15:22 2007 +0200 @@ -0,0 +1,91 @@ +/* + * Aztech AZT1605 Driver + * Copyright (C) 2007 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#define AZT1605 + +#define CRD_NAME "Aztech AZT1605" +#define DRV_NAME "AZT1605" +#define DEV_NAME "azt1605" + +#define GALAXY_DSP_MAJOR 2 +#define GALAXY_DSP_MINOR 1 + +#define GALAXY_CONFIG_SIZE 3 + +/* + * 24-bit config register + */ + +#define GALAXY_CONFIG_SBA_220 (0 << 0) +#define GALAXY_CONFIG_SBA_240 (1 << 0) +#define GALAXY_CONFIG_SBA_260 (2 << 0) +#define GALAXY_CONFIG_SBA_280 (3 << 0) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_MPUA_300 (0 << 2) +#define GALAXY_CONFIG_MPUA_330 (1 << 2) + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 3) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 4) + +#define GALAXY_CONFIG_CD_PANASONIC (1 << 5) +#define GALAXY_CONFIG_CD_MITSUMI (1 << 6) +#define GALAXY_CONFIG_CD_MASK (\ + GALAXY_CONFIG_CD_PANASONIC | GALAXY_CONFIG_CD_MITSUMI) + +#define GALAXY_CONFIG_UNUSED (1 << 7) +#define GALAXY_CONFIG_UNUSED_MASK GALAXY_CONFIG_UNUSED + +#define GALAXY_CONFIG_SBIRQ_2 (1 << 8) +#define GALAXY_CONFIG_SBIRQ_3 (1 << 9) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 10) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 11) + +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 12) +#define GALAXY_CONFIG_MPUIRQ_3 (1 << 13) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 14) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 15) + +#define GALAXY_CONFIG_WSSA_530 (0 << 16) +#define GALAXY_CONFIG_WSSA_604 (1 << 16) +#define GALAXY_CONFIG_WSSA_E80 (2 << 16) +#define GALAXY_CONFIG_WSSA_F40 (3 << 16) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 18) + +#define GALAXY_CONFIG_CDIRQ_11 (1 << 19) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 20) +#define GALAXY_CONFIG_CDIRQ_15 (1 << 21) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_11 | GALAXY_CONFIG_CDIRQ_12 |\ + GALAXY_CONFIG_CDIRQ_15) + +#define GALAXY_CONFIG_CDDMA_DISABLE (0 << 22) +#define GALAXY_CONFIG_CDDMA_0 (1 << 22) +#define GALAXY_CONFIG_CDDMA_1 (2 << 22) +#define GALAXY_CONFIG_CDDMA_3 (3 << 22) +#define GALAXY_CONFIG_CDDMA_MASK GALAXY_CONFIG_CDDMA_3 + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CD_MASK |\ + GALAXY_CONFIG_UNUSED_MASK | GALAXY_CONFIG_CDIRQ_MASK |\ + GALAXY_CONFIG_CDDMA_MASK) + +#include "galaxy.c" diff -r 0687faf0e009 isa/galaxy/azt2316.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/isa/galaxy/azt2316.c Thu Sep 20 14:15:22 2007 +0200 @@ -0,0 +1,111 @@ +/* + * Aztech AZT2316 Driver + * Copyright (C) 2007 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#define AZT2316 + +#define CRD_NAME "Aztech AZT2316" +#define DRV_NAME "AZT2316" +#define DEV_NAME "azt2316" + +#define GALAXY_DSP_MAJOR 3 +#define GALAXY_DSP_MINOR 1 + +#define GALAXY_CONFIG_SIZE 4 + +/* + * 32-bit config register + */ + +#define GALAXY_CONFIG_SBA_220 (0 << 0) +#define GALAXY_CONFIG_SBA_240 (1 << 0) +#define GALAXY_CONFIG_SBA_260 (2 << 0) +#define GALAXY_CONFIG_SBA_280 (3 << 0) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_SBIRQ_2 (1 << 2) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 3) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 4) +#define GALAXY_CONFIG_SBIRQ_10 (1 << 5) + +#define GALAXY_CONFIG_SBDMA_DISABLE (0 << 6) +#define GALAXY_CONFIG_SBDMA_0 (1 << 6) +#define GALAXY_CONFIG_SBDMA_1 (2 << 6) +#define GALAXY_CONFIG_SBDMA_3 (3 << 6) + +#define GALAXY_CONFIG_WSSA_530 (0 << 8) +#define GALAXY_CONFIG_WSSA_604 (1 << 8) +#define GALAXY_CONFIG_WSSA_E80 (2 << 8) +#define GALAXY_CONFIG_WSSA_F40 (3 << 8) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 10) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 11) + +#define GALAXY_CONFIG_MPUA_300 (0 << 12) +#define GALAXY_CONFIG_MPUA_330 (1 << 12) + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 13) + +#define GALAXY_CONFIG_CDA_310 (0 << 14) +#define GALAXY_CONFIG_CDA_320 (1 << 14) +#define GALAXY_CONFIG_CDA_340 (2 << 14) +#define GALAXY_CONFIG_CDA_350 (3 << 14) +#define GALAXY_CONFIG_CDA_MASK GALAXY_CONFIG_CDA_350 + +#define GALAXY_CONFIG_CD_DISABLE (0 << 16) +#define GALAXY_CONFIG_CD_PANASONIC (1 << 16) +#define GALAXY_CONFIG_CD_SONY (2 << 16) +#define GALAXY_CONFIG_CD_MITSUMI (3 << 16) +#define GALAXY_CONFIG_CD_AZTECH (4 << 16) +#define GALAXY_CONFIG_CD_UNUSED_5 (5 << 16) +#define GALAXY_CONFIG_CD_UNUSED_6 (6 << 16) +#define GALAXY_CONFIG_CD_UNUSED_7 (7 << 16) +#define GALAXY_CONFIG_CD_MASK GALAXY_CONFIG_CD_UNUSED_7 + +#define GALAXY_CONFIG_CDDMA8_DISABLE (0 << 20) +#define GALAXY_CONFIG_CDDMA8_0 (1 << 20) +#define GALAXY_CONFIG_CDDMA8_1 (2 << 20) +#define GALAXY_CONFIG_CDDMA8_3 (3 << 20) +#define GALAXY_CONFIG_CDDMA8_MASK GALAXY_CONFIG_CDDMA8_3 + +#define GALAXY_CONFIG_CDDMA16_DISABLE (0 << 22) +#define GALAXY_CONFIG_CDDMA16_5 (1 << 22) +#define GALAXY_CONFIG_CDDMA16_6 (2 << 22) +#define GALAXY_CONFIG_CDDMA16_7 (3 << 22) +#define GALAXY_CONFIG_CDDMA16_MASK GALAXY_CONFIG_CDDMA16_7 + +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 24) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 25) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 26) +#define GALAXY_CONFIG_MPUIRQ_10 (1 << 27) + +#define GALAXY_CONFIG_CDIRQ_5 (1 << 28) +#define GALAXY_CONFIG_CDIRQ_11 (1 << 29) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 30) +#define GALAXY_CONFIG_CDIRQ_15 (1 << 31) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_5 | GALAXY_CONFIG_CDIRQ_11 |\ + GALAXY_CONFIG_CDIRQ_12 | GALAXY_CONFIG_CDIRQ_15) + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CDA_MASK |\ + GALAXY_CONFIG_CD_MASK | GALAXY_CONFIG_CDDMA16_MASK |\ + GALAXY_CONFIG_CDDMA8_MASK | GALAXY_CONFIG_CDIRQ_MASK) + +#include "galaxy.c" diff -r 0687faf0e009 isa/galaxy/galaxy.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/isa/galaxy/galaxy.c Thu Sep 20 14:15:22 2007 +0200 @@ -0,0 +1,562 @@ +/* + * Aztech AZT1605/AZT2316 Driver + * Copyright (C) 2007 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <sound/driver.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/isa.h> +#include <linux/delay.h> +#include <asm/io.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/cs4231.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include "lib.h" + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rene Herman"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); + +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param_array(wss_port, long, NULL, 0444); +MODULE_PARM_DESC(wss_port, "WSS port # for " CRD_NAME " driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "FM port # for " CRD_NAME " driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "Playback DMA # for " CRD_NAME " driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "Capture DMA # for " CRD_NAME " driver."); + +#define GALAXY_PORT_CONFIG 1024 +#define CONFIG_PORT_SET 4 + +#define DSP_COMMAND_GALAXY_8 8 +#define GALAXY_COMMAND_GET_TYPE 5 + +#define DSP_COMMAND_GALAXY_9 9 +#define GALAXY_COMMAND_WSSMODE 0 +#define GALAXY_COMMAND_SB8MODE 1 + +#define GALAXY_MODE_WSS GALAXY_COMMAND_WSSMODE +#define GALAXY_MODE_SB8 GALAXY_COMMAND_SB8MODE + +struct snd_galaxy { + void __iomem *port; + void __iomem *config_port; + void __iomem *wss_port; + u32 config; + struct resource *res_port; + struct resource *res_config_port; + struct resource *res_wss_port; +}; + +static u32 config[SNDRV_CARDS]; +static u8 wss_config[SNDRV_CARDS]; + +static int __devinit snd_galaxy_match(struct device *dev, unsigned int n) +{ + if (!enable[n]) + return 0; + + switch (port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + return 0; + case 0x220: + config[n] |= GALAXY_CONFIG_SBA_220; + break; + case 0x240: + config[n] |= GALAXY_CONFIG_SBA_240; + break; + case 0x260: + config[n] |= GALAXY_CONFIG_SBA_260; + break; + case 0x280: + config[n] |= GALAXY_CONFIG_SBA_280; + break; + default: + snd_printk(KERN_ERR "%s: invalid port %#lx\n", dev->bus_id, + port[n]); + return 0; + } + + switch (wss_port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_ERR "%s: please specify wss_port\n", + dev->bus_id); + return 0; + case 0x530: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_530; + break; + case 0x604: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_604; + break; + case 0xe80: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_E80; + break; + case 0xf40: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_F40; + break; + default: + snd_printk(KERN_ERR "%s: invalid WSS port %#lx\n", dev->bus_id, + wss_port[n]); + return 0; + } + + switch (irq[n]) { + case SNDRV_AUTO_IRQ: + snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + return 0; + case 7: + wss_config[n] |= WSS_CONFIG_IRQ_7; + break; + case 2: + irq[n] = 9; + case 9: + wss_config[n] |= WSS_CONFIG_IRQ_9; + break; + case 10: + wss_config[n] |= WSS_CONFIG_IRQ_10; + break; + case 11: + wss_config[n] |= WSS_CONFIG_IRQ_11; + break; + default: + snd_printk(KERN_ERR "%s: invalid IRQ %d\n", dev->bus_id, + irq[n]); + return 0; + } + + switch (dma1[n]) { + case SNDRV_AUTO_DMA: + snd_printk(KERN_ERR "%s: please specify dma1\n", + dev->bus_id); + return 0; + case 0: + wss_config[n] |= WSS_CONFIG_DMA_0; + break; + case 1: + wss_config[n] |= WSS_CONFIG_DMA_1; + break; + case 3: + wss_config[n] |= WSS_CONFIG_DMA_3; + break; + default: + snd_printk(KERN_ERR "%s: invalid playback DMA %d\n", + dev->bus_id, dma1[n]); + return 0; + } + + if (dma2[n] == SNDRV_AUTO_DMA || dma2[n] == dma1[n]) { + dma2[n] = -1; + goto mpu; + } + + wss_config[n] |= WSS_CONFIG_DUPLEX; + switch (dma2[n]) { + case 1: + if (dma1[n] == 0) + case 0: + break; + default: + snd_printk(KERN_ERR "%s: invalid capture DMA %d\n", + dev->bus_id, dma2[n]); + return 0; + } + +mpu: + switch (mpu_port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_WARNING "%s: mpu_port not specified; not using " + "MPU-401\n", dev->bus_id); + mpu_port[n] = -1; + goto fm; + case 0x300: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_300; + break; + case 0x330: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_330; + break; + default: + snd_printk(KERN_ERR "%s: invalid mpu port %#lx\n", dev->bus_id, + mpu_port[n]); + return 0; + } + + switch (mpu_irq[n]) { + case SNDRV_AUTO_IRQ: + snd_printk(KERN_WARNING "%s: mpu_irq not specified; using " + "polling mode\n", dev->bus_id); + mpu_irq[n] = -1; + break; + case 2: + mpu_irq[n] = 9; + case 9: + config[n] |= GALAXY_CONFIG_MPUIRQ_2; + break; +#ifdef AZT1605 + case 3: + config[n] |= GALAXY_CONFIG_MPUIRQ_3; + break; +#endif + case 5: + config[n] |= GALAXY_CONFIG_MPUIRQ_5; + break; + case 7: + config[n] |= GALAXY_CONFIG_MPUIRQ_7; + break; +#ifdef AZT2316 + case 10: + config[n] |= GALAXY_CONFIG_MPUIRQ_10; + break; +#endif + default: + snd_printk(KERN_ERR "%s: invalid MPU IRQ %d\n", dev->bus_id, + mpu_irq[n]); + return 0; + } + + if (mpu_irq[n] == irq[n]) { + snd_printk(KERN_ERR "%s: cannot share IRQ between WSS and " + "MPU-401\n", dev->bus_id); + return 0; + } + +fm: + switch (fm_port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_WARNING "%s: fm_port not specified: not using " + "OPL3\n", dev->bus_id); + fm_port[n] = -1; + break; + case 0x388: + break; + default: + snd_printk(KERN_ERR "%s: illegal FM port %#lx\n", dev->bus_id, + fm_port[n]); + return 0; + } + + config[n] |= GALAXY_CONFIG_GAME_ENABLE; + return 1; +} + +static int __devinit __galaxy_init(struct snd_galaxy *galaxy, u8 *type) +{ + u8 major; + u8 minor; + int err; + + err = __dsp_reset(galaxy->port); + if (err < 0) + return err; + + err = __dsp_get_version(galaxy->port, &major, &minor); + if (err < 0) + return err; + + if (major != GALAXY_DSP_MAJOR || minor != GALAXY_DSP_MINOR) + return -ENODEV; + + err = __dsp_command(galaxy->port, DSP_COMMAND_GALAXY_8); + if (err < 0) + return err; + + err = __dsp_command(galaxy->port, GALAXY_COMMAND_GET_TYPE); + if (err < 0) + return err; + + err = __dsp_get_byte(galaxy->port, type); + if (err < 0) + return err; + + return 0; +} + +static int __devinit __galaxy_set_mode(struct snd_galaxy *galaxy, u8 mode) +{ + int err; + + err = __dsp_command(galaxy->port, DSP_COMMAND_GALAXY_9); + if (err < 0) + return err; + + err = __dsp_command(galaxy->port, mode); + if (err < 0) + return err; + +#ifdef AZT1605 + /* + * Needed for MPU IRQ on AZT1605, but AZT2316 loses WSS again + */ + err = __dsp_reset(galaxy->port); + if (err < 0) + return err; +#endif + + return 0; +} + +static void __galaxy_set_config(struct snd_galaxy *galaxy, u32 config) +{ + u8 tmp = ioread8(galaxy->config_port + CONFIG_PORT_SET); + int i; + + iowrite8(tmp | 0x80, galaxy->config_port + CONFIG_PORT_SET); + for (i = 0; i < GALAXY_CONFIG_SIZE; i++) { + iowrite8(config, galaxy->config_port + i); + config >>= 8; + } + iowrite8(tmp & 0x7f, galaxy->config_port + CONFIG_PORT_SET); + msleep(10); +} + +static void __devinit __galaxy_config(struct snd_galaxy *galaxy, u32 config) +{ + int i; + + for (i = GALAXY_CONFIG_SIZE; i; i--) { + u8 tmp = ioread8(galaxy->config_port + i - 1); + galaxy->config = (galaxy->config << 8) | tmp; + } + config |= galaxy->config & GALAXY_CONFIG_MASK; + __galaxy_set_config(galaxy, config); +} + +static int __devinit __galaxy_wss_config(struct snd_galaxy *galaxy, + u8 wss_config) +{ + int err; + + err = __wss_detect(galaxy->wss_port); + if (err < 0) + return err; + + __wss_set_config(galaxy->wss_port, wss_config); + + err = __galaxy_set_mode(galaxy, GALAXY_MODE_WSS); + if (err < 0) + return err; + + return 0; +} + +static void snd_galaxy_free(struct snd_card *card) +{ + struct snd_galaxy *galaxy = card->private_data; + + if (galaxy->wss_port) { + __wss_set_config(galaxy->wss_port, 0); + ioport_unmap(galaxy->wss_port); + release_and_free_resource(galaxy->res_wss_port); + } + if (galaxy->config_port) { + __galaxy_set_config(galaxy, galaxy->config); + ioport_unmap(galaxy->config_port); + release_and_free_resource(galaxy->res_config_port); + } + if (galaxy->port) { + ioport_unmap(galaxy->port); + release_and_free_resource(galaxy->res_port); + } +} + +static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) +{ + struct snd_galaxy *galaxy; + struct snd_cs4231 *chip; + struct snd_card *card; + u8 type; + int err; + + card = snd_card_new(index[n], id[n], THIS_MODULE, sizeof *galaxy); + if (!card) + return -EINVAL; + + snd_card_set_dev(card, dev); + + card->private_free = snd_galaxy_free; + galaxy = card->private_data; + + galaxy->res_port = request_region(port[n], 16, DRV_NAME); + if (!galaxy->res_port) { + snd_printk(KERN_ERR "%s: could not grab ports %#lx-%#lx\n", + dev->bus_id, port[n], port[n] + 15); + err = -EBUSY; + goto error; + } + galaxy->port = ioport_map(port[n], 16); + + err = __galaxy_init(galaxy, &type); + if (err < 0) { + snd_printk(KERN_ERR "%s: no Sound Galaxy found at %#lx\n", + dev->bus_id, port[n]); + goto error; + } + snd_printk(KERN_INFO "%s: Sound Galaxy (type %d) found at %#lx\n", + dev->bus_id, type, port[n]); + + galaxy->res_config_port = request_region(port[n] + GALAXY_PORT_CONFIG, + 16, DRV_NAME); + if (!galaxy->res_config_port) { + snd_printk(KERN_ERR "%s: could not grab ports %#lx-%#lx\n", + dev->bus_id, port[n] + GALAXY_PORT_CONFIG, + port[n] + GALAXY_PORT_CONFIG + 15); + err = -EBUSY; + goto error; + } + galaxy->config_port = ioport_map(port[n] + GALAXY_PORT_CONFIG, 16); + + __galaxy_config(galaxy, config[n]); + + galaxy->res_wss_port = request_region(wss_port[n], 4, DRV_NAME); + if (!galaxy->res_wss_port) { + snd_printk(KERN_ERR "%s: could not grab ports %#lx-%#lx\n", + dev->bus_id, wss_port[n], wss_port[n] + 3); + err = -EBUSY; + goto error; + } + galaxy->wss_port = ioport_map(wss_port[n], 4); + + err = __galaxy_wss_config(galaxy, wss_config[n]); + if (err < 0) { + snd_printk(KERN_ERR "%s: could not configure WSS\n", + dev->bus_id); + goto error; + } + + strcpy(card->driver, DRV_NAME); + strcpy(card->shortname, DRV_NAME); + sprintf(card->longname, "%s at %#lx/%#lx, irq %d, dma %d/%d", + card->shortname, port[n], wss_port[n], irq[n], dma1[n], + dma2[n]); + + err = snd_cs4231_create(card, wss_port[n] + 4, -1, irq[n], dma1[n], + dma2[n], CS4231_HW_DETECT, 0, &chip); + if (err < 0) + goto error; + + err = snd_cs4231_pcm(chip, 0, NULL); + if (err < 0) + goto error; + + err = snd_cs4231_mixer(chip); + if (err < 0) + goto error; + + err = snd_cs4231_timer(chip, 0, NULL); + if (err < 0) + goto error; + + if (mpu_port[n] >= 0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port[n], 0, mpu_irq[n], + IRQF_DISABLED, NULL); + if (err < 0) + goto error; + } + + if (fm_port[n] >= 0) { + struct snd_opl3 *opl3; + + err = snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_AUTO, 0, &opl3); + if (err < 0) { + snd_printk(KERN_ERR "%s: no OPL device at %#lx\n", + dev->bus_id, fm_port[n]); + goto error; + } + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto error; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto error; + } + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(dev, card); + return 0; + +error: + snd_card_free(card); + return err; +} + +static int __devexit snd_galaxy_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_galaxy_driver = { + .match = snd_galaxy_match, + .probe = snd_galaxy_probe, + .remove = __devexit_p(snd_galaxy_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __init alsa_card_galaxy_init(void) +{ + return isa_register_driver(&snd_galaxy_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_galaxy_exit(void) +{ + isa_unregister_driver(&snd_galaxy_driver); +} + +module_init(alsa_card_galaxy_init); +module_exit(alsa_card_galaxy_exit); diff -r 0687faf0e009 isa/galaxy/lib.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/isa/galaxy/lib.c Thu Sep 20 14:15:22 2007 +0200 @@ -0,0 +1,118 @@ +/* + * Generic SB DSP / WSS support routines + * Copyright (C) 2007 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#include <linux/delay.h> +#include <asm/io.h> +#include <asm/processor.h> +#include "lib.h" + +/* + * Generic SB DSP + */ + +#define DSP_PORT_RESET 0x6 +#define DSP_PORT_READ 0xa +#define DSP_PORT_COMMAND 0xc +#define DSP_PORT_STATUS 0xc +#define DSP_PORT_DATA_AVAIL 0xe + +#define DSP_SIGNATURE 0xaa + +#define DSP_COMMAND_GET_VERSION 0xe1 + +int __devinit __dsp_get_byte(void __iomem *port, u8 *val) +{ + int loops = 1000; + + while (!(ioread8(port + DSP_PORT_DATA_AVAIL) & 0x80)) { + if (!loops--) + return -EIO; + cpu_relax(); + } + *val = ioread8(port + DSP_PORT_READ); + return 0; +} + +int __devinit __dsp_reset(void __iomem *port) +{ + u8 val; + + iowrite8(1, port + DSP_PORT_RESET); + udelay(10); + iowrite8(0, port + DSP_PORT_RESET); + + if (__dsp_get_byte(port, &val) < 0 || val != DSP_SIGNATURE) + return -ENODEV; + + return 0; +} + +int __devinit __dsp_command(void __iomem *port, u8 cmd) +{ + int loops = 1000; + + while (ioread8(port + DSP_PORT_STATUS) & 0x80) { + if (!loops--) + return -EIO; + cpu_relax(); + } + iowrite8(cmd, port + DSP_PORT_COMMAND); + return 0; +} + +int __devinit __dsp_get_version(void __iomem *port, u8 *major, u8 *minor) +{ + int err; + + err = __dsp_command(port, DSP_COMMAND_GET_VERSION); + if (err < 0) + return err; + + err = __dsp_get_byte(port, major); + if (err < 0) + return err; + + err = __dsp_get_byte(port, minor); + if (err < 0) + return err; + + return 0; +} + +/* + * Generic WSS + */ + +#define WSS_PORT_CONFIG 0 +#define WSS_PORT_SIGNATURE 3 + +#define WSS_SIGNATURE 4 + +int __devinit __wss_detect(void __iomem *wss_port) +{ + if ((ioread8(wss_port + WSS_PORT_SIGNATURE) & 0x3f) != WSS_SIGNATURE) + return -ENODEV; + + return 0; +} + +void __wss_set_config(void __iomem *wss_port, u8 wss_config) +{ + iowrite8(wss_config, wss_port + WSS_PORT_CONFIG); +} diff -r 0687faf0e009 isa/galaxy/lib.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/isa/galaxy/lib.h Thu Sep 20 14:15:22 2007 +0200 @@ -0,0 +1,52 @@ +/* + * Generic SB DSP / WSS support routines + * Copyright (C) 2007 Rene Herman + * + * This program is free software: you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program. If not, see <http://www.gnu.org/licenses/>. + * + */ + +#ifndef __SOUND_ISA_GALAXY_LIB_H +#define __SOUND_ISA_GALAXY_LIB_H + +#include <linux/compiler.h> +#include <linux/types.h> + +/* + * Generic SB DSP support routines + */ +int __dsp_get_byte(void __iomem *port, u8 *val); +int __dsp_reset(void __iomem *port); +int __dsp_command(void __iomem *port, u8 cmd); +int __dsp_get_version(void __iomem *port, u8 *major, u8 *minor); + +/* + * WSS 8-bit configuration register + */ +#define WSS_CONFIG_DMA_0 (1 << 0) +#define WSS_CONFIG_DMA_1 (2 << 0) +#define WSS_CONFIG_DMA_3 (3 << 0) +#define WSS_CONFIG_DUPLEX (1 << 2) +#define WSS_CONFIG_IRQ_7 (1 << 3) +#define WSS_CONFIG_IRQ_9 (2 << 3) +#define WSS_CONFIG_IRQ_10 (3 << 3) +#define WSS_CONFIG_IRQ_11 (4 << 3) + +/* + * Generic WSS support routines + */ +int __wss_detect(void __iomem *wss_port); +void __wss_set_config(void __iomem *wss_port, u8 wss_config); + +#endif /* __SOUND_ISA_GALAXY_LIB_H */

4 5

[alsa-devel] race conditions in alsa core
by Trent Piepho 24 Sep '07

24 Sep '07

In the process of trying to discover that kind of locking ALSA provides to a driver and what the driver must take care of itself, I think I have found some race conditions inside the alsa core. I think there are likely quite a few, but this example was one of the easiest to prove. ALSA uses unlocked_ioctl, which means ioctl calls aren't protected by the BKL and may run at the same time. Some ioctls, for instance HW_PARAMS, don't appear to have any locking done by ALSA. It's entirely possible for two hw_params calls to run at the same time on the same substream, and the ALSA core (and probably many drivers too) can't handle this. The following patch helps to create a larger window for the race condition, so that it can be triggered consistently. --- a/sound/core/pcm_native.c Tue Sep 04 09:02:06 2007 +0000 +++ b/sound/core/pcm_native.c Sat Sep 22 17:12:37 2007 -0700 @@ -411,6 +411,7 @@ static int snd_pcm_hw_params(struct snd_ runtime->channels = params_channels(params); runtime->rate = params_rate(params); runtime->period_size = params_period_size(params); + { static unsigned long x=1; if(test_and_change_bit(0, &x)) msleep(10); } runtime->periods = params_periods(params); runtime->buffer_size = params_buffer_size(params); runtime->tick_time = params_tick_time(params); This causes every other call to snd_pcm_hw_params() to have a 10ms delay between setting the period size and period count. If there are two calls within 10ms, the period count should come from the first call (which slept), while the period size will be from the second call. My test program will call the HW_PARAMS ioctl twice from different threads running at the same time. The first call will ask for the maximum number of periods with minimum size, the second call for the minimum number of periods of maximum size. When prepare is called, we discover that the runtime is set for the maximum number of periods of maximum size, which exceeds the buffer size. Of course even if it didn't exceed the buffer size, the params should be from one call or the other, not some random combination of both. I've attaching a very simple dummy ALSA driver which shows this. All it does is print timestamps and a few parameters when the various callback functions are invoked. The output looks like this: as_open - 0.084 us as_hw_params - 552.864 us periods 128, period size 1024 as_hw_params - 589.996 us periods 1, period size 131072 as_prepare - 11065.307 us periods 128, period size 131072 as_close - 11278.229 us Notice how in prepare the runtime has parameters that are a combination of the two hw_params calls. This test program should trigger the same race with any other ALSA driver. Just adjust the period count and size. The test program doesn't use the ALSA lib. It does need the headers from the alsa lib package to compile, since this seems to be the only place where user space headers for the device interface are provided.

2 2

[alsa-devel] Fundamental principle of audio mixing?
by Nobin Mathew 24 Sep '07

24 Sep '07

Can any body throw some lights on how audio mixing is done in software? Basic principle? Or Any link to web pages expianing this? Thanks & Regards Nobin Mathew

3 3

[alsa-devel] Emux hwdep ioctl requests
by Claudio Matsuoka 24 Sep '07

24 Sep '07

Hi, I'm writing an application for Emux patch loading using asfxload as reference, and noticed that there are many ioctl request definitions specified in the application instead of in a system header file. Should I do the same redefining SNDRV_EMUX_IOCTL_VERSION, SNDRV_EMUX_IOCTL_LOAD_PATCH, etc. or there are plans to place these in a common location? Also it seems that current Ubuntu doesn't have /usr/include/linux/awe_voice.h, has that been deprecated for some reason or it's just a bug in the distro?

1 0

[alsa-devel] OSS sequencer mode and SNDCTL_SEQ_CTRLRATE
by Claudio Matsuoka 24 Sep '07

24 Sep '07

Hi, Using the OSS sequencer mode of ALSA, I noticed that a request for SNDCTL_SEQ_CTRLRATE always returns 100 regardless of the kernel HZ value. I therefore assume that the OSS sequencer mode always works at this rate, unlike real OSS which varies with the kernel HZ, is that correct?

2 1

[alsa-devel] Sound trouble with MCP61 chipset.
by Steven Young 23 Sep '07

23 Sep '07

Hi, I had sent a message a couple of weeks ago regarding a somewhat reproducible error I'm having with onboard audio for systems with the MCP61 chipset. Specifically, the left channel doesn't work, and emits a high-pitched tone whenever the audio device is open. Also, adjusting the volume disables all sound until the kernel modules for sound are all reloaded. I didn't receive any response to that message; is this the right place to ask? I have two machines, both with the MCP61 chipset, and both with the same problem. I have included the lspci -v output of one of them below. Thanks, Steve. 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0 Capabilities: [44] HyperTransport: Slave or Primary Interface Capabilities: [dc] HyperTransport: MSI Mapping 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: 66MHz, fast devsel, IRQ 10 I/O ports at dc00 [size=64] I/O ports at 0600 [size=64] I/O ports at 0700 [size=64] Capabilities: [44] Power Management version 2 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: 66MHz, fast devsel 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: 66MHz, fast devsel Memory at 40100000 (32-bit, non-prefetchable) [disabled] [size=512K] 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a2) (prog-if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 16 Memory at dfe7f000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a2) (prog-if 20 [EHCI]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 19 Memory at dfe7ec00 (32-bit, non-prefetchable) [size=256] Capabilities: [44] Debug port Capabilities: [80] Power Management version 2 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) (prog-if 01 [Subtractive decode]) Flags: bus master, 66MHz, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=64 I/O behind bridge: 0000e000-0000efff Memory behind bridge: dff00000-dfffffff Prefetchable memory behind bridge: 40000000-400fffff Capabilities: [b8] Subsystem: nVidia Corporation Unknown device cb84 Capabilities: [8c] HyperTransport: MSI Mapping 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 19 Memory at dfe78000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) (prog-if 8a [Master SecP PriP]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0 [virtual] Memory at 000001f0 (32-bit, non-prefetchable) [disabled] [size=8] [virtual] Memory at 000003f0 (type 3, non-prefetchable) [disabled] [size=1] [virtual] Memory at 00000170 (32-bit, non-prefetchable) [disabled] [size=8] [virtual] Memory at 00000370 (type 3, non-prefetchable) [disabled] [size=1] I/O ports at ffa0 [size=16] Capabilities: [44] Power Management version 2 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at dfe7d000 (32-bit, non-prefetchable) [size=4K] I/O ports at d480 [size=8] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/3 Enable- Capabilities: [6c] HyperTransport: MSI Mapping 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) (prog-if 85 [Master SecO PriO]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 20 I/O ports at d400 [size=8] I/O ports at d080 [size=4] I/O ports at d000 [size=8] I/O ports at cc00 [size=4] I/O ports at c880 [size=16] Memory at dfe7c000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 Capabilities: [b0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/2 Enable- Capabilities: [cc] HyperTransport: MSI Mapping 00:08.1 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) (prog-if 85 [Master SecO PriO]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 16 I/O ports at c800 [size=8] I/O ports at c480 [size=4] I/O ports at c400 [size=8] I/O ports at c080 [size=4] I/O ports at c000 [size=16] Memory at dfe73000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 Capabilities: [b0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/2 Enable- Capabilities: [cc] HyperTransport: MSI Mapping 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Capabilities: [40] Subsystem: nVidia Corporation Unknown device 0000 Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable- Capabilities: [60] HyperTransport: MSI Mapping Capabilities: [80] Express Root Port (Slot+) IRQ 0 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 Capabilities: [40] Subsystem: nVidia Corporation Unknown device 0000 Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable- Capabilities: [60] HyperTransport: MSI Mapping Capabilities: [80] Express Root Port (Slot+) IRQ 0 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=0 Capabilities: [40] Subsystem: nVidia Corporation Unknown device 0000 Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable- Capabilities: [60] HyperTransport: MSI Mapping Capabilities: [80] Express Root Port (Slot+) IRQ 0 00:0d.0 VGA compatible controller: nVidia Corporation GeForce 6100 nForce 430 (rev a2) (prog-if 00 [VGA]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at de000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at dd000000 (64-bit, non-prefetchable) [size=16M] [virtual] Expansion ROM at dfe40000 [disabled] [size=128K] Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration Flags: fast devsel Capabilities: [80] HyperTransport: Host or Secondary Interface 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map Flags: fast devsel 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller Flags: fast devsel 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control Flags: fast devsel Capabilities: [f0] #0f [0010] 01:06.0 Ethernet controller: ADMtek NC100 Network Everywhere Fast Ethernet 10/100 (rev 11) Subsystem: Accton Technology Corporation Unknown device 1216 Flags: bus master, medium devsel, latency 64, IRQ 18 I/O ports at e800 [size=256] Memory at dffffc00 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 40000000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 01:07.0 FireWire (IEEE 1394): NEC Corporation uPD72874 IEEE1394 OHCI 1.1 3-port PHY-Link Ctrlr (rev 01) (prog-if 10 [OHCI]) Subsystem: NEC Corporation uPD72874 IEEE1394 OHCI 1.1 3-port PHY-Link Ctrlr Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at dfffe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [60] Power Management version 2

3 6

[alsa-devel] [PATCH] opti93x: conversion to use cs4231 library
by Krzysztof Helt 22 Sep '07

22 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch converts the Opti93x driver to use the cs4231 library instead of just duplicating the code. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch requires the patch which adds opti93x support to the cs4231 library. This patch is just first pass to remove the code not used any more and just hook the driver into the cs4231 library. It can be improved as well as made more coding style correct. It was tested with full-duplex audio on the opti931 card. Regards, Krzysztof diff -urp linux-2.6.22.old/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.23/sound/isa/opti9xx/opti92x-ad1848.c --- linux-2.6.22.old/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-08 13:37:59.000000000 +0200 +++ linux-2.6.23/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-22 21:12:32.000000000 +0200 @@ -34,15 +34,10 @@ #include <asm/io.h> #include <asm/dma.h> #include <sound/core.h> -#ifdef CS4231 +#if defined(CS4231) || defined(OPTi93X) #include <sound/cs4231.h> #else -#ifndef OPTi93X #include <sound/ad1848.h> -#else -#include <sound/control.h> -#include <sound/pcm.h> -#endif /* OPTi93X */ #endif /* CS4231 */ #include <sound/mpu401.h> #include <sound/opl3.h> @@ -110,7 +105,6 @@ module_param(dma2, int, 0444); MODULE_PARM_DESC(dma2, "2nd dma # for opti9xx driver."); #endif /* CS4231 || OPTi93X */ -#define OPTi9XX_HW_DETECT 0 #define OPTi9XX_HW_82C928 1 #define OPTi9XX_HW_82C929 2 #define OPTi9XX_HW_82C924 3 @@ -124,105 +118,12 @@ MODULE_PARM_DESC(dma2, "2nd dma # for op #ifdef OPTi93X -#define OPTi93X_INDEX 0x00 -#define OPTi93X_DATA 0x01 #define OPTi93X_STATUS 0x02 -#define OPTi93X_DDATA 0x03 #define OPTi93X_PORT(chip, r) ((chip)->port + OPTi93X_##r) -#define OPTi93X_MIXOUT_LEFT 0x00 -#define OPTi93X_MIXOUT_RIGHT 0x01 -#define OPTi93X_CD_LEFT_INPUT 0x02 -#define OPTi93X_CD_RIGHT_INPUT 0x03 -#define OPTi930_AUX_LEFT_INPUT 0x04 -#define OPTi930_AUX_RIGHT_INPUT 0x05 -#define OPTi931_FM_LEFT_INPUT 0x04 -#define OPTi931_FM_RIGHT_INPUT 0x05 -#define OPTi93X_DAC_LEFT 0x06 -#define OPTi93X_DAC_RIGHT 0x07 -#define OPTi93X_PLAY_FORMAT 0x08 -#define OPTi93X_IFACE_CONF 0x09 -#define OPTi93X_PIN_CTRL 0x0a -#define OPTi93X_ERR_INIT 0x0b -#define OPTi93X_ID 0x0c -#define OPTi93X_PLAY_UPR_CNT 0x0e -#define OPTi93X_PLAY_LWR_CNT 0x0f -#define OPTi931_AUX_LEFT_INPUT 0x10 -#define OPTi931_AUX_RIGHT_INPUT 0x11 -#define OPTi93X_LINE_LEFT_INPUT 0x12 -#define OPTi93X_LINE_RIGHT_INPUT 0x13 -#define OPTi93X_MIC_LEFT_INPUT 0x14 -#define OPTi93X_MIC_RIGHT_INPUT 0x15 -#define OPTi93X_OUT_LEFT 0x16 -#define OPTi93X_OUT_RIGHT 0x17 -#define OPTi93X_CAPT_FORMAT 0x1c -#define OPTi93X_CAPT_UPR_CNT 0x1e -#define OPTi93X_CAPT_LWR_CNT 0x1f - -#define OPTi93X_TRD 0x20 -#define OPTi93X_MCE 0x40 -#define OPTi93X_INIT 0x80 - -#define OPTi93X_MIXOUT_MIC_GAIN 0x20 -#define OPTi93X_MIXOUT_LINE 0x00 -#define OPTi93X_MIXOUT_CD 0x40 -#define OPTi93X_MIXOUT_MIC 0x80 -#define OPTi93X_MIXOUT_MIXER 0xc0 - -#define OPTi93X_STEREO 0x10 -#define OPTi93X_LINEAR_8 0x00 -#define OPTi93X_ULAW_8 0x20 -#define OPTi93X_LINEAR_16_LIT 0x40 -#define OPTi93X_ALAW_8 0x60 -#define OPTi93X_ADPCM_16 0xa0 -#define OPTi93X_LINEAR_16_BIG 0xc0 - -#define OPTi93X_CAPTURE_PIO 0x80 -#define OPTi93X_PLAYBACK_PIO 0x40 -#define OPTi93X_AUTOCALIB 0x08 -#define OPTi93X_SINGLE_DMA 0x04 -#define OPTi93X_CAPTURE_ENABLE 0x02 -#define OPTi93X_PLAYBACK_ENABLE 0x01 - -#define OPTi93X_IRQ_ENABLE 0x02 - -#define OPTi93X_DMA_REQUEST 0x10 -#define OPTi93X_CALIB_IN_PROGRESS 0x20 - #define OPTi93X_IRQ_PLAYBACK 0x04 #define OPTi93X_IRQ_CAPTURE 0x08 - -struct snd_opti93x { - unsigned long port; - struct resource *res_port; - int irq; - int dma1; - int dma2; - - struct snd_opti9xx *chip; - unsigned short hardware; - unsigned char image[32]; - - unsigned char mce_bit; - unsigned short mode; - int mute; - - spinlock_t lock; - - struct snd_card *card; - struct snd_pcm *pcm; - struct snd_pcm_substream *playback_substream; - struct snd_pcm_substream *capture_substream; - unsigned int p_dma_size; - unsigned int c_dma_size; -}; - -#define OPTi93X_MODE_NONE 0x00 -#define OPTi93X_MODE_PLAY 0x01 -#define OPTi93X_MODE_CAPTURE 0x02 -#define OPTi93X_MODE_OPEN (OPTi93X_MODE_PLAY | OPTi93X_MODE_CAPTURE) - #endif /* OPTi93X */ struct snd_opti9xx { @@ -235,6 +136,7 @@ struct snd_opti9xx { unsigned long mc_base_size; #ifdef OPTi93X unsigned long mc_indir_index; + struct snd_cs4231 *codec; #endif /* OPTi93X */ unsigned long pwd_reg; @@ -645,979 +557,23 @@ __skip_mpu: #ifdef OPTi93X -static unsigned char snd_opti93x_default_image[32] = -{ - 0x00, /* 00/00 - l_mixout_outctrl */ - 0x00, /* 01/01 - r_mixout_outctrl */ - 0x88, /* 02/02 - l_cd_inctrl */ - 0x88, /* 03/03 - r_cd_inctrl */ - 0x88, /* 04/04 - l_a1/fm_inctrl */ - 0x88, /* 05/05 - r_a1/fm_inctrl */ - 0x80, /* 06/06 - l_dac_inctrl */ - 0x80, /* 07/07 - r_dac_inctrl */ - 0x00, /* 08/08 - ply_dataform_reg */ - 0x00, /* 09/09 - if_conf */ - 0x00, /* 0a/10 - pin_ctrl */ - 0x00, /* 0b/11 - err_init_reg */ - 0x0a, /* 0c/12 - id_reg */ - 0x00, /* 0d/13 - reserved */ - 0x00, /* 0e/14 - ply_upcount_reg */ - 0x00, /* 0f/15 - ply_lowcount_reg */ - 0x88, /* 10/16 - reserved/l_a1_inctrl */ - 0x88, /* 11/17 - reserved/r_a1_inctrl */ - 0x88, /* 12/18 - l_line_inctrl */ - 0x88, /* 13/19 - r_line_inctrl */ - 0x88, /* 14/20 - l_mic_inctrl */ - 0x88, /* 15/21 - r_mic_inctrl */ - 0x80, /* 16/22 - l_out_outctrl */ - 0x80, /* 17/23 - r_out_outctrl */ - 0x00, /* 18/24 - reserved */ - 0x00, /* 19/25 - reserved */ - 0x00, /* 1a/26 - reserved */ - 0x00, /* 1b/27 - reserved */ - 0x00, /* 1c/28 - cap_dataform_reg */ - 0x00, /* 1d/29 - reserved */ - 0x00, /* 1e/30 - cap_upcount_reg */ - 0x00 /* 1f/31 - cap_lowcount_reg */ -}; - - -static int snd_opti93x_busy_wait(struct snd_opti93x *chip) -{ - int timeout; - - for (timeout = 250; timeout-- > 0; udelay(10)) - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_INIT)) - return 0; - - snd_printk("chip still busy.\n"); - return -EBUSY; -} - -static unsigned char snd_opti93x_in(struct snd_opti93x *chip, unsigned char reg) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - return inb(OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_busy_wait(chip); - outb(chip->mce_bit | (reg & 0x1f), OPTi93X_PORT(chip, INDEX)); - outb(value, OPTi93X_PORT(chip, DATA)); -} - -static void snd_opti93x_out_image(struct snd_opti93x *chip, unsigned char reg, - unsigned char value) -{ - snd_opti93x_out(chip, reg, chip->image[reg] = value); -} - -static void snd_opti93x_out_mask(struct snd_opti93x *chip, unsigned char reg, - unsigned char mask, unsigned char value) -{ - snd_opti93x_out_image(chip, reg, - (chip->image[reg] & ~mask) | (value & mask)); -} - - -static void snd_opti93x_mce_up(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = OPTi93X_MCE; - if (!(inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE)) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -static void snd_opti93x_mce_down(struct snd_opti93x *chip) -{ - snd_opti93x_busy_wait(chip); - - chip->mce_bit = 0; - if (inb(OPTi93X_PORT(chip, INDEX)) & OPTi93X_MCE) - outb(chip->mce_bit, OPTi93X_PORT(chip, INDEX)); -} - -#define snd_opti93x_mute_reg(chip, reg, mute) \ - snd_opti93x_out(chip, reg, mute ? 0x80 : chip->image[reg]); - -static void snd_opti93x_mute(struct snd_opti93x *chip, int mute) -{ - mute = mute ? 1 : 0; - if (chip->mute == mute) - return; - - chip->mute = mute; - - snd_opti93x_mute_reg(chip, OPTi93X_CD_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_CD_RIGHT_INPUT, mute); - switch (chip->hardware) { - case OPTi9XX_HW_82C930: - snd_opti93x_mute_reg(chip, OPTi930_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi930_AUX_RIGHT_INPUT, mute); - break; - case OPTi9XX_HW_82C931: - case OPTi9XX_HW_82C933: - snd_opti93x_mute_reg(chip, OPTi931_FM_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_FM_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi931_AUX_RIGHT_INPUT, mute); - } - snd_opti93x_mute_reg(chip, OPTi93X_DAC_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_DAC_RIGHT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_LINE_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_LEFT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_MIC_RIGHT_INPUT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_LEFT, mute); - snd_opti93x_mute_reg(chip, OPTi93X_OUT_RIGHT, mute); -} - - -static unsigned int snd_opti93x_get_count(unsigned char format, - unsigned int size) -{ - switch (format & 0xe0) { - case OPTi93X_LINEAR_16_LIT: - case OPTi93X_LINEAR_16_BIG: - size >>= 1; - break; - case OPTi93X_ADPCM_16: - return size >> 2; - } - return (format & OPTi93X_STEREO) ? (size >> 1) : size; -} - -static unsigned int rates[] = { 5512, 6615, 8000, 9600, 11025, 16000, - 18900, 22050, 27428, 32000, 33075, 37800, - 44100, 48000 }; -#define RATES ARRAY_SIZE(rates) - -static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = RATES, - .list = rates, - .mask = 0, -}; - -static unsigned char bits[] = { 0x01, 0x0f, 0x00, 0x0e, 0x03, 0x02, - 0x05, 0x07, 0x04, 0x06, 0x0d, 0x09, - 0x0b, 0x0c}; - -static unsigned char snd_opti93x_get_freq(unsigned int rate) -{ - unsigned int i; - - for (i = 0; i < RATES; i++) { - if (rate == rates[i]) - return bits[i]; - } - snd_BUG(); - return bits[RATES-1]; -} - -static unsigned char snd_opti93x_get_format(struct snd_opti93x *chip, - unsigned int format, int channels) -{ - unsigned char retval = OPTi93X_LINEAR_8; - - switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: - retval = OPTi93X_ULAW_8; - break; - case SNDRV_PCM_FORMAT_A_LAW: - retval = OPTi93X_ALAW_8; - break; - case SNDRV_PCM_FORMAT_S16_LE: - retval = OPTi93X_LINEAR_16_LIT; - break; - case SNDRV_PCM_FORMAT_S16_BE: - retval = OPTi93X_LINEAR_16_BIG; - break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: - retval = OPTi93X_ADPCM_16; - } - return (channels > 1) ? (retval | OPTi93X_STEREO) : retval; -} - - -static void snd_opti93x_playback_format(struct snd_opti93x *chip, unsigned char fmt) -{ - unsigned char mask; - - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - mask = (chip->mode & OPTi93X_MODE_CAPTURE) ? 0xf0 : 0xff; - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, mask, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - -static void snd_opti93x_capture_format(struct snd_opti93x *chip, unsigned char fmt) -{ - snd_opti93x_mute(chip, 1); - - snd_opti93x_mce_up(chip); - if (!(chip->mode & OPTi93X_MODE_PLAY)) - snd_opti93x_out_mask(chip, OPTi93X_PLAY_FORMAT, 0x0f, fmt); - else - fmt = chip->image[OPTi93X_PLAY_FORMAT] & 0xf0; - snd_opti93x_out_image(chip, OPTi93X_CAPT_FORMAT, fmt); - snd_opti93x_mce_down(chip); - - snd_opti93x_mute(chip, 0); -} - - -static int snd_opti93x_open(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (chip->mode & mode) { - spin_unlock_irqrestore(&chip->lock, flags); - return -EAGAIN; - } - - if (!(chip->mode & OPTi93X_MODE_OPEN)) { - outb(0x00, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, - OPTi93X_IRQ_ENABLE, OPTi93X_IRQ_ENABLE); - chip->mode = mode; - } - else - chip->mode |= mode; - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static void snd_opti93x_close(struct snd_opti93x *chip, unsigned int mode) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - chip->mode &= ~mode; - if (chip->mode & OPTi93X_MODE_OPEN) { - spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - snd_opti93x_mute(chip, 1); - - outb(0, OPTi93X_PORT(chip, STATUS)); - snd_opti93x_out_mask(chip, OPTi93X_PIN_CTRL, OPTi93X_IRQ_ENABLE, - ~OPTi93X_IRQ_ENABLE); - - snd_opti93x_mce_up(chip); - snd_opti93x_out_image(chip, OPTi93X_IFACE_CONF, 0x00); - snd_opti93x_mce_down(chip); - chip->mode = 0; - - snd_opti93x_mute(chip, 0); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_trigger(struct snd_pcm_substream *substream, - unsigned char what, int cmd) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_STOP: - { - unsigned int what = 0; - struct snd_pcm_substream *s; - snd_pcm_group_for_each_entry(s, substream) { - if (s == chip->playback_substream) { - what |= OPTi93X_PLAYBACK_ENABLE; - snd_pcm_trigger_done(s, substream); - } else if (s == chip->capture_substream) { - what |= OPTi93X_CAPTURE_ENABLE; - snd_pcm_trigger_done(s, substream); - } - } - spin_lock(&chip->lock); - if (cmd == SNDRV_PCM_TRIGGER_START) { - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, what); - if (what & OPTi93X_CAPTURE_ENABLE) - udelay(50); - } else - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, what, 0x00); - spin_unlock(&chip->lock); - break; - } - default: - return -EINVAL; - } - return 0; -} - -static int snd_opti93x_playback_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_PLAYBACK_ENABLE, cmd); -} - -static int snd_opti93x_capture_trigger(struct snd_pcm_substream *substream, int cmd) -{ - return snd_opti93x_trigger(substream, - OPTi93X_CAPTURE_ENABLE, cmd); -} - -static int snd_opti93x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *hw_params) -{ - return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); -} - - -static int snd_opti93x_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - return 0; -} - - -static int snd_opti93x_playback_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->p_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO, - ~(OPTi93X_PLAYBACK_ENABLE | OPTi93X_PLAYBACK_PIO)); - - snd_dma_program(chip->dma1, runtime->dma_addr, size, - DMA_MODE_WRITE | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_playback_format(chip, format); - format = chip->image[OPTi93X_PLAY_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_PLAY_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_PLAY_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_capture_prepare(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned long flags; - unsigned char format; - unsigned int count = snd_pcm_lib_period_bytes(substream); - unsigned int size = snd_pcm_lib_buffer_bytes(substream); - - spin_lock_irqsave(&chip->lock, flags); - - chip->c_dma_size = size; - snd_opti93x_out_mask(chip, OPTi93X_IFACE_CONF, - OPTi93X_CAPTURE_ENABLE | OPTi93X_CAPTURE_PIO, 0); - - snd_dma_program(chip->dma2, runtime->dma_addr, size, - DMA_MODE_READ | DMA_AUTOINIT); - - format = snd_opti93x_get_freq(runtime->rate); - format |= snd_opti93x_get_format(chip, runtime->format, - runtime->channels); - snd_opti93x_capture_format(chip, format); - format = chip->image[OPTi93X_CAPT_FORMAT]; - - count = snd_opti93x_get_count(format, count) - 1; - snd_opti93x_out_image(chip, OPTi93X_CAPT_LWR_CNT, count); - snd_opti93x_out_image(chip, OPTi93X_CAPT_UPR_CNT, count >> 8); - - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static snd_pcm_uframes_t snd_opti93x_playback_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_PLAYBACK_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma1, chip->p_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - -static snd_pcm_uframes_t snd_opti93x_capture_pointer(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - size_t ptr; - - if (!(chip->image[OPTi93X_IFACE_CONF] & OPTi93X_CAPTURE_ENABLE)) - return 0; - - ptr = snd_dma_pointer(chip->dma2, chip->c_dma_size); - return bytes_to_frames(substream->runtime, ptr); -} - - -static void snd_opti93x_overrange(struct snd_opti93x *chip) -{ - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - - if (snd_opti93x_in(chip, OPTi93X_ERR_INIT) & (0x08 | 0x02)) - chip->capture_substream->runtime->overrange++; - - spin_unlock_irqrestore(&chip->lock, flags); -} - static irqreturn_t snd_opti93x_interrupt(int irq, void *dev_id) { - struct snd_opti93x *codec = dev_id; + struct snd_cs4231 *codec = dev_id; + struct snd_opti9xx *chip = codec->card->private_data; unsigned char status; - status = snd_opti9xx_read(codec->chip, OPTi9XX_MC_REG(11)); + status = snd_opti9xx_read(chip, OPTi9XX_MC_REG(11)); if ((status & OPTi93X_IRQ_PLAYBACK) && codec->playback_substream) snd_pcm_period_elapsed(codec->playback_substream); if ((status & OPTi93X_IRQ_CAPTURE) && codec->capture_substream) { - snd_opti93x_overrange(codec); + snd_cs4231_overrange(codec); snd_pcm_period_elapsed(codec->capture_substream); } outb(0x00, OPTi93X_PORT(codec, STATUS)); return IRQ_HANDLED; } - -static struct snd_pcm_hardware snd_opti93x_playback = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static struct snd_pcm_hardware snd_opti93x_capture = { - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_8000_48000, - .rate_min = 5512, - .rate_max = 48000, - .channels_min = 1, - .channels_max = 2, - .buffer_bytes_max = (128*1024), - .period_bytes_min = 64, - .period_bytes_max = (128*1024), - .periods_min = 1, - .periods_max = 1024, - .fifo_size = 0, -}; - -static int snd_opti93x_playback_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_PLAY)) < 0) - return error; - snd_pcm_set_sync(substream); - chip->playback_substream = substream; - runtime->hw = snd_opti93x_playback; - snd_pcm_limit_isa_dma_size(chip->dma1, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_capture_open(struct snd_pcm_substream *substream) -{ - int error; - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - - if ((error = snd_opti93x_open(chip, OPTi93X_MODE_CAPTURE)) < 0) - return error; - runtime->hw = snd_opti93x_capture; - snd_pcm_set_sync(substream); - chip->capture_substream = substream; - snd_pcm_limit_isa_dma_size(chip->dma2, &runtime->hw.buffer_bytes_max); - snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates); - return error; -} - -static int snd_opti93x_playback_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->playback_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_PLAY); - return 0; -} - -static int snd_opti93x_capture_close(struct snd_pcm_substream *substream) -{ - struct snd_opti93x *chip = snd_pcm_substream_chip(substream); - - chip->capture_substream = NULL; - snd_opti93x_close(chip, OPTi93X_MODE_CAPTURE); - return 0; -} - - -static void snd_opti93x_init(struct snd_opti93x *chip) -{ - unsigned long flags; - int i; - - spin_lock_irqsave(&chip->lock, flags); - snd_opti93x_mce_up(chip); - - for (i = 0; i < 32; i++) - snd_opti93x_out_image(chip, i, snd_opti93x_default_image[i]); - - snd_opti93x_mce_down(chip); - spin_unlock_irqrestore(&chip->lock, flags); -} - -static int snd_opti93x_probe(struct snd_opti93x *chip) -{ - unsigned long flags; - unsigned char val; - - spin_lock_irqsave(&chip->lock, flags); - val = snd_opti93x_in(chip, OPTi93X_ID) & 0x0f; - spin_unlock_irqrestore(&chip->lock, flags); - - return (val == 0x0a) ? 0 : -ENODEV; -} - -static int snd_opti93x_free(struct snd_opti93x *chip) -{ - release_and_free_resource(chip->res_port); - if (chip->dma1 >= 0) { - disable_dma(chip->dma1); - free_dma(chip->dma1); - } - if (chip->dma2 >= 0) { - disable_dma(chip->dma2); - free_dma(chip->dma2); - } - if (chip->irq >= 0) { - free_irq(chip->irq, chip); - } - kfree(chip); - return 0; -} - -static int snd_opti93x_dev_free(struct snd_device *device) -{ - struct snd_opti93x *chip = device->device_data; - return snd_opti93x_free(chip); -} - -static const char *snd_opti93x_chip_id(struct snd_opti93x *codec) -{ - switch (codec->hardware) { - case OPTi9XX_HW_82C930: return "82C930"; - case OPTi9XX_HW_82C931: return "82C931"; - case OPTi9XX_HW_82C933: return "82C933"; - default: return "???"; - } -} - -static int snd_opti93x_create(struct snd_card *card, struct snd_opti9xx *chip, - int dma1, int dma2, - struct snd_opti93x **rcodec) -{ - static struct snd_device_ops ops = { - .dev_free = snd_opti93x_dev_free, - }; - int error; - struct snd_opti93x *codec; - - *rcodec = NULL; - codec = kzalloc(sizeof(*codec), GFP_KERNEL); - if (codec == NULL) - return -ENOMEM; - codec->irq = -1; - codec->dma1 = -1; - codec->dma2 = -1; - - if ((codec->res_port = request_region(chip->wss_base + 4, 4, "OPTI93x CODEC")) == NULL) { - snd_printk(KERN_ERR "opti9xx: can't grab port 0x%lx\n", chip->wss_base + 4); - snd_opti93x_free(codec); - return -EBUSY; - } - if (request_dma(dma1, "OPTI93x - 1")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA1 %d\n", dma1); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma1 = chip->dma1; - if (request_dma(dma2, "OPTI93x - 2")) { - snd_printk(KERN_ERR "opti9xx: can't grab DMA2 %d\n", dma2); - snd_opti93x_free(codec); - return -EBUSY; - } - codec->dma2 = chip->dma2; - - if (request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec)) { - snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); - snd_opti93x_free(codec); - return -EBUSY; - } - - codec->card = card; - codec->port = chip->wss_base + 4; - codec->irq = chip->irq; - - spin_lock_init(&codec->lock); - codec->hardware = chip->hardware; - codec->chip = chip; - - if ((error = snd_opti93x_probe(codec))) { - snd_opti93x_free(codec); - return error; - } - - snd_opti93x_init(codec); - - /* Register device */ - if ((error = snd_device_new(card, SNDRV_DEV_LOWLEVEL, codec, &ops)) < 0) { - snd_opti93x_free(codec); - return error; - } - - *rcodec = codec; - return 0; -} - -static struct snd_pcm_ops snd_opti93x_playback_ops = { - .open = snd_opti93x_playback_open, - .close = snd_opti93x_playback_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_playback_prepare, - .trigger = snd_opti93x_playback_trigger, - .pointer = snd_opti93x_playback_pointer, -}; - -static struct snd_pcm_ops snd_opti93x_capture_ops = { - .open = snd_opti93x_capture_open, - .close = snd_opti93x_capture_close, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = snd_opti93x_hw_params, - .hw_free = snd_opti93x_hw_free, - .prepare = snd_opti93x_capture_prepare, - .trigger = snd_opti93x_capture_trigger, - .pointer = snd_opti93x_capture_pointer, -}; - -static int snd_opti93x_pcm(struct snd_opti93x *codec, int device, struct snd_pcm **rpcm) -{ - int error; - struct snd_pcm *pcm; - - if ((error = snd_pcm_new(codec->card, "OPTi 82C93X", device, 1, 1, &pcm)) < 0) - return error; - - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_opti93x_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_opti93x_capture_ops); - - pcm->private_data = codec; - pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; - - strcpy(pcm->name, snd_opti93x_chip_id(codec)); - - snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - snd_dma_isa_data(), - 64*1024, codec->dma1 > 3 || codec->dma2 > 3 ? 128*1024 : 64*1024); - - codec->pcm = pcm; - if (rpcm) - *rpcm = pcm; - return 0; -} - -/* - * MIXER part - */ - -static int snd_opti93x_info_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - static char *texts[4] = { - "Line1", "Aux", "Mic", "Mix" - }; - - uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; - uinfo->count = 2; - uinfo->value.enumerated.items = 4; - if (uinfo->value.enumerated.item > 3) - uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); - return 0; -} - -static int snd_opti93x_get_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.enumerated.item[0] = (chip->image[OPTi93X_MIXOUT_LEFT] & OPTi93X_MIXOUT_MIXER) >> 6; - ucontrol->value.enumerated.item[1] = (chip->image[OPTi93X_MIXOUT_RIGHT] & OPTi93X_MIXOUT_MIXER) >> 6; - spin_unlock_irqrestore(&chip->lock, flags); - return 0; -} - -static int snd_opti93x_put_mux(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - unsigned short left, right; - int change; - - if (ucontrol->value.enumerated.item[0] > 3 || - ucontrol->value.enumerated.item[1] > 3) - return -EINVAL; - left = ucontrol->value.enumerated.item[0] << 6; - right = ucontrol->value.enumerated.item[1] << 6; - spin_lock_irqsave(&chip->lock, flags); - left = (chip->image[OPTi93X_MIXOUT_LEFT] & ~OPTi93X_MIXOUT_MIXER) | left; - right = (chip->image[OPTi93X_MIXOUT_RIGHT] & ~OPTi93X_MIXOUT_MIXER) | right; - change = left != chip->image[OPTi93X_MIXOUT_LEFT] || - right != chip->image[OPTi93X_MIXOUT_RIGHT]; - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_LEFT, left); - snd_opti93x_out_image(chip, OPTi93X_MIXOUT_RIGHT, right); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#if 0 - -#define OPTi93X_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_single, \ - .get = snd_opti93x_get_single, .put = snd_opti93x_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -static int snd_opti93x_info_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 16) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 1; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - return 0; -} - -static int snd_opti93x_put_single(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int reg = kcontrol->private_value & 0xff; - int shift = (kcontrol->private_value >> 8) & 0xff; - int mask = (kcontrol->private_value >> 16) & 0xff; - int invert = (kcontrol->private_value >> 24) & 0xff; - int change; - unsigned short val; - - val = (ucontrol->value.integer.value[0] & mask); - if (invert) - val = mask - val; - val <<= shift; - spin_lock_irqsave(&chip->lock, flags); - val = (chip->image[reg] & ~(mask << shift)) | val; - change = val != chip->image[reg]; - snd_opti93x_out(chip, reg, val); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -#endif /* single */ - -#define OPTi93X_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_opti93x_info_double, \ - .get = snd_opti93x_get_double, .put = snd_opti93x_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } - -#define OPTi93X_DOUBLE_INVERT_INVERT(xctl) \ - do { xctl.private_value ^= 22; } while (0) -#define OPTi93X_DOUBLE_CHANGE_REGS(xctl, left_reg, right_reg) \ - do { xctl.private_value &= ~0x0000ffff; \ - xctl.private_value |= left_reg | (right_reg << 8); } while (0) - -static int snd_opti93x_info_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) -{ - int mask = (kcontrol->private_value >> 24) & 0xff; - - uinfo->type = mask == 1 ? SNDRV_CTL_ELEM_TYPE_BOOLEAN : SNDRV_CTL_ELEM_TYPE_INTEGER; - uinfo->count = 2; - uinfo->value.integer.min = 0; - uinfo->value.integer.max = mask; - return 0; -} - -static int snd_opti93x_get_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - - spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; - spin_unlock_irqrestore(&chip->lock, flags); - if (invert) { - ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - ucontrol->value.integer.value[1] = mask - ucontrol->value.integer.value[1]; - } - return 0; -} - -static int snd_opti93x_put_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -{ - struct snd_opti93x *chip = snd_kcontrol_chip(kcontrol); - unsigned long flags; - int left_reg = kcontrol->private_value & 0xff; - int right_reg = (kcontrol->private_value >> 8) & 0xff; - int shift_left = (kcontrol->private_value >> 16) & 0x07; - int shift_right = (kcontrol->private_value >> 19) & 0x07; - int mask = (kcontrol->private_value >> 24) & 0xff; - int invert = (kcontrol->private_value >> 22) & 1; - int change; - unsigned short val1, val2; - - val1 = ucontrol->value.integer.value[0] & mask; - val2 = ucontrol->value.integer.value[1] & mask; - if (invert) { - val1 = mask - val1; - val2 = mask - val2; - } - val1 <<= shift_left; - val2 <<= shift_right; - spin_lock_irqsave(&chip->lock, flags); - val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; - val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; - snd_opti93x_out_image(chip, left_reg, val1); - snd_opti93x_out_image(chip, right_reg, val2); - spin_unlock_irqrestore(&chip->lock, flags); - return change; -} - -static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { -OPTi93X_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), -OPTi93X_DOUBLE("PCM Playback Switch", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 7, 7, 1, 1), -OPTi93X_DOUBLE("PCM Playback Volume", 0, OPTi93X_DAC_LEFT, OPTi93X_DAC_RIGHT, 0, 0, 31, 1), -OPTi93X_DOUBLE("FM Playback Switch", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("FM Playback Volume", 0, OPTi931_FM_LEFT_INPUT, OPTi931_FM_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Line Playback Switch", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Line Playback Volume", 0, OPTi93X_LINE_LEFT_INPUT, OPTi93X_LINE_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Mic Boost", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 5, 5, 1, 1), -OPTi93X_DOUBLE("CD Playback Switch", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("CD Playback Volume", 0, OPTi93X_CD_LEFT_INPUT, OPTi93X_CD_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), -OPTi93X_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), -OPTi93X_DOUBLE("Capture Volume", 0, OPTi93X_MIXOUT_LEFT, OPTi93X_MIXOUT_RIGHT, 0, 0, 15, 0), -{ - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - .name = "Capture Source", - .info = snd_opti93x_info_mux, - .get = snd_opti93x_get_mux, - .put = snd_opti93x_put_mux, -} -}; - -static int snd_opti93x_mixer(struct snd_opti93x *chip) -{ - struct snd_card *card; - struct snd_kcontrol_new knew; - int err; - unsigned int idx; - - snd_assert(chip != NULL && chip->card != NULL, return -EINVAL); - - card = chip->card; - - strcpy(card->mixername, snd_opti93x_chip_id(chip)); - - for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { - knew = snd_opti93x_controls[idx]; - if (chip->hardware == OPTi9XX_HW_82C930) { - if (strstr(knew.name, "FM")) /* skip FM controls */ - continue; - else if (strcmp(knew.name, "Mic Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strstr(knew.name, "Aux")) - OPTi93X_DOUBLE_CHANGE_REGS(knew, OPTi930_AUX_LEFT_INPUT, OPTi930_AUX_RIGHT_INPUT); - else if (strcmp(knew.name, "PCM Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - else if (strcmp(knew.name, "Master Playback Volume")) - OPTi93X_DOUBLE_INVERT_INVERT(knew); - } - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_opti93x_controls[idx], chip))) < 0) - return err; - } - return 0; -} - #endif /* OPTi93X */ static int __devinit snd_card_opti9xx_detect(struct snd_card *card, @@ -1774,8 +731,16 @@ static void snd_card_opti9xx_free(struct { struct snd_opti9xx *chip = card->private_data; - if (chip) + if (chip) { +#ifdef OPTi93X + struct snd_cs4231 *codec = chip->codec; + if (codec->irq > 0) { + disable_irq(codec->irq); + free_irq(codec->irq, codec); + } +#endif release_and_free_resource(chip->res_mc_base); + } } static int __devinit snd_opti9xx_probe(struct snd_card *card) @@ -1783,11 +748,11 @@ static int __devinit snd_opti9xx_probe(s static long possible_ports[] = {0x530, 0xe80, 0xf40, 0x604, -1}; int error; struct snd_opti9xx *chip = card->private_data; -#if defined(OPTi93X) - struct snd_opti93x *codec; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) struct snd_cs4231 *codec; +#ifdef CS4231 struct snd_timer *timer; +#endif #else struct snd_ad1848 *codec; #endif @@ -1819,26 +784,31 @@ static int __devinit snd_opti9xx_probe(s if ((error = snd_opti9xx_configure(chip))) return error; -#if defined(OPTi93X) - if ((error = snd_opti93x_create(card, chip, chip->dma1, chip->dma2, &codec))) - return error; - if ((error = snd_opti93x_pcm(codec, 0, &pcm)) < 0) - return error; - if ((error = snd_opti93x_mixer(codec)) < 0) - return error; -#elif defined(CS4231) +#if defined(CS4231) || defined(OPTi93X) if ((error = snd_cs4231_create(card, chip->wss_base + 4, -1, chip->irq, chip->dma1, chip->dma2, - CS4231_HW_DETECT, - 0, +#ifdef CS4231 + CS4231_HW_DETECT, 0, +#else /* OPTi93x */ + CS4231_HW_OPTI93X, CS4231_HWSHARE_IRQ, +#endif &codec)) < 0) return error; + chip->codec = codec; if ((error = snd_cs4231_pcm(codec, 0, &pcm)) < 0) return error; if ((error = snd_cs4231_mixer(codec)) < 0) return error; +#ifdef CS4231 if ((error = snd_cs4231_timer(codec, 0, &timer)) < 0) return error; +#else /* OPTI93X */ + error = request_irq(chip->irq, snd_opti93x_interrupt, IRQF_DISABLED, DEV_NAME" - WSS", codec); + if (error < 0) { + snd_printk(KERN_ERR "opti9xx: can't grab IRQ %d\n", chip->irq); + return error; + } +#endif #else if ((error = snd_ad1848_create(card, chip->wss_base + 4, chip->irq, chip->dma1, ---------------------------------------------------------------------- Pokaz innym kto tu rzadzi! 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1 0

[alsa-devel] [PATCH] cs4231-lib: add support for Opti93x codec
by Krzysztof Helt 22 Sep '07

22 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch adds support for WSS compatible Opti93x codec to the cs4231-lib. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This allows to remove cs4231 duplicated code from the opti93x driver. It is almost half of the opti driver. The opti93x must use its own interrupt handler so it needs also the overrange function. That's why it is exported. The next patch will convert the opti93x driver to use the cs4231 library instead of its own code. The cs4231 library (and ad1848's) gets improved with Rene's and Trent's help so this should improve the opti driver as well. Regards, Krzysztof diff -urp linux-ref/include/sound/cs4231-regs.h linux-2.6.23/include/sound/cs4231-regs.h --- linux-ref/include/sound/cs4231-regs.h 2007-09-22 21:42:29.000000000 +0200 +++ linux-2.6.23/include/sound/cs4231-regs.h 2007-09-22 21:43:23.876402416 +0200 @@ -177,4 +177,12 @@ #define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ #define CS4236_VERSION 0x9c /* chip version and ID */ +/* definitions for extended registers - OPTI93X */ +#define OPTi931_AUX_LEFT_INPUT 0x10 +#define OPTi931_AUX_RIGHT_INPUT 0x11 +#define OPTi93X_MIC_LEFT_INPUT 0x14 +#define OPTi93X_MIC_RIGHT_INPUT 0x15 +#define OPTi93X_OUT_LEFT 0x16 +#define OPTi93X_OUT_RIGHT 0x17 + #endif /* __SOUND_CS4231_REGS_H */ diff -urp linux-ref/include/sound/cs4231.h linux-2.6.23/include/sound/cs4231.h --- linux-ref/include/sound/cs4231.h 2007-09-22 21:38:19.000000000 +0200 +++ linux-2.6.23/include/sound/cs4231.h 2007-09-22 21:43:17.784055233 +0200 @@ -58,6 +58,7 @@ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ #define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1002 /* Opti 930/931/933 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) @@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct s void snd_cs4231_mce_up(struct snd_cs4231 *chip); void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_cs4231_overrange(struct snd_cs4231 *chip); + irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); diff -urp linux-ref/sound/isa/cs423x/cs4231_lib.c linux-2.6.23/sound/isa/cs423x/cs4231_lib.c --- linux-ref/sound/isa/cs423x/cs4231_lib.c 2007-09-22 14:52:40.000000000 +0200 +++ linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-09-22 21:40:46.063409166 +0200 @@ -120,6 +120,42 @@ static unsigned char snd_cs4231_original 0x00, /* 1f/31 - cbrl */ }; +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + /* * Basic I/O functions */ @@ -897,7 +933,7 @@ static int snd_cs4231_capture_prepare(st return 0; } -static void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_cs4231_overrange(struct snd_cs4231 *chip) { unsigned long flags; unsigned char res; @@ -1056,8 +1092,11 @@ static int snd_cs4231_probe(struct snd_c chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } ptr = (unsigned char *) &chip->image; snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1378,6 +1417,7 @@ const char *snd_cs4231_chip_id(struct sn case CS4231_HW_INTERWAVE: return "AMD InterWave"; case CS4231_HW_OPL3SA2: return chip->card->shortname; case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } @@ -1403,7 +1443,12 @@ static int snd_cs4231_new(struct snd_car chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; chip->set_capture_format = snd_cs4231_capture_format; - memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); *rchip = chip; return 0; @@ -1792,6 +1837,32 @@ CS4231_SINGLE("Loopback Capture Switch", CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; +static struct snd_kcontrol_new snd_opti93x_controls[] __devinitdata = { +CS4231_DOUBLE("Master Playback Switch", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + int snd_cs4231_mixer(struct snd_cs4231 *chip) { struct snd_card *card; @@ -1804,10 +1875,22 @@ int snd_cs4231_mixer(struct snd_cs4231 * strcpy(card->mixername, chip->pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0) - return err; - } + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } return 0; } @@ -1817,6 +1900,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out); EXPORT_SYMBOL(snd_cs4236_ext_in); EXPORT_SYMBOL(snd_cs4231_mce_up); EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); EXPORT_SYMBOL(snd_cs4231_interrupt); EXPORT_SYMBOL(snd_cs4231_chip_id); EXPORT_SYMBOL(snd_cs4231_create); ---------------------------------------------------------------------- Fajne i smieszne. 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1 0

[alsa-devel] Phoenix Audio Duet
by Shane Hathaway 22 Sep '07

22 Sep '07

Hello, I just bought a set of Phoenix Audio Duets for VOIP conferencing. They are USB speakerphones with good echo cancellation. They work fine with the snd_usb_audio module, with one exception: I can't control the microphone and speaker levels. ALSA exposes some controls, but they have no effect, so I have to shout to be heard. Rather than get a hoarse voice, I want to help enhance the snd_usb_audio module to correctly support the mixer controls within the Duet. I know the device actually has working mixer controls because Phoenix Audio provides a Windows-only utility for controlling the levels, and when I use that utility turn up the mic, I can be heard quite well. Unfortunately, the levels are reset when I move the device back to a Linux box. I used SnoopyPro to capture USB packets as I adjusted the levels in Windows and I think I know what to transmit to the device in order to read and change the levels. Now I just need to figure out how to translate that knowledge into kernel code. I am a capable C coder (though rusty), I don't know much about USB, and I've dabbled in kernel code before. lsusb lists the Duet (Executive) as: Bus 001 Device 011: ID 0556:0004 Asahi Kasei Microsystems Co., Ltd Both kmix and Ekiga see it as an AK4571. There is a data sheet for that chip here: http://www.datasheetcatalog.com/asahikaseimicrosystems/3/ I would guess that the AK4571 has mixer controls built in, but that Phoenix Audio (phnxaudio.com) chose not to use those controls and instead enhanced the USB interface with nonstandard mixer controls. So, where should I begin? I don't see evidence that anyone has worked on this yet. Shane

1 1

[alsa-devel] ESI MAYA44 driver status
by Piotr Makowski 22 Sep '07

22 Sep '07

Hi, I would like to know is there any progress with writing those drivers? Anybody contacted with ESI? Regards, Piotr Makowski

6 11

Re: [alsa-devel] ESI MAYA44 driver status
by Piotr Makowski 22 Sep '07

22 Sep '07

If anybody would like to help with ESI MAYA44 PCI please bid on getacoder - we offer up to 300$ for the driver http://www.getacoder.com/projects/esi_maya_pci_alsa_driver_59542.html Please don't start working on it without bidding and before your bid is accepted. Thank you! Regards, Piotr

1 0

[alsa-devel] [PATCH -mm] sb16: Shut up uninitialized var build warning
by Satyam Sharma 22 Sep '07

22 Sep '07

sound/isa/sb16/sb16.c: In function ‘snd_sb16_isa_probe’: sound/isa/sb16/sb16.c:559: warning: ‘err’ may be used uninitialized in this function is a bogus warning, so let's shut it up. Signed-off-by: Satyam Sharma <satyam(a)infradead.org> --- sound/isa/sb/sb16.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) --- linux-2.6.23-rc4-mm1/sound/isa/sb/sb16.c‾fix 2007-09-02 21:41:51.000000000 +0530 +++ linux-2.6.23-rc4-mm1/sound/isa/sb/sb16.c 2007-09-02 21:42:56.000000000 +0530 @@ -556,7 +556,6 @@ static int __devinit snd_sb16_isa_match( static int __devinit snd_sb16_isa_probe(struct device *pdev, unsigned int dev) { - int err; static int possible_irqs[] = {5, 9, 10, 7, -1}; static int possible_dmas8[] = {1, 3, 0, -1}; static int possible_dmas16[] = {5, 6, 7, -1}; @@ -585,6 +584,8 @@ static int __devinit snd_sb16_isa_probe( else { static int possible_ports[] = {0x220, 0x240, 0x260, 0x280}; int i; + int uninitialized_var(err); + for (i = 0; i < ARRAY_SIZE(possible_ports); i++) { port[dev] = possible_ports[i]; err = snd_sb16_isa_probe1(dev, pdev);

3 4

[alsa-devel] Tascam US122 on ARM [PATCH] and error :-(
by ST 21 Sep '07

21 Sep '07

Hi I managed to get the snd_usb_usx2y driver running on an Linksys NSLU2. The whole stuff is running under debian etch. Kernel Version is 2.6.18 with Debian patches. Below is a little patch to be able to enable the driver. Unfortunatly the driver gives the error: Sequence Error!(hcd_frame=7 ep=8in;wait=65539,frame=3). Most propably some urb of usb-frame 65539 is still missing. Cause could be too long delays in usb-hcd interrupt handling. Which is the output of the function usX2Y_error_sequence. This function is only called from usbusx2yaudio.c:328. Since all numbers lie just above 65535 (They are all between 65537 and 65540) is suspect that there is some wrap around error, especially because the values are anded with a 16 bit mask (0xffff). But unfortunatly i don't know to much about usb right now. Any hints to resolve this problem? The cpu runs with 266Mhz, so it could be that it would be really to slow, but i really suspect an wrap-around error. Any hints or ideas to solve this issue? Thanks ST --- a/sound/usb/Kconfig +++ b/sound/usb/Kconfig @@ -18,7 +18,7 @@ config SND_USB_AUDIO config SND_USB_USX2Y tristate "Tascam US-122, US-224 and US-428 USB driver" - depends on SND && USB && (X86 || PPC || ALPHA) + depends on SND && USB && (X86 || PPC || ALPHA ||ARM) select SND_HWDEP select SND_RAWMIDI select SND_PCM

1 0

[alsa-devel] When to start the capture device?
by Sheng (Sean) Liu 21 Sep '07

21 Sep '07

Hello everyone, I am using ALSA to capture sound from a micrphone. I need to know when the capture device starts. In Paul Davis' Tutorial, he mentioned two approaches. But, I have questions on each of them. When to start the device When you open the audio interface, ALSA ensures that it is not active - no data is being moved to or from its external connectors. Presumably, at some point you want this data transfer to begin. There are several options for how to make this happen. 1. The control point here the start threshold, which defines the number of frames of space/data necessary to start the device automatically. If set to some value other than zero for playback, it is necessary to prefill the playback buffer before the device will start. If set to zero, the first data written to the device (or first attempt to read from a capture stream) will start the device. My question: what does he means on " If set to zero"? Does anyone know which parameter of which API should be set to zero? 2. You can also start the device explicitly using snd_pcm_start, but this requires buffer prefilling in the case of the playback stream. If you attempt to start the stream without doing this, you will get -EPIPE as a return code, indicating that there is no data waiting to deliver to the playback hardware buffer. My question: Is there anything to be done before calling snd_pcm_start in the case of capturing stream? If there is, what are they? Thanks in advance, Sheng

4 9

[alsa-devel] [PATCH] ad1848 ac loop
by Trent Piepho 21 Sep '07

21 Sep '07

New thread for the the final version of the patches for my fixes to the ad1848 auto-calibration loop. I made a couple suggested changes. These should be ok to apply, and should be independent of the other as yet un-applied patches that have been posted by Rene and Krzysztof. I'm not an expert on this chip by any means, but from looking at the code I think there are few things that could be fixed. The reg_lock spinlock isn't acquired from the irq handler, as the handler doesn't use any of the indirect registers. I think this means that is isn't necessary to use the irq disabling versions of the spin_lock functions with reg_lock. It does look like there is a different SMP race condition wrt the irq handler.

5 9

[alsa-devel] OSS app works but aplay does not with SNDRV_PCM_FMTBIT_U32_LE
by Steve Strobel 21 Sep '07

21 Sep '07

I am developing an ALSA driver for the AD1938 codec on uClinux for Blackfin (I don't believe my problem is specific to that platform). When I set up the snd_pcm_hardware_t structure with .formats = SNDRV_PCM_FMTBIT_S32_LE, I can run both "aplay < /dev/urandom" and an OSS test application ("tone") that sends a sine wave, but the sine wave is "inside out" (see <http://www.link-comm.com/temp/inside_out.gif> for a screen shot) because the codec needs unsigned data. If I set up the snd_pcm_hardware_t structure with .formats = SNDRV_PCM_FMTBIT_U32_LE, the OSS test application sends a nice sine wave, but aplay fails with the following messages: root:~> aplay < /dev/urandom ALSA sound/blackfin/ad1836.c:1050: snd_ad1836_playback_open Playing raw data 'stdin' : Unsigned 8 bit, Rate 8000 Hz, Mono snd_pcm_hw_refine() ACCESS = 00000000ffffffffffffffff -> 0000000000000008 FORMAT = 0000000000000400 -> 0000000000000000 snd_pcm_hw_refine() ACCESS = 333f8fc000a32c60c0333f8fc -> 333f8fc000a00000008 ALSA lib ALSA sound/blackfin/ad1836.c:1172: snd_ad1836_hw_free ../../../src/pcm/pcm_params.c:2152:(snd_pcm_hw_refine_slave) Slave PCM not usable aplay: set_params:882: Broken configuration for this PCM: no configurations available I am working my way through the ALSA source, but I have not yet figured out why aplay won't work with the unsigned codec configuration. Any suggestions? Steve P.S. A more complete log of the console output with both settings for those commands and most of the info from aadebug can be found at the following URLs: <http://www.link-comm.com/temp/SNDRV_PCM_FMTBIT_S32_LE.txt> <http://www.link-comm.com/temp/SNDRV_PCM_FMTBIT_U32_LE.txt> --- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

3 6

[alsa-devel] Pending patches for 1.0.15
by Takashi Iwai 21 Sep '07

21 Sep '07

Hi, we're planning to release 1.0.15rc3 tomorrow. This should be the last candidate before 1.0.15 (hopefully). If you have pending "hotfix" patches, please submit as soon as possible. Thanks. Takashi

3 4

[alsa-devel] [PATCH] alsa: Add the MCP79 support to hda_intel driver
by Peer Chen 21 Sep '07

21 Sep '07

Resend the patch for description modification and send to correct alsa maillist. Add the MCP79 support to hda driver. The patch base on kernel 2.6.23-rc7 Signed-off-by: Peer Chen <peerchen(a)gmail.com> --- --- linux-2.6.23-rc7/sound/pci/hda/hda_intel.c.orig 2007-09-20 15:40:51.000000000 -0400 +++ linux-2.6.23-rc7/sound/pci/hda/hda_intel.c 2007-09-21 13:50:33.000000000 -0400 @@ -1791,6 +1791,10 @@ static struct pci_device_id azx_ids[] = { 0x10de, 0x0775, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0x10de, 0x0776, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ { 0x10de, 0x0777, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP77 */ + { 0x10de, 0x0ac0, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0ac1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0ac2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ + { 0x10de, 0x0ac3, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_NVIDIA }, /* NVIDIA MCP79 */ { 0, } }; MODULE_DEVICE_TABLE(pci, azx_ids); - -------------- Peer Chen 2007-09-21

1 0

[alsa-devel] [PATCH] sun-cs4231: mce_down cleanup after ad1848
by Krzysztof Helt 20 Sep '07

20 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This simplifies and fixes waiting loops of the mce_down() function after Trent Piepho's patch for AD1848. It also makes busy_wait() function call not atomic. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-ref/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-ref/sound/sparc/cs4231.c 2007-09-19 23:31:30.603838619 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-19 23:23:24.552140109 +0200 @@ -392,62 +392,45 @@ static void snd_cs4231_mce_up(struct snd static void snd_cs4231_mce_down(struct snd_cs4231 *chip) { - unsigned long flags; - int timeout; + unsigned long flags, timeout; + int reg; - spin_lock_irqsave(&chip->lock, flags); snd_cs4231_busy_wait(chip); + spin_lock_irqsave(&chip->lock, flags); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) snd_printdd("mce_down [%p] - auto calibration time out (0)\n", CS4231U(chip, REGSEL)); #endif chip->mce_bit &= ~CS4231_MCE; - timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); - __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), + reg = __cs4231_readb(chip, CS4231U(chip, REGSEL)); + __cs4231_writeb(chip, chip->mce_bit | (reg & 0x1f), CS4231U(chip, REGSEL)); - if (timeout == 0x80) - snd_printdd("mce_down [%p]: serious init problem - codec still busy\n", chip->port); - if ((timeout & CS4231_MCE) == 0) { + if (reg == 0x80) + snd_printdd("mce_down [%p]: serious init problem " + "- codec still busy\n", chip->port); + if ((reg & CS4231_MCE) == 0) { spin_unlock_irqrestore(&chip->lock, flags); return; } - snd_cs4231_busy_wait(chip); - /* calibration process */ + /* + * Wait for auto-calibration (AC) process to finish, i.e. ACI to go low. + */ - snd_cs4231_ready(chip); - snd_cs4231_ready(chip); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); + timeout = jiffies + msecs_to_jiffies(250); + do { spin_unlock_irqrestore(&chip->lock, flags); - return; - } - - /* in 10ms increments, check condition, up to 250ms */ - timeout = 25; - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { - spin_unlock_irqrestore(&chip->lock, flags); - if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (2)\n"); - return; - } - msleep(10); - spin_lock_irqsave(&chip->lock, flags); - } - - /* in 10ms increments, check condition, up to 100ms */ - timeout = 10; - while (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) { - spin_unlock_irqrestore(&chip->lock, flags); - if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (3)\n"); - return; - } - msleep(10); + msleep(1); spin_lock_irqsave(&chip->lock, flags); - } + reg = snd_cs4231_in(chip, CS4231_TEST_INIT); + reg &= CS4231_CALIB_IN_PROGRESS; + } while (reg && time_before(jiffies, timeout)); spin_unlock_irqrestore(&chip->lock, flags); + + if (reg) + snd_printk(KERN_ERR + "mce_down - auto calibration time out (2)\n"); } static void snd_cs4231_advance_dma(struct cs4231_dma_control *dma_cont, ---------------------------------------------------------------------- Bedac w toalecie korzystala z ... >>> http://link.interia.pl/f1bd8

2 3

[alsa-devel] Realtek ALC660-VD jack sense
by Vladimir Pouzanov 20 Sep '07

20 Sep '07

Hello all, I wonder if it could be possible to somehow fix jack sense detection on Realtek ALC660-VD and speaker output. When I plug in headphones speakers are still working and there's no way to separately mute them.

2 1

[alsa-devel] [PARTIAL PATCH] snd-hda-intel on Medion WIM2160
by Bernhard Rosenkraenzer 20 Sep '07

20 Sep '07

Hi, On Medion WIM2160 notebooks (soundchip PCI ID 8086:27d8/17c0:4071), snd-hda-intel loads and everything acts like it's working, but there's no sound. The master volume is muted and there's no way to unmute it. The patch below fixes it, but unfortunately, Medion MD2 uses the same PCI (and subvendor/subdevice) ID -- so the patch quite likely breaks Medion MD2 support. WIM2160's soundchip has revision ID 02 - if MD2 uses something else, maybe that can be used to identify them and do the right thing. (I don't have an MD2...). ttyl bero --- linux-2.6.22/sound/pci/hda/patch_realtek.c.wim2160snd~ 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22/sound/pci/hda/patch_realtek.c 2007-09-09 20:28:55.000000000 +0200 @@ -159,6 +159,7 @@ ALC883_ACER, ALC883_MEDION, ALC883_MEDION_MD2, + ALC883_MEDION_WIM2160, ALC883_LAPTOP_EAPD, ALC883_LENOVO_101E_2ch, ALC883_LENOVO_NB0763, @@ -5995,6 +5996,32 @@ { } /* end */ }; +static struct snd_kcontrol_new alc883_medion_wim2160_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Master Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x14, 0x0, HDA_OUTPUT), + 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), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("iMic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("iMic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_chmode_mixer[] = { { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -6364,6 +6391,7 @@ [ALC883_ACER] = "acer", [ALC883_MEDION] = "medion", [ALC883_MEDION_MD2] = "medion-md2", + [ALC883_MEDION_WIM2160] = "medion-wim2160", [ALC883_LAPTOP_EAPD] = "laptop-eapd", [ALC883_LENOVO_101E_2ch] = "lenovo-101e", [ALC883_LENOVO_NB0763] = "lenovo-nb0763", @@ -6401,6 +6429,7 @@ SND_PCI_QUIRK(0x17aa, 0x3bfd, "Lenovo NB0763", ALC883_LENOVO_NB0763), SND_PCI_QUIRK(0x17aa, 0x2085, "Lenovo NB0763", ALC883_LENOVO_NB0763), - SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + //SND_PCI_QUIRK(0x17c0, 0x4071, "MEDION MD2", ALC883_MEDION_MD2), + SND_PCI_QUIRK(0x17c0, 0x4071, "Medion WIM2160", ALC883_MEDION_WIM2160), {} }; @@ -6530,6 +6559,19 @@ .unsol_event = alc883_medion_md2_unsol_event, .init_hook = alc883_medion_md2_automute, }, + [ALC883_MEDION_WIM2160] = { + .mixers = { alc883_medion_wim2160_mixer }, + .init_verbs = { alc883_init_verbs, alc883_lenovo_nb0763_verbs}, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_lenovo_nb0763_capture_source, + .unsol_event = alc883_medion_md2_unsol_event, + .init_hook = alc883_medion_md2_automute, + }, [ALC883_LAPTOP_EAPD] = { .mixers = { alc883_base_mixer, alc883_chmode_mixer },

2 3

[alsa-devel] alsa driver #2234 SoundScape Elite not support
by Krzysztof Helt 20 Sep '07

20 Sep '07

Hi, I looked into the ALSA bug tracking system and found the bug that SoundScape Elite is not supported by the sscape driver. I don't know how to contact the reporter of the bug (tjallen) so I am posting here. I would like to ask the reporter to test the current driver which support VIVO cards. One of changes is delay to wait for codec to be ready (delay suggested by Rene Herman but the OSS code had long waiting loop in the probe function as well). The current driver could work on the SoundScape Elite. The driver will be released as part of alsa-driver-1.0.15-rc3 in few days. Regards, Krzysztof ---------------------------------------------------------------------- Gra - zostan agentem ABW! Zobacz >>> http://link.interia.pl/f1ba5

2 1

[alsa-devel] resetting an audio device?
by william estrada 20 Sep '07

20 Sep '07

Hi guys, I got my alsa player code to work. Now I need to figure out how to reset the audio device? I don't want to close the device, just recycle it. Any pointers? Also, where are the documents for all of those helper subroutes? Like snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_any, etc. Thanks for your time. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum

2 1

Re: [alsa-devel] resetting an audio device?
by william estrada 20 Sep '07

20 Sep '07

I don't need item one, it works now. But I would like to see the better API docs. -- William Estrada MrUmunhum(a)popdial.com Mt-Umunhum-Wireless.net ( http://Mt-Umunhum-Wireless.net ) Ymessenger: MrUmunhum > Hi guys, > > I got my alsa player code to work. Now I need to figure out how to > reset the audio device? I don't want to close the device, just recycle > it. Any pointers? > > Also, where are the documents for all of those helper subroutes? Like > snd_pcm_hw_params_get_period_time, snd_pcm_hw_params_any, etc. > > Thanks for your time.

1 0

[alsa-devel] [PATCH] opti93x: opti931 + QS1000 wavetable
by Krzysztof Helt 19 Sep '07

19 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch adds support for a wavetable chip on the BTC 1817DW board. The QS1000 is connected through the digital input to the Opti931 chip. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This is not a hot-fix and can be postponed after 1.0.15 (before Takashi asks). A cornerstone of this patch is to set the auto-source selection for the FM DAC. This way it does not affects boards using the FM OPL synthesizers which do not use the digital input. If any other wavetable is connected to the opti chip through the digital input, the format setting (MC_REG(21)) should be revised if it is correct for the new board. However, in the whole opti92x-ad1848 this register was never touched (so probably nobody ever used the digital input ). Regards, Krzysztof diff -urp linux-ref/sound/isa/opti9xx/opti92x-ad1848.c linux-2.6.23/sound/isa/opti9xx/opti92x-ad1848.c --- linux-ref/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-12 13:33:51.000000000 +0200 +++ linux-2.6.23/sound/isa/opti9xx/opti92x-ad1848.c 2007-09-19 22:13:02.000000000 +0200 @@ -501,6 +501,16 @@ static int __devinit snd_opti9xx_configu (chip->hardware == OPTi9XX_HW_82C930 ? 0x00 : 0x04), 0x34); snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(5), 0x20, 0xbf); + /* + * The BTC 1817DW has QS1000 wavetable which is connected + * to the serial digital input of the OPTI931. + */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(21), 0x82, 0xff); + /* + * This bit sets OPTI931 to automaticaly select FM + * or digital input signal. + */ + snd_opti9xx_write_mask(chip, OPTi9XX_MC_REG(26), 0x01, 0x01); break; #endif /* OPTi93X */ ---------------------------------------------------------------------- Zaaresztuj ministra Koszmarka! http://link.interia.pl/f1ba6

1 0

[alsa-devel] [PATCH] ad1848-lib: remove open_mutex
by Krzysztof Helt 19 Sep '07

19 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch removes open_mutex from the ad1848-lib as open and close operations are called only from pcm layer and mutexed there with pcm->open_mutex. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Another step into crusade against the ad1848-lib. Inspired by Trent's comments about missing locking I have checked and open and close operations are already mutexed in the pcm layer by similarly called pcm->open_mutex. The OSS pcm layer also uses the pcm->open_mutex. Is it right thing to do? (Takashi, Jaroslav)? Regards, Krzysztof diff -urp linux-ref/include/sound/ad1848.h linux-2.6.23/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2007-09-12 13:33:51.000000000 +0200 +++ linux-2.6.23/include/sound/ad1848.h 2007-09-19 17:54:30.000000000 +0200 @@ -154,7 +154,6 @@ struct snd_ad1848 { #endif spinlock_t reg_lock; - struct mutex open_mutex; }; /* exported functions */ diff -urp linux-ref/sound/isa/ad1848/ad1848_lib.c linux-2.6.23/sound/isa/ad1848/ad1848_lib.c --- linux-ref/sound/isa/ad1848/ad1848_lib.c 2007-09-12 13:33:51.000000000 +0200 +++ linux-2.6.23/sound/isa/ad1848/ad1848_lib.c 2007-09-19 17:59:38.000000000 +0200 @@ -391,11 +391,9 @@ static int snd_ad1848_open(struct snd_ad { unsigned long flags; - mutex_lock(&chip->open_mutex); - if (chip->mode & AD1848_MODE_OPEN) { - mutex_unlock(&chip->open_mutex); + if (chip->mode & AD1848_MODE_OPEN) return -EAGAIN; - } + snd_ad1848_mce_down(chip); #ifdef SNDRV_DEBUG_MCE @@ -436,7 +434,6 @@ static int snd_ad1848_open(struct snd_ad spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = mode; - mutex_unlock(&chip->open_mutex); return 0; } @@ -445,11 +442,8 @@ static void snd_ad1848_close(struct snd_ { unsigned long flags; - mutex_lock(&chip->open_mutex); - if (!chip->mode) { - mutex_unlock(&chip->open_mutex); + if (!chip->mode) return; - } /* disable IRQ */ spin_lock_irqsave(&chip->reg_lock, flags); outb(0, AD1848P(chip, STATUS)); /* clear IRQ */ @@ -475,7 +469,6 @@ static void snd_ad1848_close(struct snd_ spin_unlock_irqrestore(&chip->reg_lock, flags); chip->mode = 0; - mutex_unlock(&chip->open_mutex); } /* @@ -893,7 +886,6 @@ int snd_ad1848_create(struct snd_card *c if (chip == NULL) return -ENOMEM; spin_lock_init(&chip->reg_lock); - mutex_init(&chip->open_mutex); chip->card = card; chip->port = port; chip->irq = -1;

2 1

[alsa-devel] ALSA C++ API
by Lasse Kärkkäinen 19 Sep '07

19 Sep '07

Even though I got no replies to my original announcement, here is a new version that wraps hw and sw params completely, has a macro for checking ALSA C function return values (throws alsa::error) and uses different naming style (similar to the C++ standard library). It is better documented now, too. The code is now largely generated with macros, so it isn't very readable anymore :/ Next I'll be trying to figure out how the PCM API itself should be wrapped, for C++ style callbacks, etc. I am using the test/pcm.c program for my tests. Does it use good style for error recovery, etc? Mine will base on it, so this is important. Example of use (a port of what test/pcm.c does using a few separate functions and 140 lines of code): alsa::pcm handle(device); unsigned int rrate = rate; alsa::hw_config(handle) .rate_resample(resample) .set(transfer_methods[method].access) .set(format) .channels(channels) .rate_near(rrate) .buffer_time_near(buffer_time) .get_buffer_size(buffer_size) .period_time_near(period_time) .get_period_size(period_size) .commit(); if (rate != rrate) { std::cout << "Requested rate " << rate << " Hz, got " << rrate << " Hz" << std::endl; rate = rrate; } alsa::sw_config(handle) .start_threshold((buffer_size / period_size) * period_size) .avail_min(period_size) .xfer_align(1) .commit(); I'd prefer to use the Boost libraries (boost::function, boost::noncopyable, etc), but I am afraid that a dependency on them might be problematic. What do you think? #ifndef ALSA_HPP_INCLUDED #define ALSA_HPP_INCLUDED /** @file alsa.hpp @brief An experimental low-level C++ API to ALSA. @version 0.3 @author Lasse Kärkkäinen <tronic> @license GNU LGPL 2.1 or later This is a header-only C++ wrapper for the ALSA library. This means that you do not need to add any new binary files to your build and you will only need to link with -lasound, as if you were using the C API directly. GCC will probably optimize all of the wrapper overhead away in any case, leaving you with a safer and easier API to ALSA, but leaving your binaries just as if you had used C. The library is designed to be very closely related to the C API, so that you can still see what is going on under the hood, and also so that porting existing applications to it is trivial. The interoperatibility between C and C++ APIs is is a major design goal. What this means is that you can freely mix C and C++ API calls with no problems. Usage example: alsa::pcm alsa_pcm; // Create a PCM object (defaults to playback and mode = 0) The above creates an equivalent for the snd_pcm_t*, which is what you need to play or record anything. Make sure that the alsa_pcm object stays alive as long as you need it (and preferrably not longer) by putting it inside a suitable class or even inside the main() function. The object cannot be copied, but you can pass references to it to other objects or functions. The alsa_pcm also automatically converts into snd_pcm_t* as required, so you can use it as an argument for the C API functions. Next you'll need to configure it: unsigned int rate = 44100; unsigned int period; alsa::hw_config(alsa_pcm) // Create a new config space .set(SND_PCM_ACCESS_RW_INTERLEAVED) .set(SND_PCM_FORMAT_S16_LE) .rate_near(rate) .channels(1) .period_size_first(period) // Get the smallest available period size .commit(); // Apply the config to the PCM alsa::hw_config(pcm) constructs a new config space, using the current settings from the PCM, if available, or the "any" space, if not set yet. The any space contains all available hardware configurations and you have to narrow it down to exactly one option by setting some parameters. Trying to narrow it too much (by asking an amount of channels that is not available, for example) causes a failure. In case of failure, an alsa::error is thrown. When this happens, the commit part never gets executed and thus the result is not stored to alsa_pcm and the failed operation will have no effect (even to the temporary hw_config object, which gets destroyed when the exception flies). However, all the operations already performed successfully remain in effect. The rate_near functions behaves like the C API *_near functions do: they take the preferred value as an argument and then they modify the argument, returning the actual rate. For example, if your sound card only supports 48000 Hz, rate will be set to that on that call, even if some later part, such as setting the number of channels, fails. In the example above, a temporary object of type alsa::hw_config was used, but you can also create a variable of it, should you need to test something in between, or if you want to call the C API functions directly (hw_config converts automatically into hw_params, which converts into snd_hw_params_t*). For this, you may use a block like this: { alsa::hw_config hw(alsa_pcm); hw.set(SND_ACCESS_(SND_PCM_ACCESS_MMAP_INTERLEAVED); hw.set(SND_PCM_FORMAT_FLOAT_BE); if (!snd_pcm_hw_params_is_full_duplex(hw)) hw.channels(2); hw.commit(); } (anyone got a better example?) Software configuration works in the same way, using alsa::sw_config, just the parameters to be adjusted are different. When constructed, both sw_config and hw_config try to load config from the given PCM. If that fails, sw_config throws, but hw_config still tries loading the any space. Alternatively, you may supply a snd_pcm_hw/sw_params_t const* as a second argument for the constructor to get a copy of the contents of that that instead. The contents may be loaded (overwrites the old contents) with these functions: .any() Get the "any" configuration (hw_config only) .current() Get the currently active configuration from PCM Once finished with the changes, you should call: .commit() Store current config space to the PCM For enum values SND_PCM_*, you may use the following functions: .get(var) Get the current setting into the given variable .set(val) Set the value requested (also supports masks) .enum_test(val) The same as .set, except that it does not set anything .enum_first(var) Set the first available option, store the selection in var .enum_last(var) Set the last available option, store the selection in var The parameter to manipulate is chosen based on the argument type. The enum_* functions and masks are only available for hardware parameters, not for sw_param. For integer values (times, sizes, counts), these functions are available: .get_X(var) Get the current setting into the given variable .X(val) Set the value requested For ranges, the following can also be used: .get_X_min(var) Get the smallest available value into var .get_X_max(var) Get the largest available value into var .X_min(var) Remove all values smaller than var and store the new smallest value into var. .X_max(var) Remove all values larger than var, store new max in var. .X_minmax(v1, v2) Limit to [v1, v2], store new range in v1, v2. .X_near(var) Set the closest available value and store it in var .X_first(var) Set the first available option, store the selection in var .X_last(var) Set the last available option, store the selection in var For booleans, these functions are available: .get_X(var) Get the current setting (var must be unsigned int or bool) .set_X(val) Set the value (val can be anything that converts into bool) Replace X with the name of the parameter that you want to set. Consult the ALSA C library reference for available options. All functions that modify their arguments require the same type as is used in the C API (often unsigned int or snd_pcm_uframes_t). The only exception is with bool types, where both bool and unsigned int are accepted. For those ranged parameters that support the dir argument (see ALSA docs), the default value is always 0 when writing and NULL (none) when reading. You may supply the preferred dir value or variable as the second argument and then the value will be used or the result will be stored in the variable supplied. For example, the following calls are equivalent: snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_FLOAT_LE) <=> hw.set(SND_PCM_FORMAT_FLOAT_LE) snd_pcm_hw_params_set_rate_resample(pcm, hw, 1) <=> hw.rate_resample(true) snd_pcm_hw_params_set_channels_near(pcm, hw, &num) <=> hw.channels_near(num) snd_pcm_hw_params_get_rate(hw, &rate, NULL) <=> hw.get_rate(rate) snd_pcm_hw_params_get_rate(hw, &rate, &dir) <=> hw.get_rate(rate, dir) ... except for the fact that the C++ versions also check the return values and throw alsa::error if anything goes wrong. alsa::error inherits from std::runtime_error and thus eventually from std::exception, so you can catch pretty much everything by catching that somewhere in your code: try { // do everything here } catch (std::exception& e) { std::cerr << "FATAL ERROR: " << e.what() << std::endl; } However, recent versions of glibc seem to be handling uncaught exceptions quite nicely, so even without a try-catch you may get a nice printout in your console: terminate called after throwing an instance of 'alsa::error' what(): ALSA snd_pcm_hw_params_set_channels failed: Invalid argument Aborted If you need to know the error code, you may call e.code() after catching alsa::error& e. When you want to get low-level, the ALSA_CHECKED macro may prove to be useful. It is used internally by the library for testing errors and throwing exceptions when calling the C API functions. It will throw alsa::error with a description of the error if the function returns a negative value. The macro is well-behaving, so it does not mess up if-elses or other structures and it evaluates its arguments exactly once. Usage: ALSA_CHECKED(snd_pcm_whatever, (arg1, arg2, arg2)); Note: a comma between function name and arguments. In case you really want to get low-level, alsa::hw_params and alsa::sw_params are offered. These only contain the corresponding snd_pcm_*_params_t, but they allocate and free memory automatically and they can also properly copy the struct contents when they get copied. Be aware that the structure contents are not initialized during construction, so you have to initialize it yourself (just like with the C API). They are used internally by hw_config and sw_config and normally it should be better to use these instead of dealing directly with the params. **/ #include <alsa/asoundlib.h> #include <stdexcept> #include <string> /** * A macro that executes func with the given args and tests for errors. * @throws alsa::error if the return value is smaller than zero. **/ #define ALSA_CHECKED(func, args) do { int err = func args; if (err < 0) throw alsa::error(#func, err); } while (0) namespace alsa { namespace util { /** * @short FOR INTERNAL USE ONLY. A utility class similar to * boost::noncopyable, duplicated here in order to avoid * a dependency on the Boost library. **/ class noncopyable { protected: noncopyable() {} ~noncopyable() {} private: noncopyable(noncopyable const&); noncopyable const& operator=(noncopyable const&); }; } /** @short Exception class **/ class error: public std::runtime_error { int err; public: error(std::string const& function, int err): std::runtime_error("ALSA " + function + " failed: " + std::string(snd_strerror(err))), err(err) {} int code() const { return err; } }; /** * @short A minimal RAII wrapper for ALSA PCM. * Automatically converts into snd_pcm_t* as needed, so the ALSA C API * can be used directly with this. **/ class pcm: util::noncopyable { snd_pcm_t* handle; public: pcm(char const* device = "default", snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK, int mode = 0) { ALSA_CHECKED(snd_pcm_open, (&handle, device, stream, mode)); } ~pcm() { snd_pcm_close(handle); } operator snd_pcm_t*() { return handle; } operator snd_pcm_t const*() const { return handle; } }; // RAII wrapper for snd_pcm_hw/sw_params_t types. #define ALSA_HPP_PARAMWRAPPER(type) \ class type##_params {\ snd_pcm_##type##_params_t* handle;\ void init() { ALSA_CHECKED(snd_pcm_##type##_params_malloc, (&handle)); }\ public:\ type##_params() { init(); }\ ~type##_params() { snd_pcm_##type##_params_free(handle); }\ type##_params(type##_params const& orig) { init(); *this = orig; }\ type##_params(snd_pcm_##type##_params_t const* orig) { init(); *this = orig; }\ type##_params& operator=(snd_pcm_##type##_params_t const* params) {\ if (handle != params) snd_pcm_##type##_params_copy(handle, params);\ return *this;\ }\ operator snd_pcm_##type##_params_t*() { return handle; }\ operator snd_pcm_##type##_params_t const*() const { return handle; }\ }; ALSA_HPP_PARAMWRAPPER(hw) ALSA_HPP_PARAMWRAPPER(sw) #undef ALSA_HPP_PARAMWRAPPER // Various helper macros used for generating code for hw_config and sw_config #define ALSA_HPP_FUNC(name, suffix) ALSA_HPP_TEMPLATE(& name(), suffix, (pcm, params)) #define ALSA_HPP_VARGET(name, type) \ ALSA_HPP_TEMPLATE(& get_##name(type& val), _get_##name, (params, &val))\ ALSA_HPP_TEMPLATE(const& get_##name(type& val) const, _get_##name, (params, &val)) #define ALSA_HPP_VAR(name, type) ALSA_HPP_VARGET(name, type)\ ALSA_HPP_TEMPLATE(& name(type val), _set_##name, (pcm, params, val)) #define ALSA_HPP_ENUMVARMINIMAL(name) \ ALSA_HPP_TEMPLATE(& get(snd_pcm_##name##_t& name), _get_##name, (params, &name))\ ALSA_HPP_TEMPLATE(const& get(snd_pcm_##name##_t& name) const, _get_##name, (params, &name))\ ALSA_HPP_TEMPLATE(& set(snd_pcm_##name##_t name), _set_##name, (pcm, params, name)) #define ALSA_HPP_ENUMVAR(name) ALSA_HPP_ENUMVARMINIMAL(name)\ ALSA_HPP_TEMPLATE(& enum_test(snd_pcm_##name##_t name), _test_##name, (pcm, params, name))\ ALSA_HPP_TEMPLATE(& enum_first(snd_pcm_##name##_t& name), _set_##name##_first, (pcm, params, &name))\ ALSA_HPP_TEMPLATE(& enum_last(snd_pcm_##name##_t& name), _set_##name##_last, (pcm, params, &name))\ ALSA_HPP_TEMPLATE(& set(snd_pcm_##name##_mask_t* mask), _set_##name##_mask, (pcm, params, mask)) #define ALSA_HPP_BOOLVAR(name) \ ALSA_HPP_CLASS& get_##name(bool& val) { unsigned int tmp; get_##name(tmp); val = tmp; return *this; }\ /*ALSA_HPP_CLASS const& get_##name(bool& val) const { unsigned int tmp; get_##name(tmp); val = tmp; return *this; }*/\ ALSA_HPP_TEMPLATE(& get_##name(unsigned int& val), _get_##name, (pcm, params, &val))\ /*ALSA_HPP_TEMPLATE(const& get_##name(unsigned int& val) const, _get_##name, (pcm, params, &val))*/\ ALSA_HPP_TEMPLATE(& name(bool val = true), _set_##name, (pcm, params, val)) #define ALSA_HPP_RANGEVAR(name, type) ALSA_HPP_VAR(name, type)\ ALSA_HPP_TEMPLATE(& get_##name##_min(type& min), _get_##name##_min, (params, &min))\ ALSA_HPP_TEMPLATE(const& get_##name##_min(type& min) const, _get_##name##_min, (params, &min))\ ALSA_HPP_TEMPLATE(& get_##name##_max(type& max), _get_##name##_max, (params, &max))\ ALSA_HPP_TEMPLATE(const& get_##name##_max(type& max) const, _get_##name##_max, (params, &max))\ ALSA_HPP_TEMPLATE(& name##_min(type& min), _set_##name##_min, (pcm, params, &min))\ ALSA_HPP_TEMPLATE(& name##_max(type& max), _set_##name##_max, (pcm, params, &max))\ ALSA_HPP_TEMPLATE(& name##_minmax(type& min, type& max), _set_##name##_minmax, (pcm, params, &min, &max))\ ALSA_HPP_TEMPLATE(& name##_near(type& val), _set_##name##_near, (pcm, params, &val))\ ALSA_HPP_TEMPLATE(& name##_first(type& val), _set_##name##_first, (pcm, params, &val))\ ALSA_HPP_TEMPLATE(& name##_last(type& val), _set_##name##_last, (pcm, params, &val)) #define ALSA_HPP_RANGEVARDIR(name, type) \ ALSA_HPP_TEMPLATE(& get_##name(type& val), _get_##name, (params, &val, NULL))\ ALSA_HPP_TEMPLATE(const& get_##name(type& val) const, _get_##name, (params, &val, NULL))\ ALSA_HPP_TEMPLATE(& get_##name(type& val, int& dir), _get_##name, (params, &val, &dir))\ ALSA_HPP_TEMPLATE(const& get_##name(type& val, int& dir) const, _get_##name, (params, &val, &dir))\ ALSA_HPP_TEMPLATE(& get_##name##_min(type& min), _get_##name##_min, (params, &min, NULL))\ ALSA_HPP_TEMPLATE(const& get_##name##_min(type& min) const, _get_##name##_min, (params, &min, NULL))\ ALSA_HPP_TEMPLATE(& get_##name##_min(type& min, int& dir), _get_##name##_min, (params, &min, &dir))\ ALSA_HPP_TEMPLATE(const& get_##name##_min(type& min, int& dir) const, _get_##name##_min, (params, &min, &dir))\ ALSA_HPP_TEMPLATE(& get_##name##_max(type& max), _get_##name##_max, (params, &max, NULL))\ ALSA_HPP_TEMPLATE(const& get_##name##_max(type& max) const, _get_##name##_max, (params, &max, NULL))\ ALSA_HPP_TEMPLATE(& get_##name##_max(type& max, int& dir), _get_##name##_max, (params, &max, &dir))\ ALSA_HPP_TEMPLATE(const& get_##name##_max(type& max, int& dir) const, _get_##name##_max, (params, &max, &dir))\ ALSA_HPP_TEMPLATE(& name(type val, int dir = 0), _set_##name, (pcm, params, val, dir))\ ALSA_HPP_TEMPLATE(& name##_min(type& min), _set_##name##_min, (pcm, params, &min, NULL))\ ALSA_HPP_TEMPLATE(& name##_min(type& min, int& dir), _set_##name##_min, (pcm, params, &min, &dir))\ ALSA_HPP_TEMPLATE(& name##_max(type& max), _set_##name##_max, (pcm, params, &max, NULL))\ ALSA_HPP_TEMPLATE(& name##_max(type& max, int& dir), _set_##name##_max, (pcm, params, &max, &dir))\ ALSA_HPP_TEMPLATE(& name##_minmax(type& min, type& max), _set_##name##_minmax, (pcm, params, &min, NULL, &max, NULL))\ ALSA_HPP_TEMPLATE(& name##_minmax(type& min, int& mindir, type& max, int& maxdir), _set_##name##_minmax, (pcm, params, &min, &mindir, &max, &maxdir))\ ALSA_HPP_TEMPLATE(& name##_near(type& val), _set_##name##_near, (pcm, params, &val, NULL))\ ALSA_HPP_TEMPLATE(& name##_near(type& val, int& dir), _set_##name##_near, (pcm, params, &val, &dir))\ ALSA_HPP_TEMPLATE(& name##_first(type& val), _set_##name##_first, (pcm, params, &val, NULL))\ ALSA_HPP_TEMPLATE(& name##_first(type& val, int& dir), _set_##name##_first, (pcm, params, &val, &dir))\ ALSA_HPP_TEMPLATE(& name##_last(type& val), _set_##name##_last, (pcm, params, &val, NULL))\ ALSA_HPP_TEMPLATE(& name##_last(type& val, int& dir), _set_##name##_last, (pcm, params, &val, &dir)) /** @short A helper object for modifying hw_params of a PCM. **/ class hw_config: util::noncopyable { snd_pcm_t* pcm; hw_params params; public: /** * Construct a new config object, initialized with the current settings * of the PCM or with the "any" configuration space, if there are none. **/ hw_config(snd_pcm_t* pcm): pcm(pcm) { try { current(); } catch (std::runtime_error&) { any(); } } /** Construct a new config object, initialized with a copy from given parameters **/ hw_config(snd_pcm_t* pcm, snd_pcm_hw_params_t const* params): pcm(pcm), params(params) {} operator hw_params&() { return params; } operator hw_params const&() const { return params; } #define ALSA_HPP_CLASS hw_config #define ALSA_HPP_TEMPLATE(proto, suffix, params) ALSA_HPP_CLASS proto { ALSA_CHECKED(snd_pcm_hw_params##suffix, params); return *this; } // Load / store functions ALSA_HPP_FUNC(commit,) ALSA_HPP_FUNC(any, _any) ALSA_HPP_FUNC(current, _current) // Enum functions ALSA_HPP_ENUMVAR(access) ALSA_HPP_ENUMVAR(format) ALSA_HPP_ENUMVAR(subformat) // Bool functions ALSA_HPP_BOOLVAR(rate_resample) ALSA_HPP_BOOLVAR(export_buffer) // Range functions ALSA_HPP_RANGEVAR(channels, unsigned int) ALSA_HPP_RANGEVAR(buffer_size, snd_pcm_uframes_t) // Range functions with direction argument ALSA_HPP_RANGEVARDIR(rate, unsigned int) ALSA_HPP_RANGEVARDIR(period_time, unsigned int) ALSA_HPP_RANGEVARDIR(period_size, snd_pcm_uframes_t) ALSA_HPP_RANGEVARDIR(periods, unsigned int) ALSA_HPP_RANGEVARDIR(buffer_time, unsigned int) ALSA_HPP_RANGEVARDIR(tick_time, unsigned int) #undef ALSA_HPP_TEMPLATE #undef ALSA_HPP_CLASS }; class sw_config: util::noncopyable { snd_pcm_t* pcm; sw_params params; public: sw_config(snd_pcm_t* pcm): pcm(pcm) { current(); } /** Construct a new config object, initialized with a copy from given parameters **/ sw_config(snd_pcm_t* pcm, snd_pcm_sw_params_t const* params): pcm(pcm), params(params) {} operator sw_params&() { return params; } operator sw_params const&() const { return params; } #define ALSA_HPP_CLASS sw_config #define ALSA_HPP_TEMPLATE(proto, suffix, params) ALSA_HPP_CLASS proto { ALSA_CHECKED(snd_pcm_sw_params##suffix, params); return *this; } // Load / store functions ALSA_HPP_FUNC(commit,) ALSA_HPP_FUNC(current, _current) // Enum functions typedef snd_pcm_tstamp_t snd_pcm_tstamp_mode_t; // Workaround for inconsistent naming in asound ALSA_HPP_ENUMVARMINIMAL(tstamp_mode) // Simple variable functions ALSA_HPP_VAR(sleep_min, unsigned int) ALSA_HPP_VAR(avail_min, snd_pcm_uframes_t) ALSA_HPP_VAR(xfer_align, snd_pcm_uframes_t) ALSA_HPP_VAR(start_threshold, snd_pcm_uframes_t) ALSA_HPP_VAR(stop_threshold, snd_pcm_uframes_t) ALSA_HPP_VAR(silence_threshold, snd_pcm_uframes_t) ALSA_HPP_VAR(silence_size, snd_pcm_uframes_t) // Get-only variable ALSA_HPP_VARGET(boundary, snd_pcm_uframes_t) #undef ALSA_HPP_TEMPLATE #undef ALSA_HPP_CLASS }; #undef ALSA_HPP_FUNC #undef ALSA_HPP_VAR #undef ALSA_HPP_VARGET #undef ALSA_HPP_ENUMVAR #undef ALSA_HPP_ENUMVARMINIMAL #undef ALSA_HPP_BOOLVAR #undef ALSA_HPP_RANGEVAR #undef ALSA_HPP_RANGEVARDIR } #endif

3 4

[alsa-devel] [PATCH 2/2] ac97: YMF743 missing controls support
by Keita Maehara 19 Sep '07

19 Sep '07

From: Keita Maehara <maehara(a)debian.org> These patches enable a few YMF743 controls (Tone/3D/IEC958) that won't be detected with the current version of ALSA. Signed-off-by: Keita Maehara <maehara(a)debian.org> --- The second one contains following changes: - A chip-specific SPDIF support for YMF743 (It doesn't have AC97 standard SPDIF registers seen on YMF753). - The implementation for "IEC958 Playback Source" and "IEC958 Mute" are identical to the ones for YMF753. But there is no "IEC958 Output Pin" for YMF743. diff -r 92978f62cc03 pci/ac97/ac97_codec.c --- a/pci/ac97/ac97_codec.c Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_codec.c Tue Sep 18 00:29:26 2007 +0900 @@ -176,7 +176,7 @@ static const struct ac97_codec_id snd_ac { 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, { 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, -{ 0x594d4800, 0xffffffff, "YMF743", NULL, NULL }, +{ 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, { 0x594d4803, 0xffffffff, "YMF753", patch_yamaha_ymf753, NULL }, { 0x83847600, 0xffffffff, "STAC9700,83,84", patch_sigmatel_stac9700, NULL }, @@ -779,6 +779,12 @@ static int snd_ac97_spdif_default_put(st change |= snd_ac97_update_bits_nolock(ac97, AC97_CXR_AUDIO_MISC, AC97_CXR_SPDIF_MASK | AC97_CXR_COPYRGT, v); + } else if (ac97->id == AC97_ID_YMF743) { + change |= snd_ac97_update_bits_nolock(ac97, + AC97_YMF7X3_DIT_CTRL, + 0xff38, + ((val << 4) & 0xff00) | + ((val << 2) & 0x0038)); } else { unsigned short extst = snd_ac97_read_cache(ac97, AC97_EXTENDED_STATUS); snd_ac97_update_bits_nolock(ac97, AC97_EXTENDED_STATUS, AC97_EA_SPDIF, 0); /* turn off */ @@ -1375,7 +1381,8 @@ static int snd_ac97_mixer_build(struct s for (idx = 0; idx < 2; idx++) { if ((err = snd_ctl_add(card, kctl = snd_ac97_cnew(&snd_ac97_controls_tone[idx], ac97))) < 0) return err; - if (ac97->id == AC97_ID_YMF753) { + if (ac97->id == AC97_ID_YMF743 || + ac97->id == AC97_ID_YMF753) { kctl->private_value &= ~(0xff << 16); kctl->private_value |= 7 << 16; } diff -r 92978f62cc03 pci/ac97/ac97_id.h --- a/pci/ac97/ac97_id.h Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_id.h Tue Sep 18 00:29:26 2007 +0900 @@ -54,6 +54,7 @@ #define AC97_ID_ALC658 0x414c4780 #define AC97_ID_ALC658D 0x414c4781 #define AC97_ID_ALC850 0x414c4790 +#define AC97_ID_YMF743 0x594d4800 #define AC97_ID_YMF753 0x594d4803 #define AC97_ID_VT1616 0x49434551 #define AC97_ID_CM9738 0x434d4941 diff -r 92978f62cc03 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_patch.c Tue Sep 18 00:29:26 2007 +0900 @@ -204,6 +204,7 @@ static inline int is_shared_micin(struct /* The following snd_ac97_ymf753_... items added by David Shust (dshust(a)shustring.com) */ +/* Modified for YMF743 by Keita Maehara <maehara(a)debian.org> */ /* It is possible to indicate to the Yamaha YMF7x3 the type of speakers being used. */ static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -287,6 +288,63 @@ static int snd_ac97_ymf7x3_spdif_source_ return -EINVAL; val = ucontrol->value.enumerated.item[0] << 1; return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val); +} + +static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) +{ + struct snd_kcontrol *kctl; + int err; + + if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0) + return err; + strcpy(kctl->id.name, "3D Control - Wide"); + kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); + snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) + return err; + snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); + return 0; +} + +static const struct snd_kcontrol_new snd_ac97_yamaha_ymf743_controls_spdif[3] = { + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), AC97_YMF7X3_DIT_CTRL, 0, 1, 0), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, + }, + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) +}; + +static int patch_yamaha_ymf743_build_spdif(struct snd_ac97 * ac97) +{ + int err; + + if ((err = patch_build_controls(ac97, &snd_ac97_controls_spdif[0], 3)) < 0) + return err; + if ((err = patch_build_controls(ac97, snd_ac97_yamaha_ymf743_controls_spdif, ARRAY_SIZE(snd_ac97_yamaha_ymf743_controls_spdif))) < 0) + return err; + /* set default PCM S/PDIF params */ + /* PCM audio,no copyright,no preemphasis,PCM coder,original */ + snd_ac97_write_cache(ac97, AC97_YMF7X3_DIT_CTRL, 0xa201); + return 0; +} + +static struct snd_ac97_build_ops patch_yamaha_ymf743_ops = { + .build_spdif = patch_yamaha_ymf743_build_spdif, + .build_3d = patch_yamaha_ymf7x3_3d, +}; + +static int patch_yamaha_ymf743(struct snd_ac97 * ac97) +{ + ac97->build_ops = &patch_yamaha_ymf743_ops; + ac97->caps |= AC97_BC_BASS_TREBLE; + ac97->caps |= 0x04 << 10; /* Yamaha 3D enhancement */ + ac97->rates[AC97_RATES_SPDIF] = SNDRV_PCM_RATE_48000; /* 48k only */ + ac97->ext_id |= AC97_EI_SPDIF; /* force the detection of spdif */ + return 0; } /* The AC'97 spec states that the S/PDIF signal is to be output at pin 48. @@ -347,22 +405,6 @@ static const struct snd_kcontrol_new snd }, AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) }; - -static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) -{ - struct snd_kcontrol *kctl; - int err; - - if ((err = snd_ctl_add(ac97->bus->card, kctl = snd_ac97_cnew(&snd_ac97_controls_3d[0], ac97))) < 0) - return err; - strcpy(kctl->id.name, "3D Control - Wide"); - kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); - snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) - return err; - snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); - return 0; -} static int patch_yamaha_ymf753_post_spdif(struct snd_ac97 * ac97) { diff -r 92978f62cc03 pci/ac97/ac97_proc.c --- a/pci/ac97/ac97_proc.c Tue Sep 18 00:28:51 2007 +0900 +++ b/pci/ac97/ac97_proc.c Tue Sep 18 00:29:26 2007 +0900 @@ -236,10 +236,14 @@ static void snd_ac97_proc_read_main(stru val = snd_ac97_read(ac97, AC97_PCM_MIC_ADC_RATE); snd_iprintf(buffer, "PCM MIC ADC : %iHz\n", val); } - if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF)) { + if ((ext & AC97_EI_SPDIF) || (ac97->flags & AC97_CS_SPDIF) || + (ac97->id == AC97_ID_YMF743)) { if (ac97->flags & AC97_CS_SPDIF) val = snd_ac97_read(ac97, AC97_CSR_SPDIF); - else + else if (ac97->id == AC97_ID_YMF743) { + val = snd_ac97_read(ac97, AC97_YMF7X3_DIT_CTRL); + val = 0x2000 | (val & 0xff00) >> 4 | (val & 0x38) >> 2; + } else val = snd_ac97_read(ac97, AC97_SPDIF); snd_iprintf(buffer, "SPDIF Control :%s%s%s%s Category=0x%x Generation=%i%s%s%s\n",

2 6

[alsa-devel] Routing multiple signals to one output with ice1712 (Delta 66)
by Mathis Hofer 19 Sep '07

19 Sep '07

Hi folks I want to setup the following audio routing: pcm + input1 (mic) + input2 (mic) -> output1 pcm + input1 (mic) + input2 (mic) -> output2 ...with pcm being the audio playback and input1&2 being two microphones. With envy24control it is possible to select one signal for each output, but not multiple signals. Is this a restriction of envy24control, the alsa driver or is it simply not supported by the hardware? Regards, Mathis -- DreamLab Technologies AG Monbijoustrasse 36 3011 Bern Switzerland Tel: +41 31 398 66 66 Fax: +41 31 398 66 69 PGP Key ID: 2462240B

2 2

[alsa-devel] Separate default devices for playback and capture
by Lasse Kärkkäinen 19 Sep '07

19 Sep '07

I've been reading the library source code and from what I gather, it seems that it just fundamentally cannot support separate config sections for playback and capture. What should the fix be? I can see three options: 1. allow specifying in asound.conf whether a section is for recording or playback (needs major redesign of conf.c and some changes in pcm.c). 2. if the name is "default", use "capture" (for example) instead, in snd_pcm_open_noupdate, maybe reverting to "default" if the other device cannot be found (dirty, but very easy to implement) 3. tell application developers to use some other device name for capture: "capture", instead of "default", for example (needs a little change in every application program that uses ALSA capture).

2 1

Re: [alsa-devel] Audiophile Delta 24/96 Midi Out continuously sends zeroes (alsa 1.0.14)
by Clemens Ladisch 19 Sep '07

19 Sep '07

Romain Bossart wrote: > Selon Clemens Ladisch <cladisch(a)fastmail.net>: > > That is, the zeroes are _not_ sent when you're not using jack? When > > nothing is connected, or when you use aplaymidi or amidi to send > > something, everything is OK? > > Here is a summary of the various tests I did. I have just one card: Audiophile > Delta 24/96. My kernel is now 2.6.22.1 RT patched & alsa 1.0.14, > Audio works perfectly. I don't think jack is the problem. > > Config #2 = Delta Midi Out -> (Midi In) Instrument (Midi Out/Thru) -> Delta Midi > In and 'amidi -p hw:0,0 -d'. Jack being started or not. > > Symptoms: tons of zeroes are shown by amidi as soon as I launch it. The stream > is so intense I can actually hear it in my speakers/monitors (need to switch to > better MIDI cables, I know!). amidi and /proc/asound/card0/midi0 give the same number > of transmitted bytes. With "transmitted", you meant "received"? When Jack is not running, there is no software to send anything. If the number of _transmitted_ bytes changes anyway, this would be a bug in ALSA. > Config #2b = Delta Midi Out -> Delta Midi In > and 'amidi -p hw:0,0 -d', Jack being started or not. > > Symptoms: identical to config#2. Moreover, if meanwhile I send some Midi commands > with amidi ('amidi -p hw:0,0 -S C01C'), I don't find this data in my dump. Apparently, the output port is broken. While it is theoretically possible that the driver doesn't initialize the card correctly, I cannot imagine a bug that would cause the port to presume that there is data to send. Could you test if the port sends data even before the driver (or Linux) is loaded, or in Windows? Regards, Clemens

2 2

[alsa-devel] [PATCH] ad1838/cs4231 -- fix MCE timeout upon initial load
by Rene Herman 19 Sep '07

19 Sep '07

Hi Takashi. When the ad1848/cs2431 is first being initialized, auto-calibration may not be set causing a timeout waiting for it in snd_ad1848/cs4231_mce_down(). This has no dire consequences other than an alarming printk, but since what we need to wait for is for the calibration to _finish_, let's just check for that instead. The early chips need a slight delay (as commented -- 5 sample periods) to be sure that _if_ calibration is going to happen, it has started when we check While the CS4231A datasheet implies it'll happen immediately on downing MCE, some testing is showing that there's a window there as well, so just do the delay everywhere. Thanks to Krysztof Helt for pinpointing this problem. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 8094282..dcd4435 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -227,16 +227,15 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) spin_unlock_irqrestore(&chip->reg_lock, flags); return; } - /* calibration process */ - for (timeout = 500; timeout > 0 && (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0; timeout--); - if ((snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0) { - snd_printd("mce_down - auto calibration time out (1)\n"); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } + /* + * Wait for (possible -- during init auto-calibration may not be set) + * calibration process to start. Needs upto 5 sample periods on AD1848 + * which at the slowest possible rate of 5.5125 kHz means 907 us. + */ + msleep(1); #if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); + printk("(2) jiffies = %li\n", jiffies); #endif time = HZ / 4; while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 914d77b..9e8e0f1 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -346,16 +346,14 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } snd_cs4231_busy_wait(chip); - /* calibration process */ - - for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--) - udelay(10); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); - return; - } + /* + * Wait for (possible -- during init auto-calibration may not be set) + * calibration process to start. Needs upto 5 sample periods on AD1848 + * which at the slowest possible rate of 5.5125 kHz means 907 us. + */ + msleep(1); #if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); + printk("(2) jiffies = %li\n", jiffies); #endif /* in 10 ms increments, check condition, up to 250 ms */ timeout = 25;

4 35

[alsa-devel] Enabling in-kernel synch for M-Audio boards
by Ludovico Verducci 19 Sep '07

19 Sep '07

Hello all! I'm developing a complex multichannel audio distribution system where multiple linux boxes will stream audio data over ethernet and then should play audio at sample level resolution synchronization. The boxes clocks are synchronized over ethernet using PTP. I need to keep in synch the audio board's clocks and I can't use an external wordclock nor s/pdif. At the moment I'm experimenting with m-audio delta 1010, based on envy24 (VT1712). I know these boards can synch their clocks over PCI signalling (in windows drivers this feature support synching up to 4 boards), but ALSA drivers don't implement that feature. I'm trying to implement a similar feature (controlling board clock synch through kernel module), but it's very difficult without a good starting point! Is there anyone that could give me some help, support or infos? I would eventually appreciate also hardware reference designs or application notes about the VIA envy24 VT1712 chip. Obviously, should I ever be successful in the implementation, those features could be eventually added to ALSA drivers. Thank you all in advance. Regards, Ludovico

3 7

Re: [alsa-devel] Intel HDA / AD1986A problems
by Takashi Iwai 18 Sep '07

18 Sep '07

At Tue, 21 Aug 2007 21:45:52 -0400, Peter Skensved wrote: > > On Tue, Aug 21, 2007 at 02:34:22PM +0200, Takashi Iwai wrote: > > At Mon, 20 Aug 2007 22:27:40 -0400, > > Peter Skensved wrote: > > > > > > > > > > > > Configuration : Lenovo 3000 N100 laptop with Intel HDA / Anlog Devices AD1986A > > > soundcard running CentOS5.0 with 2.6.18-8.1.8.el5 kernel ( uses 1.0.12rc1 based > > > alsa driver ). > > > > > > The driver works with the model=laptop-eapd option except for a few things : > > > > > > 1) Plugging in an external headset will not mute the internal speakers ( has to be done > > > manually in alsa-mixer ) > > > > That's an unimplemented feature. It shouldn't be too hard to add it > > if you help debugging well (i.e. you can build and test drivers from > > HG tree without hitch). > > > > Ok - I'm willing to give it a try. Do I need development version of theh library > and the utilities as well ? Recommended, but not mandatory. > > > 2) The PC speaker ( bell ) appears to be permanently muted. > > > > Partly intended (as I dislike it ;) > > > > It is nice to have the option ... Another opinion :) > > > 3) The external microphone input does not appear to be working no matter what I do. > > > > Do you mean for recording or analog loopback? > > > > Yes - for recording. I was hoping to get it to work well enough to use skype - Note that recording and analog-loopback are different things. > > > 4) When recording from the internal microphone the level is very, very low and almost > > > drowns in static noise. > > > > There are "Mic Boost" and "Front Mic Boost" mixer volumes. > > > > Not with model=laptop-eapd - my alas mixer only shows Master, PCM, Mic, Capture, Mix, > External Amp and Internal Amp The Mic Boost control was added _after_ 1.0.14 release. (Correction in the above text: Front Mic Boost doesn't exist for AD1986A but for other codec.) So, please try HG version first. Then let's debug further. Takashi

2 3

[alsa-devel] AD1986A stop working if adjust the volume
by zhejiang 18 Sep '07

18 Sep '07

Hi, I have a ICH8M + AD1986A machine and run alsa-driver-1.0.15rc. I found that : In 6 channel mode,if adjust the front or pcm volume when playing music, all channel will be muted,but if adjust the volume alone(don't play music at the same time),every thing is ok. I use printk to display the verb in the azx_send_cmd(), when adjusting the volume in either case, there are just two verb (0x300) sent to the nid(0x1b) [correct action]. But in the first case, the dac widget amp-out vals are all affected. Some other interesting thing, If i adjust the Sur or CLFE volume when playing music, the printk show that correct verb is sent, but the codec status don't change. If I adjust them alone,every thing is ok. The card0/codec#0 before adjust volume : Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0xbfd60000 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 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 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: [0x11 0x11] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x11 0x11] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: 0x0 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 Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] 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: [0x80 0x80] 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: [0x8f] 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: [0x80 0x80] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP 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: [0x10 0x10] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT 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: [0x10 0x10] Pincap 0x0837: IN OUT Detect Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x0e 0x0f] Pincap 0x081737: IN OUT Detect Pin Default 0x01019011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT 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: [0x80] Pincap 0x0810: OUT Pin Default 0x901301f0: [Fixed] Speaker at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x99930123: [Fixed] Aux at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330122: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 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 The card0/codec#0 after adjust "Front " volume: Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0xbfd60000 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 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x60]: 44100 48000 bits [0x2]: 16 formats [0x5]: PCM AC3 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: [0x97 0x97] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates [0x7f]: 8000 11025 16000 22050 32000 44100 48000 bits [0x6]: 16 20 formats [0x1]: PCM Power: 0x0 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 Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] 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: [0x97 0x97] 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: [0x97] 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: [0x97 0x97] 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: [0x97 0x97] 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: [0x97 0x97] 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: [0x8f] 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: [0x9f 0x9f] Pincap 0x081f: OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP 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: [0x9f 0x9f] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT 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: [0x9f 0x9f] Pincap 0x0837: IN OUT Detect Pin Default 0x01013012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Pincap 0x081737: IN OUT Detect Pin Default 0x01019011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT 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: [0x9f] Pincap 0x0810: OUT Pin Default 0x901301f0: [Fixed] Speaker at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a19020: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x01813021: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x99930123: [Fixed] Aux at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x99330122: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 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

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[alsa-devel] [PATCH 1/2] ac97: YMF743 missing controls support
by Keita Maehara 18 Sep '07

18 Sep '07

From: Keita Maehara <maehara(a)debian.org> These patches enable some YMF743 controls (Tone/3D/IEC958) that won't be detected with the current version of ALSA. Signed-off-by: Keita Maehara <maehara(a)debian.org> --- The first one contains only cosmetic changes to share a few YMF753-specific symbols with YMF743. diff -r 4a3f9f77e676 include/ac97_codec.h --- a/include/ac97_codec.h Mon Sep 17 16:26:16 2007 +0200 +++ b/include/ac97_codec.h Tue Sep 18 00:27:24 2007 +0900 @@ -345,9 +345,9 @@ #define AC97_ALC650_GPIO_STATUS 0x78 #define AC97_ALC650_CLOCK 0x7a -/* specific - Yamaha YMF753 */ -#define AC97_YMF753_DIT_CTRL2 0x66 /* DIT Control 2 */ -#define AC97_YMF753_3D_MODE_SEL 0x68 /* 3D Mode Select */ +/* specific - Yamaha YMF7x3 */ +#define AC97_YMF7X3_DIT_CTRL 0x66 /* DIT Control (YMF743) / 2 (YMF753) */ +#define AC97_YMF7X3_3D_MODE_SEL 0x68 /* 3D Mode Select */ /* specific - C-Media */ #define AC97_CM9738_VENDOR_CTRL 0x5a diff -r 4a3f9f77e676 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Mon Sep 17 16:26:16 2007 +0200 +++ b/pci/ac97/ac97_patch.c Tue Sep 18 00:27:24 2007 +0900 @@ -205,8 +205,8 @@ static inline int is_shared_micin(struct /* The following snd_ac97_ymf753_... items added by David Shust (dshust(a)shustring.com) */ -/* It is possible to indicate to the Yamaha YMF753 the type of speakers being used. */ -static int snd_ac97_ymf753_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) +/* It is possible to indicate to the Yamaha YMF7x3 the type of speakers being used. */ +static int snd_ac97_ymf7x3_info_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[3] = { "Standard", "Small", "Smaller" @@ -221,12 +221,12 @@ static int snd_ac97_ymf753_info_speaker( return 0; } -static int snd_ac97_ymf753_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_get_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_3D_MODE_SEL]; + val = ac97->regs[AC97_YMF7X3_3D_MODE_SEL]; val = (val >> 10) & 3; if (val > 0) /* 0 = invalid */ val--; @@ -234,7 +234,7 @@ static int snd_ac97_ymf753_get_speaker(s return 0; } -static int snd_ac97_ymf753_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_put_speaker(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -242,20 +242,20 @@ static int snd_ac97_ymf753_put_speaker(s if (ucontrol->value.enumerated.item[0] > 2) return -EINVAL; val = (ucontrol->value.enumerated.item[0] + 1) << 10; - return snd_ac97_update(ac97, AC97_YMF753_3D_MODE_SEL, val); -} - -static const struct snd_kcontrol_new snd_ac97_ymf753_controls_speaker = + return snd_ac97_update(ac97, AC97_YMF7X3_3D_MODE_SEL, val); +} + +static const struct snd_kcontrol_new snd_ac97_ymf7x3_controls_speaker = { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "3D Control - Speaker", - .info = snd_ac97_ymf753_info_speaker, - .get = snd_ac97_ymf753_get_speaker, - .put = snd_ac97_ymf753_put_speaker, -}; - -/* It is possible to indicate to the Yamaha YMF753 the source to direct to the S/PDIF output. */ -static int snd_ac97_ymf753_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) + .info = snd_ac97_ymf7x3_info_speaker, + .get = snd_ac97_ymf7x3_get_speaker, + .put = snd_ac97_ymf7x3_put_speaker, +}; + +/* It is possible to indicate to the Yamaha YMF7x3 the source to direct to the S/PDIF output. */ +static int snd_ac97_ymf7x3_spdif_source_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static char *texts[2] = { "AC-Link", "A/D Converter" }; @@ -268,17 +268,17 @@ static int snd_ac97_ymf753_spdif_source_ return 0; } -static int snd_ac97_ymf753_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val >> 1) & 1; return 0; } -static int snd_ac97_ymf753_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) +static int snd_ac97_ymf7x3_spdif_source_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; @@ -286,7 +286,7 @@ static int snd_ac97_ymf753_spdif_source_ if (ucontrol->value.enumerated.item[0] > 1) return -EINVAL; val = ucontrol->value.enumerated.item[0] << 1; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0002, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0002, val); } /* The AC'97 spec states that the S/PDIF signal is to be output at pin 48. @@ -311,7 +311,7 @@ static int snd_ac97_ymf753_spdif_output_ struct snd_ac97 *ac97 = snd_kcontrol_chip(kcontrol); unsigned short val; - val = ac97->regs[AC97_YMF753_DIT_CTRL2]; + val = ac97->regs[AC97_YMF7X3_DIT_CTRL]; ucontrol->value.enumerated.item[0] = (val & 0x0008) ? 2 : (val & 0x0020) ? 1 : 0; return 0; } @@ -325,7 +325,7 @@ static int snd_ac97_ymf753_spdif_output_ return -EINVAL; val = (ucontrol->value.enumerated.item[0] == 2) ? 0x0008 : (ucontrol->value.enumerated.item[0] == 1) ? 0x0020 : 0; - return snd_ac97_update_bits(ac97, AC97_YMF753_DIT_CTRL2, 0x0028, val); + return snd_ac97_update_bits(ac97, AC97_YMF7X3_DIT_CTRL, 0x0028, val); /* The following can be used to direct S/PDIF output to pin 47 (EAPD). snd_ac97_write_cache(ac97, 0x62, snd_ac97_read(ac97, 0x62) | 0x0008); */ } @@ -334,9 +334,9 @@ static const struct snd_kcontrol_new snd { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", - .info = snd_ac97_ymf753_spdif_source_info, - .get = snd_ac97_ymf753_spdif_source_get, - .put = snd_ac97_ymf753_spdif_source_put, + .info = snd_ac97_ymf7x3_spdif_source_info, + .get = snd_ac97_ymf7x3_spdif_source_get, + .put = snd_ac97_ymf7x3_spdif_source_put, }, { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, @@ -345,10 +345,10 @@ static const struct snd_kcontrol_new snd .get = snd_ac97_ymf753_spdif_output_pin_get, .put = snd_ac97_ymf753_spdif_output_pin_put, }, - AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF753_DIT_CTRL2, 2, 1, 1) -}; - -static int patch_yamaha_ymf753_3d(struct snd_ac97 * ac97) + AC97_SINGLE(SNDRV_CTL_NAME_IEC958("",NONE,NONE) "Mute", AC97_YMF7X3_DIT_CTRL, 2, 1, 1) +}; + +static int patch_yamaha_ymf7x3_3d(struct snd_ac97 * ac97) { struct snd_kcontrol *kctl; int err; @@ -358,9 +358,9 @@ static int patch_yamaha_ymf753_3d(struct strcpy(kctl->id.name, "3D Control - Wide"); kctl->private_value = AC97_SINGLE_VALUE(AC97_3D_CONTROL, 9, 7, 0); snd_ac97_write_cache(ac97, AC97_3D_CONTROL, 0x0000); - if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf753_controls_speaker, ac97))) < 0) + if ((err = snd_ctl_add(ac97->bus->card, snd_ac97_cnew(&snd_ac97_ymf7x3_controls_speaker, ac97))) < 0) return err; - snd_ac97_write_cache(ac97, AC97_YMF753_3D_MODE_SEL, 0x0c00); + snd_ac97_write_cache(ac97, AC97_YMF7X3_3D_MODE_SEL, 0x0c00); return 0; } @@ -374,7 +374,7 @@ static int patch_yamaha_ymf753_post_spdi } static struct snd_ac97_build_ops patch_yamaha_ymf753_ops = { - .build_3d = patch_yamaha_ymf753_3d, + .build_3d = patch_yamaha_ymf7x3_3d, .build_post_spdif = patch_yamaha_ymf753_post_spdif };

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