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[alsa-devel] Why there is no SNDRV_PCM_RATE_24000 in pcm.h
by Wei Kai 10 Aug '07

10 Aug '07

hi, guys I checked the include/sound/pcm.h in kernel and find this problem, it seemed strange, so many codecs support the 24000 sample rate, why there is no such definition. Any special concern?

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[alsa-devel] [PATCH] Add quirk for Stanton ScratchAmp v1
by Mark Hills 10 Aug '07

10 Aug '07

Hello, This patch is a USB quirk to ensure the Stanton Scratchamp v1 is detected (bugtrack #2932). The interface is two USB devices in the same physical box. Note that this is the USB ScratchAmp v1 and not the later v2 (firewire) model. Signed-off-by: Mark Hills <mark(a)pogo.org.uk> diff -r d19ec80205a5 usb/usbquirks.h --- a/usb/usbquirks.h Fri Jul 27 19:15:54 2007 +0200 +++ b/usb/usbquirks.h Sun Jul 29 23:45:00 2007 +0100 @@ -1774,4 +1774,22 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Stanton/N2IT Final Scratch v1 device ('Scratchamp') */ +{ + USB_DEVICE(0x103d, 0x0100), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Stanton", + .product_name = "ScratchAmp", + .ifnum = QUIRK_NO_INTERFACE + } +}, +{ + USB_DEVICE(0x103d, 0x0101), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Stanton", + .product_name = "ScratchAmp", + .ifnum = QUIRK_NO_INTERFACE + } +}, + #undef USB_DEVICE_VENDOR_SPEC

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[alsa-devel] [PATCH 2/2] [VIRTEX] Register AC97 Controller Reference with the platform bus
by Joachim Förster 09 Aug '07

09 Aug '07

From: Joachim Foerster <JOFT(a)gmx.de> (Patch for Linus' master branch, date 2007/08/08) Signed-off-by: Joachim Foerster <JOFT(a)gmx.de> --- arch/ppc/syslib/virtex_devices.c | 28 ++++++++++++++++++++++++++++ 1 files changed, 28 insertions(+), 0 deletions(-) diff --git a/arch/ppc/syslib/virtex_devices.c b/arch/ppc/syslib/virtex_devices.c index 62a9495..0d6f0ac 100644 --- a/arch/ppc/syslib/virtex_devices.c +++ b/arch/ppc/syslib/virtex_devices.c @@ -121,6 +121,29 @@ }, \ } +#define XPAR_AC97_CONTROLLER_REFERENCE(num) { \ + .name = "ml403_ac97cr", \ + .id = num, \ + .num_resources = 3, \ + .resource = (struct resource[]) { \ + { \ + .start = XPAR_OPB_AC97_CONTROLLER_REF_0_BASEADDR, \ + .end = XPAR_OPB_AC97_CONTROLLER_REF_0_HIGHADDR, \ + .flags = IORESOURCE_MEM, \ + }, \ + { \ + .start = XPAR_OPB_INTC_0_OPB_AC97_CONTROLLER_REF_0_PLAYBACK_INTERRUPT_INTR, \ + .end = XPAR_OPB_INTC_0_OPB_AC97_CONTROLLER_REF_0_PLAYBACK_INTERRUPT_INTR, \ + .flags = IORESOURCE_IRQ, \ + }, \ + { \ + .start = XPAR_OPB_INTC_0_OPB_AC97_CONTROLLER_REF_0_RECORD_INTERRUPT_INTR, \ + .end = XPAR_OPB_INTC_0_OPB_AC97_CONTROLLER_REF_0_RECORD_INTERRUPT_INTR, \ + .flags = IORESOURCE_IRQ, \ + }, \ + }, \ +} + /* UART 8250 driver platform data table */ struct plat_serial8250_port virtex_serial_platform_data[] = { #if defined(XPAR_UARTNS550_0_BASEADDR) @@ -221,6 +244,11 @@ struct platform_device virtex_platform_devices[] = { #if defined(XPAR_TFT_3_BASEADDR) XPAR_TFT(3), #endif + + /* AC97 Controller Reference instances */ +#if defined(XPAR_OPB_AC97_CONTROLLER_REF_0_BASEADDR) + XPAR_AC97_CONTROLLER_REFERENCE(0), +#endif }; /* Early serial support functions */ -- 1.5.2.4

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[alsa-devel] [PATCH] asihpi unify play/capture timer functions
by Eliot Blennerhassett 09 Aug '07

09 Aug '07

Unify the playback and capture timer function. A group of linked streams only uses one timer. Cope with linked streams on more than one card (untested) Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com>

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Re: [alsa-devel] Website to Wiki migration
by Ash Willis 09 Aug '07

09 Aug '07

> Anything missing - tell about it here: > http://bugtrack.alsa-project.org/main/index.php/MigrationPlan > > One proposal for prettier front page is here: > http://bugtrack.alsa-project.org/main/index.php/Talk:Main_Page > Go ahead - criticise, praise, make your own... > The Wiki's looking great. It's obvious that you've thought about where everything should be and how it should all tie in instead of blinding porting the old pages. It's looking very fresh and making much more sense. Is it just me though, or is "Vendor-A-trend Vendor-ALi Vendor-AMD" much harder to read than "A-trend ALi AMD"? Am I OK to go ahead and remove those prefixes or do we want those kept? Same with the modules, although I find the module prefixes much easier to deal with visually for some odd reason. Also, two of the modules that don't have pages were written by myself. I'd be glad to make the pages for those modules but I don't really know what should go on those pages TBH. IMO, most of the info on the module pages is outdated and pointless. At the very least, it's heavily repeated. Is this something we should look at? Personally I find that it's letting down the rather nice new site. Ash -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains

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[alsa-devel] ALSA on MIPS platform
by Songmao Tian 09 Aug '07

09 Aug '07

Hi, In http://www.linux-mips.org/archives/linux-mips/2007-04/msg00114.html, Atsushi Nemoto pointed out the problem is discussed before, the thread can be found at http://lkml.org/lkml/2006/1/25/117. Thanks Atsushi Nemoto:) The problem is clear: 1. dma_alloc_noncoherent() return a non-cached address, and virt_to_page() need a cached logical addr (Have I named it right?) 2. mmaped dam buffer should be non-cached. We have a ugly patch, but we want to solve the problem cleanly, so can anyone show me the way? Regards, Tian diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59b29cd..fd0aa66 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3138,7 +3138,11 @@ static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, return NOPAGE_OOM; /* XXX: is this really due to OOM? */ } else { vaddr = runtime->dma_area + offset; +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) + page = virt_to_page(CAC_ADDR(vaddr)); +#else page = virt_to_page(vaddr); +#endif } get_page(page); if (type) @@ -3254,6 +3258,11 @@ static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) substream = pcm_file->substream; snd_assert(substream != NULL, return -ENXIO); +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) + /* all mmap using uncached mode */ + area->vm_page_prot = pgprot_noncached(area->vm_page_prot); +#endif + offset = area->vm_pgoff << PAGE_SHIFT; switch (offset) { case SNDRV_PCM_MMAP_OFFSET_STATUS: diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index cefd228..2251e70 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -91,12 +91,21 @@ void *snd_malloc_sgbuf_pages(struct device *device, } sgbuf->table[i].buf = tmpb.area; sgbuf->table[i].addr = tmpb.addr; +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) + sgbuf->page_table[i] = virt_to_page(CAC_ADDR(tmpb.area)); +#else sgbuf->page_table[i] = virt_to_page(tmpb.area); +#endif sgbuf->pages++; } sgbuf->size = size; +#if defined(__mips__) && defined(CONFIG_DMA_NONCOHERENT) + /* maybe we should use uncached accelerated mode */ + dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, pgprot_noncached(PAGE_KERNEL)); +#else dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); +#endif if (! dmab->area) goto _failed; return dmab->area;

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[alsa-devel] Linux 2.6.22 and SB-Live! - broken microphone capture?
by Chris Rankin 09 Aug '07

09 Aug '07

Hi, I have been testing my old SB Live! card with Linux 2.6.22 and the 1.0.14 alsa lib and utils, and I have discovered that the following no longer works: $ arecord -f cd -t wav -d 4 > testin.wav Recording WAVE 'stdin' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo arecord: pcm_read:1346: read error: Input/output error The testin.wav file is always 44 bytes long. The microphone works, is plugged into the correct slot, and (AFAICT) all the mixer settings are correct: - playback AC97 = 0 - capture AC97 = 100 - mic select = mic1 - mic set to capture - capture volume = 100, and set to capture Has something broken in ALSA, please? Playback seems fine, and I get some kind of noise from the speakers if I raise the Microphone playback volume to 100 and then blow on the microphone. I also have an Audigy 2 card, and microphone capture works fine there. Can anyone confirm this, please? Thanks, Chris ___________________________________________________________ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.ht…

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[alsa-devel] Website to Wiki migration
by Eliot Blennerhassett 08 Aug '07

08 Aug '07

Greetings, IMO the basic content of the static website has been migrated to the wiki. This doesn't include the following: * http://alsa-project.org/changes/ Changelogs .txt files are referenced, assume they are auto generated? * Contents of http://www.alsa-project.org/alsa/ftp/ * Contents of http://www.alsa-project.org/~iwai/ include "Writing an alsa driver" * User added comments in the soundcard matrix are referenced. (Is there any way to find out which modules in the matrix have user comments attached so that they can be copied across to the wiki?) Anything missing - tell about it here: http://bugtrack.alsa-project.org/main/index.php/MigrationPlan One proposal for prettier front page is here: http://bugtrack.alsa-project.org/main/index.php/Talk:Main_Page Go ahead - criticise, praise, make your own... cheers Eliot

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[alsa-devel] Possible problems with alsa soc
by Nobin Mathew 08 Aug '07

08 Aug '07

Hi Liam, There are some players which will call snd-pcm_close() before any open call, i dont know why they are calling this. So in ASoC it will put the stream in SNDRV_CTL_POWER_D3hot and pop_wait equal to 1. This state will make problems in snd_pcm_prepare (i.e soc_pcm_prepare). Trouble creating players and libraries are mpg123 and gstreamer library. Takashi, We now have priority based alsa plugin, as I explained earlier. all these credits goes to Sudhanshu. This will be usefull for many gps, bluetooth and audio SoCs. But we need to make some changes to make it generic, later we will summit this as a patch. Thanks Nobin Mathew

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[alsa-devel] how to 'alias' a mixer element?
by John L. Utz III 08 Aug '07

08 Aug '07

Hello; If one had a mixer item called 'Surround' and one really wanted to access it as an item called 'Rear' instead of using the existing name 'Surround', would it be possible to create a virtual control called Rear that just maps directly to Surround? tnx! johnu

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[alsa-devel] trouble compiling alsa-drivers
by Wallace, Brooke 08 Aug '07

08 Aug '07

The alsa-drivers INSTALL file tells me: On 2.6 kernels, the build directory has to be given via --with-build=<kernel_build_dir> option additionaly, too. What exactly is mean by kernel_build_dir? I should not have to recompile my kernel because I already have the sourcecore module. Configure seems to work fine and locates my kernel source code: ./configure --with-cards=hda-intel --with-sequencer=yes But when I compile, it fails in acore/pcm_native.c line #75, acore/pcm_native.c line #75 uses the macro DEFINE_RWLOCK which apparently has not been defined yet. Searching thu the alsa-drivers source, I find that include/adriver.h defines this macro. What am I doing wrong? Why doesn't adriver.h get included when I compile acore/pcm_native.c? I'm using the latest release - alsa-driver-1.0.14rc4 My Kernel sources are: 2.6.9-34.EL-smp-i686 Alsa is already working fine on this system using the HDA Intel driver, its a RHEL- 3.4.5-2 system. I just want to re-compile the latest ALSA software and start writing a dummy user space driver that prints to a file or something. I should be able to cmopile all of the latest ALSA software... Any help is appreciated. -Brooke

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[alsa-devel] Noise on rear speaker when switching from 2.0 to 5.1 sound
by Felix Rauscher 07 Aug '07

07 Aug '07

Hello, I have a Hercules Muse 5.1 DVD (C-Media PCI CMI8738-MC6). I have a problem when a running xine (e.g xineliboutput from vdr) switches from 2.0 to 5.1 output. Then the sound quality of the rear speakers is terrible. There is a noise on the rear speakers. If I start the xineliboutput-client when vdr already plays a programme with a 5.1 sound track, the quality is ok. It looks like the switching between 2.0 to 5.1 is a problem. Thank you for your help! Felix

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Re: [alsa-devel] SOLVED Re: snd_pcm_hw_params_set_format fails with invalid argument
by stan 07 Aug '07

07 Aug '07

I sent this directly to Takashi by mistake. I could have sworn I saw the list in the to field but I must be mistaken. Here it is for the list. On Tue, 7 Aug 2007 08:02:42 -0700 stan <stanl(a)cox.net> wrote: > On Tue, 07 Aug 2007 15:38:40 +0200 > Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Mon, 6 Aug 2007 09:22:01 -0700, > > stan wrote: > > > > > > On Mon, 6 Aug 2007 09:04:57 -0700 > > > stan <stanl(a)cox.net> wrote: > > > > > > > I downloaded the hg alsa-lib snapshot of July 14, 2007, compiled > > > > it, (on Fedora 7 you will have to make a link to compile, > > > > cd /usr/include > > > > ln -s python2.5 python) > > > > and linked my program against it. It works again so the issue > > > > has been fixed in the latest version of alsa lib. Any estimate > > > > of when this version will be released for public consumption? > > > > My app is dead in the water with the current version of alsa, so > > > > anyone trying to use it will have a terrible user > > > > experience. :-( I want to give some indication of when people > > > > can expect it to start working again. In the meantime I'll > > > > create instructions on how to install the hg version to > > > > usr/local and change the library linking. > > > > > > > > Thanks. > > > > > > > For anyone interested, these are the formats that fail in 1.0.14 > > > final > > > > > > Value of last format 43 > > > test of sample format 14 failed (Invalid argument) > > > test of sample format 15 failed (Invalid argument) > > > test of sample format 16 failed (Invalid argument) > > > test of sample format 17 failed (Invalid argument) > > > test of sample format 18 failed (Invalid argument) > > > test of sample format 19 failed (Invalid argument) > > > test of sample format 23 failed (Invalid argument) > > > test of sample format 24 failed (Invalid argument) > > > test of sample format 25 failed (Invalid argument) > > > test of sample format 26 failed (Invalid argument) > > > test of sample format 27 failed (Invalid argument) > > > test of sample format 28 failed (Invalid argument) > > > test of sample format 29 failed (Invalid argument) > > > test of sample format 30 failed (Invalid argument) > > > test of sample format 31 failed (Invalid argument) > > > Format returned (0) > > > > > > > > > and these are the formats that fail in the hg version. > > > > > > Value of last format 43 > > > test of sample format 18 failed (Invalid argument) > > > test of sample format 19 failed (Invalid argument) > > > test of sample format 23 failed (Invalid argument) > > > test of sample format 24 failed (Invalid argument) > > > test of sample format 25 failed (Invalid argument) > > > test of sample format 26 failed (Invalid argument) > > > test of sample format 27 failed (Invalid argument) > > > test of sample format 28 failed (Invalid argument) > > > test of sample format 29 failed (Invalid argument) > > > test of sample format 30 failed (Invalid argument) > > > test of sample format 31 failed (Invalid argument) > > > Format returned (16) > > > > Could you give me the simple test case for checking the same thing > > on my system? > > > > > > thanks, > > > > Takashi > > They are attached as file test_case.c and shell script mktest. > > Running ./mktest should generate an executable named test_case. > Running that should generate the above output. > >

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[alsa-devel] snd_pcm_hw_params_set_format fails with invalid argument
by stan 07 Aug '07

07 Aug '07

Fedora 7 upgraded alsa to 1.0.14 from RC3. I was going to test the default device sound for my prior problem (as requested by Takashi weeks ago). Unfortunately when I try to run my app it won't start. The snd_pcm_hw_params_set_format fails when I try to use it as follows. err = snd_pcm_hw_params_set_access (alsa_dev, hw_params, ND_PCM_ACCESS_RW_INTERLEAVED); if (err < 0) { fprintf (stderr, "cannot set access type (%s)\n", snd_strerror (err)) ; goto catch_error ; } ; err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_FLOAT64_LE); //err = snd_pcm_hw_params_set_format (alsa_dev, hw_params, SND_PCM_FORMAT_S32_LE); if (err < 0) { fprintf (stderr, "cannot set sample format (%s)\n", snd_strerror (err)) ; goto catch_error ; } ; cannot set sample format (Invalid argument) It compiles cleanly. When I replace the SND_PCM_FORMAT_FLOAT64_LE with the S32_LE, it sets it, but of course the sound is garbage as it is still receiving doubles. This code worked fine under RC3. Is there some change that occurred in the step from 1.0.14.rc3 to 1.0.14 that would explain the above and suggest a fix? Thank you. $ uname -r 2.6.22.1-41.fc7 $ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). $ ls /proc/asound/card0 ice1724 id oss_mixer pcm0c pcm0p pcm1p pcm2p

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[alsa-devel] HDA: Add support for Acer Aspire laptops.
by Tobin Davis 07 Aug '07

07 Aug '07

This patch adds support for some Acer Aspire systems. Signed off by Tobin Davis <tdavis(a)dsl-only.net> I was actually sent this patch by someone on IRC. They approved for it to be submitted, and I don't see anything in the code to cause it to be kept out.

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[alsa-devel] HDA: Add more Dell systems
by Tobin Davis 07 Aug '07

07 Aug '07

This patch adds support for Dell E520 and a couple of other 965 based systems. Signed off by Tobin Davis <tdavis(a)dsl-only.net> On a side note, I start back to work tomorrow, getting paid to do these patches (well, sort of).

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[alsa-devel] Accessing alsa device in kernel space.
by Vasily Khoruzhick 07 Aug '07

07 Aug '07

Hi, how to access to alsa device from kernel space? I want to create driver for virtual soundcard, that will perform mixing and rate converting in kernelspace, and put result in real hardware. But I found no documentation how to open alsa device in kernel space :( Best regards, Vasily.

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[alsa-devel] Some functionality of ice1724 broken between 1.0.14.RC1 and 1.0.14.RC3
by stan 06 Aug '07

06 Aug '07

Hi, This is going to be fairly long because of the explanations needed, but there are three problems I've found on my Revolution 5.1 between Alsa 1.0.14.RC1 and 1.0.14.RC3. 1. Support for high sample rates 96000 and 192000 was lost. 2. Sound distortion at high sound frequencies was introduced. 3. The maximum buffer size seems too small for high sample rates (not related to release candidate). Overview On Fedora 6 the alsa version is 1.0.14.RC1. Using that version, my application can use the high frequencies and there is no distortion introduced at high frequencies. On Fedora 7 the alsa version is 1.0.14.RC3. I can't use the high frequencies on the Revolution 5.1 and there is distortion introduced into the sound at high frequencies. In investigating this I noticed that the maximum buffer sizes seem small. 1. Here is the output from my app on Fedora 6 at 192000. (1.0.14.RC1) This works great! Minimum channels (1) Maximum channels (10000) Minimum rate (4000) Direction = 0 Maximum rate (4294967295) Direction = -1 Minimum period_time (10) Direction = 1 Maximum period_time (2048000) Direction = 0 Minimum period_size (0) Direction = 1 Maximum period_size (4294967295) Direction = -1 Minimum periods (0) Direction = 1 Maximum periods (4294967295) Direction = 0 Minimum buffer_time (1) Direction = 0 Maximum buffer_time (4294967295) Direction = 0 Minimum buffer_size (1) Maximum buffer_size (4294967294) Minimum tick_time (1000) Direction = 0 Maximum tick_time (1000) Direction = 0 Actual rate (192000) Direction = 0 Actual channels (2) Actual period_size (8) Direction = 0 Actual buffer_size (8192) <--- notice that this is reasonable Here is the output from my app on Fedora 7 at 192000. (1.0.14.RC3) This aborts while setting the hardware parameters with invalid argument. Minimum channels (1) Maximum channels (10000) Minimum rate (4000) Direction = 0 Maximum rate (4294967295) Direction = -1 Rate (48000) Direction = -1 Result = 0 <-- from test rate Rate (96000) Direction = -1 Result = 0 Rate (192000) Direction = -1 Result = 0 <-- maximum for card hw Rate (384000) Direction = -1 Result = 0 <-- accepts invalid 384000 Minimum period_time (21333) Direction = 1 <-- seems strange the same Maximum period_time (21334) Direction = -1 Minimum period_size (85) Direction = 1 Maximum period_size (91628833) Direction = -1 Minimum periods (0) Direction = 1 Maximum periods (17247242) Direction = -1 Minimum buffer_time (1) Direction = 0 Maximum buffer_time (4294967295) Direction = 0 Minimum buffer_size (170) Maximum buffer_size (1466015503) Minimum tick_time (0) Direction = 0 Maximum tick_time (4294967295) Direction = 0 Actual rate (192000) Direction = 0 Actual channels (2) Actual buffer_time (341333) Direction = 1 Actual buffer_size (65536) Actual buffer_size (65536) Actual period_time (21333) Direction = 1 Actual period_size (807872295) Direction = 1 <--- Note invalid period size Actual periods (21333) Direction = 1 cannot set parameters (Invalid argument) Could not open the sound device The two are on the same hardware, just different OS versions. While investigating I switched from using size to time as the allocation mechanism without any effect (using the sample from pcm.c on alsa site). I did a diff on the ice1724.c driver (attached below) and noticed that there was a lot of cleanup done by making the structs and arrays of structs const. Could that possibly cause this problem by not allowing calculated values to be set? Don't know. 44100 and 48000 work. I installed the alsa-debuginfo package to try to get a handle on this in gdb. But with the extreme indirection in alsa in order to deal with all the options and the compiler optimization, gdb left my head spinning so I thought I would ask here before going any further. Can you think of an explanation for this behavior? Can you fix it? 2. I ran a frequency loudness test that is part of the app. It drops the frequency at 5 seconds intervals from 20 KHz to 5 Hz alternating. On Fedora 6 with 1.0.14.RC1 it works as expected. When above my hearing range I hear silence. When it gets to frequencies I can hear the sound is pure. On Fedora 7, this same app produces noise at frequencies I can't hear. This behavior is the same as the behavior I noticed with my emu10k1, ca0106 and CK8S sound cards on 1.0.14.RC1 as well as 1.0.14.RC3. I previously had ascribed this to bad/low quality sound chips; now I'm not so sure. In terms of sound quality the Rev 5.1 ranks well. See the link http://compreviews.about.com/od/multimedia/tp/SoundCards.htm Exact same hardware produces different sound. Can't explain it. Can you? Can you fix it? 3. While looking at the ice1724.c code I noticed that the maximum buffer size is 2**18. This seems small for an application today. I'm producing sound at 192000 frames per second (admittedly overkill though I like very smooth sound) using stereo doubles (16 bytes per frame). The maximum buffer size is only a fraction of my per second byte rate. Could you increase this? Discussion. The app is a binaural / chronaural beat generator and is on sourceforge at http://sourceforge.net/projects/discord/ so you can download and run the tests yourself (about a meg). You will need libsndfile and libsamplerate installed to compile it. The version there won't have the conversion to time based allocation as I won't release that until this issue is resolved. To change the rate you change the -r option in the script file high_noon.discord (-r 192000 or --rate=192000). The run command to use is "./discord high_noon.discord". For issue 2 you have to run "./discord frequency_loudness_test.discord". Again change the rate using the -r option (-r 192000 or -r 96000 fail). The open_alsa function starts at around line 5610 in discord.c if you want to play around with the initialization. I basically cloned it from the libsndfile samples. If you aren't familiar with binaural or chronaural beats you can search for "binaural beat" or "brain wave entrainment" and find more than you probably want to know. :-) Thanks for any help.

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Re: [alsa-devel] PROBLEM: snd-au8830: dead after software suspend
by Gert Robben 06 Aug '07

06 Aug '07

>>> Have you reported this to the ALSA people? >> No, I thought this might as well be something for PCI or PM people, and >> I expected some ALSA people are also reading this list. >> If not, where else should I report this exactly? > alsa-devel [...] the author of the driver Done (original message below). Thanks again :) Gert Robben wrote: > After a suspend-resume cycle (using Suspend2), my sound card doesn't > work anymore. It works again after reloading the module. > > dmesg, initial boot: >> Linux version 2.6.22-ck1 > ----8<---- >> PCI driver au8830 lacks driver specific resume support. > > dmesg, after resume, trying to use the card: >> vortex: ac97 codec stuck busy > > Thanks for any help! > > Gert Robben

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[alsa-devel] a newbie overwhelmed with this API
by Adam Seychell 06 Aug '07

06 Aug '07

I'm writing simple audio application and I have successfully used the DirectSound API under Windows. Now I'm looking at porting to Linux using ALSA. I have tried reading the online ALSA API documentation, but it seems too early in development and insufficient compared what one can learn reading Microsoft's DirectX SDK documentation. Excuse my ignorance for a minute, but the API seems quite large compared to DirectSound and so I'm wondering if there is good reason why so many functions exists ? Was there an early design decision not to write ALSA in C++ ? I would like to help out with documentation but of course I thats impossible for someone who is a complete novice to the ALSA project. I know that writing good documentation takes a lot of hard work, especially for software developers who spend enough time on coding. Its an unfortunate paradox among all open source projects. Software developers who posses the knowledge can't readily communicate it to potential volunteer documentation writers in an open source environment. Adam

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[alsa-devel] [PATCH] au88x0: mem leak fix in snd_vortex_create()
by Jesper Juhl 06 Aug '07

06 Aug '07

Hi, In sound/pci/au88x0/au88x0.c::snd_vortex_create() : The Coverity checker found that if we allocate storage for 'chip' but then leave via the regions_out: label, then we end up leaking the storage allocated for 'chip'. I believe simply freeing 'chip' before the "return err;" line is all we need to fix this, but please double-check me :) Signed-off-by: Jesper Juhl <jesper.juhl(a)gmail.com> --- diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 5ec1b6f..f70286a 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -232,6 +232,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) pci_disable_device(chip->pci_dev); //FIXME: this not the right place to unregister the gameport vortex_gameport_unregister(chip); + kfree(chip); return err; }

3 2

[alsa-devel] MPC5200 I2S driver for ASoC
by lokowich 05 Aug '07

05 Aug '07

I'm integrating a ALSA driver for an MPC5200 into the ASoC layer, similar to my successful effort for a PXA255 board. I'm modeling the DMA and ALSA interfaces per ac97mgt driver found in the Lite5200B BSP, rather than the BestComm API example in the 2.4 kernel. The speaker-test with 400Hz sine wave is working, but has a slight audible jitter. A wav file output using aplay produces white noise. Debugging shows I get two requests for buffer pointer per service of my transmit ISR (after the single call to snd_pcm_period_elapsed). I expect only one pointer request per interrupt. Could this be the noise problem? This email was Anti Virus checked by Astaro Security Gateway. http://www.astaro.com

3 7

[alsa-devel] [PATCH][RESEND] au88x0: mem leak fix in snd_vortex_create()
by Jesper Juhl 04 Aug '07

04 Aug '07

(resend of patch previously submitted on 04-Aug-2007 02:09) Hi, In sound/pci/au88x0/au88x0.c::snd_vortex_create() : The Coverity checker found that if we allocate storage for 'chip' but then leave via the regions_out: label, then we end up leaking the storage allocated for 'chip'. I believe simply freeing 'chip' before the "return err;" line is all we need to fix this, but please double-check me :) Signed-off-by: Jesper Juhl <jesper.juhl(a)gmail.com> Acked-by: Rene Herman <rene.herman(a)gmail.com> --- diff --git a/sound/pci/au88x0/au88x0.c b/sound/pci/au88x0/au88x0.c index 5ec1b6f..f70286a 100644 --- a/sound/pci/au88x0/au88x0.c +++ b/sound/pci/au88x0/au88x0.c @@ -232,6 +232,7 @@ snd_vortex_create(struct snd_card *card, struct pci_dev *pci, vortex_t ** rchip) pci_disable_device(chip->pci_dev); //FIXME: this not the right place to unregister the gameport vortex_gameport_unregister(chip); + kfree(chip); return err; }

1 0

[alsa-devel] SALSA and aplay
by Alan Horstmann 04 Aug '07

04 Aug '07

Quick questions: 1. Does SALSA provide everything needed by aplay? 2. Is there a version or equivalent of aplay that is a function that can be called from other code rather than the command line? Otherwise I shall hack it so for the purpose of some testing only, but ask to avoid re-inventing (no doubt poorly) something that exists. BTW SALSA compiled great (thanks Takashi), but with the following warning (which I ignore apart from mentioning here):- In file included from hcontrol.h:36, from mixer.h:25, from mixer.c:30: hctl_macros.h: In function `snd_hctl_elem_tlv_read': hctl_macros.h:136: warning: `snd_ctl_elem_tlv_read' is deprecated (declared at ctl_macros.h:1134) hctl_macros.h: In function `snd_hctl_elem_tlv_write': hctl_macros.h:142: warning: `snd_ctl_elem_tlv_write' is deprecated (declared at ctl_macros.h:1141) hctl_macros.h: In function `snd_hctl_elem_tlv_command': hctl_macros.h:148: warning: `snd_ctl_elem_tlv_command' is deprecated (declared at ctl_macros.h:1148) Alan

7 34

Re: [alsa-devel] trouble compiling alsa-drivers
by Wallace, Brooke 04 Aug '07

04 Aug '07

Package alsa-driver-1.0.14.tar.b2z also fails to compile with the same error. Is there a particualr version of the Kernel that is recommended to use? I don't think the issue is with including the macro itself, it is defined in adriver.h and including that file directly generates the same error. Seems that MARCO depends on another MACRO "RW_LOCK_UNLOCKED" and somewhere down the line there is a syntax error.... make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/memallo c.o CC [M] /local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/sgbuf.o CC [M] /local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/memory_ wrapper.o CC [M] /local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/pcm.o CC [M] /local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/pcm_nat ive.o /local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/pcm_nat ive.c:75: error: syntax error before "snd_pcm_link_rwlock" ... [Many more similar errors] ... make[4]: *** [/local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore/pcm_na tive.o] Error 1 make[3]: *** [/local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14/acore] Error 2 make[2]: *** [_module_/local/mnt/views/brookew/alsa/workspace/alsa-driver-1.0.14] Error 2 make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' make: *** [compile] Error 2 I will do some more digging around, but I don't want to go off the deep-end hacking software I don't understand, at least not yet. -Brooke ________________________________ From: Tobin Davis [mailto:tdavis@dsl-only.net] Sent: Thursday, August 02, 2007 6:21 PM To: Wallace, Brooke Subject: Re: [alsa-devel] trouble compiling alsa-drivers Alsa 1.0.14rc4 is NOT the latest. 1.0.14 went final a few weeks after, and there was a patch to fix build issues on Redhat's broken kernels that went into it. Tobin On Thu, 2007-08-02 at 16:53 -0700, Wallace, Brooke wrote: The alsa-drivers INSTALL file tells me: On 2.6 kernels, the build directory has to be given via --with-build=<kernel_build_dir> option additionaly, too. What exactly is mean by kernel_build_dir? I should not have to recompile my kernel because I already have the sourcecore module. Configure seems to work fine and locates my kernel source code: ./configure --with-cards=hda-intel --with-sequencer=yes But when I compile, it fails in acore/pcm_native.c line #75, acore/pcm_native.c line #75 uses the macro DEFINE_RWLOCK which apparently has not been defined yet. Searching thu the alsa-drivers source, I find that include/adriver.h defines this macro. What am I doing wrong? Why doesn't adriver.h get included when I compile acore/pcm_native.c? I'm using the latest release - alsa-driver-1.0.14rc4 My Kernel sources are: 2.6.9-34.EL-smp-i686 Alsa is already working fine on this system using the HDA Intel driver, its a RHEL- 3.4.5-2 system. I just want to re-compile the latest ALSA software and start writing a dummy user space driver that prints to a file or something. I should be able to cmopile all of the latest ALSA software... Any help is appreciated. -Brooke _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel -- Tobin Davis Laetrile is the pits.

1 0

Re: [alsa-devel] Will the real alsa wiki please stand up?
by Eliot Blennerhassett 03 Aug '07

03 Aug '07

I'm confused again. Which is the real official wiki? This one "AlsaWiki" http://bugtrack.alsa-project.org/main/index.php/Main_Page (MediaWiki, last change 25 July) OR This one "Official Alsa Wiki for Developers" https://bugtrack.alsa-project.org/wiki/wikka.php?wakka=AlsaWiki (Powered by Wikka Wakka Wiki) approx 28 content pages, last change 24 July) It was made clear a while ago that this is the UNofficial wiki: http://alsa.opensrc.org/index.php/Main_Page (MediaWiki) Gosh!, how much fragmentation do we need???? If 3 wikis really are necessary, can we at least have links to the others on the main page of each wiki, and explanation of which is most approriate for different content? thanks Eliot

5 9

[alsa-devel] User space drivers
by Wallace, Brooke 03 Aug '07

03 Aug '07

Hi, I've been trying to sort thru the ALSA docs and am starting to get the impression that all of the existing drivers are written as Kernel Modules. Doesn't this mean that they are running in Kernel space and infact are not User Space drivers? Shouldn't there be an ALSA Drivers API document that covers an API that User Space Drivers can use? I must be mssing something here... I'm looking into ALSA because my company wants to use this standard for writing User Space drivers for our product running on various linux kernels. The reasoning for this is strictly proprietary licensing - our leagl department is not comfortable with the GPL and the possiblitiy that something written for Kernel space could at some time be considered as required to be GPL - not my decision... -Brooke

4 3

[alsa-devel] Output to Input routing
by Vadim Lebedev 02 Aug '07

02 Aug '07

Hello, I remember reading AC97 specs that there si possibility to configure it in such a way, that when doing stereo capture, left input channel will get actual MIC input and right input channel will get the copy of speaker output. This setting is ideal for echo cancellation algorithm. To my grand dismay i'm unable to find a doc of ALSA mixer API so i'm unsucessuful writing code to put mixer in such mode. Can some kind soul please contibuite a code snippet.... Thanks Vadim

2 3

Re: [alsa-devel] Intel HDA Driver - Bug
by Takashi Iwai 02 Aug '07

02 Aug '07

At Tue, 31 Jul 2007 13:58:32 +0100, Steven King wrote: > > Hi Takashi, > > I still have not had much success getting the front microphone jack > working correctly. The internal array microphone works by setting the > "Digital Input Source" to "Digital Mic 1" which corresponds to the > chipset configuration loaded by BIOS and then by the driver in > patch_sigmatel.c. So, im not without a microphone for the time being. Aha, so it's indeed connected to a digital mic (0x17). Good to know. > I tried setting Port C,D,E,F (Port A and B are HP and internal > speaker) to be microphone inputs in the nid configuration, however it > still did not get any input audio. Would it have something to do with > the jack detect pin on the microphone jack? What else may I have a > look at? I guess 0x0b is the mic input (although it's labelled as "Line"). You might have "Line In as Output Switch". This will change the I/O direction of the pin. There have been bugs regarding this switch in the recent version and it got fixed after 1.0.14 release. So, please use the latest ALSA HG version for debugging the problems. thanks, Takashi

1 0

[alsa-devel] problem with ATI SB600 Azalia
by Rafał Zawadzki 02 Aug '07

02 Aug '07

Hello. I tried earlier on alsa-user, but nobody answer. Propably this is the better place for this post. I have the same problem as mentioned here: http://thread.gmane.org/gmane.linux.alsa.devel/47688/focus=47827 (the same laptop). I tried this solution, but unfortunately i had below error during kernel compilation (kernel source ubuntu 2.6.20-16). What is proper way to make upgrade alsa to alsa-hg? Building modules, stage 2. MODPOST 1855 modules WARNING: vmlinux - Section mismatch: reference to .init.data:boot_params from .text between '_text' (at offset 0xc0100036) and 'startup_32_smp' WARNING: vmlinux - Section mismatch: reference to .init.data:boot_params from .text between '_text' (at offset 0xc0100044) and 'startup_32_smp' WARNING: vmlinux - Section mismatch: reference to .init.data:init_pg_tables_end from .text between '_text' (at offset 0xc01000a6) and 'startup_32_smp' WARNING: vmlinux - Section mismatch: reference to .init.text:start_kernel from .text between 'is386' (at offset 0xc0100221) and 'check_x87' WARNING: vmlinux - Section mismatch: reference to .init.text:smp_prepare_cpus from .text between 'init' (at offset 0xc0100437) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:migration_init from .text between 'init' (at offset 0xc010043c) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:spawn_ksoftirqd from .text between 'init' (at offset 0xc0100441) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:spawn_softlockup_task from .text between 'init' (at offset 0xc0100446) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:smp_cpus_done from .text between 'init' (at offset 0xc01004c2) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:sched_init_smp from .text between 'init' (at offset 0xc01004c7) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:cpuset_init_smp from .text between 'init' (at offset 0xc01004cc) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:usermodehelper_init from .text between 'init' (at offset 0xc01004d6) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:driver_init from .text between 'init' (at offset 0xc01004db) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:sysctl_init from .text between 'init' (at offset 0xc01004e1) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.data: from .text between 'init' (at offset 0xc0100503) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.data: from .text between 'init' (at offset 0xc010051c) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:prepare_namespace from .text between 'init' (at offset 0xc01006ff) and 'rest_init' WARNING: vmlinux - Section mismatch: reference to .init.text:__alloc_bootmem from .text between 'init_gdt' (at offset 0xc010a0cb) and 'cpu_init' WARNING: vmlinux - Section mismatch: reference to .init.text:__alloc_bootmem from .text between 'init_gdt' (at offset 0xc010a0e1) and 'cpu_init' WARNING: vmlinux - Section mismatch: reference to .init.text:sysenter_setup from .text between 'identify_cpu' (at offset 0xc010a73b) and 'display_cacheinfo' WARNING: vmlinux - Section mismatch: reference to .init.text:mtrr_bp_init from .text between 'identify_cpu' (at offset 0xc010a745) and 'display_cacheinfo' WARNING: vmlinux - Section mismatch: reference to .init.text:trap_init_f00f_bug from .text between 'init_intel' (at offset 0xc010c796) and 'cpuid4_cache_lookup' WARNING: vmlinux - Section mismatch: reference to .init.data:initkmem_list3 from .text between 'set_up_list3s' (at offset 0xc017238f) and 's_start' WARNING: vmlinux - Section mismatch: reference to .init.text:eisa_root_register from .text between 'pci_eisa_init' (at offset 0xc026f71b) and 'virtual_eisa_release' WARNING: vmlinux - Section mismatch: reference to .init.text:eisa_root_register from .text between 'virtual_eisa_root_init' (at offset 0xc026f77f) and 'cpufreq_unregister_driver' WARNING: vmlinux - Section mismatch: reference to .init.text: from .text between 'iret_exc' (at offset 0xc02f2330) and '_etext' WARNING: vmlinux - Section mismatch: reference to .init.text:start_kernel from .paravirtprobe between '__start_paravirtprobe' (at offset 0xc03c4790) and '__stop_paravirtprobe' WARNING: "is_power_of_2" [sound/core/snd-rtctimer.ko] undefined! make[1]: *** [__modpost] Błąd 1 make: *** [modules] Błąd 2 root@antichrist:/usr/src/linux# Cheers, -- Rafał bluszcz Zawadzki http://my.jabberpl.org - use jabber http://glam.pl - try secondhand clothes http://bluszcz.net - my homepage

2 6

[alsa-devel] [RFC][PATCH] Get rid of dead code in sound/arm/sa11xx-uda1341.c
by Michal Piotrowski 02 Aug '07

02 Aug '07

Hi, File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 82 Unknown CONFIG option! CONFIG_H3600_HAL File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 103 Unknown CONFIG option! CONFIG_H3600_HAL File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 241 Unknown CONFIG option! CONFIG_H3600_HAL File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 310 Unknown CONFIG option! CONFIG_H3600_HAL File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 334 Unknown CONFIG option! CONFIG_H3600_HAL File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 344 Unknown CONFIG option! CONFIG_H3600_HAL File /home/devel/linux-rdc/sound/arm/sa11xx-uda1341.c line 357 Unknown CONFIG option! CONFIG_H3600_HAL Regards, Michal -- LOG http://www.stardust.webpages.pl/log/ Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-rdc-clean/sound/arm/sa11xx-uda1341.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-rdc/sound/arm/sa11xx-uda1341.c 2007-07-31 17:43:12.000000000 +0200 @@ -79,12 +79,6 @@ #include <asm/mach-types.h> #include <asm/dma.h> -#ifdef CONFIG_H3600_HAL -#include <asm/semaphore.h> -#include <asm/uaccess.h> -#include <asm/arch/h3600_hal.h> -#endif - #include <sound/core.h> #include <sound/pcm.h> #include <sound/initval.h> @@ -100,9 +94,6 @@ * We use DMA stuff from 2.4.18-rmk3-hh24 here to be able to compile this * module for Familiar 0.6.1 */ -#ifdef CONFIG_H3600_HAL -#define HH_VERSION 1 -#endif /* {{{ Type definitions */ @@ -238,11 +229,8 @@ static void sa11xx_uda1341_set_samplerat rate = 8000; /* Set the external clock generator */ -#ifdef CONFIG_H3600_HAL - h3600_audio_clock(rate); -#else + sa11xx_uda1341_set_audio_clock(rate); -#endif /* Select the clock divisor */ switch (rate) { @@ -307,13 +295,10 @@ static void sa11xx_uda1341_audio_init(st local_irq_restore(flags); /* Enable the audio power */ -#ifdef CONFIG_H3600_HAL - h3600_audio_power(AUDIO_RATE_DEFAULT); -#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_CODEC_NRESET); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif /* Wait for the UDA1341 to wake up */ mdelay(1); //FIXME - was removed by Perex - Why? @@ -331,21 +316,13 @@ static void sa11xx_uda1341_audio_init(st /* make the left and right channels unswapped (flip the WS latch) */ Ser4SSDR = 0; -#ifdef CONFIG_H3600_HAL - h3600_audio_mute(0); -#else - clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); } static void sa11xx_uda1341_audio_shutdown(struct sa11xx_uda1341 *sa11xx_uda1341) { /* mute on */ -#ifdef CONFIG_H3600_HAL - h3600_audio_mute(1); -#else set_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif /* disable the audio power and all signals leading to the audio chip */ l3_close(sa11xx_uda1341->uda1341); @@ -354,13 +331,9 @@ static void sa11xx_uda1341_audio_shutdow /* power off and mute off */ /* FIXME - is muting off necesary??? */ -#ifdef CONFIG_H3600_HAL - h3600_audio_power(0); - h3600_audio_mute(0); -#else + clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_AUDIO_ON); clr_sa11xx_uda1341_egpio(IPAQ_EGPIO_QMUTE); -#endif } /* }}} */

2 1

[alsa-devel] [RFC][PATCH] Coding style fix sound/pci/ca0106/ca_midi.h
by Michal Piotrowski 02 Aug '07

02 Aug '07

Hi, Coding style fix Regards, Michal -- LOG http://www.stardust.webpages.pl/log/ Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-mm-clean/sound/pci/ca0106/ca_midi.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-mm/sound/pci/ca0106/ca_midi.h 2007-08-01 16:38:54.000000000 +0200 @@ -22,9 +22,9 @@ * */ -#include<linux/spinlock.h> -#include<sound/rawmidi.h> -#include<sound/mpu401.h> +#include <linux/spinlock.h> +#include <sound/rawmidi.h> +#include <sound/mpu401.h> #define CA_MIDI_MODE_INPUT MPU401_MODE_INPUT #define CA_MIDI_MODE_OUTPUT MPU401_MODE_OUTPUT

2 1

[alsa-devel] [PATCH alsa-driver asihpi] Fix StreamGetInfo
by Eliot Blennerhassett 02 Aug '07

02 Aug '07

StreamGetInfo queried wrong stream type. (minor) correct case of Playback and Capture control names Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> --- diff -r a2861559e4ed pci/asihpi/asihpi.c --- a/pci/asihpi/asihpi.c Fri Jul 27 08:54:15 2007 +1200 +++ b/pci/asihpi/asihpi.c Thu Aug 02 10:25:24 2007 +1200 @@ -228,11 +228,11 @@ static inline u16 HPI_StreamGetInfoEx(HP u32 * pdwAuxiliaryData) { if (HPI_HandleObject(hStream) == HPI_OBJ_OSTREAM) - return HPI_InStreamGetInfoEx(hS, hStream, pwState, + return HPI_OutStreamGetInfoEx(hS, hStream, pwState, pdwBufferSize, pdwDataInBuffer, pdwSampleCount, pdwAuxiliaryData); else - return HPI_OutStreamGetInfoEx(hS, hStream, pwState, + return HPI_InStreamGetInfoEx(hS, hStream, pwState, pdwBufferSize, pdwDataInBuffer, pdwSampleCount, pdwAuxiliaryData); } @@ -1201,7 +1201,7 @@ struct hpi_control { #define ASIHPI_SOURCENODE_STRINGS \ { \ TEXT("no source"), \ - TEXT("PCM playback"), \ + TEXT("PCM Playback"), \ TEXT("Line in"), \ TEXT("Digital in"), \ TEXT("Tuner"), \ @@ -1230,7 +1230,7 @@ struct hpi_control { #define ASIHPI_DESTNODE_STRINGS \ { \ TEXT("no destination"), \ - TEXT("PCM capture"), \ + TEXT("PCM Capture"), \ TEXT("Line out"), \ TEXT("Digital out"), \ TEXT("RF"), \

2 1

[alsa-devel] problem while reading the data from device(phone)
by Susovan Ghosh 02 Aug '07

02 Aug '07

Hi All, We are using ALSA for a application.We have a phone and RJ11 phone adaptor-duel.I connect the phone to PC through the adaptor amd use the ALSA function snd_pcm_readi() to capture what ever spoke in the phone.Adapter is a USB device.we mention "plughw:0,0" as a capture device.When we run the program we get an error "device is unavailable ".what may be the problem? or de we required any thing more? Thanx, Susovan Ghosh Engineer (D&D ) PRDE Ph No:-998667320 Bangalore-560048

2 1

[alsa-devel] [PATCH] add the ESS1879 pnpbios ID to the es18xx driver
by Rene Herman 02 Aug '07

02 Aug '07

Hi Takashi. As reported by Troy Heidner, the "Gateway Solo 5150" laptop (for one) has an onboard ESS1879 that identifies itself through PNPBIOS as just that. He also confirmed that other than not knowing about it, snd-es18xx drives the chip fine, so this adds the ID to the driver. Signed-off-by: Rene Herman <rene.herman(a)gmail.com> diff --git a/sound/isa/es18xx.c b/sound/isa/es18xx.c index f7732bf..69b1c8a 100644 --- a/sound/isa/es18xx.c +++ b/sound/isa/es18xx.c @@ -2042,6 +2042,7 @@ static int pnpc_registered; static struct pnp_device_id snd_audiodrive_pnpbiosids[] = { { .id = "ESS1869" }, + { .id = "ESS1879" }, { .id = "" } /* end */ };

2 1

[alsa-devel] [PATCH] CS4270 driver does not compile with I2C disabled
by Timur Tabi 01 Aug '07

01 Aug '07

Fix compilation errors with the CS4270 when I2C is not enabled. Updated some comments to indicate that that stand-alone mode is not fully implemented, because there is no mechanism for the CS4270 driver and the machine driver to communicate the values of various input pins. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch applies on top of the initial CS4270 patch, labeled "[PATCH v3] ASoC CS4270 codec device driver". sound/soc/codecs/cs4270.c | 319 ++++++++++++++++++++++++--------------------- 1 files changed, 172 insertions(+), 147 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index b659a5d..8beae65 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -12,9 +12,11 @@ * * Current features/limitations: * - * 1) Stand-alone and software mode is supported. Stand-alone is - * automatically selected if I2C is disabled or if a CS4270 is not found - * on the I2C bus. + * 1) Software mode is supported. Stand-alone mode is automatically + * selected if I2C is disabled or if a CS4270 is not found on the I2C + * bus. However, stand-alone mode is only partially implemented because + * there is no mechanism yet for this driver and the machine driver to + * communicate the values of the M0, M1, MCLK1, and MCLK2 pins. * 2) Only I2C is supported, not SPI * 3) Only Master mode is supported, not Slave. * 4) The machine driver's 'startup' function must call @@ -34,12 +36,157 @@ #include "cs4270.h" +/* If I2C is defined, then we support software mode. However, if we're + not compiled as module but I2C is, then we can't use I2C calls. */ +#if defined(CONFIG_I2C) || (defined(CONFIG_I2C_MODULE) && defined(MODULE)) +#define USE_I2C +#endif + /* Private data for the CS4270 */ struct cs4270_private { unsigned int mclk; /* Input frequency of the MCLK pin */ unsigned int mode; /* The mode (I2S or left-justified) */ }; +/* The number of MCLK/LRCK ratios supported by the CS4270 */ +#define NUM_MCLK_RATIOS 9 + +/* The actual MCLK/LRCK ratios, in increasing numerical order */ +static unsigned int mclk_ratios[NUM_MCLK_RATIOS] = + {64, 96, 128, 192, 256, 384, 512, 768, 1024}; + +/* + * Sampling rate <-> bit patter mapping + * + * This array maps sampling rates to their SNDRV_PCM_RATE_x equivalent. + * + * This is really something that ALSA should provide. + * + * This table is used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + */ +static struct { + unsigned int rate; + unsigned int bit; +} rate_map[] = { + {5512, SNDRV_PCM_RATE_5512}, + {8000, SNDRV_PCM_RATE_8000}, + {11025, SNDRV_PCM_RATE_11025}, + {16000, SNDRV_PCM_RATE_16000}, + {22050, SNDRV_PCM_RATE_22050}, + {32000, SNDRV_PCM_RATE_32000}, + {44100, SNDRV_PCM_RATE_44100}, + {48000, SNDRV_PCM_RATE_48000}, + {64000, SNDRV_PCM_RATE_64000}, + {88200, SNDRV_PCM_RATE_88200}, + {96000, SNDRV_PCM_RATE_96000}, + {176400, SNDRV_PCM_RATE_176400}, + {192000, SNDRV_PCM_RATE_192000} +}; + +/* + * Determine the CS4270 samples rates. + * + * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * + * The value of MCLK is used to determine which sample rates are supported + * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine + * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * + * This function calculates the nine ratios and determines which ones match + * a standard sample rate. If there's a match, then it is added to the list + * of support sample rates. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + * + * Note that in stand-alone mode, the sample rate is determined by input + * pins M0, M1, MDIV1, and MDIV2. Also in stand-alone mode, divide-by-3 + * is not a programmable option. However, divide-by-3 is not an available + * option in stand-alone mode. This cases two problems: a ratio of 768 is + * not available (it requires divide-by-3) and B) ratios 192 and 384 can + * only be selected with divide-by-1.5, but there is an errate that make + * this selection difficult. + * + * In addition, there is no mechanism for communicating with the machine + * driver what the input settings can be. This would need to be implemented + * for stand-alone mode to work. + */ +static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + unsigned int i; + + cs4270->mclk = freq; + + for (i = 0; i < NUM_MCLK_RATIOS; i++) { + unsigned int rate; + unsigned int j; + rate = freq / mclk_ratios[i]; + for (j = 0; j < ARRAY_SIZE(rate_map); j++) { + if (rate == rate_map[j].rate) { + rates |= rate_map[j].bit; + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + } + } + + if (!rates) { + printk(KERN_ERR "cs4270: could not find a valid sample rate\n"); + return -EINVAL; + } + + codec_dai->playback.rates = rates; + codec_dai->playback.rate_min = rate_min; + codec_dai->playback.rate_max = rate_max; + + codec_dai->capture.rates = rates; + codec_dai->capture.rate_min = rate_min; + codec_dai->capture.rate_max = rate_max; + + return 0; +} + +/* + * Configure the codec for the selected audio format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + * + * Currently, this function only supports SND_SOC_DAIFMT_I2S and + * SND_SOC_DAIFMT_LEFT_J. The CS4270 codec also supports right-justified + * data for playback only, but ASoC currently does not support different + * formats for playback vs. record. + */ +static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + printk(KERN_ERR "cs4270: invalid DAI format\n"); + ret = -EINVAL; + } + + return ret; +} + /* * The codec isn't really big-endian or little-endian, since the I2S * interface requires data to be sent serially with the MSbit first. @@ -53,7 +200,7 @@ struct cs4270_private { SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -#ifdef CONFIG_I2C +#ifdef USE_I2C /* CS4270 registers addresses */ #define CS4270_CHIPID 0x01 /* Chip ID */ @@ -141,7 +288,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec) CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); if (length != CS4270_NUMREGS) { - printk(KERN_ERR "cs4270: I2C read failure, addr=%u\n", + printk(KERN_ERR "cs4270: I2C read failure, addr=0x%x\n", i2c_client->addr); return -EIO; } @@ -189,13 +336,14 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, cache[reg - CS4270_FIRSTREG] = value; return 0; } else { - printk(KERN_ERR "cs4270: I2C write failed\n"); + printk(KERN_ERR "cs4270: I2C write of register %u failed\n", + reg); return -EIO; } } /* - * Clock Ratio Selection for Master Mode. + * Clock Ratio Selection for Master Mode with I2C enabled * * The data for this chart is taken from Table 5 of the CS4270 reference * manual. @@ -204,10 +352,8 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling * rates the CS4270 currently supports. * - * 'ratio' is the MCLK/LRCK ratio. MCLK is usually a fixed input frequency, - * and LRCK is equal to the sampling rate. The CS4270 only supports sampling - * rates where this ratio is one of: 64, 96, 128, 192, 256, 384, 512, 768 or - * 1024. + * Each element in this array corresponds to the ratios in mclk_ratios[]. + * These two arrays need to be in sync. * * 'speed_mode' is the corresponding bit pattern to be written to the * MODE bits of the Mode Control Register @@ -227,21 +373,20 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, * never select any sample rates that require divide-by-1.5. */ static struct { - unsigned int ratio; u8 speed_mode; u8 mclk; -} cs4270_mode_ratios[] = { - {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +} cs4270_mode_ratios[NUM_MCLK_RATIOS] = { + {CS4270_MODE_4X, CS4270_MODE_DIV1}, /* 64 */ #ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA - {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, + {CS4270_MODE_4X, CS4270_MODE_DIV15}, /* 96 */ #endif - {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, - {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, - {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, - {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, - {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, - {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, - {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} + {CS4270_MODE_2X, CS4270_MODE_DIV1}, /* 128 */ + {CS4270_MODE_4X, CS4270_MODE_DIV3}, /* 192 */ + {CS4270_MODE_1X, CS4270_MODE_DIV1}, /* 256 */ + {CS4270_MODE_2X, CS4270_MODE_DIV3}, /* 384 */ + {CS4270_MODE_1X, CS4270_MODE_DIV2}, /* 512 */ + {CS4270_MODE_1X, CS4270_MODE_DIV3}, /* 768 */ + {CS4270_MODE_1X, CS4270_MODE_DIV4} /* 1024 */ }; /* @@ -270,12 +415,12 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); /* Sampling rate, in Hz */ ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ - for (i = 0; i < ARRAY_SIZE(cs4270_mode_ratios); i++) { - if (cs4270_mode_ratios[i].ratio == ratio) + for (i = 0; i < NUM_MCLK_RATIOS; i++) { + if (mclk_ratios[i] == ratio) break; } - if (i == ARRAY_SIZE(cs4270_mode_ratios)) { + if (i == NUM_MCLK_RATIOS) { /* We did not find a matching ratio */ printk(KERN_ERR "cs4270: could not find matching ratio\n"); return -EINVAL; @@ -377,126 +522,6 @@ static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) #endif -/* - * Sampling rate <-> bit patter mapping - * - * This array maps sampling rates to their SNDRV_PCM_RATE_x equivalent. - * - * This is really something that ALSA should provide. - * - * This table is used by cs4270_set_dai_sysclk() to tell ALSA which sampling - * rates the CS4270 currently supports. - */ -static struct { - unsigned int rate; - unsigned int bit; -} rate_map[] = { - {5512, SNDRV_PCM_RATE_5512}, - {8000, SNDRV_PCM_RATE_8000}, - {11025, SNDRV_PCM_RATE_11025}, - {16000, SNDRV_PCM_RATE_16000}, - {22050, SNDRV_PCM_RATE_22050}, - {32000, SNDRV_PCM_RATE_32000}, - {44100, SNDRV_PCM_RATE_44100}, - {48000, SNDRV_PCM_RATE_48000}, - {64000, SNDRV_PCM_RATE_64000}, - {88200, SNDRV_PCM_RATE_88200}, - {96000, SNDRV_PCM_RATE_96000}, - {176400, SNDRV_PCM_RATE_176400}, - {192000, SNDRV_PCM_RATE_192000} -}; - -/* - * Determine the CS4270 samples rates. - * - * 'freq' is the input frequency to MCLK. The other parameters are ignored. - * - * The value of MCLK is used to determine which sample rates are supported - * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine - * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. - * - * This function calculates the nine ratios and determines which ones match - * a standard sample rate. If there's a match, then it is added to the list - * of support sample rates. - * - * This function must be called by the machine driver's 'startup' function, - * otherwise the list of supported sample rates will not be available in - * time for ALSA. - */ -static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, - int clk_id, unsigned int freq, int dir) -{ - struct snd_soc_codec *codec = codec_dai->codec; - struct cs4270_private *cs4270 = codec->private_data; - unsigned int rates = 0; - unsigned int rate_min = -1; - unsigned int rate_max = 0; - unsigned int i; - - cs4270->mclk = freq; - - for (i = 0; i < ARRAY_SIZE(cs4270_mode_ratios); i++) { - unsigned int rate; - unsigned int j; - rate = freq / cs4270_mode_ratios[i].ratio; - for (j = 0; j < ARRAY_SIZE(rate_map); j++) { - if (rate == rate_map[j].rate) { - rates |= rate_map[j].bit; - if (rate < rate_min) - rate_min = rate; - if (rate > rate_max) - rate_max = rate; - } - } - } - - if (!rate_max) { - printk(KERN_ERR "cs4270: could not find a valid rate\n"); - return -EINVAL; - } - - codec_dai->playback.rates = rates; - codec_dai->playback.rate_min = rate_min; - codec_dai->playback.rate_max = rate_max; - - codec_dai->capture.rates = rates; - codec_dai->capture.rate_min = rate_min; - codec_dai->capture.rate_max = rate_max; - - return 0; -} - -/* - * Configure the codec for the selected audio format - * - * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the - * codec accordingly. - * - * Currently, this function only supports SND_SOC_DAIFMT_I2S and - * SND_SOC_DAIFMT_LEFT_J. The CS4270 codec also supports right-justified - * data for playback only, but ASoC currently does not support different - * formats for playback vs. record. - */ -static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, - unsigned int format) -{ - struct snd_soc_codec *codec = codec_dai->codec; - struct cs4270_private *cs4270 = codec->private_data; - int ret = 0; - - switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - case SND_SOC_DAIFMT_LEFT_J: - cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; - break; - default: - printk(KERN_ERR "cs4270: invalid DAI format\n"); - ret = -EINVAL; - } - - return ret; -} - static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); /* @@ -740,7 +765,7 @@ static int cs4270_probe(struct platform_device *pdev) return ret; } -#ifdef CONFIG_I2C +#ifdef USE_I2C cs4270_socdev = socdev; ret = i2c_add_driver(&cs4270_i2c_driver); @@ -780,7 +805,7 @@ static int cs4270_remove(struct platform_device *pdev) snd_soc_free_pcms(socdev); -#ifdef CONFIG_I2C +#ifdef USE_I2C if (socdev->codec->control_data) i2c_del_driver(&cs4270_i2c_driver); #endif -- 1.5.2.4

2 3

[alsa-devel] [patch] Add a software amplifier plugin
by Stas Sergeev 01 Aug '07

01 Aug '07

Hi. Most of the modern built-in cards do not have an internal amplifier. This makes it difficult to use them with those cheap small passive speakers that were in a heavy use 15 years ago when the cards had an amplifier builtin. I wrote a simple software amplifier plugin, it is attached. Of course you trade a sound quality, but whoever uses the passive speakers, have already made his choice anyway. Would it be possible to apply such a patch to the repository? Signed-off-by: Stas Sergeev <stsp(a)aknet.ru>

2 2

[alsa-devel] [RFC][PATCH] Get rid of dead code in sound/pci/ac97/ac97_patch.c
by Michal Piotrowski 01 Aug '07

01 Aug '07

Hi, File /home/devel/linux-rdc/sound/pci/ac97/ac97_patch.c line 484 Unknown CONFIG option! CONFIG_TOUCHSCREEN_WM9705 Regards, Michal -- LOG http://www.stardust.webpages.pl/log/ Signed-off-by: Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> --- linux-rdc-clean/sound/pci/ac97/ac97_patch.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-rdc/sound/pci/ac97/ac97_patch.c 2007-07-31 17:26:32.000000000 +0200 @@ -481,10 +481,7 @@ static int patch_wolfson05(struct snd_ac { /* WM9705, WM9710 */ ac97->build_ops = &patch_wolfson_wm9705_ops; -#ifdef CONFIG_TOUCHSCREEN_WM9705 - /* WM9705 touchscreen uses AUX and VIDEO for touch */ - ac97->flags |= AC97_HAS_NO_VIDEO | AC97_HAS_NO_AUX; -#endif + return 0; }

2 1

[alsa-devel] [PATCH v3] ASoC CS4270 codec device driver
by Timur Tabi 31 Jul '07

31 Jul '07

This patch adds ALSA SoC support for the Cirrus Logic CS4270 codec. The following features are suppored: 1) Stand-alone and software mode 2) Software mode via I2C only 3) Master mode, not Slave 4) No power management Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This code is based on git.kernel.org/pub/scm/linux/kernel/git/perex/alsa.git. include/linux/i2c-id.h | 1 + sound/soc/codecs/Kconfig | 20 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 808 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs4270.h | 28 ++ 5 files changed, 859 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs4270.c create mode 100644 sound/soc/codecs/cs4270.h diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index b690148..c4aadc6 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -119,6 +119,7 @@ #define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_LM4857 92 /* LM4857 Audio Amplifier */ +#define I2C_DRIVERID_CS4270 93 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e5fb437..7824880 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -17,3 +17,23 @@ config SND_SOC_WM8753 config SND_SOC_WM9712 tristate depends on SND_SOC + +# Cirrus Logic CS4270 Codec +config SND_SOC_CS4270 + tristate + depends on SND_SOC + +# Cirrus Logic CS4270 Codec Hardware Mute Support +# Select if you have external muting circuitry attached to your CS4270. +config SND_SOC_CS4270_HWMUTE + bool + depends on SND_SOC_CS4270 + +# Cirrus Logic CS4270 Codec VD = 3.3V Errata +# Select if you are affected by the errata where the part will not function +# if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will +# not select any sample rates that require MCLK to be divided by 1.5. +config SND_SOC_CS4270_VD33_ERRATA + bool + depends on SND_SOC_CS4270 + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index e39a747..7ad78e3 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,9 +3,11 @@ snd-soc-wm8731-objs := wm8731.o snd-soc-wm8750-objs := wm8750.o snd-soc-wm8753-objs := wm8753.o snd-soc-wm9712-objs := wm9712.o +snd-soc-cs4270-objs := cs4270.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_WM8731) += snd-soc-wm8731.o obj-$(CONFIG_SND_SOC_WM8750) += snd-soc-wm8750.o obj-$(CONFIG_SND_SOC_WM8753) += snd-soc-wm8753.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o +obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c new file mode 100644 index 0000000..b659a5d --- /dev/null +++ b/sound/soc/codecs/cs4270.c @@ -0,0 +1,808 @@ +/* + * CS4270 ALSA SoC (ASoC) codec driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + * + * This is an ASoC device driver for the Cirrus Logic CS4270 codec. + * + * Current features/limitations: + * + * 1) Stand-alone and software mode is supported. Stand-alone is + * automatically selected if I2C is disabled or if a CS4270 is not found + * on the I2C bus. + * 2) Only I2C is supported, not SPI + * 3) Only Master mode is supported, not Slave. + * 4) The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * 5) Only I2S and left-justified modes are supported + * 6) Power management is not supported + * 7) The only supported control is volume and hardware mute (if enabled) + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <linux/i2c.h> + +#include "cs4270.h" + +/* Private data for the CS4270 */ +struct cs4270_private { + unsigned int mclk; /* Input frequency of the MCLK pin */ + unsigned int mode; /* The mode (I2S or left-justified) */ +}; + +/* + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. + */ +#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#ifdef CONFIG_I2C + +/* CS4270 registers addresses */ +#define CS4270_CHIPID 0x01 /* Chip ID */ +#define CS4270_PWRCTL 0x02 /* Power Control */ +#define CS4270_MODE 0x03 /* Mode Control */ +#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ +#define CS4270_TRANS 0x05 /* Transition Control */ +#define CS4270_MUTE 0x06 /* Mute Control */ +#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ +#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ + +#define CS4270_FIRSTREG 0x01 +#define CS4270_LASTREG 0x08 +#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) + +/* Bit masks for the CS4270 registers */ +#define CS4270_CHIPID_ID 0xF0 +#define CS4270_CHIPID_REV 0x0F +#define CS4270_PWRCTL_FREEZE 0x80 +#define CS4270_PWRCTL_PDN_ADC 0x20 +#define CS4270_PWRCTL_PDN_DAC 0x02 +#define CS4270_PWRCTL_PDN 0x01 +#define CS4270_MODE_SPEED_MASK 0x30 +#define CS4270_MODE_1X 0x00 +#define CS4270_MODE_2X 0x10 +#define CS4270_MODE_4X 0x20 +#define CS4270_MODE_SLAVE 0x30 +#define CS4270_MODE_DIV_MASK 0x0E +#define CS4270_MODE_DIV1 0x00 +#define CS4270_MODE_DIV15 0x02 +#define CS4270_MODE_DIV2 0x04 +#define CS4270_MODE_DIV3 0x06 +#define CS4270_MODE_DIV4 0x08 +#define CS4270_MODE_POPGUARD 0x01 +#define CS4270_FORMAT_FREEZE_A 0x80 +#define CS4270_FORMAT_FREEZE_B 0x40 +#define CS4270_FORMAT_LOOPBACK 0x20 +#define CS4270_FORMAT_DAC_MASK 0x18 +#define CS4270_FORMAT_DAC_LJ 0x00 +#define CS4270_FORMAT_DAC_I2S 0x08 +#define CS4270_FORMAT_DAC_RJ16 0x18 +#define CS4270_FORMAT_DAC_RJ24 0x10 +#define CS4270_FORMAT_ADC_MASK 0x01 +#define CS4270_FORMAT_ADC_LJ 0x00 +#define CS4270_FORMAT_ADC_I2S 0x01 +#define CS4270_TRANS_ONE_VOL 0x80 +#define CS4270_TRANS_SOFT 0x40 +#define CS4270_TRANS_ZERO 0x20 +#define CS4270_TRANS_INV_ADC_A 0x08 +#define CS4270_TRANS_INV_ADC_B 0x10 +#define CS4270_TRANS_INV_DAC_A 0x02 +#define CS4270_TRANS_INV_DAC_B 0x04 +#define CS4270_TRANS_DEEMPH 0x01 +#define CS4270_MUTE_AUTO 0x20 +#define CS4270_MUTE_ADC_A 0x08 +#define CS4270_MUTE_ADC_B 0x10 +#define CS4270_MUTE_POLARITY 0x04 +#define CS4270_MUTE_DAC_A 0x01 +#define CS4270_MUTE_DAC_B 0x02 + +/* + * A list of addresses on which this CS4270 could use. I2C addresses are + * 7 bits. For the CS4270, the upper four bits are always 1001, and the + * lower three bits are determined via the AD2, AD1, and AD0 pins + * (respectively). + */ +static unsigned short normal_i2c[] = { + 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END +}; +I2C_CLIENT_INSMOD; + +/* + * Pre-fill the CS4270 register cache. + * + * We use the auto-increment feature of the CS4270 to read all registers in + * one shot. + */ +static int cs4270_fill_cache(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + struct i2c_client *i2c_client = codec->control_data; + s32 length; + + length = i2c_smbus_read_i2c_block_data(i2c_client, + CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); + + if (length != CS4270_NUMREGS) { + printk(KERN_ERR "cs4270: I2C read failure, addr=%u\n", + i2c_client->addr); + return -EIO; + } + + return 0; +} + +/* + * Read from the CS4270 register cache. + * + * This CS4270 registers are cached to avoid excessive I2C I/O operations. + * After the initial read to pre-fill the cache, the CS4270 never updates + * the register values, so we won't have a cache coherncy problem. + */ +static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + return -EIO; + + return cache[reg - CS4270_FIRSTREG]; +} + +/* + * Write to a CS4270 register via the I2C bus. + * + * This function writes the given value to the given CS4270 register, and + * also updates the register cache. + * + * Note that we don't use the hw_write function pointer of snd_soc_codec. + * That's because it's too clunky: the hw_write_t prototype does not match + * i2c_smbus_write_byte_data(), and it's just another layer of overhead. + */ +static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + return -EIO; + + if (i2c_smbus_write_byte_data(codec->control_data, reg, value) == 0) { + /* We've written to the hardware, so update the cache */ + u8 *cache = codec->reg_cache; + cache[reg - CS4270_FIRSTREG] = value; + return 0; + } else { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return -EIO; + } +} + +/* + * Clock Ratio Selection for Master Mode. + * + * The data for this chart is taken from Table 5 of the CS4270 reference + * manual. + * + * This table is used to determine how to program the Mode Control register. + * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + * + * 'ratio' is the MCLK/LRCK ratio. MCLK is usually a fixed input frequency, + * and LRCK is equal to the sampling rate. The CS4270 only supports sampling + * rates where this ratio is one of: 64, 96, 128, 192, 256, 384, 512, 768 or + * 1024. + * + * 'speed_mode' is the corresponding bit pattern to be written to the + * MODE bits of the Mode Control Register + * + * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * the Mode Control Register. + * + * In situations where a single ratio is represented by multiple speed + * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick + * double-speed instead of quad-speed. However, the CS4270 errata states + * that Divide-By-1.5 can cause failures, so we avoid that mode where + * possible. + * + * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not + * work if VD = 3.3V. If this effects you, select the + * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will + * never select any sample rates that require divide-by-1.5. + */ +static struct { + unsigned int ratio; + u8 speed_mode; + u8 mclk; +} cs4270_mode_ratios[] = { + {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA + {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, +#endif + {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, + {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, + {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, + {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, + {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, + {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, + {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} +}; + +/* + * Program the CS4270 with the given hardware parameters. + * + * The .dai_ops functions are used to provide board-specific data, like + * input frequencies, to this driver. This function takes that information, + * combines it with the hardware parameters provided, and programs the + * hardware accordingly. + */ +static int cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int ret = 0; + unsigned int i; + unsigned int rate; + unsigned int ratio; + int reg; + + /* Figure out which MCLK/LRCK ratio to use */ + + rate = params_rate(params); /* Sampling rate, in Hz */ + ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ + + for (i = 0; i < ARRAY_SIZE(cs4270_mode_ratios); i++) { + if (cs4270_mode_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs4270_mode_ratios)) { + /* We did not find a matching ratio */ + printk(KERN_ERR "cs4270: could not find matching ratio\n"); + return -EINVAL; + } + + /* Freeze and power-down the codec */ + + ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | + CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | + CS4270_PWRCTL_PDN); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Program the mode control register */ + + reg = snd_soc_read(codec, CS4270_MODE); + reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); + reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + + ret = snd_soc_write(codec, CS4270_MODE, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Program the format register */ + + reg = snd_soc_read(codec, CS4270_FORMAT); + reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); + + switch (cs4270->mode) { + case SND_SOC_DAIFMT_I2S: + reg |= CS4270_FORMAT_DAC_I2S | CS4270_FORMAT_ADC_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; + break; + default: + printk(KERN_ERR "cs4270: unknown format\n"); + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4270_FORMAT, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Disable auto-mute. This feature appears to be buggy, because in + some situations, auto-mute will not deactivate when it should. */ + + reg = snd_soc_read(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = snd_soc_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Thaw and power-up the codec */ + + ret = snd_soc_write(codec, CS4270_PWRCTL, 0); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + return ret; +} + +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + +/* + * Set the CS4270 external mute + * + * This function toggles the mute bits in the MUTE register. The CS4270's + * mute capability is intended for external muting circuitry, so if the + * board does not have the MUTEA or MUTEB pins connected to such circuitry, + * then this function will do nothing. + */ +static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg6; + + reg6 = snd_soc_read(codec, CS4270_MUTE); + + if (mute) + reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | + CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + else + reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | + CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + + return snd_soc_write(codec, CS4270_MUTE, reg6); +} + +#endif + +/* + * Sampling rate <-> bit patter mapping + * + * This array maps sampling rates to their SNDRV_PCM_RATE_x equivalent. + * + * This is really something that ALSA should provide. + * + * This table is used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + */ +static struct { + unsigned int rate; + unsigned int bit; +} rate_map[] = { + {5512, SNDRV_PCM_RATE_5512}, + {8000, SNDRV_PCM_RATE_8000}, + {11025, SNDRV_PCM_RATE_11025}, + {16000, SNDRV_PCM_RATE_16000}, + {22050, SNDRV_PCM_RATE_22050}, + {32000, SNDRV_PCM_RATE_32000}, + {44100, SNDRV_PCM_RATE_44100}, + {48000, SNDRV_PCM_RATE_48000}, + {64000, SNDRV_PCM_RATE_64000}, + {88200, SNDRV_PCM_RATE_88200}, + {96000, SNDRV_PCM_RATE_96000}, + {176400, SNDRV_PCM_RATE_176400}, + {192000, SNDRV_PCM_RATE_192000} +}; + +/* + * Determine the CS4270 samples rates. + * + * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * + * The value of MCLK is used to determine which sample rates are supported + * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine + * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * + * This function calculates the nine ratios and determines which ones match + * a standard sample rate. If there's a match, then it is added to the list + * of support sample rates. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + unsigned int i; + + cs4270->mclk = freq; + + for (i = 0; i < ARRAY_SIZE(cs4270_mode_ratios); i++) { + unsigned int rate; + unsigned int j; + rate = freq / cs4270_mode_ratios[i].ratio; + for (j = 0; j < ARRAY_SIZE(rate_map); j++) { + if (rate == rate_map[j].rate) { + rates |= rate_map[j].bit; + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + } + } + + if (!rate_max) { + printk(KERN_ERR "cs4270: could not find a valid rate\n"); + return -EINVAL; + } + + codec_dai->playback.rates = rates; + codec_dai->playback.rate_min = rate_min; + codec_dai->playback.rate_max = rate_max; + + codec_dai->capture.rates = rates; + codec_dai->capture.rate_min = rate_min; + codec_dai->capture.rate_max = rate_max; + + return 0; +} + +/* + * Configure the codec for the selected audio format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + * + * Currently, this function only supports SND_SOC_DAIFMT_I2S and + * SND_SOC_DAIFMT_LEFT_J. The CS4270 codec also supports right-justified + * data for playback only, but ASoC currently does not support different + * formats for playback vs. record. + */ +static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + printk(KERN_ERR "cs4270: invalid DAI format\n"); + ret = -EINVAL; + } + + return ret; +} + +static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); + +/* + * Notify the driver that a new I2C bus has been found. + * + * This function is called for each I2C bus in the system. The function + * then asks the I2C subsystem to probe that bus at the addresses on which + * our device (the CS4270) could exist. If a device is found at one of + * those addresses, then our probe function (cs4270_i2c_probe) is called. + */ +static int cs4270_i2c_attach(struct i2c_adapter *adapter) +{ + return i2c_probe(adapter, &addr_data, cs4270_i2c_probe); +} + +static int cs4270_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + + i2c_detach_client(client); + codec->control_data = NULL; + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + kfree(client); + return 0; +} + +/* A list of non-DAPM controls that the CS4270 supports */ +static const struct snd_kcontrol_new cs4270_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 0) +}; + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "CS4270 I2C", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_CS4270, + .attach_adapter = cs4270_i2c_attach, + .detach_client = cs4270_i2c_detach, +}; + +/* + * Global variable to store socdev for i2c probe function. + * + * If struct i2c_driver had a private_data field, we wouldn't need to use + * cs4270_socdec. This is the only way to pass the socdev structure to + * cs4270_i2c_probe(). + * + * The real solution to cs4270_socdev is to create a mechanism + * that maps I2C addresses to snd_soc_device structures. Perhaps the + * creation of the snd_soc_device object should be moved out of + * cs4270_probe() and into cs4270_i2c_probe(), but that would make this + * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby + * the chip is *not* connected to the I2C bus, but is instead configured via + * input pins. + */ +static struct snd_soc_device *cs4270_socdev; + +/* + * Initialize the I2C interface of the CS4270 + * + * This function is called for whenever the I2C subsystem finds a device + * at a particular address. + * + * Note: snd_soc_new_pcms() must be called before this function can be called, + * because of snd_ctl_add(). + */ +static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) +{ + struct snd_soc_device *socdev = cs4270_socdev; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c_client = NULL; + int i; + int ret = 0; + + /* Probing all possible addresses has one drawback: if there are + multiple CS4270s on the bus, then you cannot specify which + socdev is matched with which CS4270. For now, we just reject + this I2C device if the socdev already has one attached. */ + if (codec->control_data) + return -ENODEV; + + /* Note: codec_dai->codec is NULL here */ + + i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); + if (!i2c_client) { + printk(KERN_ERR "cs4270: could not allocate I2C client\n"); + return -ENOMEM; + } + + codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); + if (!codec->reg_cache) { + printk(KERN_ERR "cs4270: could not allocate register cache\n"); + ret = -ENOMEM; + goto error; + } + + i2c_set_clientdata(i2c_client, codec); + strcpy(i2c_client->name, "CS4270"); + + i2c_client->driver = &cs4270_i2c_driver; + i2c_client->adapter = adapter; + i2c_client->addr = addr; + + /* Verify that we have a CS4270 */ + + ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to read I2C\n"); + goto error; + } + /* The top four bits of the chip ID should be 1100. */ + if ((ret & 0xF0) != 0xC0) { + /* The device at this address is not a CS4270 codec */ + ret = -ENODEV; + goto error; + } + + printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr); + printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + + /* Tell the I2C layer a new client has arrived */ + + ret = i2c_attach_client(i2c_client); + if (ret) { + printk(KERN_ERR "cs4270: could not attach codec, " + "I2C address %x, error code %i\n", addr, ret); + goto error; + } + + codec->control_data = i2c_client; + codec->read = cs4270_read_reg_cache; + codec->write = cs4270_i2c_write; + codec->reg_cache_size = CS4270_NUMREGS; + + /* The I2C interface is set up, so pre-fill our register cache */ + + ret = cs4270_fill_cache(codec); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to fill register cache\n"); + goto error; + } + + /* Add the non-DAPM controls */ + + for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + + ret = snd_ctl_add(codec->card, kctrl); + if (ret < 0) + goto error; + } + + return 0; + +error: + if (codec->control_data) { + i2c_detach_client(i2c_client); + codec->control_data = NULL; + } + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + codec->reg_cache_size = 0; + + kfree(i2c_client); + + return ret; +} + +#endif + +struct snd_soc_codec_dai cs4270_dai = { + .name = "CS4270", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .dai_ops = { + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(cs4270_dai); + +/* + * ASoC probe function + * + * This function is called when the machine driver calls + * platform_device_add(). + */ +static int cs4270_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "CS4270 ALSA SoC Codec\n"); + + /* Allocate enough space for the snd_soc_codec structure + and our private data together. */ + codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) + + sizeof(struct cs4270_private), GFP_KERNEL); + if (!codec) { + printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + return -ENOMEM; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CS4270"; + codec->owner = THIS_MODULE; + codec->dai = &cs4270_dai; + codec->num_dai = 1; + codec->private_data = codec + ALIGN(sizeof(struct snd_soc_codec), 4); + + socdev->codec = codec; + + /* Register PCMs */ + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + return ret; + } + +#ifdef CONFIG_I2C + cs4270_socdev = socdev; + + ret = i2c_add_driver(&cs4270_i2c_driver); + if (ret) { + printk(KERN_ERR "cs4270: failed to attach driver"); + snd_soc_free_pcms(socdev); + return ret; + } + + /* Did we find a CS4270 on the I2C bus? */ + if (codec->control_data) { + /* Initialize codec ops */ + cs4270_dai.ops.hw_params = cs4270_hw_params; +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + cs4270_dai.dai_ops.digital_mute = cs4270_mute; +#endif + } else + printk(KERN_INFO "cs4270: no I2C device found, " + "using stand-alone mode\n"); +#else + printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); +#endif + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return ret; +} + +static int cs4270_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + +#ifdef CONFIG_I2C + if (socdev->codec->control_data) + i2c_del_driver(&cs4270_i2c_driver); +#endif + + kfree(socdev->codec); + socdev->codec = NULL; + + return 0; +} + +/* + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +struct snd_soc_codec_device soc_codec_device_cs4270 = { + .probe = cs4270_probe, + .remove = cs4270_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h new file mode 100644 index 0000000..0ced49b --- /dev/null +++ b/sound/soc/codecs/cs4270.h @@ -0,0 +1,28 @@ +/* + * Cirrus Logic CS4270 ALSA SoC Codec Driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#ifndef _CS4270_H +#define _CS4270_H + +/* + * The ASoC codec DAI structure for the CS4270. Assign this structure to + * the .codec_dai field of your machine driver's snd_soc_dai_link structure. + */ +extern struct snd_soc_codec_dai cs4270_dai; + +/* + * The ASoC codec device structure for the CS4270. Assign this structure + * to the .codec_dev field of your machine driver's snd_soc_device + * structure. + */ +extern struct snd_soc_codec_device soc_codec_device_cs4270; + +#endif -- 1.5.2.4

3 7

[alsa-devel] No sound card found for ATI SB600 Azalia
by Marek Zelem 31 Jul '07

31 Jul '07

Hello I have Dell Latitude D531 notebook with 00:14.2 Audio device: ATI Technologies Inc SB600 Azalia sound card. Module snd_hda_intel is automaticaly loaded but no sound card is detected. Here is the dmesg output: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 16 ALSA /usr/src/modules/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:558: hda_intel: azx_get_response timeout, switching to polling mode... ALSA /usr/src/modules/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:564: hda_intel: azx_get_response timeout, switching to single_cmd mode... hda_codec: No auto-config is available, default to model=ref ALSA /usr/src/modules/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1020: hda-intel: no codecs initialized ACPI: PCI interrupt for device 0000:00:14.2 disabled I have linux kernel 2.6.22.1. Also tried alsa 1.0.14 kernel modules built from alsa sources. The result was the same. I have also tried all model=xxx arguments, nothing helped. A also tried pci=noacpi but that caused boot failure. Am I doing something wrong? Can you please help me with that? Btw, the system is 64bit, could that be a problem? lspci -nnv for sound card is: 00:14.2 Audio device [0403]: ATI Technologies Inc SB600 Azalia [1002:4383] Subsystem: Dell Unknown device [1028:0206] Flags: slow devsel, IRQ 16 Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Thanks in advance Marek

3 5

[alsa-devel] [PATCH v2] ASoC CS4270 codec device driver
by Timur Tabi 31 Jul '07

31 Jul '07

This patch adds ALSA SoC support for the Cirrus Logic CS4270 codec. The following features are suppored: 1) Stand-alone and software mode 2) Software mode via I2C only 3) Master mode, not Slave 4) No power management Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This code is based on the Liam Girdwood's ASoC tree. Please note that I was unable to clone Liam's ASoC git repositories because the http interface does not work for me, and my corporate firewall blocks the git protocol. I had to download a snapshot and create a bare git repository from it. Therefore, git may complain when trying to apply this patch. include/linux/i2c-id.h | 1 + sound/soc/codecs/Kconfig | 20 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 812 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs4270.h | 28 ++ 5 files changed, 863 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs4270.c create mode 100644 sound/soc/codecs/cs4270.h diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index 0051203..6b3d207 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -119,6 +119,7 @@ #define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_WM8951 92 /* Wolfson WM8753 audio codec */ +#define I2C_DRIVERID_CS4270 93 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e71667f..9c84eb2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,4 +78,24 @@ config SND_SOC_WM9713 tristate depends on SND_SOC +# Cirrus Logic CS4270 Codec +config SND_SOC_CS4270 + tristate + depends on SND_SOC + +# Cirrus Logic CS4270 Codec Hardware Mute Support +# Select if you have external muting circuitry attached to your CS4270. +config SND_SOC_CS4270_HWMUTE + bool + depends on SND_SOC_CS4270 + +# Cirrus Logic CS4270 Codec VD = 3.3V Errata +# Select if you are affected by the errata where the part will not function +# if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will +# not select any sample rates that require MCLK to be divided by 1.5. +config SND_SOC_CS4270_VD33_ERRATA + bool + depends on SND_SOC_CS4270 + + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 07d0b59..d9237a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-wm8976-objs := wm8976.o snd-soc-wm8980-objs := wm8980.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +snd-soc-cs4270-objs := cs4270.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_SOC_WM8976) += snd-soc-wm8976.o obj-$(CONFIG_SND_SOC_WM8980) += snd-soc-wm8980.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o +obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c new file mode 100644 index 0000000..e1cf104 --- /dev/null +++ b/sound/soc/codecs/cs4270.c @@ -0,0 +1,812 @@ +/** + * CS4270 ALSA SoC (ASoC) codec driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + * + * This is an ASoC device driver for the Cirrus Logic CS4270 codec. + * + * Current features/limitations: + * + * 1) Stand-alone and software mode is supported. Stand-alone is + * automatically selected if I2C is disabled or if a CS4270 is not found + * on the I2C bus. + * 2) Only I2C is supported, not SPI + * 3) Only Master mode is supported, not Slave. + * 4) The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * 5) Only I2S and left-justified modes are supported + * 6) Power management is not supported + * 7) The only supported control is volume and hardware mute (if enabled) + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <linux/i2c.h> + +#include "cs4270.h" + +/* Private data for the CS4270 */ +struct cs4270_private { + unsigned int mclk; /* Input frequency of the MCLK pin */ + unsigned int mode; /* The mode (I2S or left-justified) */ +}; + +/** + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. Also, it appears that the + * CS4270 doesn't like S8. + * + * FIXME: This list should contain only signed values, not unsigned. The I2S + * protocol requires signed samples. + */ +#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#ifdef CONFIG_I2C + +/* CS4270 registers addresses */ +#define CS4270_CHIPID 0x01 /* Chip ID */ +#define CS4270_PWRCTL 0x02 /* Power Control */ +#define CS4270_MODE 0x03 /* Mode Control */ +#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ +#define CS4270_TRANS 0x05 /* Transition Control */ +#define CS4270_MUTE 0x06 /* Mute Control */ +#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ +#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ + +#define CS4270_FIRSTREG 0x01 +#define CS4270_LASTREG 0x08 +#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) + +/* Bit masks for the CS4270 registers */ +#define CS4270_CHIPID_ID 0xF0 +#define CS4270_CHIPID_REV 0x0F +#define CS4270_PWRCTL_FREEZE 0x80 +#define CS4270_PWRCTL_PDN_ADC 0x20 +#define CS4270_PWRCTL_PDN_DAC 0x02 +#define CS4270_PWRCTL_PDN 0x01 +#define CS4270_MODE_SPEED_MASK 0x30 +#define CS4270_MODE_1X 0x00 +#define CS4270_MODE_2X 0x10 +#define CS4270_MODE_4X 0x20 +#define CS4270_MODE_SLAVE 0x30 +#define CS4270_MODE_DIV_MASK 0x0E +#define CS4270_MODE_DIV1 0x00 +#define CS4270_MODE_DIV15 0x02 +#define CS4270_MODE_DIV2 0x04 +#define CS4270_MODE_DIV3 0x06 +#define CS4270_MODE_DIV4 0x08 +#define CS4270_MODE_POPGUARD 0x01 +#define CS4270_FORMAT_FREEZE_A 0x80 +#define CS4270_FORMAT_FREEZE_B 0x40 +#define CS4270_FORMAT_LOOPBACK 0x20 +#define CS4270_FORMAT_DAC_MASK 0x18 +#define CS4270_FORMAT_DAC_LJ 0x00 +#define CS4270_FORMAT_DAC_I2S 0x08 +#define CS4270_FORMAT_DAC_RJ16 0x18 +#define CS4270_FORMAT_DAC_RJ24 0x10 +#define CS4270_FORMAT_ADC_MASK 0x01 +#define CS4270_FORMAT_ADC_LJ 0x00 +#define CS4270_FORMAT_ADC_I2S 0x01 +#define CS4270_TRANS_ONE_VOL 0x80 +#define CS4270_TRANS_SOFT 0x40 +#define CS4270_TRANS_ZERO 0x20 +#define CS4270_TRANS_INV_ADC_A 0x08 +#define CS4270_TRANS_INV_ADC_B 0x10 +#define CS4270_TRANS_INV_DAC_A 0x02 +#define CS4270_TRANS_INV_DAC_B 0x04 +#define CS4270_TRANS_DEEMPH 0x01 +#define CS4270_MUTE_AUTO 0x20 +#define CS4270_MUTE_ADC_A 0x08 +#define CS4270_MUTE_ADC_B 0x10 +#define CS4270_MUTE_POLARITY 0x04 +#define CS4270_MUTE_DAC_A 0x01 +#define CS4270_MUTE_DAC_B 0x02 + +/** + * A list of addresses on which this CS4270 could use. I2C addresses are + * 7 bits. For the CS4270, the upper four bits are always 1001, and the + * lower three bits are determined via the AD2, AD1, and AD0 pins + * (respectively). + */ +static unsigned short normal_i2c[] = { + 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END +}; +I2C_CLIENT_INSMOD; + +/** + * Pre-fill the CS4270 register cache. + * + * We use the auto-increment feature of the CS4270 to read all registers in + * one shot. + */ +static int cs4270_fill_cache(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + struct i2c_client *i2c_client = codec->control_data; + s32 length; + + length = i2c_smbus_read_i2c_block_data(i2c_client, + CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); + + if (length != CS4270_NUMREGS) { + printk(KERN_ERR "cs4270: I2C read failure, addr=%u\n", + i2c_client->addr); + return -EIO; + } + + return 0; +} + +/** + * Read from the CS4270 register cache. + * + * This CS4270 registers are cached to avoid excessive I2C I/O operations. + * After the initial read to pre-fill the cache, the CS4270 never updates + * the register values, so we won't have a cache coherncy problem. + */ +static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + return -EIO; + + return cache[reg - CS4270_FIRSTREG]; +} + +/** + * Write to a CS4270 register via the I2C bus. + * + * This function writes the given value to the given CS4270 register, and + * also updates the register cache. + * + * Note that we don't use the hw_write function pointer of snd_soc_codec. + * That's because it's too clunky: the hw_write_t prototype does not match + * i2c_smbus_write_byte_data(), and it's just another layer of overhead. + */ +static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + return -EIO; + + if (i2c_smbus_write_byte_data(codec->control_data, reg, value) == 0) { + /* We've written to the hardware, so update the cache */ + u8 *cache = codec->reg_cache; + cache[reg - CS4270_FIRSTREG] = value; + return 0; + } else { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return -EIO; + } +} + +/** + * Clock Ratio Selection for Master Mode. + * + * The data for this chart is taken from Table 5 of the CS4270 reference + * manual. + * + * This table is used to determine how to program the Mode Control register. + * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + * + * 'ratio' is the MCLK/LRCK ratio. MCLK is usually a fixed input frequency, + * and LRCK is equal to the sampling rate. The CS4270 only supports sampling + * rates where this ratio is one of: 64, 96, 128, 192, 256, 384, 512, 768 or + * 1024. + * + * 'speed_mode' is the corresponding bit pattern to be written to the + * MODE bits of the Mode Control Register + * + * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * the Mode Control Register. + * + * In situations where a single ratio is represented by multiple speed + * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick + * double-speed instead of quad-speed. However, the CS4270 errata states + * that Divide-By-1.5 can cause failures, so we avoid that mode where + * possible. + * + * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not + * work if VD = 3.3V. If this effects you, select the + * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will + * never select any sample rates that require divide-by-1.5. + */ +static struct { + unsigned int ratio; + u8 speed_mode; + u8 mclk; +} cs4270_mode_ratios[] = { + {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA + {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, +#endif + {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, + {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, + {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, + {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, + {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, + {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, + {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} +}; + +/** + * Program the CS4270 with the given hardware parameters. + * + * The .dai_ops functions are used to provide board-specific data, like + * input frequencies, to this driver. This function takes that information, + * combines it with the hardware parameters provided, and programs the + * hardware accordingly. + */ +static int cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int ret = 0; + unsigned int i; + unsigned int rate; + unsigned int ratio; + int reg; + + /* Figure out which MCLK/LRCK ratio to use */ + + rate = params_rate(params); /* Sampling rate, in Hz */ + ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ + + for (i = 0; i < ARRAY_SIZE(cs4270_mode_ratios); i++) { + if (cs4270_mode_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs4270_mode_ratios)) { + /* We did not find a matching ratio */ + printk(KERN_ERR "cs4270: could not find matching ratio\n"); + return -EINVAL; + } + + /* Freeze and power-down the codec */ + + ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | + CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | + CS4270_PWRCTL_PDN); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Program the mode control register */ + + reg = snd_soc_read(codec, CS4270_MODE); + reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); + reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + + ret = snd_soc_write(codec, CS4270_MODE, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Program the format register */ + + reg = snd_soc_read(codec, CS4270_FORMAT); + reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); + + switch (cs4270->mode) { + case SND_SOC_DAIFMT_I2S: + reg |= CS4270_FORMAT_DAC_I2S | CS4270_FORMAT_ADC_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; + break; + default: + printk(KERN_ERR "cs4270: unknown format\n"); + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4270_FORMAT, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Disable auto-mute. This feature appears to be buggy, because in + some situations, auto-mute will not deactivate when it should. */ + + reg = snd_soc_read(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = snd_soc_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Thaw and power-up the codec */ + + ret = snd_soc_write(codec, CS4270_PWRCTL, 0); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + return ret; +} + +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + +/** + * Set the CS4270 external mute + * + * This function toggles the mute bits in the MUTE register. The CS4270's + * mute capability is intended for external muting circuitry, so if the + * board does not have the MUTEA or MUTEB pins connected to such circuitry, + * then this function will do nothing. + */ +static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg6; + + reg6 = snd_soc_read(codec, CS4270_MUTE); + + if (mute) + reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | + CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + else + reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | + CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + + return snd_soc_write(codec, CS4270_MUTE, reg6); +} + +#endif + +/** + * Sampling rate <-> bit patter mapping + * + * This array maps sampling rates to their SNDRV_PCM_RATE_x equivalent. + * + * This is really something that ALSA should provide. + * + * This table is used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + */ +static struct { + unsigned int rate; + unsigned int bit; +} rate_map[] = { + {5512, SNDRV_PCM_RATE_5512}, + {8000, SNDRV_PCM_RATE_8000}, + {11025, SNDRV_PCM_RATE_11025}, + {16000, SNDRV_PCM_RATE_16000}, + {22050, SNDRV_PCM_RATE_22050}, + {32000, SNDRV_PCM_RATE_32000}, + {44100, SNDRV_PCM_RATE_44100}, + {48000, SNDRV_PCM_RATE_48000}, + {64000, SNDRV_PCM_RATE_64000}, + {88200, SNDRV_PCM_RATE_88200}, + {96000, SNDRV_PCM_RATE_96000}, + {176400, SNDRV_PCM_RATE_176400}, + {192000, SNDRV_PCM_RATE_192000} +}; + +/** + * Determine the CS4270 samples rates. + * + * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * + * The value of MCLK is used to determine which sample rates are supported + * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine + * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * + * This function calculates the nine ratios and determines which ones match + * a standard sample rate. If there's a match, then it is added to the list + * of support sample rates. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int rates = 0; + unsigned int rate_min = -1; + unsigned int rate_max = 0; + unsigned int i; + + cs4270->mclk = freq; + + for (i = 0; i < ARRAY_SIZE(cs4270_mode_ratios); i++) { + unsigned int rate; + unsigned int j; + rate = freq / cs4270_mode_ratios[i].ratio; + for (j = 0; j < ARRAY_SIZE(rate_map); j++) { + if (rate == rate_map[j].rate) { + rates |= rate_map[j].bit; + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + } + } + + if (!rate_max) { + printk(KERN_ERR "cs4270: could not find a valid rate\n"); + return -EINVAL; + } + + codec_dai->playback.rates = rates; + codec_dai->playback.rate_min = rate_min; + codec_dai->playback.rate_max = rate_max; + + codec_dai->capture.rates = rates; + codec_dai->capture.rate_min = rate_min; + codec_dai->capture.rate_max = rate_max; + + return 0; +} + +/** + * Configure the codec for the selected audio format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + * + * Currently, this function only supports SND_SOC_DAIFMT_I2S and + * SND_SOC_DAIFMT_LEFT_J. The CS4270 codec also supports right-justified + * data for playback only, but ASoC currently does not support different + * formats for playback vs. record. + */ +static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + printk(KERN_ERR "cs4270: invalid DAI format\n"); + ret = -EINVAL; + } + + return ret; +} + +static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); + +/** + * Notify the driver that a new I2C bus has been found. + * + * This function is called for each I2C bus in the system. The function + * then asks the I2C subsystem to probe that bus at the addresses on which + * our device (the CS4270) could exist. If a device is found at one of + * those addresses, then our probe function (cs4270_i2c_probe) is called. + */ +static int cs4270_i2c_attach(struct i2c_adapter *adapter) +{ + return i2c_probe(adapter, &addr_data, cs4270_i2c_probe); +} + +static int cs4270_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec *codec = i2c_get_clientdata(client); + + i2c_detach_client(client); + codec->control_data = NULL; + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + kfree(client); + return 0; +} + +/* A list of non-DAPM controls that the CS4270 supports */ +static const struct snd_kcontrol_new cs4270_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 0) +}; + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "CS4270 I2C", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_CS4270, + .attach_adapter = cs4270_i2c_attach, + .detach_client = cs4270_i2c_detach, +}; + +/** + * Global variable to store socdev for i2c probe function. + * + * If struct i2c_driver had a private_data field, we wouldn't need to use + * cs4270_socdec. This is the only way to pass the socdev structure to + * cs4270_i2c_probe(). + * + * The real solution to cs4270_socdev is to create a mechanism + * that maps I2C addresses to snd_soc_device structures. Perhaps the + * creation of the snd_soc_device object should be moved out of + * cs4270_probe() and into cs4270_i2c_probe(), but that would make this + * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby + * the chip is *not* connected to the I2C bus, but is instead configured via + * input pins. + */ +static struct snd_soc_device *cs4270_socdev; + +/** + * Initialize the I2C interface of the CS4270 + * + * This function is called for whenever the I2C subsystem finds a device + * at a particular address. + * + * Note: snd_soc_new_pcms() must be called before this function can be called, + * because of snd_ctl_add(). + */ +static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) +{ + struct snd_soc_device *socdev = cs4270_socdev; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c_client = NULL; + int i; + int ret = 0; + + /* Probing all possible addresses has one drawback: if there are + multiple CS4270s on the bus, then you cannot specify which + socdev is matched with which CS4270. For now, we just reject + this I2C device if the socdev already has one attached. */ + if (codec->control_data) + return -ENODEV; + + /* Note: codec_dai->codec is NULL here */ + + i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); + if (!i2c_client) { + printk(KERN_ERR "cs4270: could not allocate I2C client\n"); + return -ENOMEM; + } + + codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); + if (!codec->reg_cache) { + printk(KERN_ERR "cs4270: could not allocate register cache\n"); + ret = -ENOMEM; + goto error; + } + + i2c_set_clientdata(i2c_client, codec); + strcpy(i2c_client->name, "CS4270"); + + i2c_client->driver = &cs4270_i2c_driver; + i2c_client->adapter = adapter; + i2c_client->addr = addr; + + /* Verify that we have a CS4270 */ + + ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to read I2C\n"); + goto error; + } + /* The top four bits of the chip ID should be 1100. */ + if ((ret & 0xF0) != 0xC0) { + /* The device at this address is not a CS4270 codec */ + ret = -ENODEV; + goto error; + } + + printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr); + printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + + /* Tell the I2C layer a new client has arrived */ + + ret = i2c_attach_client(i2c_client); + if (ret) { + printk(KERN_ERR "cs4270: could not attach codec, " + "I2C address %x, error code %i\n", addr, ret); + goto error; + } + + codec->control_data = i2c_client; + codec->read = cs4270_read_reg_cache; + codec->write = cs4270_i2c_write; + codec->reg_cache_size = CS4270_NUMREGS; + + /* The I2C interface is set up, so pre-fill our register cache */ + + ret = cs4270_fill_cache(codec); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to fill register cache\n"); + goto error; + } + + /* Add the non-DAPM controls */ + + for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + + ret = snd_ctl_add(codec->card, kctrl); + if (ret < 0) + goto error; + } + + return 0; + +error: + if (codec->control_data) { + i2c_detach_client(i2c_client); + codec->control_data = NULL; + } + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + codec->reg_cache_size = 0; + + kfree(i2c_client); + + return ret; +} + +#endif + +struct snd_soc_codec_dai cs4270_dai = { + .name = "CS4270", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .dai_ops = { + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(cs4270_dai); + +/** + * ASoC probe function + * + * This function is called when the machine driver calls + * platform_device_add(). + */ +static int cs4270_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "CS4270 ALSA SoC Codec\n"); + + /* Allocate enough space for the snd_soc_codec structure + and our private data together. */ + codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) + + sizeof(struct cs4270_private), GFP_KERNEL); + if (!codec) { + printk(KERN_ERR "cs4270: Could not allocate codec structure\n"); + return -ENOMEM; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CS4270"; + codec->owner = THIS_MODULE; + codec->dai = &cs4270_dai; + codec->num_dai = 1; + codec->private_data = codec + ALIGN(sizeof(struct snd_soc_codec), 4); + + socdev->codec = codec; + + /* Register PCMs */ + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + return ret; + } + +#ifdef CONFIG_I2C + cs4270_socdev = socdev; + + ret = i2c_add_driver(&cs4270_i2c_driver); + if (ret) { + printk(KERN_ERR "cs4270: failed to attach driver"); + snd_soc_free_pcms(socdev); + return ret; + } + + /* Did we find a CS4270 on the I2C bus? */ + if (codec->control_data) { + /* Initialize codec ops */ + cs4270_dai.ops.hw_params = cs4270_hw_params; +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + cs4270_dai.dai_ops.digital_mute = cs4270_mute; +#endif + } else + printk(KERN_INFO "cs4270: no I2C device found, " + "using stand-alone mode\n"); +#else + printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); +#endif + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return ret; +} + +static int cs4270_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + +#ifdef CONFIG_I2C + if (socdev->codec->control_data) + i2c_del_driver(&cs4270_i2c_driver); +#endif + + kfree(socdev->codec); + socdev->codec = NULL; + + return 0; +} + +/** + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +struct snd_soc_codec_device soc_codec_device_cs4270 = { + .probe = cs4270_probe, + .remove = cs4270_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h new file mode 100644 index 0000000..35e5fe3 --- /dev/null +++ b/sound/soc/codecs/cs4270.h @@ -0,0 +1,28 @@ +/* + * Cirrus Logic CS4270 ALSA SoC Codec Driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#ifndef _CS4270_H +#define _CS4270_H + +/** + * The ASoC codec DAI structure for the CS4270. Assign this structure to + * the .codec_dai field of your machine driver's snd_soc_dai_link structure. + */ +extern struct snd_soc_codec_dai cs4270_dai; + +/** + * The ASoC codec device structure for the CS4270. Assign this structure + * to the .codec_dev field of your machine driver's snd_soc_device + * structure. + */ +extern struct snd_soc_codec_device soc_codec_device_cs4270; + +#endif -- 1.5.2.4

2 2

[alsa-devel] Known STAC9200 input selection problem?
by Claudio Matsuoka 31 Jul '07

31 Jul '07

Hi, I've received the following bug report about problems in the internal mic of a Dell Latitude D820 (STAC9200). Is this a known problem? The used tested it with a fairly updated (but non-HG) driver. I've seen a similar problem in an HP dx2250 (ALC862) with an outdated ALSA lib, but this doesn't seem to be the case here. --- Excerpt of original report follows --- On laptops which use the snd-hda-intel driver the microphone often cannot be activated (in particular, when the system comes out of system installation). A workaround is the following script alway helps to make the microphone work (except after a wake-up from suspend-to-disk): #!/bin/sh /usr/sbin/alsactl restore amixer -c 0 sset 'Input Source',0 Line amixer -c 0 sset 'Input Source',0 Mic Comments 1. After a bootstrap, the "capture volumn" control of alsamixer always has the value 0, irrespective of what is stored in /etc/asound.state. The previously defined value of this control can be re-established with "alsctl restore", but that does not make the microphone work. 2. Also, when the volume settings are changed with any of the mixers, very often the microphone will not work any more; but I have not found a reproducable sequence that provokes the error this way. 3. The microphone can be made work if, afterwords, the sound-input switch is toggled to "line" and back to "mic".

2 1

[alsa-devel] [PATCH] ASoC CS4270 codec device driver
by Timur Tabi 30 Jul '07

30 Jul '07

This patch adds ALSA SoC support for the Cirrus Logic CS4270 codec. The following features are suppored: 1) Stand-alone and software mode 2) Software mode via I2C only 3) Master mode, not Slave 4) No power management Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This code is based on the Liam Girdwood's ASoC tree. Please note that I was unable to clone Liam's ASoC git repositories because the http interface does not work for me, and my corporate firewall blocks the git protocol. I had to download a snapshot and create a bare git repository from it. Therefore, git may complain when trying to apply this patch. include/linux/i2c-id.h | 1 + sound/soc/codecs/Kconfig | 20 ++ sound/soc/codecs/Makefile | 2 + sound/soc/codecs/cs4270.c | 803 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/cs4270.h | 28 ++ 5 files changed, 854 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/cs4270.c create mode 100644 sound/soc/codecs/cs4270.h diff --git a/include/linux/i2c-id.h b/include/linux/i2c-id.h index 0051203..6b3d207 100644 --- a/include/linux/i2c-id.h +++ b/include/linux/i2c-id.h @@ -119,6 +119,7 @@ #define I2C_DRIVERID_WM8750 90 /* Wolfson WM8750 audio codec */ #define I2C_DRIVERID_WM8753 91 /* Wolfson WM8753 audio codec */ #define I2C_DRIVERID_WM8951 92 /* Wolfson WM8753 audio codec */ +#define I2C_DRIVERID_CS4270 93 /* Cirrus Logic 4270 audio codec */ #define I2C_DRIVERID_I2CDEV 900 #define I2C_DRIVERID_ARP 902 /* SMBus ARP Client */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index e71667f..9c84eb2 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,4 +78,24 @@ config SND_SOC_WM9713 tristate depends on SND_SOC +# Cirrus Logic CS4270 Codec +config SND_SOC_CS4270 + tristate + depends on SND_SOC + +# Cirrus Logic CS4270 Codec Hardware Mute Support +# Select if you have external muting circuitry attached to your CS4270. +config SND_SOC_CS4270_HWMUTE + bool + depends on SND_SOC_CS4270 + +# Cirrus Logic CS4270 Codec VD = 3.3V Errata +# Select if you are affected by the errata where the part will not function +# if MCLK divide-by-1.5 is selected and VD is set to 3.3V. The driver will +# not select any sample rates that require MCLK to be divided by 1.5. +config SND_SOC_CS4270_VD33_ERRATA + bool + depends on SND_SOC_CS4270 + + diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 07d0b59..d9237a1 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -18,6 +18,7 @@ snd-soc-wm8976-objs := wm8976.o snd-soc-wm8980-objs := wm8980.o snd-soc-wm9712-objs := wm9712.o snd-soc-wm9713-objs := wm9713.o +snd-soc-cs4270-objs := cs4270.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o @@ -39,3 +40,4 @@ obj-$(CONFIG_SND_SOC_WM8976) += snd-soc-wm8976.o obj-$(CONFIG_SND_SOC_WM8980) += snd-soc-wm8980.o obj-$(CONFIG_SND_SOC_WM9712) += snd-soc-wm9712.o obj-$(CONFIG_SND_SOC_WM9713) += snd-soc-wm9713.o +obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c new file mode 100644 index 0000000..6719590 --- /dev/null +++ b/sound/soc/codecs/cs4270.c @@ -0,0 +1,803 @@ +/** + * CS4270 ALSA SoC (ASoC) codec driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + * + * This is an ASoC device driver for the Cirrus Logic CS4270 codec. + * + * Current features/limitations: + * + * 1) Stand-alone and software mode is supported. Stand-alone is + * automatically selected if I2C is disabled or if a CS4270 is not found + * on the I2C bus. + * 2) Only I2C is supported, not SPI + * 3) Only Master mode is supported, not Slave. + * 4) The machine driver's 'startup' function must call + * cs4270_set_dai_sysclk() with the value of MCLK. + * 5) Only I2S and left-justified modes are supported + * 6) Power management is not supported + * 7) The only supported control is volume and hardware mute (if enabled) + */ + +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/soc.h> +#include <sound/initval.h> +#include <linux/i2c.h> + +#include "cs4270.h" + +/* Private data for the CS4270 */ +struct cs4270_private { + unsigned int mclk; /* Input frequency of the MCLK pin */ + unsigned int mode; /* The mode (I2S or left-justified) */ +}; + +/** + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. Also, it appears that the + * CS4270 doesn't like S8. + * + * FIXME: This list should contain only signed values, not unsigned. The I2S + * protocol requires signed samples. + */ +#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#ifdef CONFIG_I2C + +/* CS4270 registers addresses */ +#define CS4270_CHIPID 0x01 /* Chip ID */ +#define CS4270_PWRCTL 0x02 /* Power Control */ +#define CS4270_MODE 0x03 /* Mode Control */ +#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ +#define CS4270_TRANS 0x05 /* Transition Control */ +#define CS4270_MUTE 0x06 /* Mute Control */ +#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ +#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ + +#define CS4270_FIRSTREG 0x01 +#define CS4270_LASTREG 0x08 +#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) + +/* Bit masks for the CS4270 registers */ +#define CS4270_CHIPID_ID 0xF0 +#define CS4270_CHIPID_REV 0x0F +#define CS4270_PWRCTL_FREEZE 0x80 +#define CS4270_PWRCTL_PDN_ADC 0x20 +#define CS4270_PWRCTL_PDN_DAC 0x02 +#define CS4270_PWRCTL_PDN 0x01 +#define CS4270_MODE_SPEED_MASK 0x30 +#define CS4270_MODE_1X 0x00 +#define CS4270_MODE_2X 0x10 +#define CS4270_MODE_4X 0x20 +#define CS4270_MODE_SLAVE 0x30 +#define CS4270_MODE_DIV_MASK 0x0E +#define CS4270_MODE_DIV1 0x00 +#define CS4270_MODE_DIV15 0x02 +#define CS4270_MODE_DIV2 0x04 +#define CS4270_MODE_DIV3 0x06 +#define CS4270_MODE_DIV4 0x08 +#define CS4270_MODE_POPGUARD 0x01 +#define CS4270_FORMAT_FREEZE_A 0x80 +#define CS4270_FORMAT_FREEZE_B 0x40 +#define CS4270_FORMAT_LOOPBACK 0x20 +#define CS4270_FORMAT_DAC_MASK 0x18 +#define CS4270_FORMAT_DAC_LJ 0x00 +#define CS4270_FORMAT_DAC_I2S 0x08 +#define CS4270_FORMAT_DAC_RJ16 0x18 +#define CS4270_FORMAT_DAC_RJ24 0x10 +#define CS4270_FORMAT_ADC_MASK 0x01 +#define CS4270_FORMAT_ADC_LJ 0x00 +#define CS4270_FORMAT_ADC_I2S 0x01 +#define CS4270_TRANS_ONE_VOL 0x80 +#define CS4270_TRANS_SOFT 0x40 +#define CS4270_TRANS_ZERO 0x20 +#define CS4270_TRANS_INV_ADC_A 0x08 +#define CS4270_TRANS_INV_ADC_B 0x10 +#define CS4270_TRANS_INV_DAC_A 0x02 +#define CS4270_TRANS_INV_DAC_B 0x04 +#define CS4270_TRANS_DEEMPH 0x01 +#define CS4270_MUTE_AUTO 0x20 +#define CS4270_MUTE_ADC_A 0x08 +#define CS4270_MUTE_ADC_B 0x10 +#define CS4270_MUTE_POLARITY 0x04 +#define CS4270_MUTE_DAC_A 0x01 +#define CS4270_MUTE_DAC_B 0x02 + +/** + * A list of addresses on which this CS4270 could use. I2C addresses are + * 7 bits. For the CS4270, the upper four bits are always 1001, and the + * lower three bits are determined via the AD2, AD1, and AD0 pins + * (respectively). + */ +static unsigned short normal_i2c[] = { + 0x48, 0x49, 0x4A, 0x4B, 0x4C, 0x4D, 0x4E, 0x4F, I2C_CLIENT_END +}; +I2C_CLIENT_INSMOD; + +/** + * Pre-fill the CS4270 register cache. + * + * We use the auto-increment feature of the CS4270 to read all registers in + * one shot. + */ +static int cs4270_fill_cache(struct snd_soc_codec *codec) +{ + u8 *cache = codec->reg_cache; + struct i2c_client *i2c_client = codec->control_data; + s32 length; + + length = i2c_smbus_read_i2c_block_data(i2c_client, + CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache); + + if (length != CS4270_NUMREGS) { + printk(KERN_ERR "cs4270: I2C read failure, addr=%u\n", + i2c_client->addr); + return -EIO; + } + + return 0; +} + +/** + * Read from the CS4270 register cache. + * + * This CS4270 registers are cached to avoid excessive I2C I/O operations. + * After the initial read to pre-fill the cache, the CS4270 never updates + * the register values, so we won't have a cache coherncy problem. + */ +static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, unsigned int reg) +{ + u8 *cache = codec->reg_cache; + + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + return -EIO; + + return cache[reg - CS4270_FIRSTREG]; +} + +/** + * Write to a CS4270 register via the I2C bus. + * + * This function writes the given value to the given CS4270 register, and + * also updates the register cache. + * + * Note that we don't use the hw_write function pointer of snd_soc_codec. + * That's because it's too clunky: the hw_write_t prototype does not match + * i2c_smbus_write_byte_data(), and it's just another layer of overhead. + */ +static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) + return -EIO; + + if (i2c_smbus_write_byte_data(codec->control_data, reg, value) == 0) { + /* We've written to the hardware, so update the cache */ + u8 *cache = codec->reg_cache; + cache[reg - CS4270_FIRSTREG] = value; + return 0; + } else { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return -EIO; + } +} + +/** + * Clock Ratio Selection for Master Mode. + * + * The data for this chart is taken from Table 5 of the CS4270 reference + * manual. + * + * This table is used to determine how to program the Mode Control register. + * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + * + * 'ratio' is the MCLK/LRCK ratio. MCLK is usually a fixed input frequency, + * and LRCK is equal to the sampling rate. The CS4270 only supports sampling + * rates where this ratio is one of: 64, 96, 128, 192, 256, 384, 512, 768 or + * 1024. + * + * 'speed_mode' is the corresponding bit pattern to be written to the + * MODE bits of the Mode Control Register + * + * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * the Mode Control Register. + * + * In situations where a single ratio is represented by multiple speed + * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick + * double-speed instead of quad-speed. However, the CS4270 errata states + * that Divide-By-1.5 can cause failures, so we avoid that mode where + * possible. + * + * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not + * work if VD = 3.3V. If this effects you, select the + * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will + * never select any sample rates that require divide-by-1.5. + */ +static struct { + unsigned int ratio; + u8 speed_mode; + u8 mclk; +} cs4270_mode_ratios[] = { + {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA + {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, +#endif + {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, + {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, + {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, + {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, + {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, + {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, + {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} +}; + +/** + * Program the CS4270 with the given hardware parameters. + * + * The .dai_ops functions are used to provide board-specific data, like + * input frequencies, to this driver. This function takes that information, + * combines it with the hardware parameters provided, and programs the + * hardware accordingly. + */ +static int cs4270_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int ret = 0, i, rate, ratio; + int reg; + + /* Figure out which MCLK/LRCK ratio to use */ + + rate = params_rate(params); /* Sampling rate, in Hz */ + ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ + + for (i=0; i<ARRAY_SIZE(cs4270_mode_ratios); i++) { + if (cs4270_mode_ratios[i].ratio == ratio) { + break; + } + } + + if (i == ARRAY_SIZE(cs4270_mode_ratios)) { + /* We did not find a matching ratio */ + printk(KERN_ERR "cs4270: could not find matching ratio\n"); + return -EINVAL; + } + + /* Freeze and power-down the codec */ + + ret = snd_soc_write(codec, CS4270_PWRCTL, CS4270_PWRCTL_FREEZE | + CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | + CS4270_PWRCTL_PDN); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Program the mode control register */ + + reg = snd_soc_read(codec, CS4270_MODE); + reg &= ~(CS4270_MODE_SPEED_MASK | CS4270_MODE_DIV_MASK); + reg |= cs4270_mode_ratios[i].speed_mode | cs4270_mode_ratios[i].mclk; + + ret = snd_soc_write(codec, CS4270_MODE, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Program the format register */ + + reg = snd_soc_read(codec, CS4270_FORMAT); + reg &= ~(CS4270_FORMAT_DAC_MASK | CS4270_FORMAT_ADC_MASK); + + switch (cs4270->mode) { + case SND_SOC_DAIFMT_I2S: + reg |= CS4270_FORMAT_DAC_I2S | CS4270_FORMAT_ADC_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + reg |= CS4270_FORMAT_DAC_LJ | CS4270_FORMAT_ADC_LJ; + break; + default: + printk(KERN_ERR "cs4270: unknown format\n"); + return -EINVAL; + } + + ret = snd_soc_write(codec, CS4270_FORMAT, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Disable auto-mute. This feature appears to be buggy, because in + some situations, auto-mute will not deactivate when it should. */ + + reg = snd_soc_read(codec, CS4270_MUTE); + reg &= ~CS4270_MUTE_AUTO; + ret = snd_soc_write(codec, CS4270_MUTE, reg); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + /* Thaw and power-up the codec */ + + ret = snd_soc_write(codec, CS4270_PWRCTL, 0); + if (ret < 0) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return ret; + } + + return ret; +} + +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + +/** + * Set the CS4270 external mute + * + * This function toggles the mute bits in the MUTE register. The CS4270's + * mute capability is intended for external muting circuitry, so if the + * board does not have the MUTEA or MUTEB pins connected to such circuitry, + * then this function will do nothing. + */ +static int cs4270_mute(struct snd_soc_codec_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + int reg6; + + reg6 = snd_soc_read(codec, CS4270_MUTE); + + if (mute) + reg6 |= CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | + CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B; + else + reg6 &= ~(CS4270_MUTE_ADC_A | CS4270_MUTE_ADC_B | + CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B); + + return snd_soc_write(codec, CS4270_MUTE, reg6); +} + +#endif + +/** + * Sampling rate <-> bit patter mapping + * + * This array maps sampling rates to their SNDRV_PCM_RATE_x equivalent. + * + * This is really something that ALSA should provide. + * + * This table is used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + */ +static struct { + unsigned int rate; + unsigned int bit; +} rate_map[] = { + {5512, SNDRV_PCM_RATE_5512}, + {8000, SNDRV_PCM_RATE_8000}, + {11025, SNDRV_PCM_RATE_11025}, + {16000, SNDRV_PCM_RATE_16000}, + {22050, SNDRV_PCM_RATE_22050}, + {32000, SNDRV_PCM_RATE_32000}, + {44100, SNDRV_PCM_RATE_44100}, + {48000, SNDRV_PCM_RATE_48000}, + {64000, SNDRV_PCM_RATE_64000}, + {88200, SNDRV_PCM_RATE_88200}, + {96000, SNDRV_PCM_RATE_96000}, + {176400, SNDRV_PCM_RATE_176400}, + {192000, SNDRV_PCM_RATE_192000} +}; + +/** + * Determine the CS4270 samples rates. + * + * 'freq' is the input frequency to MCLK. The other parameters are ignored. + * + * The value of MCLK is used to determine which sample rates are supported + * by the CS4270. The ratio of MCLK / Fs must be equal to one of nine + * support values: 64, 96, 128, 192, 256, 384, 512, 768, and 1024. + * + * This function calculates the nine ratios and determines which ones match + * a standard sample rate. If there's a match, then it is added to the list + * of support sample rates. + * + * This function must be called by the machine driver's 'startup' function, + * otherwise the list of supported sample rates will not be available in + * time for ALSA. + */ +static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + unsigned int rates = 0, rate_min = -1, rate_max = 0, i, j; + + cs4270->mclk = freq; + + for (i=0; i<ARRAY_SIZE(cs4270_mode_ratios); i++) { + unsigned int rate; + rate = freq / cs4270_mode_ratios[i].ratio; + for (j = 0; j<ARRAY_SIZE(rate_map); j++) { + if (rate == rate_map[j].rate) { + rates |= rate_map[j].bit; + if (rate < rate_min) + rate_min = rate; + if (rate > rate_max) + rate_max = rate; + } + } + } + + if (!rate_max) { + printk(KERN_ERR "cs4270: could not find a valid rate\n"); + return -EINVAL; + } + + codec_dai->playback.rates = rates; + codec_dai->playback.rate_min = rate_min; + codec_dai->playback.rate_max = rate_max; + + codec_dai->capture.rates = rates; + codec_dai->capture.rate_min = rate_min; + codec_dai->capture.rate_max = rate_max; + + return 0; +} + +/** + * Configure the codec for the selected audio format + * + * This function takes a bitmask of SND_SOC_DAIFMT_x bits and programs the + * codec accordingly. + * + * Currently, this function only supports SND_SOC_DAIFMT_I2S and + * SND_SOC_DAIFMT_LEFT_J. The CS4270 codec also supports right-justified + * data for playback only, but ASoC currently does not support different + * formats for playback vs. record. + */ +static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs4270_private *cs4270 = codec->private_data; + int ret = 0; + + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + case SND_SOC_DAIFMT_LEFT_J: + cs4270->mode = format & SND_SOC_DAIFMT_FORMAT_MASK; + break; + default: + printk(KERN_ERR "cs4270: invalid DAI format\n"); + ret = -EINVAL; + } + + return ret; +} + +static int cs4270_i2c_probe(struct i2c_adapter *adap, int addr, int kind); + +/** + * Notify the driver that a new I2C bus has been found. + * + * This function is called for each I2C bus in the system. The function + * then asks the I2C subsystem to probe that bus at the addresses on which + * our device (the CS4270) could exist. If a device is found at one of + * those addresses, then our probe function (cs4270_i2c_probe) is called. + */ +static int cs4270_i2c_attach(struct i2c_adapter *adapter) +{ + return i2c_probe(adapter, &addr_data, cs4270_i2c_probe); +} + +static int cs4270_i2c_detach(struct i2c_client *client) +{ + struct snd_soc_codec* codec = i2c_get_clientdata(client); + + i2c_detach_client(client); + codec->control_data = NULL; + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + + kfree(client); + return 0; +} + +/* A list of non-DAPM controls that the CS4270 supports */ +static const struct snd_kcontrol_new cs4270_snd_controls[] = { + SOC_DOUBLE_R("Master Playback Volume", CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 0) +}; + +static struct i2c_driver cs4270_i2c_driver = { + .driver = { + .name = "CS4270 I2C", + .owner = THIS_MODULE, + }, + .id = I2C_DRIVERID_CS4270, + .attach_adapter = cs4270_i2c_attach, + .detach_client = cs4270_i2c_detach, +}; + +/** + * Global variable to store socdev for i2c probe function. + * + * If struct i2c_driver had a private_data field, we wouldn't need to use + * cs4270_socdec. This is the only way to pass the socdev structure to + * cs4270_i2c_probe(). + * + * The real solution to cs4270_socdev is to create a mechanism + * that maps I2C addresses to snd_soc_device structures. Perhaps the + * creation of the snd_soc_device object should be moved out of + * cs4270_probe() and into cs4270_i2c_probe(), but that would make this + * driver dependent on I2C. The CS4270 supports "stand-alone" mode, whereby + * the chip is *not* connected to the I2C bus, but is instead configured via + * input pins. + */ +static struct snd_soc_device *cs4270_socdev; + +/** + * Initialize the I2C interface of the CS4270 + * + * This function is called for whenever the I2C subsystem finds a device + * at a particular address. + * + * Note: snd_soc_new_pcms() must be called before this function can be called, + * because of snd_ctl_add(). + */ +static int cs4270_i2c_probe(struct i2c_adapter *adapter, int addr, int kind) +{ + struct snd_soc_device *socdev = cs4270_socdev; + struct snd_soc_codec *codec = socdev->codec; + struct i2c_client *i2c_client = NULL; + int i, ret = 0; + + /* Probing all possible addresses has one drawback: if there are + multiple CS4270s on the bus, then you cannot specify which + socdev is matched with which CS4270. For now, we just reject + this I2C device if the socdev already has one attached. */ + if (codec->control_data) { + return -ENODEV; + } + + /* Note: codec_dai->codec is NULL here */ + + i2c_client = kzalloc(sizeof(struct i2c_client), GFP_KERNEL); + if (!i2c_client) { + printk(KERN_ERR "cs4270: could not allocate I2C client\n"); + return -ENOMEM; + } + + codec->reg_cache = kzalloc(CS4270_NUMREGS, GFP_KERNEL); + if (!codec->reg_cache) { + printk(KERN_ERR "cs4270: could not allocate register cache\n"); + ret = -ENOMEM; + goto error; + } + + i2c_set_clientdata(i2c_client, codec); + strcpy(i2c_client->name, "CS4270"); + + i2c_client->driver = &cs4270_i2c_driver; + i2c_client->adapter = adapter; + i2c_client->addr = addr; + + /* Verify that we have a CS4270 */ + + ret = i2c_smbus_read_byte_data(i2c_client, CS4270_CHIPID); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to read I2C\n"); + goto error; + } + /* The top four bits of the chip ID should be 1100. */ + if ((ret & 0xF0) != 0xC0) { + /* The device at this address is not a CS4270 codec */ + ret = -ENODEV; + goto error; + } + + printk(KERN_INFO "cs4270: found device at I2C address %X\n", addr); + printk(KERN_INFO "cs4270: hardware revision %X\n", ret & 0xF); + + /* Tell the I2C layer a new client has arrived */ + + ret = i2c_attach_client(i2c_client); + if (ret) { + printk(KERN_ERR "cs4270: failed to attach codec at I2C addr %x\n", addr); + goto error; + } + + codec->control_data = i2c_client; + codec->read = cs4270_read_reg_cache; + codec->write = cs4270_i2c_write; + codec->reg_cache_size = CS4270_NUMREGS; + + /* The I2C interface is set up, so pre-fill our register cache */ + + ret = cs4270_fill_cache(codec); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to fill register cache\n"); + goto error; + } + + /* Add the non-DAPM controls */ + + for (i = 0; i < ARRAY_SIZE(cs4270_snd_controls); i++) { + struct snd_kcontrol *kctrl = + snd_soc_cnew(&cs4270_snd_controls[i], codec, NULL); + + ret = snd_ctl_add(codec->card, kctrl); + if (ret < 0) + goto error; + } + + return 0; + +error: + if (codec->control_data) { + i2c_detach_client(i2c_client); + codec->control_data = NULL; + } + + kfree(codec->reg_cache); + codec->reg_cache = NULL; + codec->reg_cache_size = 0; + + kfree(i2c_client); + + return ret; +} + +#endif + +struct snd_soc_codec_dai cs4270_dai = { + .name = "CS4270", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = 0, + .formats = CS4270_FORMATS, + }, + .dai_ops = { + .set_sysclk = cs4270_set_dai_sysclk, + .set_fmt = cs4270_set_dai_fmt, + } +}; +EXPORT_SYMBOL_GPL(cs4270_dai); + +/** + * ASoC probe function + * + * This function is called when the machine driver calls + * platform_device_add(). + */ +static int cs4270_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + printk(KERN_INFO "CS4270 ALSA SoC Codec\n"); + + /* Allocate enough space for the snd_soc_codec structure + and our private data together. */ + codec = kzalloc(ALIGN(sizeof(struct snd_soc_codec), 4) + + sizeof(struct cs4270_private), GFP_KERNEL); + if (!codec) { + printk(KERN_ERR "cs4270: Could not allocate codec structure.\n"); + return -ENOMEM; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->name = "CS4270"; + codec->owner = THIS_MODULE; + codec->dai = &cs4270_dai; + codec->num_dai = 1; + codec->private_data = codec + ALIGN(sizeof(struct snd_soc_codec), 4); + + socdev->codec = codec; + + /* Register PCMs */ + + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to create PCMs\n"); + return ret; + } + +#ifdef CONFIG_I2C + cs4270_socdev = socdev; + + ret = i2c_add_driver(&cs4270_i2c_driver); + if (ret) { + printk(KERN_ERR "cs4270: failed to attach driver"); + snd_soc_free_pcms(socdev); + return ret; + } + + /* Did we find a CS4270 on the I2C bus? */ + if (codec->control_data) { + /* Initialize codec ops */ + cs4270_dai.ops.hw_params = cs4270_hw_params; +#ifdef CONFIG_SND_SOC_CS4270_HWMUTE + cs4270_dai.dai_ops.digital_mute = cs4270_mute; +#endif + } else { + printk(KERN_INFO "cs4270: no I2C device found, using stand-alone mode\n"); + } +#else + printk(KERN_INFO "cs4270: I2C disabled, using stand-alone mode\n"); +#endif + + ret = snd_soc_register_card(socdev); + if (ret < 0) { + printk(KERN_ERR "cs4270: failed to register card\n"); + snd_soc_free_pcms(socdev); + return ret; + } + + return ret; +} + +static int cs4270_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + +#ifdef CONFIG_I2C + if (socdev->codec->control_data) + i2c_del_driver(&cs4270_i2c_driver); +#endif + + kfree(socdev->codec); + socdev->codec = NULL; + + return 0; +} + +/** + * ASoC codec device structure + * + * Assign this variable to the codec_dev field of the machine driver's + * snd_soc_device structure. + */ +struct snd_soc_codec_device soc_codec_device_cs4270 = { + .probe = cs4270_probe, + .remove = cs4270_remove +}; +EXPORT_SYMBOL_GPL(soc_codec_device_cs4270); + +MODULE_AUTHOR("Timur Tabi <timur(a)freescale.com>"); +MODULE_DESCRIPTION("Cirrus Logic CS4270 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs4270.h b/sound/soc/codecs/cs4270.h new file mode 100644 index 0000000..35e5fe3 --- /dev/null +++ b/sound/soc/codecs/cs4270.h @@ -0,0 +1,28 @@ +/* + * Cirrus Logic CS4270 ALSA SoC Codec Driver + * + * Author: Timur Tabi <timur(a)freescale.com> + * + * Copyright 2007 Freescale Semiconductor, Inc. This file is licensed under + * the terms of the GNU General Public License version 2. This program + * is licensed "as is" without any warranty of any kind, whether express + * or implied. + */ + +#ifndef _CS4270_H +#define _CS4270_H + +/** + * The ASoC codec DAI structure for the CS4270. Assign this structure to + * the .codec_dai field of your machine driver's snd_soc_dai_link structure. + */ +extern struct snd_soc_codec_dai cs4270_dai; + +/** + * The ASoC codec device structure for the CS4270. Assign this structure + * to the .codec_dev field of your machine driver's snd_soc_device + * structure. + */ +extern struct snd_soc_codec_device soc_codec_device_cs4270; + +#endif -- 1.5.2.4

2 2

[alsa-devel] problem to build alsa shared library for arm platform
by Susovan Ghosh 30 Jul '07

30 Jul '07

Hi All, we want to use alsa library for arm platform. while configuration we specified option for shared library. but its alwaye creating static library after compilation. when we build the test application pcm_min.c we got the following error. /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0x194): In function `snd_pcm_ladspa_free_plugins': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:188: undefined reference to `dlclose' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0xd28): In function `snd_pcm_ladspa_connect_controls': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:515: undefined reference to `log' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0xd50):/home/susovan/alsa/lib/ alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:515: undefined reference to `log' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0xd74):/home/susovan/alsa/lib/ alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:536: undefined reference to `exp' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0xd9c):/home/susovan/alsa/lib/ alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:526: undefined reference to `sqrtf' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0xdac):/home/susovan/alsa/lib/ alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:536: undefined reference to `log' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0xdd4):/home/susovan/alsa/lib/ alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:536: undefined reference to `log' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0x1fac): In function `snd_pcm_ladspa_check_file': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:1087: undefined reference to `dlopen' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0x1fc0):/home/susovan/alsa/lib /alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:1089: undefined reference to `dlsym' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0x20b8):/home/susovan/alsa/lib /alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:1132: undefined reference to `dlclose' /usr/lib_alsa/libasound.a(pcm_ladspa.o)(.text+0x20e0):/home/susovan/alsa/lib /alsa-lib-1.0.14rc4/src/pcm/pcm_ladspa.c:1124: undefined reference to `dlclose' /usr/lib_alsa/libasound.a(dlmisc.o)(.text+0xc): In function `snd_dlopen': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/dlmisc.c:64: undefined reference to `dlopen' /usr/lib_alsa/libasound.a(dlmisc.o)(.text+0x24): In function `snd_dlclose': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/dlmisc.c:85: undefined reference to `dlclose' /usr/lib_alsa/libasound.a(dlmisc.o)(.text+0xa8): In function `snd_dlsym_verify': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/dlmisc.c:115: undefined reference to `dlsym' /usr/lib_alsa/libasound.a(dlmisc.o)(.text+0x144): In function `snd_dlsym': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/dlmisc.c:161: undefined reference to `dlsym' /usr/lib_alsa/libasound.a(pcm_meter.o)(.text+0x74c): In function `snd_pcm_meter_avail_update': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_meter.c:106: undefined reference to `pthread_mutex_trylock' /usr/lib_alsa/libasound.a(pcm_meter.o)(.text+0xa54): In function `snd_pcm_meter_hw_params': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_meter.c:470: undefined reference to `pthread_create' /usr/lib_alsa/libasound.a(pcm_meter.o)(.text+0xaec): In function `snd_pcm_meter_hw_free': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_meter.c:483: undefined reference to `pthread_join' /usr/lib_alsa/libasound.a(pcm_meter.o)(.text+0xf84): In function `snd_pcm_meter_add_scope_conf': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_meter.c:664: undefined reference to `dlsym' /usr/lib_alsa/libasound.a(pcm_route.o)(.text+0x12c8): In function `snd_pcm_route_convert1_many': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_route.c:453: undefined reference to `rint' /usr/lib_alsa/libasound.a(pcm_share.o)(.text+0x1dd0): In function `snd_pcm_share_close': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_share.c:1243: undefined reference to `pthread_join' /usr/lib_alsa/libasound.a(pcm_share.o)(.text+0x21b8): In function `snd_pcm_share_open': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_share.c:1474: undefined reference to `pthread_create' /usr/lib_alsa/libasound.a(pcm_softvol.o)(.text+0x18a4): In function `softvol_load_control': /home/susovan/alsa/lib/alsa-lib-1.0.14rc4/src/pcm/pcm_softvol.c:665: undefined reference to `pow' we tried with the following releases- 1)alsa-lib-1.0.14 2) alsa-lib-1.0.14a 3)alsa-lib-1.0.14rc4 But when we tried to build alsa shared library for x68 platform we did not face this problem , but when we built the alsa static library for x86 and tried to compile the test application we face the above "undefine reference " error. Can any one help me to understand and solve this problem? Thank you Susovan Ghosh Engineer (D&D ) PRDE Ph No:-998667320 Bangalore-560048

2 1

[alsa-devel] humming sound/distorted sound
by Hemanth Kumar 30 Jul '07

30 Jul '07

Hi All, I am new to alsa , I am using the Texas Instrument Davinci Board (ARM arch) with linux-2.6.10 with alsa lib 1.0.6, ALSA was working fine the sound was good,Then I changed the kernel to 2.6.20 with out changeing the alsa lib no when play sound I get humming sound/distorted sound I have few question, 1 .Does I need to change the alsa lib also 2 .Does I need to change/ Select addtional option's in kernel(I have copied the linux-2.6.10/.config file to linux-2.6.20) Can any please point me where I went wrong

2 1

[alsa-devel] Patch: HDA-Codec: Add support for the ASRock K8NF6G-VSTA motherboard.
by Tobin Davis 30 Jul '07

30 Jul '07

This patch adds ALC861VD support for the ASRock K8NF6G-VSTA motherboard. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] [PATCH] DDB5477: Remove driver bits of support
by Ralf Baechle 30 Jul '07

30 Jul '07

From: Yoichi Yuasa <yoichi_yuasa(a)tripeaks.co.jp> Signed-off-by: Yoichi Yuasa <yoichi_yuasa(a)tripeaks.co.jp> Signed-off-by: Ralf Baechle <ralf(a)linux-mips.org> drivers/net/tulip/tulip_core.c | 8 drivers/serial/8250_pci.c | 20 include/linux/pci_ids.h | 1 sound/oss/Kconfig | 8 sound/oss/Makefile | 1 sound/oss/nec_vrc5477.c | 2060 ----------------------------------------- 6 files changed, 2098 deletions(-) Index: upstream-linus/drivers/net/tulip/tulip_core.c =================================================================== --- upstream-linus.orig/drivers/net/tulip/tulip_core.c +++ upstream-linus/drivers/net/tulip/tulip_core.c @@ -1471,14 +1471,6 @@ static int __devinit tulip_init_one (str sa_offset = 2; /* Grrr, damn Matrox boards. */ multiport_cnt = 4; } -#ifdef CONFIG_DDB5477 - if ((pdev->bus->number == 0) && (PCI_SLOT(pdev->devfn) == 4)) { - /* DDB5477 MAC address in first EEPROM locations. */ - sa_offset = 0; - /* No media table either */ - tp->flags &= ~HAS_MEDIA_TABLE; - } -#endif #ifdef CONFIG_MIPS_COBALT if ((pdev->bus->number == 0) && ((PCI_SLOT(pdev->devfn) == 7) || Index: upstream-linus/drivers/serial/8250_pci.c =================================================================== --- upstream-linus.orig/drivers/serial/8250_pci.c +++ upstream-linus/drivers/serial/8250_pci.c @@ -976,7 +976,6 @@ enum pci_board_num_t { pbn_oxsemi, pbn_intel_i960, pbn_sgi_ioc3, - pbn_nec_nile4, pbn_computone_4, pbn_computone_6, pbn_computone_8, @@ -1443,18 +1442,6 @@ static struct pciserial_board pci_boards }, /* - * NEC Vrc-5074 (Nile 4) builtin UART. - */ - [pbn_nec_nile4] = { - .flags = FL_BASE0, - .num_ports = 1, - .base_baud = 520833, - .uart_offset = 8 << 3, - .reg_shift = 3, - .first_offset = 0x300, - }, - - /* * Computone - uses IOMEM. */ [pbn_computone_4] = { @@ -2345,13 +2332,6 @@ static struct pci_device_id serial_pci_t PCI_ANY_ID, PCI_ANY_ID, 0, 0, pbn_b2_1_115200 }, - /* - * NEC Vrc-5074 (Nile 4) builtin UART. - */ - { PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_NILE4, - PCI_ANY_ID, PCI_ANY_ID, 0, 0, - pbn_nec_nile4 }, - { PCI_VENDOR_ID_DCI, PCI_DEVICE_ID_DCI_PCCOM2, PCI_ANY_ID, PCI_ANY_ID, 0, 0, pbn_b3_2_115200 }, Index: upstream-linus/sound/oss/Kconfig =================================================================== --- upstream-linus.orig/sound/oss/Kconfig +++ upstream-linus/sound/oss/Kconfig @@ -31,14 +31,6 @@ config SOUND_HAL2 Say Y or M if you have an SGI Indy or Indigo2 system and want to be able to use its on-board A2 audio system. -config SOUND_VRC5477 - tristate "NEC Vrc5477 AC97 sound" - depends on SOUND_PRIME && DDB5477 - help - Say Y here to enable sound support for the NEC Vrc5477 chip, an - integrated, multi-function controller chip for MIPS CPUs. Works - with the AC97 codec. - config SOUND_AU1550_AC97 tristate "Au1550/Au1200 AC97 Sound" select SND_AC97_CODEC Index: upstream-linus/sound/oss/Makefile =================================================================== --- upstream-linus.orig/sound/oss/Makefile +++ upstream-linus/sound/oss/Makefile @@ -37,7 +37,6 @@ obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o ms obj-$(CONFIG_SOUND_VWSND) += vwsnd.o obj-$(CONFIG_SOUND_ICH) += i810_audio.o ac97_codec.o obj-$(CONFIG_SOUND_ES1371) += es1371.o ac97_codec.o -obj-$(CONFIG_SOUND_VRC5477) += nec_vrc5477.o ac97_codec.o obj-$(CONFIG_SOUND_AU1550_AC97) += au1550_ac97.o ac97_codec.o obj-$(CONFIG_SOUND_TRIDENT) += trident.o ac97_codec.o obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o Index: upstream-linus/sound/oss/nec_vrc5477.c =================================================================== --- upstream-linus.orig/sound/oss/nec_vrc5477.c +++ /dev/null @@ -1,2060 +0,0 @@ -/*********************************************************************** - * Copyright 2001 MontaVista Software Inc. - * Author: Jun Sun, jsun(a)mvista.com or jsun(a)junsun.net - * - * drivers/sound/nec_vrc5477.c - * AC97 sound dirver for NEC Vrc5477 chip (an integrated, - * multi-function controller chip for MIPS CPUs) - * - * VRA support Copyright 2001 Bradley D. LaRonde <brad(a)ltc.com> - * - * 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 code is derived from ite8172.c, which is written by Steve Longerbeam. - * - * Features: - * Currently we only support the following capabilities: - * . mono output to PCM L/R (line out). - * . stereo output to PCM L/R (line out). - * . mono input from PCM L (line in). - * . stereo output from PCM (line in). - * . sampling rate at 48k or variable sampling rate - * . support /dev/dsp, /dev/mixer devices, standard OSS devices. - * . only support 16-bit PCM format (hardware limit, no software - * translation) - * . support duplex, but no trigger or realtime. - * - * Specifically the following are not supported: - * . app-set frag size. - * . mmap'ed buffer access - */ - -/* - * Original comments from ite8172.c file. - */ - -/* - * - * Notes: - * - * 1. Much of the OSS buffer allocation, ioctl's, and mmap'ing are - * taken, slightly modified or not at all, from the ES1371 driver, - * so refer to the credits in es1371.c for those. The rest of the - * code (probe, open, read, write, the ISR, etc.) is new. - * 2. The following support is untested: - * * Memory mapping the audio buffers, and the ioctl controls that go - * with it. - * * S/PDIF output. - * 3. The following is not supported: - * * I2S input. - * * legacy audio mode. - * 4. Support for volume button interrupts is implemented but doesn't - * work yet. - * - * Revision history - * 02.08.2001 0.1 Initial release - */ - -#include <linux/module.h> -#include <linux/string.h> -#include <linux/kernel.h> -#include <linux/ioport.h> -#include <linux/sched.h> -#include <linux/delay.h> -#include <linux/sound.h> -#include <linux/slab.h> -#include <linux/soundcard.h> -#include <linux/pci.h> -#include <linux/init.h> -#include <linux/poll.h> -#include <linux/bitops.h> -#include <linux/proc_fs.h> -#include <linux/spinlock.h> -#include <linux/smp_lock.h> -#include <linux/ac97_codec.h> -#include <linux/mutex.h> - -#include <asm/io.h> -#include <asm/dma.h> -#include <asm/uaccess.h> - -/* -------------------debug macros -------------------------------------- */ -/* #undef VRC5477_AC97_DEBUG */ -#define VRC5477_AC97_DEBUG - -#undef VRC5477_AC97_VERBOSE_DEBUG -/* #define VRC5477_AC97_VERBOSE_DEBUG */ - -#if defined(VRC5477_AC97_VERBOSE_DEBUG) -#define VRC5477_AC97_DEBUG -#endif - -#if defined(VRC5477_AC97_DEBUG) -#define ASSERT(x) if (!(x)) { \ - panic("assertion failed at %s:%d: %s\n", __FILE__, __LINE__, #x); } -#else -#define ASSERT(x) -#endif /* VRC5477_AC97_DEBUG */ - -#if defined(VRC5477_AC97_VERBOSE_DEBUG) -static u16 inTicket; /* check sync between intr & write */ -static u16 outTicket; -#endif - -/* --------------------------------------------------------------------- */ - -#undef OSS_DOCUMENTED_MIXER_SEMANTICS - -static const unsigned sample_shift[] = { 0, 1, 1, 2 }; - -#define VRC5477_INT_CLR 0x0 -#define VRC5477_INT_STATUS 0x0 -#define VRC5477_CODEC_WR 0x4 -#define VRC5477_CODEC_RD 0x8 -#define VRC5477_CTRL 0x18 -#define VRC5477_ACLINK_CTRL 0x1c -#define VRC5477_INT_MASK 0x24 - -#define VRC5477_DAC1_CTRL 0x30 -#define VRC5477_DAC1L 0x34 -#define VRC5477_DAC1_BADDR 0x38 -#define VRC5477_DAC2_CTRL 0x3c -#define VRC5477_DAC2L 0x40 -#define VRC5477_DAC2_BADDR 0x44 -#define VRC5477_DAC3_CTRL 0x48 -#define VRC5477_DAC3L 0x4c -#define VRC5477_DAC3_BADDR 0x50 - -#define VRC5477_ADC1_CTRL 0x54 -#define VRC5477_ADC1L 0x58 -#define VRC5477_ADC1_BADDR 0x5c -#define VRC5477_ADC2_CTRL 0x60 -#define VRC5477_ADC2L 0x64 -#define VRC5477_ADC2_BADDR 0x68 -#define VRC5477_ADC3_CTRL 0x6c -#define VRC5477_ADC3L 0x70 -#define VRC5477_ADC3_BADDR 0x74 - -#define VRC5477_CODEC_WR_RWC (1 << 23) - -#define VRC5477_CODEC_RD_RRDYA (1 << 31) -#define VRC5477_CODEC_RD_RRDYD (1 << 30) - -#define VRC5477_ACLINK_CTRL_RST_ON (1 << 15) -#define VRC5477_ACLINK_CTRL_RST_TIME 0x7f -#define VRC5477_ACLINK_CTRL_SYNC_ON (1 << 30) -#define VRC5477_ACLINK_CTRL_CK_STOP_ON (1 << 31) - -#define VRC5477_CTRL_DAC2ENB (1 << 15) -#define VRC5477_CTRL_ADC2ENB (1 << 14) -#define VRC5477_CTRL_DAC1ENB (1 << 13) -#define VRC5477_CTRL_ADC1ENB (1 << 12) - -#define VRC5477_INT_MASK_NMASK (1 << 31) -#define VRC5477_INT_MASK_DAC1END (1 << 5) -#define VRC5477_INT_MASK_DAC2END (1 << 4) -#define VRC5477_INT_MASK_DAC3END (1 << 3) -#define VRC5477_INT_MASK_ADC1END (1 << 2) -#define VRC5477_INT_MASK_ADC2END (1 << 1) -#define VRC5477_INT_MASK_ADC3END (1 << 0) - -#define VRC5477_DMA_ACTIVATION (1 << 31) -#define VRC5477_DMA_WIP (1 << 30) - - -#define VRC5477_AC97_MODULE_NAME "NEC_Vrc5477_audio" -#define PFX VRC5477_AC97_MODULE_NAME ": " - -/* --------------------------------------------------------------------- */ - -struct vrc5477_ac97_state { - /* list of vrc5477_ac97 devices */ - struct list_head devs; - - /* the corresponding pci_dev structure */ - struct pci_dev *dev; - - /* soundcore stuff */ - int dev_audio; - - /* hardware resources */ - unsigned long io; - unsigned int irq; - -#ifdef VRC5477_AC97_DEBUG - /* debug /proc entry */ - struct proc_dir_entry *ps; - struct proc_dir_entry *ac97_ps; -#endif /* VRC5477_AC97_DEBUG */ - - struct ac97_codec *codec; - - unsigned dacChannels, adcChannels; - unsigned short dacRate, adcRate; - unsigned short extended_status; - - spinlock_t lock; - struct mutex open_mutex; - mode_t open_mode; - wait_queue_head_t open_wait; - - struct dmabuf { - void *lbuf, *rbuf; - dma_addr_t lbufDma, rbufDma; - unsigned bufOrder; - unsigned numFrag; - unsigned fragShift; - unsigned fragSize; /* redundant */ - unsigned fragTotalSize; /* = numFrag * fragSize(real) */ - unsigned nextIn; - unsigned nextOut; - int count; - unsigned error; /* over/underrun */ - wait_queue_head_t wait; - /* OSS stuff */ - unsigned stopped:1; - unsigned ready:1; - } dma_dac, dma_adc; - - #define WORK_BUF_SIZE 2048 - struct { - u16 lchannel; - u16 rchannel; - } workBuf[WORK_BUF_SIZE/4]; -}; - -/* --------------------------------------------------------------------- */ - -static LIST_HEAD(devs); - -/* --------------------------------------------------------------------- */ - -static inline unsigned ld2(unsigned int x) -{ - unsigned r = 0; - - if (x >= 0x10000) { - x >>= 16; - r += 16; - } - if (x >= 0x100) { - x >>= 8; - r += 8; - } - if (x >= 0x10) { - x >>= 4; - r += 4; - } - if (x >= 4) { - x >>= 2; - r += 2; - } - if (x >= 2) - r++; - return r; -} - -/* --------------------------------------------------------------------- */ - -static u16 rdcodec(struct ac97_codec *codec, u8 addr) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)codec->private_data; - unsigned long flags; - u32 result; - - spin_lock_irqsave(&s->lock, flags); - - /* wait until we can access codec registers */ - while (inl(s->io + VRC5477_CODEC_WR) & 0x80000000); - - /* write the address and "read" command to codec */ - addr = addr & 0x7f; - outl((addr << 16) | VRC5477_CODEC_WR_RWC, s->io + VRC5477_CODEC_WR); - - /* get the return result */ - udelay(100); /* workaround hardware bug */ - while ( (result = inl(s->io + VRC5477_CODEC_RD)) & - (VRC5477_CODEC_RD_RRDYA | VRC5477_CODEC_RD_RRDYD) ) { - /* we get either addr or data, or both */ - if (result & VRC5477_CODEC_RD_RRDYA) { - ASSERT(addr == ((result >> 16) & 0x7f) ); - } - if (result & VRC5477_CODEC_RD_RRDYD) { - break; - } - } - - spin_unlock_irqrestore(&s->lock, flags); - - return result & 0xffff; -} - - -static void wrcodec(struct ac97_codec *codec, u8 addr, u16 data) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)codec->private_data; - unsigned long flags; - - spin_lock_irqsave(&s->lock, flags); - - /* wait until we can access codec registers */ - while (inl(s->io + VRC5477_CODEC_WR) & 0x80000000); - - /* write the address and value to codec */ - outl((addr << 16) | data, s->io + VRC5477_CODEC_WR); - - spin_unlock_irqrestore(&s->lock, flags); -} - - -static void waitcodec(struct ac97_codec *codec) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)codec->private_data; - - /* wait until we can access codec registers */ - while (inl(s->io + VRC5477_CODEC_WR) & 0x80000000); -} - -static int ac97_codec_not_present(struct ac97_codec *codec) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)codec->private_data; - unsigned long flags; - unsigned short count = 0xffff; - - spin_lock_irqsave(&s->lock, flags); - - /* wait until we can access codec registers */ - do { - if (!(inl(s->io + VRC5477_CODEC_WR) & 0x80000000)) - break; - } while (--count); - - if (count == 0) { - spin_unlock_irqrestore(&s->lock, flags); - return -1; - } - - /* write 0 to reset */ - outl((AC97_RESET << 16) | 0, s->io + VRC5477_CODEC_WR); - - /* test whether we get a response from ac97 chip */ - count = 0xffff; - do { - if (!(inl(s->io + VRC5477_CODEC_WR) & 0x80000000)) - break; - } while (--count); - - if (count == 0) { - spin_unlock_irqrestore(&s->lock, flags); - return -1; - } - spin_unlock_irqrestore(&s->lock, flags); - return 0; -} - -/* --------------------------------------------------------------------- */ - -static void vrc5477_ac97_delay(int msec) -{ - unsigned long tmo; - signed long tmo2; - - if (in_interrupt()) - return; - - tmo = jiffies + (msec*HZ)/1000; - for (;;) { - tmo2 = tmo - jiffies; - if (tmo2 <= 0) - break; - schedule_timeout(tmo2); - } -} - - -static void set_adc_rate(struct vrc5477_ac97_state *s, unsigned rate) -{ - wrcodec(s->codec, AC97_PCM_LR_ADC_RATE, rate); - s->adcRate = rate; -} - - -static void set_dac_rate(struct vrc5477_ac97_state *s, unsigned rate) -{ - if(s->extended_status & AC97_EXTSTAT_VRA) { - wrcodec(s->codec, AC97_PCM_FRONT_DAC_RATE, rate); - s->dacRate = rdcodec(s->codec, AC97_PCM_FRONT_DAC_RATE); - } -} - -static int ac97_codec_not_present(struct ac97_codec *codec) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)codec->private_data; - unsigned long flags; - unsigned short count = 0xffff; - - spin_lock_irqsave(&s->lock, flags); - - /* wait until we can access codec registers */ - do { - if (!(inl(s->io + VRC5477_CODEC_WR) & 0x80000000)) - break; - } while (--count); - - if (count == 0) { - spin_unlock_irqrestore(&s->lock, flags); - return -1; - } - - /* write 0 to reset */ - outl((AC97_RESET << 16) | 0, s->io + VRC5477_CODEC_WR); - - /* test whether we get a response from ac97 chip */ - count = 0xffff; - do { - if (!(inl(s->io + VRC5477_CODEC_WR) & 0x80000000)) - break; - } while (--count); - - if (count == 0) { - spin_unlock_irqrestore(&s->lock, flags); - return -1; - } - spin_unlock_irqrestore(&s->lock, flags); - return 0; -} - -/* --------------------------------------------------------------------- */ - -static inline void -stop_dac(struct vrc5477_ac97_state *s) -{ - struct dmabuf* db = &s->dma_dac; - unsigned long flags; - u32 temp; - - spin_lock_irqsave(&s->lock, flags); - - if (db->stopped) { - spin_unlock_irqrestore(&s->lock, flags); - return; - } - - /* deactivate the dma */ - outl(0, s->io + VRC5477_DAC1_CTRL); - outl(0, s->io + VRC5477_DAC2_CTRL); - - /* wait for DAM completely stop */ - while (inl(s->io + VRC5477_DAC1_CTRL) & VRC5477_DMA_WIP); - while (inl(s->io + VRC5477_DAC2_CTRL) & VRC5477_DMA_WIP); - - /* disable dac slots in aclink */ - temp = inl(s->io + VRC5477_CTRL); - temp &= ~ (VRC5477_CTRL_DAC1ENB | VRC5477_CTRL_DAC2ENB); - outl (temp, s->io + VRC5477_CTRL); - - /* disable interrupts */ - temp = inl(s->io + VRC5477_INT_MASK); - temp &= ~ (VRC5477_INT_MASK_DAC1END | VRC5477_INT_MASK_DAC2END); - outl (temp, s->io + VRC5477_INT_MASK); - - /* clear pending ones */ - outl(VRC5477_INT_MASK_DAC1END | VRC5477_INT_MASK_DAC2END, - s->io + VRC5477_INT_CLR); - - db->stopped = 1; - - spin_unlock_irqrestore(&s->lock, flags); -} - -static void start_dac(struct vrc5477_ac97_state *s) -{ - struct dmabuf* db = &s->dma_dac; - unsigned long flags; - u32 dmaLength; - u32 temp; - - spin_lock_irqsave(&s->lock, flags); - - if (!db->stopped) { - spin_unlock_irqrestore(&s->lock, flags); - return; - } - - /* we should have some data to do the DMA trasnfer */ - ASSERT(db->count >= db->fragSize); - - /* clear pending fales interrupts */ - outl(VRC5477_INT_MASK_DAC1END | VRC5477_INT_MASK_DAC2END, - s->io + VRC5477_INT_CLR); - - /* enable interrupts */ - temp = inl(s->io + VRC5477_INT_MASK); - temp |= VRC5477_INT_MASK_DAC1END | VRC5477_INT_MASK_DAC2END; - outl(temp, s->io + VRC5477_INT_MASK); - - /* setup dma base addr */ - outl(db->lbufDma + db->nextOut, s->io + VRC5477_DAC1_BADDR); - if (s->dacChannels == 1) { - outl(db->lbufDma + db->nextOut, s->io + VRC5477_DAC2_BADDR); - } else { - outl(db->rbufDma + db->nextOut, s->io + VRC5477_DAC2_BADDR); - } - - /* set dma length, in the unit of 0x10 bytes */ - dmaLength = db->fragSize >> 4; - outl(dmaLength, s->io + VRC5477_DAC1L); - outl(dmaLength, s->io + VRC5477_DAC2L); - - /* activate dma */ - outl(VRC5477_DMA_ACTIVATION, s->io + VRC5477_DAC1_CTRL); - outl(VRC5477_DMA_ACTIVATION, s->io + VRC5477_DAC2_CTRL); - - /* enable dac slots - we should hear the music now! */ - temp = inl(s->io + VRC5477_CTRL); - temp |= (VRC5477_CTRL_DAC1ENB | VRC5477_CTRL_DAC2ENB); - outl (temp, s->io + VRC5477_CTRL); - - /* it is time to setup next dma transfer */ - ASSERT(inl(s->io + VRC5477_DAC1_CTRL) & VRC5477_DMA_WIP); - ASSERT(inl(s->io + VRC5477_DAC2_CTRL) & VRC5477_DMA_WIP); - - temp = db->nextOut + db->fragSize; - if (temp >= db->fragTotalSize) { - ASSERT(temp == db->fragTotalSize); - temp = 0; - } - - outl(db->lbufDma + temp, s->io + VRC5477_DAC1_BADDR); - if (s->dacChannels == 1) { - outl(db->lbufDma + temp, s->io + VRC5477_DAC2_BADDR); - } else { - outl(db->rbufDma + temp, s->io + VRC5477_DAC2_BADDR); - } - - db->stopped = 0; - -#if defined(VRC5477_AC97_VERBOSE_DEBUG) - outTicket = *(u16*)(db->lbuf+db->nextOut); - if (db->count > db->fragSize) { - ASSERT((u16)(outTicket+1) == *(u16*)(db->lbuf+temp)); - } -#endif - - spin_unlock_irqrestore(&s->lock, flags); -} - -static inline void stop_adc(struct vrc5477_ac97_state *s) -{ - struct dmabuf* db = &s->dma_adc; - unsigned long flags; - u32 temp; - - spin_lock_irqsave(&s->lock, flags); - - if (db->stopped) { - spin_unlock_irqrestore(&s->lock, flags); - return; - } - - /* deactivate the dma */ - outl(0, s->io + VRC5477_ADC1_CTRL); - outl(0, s->io + VRC5477_ADC2_CTRL); - - /* disable adc slots in aclink */ - temp = inl(s->io + VRC5477_CTRL); - temp &= ~ (VRC5477_CTRL_ADC1ENB | VRC5477_CTRL_ADC2ENB); - outl (temp, s->io + VRC5477_CTRL); - - /* disable interrupts */ - temp = inl(s->io + VRC5477_INT_MASK); - temp &= ~ (VRC5477_INT_MASK_ADC1END | VRC5477_INT_MASK_ADC2END); - outl (temp, s->io + VRC5477_INT_MASK); - - /* clear pending ones */ - outl(VRC5477_INT_MASK_ADC1END | VRC5477_INT_MASK_ADC2END, - s->io + VRC5477_INT_CLR); - - db->stopped = 1; - - spin_unlock_irqrestore(&s->lock, flags); -} - -static void start_adc(struct vrc5477_ac97_state *s) -{ - struct dmabuf* db = &s->dma_adc; - unsigned long flags; - u32 dmaLength; - u32 temp; - - spin_lock_irqsave(&s->lock, flags); - - if (!db->stopped) { - spin_unlock_irqrestore(&s->lock, flags); - return; - } - - /* we should at least have some free space in the buffer */ - ASSERT(db->count < db->fragTotalSize - db->fragSize * 2); - - /* clear pending ones */ - outl(VRC5477_INT_MASK_ADC1END | VRC5477_INT_MASK_ADC2END, - s->io + VRC5477_INT_CLR); - - /* enable interrupts */ - temp = inl(s->io + VRC5477_INT_MASK); - temp |= VRC5477_INT_MASK_ADC1END | VRC5477_INT_MASK_ADC2END; - outl(temp, s->io + VRC5477_INT_MASK); - - /* setup dma base addr */ - outl(db->lbufDma + db->nextIn, s->io + VRC5477_ADC1_BADDR); - outl(db->rbufDma + db->nextIn, s->io + VRC5477_ADC2_BADDR); - - /* setup dma length */ - dmaLength = db->fragSize >> 4; - outl(dmaLength, s->io + VRC5477_ADC1L); - outl(dmaLength, s->io + VRC5477_ADC2L); - - /* activate dma */ - outl(VRC5477_DMA_ACTIVATION, s->io + VRC5477_ADC1_CTRL); - outl(VRC5477_DMA_ACTIVATION, s->io + VRC5477_ADC2_CTRL); - - /* enable adc slots */ - temp = inl(s->io + VRC5477_CTRL); - temp |= (VRC5477_CTRL_ADC1ENB | VRC5477_CTRL_ADC2ENB); - outl (temp, s->io + VRC5477_CTRL); - - /* it is time to setup next dma transfer */ - temp = db->nextIn + db->fragSize; - if (temp >= db->fragTotalSize) { - ASSERT(temp == db->fragTotalSize); - temp = 0; - } - outl(db->lbufDma + temp, s->io + VRC5477_ADC1_BADDR); - outl(db->rbufDma + temp, s->io + VRC5477_ADC2_BADDR); - - db->stopped = 0; - - spin_unlock_irqrestore(&s->lock, flags); -} - -/* --------------------------------------------------------------------- */ - -#define DMABUF_DEFAULTORDER (16-PAGE_SHIFT) -#define DMABUF_MINORDER 1 - -static inline void dealloc_dmabuf(struct vrc5477_ac97_state *s, - struct dmabuf *db) -{ - if (db->lbuf) { - ASSERT(db->rbuf); - pci_free_consistent(s->dev, PAGE_SIZE << db->bufOrder, - db->lbuf, db->lbufDma); - pci_free_consistent(s->dev, PAGE_SIZE << db->bufOrder, - db->rbuf, db->rbufDma); - db->lbuf = db->rbuf = NULL; - } - db->nextIn = db->nextOut = 0; - db->ready = 0; -} - -static int prog_dmabuf(struct vrc5477_ac97_state *s, - struct dmabuf *db, - unsigned rate) -{ - int order; - unsigned bufsize; - - if (!db->lbuf) { - ASSERT(!db->rbuf); - - db->ready = 0; - for (order = DMABUF_DEFAULTORDER; - order >= DMABUF_MINORDER; - order--) { - db->lbuf = pci_alloc_consistent(s->dev, - PAGE_SIZE << order, - &db->lbufDma); - db->rbuf = pci_alloc_consistent(s->dev, - PAGE_SIZE << order, - &db->rbufDma); - if (db->lbuf && db->rbuf) break; - if (db->lbuf) { - ASSERT(!db->rbuf); - pci_free_consistent(s->dev, - PAGE_SIZE << order, - db->lbuf, - db->lbufDma); - } - } - if (!db->lbuf) { - ASSERT(!db->rbuf); - return -ENOMEM; - } - - db->bufOrder = order; - } - - db->count = 0; - db->nextIn = db->nextOut = 0; - - bufsize = PAGE_SIZE << db->bufOrder; - db->fragShift = ld2(rate * 2 / 100); - if (db->fragShift < 4) db->fragShift = 4; - - db->numFrag = bufsize >> db->fragShift; - while (db->numFrag < 4 && db->fragShift > 4) { - db->fragShift--; - db->numFrag = bufsize >> db->fragShift; - } - db->fragSize = 1 << db->fragShift; - db->fragTotalSize = db->numFrag << db->fragShift; - memset(db->lbuf, 0, db->fragTotalSize); - memset(db->rbuf, 0, db->fragTotalSize); - - db->ready = 1; - - return 0; -} - -static inline int prog_dmabuf_adc(struct vrc5477_ac97_state *s) -{ - stop_adc(s); - return prog_dmabuf(s, &s->dma_adc, s->adcRate); -} - -static inline int prog_dmabuf_dac(struct vrc5477_ac97_state *s) -{ - stop_dac(s); - return prog_dmabuf(s, &s->dma_dac, s->dacRate); -} - - -/* --------------------------------------------------------------------- */ -/* hold spinlock for the following! */ - -static inline void vrc5477_ac97_adc_interrupt(struct vrc5477_ac97_state *s) -{ - struct dmabuf* adc = &s->dma_adc; - unsigned temp; - - /* we need two frags avaiable because one is already being used - * and the other will be used when next interrupt happens. - */ - if (adc->count >= adc->fragTotalSize - adc->fragSize) { - stop_adc(s); - adc->error++; - printk(KERN_INFO PFX "adc overrun\n"); - return; - } - - /* set the base addr for next DMA transfer */ - temp = adc->nextIn + 2*adc->fragSize; - if (temp >= adc->fragTotalSize) { - ASSERT( (temp == adc->fragTotalSize) || - (temp == adc->fragTotalSize + adc->fragSize) ); - temp -= adc->fragTotalSize; - } - outl(adc->lbufDma + temp, s->io + VRC5477_ADC1_BADDR); - outl(adc->rbufDma + temp, s->io + VRC5477_ADC2_BADDR); - - /* adjust nextIn */ - adc->nextIn += adc->fragSize; - if (adc->nextIn >= adc->fragTotalSize) { - ASSERT(adc->nextIn == adc->fragTotalSize); - adc->nextIn = 0; - } - - /* adjust count */ - adc->count += adc->fragSize; - - /* wake up anybody listening */ - if (waitqueue_active(&adc->wait)) { - wake_up_interruptible(&adc->wait); - } -} - -static inline void vrc5477_ac97_dac_interrupt(struct vrc5477_ac97_state *s) -{ - struct dmabuf* dac = &s->dma_dac; - unsigned temp; - - /* next DMA transfer should already started */ - // ASSERT(inl(s->io + VRC5477_DAC1_CTRL) & VRC5477_DMA_WIP); - // ASSERT(inl(s->io + VRC5477_DAC2_CTRL) & VRC5477_DMA_WIP); - - /* let us set for next next DMA transfer */ - temp = dac->nextOut + dac->fragSize*2; - if (temp >= dac->fragTotalSize) { - ASSERT( (temp == dac->fragTotalSize) || - (temp == dac->fragTotalSize + dac->fragSize) ); - temp -= dac->fragTotalSize; - } - outl(dac->lbufDma + temp, s->io + VRC5477_DAC1_BADDR); - if (s->dacChannels == 1) { - outl(dac->lbufDma + temp, s->io + VRC5477_DAC2_BADDR); - } else { - outl(dac->rbufDma + temp, s->io + VRC5477_DAC2_BADDR); - } - -#if defined(VRC5477_AC97_VERBOSE_DEBUG) - if (*(u16*)(dac->lbuf + dac->nextOut) != outTicket) { - printk("assert fail: - %d vs %d\n", - *(u16*)(dac->lbuf + dac->nextOut), - outTicket); - ASSERT(1 == 0); - } -#endif - - /* adjust nextOut pointer */ - dac->nextOut += dac->fragSize; - if (dac->nextOut >= dac->fragTotalSize) { - ASSERT(dac->nextOut == dac->fragTotalSize); - dac->nextOut = 0; - } - - /* adjust count */ - dac->count -= dac->fragSize; - if (dac->count <=0 ) { - /* buffer under run */ - dac->count = 0; - dac->nextIn = dac->nextOut; - stop_dac(s); - } - -#if defined(VRC5477_AC97_VERBOSE_DEBUG) - if (dac->count) { - outTicket ++; - ASSERT(*(u16*)(dac->lbuf + dac->nextOut) == outTicket); - } -#endif - - /* we cannot have both under run and someone is waiting on us */ - ASSERT(! (waitqueue_active(&dac->wait) && (dac->count <= 0)) ); - - /* wake up anybody listening */ - if (waitqueue_active(&dac->wait)) - wake_up_interruptible(&dac->wait); -} - -static irqreturn_t vrc5477_ac97_interrupt(int irq, void *dev_id) -{ - struct vrc5477_ac97_state *s = (struct vrc5477_ac97_state *)dev_id; - u32 irqStatus; - u32 adcInterrupts, dacInterrupts; - - spin_lock(&s->lock); - - /* get irqStatus and clear the detected ones */ - irqStatus = inl(s->io + VRC5477_INT_STATUS); - outl(irqStatus, s->io + VRC5477_INT_CLR); - - /* let us see what we get */ - dacInterrupts = VRC5477_INT_MASK_DAC1END | VRC5477_INT_MASK_DAC2END; - adcInterrupts = VRC5477_INT_MASK_ADC1END | VRC5477_INT_MASK_ADC2END; - if (irqStatus & dacInterrupts) { - /* we should get both interrupts, but just in case ... */ - if (irqStatus & VRC5477_INT_MASK_DAC1END) { - vrc5477_ac97_dac_interrupt(s); - } - if ( (irqStatus & dacInterrupts) != dacInterrupts ) { - printk(KERN_WARNING "vrc5477_ac97 : dac interrupts not in sync!!!\n"); - stop_dac(s); - start_dac(s); - } - } else if (irqStatus & adcInterrupts) { - /* we should get both interrupts, but just in case ... */ - if(irqStatus & VRC5477_INT_MASK_ADC1END) { - vrc5477_ac97_adc_interrupt(s); - } - if ( (irqStatus & adcInterrupts) != adcInterrupts ) { - printk(KERN_WARNING "vrc5477_ac97 : adc interrupts not in sync!!!\n"); - stop_adc(s); - start_adc(s); - } - } - - spin_unlock(&s->lock); - return IRQ_HANDLED; -} - -/* --------------------------------------------------------------------- */ - -static int vrc5477_ac97_open_mixdev(struct inode *inode, struct file *file) -{ - int minor = iminor(inode); - struct list_head *list; - struct vrc5477_ac97_state *s; - - for (list = devs.next; ; list = list->next) { - if (list == &devs) - return -ENODEV; - s = list_entry(list, struct vrc5477_ac97_state, devs); - if (s->codec->dev_mixer == minor) - break; - } - file->private_data = s; - return nonseekable_open(inode, file); -} - -static int vrc5477_ac97_release_mixdev(struct inode *inode, struct file *file) -{ - return 0; -} - - -static int mixdev_ioctl(struct ac97_codec *codec, unsigned int cmd, - unsigned long arg) -{ - return codec->mixer_ioctl(codec, cmd, arg); -} - -static int vrc5477_ac97_ioctl_mixdev(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)file->private_data; - struct ac97_codec *codec = s->codec; - - return mixdev_ioctl(codec, cmd, arg); -} - -static /*const*/ struct file_operations vrc5477_ac97_mixer_fops = { - .owner = THIS_MODULE, - .llseek = no_llseek, - .ioctl = vrc5477_ac97_ioctl_mixdev, - .open = vrc5477_ac97_open_mixdev, - .release = vrc5477_ac97_release_mixdev, -}; - -/* --------------------------------------------------------------------- */ - -static int drain_dac(struct vrc5477_ac97_state *s, int nonblock) -{ - unsigned long flags; - int count, tmo; - - if (!s->dma_dac.ready) - return 0; - - for (;;) { - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - spin_unlock_irqrestore(&s->lock, flags); - if (count <= 0) - break; - if (signal_pending(current)) - break; - if (nonblock) - return -EBUSY; - tmo = 1000 * count / s->dacRate / 2; - vrc5477_ac97_delay(tmo); - } - if (signal_pending(current)) - return -ERESTARTSYS; - return 0; -} - -/* --------------------------------------------------------------------- */ - -static inline int -copy_two_channel_adc_to_user(struct vrc5477_ac97_state *s, - char *buffer, - int copyCount) -{ - struct dmabuf *db = &s->dma_adc; - int bufStart = db->nextOut; - for (; copyCount > 0; ) { - int i; - int count = copyCount; - if (count > WORK_BUF_SIZE/2) count = WORK_BUF_SIZE/2; - for (i=0; i< count/2; i++) { - s->workBuf[i].lchannel = - *(u16*)(db->lbuf + bufStart + i*2); - s->workBuf[i].rchannel = - *(u16*)(db->rbuf + bufStart + i*2); - } - if (copy_to_user(buffer, s->workBuf, count*2)) { - return -1; - } - - copyCount -= count; - bufStart += count; - ASSERT(bufStart <= db->fragTotalSize); - buffer += count *2; - } - return 0; -} - -/* return the total bytes that is copied */ -static inline int -copy_adc_to_user(struct vrc5477_ac97_state *s, - char * buffer, - size_t count, - int avail) -{ - struct dmabuf *db = &s->dma_adc; - int copyCount=0; - int copyFragCount=0; - int totalCopyCount = 0; - int totalCopyFragCount = 0; - unsigned long flags; - - /* adjust count to signel channel byte count */ - count >>= s->adcChannels - 1; - - /* we may have to "copy" twice as ring buffer wraps around */ - for (; (avail > 0) && (count > 0); ) { - /* determine max possible copy count for single channel */ - copyCount = count; - if (copyCount > avail) { - copyCount = avail; - } - if (copyCount + db->nextOut > db->fragTotalSize) { - copyCount = db->fragTotalSize - db->nextOut; - ASSERT((copyCount % db->fragSize) == 0); - } - - copyFragCount = (copyCount-1) >> db->fragShift; - copyFragCount = (copyFragCount+1) << db->fragShift; - ASSERT(copyFragCount >= copyCount); - - /* we copy differently based on adc channels */ - if (s->adcChannels == 1) { - if (copy_to_user(buffer, - db->lbuf + db->nextOut, - copyCount)) - return -1; - } else { - /* *sigh* we have to mix two streams into one */ - if (copy_two_channel_adc_to_user(s, buffer, copyCount)) - return -1; - } - - count -= copyCount; - totalCopyCount += copyCount; - avail -= copyFragCount; - totalCopyFragCount += copyFragCount; - - buffer += copyCount << (s->adcChannels-1); - - db->nextOut += copyFragCount; - if (db->nextOut >= db->fragTotalSize) { - ASSERT(db->nextOut == db->fragTotalSize); - db->nextOut = 0; - } - - ASSERT((copyFragCount % db->fragSize) == 0); - ASSERT( (count == 0) || (copyCount == copyFragCount)); - } - - spin_lock_irqsave(&s->lock, flags); - db->count -= totalCopyFragCount; - spin_unlock_irqrestore(&s->lock, flags); - - return totalCopyCount << (s->adcChannels-1); -} - -static ssize_t -vrc5477_ac97_read(struct file *file, - char *buffer, - size_t count, - loff_t *ppos) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)file->private_data; - struct dmabuf *db = &s->dma_adc; - ssize_t ret = 0; - unsigned long flags; - int copyCount; - size_t avail; - - if (!access_ok(VERIFY_WRITE, buffer, count)) - return -EFAULT; - - ASSERT(db->ready); - - while (count > 0) { - // wait for samples in capture buffer - do { - spin_lock_irqsave(&s->lock, flags); - if (db->stopped) - start_adc(s); - avail = db->count; - spin_unlock_irqrestore(&s->lock, flags); - if (avail <= 0) { - if (file->f_flags & O_NONBLOCK) { - if (!ret) - ret = -EAGAIN; - return ret; - } - interruptible_sleep_on(&db->wait); - if (signal_pending(current)) { - if (!ret) - ret = -ERESTARTSYS; - return ret; - } - } - } while (avail <= 0); - - ASSERT( (avail % db->fragSize) == 0); - copyCount = copy_adc_to_user(s, buffer, count, avail); - if (copyCount <=0 ) { - if (!ret) ret = -EFAULT; - return ret; - } - - count -= copyCount; - buffer += copyCount; - ret += copyCount; - } // while (count > 0) - - return ret; -} - -static inline int -copy_two_channel_dac_from_user(struct vrc5477_ac97_state *s, - const char *buffer, - int copyCount) -{ - struct dmabuf *db = &s->dma_dac; - int bufStart = db->nextIn; - - ASSERT(db->ready); - - for (; copyCount > 0; ) { - int i; - int count = copyCount; - if (count > WORK_BUF_SIZE/2) count = WORK_BUF_SIZE/2; - if (copy_from_user(s->workBuf, buffer, count*2)) { - return -1; - } - for (i=0; i< count/2; i++) { - *(u16*)(db->lbuf + bufStart + i*2) = - s->workBuf[i].lchannel; - *(u16*)(db->rbuf + bufStart + i*2) = - s->workBuf[i].rchannel; - } - - copyCount -= count; - bufStart += count; - ASSERT(bufStart <= db->fragTotalSize); - buffer += count *2; - } - return 0; - -} - -/* return the total bytes that is copied */ -static inline int -copy_dac_from_user(struct vrc5477_ac97_state *s, - const char *buffer, - size_t count, - int avail) -{ - struct dmabuf *db = &s->dma_dac; - int copyCount=0; - int copyFragCount=0; - int totalCopyCount = 0; - int totalCopyFragCount = 0; - unsigned long flags; -#if defined(VRC5477_AC97_VERBOSE_DEBUG) - int i; -#endif - - /* adjust count to signel channel byte count */ - count >>= s->dacChannels - 1; - - /* we may have to "copy" twice as ring buffer wraps around */ - for (; (avail > 0) && (count > 0); ) { - /* determine max possible copy count for single channel */ - copyCount = count; - if (copyCount > avail) { - copyCount = avail; - } - if (copyCount + db->nextIn > db->fragTotalSize) { - copyCount = db->fragTotalSize - db->nextIn; - ASSERT(copyCount > 0); - } - - copyFragCount = copyCount; - ASSERT(copyFragCount >= copyCount); - - /* we copy differently based on the number channels */ - if (s->dacChannels == 1) { - if (copy_from_user(db->lbuf + db->nextIn, - buffer, - copyCount)) - return -1; - /* fill gaps with 0 */ - memset(db->lbuf + db->nextIn + copyCount, - 0, - copyFragCount - copyCount); - } else { - /* we have demux the stream into two separate ones */ - if (copy_two_channel_dac_from_user(s, buffer, copyCount)) - return -1; - /* fill gaps with 0 */ - memset(db->lbuf + db->nextIn + copyCount, - 0, - copyFragCount - copyCount); - memset(db->rbuf + db->nextIn + copyCount, - 0, - copyFragCount - copyCount); - } - -#if defined(VRC5477_AC97_VERBOSE_DEBUG) - for (i=0; i< copyFragCount; i+= db->fragSize) { - *(u16*)(db->lbuf + db->nextIn + i) = inTicket ++; - } -#endif - - count -= copyCount; - totalCopyCount += copyCount; - avail -= copyFragCount; - totalCopyFragCount += copyFragCount; - - buffer += copyCount << (s->dacChannels - 1); - - db->nextIn += copyFragCount; - if (db->nextIn >= db->fragTotalSize) { - ASSERT(db->nextIn == db->fragTotalSize); - db->nextIn = 0; - } - - ASSERT( (count == 0) || (copyCount == copyFragCount)); - } - - spin_lock_irqsave(&s->lock, flags); - db->count += totalCopyFragCount; - if (db->stopped) { - start_dac(s); - } - - /* nextIn should not be equal to nextOut unless we are full */ - ASSERT( ( (db->count == db->fragTotalSize) && - (db->nextIn == db->nextOut) ) || - ( (db->count < db->fragTotalSize) && - (db->nextIn != db->nextOut) ) ); - - spin_unlock_irqrestore(&s->lock, flags); - - return totalCopyCount << (s->dacChannels-1); - -} - -static ssize_t vrc5477_ac97_write(struct file *file, const char *buffer, - size_t count, loff_t *ppos) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)file->private_data; - struct dmabuf *db = &s->dma_dac; - ssize_t ret; - unsigned long flags; - int copyCount, avail; - - if (!access_ok(VERIFY_READ, buffer, count)) - return -EFAULT; - ret = 0; - - while (count > 0) { - // wait for space in playback buffer - do { - spin_lock_irqsave(&s->lock, flags); - avail = db->fragTotalSize - db->count; - spin_unlock_irqrestore(&s->lock, flags); - if (avail <= 0) { - if (file->f_flags & O_NONBLOCK) { - if (!ret) - ret = -EAGAIN; - return ret; - } - interruptible_sleep_on(&db->wait); - if (signal_pending(current)) { - if (!ret) - ret = -ERESTARTSYS; - return ret; - } - } - } while (avail <= 0); - - copyCount = copy_dac_from_user(s, buffer, count, avail); - if (copyCount < 0) { - if (!ret) ret = -EFAULT; - return ret; - } - - count -= copyCount; - buffer += copyCount; - ret += copyCount; - } // while (count > 0) - - return ret; -} - -/* No kernel lock - we have our own spinlock */ -static unsigned int vrc5477_ac97_poll(struct file *file, - struct poll_table_struct *wait) -{ - struct vrc5477_ac97_state *s = (struct vrc5477_ac97_state *)file->private_data; - unsigned long flags; - unsigned int mask = 0; - - if (file->f_mode & FMODE_WRITE) - poll_wait(file, &s->dma_dac.wait, wait); - if (file->f_mode & FMODE_READ) - poll_wait(file, &s->dma_adc.wait, wait); - spin_lock_irqsave(&s->lock, flags); - if (file->f_mode & FMODE_READ) { - if (s->dma_adc.count >= (signed)s->dma_adc.fragSize) - mask |= POLLIN | POLLRDNORM; - } - if (file->f_mode & FMODE_WRITE) { - if ((signed)s->dma_dac.fragTotalSize >= - s->dma_dac.count + (signed)s->dma_dac.fragSize) - mask |= POLLOUT | POLLWRNORM; - } - spin_unlock_irqrestore(&s->lock, flags); - return mask; -} - -#ifdef VRC5477_AC97_DEBUG -static struct ioctl_str_t { - unsigned int cmd; - const char* str; -} ioctl_str[] = { - {SNDCTL_DSP_RESET, "SNDCTL_DSP_RESET"}, - {SNDCTL_DSP_SYNC, "SNDCTL_DSP_SYNC"}, - {SNDCTL_DSP_SPEED, "SNDCTL_DSP_SPEED"}, - {SNDCTL_DSP_STEREO, "SNDCTL_DSP_STEREO"}, - {SNDCTL_DSP_GETBLKSIZE, "SNDCTL_DSP_GETBLKSIZE"}, - {SNDCTL_DSP_SETFMT, "SNDCTL_DSP_SETFMT"}, - {SNDCTL_DSP_SAMPLESIZE, "SNDCTL_DSP_SAMPLESIZE"}, - {SNDCTL_DSP_CHANNELS, "SNDCTL_DSP_CHANNELS"}, - {SOUND_PCM_WRITE_CHANNELS, "SOUND_PCM_WRITE_CHANNELS"}, - {SOUND_PCM_WRITE_FILTER, "SOUND_PCM_WRITE_FILTER"}, - {SNDCTL_DSP_POST, "SNDCTL_DSP_POST"}, - {SNDCTL_DSP_SUBDIVIDE, "SNDCTL_DSP_SUBDIVIDE"}, - {SNDCTL_DSP_SETFRAGMENT, "SNDCTL_DSP_SETFRAGMENT"}, - {SNDCTL_DSP_GETFMTS, "SNDCTL_DSP_GETFMTS"}, - {SNDCTL_DSP_GETOSPACE, "SNDCTL_DSP_GETOSPACE"}, - {SNDCTL_DSP_GETISPACE, "SNDCTL_DSP_GETISPACE"}, - {SNDCTL_DSP_NONBLOCK, "SNDCTL_DSP_NONBLOCK"}, - {SNDCTL_DSP_GETCAPS, "SNDCTL_DSP_GETCAPS"}, - {SNDCTL_DSP_GETTRIGGER, "SNDCTL_DSP_GETTRIGGER"}, - {SNDCTL_DSP_SETTRIGGER, "SNDCTL_DSP_SETTRIGGER"}, - {SNDCTL_DSP_GETIPTR, "SNDCTL_DSP_GETIPTR"}, - {SNDCTL_DSP_GETOPTR, "SNDCTL_DSP_GETOPTR"}, - {SNDCTL_DSP_MAPINBUF, "SNDCTL_DSP_MAPINBUF"}, - {SNDCTL_DSP_MAPOUTBUF, "SNDCTL_DSP_MAPOUTBUF"}, - {SNDCTL_DSP_SETSYNCRO, "SNDCTL_DSP_SETSYNCRO"}, - {SNDCTL_DSP_SETDUPLEX, "SNDCTL_DSP_SETDUPLEX"}, - {SNDCTL_DSP_GETODELAY, "SNDCTL_DSP_GETODELAY"}, - {SNDCTL_DSP_GETCHANNELMASK, "SNDCTL_DSP_GETCHANNELMASK"}, - {SNDCTL_DSP_BIND_CHANNEL, "SNDCTL_DSP_BIND_CHANNEL"}, - {OSS_GETVERSION, "OSS_GETVERSION"}, - {SOUND_PCM_READ_RATE, "SOUND_PCM_READ_RATE"}, - {SOUND_PCM_READ_CHANNELS, "SOUND_PCM_READ_CHANNELS"}, - {SOUND_PCM_READ_BITS, "SOUND_PCM_READ_BITS"}, - {SOUND_PCM_READ_FILTER, "SOUND_PCM_READ_FILTER"} -}; -#endif - -static int vrc5477_ac97_ioctl(struct inode *inode, struct file *file, - unsigned int cmd, unsigned long arg) -{ - struct vrc5477_ac97_state *s = (struct vrc5477_ac97_state *)file->private_data; - unsigned long flags; - audio_buf_info abinfo; - int count; - int val, ret; - -#ifdef VRC5477_AC97_DEBUG - for (count = 0; count < ARRAY_SIZE(ioctl_str); count++) { - if (ioctl_str[count].cmd == cmd) - break; - } - if (count < ARRAY_SIZE(ioctl_str)) - printk(KERN_INFO PFX "ioctl %s\n", ioctl_str[count].str); - else - printk(KERN_INFO PFX "ioctl unknown, 0x%x\n", cmd); -#endif - - switch (cmd) { - case OSS_GETVERSION: - return put_user(SOUND_VERSION, (int *)arg); - - case SNDCTL_DSP_SYNC: - if (file->f_mode & FMODE_WRITE) - return drain_dac(s, file->f_flags & O_NONBLOCK); - return 0; - - case SNDCTL_DSP_SETDUPLEX: - return 0; - - case SNDCTL_DSP_GETCAPS: - return put_user(DSP_CAP_DUPLEX, (int *)arg); - - case SNDCTL_DSP_RESET: - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - synchronize_irq(s->irq); - s->dma_dac.count = 0; - s->dma_dac.nextIn = s->dma_dac.nextOut = 0; - } - if (file->f_mode & FMODE_READ) { - stop_adc(s); - synchronize_irq(s->irq); - s->dma_adc.count = 0; - s->dma_adc.nextIn = s->dma_adc.nextOut = 0; - } - return 0; - - case SNDCTL_DSP_SPEED: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val >= 0) { - if (file->f_mode & FMODE_READ) { - stop_adc(s); - set_adc_rate(s, val); - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - set_dac_rate(s, val); - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } - return put_user((file->f_mode & FMODE_READ) ? - s->adcRate : s->dacRate, (int *)arg); - - case SNDCTL_DSP_STEREO: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - if (val) - s->adcChannels = 2; - else - s->adcChannels = 1; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - if (val) - s->dacChannels = 2; - else - s->dacChannels = 1; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - return 0; - - case SNDCTL_DSP_CHANNELS: - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val != 0) { - if ( (val != 1) && (val != 2)) val = 2; - - if (file->f_mode & FMODE_READ) { - stop_adc(s); - s->dacChannels = val; - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - s->dacChannels = val; - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } - return put_user(val, (int *)arg); - - case SNDCTL_DSP_GETFMTS: /* Returns a mask */ - return put_user(AFMT_S16_LE, (int *)arg); - - case SNDCTL_DSP_SETFMT: /* Selects ONE fmt*/ - if (get_user(val, (int *)arg)) - return -EFAULT; - if (val != AFMT_QUERY) { - if (val != AFMT_S16_LE) return -EINVAL; - if (file->f_mode & FMODE_READ) { - stop_adc(s); - if ((ret = prog_dmabuf_adc(s))) - return ret; - } - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - if ((ret = prog_dmabuf_dac(s))) - return ret; - } - } else { - val = AFMT_S16_LE; - } - return put_user(val, (int *)arg); - - case SNDCTL_DSP_POST: - return 0; - - case SNDCTL_DSP_GETTRIGGER: - case SNDCTL_DSP_SETTRIGGER: - /* NO trigger */ - return -EINVAL; - - case SNDCTL_DSP_GETOSPACE: - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - abinfo.fragsize = s->dma_dac.fragSize << (s->dacChannels-1); - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - spin_unlock_irqrestore(&s->lock, flags); - abinfo.bytes = (s->dma_dac.fragTotalSize - count) << - (s->dacChannels-1); - abinfo.fragstotal = s->dma_dac.numFrag; - abinfo.fragments = abinfo.bytes >> s->dma_dac.fragShift >> - (s->dacChannels-1); - return copy_to_user((void *)arg, &abinfo, sizeof(abinfo)) ? -EFAULT : 0; - - case SNDCTL_DSP_GETISPACE: - if (!(file->f_mode & FMODE_READ)) - return -EINVAL; - abinfo.fragsize = s->dma_adc.fragSize << (s->adcChannels-1); - spin_lock_irqsave(&s->lock, flags); - count = s->dma_adc.count; - spin_unlock_irqrestore(&s->lock, flags); - if (count < 0) - count = 0; - abinfo.bytes = count << (s->adcChannels-1); - abinfo.fragstotal = s->dma_adc.numFrag; - abinfo.fragments = (abinfo.bytes >> s->dma_adc.fragShift) >> - (s->adcChannels-1); - return copy_to_user((void *)arg, &abinfo, sizeof(abinfo)) ? -EFAULT : 0; - - case SNDCTL_DSP_NONBLOCK: - file->f_flags |= O_NONBLOCK; - return 0; - - case SNDCTL_DSP_GETODELAY: - if (!(file->f_mode & FMODE_WRITE)) - return -EINVAL; - spin_lock_irqsave(&s->lock, flags); - count = s->dma_dac.count; - spin_unlock_irqrestore(&s->lock, flags); - return put_user(count, (int *)arg); - - case SNDCTL_DSP_GETIPTR: - case SNDCTL_DSP_GETOPTR: - /* we cannot get DMA ptr */ - return -EINVAL; - - case SNDCTL_DSP_GETBLKSIZE: - if (file->f_mode & FMODE_WRITE) - return put_user(s->dma_dac.fragSize << (s->dacChannels-1), (int *)arg); - else - return put_user(s->dma_adc.fragSize << (s->adcChannels-1), (int *)arg); - - case SNDCTL_DSP_SETFRAGMENT: - /* we ignore fragment size request */ - return 0; - - case SNDCTL_DSP_SUBDIVIDE: - /* what is this for? [jsun] */ - return 0; - - case SOUND_PCM_READ_RATE: - return put_user((file->f_mode & FMODE_READ) ? - s->adcRate : s->dacRate, (int *)arg); - - case SOUND_PCM_READ_CHANNELS: - if (file->f_mode & FMODE_READ) - return put_user(s->adcChannels, (int *)arg); - else - return put_user(s->dacChannels ? 2 : 1, (int *)arg); - - case SOUND_PCM_READ_BITS: - return put_user(16, (int *)arg); - - case SOUND_PCM_WRITE_FILTER: - case SNDCTL_DSP_SETSYNCRO: - case SOUND_PCM_READ_FILTER: - return -EINVAL; - } - - return mixdev_ioctl(s->codec, cmd, arg); -} - - -static int vrc5477_ac97_open(struct inode *inode, struct file *file) -{ - int minor = iminor(inode); - DECLARE_WAITQUEUE(wait, current); - unsigned long flags; - struct list_head *list; - struct vrc5477_ac97_state *s; - int ret=0; - - nonseekable_open(inode, file); - for (list = devs.next; ; list = list->next) { - if (list == &devs) - return -ENODEV; - s = list_entry(list, struct vrc5477_ac97_state, devs); - if (!((s->dev_audio ^ minor) & ~0xf)) - break; - } - file->private_data = s; - - /* wait for device to become free */ - mutex_lock(&s->open_mutex); - while (s->open_mode & file->f_mode) { - - if (file->f_flags & O_NONBLOCK) { - mutex_unlock(&s->open_mutex); - return -EBUSY; - } - add_wait_queue(&s->open_wait, &wait); - __set_current_state(TASK_INTERRUPTIBLE); - mutex_unlock(&s->open_mutex); - schedule(); - remove_wait_queue(&s->open_wait, &wait); - set_current_state(TASK_RUNNING); - if (signal_pending(current)) - return -ERESTARTSYS; - mutex_lock(&s->open_mutex); - } - - spin_lock_irqsave(&s->lock, flags); - - if (file->f_mode & FMODE_READ) { - /* set default settings */ - set_adc_rate(s, 48000); - s->adcChannels = 2; - - ret = prog_dmabuf_adc(s); - if (ret) goto bailout; - } - if (file->f_mode & FMODE_WRITE) { - /* set default settings */ - set_dac_rate(s, 48000); - s->dacChannels = 2; - - ret = prog_dmabuf_dac(s); - if (ret) goto bailout; - } - - s->open_mode |= file->f_mode & (FMODE_READ | FMODE_WRITE); - - bailout: - spin_unlock_irqrestore(&s->lock, flags); - - mutex_unlock(&s->open_mutex); - return ret; -} - -static int vrc5477_ac97_release(struct inode *inode, struct file *file) -{ - struct vrc5477_ac97_state *s = - (struct vrc5477_ac97_state *)file->private_data; - - lock_kernel(); - if (file->f_mode & FMODE_WRITE) - drain_dac(s, file->f_flags & O_NONBLOCK); - mutex_lock(&s->open_mutex); - if (file->f_mode & FMODE_WRITE) { - stop_dac(s); - dealloc_dmabuf(s, &s->dma_dac); - } - if (file->f_mode & FMODE_READ) { - stop_adc(s); - dealloc_dmabuf(s, &s->dma_adc); - } - s->open_mode &= (~file->f_mode) & (FMODE_READ|FMODE_WRITE); - mutex_unlock(&s->open_mutex); - wake_up(&s->open_wait); - unlock_kernel(); - return 0; -} - -static /*const*/ struct file_operations vrc5477_ac97_audio_fops = { - .owner = THIS_MODULE, - .llseek = no_llseek, - .read = vrc5477_ac97_read, - .write = vrc5477_ac97_write, - .poll = vrc5477_ac97_poll, - .ioctl = vrc5477_ac97_ioctl, - // .mmap = vrc5477_ac97_mmap, - .open = vrc5477_ac97_open, - .release = vrc5477_ac97_release, -}; - - -/* --------------------------------------------------------------------- */ - - -/* --------------------------------------------------------------------- */ - -/* - * for debugging purposes, we'll create a proc device that dumps the - * CODEC chipstate - */ - -#ifdef VRC5477_AC97_DEBUG - -struct { - const char *regname; - unsigned regaddr; -} vrc5477_ac97_regs[] = { - {"VRC5477_INT_STATUS", VRC5477_INT_STATUS}, - {"VRC5477_CODEC_WR", VRC5477_CODEC_WR}, - {"VRC5477_CODEC_RD", VRC5477_CODEC_RD}, - {"VRC5477_CTRL", VRC5477_CTRL}, - {"VRC5477_ACLINK_CTRL", VRC5477_ACLINK_CTRL}, - {"VRC5477_INT_MASK", VRC5477_INT_MASK}, - {"VRC5477_DAC1_CTRL", VRC5477_DAC1_CTRL}, - {"VRC5477_DAC1L", VRC5477_DAC1L}, - {"VRC5477_DAC1_BADDR", VRC5477_DAC1_BADDR}, - {"VRC5477_DAC2_CTRL", VRC5477_DAC2_CTRL}, - {"VRC5477_DAC2L", VRC5477_DAC2L}, - {"VRC5477_DAC2_BADDR", VRC5477_DAC2_BADDR}, - {"VRC5477_DAC3_CTRL", VRC5477_DAC3_CTRL}, - {"VRC5477_DAC3L", VRC5477_DAC3L}, - {"VRC5477_DAC3_BADDR", VRC5477_DAC3_BADDR}, - {"VRC5477_ADC1_CTRL", VRC5477_ADC1_CTRL}, - {"VRC5477_ADC1L", VRC5477_ADC1L}, - {"VRC5477_ADC1_BADDR", VRC5477_ADC1_BADDR}, - {"VRC5477_ADC2_CTRL", VRC5477_ADC2_CTRL}, - {"VRC5477_ADC2L", VRC5477_ADC2L}, - {"VRC5477_ADC2_BADDR", VRC5477_ADC2_BADDR}, - {"VRC5477_ADC3_CTRL", VRC5477_ADC3_CTRL}, - {"VRC5477_ADC3L", VRC5477_ADC3L}, - {"VRC5477_ADC3_BADDR", VRC5477_ADC3_BADDR}, - {NULL, 0x0} -}; - -static int proc_vrc5477_ac97_dump (char *buf, char **start, off_t fpos, - int length, int *eof, void *data) -{ - struct vrc5477_ac97_state *s; - int cnt, len = 0; - - if (list_empty(&devs)) - return 0; - s = list_entry(devs.next, struct vrc5477_ac97_state, devs); - - /* print out header */ - len += sprintf(buf + len, "\n\t\tVrc5477 Audio Debug\n\n"); - - // print out digital controller state - len += sprintf (buf + len, "NEC Vrc5477 Audio Controller registers\n"); - len += sprintf (buf + len, "---------------------------------\n"); - for (cnt=0; vrc5477_ac97_regs[cnt].regname != NULL; cnt++) { - len+= sprintf (buf + len, "%-20s = %08x\n", - vrc5477_ac97_regs[cnt].regname, - inl(s->io + vrc5477_ac97_regs[cnt].regaddr)); - } - - /* print out driver state */ - len += sprintf (buf + len, "NEC Vrc5477 Audio driver states\n"); - len += sprintf (buf + len, "---------------------------------\n"); - len += sprintf (buf + len, "dacChannels = %d\n", s->dacChannels); - len += sprintf (buf + len, "adcChannels = %d\n", s->adcChannels); - len += sprintf (buf + len, "dacRate = %d\n", s->dacRate); - len += sprintf (buf + len, "adcRate = %d\n", s->adcRate); - - len += sprintf (buf + len, "dma_dac is %s ready\n", - s->dma_dac.ready? "" : "not"); - if (s->dma_dac.ready) { - len += sprintf (buf + len, "dma_dac is %s stopped.\n", - s->dma_dac.stopped? "" : "not"); - len += sprintf (buf + len, "dma_dac.fragSize = %x\n", - s->dma_dac.fragSize); - len += sprintf (buf + len, "dma_dac.fragShift = %x\n", - s->dma_dac.fragShift); - len += sprintf (buf + len, "dma_dac.numFrag = %x\n", - s->dma_dac.numFrag); - len += sprintf (buf + len, "dma_dac.fragTotalSize = %x\n", - s->dma_dac.fragTotalSize); - len += sprintf (buf + len, "dma_dac.nextIn = %x\n", - s->dma_dac.nextIn); - len += sprintf (buf + len, "dma_dac.nextOut = %x\n", - s->dma_dac.nextOut); - len += sprintf (buf + len, "dma_dac.count = %x\n", - s->dma_dac.count); - } - - len += sprintf (buf + len, "dma_adc is %s ready\n", - s->dma_adc.ready? "" : "not"); - if (s->dma_adc.ready) { - len += sprintf (buf + len, "dma_adc is %s stopped.\n", - s->dma_adc.stopped? "" : "not"); - len += sprintf (buf + len, "dma_adc.fragSize = %x\n", - s->dma_adc.fragSize); - len += sprintf (buf + len, "dma_adc.fragShift = %x\n", - s->dma_adc.fragShift); - len += sprintf (buf + len, "dma_adc.numFrag = %x\n", - s->dma_adc.numFrag); - len += sprintf (buf + len, "dma_adc.fragTotalSize = %x\n", - s->dma_adc.fragTotalSize); - len += sprintf (buf + len, "dma_adc.nextIn = %x\n", - s->dma_adc.nextIn); - len += sprintf (buf + len, "dma_adc.nextOut = %x\n", - s->dma_adc.nextOut); - len += sprintf (buf + len, "dma_adc.count = %x\n", - s->dma_adc.count); - } - - /* print out CODEC state */ - len += sprintf (buf + len, "\nAC97 CODEC registers\n"); - len += sprintf (buf + len, "----------------------\n"); - for (cnt=0; cnt <= 0x7e; cnt = cnt +2) - len+= sprintf (buf + len, "reg %02x = %04x\n", - cnt, rdcodec(s->codec, cnt)); - - if (fpos >=len){ - *start = buf; - *eof =1; - return 0; - } - *start = buf + fpos; - if ((len -= fpos) > length) - return length; - *eof =1; - return len; - -} -#endif /* VRC5477_AC97_DEBUG */ - -/* --------------------------------------------------------------------- */ - -/* maximum number of devices; only used for command line params */ -#define NR_DEVICE 5 - -static unsigned int devindex; - -MODULE_AUTHOR("Monta Vista Software, jsun(a)mvista.com or jsun(a)junsun.net"); -MODULE_DESCRIPTION("NEC Vrc5477 audio (AC97) Driver"); -MODULE_LICENSE("GPL"); - -static int __devinit vrc5477_ac97_probe(struct pci_dev *pcidev, - const struct pci_device_id *pciid) -{ - struct vrc5477_ac97_state *s; -#ifdef VRC5477_AC97_DEBUG - char proc_str[80]; -#endif - - if (pcidev->irq == 0) - return -1; - - if (!(s = kzalloc(sizeof(struct vrc5477_ac97_state), GFP_KERNEL))) { - printk(KERN_ERR PFX "alloc of device struct failed\n"); - return -1; - } - - init_waitqueue_head(&s->dma_adc.wait); - init_waitqueue_head(&s->dma_dac.wait); - init_waitqueue_head(&s->open_wait); - mutex_init(&s->open_mutex); - spin_lock_init(&s->lock); - - s->dev = pcidev; - s->io = pci_resource_start(pcidev, 0); - s->irq = pcidev->irq; - - s->codec = ac97_alloc_codec(); - - s->codec->private_data = s; - s->codec->id = 0; - s->codec->codec_read = rdcodec; - s->codec->codec_write = wrcodec; - s->codec->codec_wait = waitcodec; - - /* setting some other default values such as - * adcChannels, adcRate is done in open() so that - * no persistent state across file opens. - */ - - /* test if get response from ac97, if not return */ - if (ac97_codec_not_present(s->codec)) { - printk(KERN_ERR PFX "no ac97 codec\n"); - goto err_region; - - } - - /* test if get response from ac97, if not return */ - if (ac97_codec_not_present(&(s->codec))) { - printk(KERN_ERR PFX "no ac97 codec\n"); - goto err_region; - - } - - if (!request_region(s->io, pci_resource_len(pcidev,0), - VRC5477_AC97_MODULE_NAME)) { - printk(KERN_ERR PFX "io ports %#lx->%#lx in use\n", - s->io, s->io + pci_resource_len(pcidev,0)-1); - goto err_region; - } - if (request_irq(s->irq, vrc5477_ac97_interrupt, IRQF_DISABLED, - VRC5477_AC97_MODULE_NAME, s)) { - printk(KERN_ERR PFX "irq %u in use\n", s->irq); - goto err_irq; - } - - printk(KERN_INFO PFX "IO at %#lx, IRQ %d\n", s->io, s->irq); - - /* register devices */ - if ((s->dev_audio = register_sound_dsp(&vrc5477_ac97_audio_fops, -1)) < 0) - goto err_dev1; - if ((s->codec->dev_mixer = - register_sound_mixer(&vrc5477_ac97_mixer_fops, -1)) < 0) - goto err_dev2; - -#ifdef VRC5477_AC97_DEBUG - /* initialize the debug proc device */ - s->ps = create_proc_read_entry(VRC5477_AC97_MODULE_NAME, 0, NULL, - proc_vrc5477_ac97_dump, NULL); -#endif /* VRC5477_AC97_DEBUG */ - - /* enable pci io and bus mastering */ - if (pci_enable_device(pcidev)) - goto err_dev3; - pci_set_master(pcidev); - - /* cold reset the AC97 */ - outl(VRC5477_ACLINK_CTRL_RST_ON | VRC5477_ACLINK_CTRL_RST_TIME, - s->io + VRC5477_ACLINK_CTRL); - while (inl(s->io + VRC5477_ACLINK_CTRL) & VRC5477_ACLINK_CTRL_RST_ON); - - /* codec init */ - if (!ac97_probe_codec(s->codec)) - goto err_dev3; - -#ifdef VRC5477_AC97_DEBUG - sprintf(proc_str, "driver/%s/%d/ac97", - VRC5477_AC97_MODULE_NAME, s->codec->id); - s->ac97_ps = create_proc_read_entry (proc_str, 0, NULL, - ac97_read_proc, s->codec); - /* TODO : why this proc file does not show up? */ -#endif - - /* Try to enable variable rate audio mode. */ - wrcodec(s->codec, AC97_EXTENDED_STATUS, - rdcodec(s->codec, AC97_EXTENDED_STATUS) | AC97_EXTSTAT_VRA); - /* Did we enable it? */ - if(rdcodec(s->codec, AC97_EXTENDED_STATUS) & AC97_EXTSTAT_VRA) - s->extended_status |= AC97_EXTSTAT_VRA; - else { - s->dacRate = 48000; - printk(KERN_INFO PFX "VRA mode not enabled; rate fixed at %d.", - s->dacRate); - } - - /* let us get the default volumne louder */ - wrcodec(s->codec, 0x2, 0x1010); /* master volume, middle */ - wrcodec(s->codec, 0xc, 0x10); /* phone volume, middle */ - // wrcodec(s->codec, 0xe, 0x10); /* misc volume, middle */ - wrcodec(s->codec, 0x10, 0x8000); /* line-in 2 line-out disable */ - wrcodec(s->codec, 0x18, 0x0707); /* PCM out (line out) middle */ - - - /* by default we select line in the input */ - wrcodec(s->codec, 0x1a, 0x0404); - wrcodec(s->codec, 0x1c, 0x0f0f); - wrcodec(s->codec, 0x1e, 0x07); - - /* enable the master interrupt but disable all others */ - outl(VRC5477_INT_MASK_NMASK, s->io + VRC5477_INT_MASK); - - /* store it in the driver field */ - pci_set_drvdata(pcidev, s); - pcidev->dma_mask = 0xffffffff; - /* put it into driver list */ - list_add_tail(&s->devs, &devs); - /* increment devindex */ - if (devindex < NR_DEVICE-1) - devindex++; - return 0; - - err_dev3: - unregister_sound_mixer(s->codec->dev_mixer); - err_dev2: - unregister_sound_dsp(s->dev_audio); - err_dev1: - printk(KERN_ERR PFX "cannot register misc device\n"); - free_irq(s->irq, s); - err_irq: - release_region(s->io, pci_resource_len(pcidev,0)); - err_region: - ac97_release_codec(codec); - kfree(s); - return -1; -} - -static void __devexit vrc5477_ac97_remove(struct pci_dev *dev) -{ - struct vrc5477_ac97_state *s = pci_get_drvdata(dev); - - if (!s) - return; - list_del(&s->devs); - -#ifdef VRC5477_AC97_DEBUG - if (s->ps) - remove_proc_entry(VRC5477_AC97_MODULE_NAME, NULL); -#endif /* VRC5477_AC97_DEBUG */ - - synchronize_irq(); - free_irq(s->irq, s); - release_region(s->io, pci_resource_len(dev,0)); - unregister_sound_dsp(s->dev_audio); - unregister_sound_mixer(s->codec->dev_mixer); - ac97_release_codec(s->codec); - kfree(s); - pci_set_drvdata(dev, NULL); -} - - -static struct pci_device_id id_table[] = { - { PCI_VENDOR_ID_NEC, PCI_DEVICE_ID_NEC_VRC5477_AC97, - PCI_ANY_ID, PCI_ANY_ID, 0, 0 }, - { 0, } -}; - -MODULE_DEVICE_TABLE(pci, id_table); - -static struct pci_driver vrc5477_ac97_driver = { - .name = VRC5477_AC97_MODULE_NAME, - .id_table = id_table, - .probe = vrc5477_ac97_probe, - .remove = __devexit_p(vrc5477_ac97_remove) -}; - -static int __init init_vrc5477_ac97(void) -{ - printk("Vrc5477 AC97 driver: version v0.2 time " __TIME__ " " __DATE__ " by Jun Sun\n"); - return pci_register_driver(&vrc5477_ac97_driver); -} - -static void __exit cleanup_vrc5477_ac97(void) -{ - printk(KERN_INFO PFX "unloading\n"); - pci_unregister_driver(&vrc5477_ac97_driver); -} - -module_init(init_vrc5477_ac97); -module_exit(cleanup_vrc5477_ac97); - Index: upstream-linus/include/linux/pci_ids.h =================================================================== --- upstream-linus.orig/include/linux/pci_ids.h +++ upstream-linus/include/linux/pci_ids.h @@ -611,7 +611,6 @@ #define PCI_DEVICE_ID_NEC_CBUS_3 0x003b #define PCI_DEVICE_ID_NEC_NAPCCARD 0x003e #define PCI_DEVICE_ID_NEC_PCX2 0x0046 /* PowerVR */ -#define PCI_DEVICE_ID_NEC_NILE4 0x005a #define PCI_DEVICE_ID_NEC_VRC5476 0x009b #define PCI_DEVICE_ID_NEC_VRC4173 0x00a5 #define PCI_DEVICE_ID_NEC_VRC5477_AC97 0x00a6

2 1

[alsa-devel] Gateway MP-6954 laptop, FC6, no sound
by Bill McKie 30 Jul '07

30 Jul '07

I've been running FC6 on a Gateway MP-6954 laptop computer since Oct-2006, and have been hoping that the regular FC6 yum updates will eventually provide an alsa update that will support the MP-6954's sound hardware. So far, that hasn't happened, and my system has never had sound capability. Current system stuff: uname -r 2.6.20-1.2962.fc6 rpm -q -a | grep -i alsa alsa-lib-1.0.14-0.1.rc1.fc6 alsa-lib-devel-1.0.14-0.1.rc1.fc6 alsa-utils-1.0.14-0.2.rc1.fc6 cat /etc/modprobe.conf alias eth0 sky2 alias scsi_hostadapter ahci remove snd-hda-intel { /usr/sbin/alsactl store 0 >/dev/null 2>&1 || : ; }; /sbin/modprobe -r --ignore-remove snd-hda-intel options snd cards_limit=8 alias snd-card-0 snd-hda-intel options snd-hda-intel index=0 alias snd-card-7 snd-usb-audio options snd-usb-audio index=7 alsamixer (v1.0.14rc1) reports: Card: HDA Intel Chip: SigmaTel ID 7634 Some Internet googling showed that Tobin Davis released a source level Sigmatel STAC9250 (7634) test patch 6 (gateway-mp6954.patch.6) that appears to address this problem back in December 2006. http://www.spinics.net/lists/alsa-devel/msg05113.html Is this gateway-mp6954.patch.6 still appropriate to be used on my FC6 system? (My FC6 sys has all the current yum updates applied.) If this patch is appropriate, should I first remove the existing alsa rpms before installing alsa from source with the patch. I.e. should I remove all 3 packages: alsa-lib-1.0.14-0.1.rc1.fc6 alsa-lib-devel-1.0.14-0.1.rc1.fc6 alsa-utils-1.0.14-0.2.rc1.fc6 Will subsequent FC6 yum updates involving alsa then disturb the patched alsa? Thanks much, Bill

2 1

Re: [alsa-devel] USB Audio out PS3
by gt bradley 30 Jul '07

30 Jul '07

I have connected a soundblaster mp3+ via usb to my ps3. it shows up in the sound-config applet and plays stereo just fine using the various test commands. I was using Fedora 7, (which has alsa ver. 1.0.14), and had been compiling bleeding edge kernels, with the sound modules set up as modules. The stereo part was almost plug-and-play, except that to install the USB, I did have to enable the "CONFIG_USB_EHCI_SPLIT_ISO (Experimental) " option in the kernel. I gave up trying to configure the SPDIF to do surround sound (long story). GT

1 0

[alsa-devel] aoss never returns POLLIN
by Mark Hills 29 Jul '07

29 Jul '07

Hi, When using the aoss wrapper, poll() is never woken by POLLIN. This happens with devices opened in either O_RDWR or O_RDONLY and is demonstrated by the trivial example below, which never prints "POLLIN". The example performs correctly using the kernel OSS emulation. Is this a (known) bug in aoss? I've now seen it on different distributions and in the latest code from the repository. Enabling ALSA_OSS_DEBUG shows that the struct pollfd is being correctly manipulated: Orig enter POLL nfds: 1, timeout: 1000 fd=3, events=d, revents=0 Changed enter POLL nfds: 2, timeout: 1000 fd=5, events=2c, revents=0 fd=6, events=29, revents=0 Changed exit POLL nfds: 2, timeout: 1000 fd=5, events=2c, revents=4 fd=6, events=29, revents=0 Orig exit POLL nfds: 1, timeout: 1000 fd=3, events=d, revents=4 The file descriptors are: lrwx------ 1 mark users 64 Jul 29 00:46 /proc/5399/fd/3 -> /dev/null lrwx------ 1 mark users 64 Jul 29 00:46 /proc/5399/fd/5 -> /dev/snd/pcmC0D0p lrwx------ 1 mark users 64 Jul 29 00:46 /proc/5399/fd/6 -> /dev/snd/pcmC0D0c The pass through to the _poll() call inside poll_with_pcm() in alsa-oss.c isn't being woken by POLLIN for the capture device. Mark #include <errno.h> #include <fcntl.h> #include <stdio.h> #include <unistd.h> #include <sys/ioctl.h> #include <sys/poll.h> #include <sys/soundcard.h> #define CHANNELS 2 #define SAMPLES 64 int main(int argc, char *argv[]) { int fd, p, r; struct pollfd pe; char buf[CHANNELS * SAMPLES]; /* Change below to O_RDWR, O_RDONLY or O_WRONLY */ fd = open("/dev/dsp", O_RDWR, 0); if(fd == -1) { perror("open"); return -1; } p = AFMT_S16_LE; if(ioctl(fd, SNDCTL_DSP_SETFMT, &p) == -1) { perror("SNDCTL_DSP_SETFMT"); return -1; } p = CHANNELS; if(ioctl(fd, SNDCTL_DSP_CHANNELS, &p) == -1) { perror("SNDCTL_DSP_CHANNELS"); return -1; } p = 44100; if(ioctl(fd, SNDCTL_DSP_SPEED, &p) == -1) { perror("SNDCTL_DSP_SPEED"); return -1; } if(fcntl(fd, F_SETFL, O_NONBLOCK) == -1) { perror("fcntl"); return -1; } pe.fd = fd; pe.events = POLLERR | POLLIN | POLLOUT; for(;;) { fputs("poll", stderr); pe.revents = 0; poll(&pe, 1, 1000); if(pe.revents & POLLOUT) { fputs(" POLLOUT", stderr); r = write(fd, buf, CHANNELS * SAMPLES); if(r == -1 && errno != EAGAIN) { perror("read"); return -1; } } if(pe.revents & POLLIN) { fputs(" POLLIN", stderr); r = read(fd, buf, CHANNELS * SAMPLES); if(r == -1 && errno != EAGAIN) { perror("read"); return -1; } } fputc('\n', stderr); } close(fd); return 0; }

1 0

[alsa-devel] USB Audio out PS3
by midtown123＠gmail.com 29 Jul '07

29 Jul '07

Is there a way I can get audio out from the USB on my ps3? Thanks Martin

1 0

[alsa-devel] IDing stac9200 vs 92HD001 vs 92HD005, etc.
by Brian Hinz 29 Jul '07

29 Jul '07

Hi All, Is there any way (other than visually ID'ing) of differentiating these chips? They all have the same device ID (0x8384, 0x7690)! Thanks, -brian

1 0

[alsa-devel] snd-hda-intel & ALC268 codec
by Meelis Roos 28 Jul '07

28 Jul '07

Hi, I have a laptop with Realtek ALC268 codec connected to Intel HDA controller. The laptop is Compal OEM, specific module currently unknown (it's assembled by a local reseller as Ordi Enduro 7300BF). 2.6.22 didn't much understand the codec. 2.6.23-rc1 seems to know much more about it and it almost works - but not quite. On bootup, the following line is in dmesg: hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... It initializes fine but the sound is so quiet that I can only hear it with active speakers when I turn the volume to max. No mixer sliders can make it louder. Additionally, there are some strange mixer switches like Off-hook and Caller ID. The only strings identifying the base board are from dmidecode: Handle 0x0002, DMI type 2, 8 bytes Base Board Information Manufacturer: COMPAL Product Name: IFT00 Version: REFERENCE Serial Number: N/A Additionally, the system has identified most PCI devices by subsystem vendor & ID 14c0:0023, like this: 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 14c0:0023 Flags: bus master, fast devsel, latency 0, IRQ 23 Memory at f8400000 (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 ls -lR /proc/asound/Intel/: /proc/asound/Intel/: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 codec#0 -r--r--r-- 1 root root 0 2007-07-28 17:26 codec#1 -r--r--r-- 1 root root 0 2007-07-28 17:26 id -rw-r--r-- 1 root root 0 2007-07-28 17:26 oss_mixer dr-xr-xr-x 4 root root 0 2007-07-28 17:26 pcm0c dr-xr-xr-x 3 root root 0 2007-07-28 17:26 pcm0p dr-xr-xr-x 4 root root 0 2007-07-28 17:26 pcm2c dr-xr-xr-x 3 root root 0 2007-07-28 17:26 pcm6c dr-xr-xr-x 3 root root 0 2007-07-28 17:26 pcm6p /proc/asound/Intel/pcm0c: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 oss dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub1 /proc/asound/Intel/pcm0c/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm0c/sub1: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm0p: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 oss dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 /proc/asound/Intel/pcm0p/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm2c: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub1 /proc/asound/Intel/pcm2c/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm2c/sub1: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm6c: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 /proc/asound/Intel/pcm6c/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params /proc/asound/Intel/pcm6p: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 info dr-xr-xr-x 2 root root 0 2007-07-28 17:26 sub0 /proc/asound/Intel/pcm6p/sub0: total 0 -r--r--r-- 1 root root 0 2007-07-28 17:26 hw_params -r--r--r-- 1 root root 0 2007-07-28 17:26 info -rw-r--r-- 1 root root 0 2007-07-28 17:26 prealloc -r--r--r-- 1 root root 0 2007-07-28 17:26 prealloc_max -r--r--r-- 1 root root 0 2007-07-28 17:26 status -r--r--r-- 1 root root 0 2007-07-28 17:26 sw_params amixer controls gives the following output: numid=4,iface=MIXER,name='Headphone Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Headphone Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=45,45 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=13,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=255,255 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=2,iface=MIXER,name='Front Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Front Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=64,step=0 : values=52,52 | dBscale-min=-64.00dB,step=1.00dB,mute=0 numid=6,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=8,iface=MIXER,name='Capture Switch',index=1 ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=5,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=1,1 | dBscale-min=-16.50dB,step=1.50dB,mute=0 numid=7,iface=MIXER,name='Capture Volume',index=1 ; type=INTEGER,access=rw---R--,values=2,min=0,max=31,step=0 : values=1,1 | dBscale-min=-16.50dB,step=1.50dB,mute=0 numid=12,iface=MIXER,name='Caller ID Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=14,iface=MIXER,name='Digital Capture Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=120,step=0 : values=60,60 | dBscale-min=-30.00dB,step=0.50dB,mute=0 numid=9,iface=MIXER,name='Input Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Mic' ; Item #1 'Front Mic' : values=0 numid=10,iface=MIXER,name='Input Source',index=1 ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Mic' ; Item #1 'Front Mic' : values=0 numid=11,iface=MIXER,name='Off-hook Switch' ; type=BOOLEAN,access=rw------,values=1 : values=off -- Meelis Roos (mroos(a)linux.ee)

1 0

[alsa-devel] Mixing.
by Vasily Khoruzhick 28 Jul '07

28 Jul '07

Hi 2 all. I've just read many discussions about kernelspace and userspace sound mixing. Both sides have their arguments, etc. But why not to implement both? If you don't want to rewrite something there's easy way: just create a driver for dummy sound card, that will perform kernelspace mixing and will use default sound card for sound output. Kernelspace mixing will be very usefull for example for oss kernelspace oss emulation (as you know, userspace one doesn't work for huge amount of apps), or for dumb apps, that open hw:0 device for sound and have no option to change output device. Please, think about your users ;) Let them decide which mixing they wants Cheers, Vasily.

1 0

[alsa-devel] ALSA support for Teak DSP used in NXP LPC3180
by Jon Smirl 27 Jul '07

27 Jul '07

Is there an Alsa driver for the Teak DSP used in some ARM chips like the NXP LPC3180? I've been googling around and can't locate one. I have a GPL'd OSS driver if anyone is interested. It's 3,581 lines. -- Jon Smirl jonsmirl(a)gmail.com

1 0

[alsa-devel] Intel HDA Driver - Bug
by Steven King 27 Jul '07

27 Jul '07

Hi, I have been trying to get the sound on my Dell Inspiron 1720 laptop running under Debian Etch (STAC 9205). Originally, I could only get sound through the main speakers, there was no headphone sound or mic. I downloaded the latest alsa-drivers package (1.0.14), compiled it and installed it. I received the following messages (and the device would not register itself): hda_intel: azx_get_response timeout, switching to polling mode... hda_intel: azx_get_response timeout, switching to single_cmd mode... I started digging into the driver code and found an issue with one of the routines going outside an array. The offending code is (all in patch_sigmatel): static void stac92xx_set_config_regs(struct hda_codec *codec) { // ... for (i = 0; i < spec->num_pins; i++) { // ... } And in: static int patch_stac9205(struct hda_codec *codec) { // .. spec->num_pins = 14 ; //ERROR // .. } However, there are only 12 pins with 12 corresponding NIDs and default configuration so in stac92xx_set_config_regs it accesses spec->pin_configs[i] and spec->pin_nids[i] outside of bounds. Fix: Add a define (STAC_9205_MAX_PINS = 12) and use for array defines and num_pins. The driver loads now, but I still only have sound out the main speaker. If I get rid of the configuration sent to the chipset altogether. I get sound through Headphone and Speakers (with detect working on HP), but still no MIC. How can I add support for my Dell as I don't know the configuration to use or am I missing something here? Thanks, Steve

2 4

[alsa-devel] BT87x ALSA driver patches
by Trent Piepho 27 Jul '07

27 Jul '07

On Mon, 23 Jul 2007, Mauro Carvalho Chehab wrote: > Em Sáb, 2007-07-21 às 17:36 -0700, Trent Piepho escreveu: > > 01/01: bttv: Fix Viewcast Osprey 440 support > > http://linuxtv.org/hg/~tap/v4l-dvb?cmd=changeset;node=8e72fe5675b3 > > > > I have two other changes that only affect the ALSA driver part. Should I > > send those to ALSA or to you? > > Please send to ALSA guys, c/c me. Ok, I'll do that. What about the two lines to bt87x.c that this patch added? I seemed logical to me that they should be part of the same patch at git as the changes to bttv-cards.c, but I suppose it's not 100% necessary to do so, the kernel will still compile and not crash. Do you plan to send this to git as one patch, or to strip out the ALSA change? I'm just wondering if I should send these two lines to the ALSA people as a separate patch, of if they will get them into their hg via a kernel-sync patch. My bt87x patches depend on this one, so somehow the ALSA hg needs to get this change before I can send them the next two. The problem really, is that having two divers for the same chip split into v4l-dvb and ALSA like this a pain. IMHO, it would make a lot more sense to put the bttv alsa driver into v4l-dvb, like the saa7134 alsa and oss and cx88 alsa drivers are. > > drivers/media/video/bt8xx/bttv-cards.c | 240 ++++++++++++++++++--------------- > > sound/pci/bt87x.c | 2 > > 2 files changed, 137 insertions(+), 105 deletions(-)

3 4

[alsa-devel] another device for opl3sa2
by Paul Vojta 27 Jul '07

27 Jul '07

Folks: Here's a patch to add support for an (admittedly old) device: --- linux-2.6.21.5/sound/isa/opl3sa2.c.orig 2007-06-11 11:37:06.000000000 -0700 +++ linux-2.6.21.5/sound/isa/opl3sa2.c 2007-06-30 23:29:43.000000000 -0700 @@ -165,6 +165,8 @@ { .id = "YMH0801", .devs = { { "YMH0021" } } }, /* NeoMagic MagicWave 3DX */ { .id = "NMX2200", .devs = { { "YMH2210" } } }, + /* NeoMagic MagicWave 3D */ + { .id = "NMX2200", .devs = { { "NMX2210" } } }, /* --- */ { .id = "" } /* end */ }; (There may be a bug in the initialization code, though -- it doesn't work the first time after booting, just the second and subsequent times. For example, the first time I try running "aplay file.wav" it'll sit there silently instead of playing the file, but if I type "aplay file.wav" again then it'll play.) I don't know whether snd_opl3sa2_pnpbiosids[] should also be changed. Also, why does nm256.c direct users to the cs423x drivers and not opl3sa2? If it had mentioned opl3sa2 it would have saved me many hours. Sincerely, Paul Vojta vojta(a)math.berkeley.edu

2 12

[alsa-devel] [PATCH] snd_usb_caiaq_input_free() fix
by Karsten Wiese 27 Jul '07

27 Jul '07

input_free_device()'s comment says: input_free_device() should only be used if input_register_device() was not called yet or if it failed. Once device was registered use input_unregister_device() and memory will be freed once last refrence to the device is dropped. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- diff -pur rt9/sound/usb/caiaq/caiaq-input.c rt9-kw/sound/usb/caiaq/caiaq-input.c --- rt9/sound/usb/caiaq/caiaq-input.c 2007-07-10 20:56:30.000000000 +0200 +++ rt9-kw/sound/usb/caiaq/caiaq-input.c 2007-07-27 11:45:52.000000000 +0200 @@ -238,7 +238,6 @@ void snd_usb_caiaq_input_free(struct snd return; input_unregister_device(dev->input_dev); - input_free_device(dev->input_dev); dev->input_dev = NULL; }

2 1

[alsa-devel] [PATCH] snd_usb_caiaq_input_free() fix
by Karsten Wiese 27 Jul '07

27 Jul '07

input_free_device()'s comment says: input_free_device() should only be used if input_register_device() was not called yet or if it failed. Once device was registered use input_unregister_device() and memory will be freed once last refrence to the device is dropped. Signed-off-by: Karsten Wiese <fzu(a)wemgehoertderstaat.de> --- diff -pur rt9/sound/usb/caiaq/caiaq-input.c rt9-kw/sound/usb/caiaq/caiaq-input.c --- rt9/sound/usb/caiaq/caiaq-input.c 2007-07-10 20:56:30.000000000 +0200 +++ rt9-kw/sound/usb/caiaq/caiaq-input.c 2007-07-27 11:45:52.000000000 +0200 @@ -238,7 +238,6 @@ void snd_usb_caiaq_input_free(struct snd return; input_unregister_device(dev->input_dev); - input_free_device(dev->input_dev); dev->input_dev = NULL; }

1 0

[alsa-devel] Arecord not working
by Pharaoh . 27 Jul '07

27 Jul '07

Hi The arecord says following: arecord -f cd -t wav -c 2 -D dsnoop 16bit.raw ALSA lib confmisc.c:670:(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:391:(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:1070:(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:3968:(snd_config_expand) Evaluate error: No such device ALSA lib pcm.c:2143:(snd_pcm_open_noupdate) Unknown PCM dsnoop arecord: main:550: audio open error: No such device #arecord -L arecord -L PCM list: hw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type hw card $CARD device $DEV subdevice $SUBDEV } plughw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type plug slave.pcm { type hw card $CARD device $DEV subdevice $SUBDEV } } plug { @args.0 SLAVE @args.SLAVE { type string } type plug slave.pcm $SLAVE } shm { @args.0 SOCKET @args.1 PCM @args.SOCKET { type string } @args.PCM { type string } type shm server $SOCKET pcm $PCM } tee { @args.0 SLAVE @args.1 FILE @args.2 FORMAT @args.SLAVE { type string } @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm $SLAVE file $FILE format $FORMAT } file { @args.0 FILE @args.1 FORMAT @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT } null { type null } cards 'cards.pcm' front 'cards.pcm.front' rear 'cards.pcm.rear' center_lfe 'cards.pcm.center_lfe' side 'cards.pcm.side' surround40 'cards.pcm.surround40' surround41 'cards.pcm.surround41' surround50 'cards.pcm.surround50' surround51 'cards.pcm.surround51' surround71 'cards.pcm.surround71' iec958 'cards.pcm.iec958' spdif 'cards.pcm.iec958' modem 'cards.pcm.modem' phoneline 'cards.pcm.phoneline' dmix 'cards.pcm.dmix' dsnoop 'cards.pcm.dsnoop' default { type plug slave.pcm dmix } headset { @args.0 BDADDR @args.1 TIMEOUT @args.BDADDR { type string default '00:07:A4:5C:64:F0' } @args.TIMEOUT { type integer default 6000 } type sco bdaddr $BDADDR timeout $TIMEOUT } a2dpd { type a2dpd } I tried to incorporate dsnoop in this, but it doesnt seem to work. How should I merge it in the alsa.conf?

1 0

[alsa-devel] Effect of CONFIG_SND_PCM_OSS_PLUGINS
by Adrian McMenamin 27 Jul '07

27 Jul '07

Unless CONFIG_SND_PCM_OSS_PLUGINS is set then with my driver the userspace (and OSS) app mpg123 doesn't work properly - not outputting any sound. Is this likely to be a problem with my driver or with the app? Adrian

3 7

[alsa-devel] VT1618 driver
by Niels Gram Jeppesen 27 Jul '07

27 Jul '07

Hi, I have a VIA CN1000G mini-ITX board with a VT1618 that I would like to have up and running with sound next week. I see (https://bugtrack.alsa-project.org/wiki/wikka.php?wakka=VT1618&show_comments…) that there exist a still unreleased driver for the chip. If possible, I would like to play arround with that driver next week. Best regards and thanks in advance, Niels Gram Jeppesen

1 0

[alsa-devel] Intel HDA Driver - Bug
by Steven King 26 Jul '07

26 Jul '07

Hi, I have been trying to get the sound on my Dell Inspiron 1720 laptop running under Debian Etch (STAC 9205). Originally, I could only get sound through the main speakers, there was no headphone sound or mic. I downloaded the latest alsa-drivers package (1.0.14), compiled it and installed it. I received the following messages (and the device would not register itself): hda_intel: azx_get_response timeout, switching to polling mode... hda_intel: azx_get_response timeout, switching to single_cmd mode... I started digging into the driver code and found an issue with one of the routines going outside an array. The offending code is (all in patch_sigmatel): static void stac92xx_set_config_regs(struct hda_codec *codec) { // ... for (i = 0; i < spec->num_pins; i++) { // ... } And in: static int patch_stac9205(struct hda_codec *codec) { // .. spec->num_pins = 14 ; //ERROR // .. } However, there are only 12 pins with 12 corresponding NIDs and default configuration so in stac92xx_set_config_regs it accesses spec->pin_configs[i] and spec->pin_nids[i] outside of bounds. Fix: Add a define (STAC_9205_MAX_PINS = 12) and use for array defines and num_pins. The driver loads now, but I still only have sound out the main speaker. If I get rid of the configuration sent to the chipset altogether. I get sound through Headphone and Speakers (with detect working on HP), but still no MIC. How can I add support for my Dell as I don't know the configuration to use or am I missing something here? Thanks, Steve

1 0

[alsa-devel] [patch] asihpi: mmap and link streams
by Eliot Blennerhassett 26 Jul '07

26 Jul '07

Greetings, asihpi version 3.07.13 : Enable MMAP dma capable cards. Enable linked streams for cards that have hardware support for this. Add TLV for level and volume controls. Register asihpi class for HPI part of driver Add documentation, remove unused code. Signed-off-by: Eliot Blennerhassett <eblennerhassett(a)audioscience.com> ---- Patch (4798 lines) is here: http://audioscience.com/internet/download/beta/alsa_driver3349_asihpi30713.… Matching firmware is here: http://audioscience.com/internet/download/firmware/dspbins30713.zip FYI: Original source package here: http://audioscience.com/internet/download/hpklinux-3.07.13.tar.bz2 asihpi.c | 1517 ++++++++++++++++++++++++++------------------------- Implement MMAP and linked streams. Merge some capture and playback functions Add dB TLV for level and volume controls hpi6000.c | 7 Clean hpi6205.c | 498 +++++++++------- Changes to allow attaching external DMA buffer (for ALSA) hpidebug.c | 67 +- hpidebug.h | 349 +++++------ Changes to support unicode hpimod.c | 283 ++------- Cleanup, add class registration hpimsgx.c | 50 + Add 2 new subsys functions hpios_linux_kernel.c | 36 - hpios_linux_kernel.h | 15 Minor debug changes. Add pci dev ptr to allow alsa to allocate dma buffer hpi.h | 54 - hpi56301.c | 1 hpidspcd.c | 2 hpifunc.c | 121 ++-- Documentation updates 13 files changed, 1506 insertions(+), 1494 deletions(-)

2 1

[alsa-devel] ALC262 - HDA-Intel
by Andy 26 Jul '07

26 Jul '07

I have a ALC262 hda-intel card which doesn't function quite correctly. Details can be found at https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/110145 I got bored of waiting for it to be fixed, and had a go myself. My knowledge of ALSA is very limited, but nevertheless it was really simple to add my device ids to the quirks list in the patch_realtek.c file - eg: ... SND_PCI_QUIRK(0x104d, 0x900e, "Sony ASSAMD", ALC262_SONY_ASSAMD), SND_PCI_QUIRK(0x104d, 0x1f00, "Sony ASSAMD", ALC262_SONY_ASSAMD), /* My card... */ SND_PCI_QUIRK(0x104d, 0x820f, "Sony ASSAMD", ALC262_SONY_ASSAMD), ... I found by trial and error that ALC262_SONY_ASSAMD almost works. Connecting headphones disables built-in speakers and unplugging them re-enables them. However, something does not get initialised until this is done. That is, every bootup using ALC262_SONY_ASSAMD, I must plug in and remove my headphones before the device is initialised properly. Until this is done, I get no sound and the little GUI mixer in XFCE doesn't show the list of supported outputs on the device. I'm posting this to the devel list as I'm more interested the code side of things rather than actually having working sound :) Thanks Andy. ___________________________________________________________ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.ht…

2 7

[alsa-devel] [PATCH 1/2] Add SPI devices to ALSA Kconfig and Makefile
by Hans-Christian Egtvedt 26 Jul '07

26 Jul '07

This patch adds SPI devices in the ALSA diretory, including the Kconfig and Makefile. Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- sound/Kconfig | 6 +++++- sound/Makefile | 3 ++- sound/spi/Kconfig | 6 ++++++ sound/spi/Makefile | 2 ++ 4 files changed, 15 insertions(+), 2 deletions(-) create mode 100644 sound/spi/Kconfig create mode 100644 sound/spi/Makefile diff --git a/sound/Kconfig b/sound/Kconfig index 9ea4738..9217739 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -63,9 +63,13 @@ source "sound/aoa/Kconfig" source "sound/arm/Kconfig" +if SPI +source "sound/spi/Kconfig" +endif + source "sound/mips/Kconfig" -# the following will depend on the order of config. +# tee following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index b7c7fb7..581ea5e 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -5,7 +5,8 @@ obj-$(CONFIG_SOUND) += soundcore.o obj-$(CONFIG_SOUND_PRIME) += sound_firmware.o obj-$(CONFIG_SOUND_PRIME) += oss/ obj-$(CONFIG_DMASOUND) += oss/ -obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ +obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ \ + sparc/ spi/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig new file mode 100644 index 0000000..60fbb2a --- /dev/null +++ b/sound/spi/Kconfig @@ -0,0 +1,6 @@ +#SPI drivers + +menu "SPI devices" + depends on SND != n + +endmenu diff --git a/sound/spi/Makefile b/sound/spi/Makefile new file mode 100644 index 0000000..30d8ec9 --- /dev/null +++ b/sound/spi/Makefile @@ -0,0 +1,2 @@ +# Makefile for SPI drivers + -- 1.5.2.3

3 20

[alsa-devel] How to get correct midi timings from ALSA using the library only
by Carlo Florendo 26 Jul '07

26 Jul '07

Hi, (1) I've written a command line MIDI sequencer for lightweight systems and am successful in making it work using the ALSA queue API. However, one drawback of the API is its lack of callback functions. I wish to be able to track events as they are drained by the queue. (2) I know and have successfully worked on a work around whereby the application itself subscribes to the output port so as to see events as they are played. However, I wish to be able to make the sequencer or player work without the use of the ALSA queue nor the workaround in (2). Here's the pseudo-code of the relevant MIDI player routine: for (i = 0; i < number_of_events; i++) { usleep(event[i].delta_time_in_microseconds); output_and_drain_event(event[i]); } This routine gives a non-bearable latency on 2.4 kernels but not so much on 2.6 kernels. How could I get the app to <u|nano>sleep() in the most accurate way in userspace without using the ALSA queue nor the extra subscription to an output port? Or, is there a drain or output routine that supports callbacks? If so, I will be grateful if you could point them out. I seem not to find any output callback routine under the docs. Thank you very much. Best Regards, Carlo -- Carlo Florendo Softare Engineer/Network Co-Administrator Astra Philippines Inc. UP-Ayala Technopark, Diliman 1101, Quezon City Philippines http://www.astra.ph -- The Astra Group of Companies 5-3-11 Sekido, Tama City Tokyo 206-0011, Japan http://www.astra.co.jp

3 9

[alsa-devel] dmix and sounds less than 64kb
by Aaron "Caustik" Robinson 25 Jul '07

25 Jul '07

We have been having one heck of a time tracking down the cause of some ALSA audio playback issues. It appears that any time we play a very short sound (<64kb), either the sound doesn't play at all, or only a fragment of it plays. We haven't had much luck finding a solution to this issue, then I found this thread on the history of this email group: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-April/000526.html Our problem is pretty much exactly like his issues relating to <64k sounds. My question is - has anybody else run into this problem? To the OP of the above thread, if you're still listening - did you ever find a solution to your problem? I'd be very interested to hear any help you can provide. Thanks much, caustik

3 9

[alsa-devel] Testing play and record simultaneously.
by Pharaoh . 25 Jul '07

25 Jul '07

Very naive questions. 1. I want to test the recording using arecord, but how will I make sure that the data is actually being recorded? I mean is there any way of storing the data recorded, say in a file? Off course there must be some way, its just that I dont know it. Can some one throw some light on this? 2. Also, I want to test my card in full duplex mode as well, i.e. playing and recording simultaneously. How can test this using aplay and arecord? -Pharaoh.

2 1

[alsa-devel] [PATCH] CA0106 SPI DAC powerdown
by Trent Piepho 25 Jul '07

25 Jul '07

Here are three patches for the CA0106 SPI DAC. The first patch powers down the DACs (other than the front dac) when they are not in use. The front dac serves as the output for the line-in pass-through / capture feedback, so it's not powered down. The second patch adds a bunch of symbol names for SPI registers, and converts the existing init sequence from magic numbers to these names. The sequence doesn't change at all. The third patch cleans up some unnecessary repeated sets to the same registers in the SPI init sequence. Of course, I have tested all these patches and they work.

2 1

[alsa-devel] How do I tell ALSA that my driver supports every possible sample rate?
by Timur Tabi 25 Jul '07

25 Jul '07

I'm building an ASOC driver, and I want to tell ALSA that my PCM and my I2S drivers support every possible sample rate, because it's the codec that determines the real limitations. What SNDRV_PCM_RATE_xxx bit-pattern do I specify that will always match? -- Timur Tabi Linux Kernel Developer @ Freescale

4 14

Re: [alsa-devel] support for Medion laptops
by albox＠free.fr 25 Jul '07

25 Jul '07

Realtek published a driver using ALSA 1.0.14-4. It seems that they added a new model for medion ("medion-md2"). I tried this model but it doesn't work either on my laptop. I sent them an email to know more about that. Does anybody has more information ? I updated the bug report : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3160 Thanks

1 0

[alsa-devel] YMFPCI: IEC958 volume does not work anymore (2.6.18->2.6.21)
by Jan Zuchhold 24 Jul '07

24 Jul '07

Hello, i'm using a Hoontech Soundtrack Digital-XG (YMF744) and after i've upgraded to kernel 2.6.21, i can't control the IEC958 volume (using "Wave volume") any longer. It seems to be always at maximum volume. The volume control of the analog output port works fine. This is working in 2.6.18, that i've used before. Something must have changed in between. Is there anything I can do to help to get this working again? Thanks alot, Jan

2 2

Re: [alsa-devel] snd-hda-intel, AD 1988b: SPDIF output not working correctly, volume low for analog output
by Takashi Iwai 24 Jul '07

24 Jul '07

At Mon, 23 Jul 2007 23:02:57 +0200, Matthias Dahl wrote: > > Hello Takashi. > > On Friday 20 July 2007 15:47:55 Takashi Iwai wrote: > > > This is expected behavior. The PCM volume is implemented via softvol > > plugin, and it creates dynamically at the first run. But, usually, > > it's restored via alsactl at the next boot. > > It is. Thanks for the info. > > > Yes, it'll involve with patch_analog.c deeply. Basically you need to > > modify init verb table and the mixer element table. Check the AD1988B > > datasheet, and try to assign the different PIN (and corresponding > > mixer / selector widgets) in init verb and mixer tables. Then, > > rebuild driver, and try the driver until you get the complete mapping > > of pin widgets. > > I am currently a bit limited in time (end of semester exams) but after a few > hours of playing around (and reading the specs) I figured the following: > > 1) From time to time no analog output is routed to the spdif anymore. The only > clue I could get is the "IEC958 Playback Default" which is set to > > 06820002 [... many more zeros ...] > > and thus restored to that value after reboots. When things work like they > should (analog output is routed to spdif), its value is > > 04820002 [... many more zeros ...] > > Manually setting it to this value naturally solves the problem. I have yet > to figure out how to trigger this problem?! It seems to happen rather > randomly. In both cases, ac3 passthrough works just fine btw. Maybe some apps change the IEC958 status bits and doesn't restore the original value at close. (Or interrupting via ctrl-C may cause it.) You can see the meaning of these bits via iecset utility included in alsa-utils package. > 2) I have come a step closer in solving the analog-to-spdif volume/distortions > problem... at least I hope. > > The following patch fixes the volume problem for me. Without it, I had to > set the IEC958 volume somewhere around 58... increasing it would result in > noticable clipping and even short audio gaps (noise). > > Honestly, I don't quite understand the purpose of the code I removed in the > first place. From what I figured reading the specs and the code, the spdif > in is mixed with the spdif out stream... but that's all I could gather. But > still it doesn't seem quite right to me. > > --- patch_analog.c.orig 2007-07-23 22:44:25.000000000 +0200 > +++ patch_analog.c 2007-07-23 22:45:15.000000000 +0200 > @@ -2037,12 +2037,8 @@ > static struct hda_verb ad1988_spdif_init_verbs[] = { > /* SPDIF out sel */ > {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ > - {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ > - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, > - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, > /* SPDIF out pin */ > {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ > - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */ > > { } > }; > > The audio distortions in the background have become a little less noisy > but are still there. (a bass-like pumping, some white noise and sometimes > some high frequency tone) But the volume works just fine now... (iec958 > vol at 100%) and the audio sounds somewhat "clearer"! So, the culprit looks like the analog loopback. Could you try the patch below, and make sure that you set "PCM" as the "IEC958 Playback Source"? thanks, Takashi diff -r c2078875c0cd pci/hda/patch_analog.c --- a/pci/hda/patch_analog.c Tue Jul 24 12:10:34 2007 +0200 +++ b/pci/hda/patch_analog.c Tue Jul 24 12:47:42 2007 +0200 @@ -1889,16 +1889,19 @@ static int ad1988_spdif_playback_source_ struct hda_codec *codec = snd_kcontrol_chip(kcontrol); unsigned int sel; - sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); - if (sel > 0) { + sel = snd_hda_codec_read(codec, 0x1d, 0, AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + if (!(sel & 0x80)) + ucontrol->value.enumerated.item[0] = 0; + else { sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0); if (sel < 3) sel++; else sel = 0; + ucontrol->value.enumerated.item[0] = sel; } - ucontrol->value.enumerated.item[0] = sel; return 0; } @@ -1910,17 +1913,32 @@ static int ad1988_spdif_playback_source_ int change; val = ucontrol->value.enumerated.item[0]; - sel = snd_hda_codec_read(codec, 0x02, 0, AC_VERB_GET_CONNECT_SEL, 0); if (!val) { - change = sel != 0; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x02, 0, - AC_VERB_SET_CONNECT_SEL, 0); + sel = snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT); + change = sel & 0x80; + if (change || codec->in_resume) { + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(0)); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(1)); + } } else { - change = sel == 0; - if (change || codec->in_resume) - snd_hda_codec_write(codec, 0x02, 0, - AC_VERB_SET_CONNECT_SEL, 1); + sel = snd_hda_codec_read(codec, 0x1d, 0, + AC_VERB_GET_AMP_GAIN_MUTE, + AC_AMP_GET_INPUT | 0x01); + change = sel & 0x80; + if (change || codec->in_resume) { + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_MUTE(0)); + snd_hda_codec_write(codec, 0x1d, 0, + AC_VERB_SET_AMP_GAIN_MUTE, + AMP_IN_UNMUTE(1)); + } sel = snd_hda_codec_read(codec, 0x0b, 0, AC_VERB_GET_CONNECT_SEL, 0) + 1; change |= sel != val; @@ -2039,10 +2057,9 @@ static struct hda_verb ad1988_spdif_init {0x02, AC_VERB_SET_CONNECT_SEL, 0x0}, /* PCM */ {0x0b, AC_VERB_SET_CONNECT_SEL, 0x0}, /* ADC1 */ {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, + {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)}, /* SPDIF out pin */ {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE | 0x27}, /* 0dB */ - {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0) | 0x17}, /* 0dB */ { } };

2 2

[alsa-devel] Query regarding ALSA control interface
by Pharaoh . 24 Jul '07

24 Jul '07

Hi I am writing a pcm control interface for my alsa driver. I am confused about how the switches are used. I have implemented simple cases like following and they are working well. static int audio_path_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { /*Count is 256 since the values can vary from 0-255. But values fall in this range only*/ uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 256; uinfo->value.integer.min = 0; uinfo->value.integer.max = 255; FN_EXIT; return 0; } static int audio_path_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { my_mixer_t *mixer_chip = snd_kcontrol_chip(kcontrol); ucontrol->value.integer.value[0] = mixer_chip->echo; ucontrol->value.integer.value[1] = mixer_chip->echo; return 0; } static int audio_path_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { my_mixer_t *mixer_chip = snd_kcontrol_chip(kcontrol); ----- ---- } Now, I have to implement a control which takes four different values..namely 0,1.100 and 101. In this case what shoud be the type of control? It should be ENUMERATED or INTEGER? I am confused here since the values are not exactly in a range. -pharaoh.

2 2

[alsa-devel] [PATCH] Patch for AICA
by Adrian McMenamin 24 Jul '07

24 Jul '07

I submitted this yesterday via lkml and it was accepted by the SH maintainer. Probably should have copied in the ALSA list too - though the change is so small and consequential of another issue I was dealing with I forgot. The change is necessary because of a fix in the SH DMA API (see http://lkml.org/lkml/2007/7/23/445 for the larger patch) Signed-off by: Adrian McMenamin ,adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.h b/sound/sh/aica.h index 8c11e3d..d098baa 100644 --- a/sound/sh/aica.h +++ b/sound/sh/aica.h @@ -52,7 +52,7 @@ #define AICA_CHANNEL1_OFFSET 0x21000 #define CHANNEL_OFFSET 0x10000 -#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_CHANNEL 5 #define AICA_DMA_MODE 5

3 2

[alsa-devel] [PATCH] Support for muting channels DAC with CA0106 - SB0570
by Trent Piepho 24 Jul '07

24 Jul '07

This patch adds mute controls for the analog output channels. It does this by sending SPI commands to the DAC on cards like the SB live 24-bit / Audigy SE that have an SPI DAC.

2 3

[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

2 1

[alsa-devel] SuSE 9.1 on Toshiba Satellite A105 - S4004
by Neeraj Vaidya 24 Jul '07

24 Jul '07

Hi, I've installed this distro on my toshiba laptop. In YaST , when I try to configure the sound card , it says no sound card detected. To cut a long story short , I do not get any sounds to work on my laptop. Sound works beautifully on XP , but I am trying to migrate all of my routine activities to linux ;) I am a newbie to the ALSA mailing list. Have a feeling that something to do with ALSA configuration can help me sort out this problem. When I run lspci , I do not see any audio card as being detected. Can someone please help me out ? Also , send me some diagnostic steps , if not the entire solution , so that we can sort this out in a series of emails , if the entire solution can not be provided in one email. Thanks in Advance! -Neeraj --------------------------------- Get the freedom to save as many mails as you wish. Click here to know how.

2 1

[alsa-devel] [PATCH] s3c24xx-pcm: fix hw_params dma handling
by Arnaud Patard 24 Jul '07

24 Jul '07

Hi, While testing the recently merged driver for the neo1973, I found out that it was not working due to dma allocation failure. The problem is that there's only one dma channel available on s3c2410 and there may be more than one call to pcm_ops->hw_params(), resulting in dma request failures. This patch from Harald Welte fix this. Arnaud

3 2

[alsa-devel] Intel HDA digital out distortion
by Ingmar Stieger 24 Jul '07

24 Jul '07

Hi, 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 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. In patch_analog.c, in static struct hda_verb ad1988_spdif_init_verbs[], changeing the forth entry from {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, to {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(1)} mutes what might be an feedback loop from the analog inputs. The distortion is gone if the input to this widget is muted. On a side note, the settings for the ADCs with NID:08, 09, and 0F looks fishy, too. According to the functional diagram, those should have no amps, and thus should not be muteable. The driver mutes them anyway... I'd like to help somebody who really understands this stuff to solve the problem for good. What would the next step be ? Thanks, Ingmar

2 2

[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

1 0

[alsa-devel] basic workflow when playing some sounddata
by Markus Franke 24 Jul '07

24 Jul '07

Dear Alsa-Developers, I am very new to Alsa driver development and have some problems to get a clue how the process of playing sounddata works. Let me explain my point of view and it would be nice if anybody could give some clarifications and corrections on it. Suppose I want to play a *.wav file from userspace with "aplay". After triggering all the configuration functions for PCM and the DAI at some point the chip_pcm_trigger() function get's called. As per "Writing Alsa Drivers" the TRIGGER_START command should now do something for "enabling the PCM engine". As per my understanding this could be for example to enable the interrupts of my DAI, which is (I2S over GSI) in my case. The interrupt handler by itself should call snd_pcm_period_elapsed() whenever a period of sounddata is copied to the device. Furthermore the so called "hw_pointer" should be updated in the interrupt handler, right? To make it easy I don't want to use DMA in the beginning. I simply want to copy the data to the hardware buffer of my DAI (I2S via GSI) and from there it is copied automatically to my codec. The question is now, at which place in the alsa framework I have to put the copy process of the sounddata. It should not be done in the interrupt handler because this would take to much time. But whereelse could this be done? Also I don't know the meaning of chip_pcm_pointer() callback. As far as I could figure out from the source, this function is called whenever snd_pcm_period_elapsed() is called, right? But what should be done exactly in this callback. Again I want to say that I don't want to use DMA at all by now. I've read already "Writing Alsa Drivers" but the questions above still remain for me and I am quite confused as you maybe noticed already from my questions above. ;-) It would be nice if anybody can give me a basic workflow which functions/callbacks are necessary for the playing part of an Alsa driver. Thanks in advance for any help, Regards, Markus

1 0

[alsa-devel] SNDRV_PCM_FMTBIT_S24_LE vs. SNDRV_PCM_FMTBIT_S24_3LE
by Timur Tabi 24 Jul '07

24 Jul '07

What is the difference between SNDRV_PCM_FMTBIT_S24_LE and SNDRV_PCM_FMTBIT_S24_3LE? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] Fix Kconfig entry for SND_S3C24XX_SOC_NEO1973_WM8753
by Arnaud Patard 23 Jul '07

23 Jul '07

SND_S3C24XX_SOC_NEO1973_WM8753 depends on MACH_GTA01 but the Kconfig entry which is going to be merged is MACH_NEO1973_GTA01. signed-off-by: Arnaud Patard <arnaud.patard(a)rtp-net.org>

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[alsa-devel] [PATCH] Fix tas_suspend/resume build warning
by Stephen Rothwell 23 Jul '07

23 Jul '07

sound/aoa/codecs/snd-aoa-codec-tas.c:750: warning: 'tas_suspend' defined but not used sound/aoa/codecs/snd-aoa-codec-tas.c:760: warning: 'tas_resume' defined but not used Signed-off-by: Stephen Rothwell <sfr(a)canb.auug.org.au> --- sound/aoa/codecs/snd-aoa-codec-tas.c | 9 +++++---- 1 files changed, 5 insertions(+), 4 deletions(-) -- Cheers, Stephen Rothwell sfr(a)canb.auug.org.au diff --git a/sound/aoa/codecs/snd-aoa-codec-tas.c b/sound/aoa/codecs/snd-aoa-codec-tas.c index 2f771f5..3cbfe46 100644 --- a/sound/aoa/codecs/snd-aoa-codec-tas.c +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c @@ -743,6 +743,7 @@ static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock return 0; } +#ifdef CONFIG_PM /* we are controlled via i2c and assume that is always up * If that wasn't the case, we'd have to suspend once * our i2c device is suspended, and then take note of that! */ @@ -768,7 +769,6 @@ static int tas_resume(struct tas *tas) return 0; } -#ifdef CONFIG_PM static int _tas_suspend(struct codec_info_item *cii, pm_message_t state) { return tas_suspend(cii->codec_data); @@ -778,7 +778,10 @@ static int _tas_resume(struct codec_info_item *cii) { return tas_resume(cii->codec_data); } -#endif +#else /* CONFIG_PM */ +#define _tas_suspend NULL +#define _tas_resume NULL +#endif /* CONFIG_PM */ static struct codec_info tas_codec_info = { .transfers = tas_transfers, @@ -791,10 +794,8 @@ static struct codec_info tas_codec_info = { .owner = THIS_MODULE, .usable = tas_usable, .switch_clock = tas_switch_clock, -#ifdef CONFIG_PM .suspend = _tas_suspend, .resume = _tas_resume, -#endif }; static int tas_init_codec(struct aoa_codec *codec) -- 1.5.2.3

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[alsa-devel] [PATCH] hda-intel: support for iMac 24'' released on 09/2006
by Nicola Fagnani 23 Jul '07

23 Jul '07

Basic audio support for the iMac 24'' model released on 09/2006, including headphone jack detection with automatic speaker muting. This iMac uses the Realtek ALC885 codec, not a Sigmatel one as in other models. Functionality has been tested for internal speakers, headphone and microphone. Signed-off-by: Nicola Fagnani <nicfagn(a)iol.it> --- diff -ru alsa-driver-1.0.14/alsa-kernel/Documentation/ALSA- Configuration.txt alsa-driver-1.0.14-patched/alsa-kernel/ Documentation/ALSA-Configuration.txt --- alsa-driver-1.0.14/alsa-kernel/Documentation/ALSA- Configuration.txt 2007-06-04 09:28:52.000000000 +0200 +++ alsa-driver-1.0.14-patched/alsa-kernel/Documentation/ALSA- Configuration.txt 2007-07-19 20:16:24.000000000 +0200 @@ -827,6 +827,7 @@ 6stack-dig 6-jack digital with SPDIF I/O arima Arima W820Di1 macpro MacPro support + imac24 iMac 24'' with jack detection w2jc ASUS W2JC auto auto-config reading BIOS (default) diff -ru alsa-driver-1.0.14/alsa-kernel/pci/hda/patch_realtek.c alsa- driver-1.0.14-patched/alsa-kernel/pci/hda/patch_realtek.c --- alsa-driver-1.0.14/alsa-kernel/pci/hda/patch_realtek.c 2007-06-04 09:28:54.000000000 +0200 +++ alsa-driver-1.0.14-patched/alsa-kernel/pci/hda/patch_realtek.c 2007-07-06 12:19:30.000000000 +0200 @@ -144,6 +144,7 @@ ALC882_TARGA, ALC882_ASUS_A7J, ALC885_MACPRO, + ALC885_IMAC24, ALC882_AUTO, ALC882_MODEL_LAST, }; @@ -5054,6 +5055,59 @@ { } }; +/* iMac 24 mixer. */ +static struct snd_kcontrol_new alc885_imac24_mixer[] = { + HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x00, HDA_OUTPUT), + HDA_CODEC_MUTE("Master Playback Switch", 0x0c, 0x00, HDA_INPUT), + { } /* end */ +}; +/* iMac 24 init verbs. */ +static struct hda_verb alc885_imac24_init_verbs[] = { + /* Internal speakers: output 0 (0x0c) */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x18, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Internal speakers: output 0 (0x0c) */ + {0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1a, AC_VERB_SET_CONNECT_SEL, 0x00}, + /* Headphone: output 0 (0x0c) */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* Front Mic: input vref at 80% */ + {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, + {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, + { } +}; + +/* Toggle speaker-output according to the hp-jack state */ +static void alc885_imac24_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_update(codec, 0x18, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x18, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 0, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); + snd_hda_codec_amp_update(codec, 0x1a, 1, HDA_OUTPUT, 0, + 0x80, present ? 0x80 : 0); +} +/* Processes unsolicited events. */ +static void alc885_imac24_unsol_event(struct hda_codec *codec, unsigned int res) +{ + /* Headphone insertion or removal. */ + if (((res >> 26) == ALC880_HP_EVENT)) { + alc885_imac24_automute(codec); + } +} + + static struct hda_verb alc882_targa_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -5274,6 +5328,7 @@ [ALC882_ARIMA] = "arima", [ALC882_W2JC] = "w2jc", [ALC885_MACPRO] = "macpro", + [ALC885_IMAC24] = "imac24", [ALC882_AUTO] = "auto", }; @@ -5345,6 +5400,19 @@ .channel_mode = alc882_ch_modes, .input_mux = &alc882_capture_source, }, + [ALC885_IMAC24] = { + .mixers = { alc885_imac24_mixer }, + .init_verbs = { alc885_imac24_init_verbs }, + .num_dacs = ARRAY_SIZE(alc882_dac_nids), + .dac_nids = alc882_dac_nids, + .dig_out_nid = ALC882_DIGOUT_NID, + .dig_in_nid = ALC882_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc882_ch_modes), + .channel_mode = alc882_ch_modes, + .input_mux = &alc882_capture_source, + .unsol_event = alc885_imac24_unsol_event, + .init_hook = alc885_imac24_automute, + }, [ALC882_TARGA] = { .mixers = { alc882_targa_mixer, alc882_chmode_mixer, alc882_capture_mixer }, @@ -5494,6 +5562,9 @@ case 0x106b0c00: /* Mac Pro */ board_config = ALC885_MACPRO; break; + case 0x106b1000: /* iMac 24 */ + board_config = ALC885_IMAC24; + break; default: printk(KERN_INFO "hda_codec: Unknown model for ALC882, " "trying auto-probe from BIOS...\n"); @@ -5518,7 +5589,7 @@ if (board_config != ALC882_AUTO) setup_preset(spec, &alc882_presets[board_config]); - if (board_config == ALC885_MACPRO) { + if (board_config == ALC885_MACPRO || board_config == ALC885_IMAC24) { alc882_gpio_mute(codec, 0, 0); alc882_gpio_mute(codec, 1, 0); }

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[alsa-devel] Close bug 3113?
by Alan Horstmann 22 Jul '07

22 Jul '07

Since bug-track notes don't come to alsa-devel, I'm not sure where/who to send this to. I came across the link to this tidying my bookmarks. Looks like it can be closed now, which = one less. Alan

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[alsa-devel] [BUG] Open Kernel Bugs
by Natalie Protasevich 21 Jul '07

21 Jul '07

This is the first listing of the open bugs that will be sent out and will contain bugs on specific areas. The bugs on the list below are about a year old and most of them confirmed (or claimed) to still be a problem. It would be appreciated if the corresponding maintenance team could take a look, close off any which are fixed and see if we can fix any which aren't. NOTE: when replying to this email, please add the bug number to the Subject in the form [Bug 1234] so that bugzilla will capture the discussion. Thanks. bugzilla: timers [BUG 7967] TIMER_ABSTIME cleared resulting in userland deadlock http://bugzilla.kernel.org/show_bug.cgi?id=7967 bugzilla: parallel [BUG 7492] parport doesn't work if DMA is used http://bugzilla.kernel.org/show_bug.cgi?id=7492 bugzilla: network [BUG 8106] tx_errors and tx_fifo_errors not updated consistantly http://bugzilla.kernel.org/show_bug.cgi?id=8106 - error reporting [BUG 7856] b44 WOL problem (bcm4401) http://bugzilla.kernel.org/show_bug.cgi?id=7856 [BUG 5839] uli526x partially recognizing interface http://bugzilla.kernel.org/show_bug.cgi?id=5839 [BUG 6613] iptables broken on 32-bit PReP (ARCH=ppc) http://bugzilla.kernel.org/show_bug.cgi?id=6613 bugzilla: sound (ALSA) [BUG 7195] pc doesn't correctly shut down because of hda_intel http://bugzilla.kernel.org/show_bug.cgi?id=7195 [BUG 6244] cs46xx in Thinkapad A21m after suspend/resume doesn't play http://bugzilla.kernel.org/process_bug.cgi?id=6244 bugzilla: scheduler [BUG 7372] Unusable system (ie slow) when copying large files. http://bugzilla.kernel.org/show_bug.cgi?id=7372 bugzilla: input [BUG 7746] Support for unicode dead keys http://bugzilla.kernel.org/show_bug.cgi?id=7746 [BUG 7941] The atkbd module makes system unusable http://bugzilla.kernel.org/process_bug.cgi?id=7941 bugzilla: dvb-usb [BUG 8179] Yuan ub300 card not working/tuning http://bugzilla.kernel.org/show_bug.cgi?id=8179 bugzilla: PCI [BUG 7528] pci=assign-busses needed on DELL Latitude D810 http://bugzilla.kernel.org/show_bug.cgi?id=7528 [BUG 8187] 2.6.20 "PCI: Quirks" patch breaks X11 on I82801 http://bugzilla.kernel.org/show_bug.cgi?id=8187 [BUG 7917] "PCI: Failed to allocate mem resource" for PCI E-to-PCI Bridge http://bugzilla.kernel.org/show_bug.cgi?id=7917 bugzilla: i386 [BUG 5565] Guess of i386 APIC PTE area scribble http://bugzilla.kernel.org/show_bug.cgi?id=5565 [BUG 7589] libata: does not work anymore on VIA VT8251 AHCI/SATA 4-Port Controller http://bugzilla.kernel.org/show_bug.cgi?id=7589 bugzilla: Interrupts [BUG 7673] Memory card reader 5-in-1 don't working on Lenovo N100 3000 http://bugzilla.kernel.org/show_bug.cgi?id=7673 bugzilla: I/O - Storage [BUG 7921] oops in 2.6.20-rc6-git3 &&udf && pktcdvd http://bugzilla.kernel.org/show_bug.cgi?id=7921 [BUG 7507] pata_it821x: system freezes http://bugzilla.kernel.org/show_bug.cgi?id=7507 [BUG 7506] it821x: long wait time due to dma_timer_expiry http://bugzilla.kernel.org/show_bug.cgi?id=7506 [BUG 7703] hpt366 does not find drives http://bugzilla.kernel.org/show_bug.cgi?id=7703 [BUG 6516] wrong value for avgqu-sz http://bugzilla.kernel.org/show_bug.cgi?id=6516 bugzilla: serial [BUG 7750] open/close loop causes disabled irq http://bugzilla.kernel.org/show_bug.cgi?id=7750 bugzilla: PCMCIA [BUG 7788] System freeze as soon as PCMCIA card is inserted http://bugzilla.kernel.org/show_bug.cgi?id=7788 [BUG 8118] Wrong device identifiers returned in lspci for pccard with RL5c476 II (rev 80) http://bugzilla.kernel.org/show_bug.cgi?id=8118 [BUG 8262] PCMCIA: socket *** DANGER *** unable to remove socket power http://bugzilla.kernel.org/show_bug.cgi?id=8262 [BUG 7215] PCMCIA network card causes either X or kernel to freeze http://bugzilla.kernel.org/show_bug.cgi?id=7215 bugzilla: SCSI [BUG 4880] dpt_i2o.c does not register itself with sysfs http://bugzilla.kernel.org/show_bug.cgi?id=4880 bugzilla: File Systems [BUG 6124] mount of UDF fs ignores UID and GID options http://bugzilla.kernel.org/show_bug.cgi?id=6124 bugzilla: USB [BUG 8104] kernel floods console with "connect-debounce failed" http://bugzilla.kernel.org/show_bug.cgi?id=8104 ----------------------------------------------------------------------------- ----------------------------------------------------------------------------- ----------------------------- -------------------------------------------------------

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[alsa-devel] ALSA-USB driver issues with C-Media chipset
by Saurav SAHU 20 Jul '07

20 Jul '07

We have encountered some audio recording quality issues on the recent ALSA-USB drivers. These problems are faced on C-Media chipset based external USB audio devices. The recorded audio with 'arecord' is quite noisy, unclear and jerky. The version of ALSA I encountered are 1.0.12 and beyond. It seemed to work well previously on ALSA version 1.0.8. If I record with a Philips webcam mic (Philips chipset), the audio recording is very clear on across all ALSA versions. In case you have observed similar issues with C-Media drivers and aware of them, please let me know. Best Regards Saurav

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[alsa-devel] Noise Problem with rate convert plugins
by Andreas Rumpler 20 Jul '07

20 Jul '07

Hello, I'm new to this list.I hope someone could help me with a problem concerning the samplerate converter plugins of the alsa driver. Firstly some facts about my application. - Envy24 (VT1270) pci audio chip on a custom mainboard - playback of sounds/music with different sample rates from 8kHz to 48kHz (sample rate of the files) - the hardware (Envy24 chip) must always run with 44.1kHz or 48kHz (selectable by user), because the audio is outputed by a DAC and SPDIF and the SPDIF clock must not be changed according to the sample rate of the files. - Kernel is 2.6.22 and alsa is 1.0.14 So I'm using the samplerate (libsample rate based) plugin from the alsa-plugin package. Generaly it works very good. My problem is: Every time a playback is started a short plop noise is heard. The noise is on analog and SPDIF output. If I don't use the samplerate converter the noise is not heard at all. So it comes definitely from the rate converter plugin, I think. Changing the quality level of the plugin don't solve the problem. I've also switched to the new Speex rate converter plugin. With this plugin there is no noise at the start of playback. But here I have trouble with awful noise at some rate conversions (6kHz(source) -> 48kHz(output); 11.025kHz -> 44.1kHz; 22.05kHz -> 44.1kHz). So it's even worse than using the libsamplerate plugin. Finaly I've tried the libavcodec plugin, which is the worst according to noise. Could any comment this problem, please? This is my asound.conf: ############################################## # Playback over Line-Out and SPDIF simultaniously as default pcm.!default { type plug slave.pcm { type multi slaves.a.pcm "plughw:0,0" slaves.a.channels 2 slaves.b.pcm "hw:0,1" slaves.b.channels 2 bindings.0.slave a bindings.0.channel 0 bindings.1.slave a bindings.1.channel 1 bindings.2.slave b bindings.2.channel 0 bindings.3.slave b bindings.3.channel 1 } ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 } pcm_slave.s1 { pcm default rate 44100 } pcm_slave.s2 { pcm default rate 48000 } pcm.rate_convert_44100 { type rate slave s1 converter samplerate } pcm.rate_convert_48000 { type rate slave s2 converter samplerate } ##################################################### Regards Andreas Rumpler

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[alsa-devel] The users view of an audio card
by David Webb 20 Jul '07

20 Jul '07

I've been editing a couple of the unofficial wiki pages and a number of issues have arisen. My own system has the NVidia/Realtek motherboard chips so I may be in the dark about what is available on other chips. Some of the questions are background to ensure that what I write later in the wiki is correct. Takashi Iwai suggested I raise the issues here: 1. A large number of users seem to be reporting problems with Nvidia and Via soundcards. As far as I can tell Nvidia and Via do not produce soundcards but they do produce chipsets with southbridges which can connect to a motherboard audio chip. The when alsamixer (or Kmix) runs it reports something like: Card: NVidia CK804 Chip: Realtek AL650 rev 0 Presumably because the southbridge is the primary PCI device. So: 1.1 Is the above always correct (so I can add it to the wiki)? 1.2 Could alsamixer etc be modified to give a more sensible answer for non-experts. Something like: Card: Motherboard via NVidia CK804 Chip: Realtek AL650 rev 0 might work. I presume a look-up table is used to convert the PCI 2-byte integers to the above names so presumably the extra information needed could be obtained at this level. 2. On different motherboards a Realtek 650 may be connected using a NVidia CK804 with the intel8x0 alsa driver or using a VIA VT8233 with the vt82xx driver. 2.1 Given that the audio chip registers are the same in both cases why are two different drivers needed? (Are they?) 2.2 If it is something to do with the southbridges, where may the manuals that describe the differences be found? 2.3 What would happen if the motherboard manufacturers kept the same southbridges but started using an audio chip with a different set of registers? (2.2a Where can an AD'97 manual be found? - I could not find one on the intel site) 3. The Realtek 650 datasheet implies that independently of the mixing in the main input -> line-out pathway, the input to the ADC (i.e. on capture) may include more than one source (i.e. both audio and microphone). However alsamixer (and kmix) only allows me one input stream. So: 3.1 Is the manual misleading or have I misread it? 3.2 Is the manual correct but the driver is based on an earlier intel chip which did not have this facility? Related to this the manual seems to show a number of options which I cannot identify in alsamixer. For example the manual shows two surroundsound volume stages between which there are switched links, one to and two from the main stero path. Alsamixer shows a single amplifier. It has one link 'from' the main path but I could not find a link 'to' the main path. 3.3 Is the driver assuming a old layout which fits most of the new audio chip layout but not all? (This is really a question about what sort of warnings/advice should my write up include). 3.4 At which stage does alsa relate the registers to the descriptions, aux-volume, aux-mute etc? Again does it assume a layout which fits most aspects of the chips listed? Thats enough for the moment. I presume I could sit down for a year and read and understand the code - but in the meantime any help will be welcome. Regards, David Webb.

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[alsa-devel] eliminating sequencer and midi from my alsa module dependencies
by John L. Utz III 20 Jul '07

20 Jul '07

Hello; What arguments should give to the current version of alsa to eliminate the aforementioned things? here is my current conf string from hgcompile, am i missing something? args="--with-debug=full --with-isapnp=no --with-sequencer=no --with-oss=no --with-cards=via82xx" tnx! johnu

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[alsa-devel] Problem and workaround for Intel HDA
by Yann Le Brech 20 Jul '07

20 Jul '07

The problem : missing controls of the sound cards for an Intel HDA, Toshiba Satellite Pro U200 Full description : I was using the kernel 2.6.19 for a while, and I got some interesting functions like controlling the sound level on headphones or speakers separately. Then I upgraded to the kernel 2.6.21.1 and suddently the level controls changed, to useless level controls, like Master and PCM only (and some other useless). I wanted to get back to the previous control set, and I've found a solution by checking what was done in debugging mode. The previous kernel was not detecting the model config, but the new one did. So I simply strapped the detection routine by a return -1 and it did the trick. I know its not the best thing to do, but it was the simplest thing I thought about. Outputs : 2.6.19's Alsa output : ALSA sound/pci/hda/hda_intel.c:695: codec_mask = 0x3 2.6.21.1 : ALSA sound/pci/hda/hda_intel.c:695: codec_mask = 0x3 ALSA sound/pci/hda/hda_codec.c:1757: hda_codec: model 'toshiba' is selected for config 1179:1 (Toshiba U205) How to see the problem : Just see the list of controls in any mixer software. Environment : Toshiba Satellite Pro U200 Software : KateOS 3.2 with Linux 2.6.19 (working) and 2.6.21.1 (not working) Workaround : int __devinit snd_hda_check_board_config(struct hda_codec *codec, int num_configs, const char **models, const struct snd_pci_quirk *tbl) { return -1; // WORKAROUND if (codec->bus->modelname && models) { int i; for (i = 0; i < num_configs; i++) { if (models[i] && -- a++ Y.

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[alsa-devel] [PATCH] Add quirk for Asus P5LD2
by Claudio Matsuoka 20 Jul '07

20 Jul '07

Another simple patch that adds a system to the ALC882 quirk table. Tested with a week-old HG version, buildsystem of the current alsa-driver seems to be broken.

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[alsa-devel] a new synthesizer driver
by Steve Longerbeam 19 Jul '07

19 Jul '07

Hi all, I'm embarking on a big task: writing a new synthesizer driver. My device has all the ingredients for a MIDI wavetable sythesizer: 64 voices, each voice with envelope, pitch, and LFO controls. Also global effects like reverb. It's similar to the emu10k1. So my first question would be, is the synth code under synth/emux/ general enough to use for a different synthesizer chip? Ie, can I implement my own operators in 'struct snd_emux_operators' and pass that to snd_emux_register() ? Would it be a simple matter to make synth/emux/ a generic synth layer, by moving all the source in synth/emux/ up one level to synth/, and then rename all functions and data structures starting with "snd_emux" to "snd_synth" ? For instance, snd_emux_operators becomes snd_synth_operators, snd_emux_control() becomes snd_synth_control(), etc. It seems this path is already partially completed, since synth/emux/soundfont.c is a generic soundfont loader. The thing I want to avoid doing is creating a synth/my_synth/ directory and duplicating lots of code from synth/emux/ (including just copying soundfont.c verbatim!). Steve

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[alsa-devel] Re-uploaded patch for bug #2708
by Nicola Fagnani 19 Jul '07

19 Jul '07

I re-uploaded the patch with the requested changes some time ago, and I was wondering if there is still something wrong with it. Thanks Nicola Fagnani

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[alsa-devel] underrun observed when aplay is used with mmap
by Pharaoh . 19 Jul '07

19 Jul '07

Hi , Finally I have got my driver working and now testing it using aplay. I am able to play a song without getting any underruns when I use read/write mode but when I use mmapped mode I get underrun warnings. After this if I use read/write mode again then I observe 1 warning of underrun. Any idea what could be wrong? I know why underruns occur etc. Does it have to do with mmapped mode? I dont think so... Here is the log. # aplay -Dplughw:0,0 --rate=44100 --format=s16 --channels=2 --mmap 44.wav Playing WAVE '44.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 0.384 ms long) underrun!!! (at least 0.118 ms long) underrun!!! (at least 0.124 ms long) underrun!!! (at least 0.117 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.128 ms long) underrun!!! (at least 0.116 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.121 ms long) underrun!!! (at least 0.125 ms long) ....... and it goes on till the song finishes. And after this I play without using mmap and log says.. # aplay -Dplughw:0,0 --rate=44100 --format=s16 --channels=2 44.wav Playing WAVE '44.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo underrun!!! (at least 516.424 ms long) and then the song plays without any underrun. -pharaoh.

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[alsa-devel] cs46xx thinkpad resume fix - #bug1946
by Pavel Kysilka 19 Jul '07

19 Jul '07

Hi, attached patch fix resume on thinkpad notebooks with cs46xx chip. Tested with Thinkpad A21m. Patch is opposite latest linus git 2.6.22. Bug #1946 may be closed. Signed-off-by: Pavel Kysilka <pavelk(a)bsys.cz> Pavel Kysilka ________ Information from NOD32 ________ This message was checked by NOD32 Antivirus System for Linux Mail Servers. http://www.eset.com

3 6

[alsa-devel] PS3 - Surround sound on SPDIF set-up
by gt bradley 19 Jul '07

19 Jul '07

Hi, I've been beating my had against this for several weeks now. (started with fedora 6, tried adding in the plugings etc (version 1.0.12) -- also tried upgrading to the current 1.0.14 version, without success (couldn't produce surround-sound on teh spdif output). I have now switched to fedorea 7 and was able to have rythmbox play surround sound via spdif, but have since lost that ability. (but I'm convinced the hardware and low-level drivers support it). In order to limit the number of moving parts, I have reduced to just trying to get speaker-test to play on all 6 speakers. from pirut & I have alsa-lib 1.0.14-0.4.rc3.fc7.ppc(64) and alsa-utils 1.0.14-0.7.rc2.fc7.ppc The default install has no asound.conf file. (attempts to create a "better" one didn't improve the situation). I do not see a file matching the ps3 device in etc/alsa/cards nor etc/alsa/pcm, but I have not found any documentation on adding entries to the above directories. [root@ps3 pcm0p]# cat /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: SPDIF name: SPDIF subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 [root@ps3 ~]# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: PS3 [PS3], device 0: SPDIF [SPDIF] Subdevices: 1/1 Subdevice #0: subdevice #0 [root@ps3 ~]# aplay -L null Discard all samples (playback) or generate zero samples (capture) [root@ps3 ~]# amixer [root@ps3 ~]# amixer scontrols [root@ps3 ~]# speaker-test -c 6 speaker-test 1.0.14rc2 Playback device is default Stream parameters are 48000Hz, S16_BE, 6 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 4096 to 8192 Period size range from 256 to 256 Using max buffer size 8192 Periods = 4 was set period_size = 256 was set buffer_size = 8192 0 - Front Left 4 - Center 1 - Front Right 3 - Rear Right 2 - Rear Left 5 - LFE Time per period = 17.821324 * above test only produces sound on channels 0 & 1. xmms produces no sound and emits the following: ** WARNING **: alsa_setup_mixer(): Failed to find mixer element: PCM Is it possible to get some help/pointers in the right direction? thanks GT* *

4 7

Re: [alsa-devel] ALSA driver for RealTek ALC268 chip
by Jonathan Woithe 18 Jul '07

18 Jul '07

Hi Amitay Amitay Isaacs (A.Isaacs(a)adfa.edu.au) wrote: > I found your names from the ALSA RealTek driver available for download > on the realtek website. Unfortunately the driver package does not have > support for the ALC268 chip. > > I have an ACER laptop with ALC268 HDA sound and would like to run linux > on it. > > Is the ALC268 chip driver available yet? Even the alpha/test version? Firstly, please submit queries like this to the ALSA development list. While I'm not fussed, other developers may not respond to emails sent only to their private addresses. It also means you don't have to guess who would be the most appropriate person to ask. Consequently I have added alsa-devel to the CC of this email. It would probably not take a lot of effort to add support for the ALC268 chip given that the ALC260 and ALC262 are supported. Furthermore, it seems that some work has already gone into this - see for example https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 So I would say that ALC268 support is being worked on. Pshou can probably give more details or provide further assistance. Note that my involvement in the ALSA RealTek driver was limited to the extended ALC260 support I added to take full advantage of this chip in my laptop. I don't have access to machines with other Realtek chips in them and so I'm not in the best position to provide direct assistance for other chips. Regards jonathan

1 0

[alsa-devel] Cirrus Logic CS4270 support coming soon
by Timur Tabi 18 Jul '07

18 Jul '07

Hi Everyone, I just wanted to let everyone know that I will soon be submitting an ASoC driver for the Cirrus Logic CS4270 codec. I have the driver working in-house on an older version of ALSA. As soon as I finish work on my current project, I will clean up the driver, update to the latest code, and submit it. If anyone is using this codec on their hardware, please let me know. The CS4270 supports many different configurations, but my board only enables some of them.

1 0

[alsa-devel] substream->private_data == NULL in hw_params
by Florian Demski 17 Jul '07

17 Jul '07

Hi there, after many reboots I finally fixed an error in a driver but I am not sure if there ain't an underlying problem which causes this error. The initialization is done with struct snd_pcm *pcm; int err; if((err = snd_pcm_new(chip->pod->line6.card, (char *)chip->pod->line6.devname, 0, 1, 1, &pcm)) < 0) return err; pcm->private_data = chip; pcm->private_free = pod_cleanup_pcm; chip->pcm = pcm; However in hw_param, substream->private_data is NULL and I have to get the right value with this construct: static int snd_pod_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { int ret; struct snd_pod_pcm *chip = snd_pcm_substream_chip(substream); /* -- Florian Demski [FD] */ /* don't ask me why, but this fixes the bug on my machine */ if ( chip == NULL ) { if ( substream->pcm == NULL ) return -ENOMEM; if ( substream->pcm->private_data == NULL ) return -ENOMEM; substream->private_data = substream->pcm->private_data; chip = snd_pcm_substream_chip( substream ); } /* -- [FD] end */ Somehow only my machine produces this error, I tried switching off SMP and PREEMPT, but with no other result. Or is this possibly an (undocumented) feature in alsa? I'd be happy if actually some of you experienced guys might take a look at that driver: SVN https://line6linux.svn.sourceforge.net/svnroot/line6linux/driver/trunk Browse https://line6linux.svn.sourceforge.net/viewvc/line6linux/driver/trunk/ Thanks in advance, Florian

2 1

[alsa-devel] [PATCH] hda: Enable SPDIF in/out on some stac9205 boards
by Matthew Ranostay 17 Jul '07

17 Jul '07

Added support for some boards with SPDIF in/out, and cleaned up the GPIO enable function. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 631a50e624df pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jul 16 11:01:30 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Mon Jul 16 16:34:46 2007 -0400 @@ -44,6 +44,7 @@ enum { enum { STAC_9205_REF, + STAC_M43xx, STAC_9205_MODELS }; @@ -218,7 +219,6 @@ static hda_nid_t stac9205_pin_nids[12] = 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x14, 0x16, 0x17, 0x18, 0x21, 0x22, - }; static int stac92xx_dmux_enum_info(struct snd_kcontrol *kcontrol, @@ -773,7 +773,8 @@ static unsigned int ref9205_pin_configs[ }; static unsigned int *stac9205_brd_tbl[STAC_9205_MODELS] = { - ref9205_pin_configs, + [STAC_REF] = ref9205_pin_configs, + [STAC_M43xx] = NULL, }; static const char *stac9205_models[STAC_9205_MODELS] = { @@ -784,6 +785,10 @@ static struct snd_pci_quirk stac9205_cfg /* SigmaTel reference board */ SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, "DFI LanParty", STAC_9205_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01f8, + "Dell Precision", STAC_M43xx), + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x01ff, + "Dell Precision", STAC_M43xx), {} /* terminator */ }; @@ -813,48 +818,52 @@ static int stac92xx_save_bios_config_reg return 0; } +static void stac92xx_set_config_reg(struct hda_codec *codec, hda_nid_t pin_nid, unsigned int pin_config) +{ + int i; + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, + pin_config & 0x000000ff); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, + (pin_config & 0x0000ff00) >> 8); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, + (pin_config & 0x00ff0000) >> 16); + snd_hda_codec_write(codec, pin_nid, 0, + AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + pin_config >> 24); + i = snd_hda_codec_read(codec, pin_nid, 0, + AC_VERB_GET_CONFIG_DEFAULT, + 0x00); + snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", pin_nid, i); +} + static void stac92xx_set_config_regs(struct hda_codec *codec) { int i; struct sigmatel_spec *spec = codec->spec; - unsigned int pin_cfg; - - if (! spec->pin_nids || ! spec->pin_configs) - return; - - for (i = 0; i < spec->num_pins; i++) { - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_0, - spec->pin_configs[i] & 0x000000ff); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_1, - (spec->pin_configs[i] & 0x0000ff00) >> 8); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_2, - (spec->pin_configs[i] & 0x00ff0000) >> 16); - snd_hda_codec_write(codec, spec->pin_nids[i], 0, - AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, - spec->pin_configs[i] >> 24); - pin_cfg = snd_hda_codec_read(codec, spec->pin_nids[i], 0, - AC_VERB_GET_CONFIG_DEFAULT, - 0x00); - snd_printdd(KERN_INFO "hda_codec: pin nid %2.2x pin config %8.8x\n", spec->pin_nids[i], pin_cfg); - } -} - -static void stac92xx_enable_eapd(struct hda_codec *codec) -{ - /* Configure GPIO0 as output */ + if (!spec->pin_configs) + return; + + for (i=0; i < spec->num_pins; i++) + stac92xx_set_config_reg(codec, spec->pin_nids[i], + spec->pin_configs[i]); +} + +static void stac92xx_enable_gpio_mask(struct hda_codec *codec,int gpio_mask,int gpio_data) +{ + /* Configure GPIOx as output */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DIRECTION, 0x00000001); - /* Configure GPIO0 as CMOS */ + AC_VERB_SET_GPIO_DIRECTION, gpio_mask); + /* Configure GPIOx as CMOS */ snd_hda_codec_write(codec, codec->afg, 0, 0x7e7, 0x00000000); - /* Assert GPIO0 high */ + /* Assert GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_DATA, 0x00000001); - /* Enable GPIO0 */ + AC_VERB_SET_GPIO_DATA, gpio_data); + /* Enable GPIOx */ snd_hda_codec_write(codec, codec->afg, 0, - AC_VERB_SET_GPIO_MASK, 0x00000001); + AC_VERB_SET_GPIO_MASK, gpio_mask); } /* @@ -2233,7 +2242,7 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); + stac92xx_enable_gpio_mask(codec, 0x00000001, 0x00000001); /* GPIO0 High = Enable EAPD */ err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { @@ -2258,7 +2267,7 @@ static int patch_stac9205(struct hda_cod static int patch_stac9205(struct hda_codec *codec) { struct sigmatel_spec *spec; - int err; + int err, gpio_mask, gpio_data; spec = kzalloc(sizeof(*spec), GFP_KERNEL); if (spec == NULL) @@ -2295,8 +2304,19 @@ static int patch_stac9205(struct hda_cod spec->mixer = stac9205_mixer; spec->multiout.dac_nids = spec->dac_nids; - stac92xx_enable_eapd(codec); - + + if (spec->board_config == STAC_M43xx) { + /* Enable SPDIF in/out */ + stac92xx_set_config_reg(codec, 0x1f, 0x01441030); + stac92xx_set_config_reg(codec, 0x20, 0x1c410030); + + gpio_mask = 0x00000007; /* GPIO0-2 */ + gpio_data = 0x00000005; /* GPIO0 High = EAPD, GPIO1 Low = DRM, GPIO2 High = Headphone Mute */ + } + else + gpio_mask = gpio_data = 0x00000001; /* GPIO0 High = EAPD */ + + stac92xx_enable_gpio_mask(codec, gpio_mask, gpio_data); err = stac92xx_parse_auto_config(codec, 0x1f, 0x20); if (!err) { if (spec->board_config < 0) {

2 1

[alsa-devel] Input/output error - Confusion regarding period size and interrupts
by Pharaoh . 16 Jul '07

16 Jul '07

Hi The hardware I am working on expects 8K bytes to be DMAed before giving an interrupt. Only after a block of 8K bytes is DMAed, I get an end of the block interrupt and I enqueue next 8K bytes for processing. The params look like these: static snd_pcm_hardware_t snd_omap_alsa_playback = { .info = (SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_MMAP), .formats = (SNDRV_PCM_FMTBIT_S16_LE), .rates = (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 ), .rate_min = 44100, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 8192*8, .period_bytes_min = 32, .period_bytes_max = 8 * 1024, .periods_min = 2, //ping pong buffer .periods_max = 128, .fifo_size = 0, }; I just copied it from another driver ;-). But now I am getting an input/output error after first block of data i.e. 8K bytes are DMAed. Does it have anything to do with the above struct? I have discussed this earlier too, but now attacking it once again after long gap. -pharaoh.

1 1

[alsa-devel] [Alsa-devel] xrun problem
by Gao, Jonathan 16 Jul '07

16 Jul '07

Hi, I am writing a PCM driver on a Soc for 2.6.10 kernel. The test code I'm using is from alsa-lib-1.0.6rc1/test/pcm.c. The runtime log gives me like this... ----------------------------------------------------------------- open callback: hw params callback: alloc result = 1, buffer bytes = 4608 /* --------- HW params ------------- */ format = 2, rate = 44100, channels= 2, period_size = 288, periods = 4, buffer_size = 1152 prepare callback /* --------- Runtime Info ---------- */ state = 1 (SNDRV_PCM_STATE_SETUP), hw_ptr = 0, appl_ptr = 0, avail_min = 288 avail_max = 0, hw_ptr_base = 0, hw_ptr_interrupt = 0 trigger callback: START Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 0, appl_ptr = 1152, avail_min = 288 avail_max = 0, hw_ptr_base = 0, hw_ptr_interrupt = 0 pointer callback: hw pointer: 288 (return value of pointer callback) Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 288, appl_ptr = 1152, avail_min = 288 avail_max = 288, hw_ptr_base = 0, hw_ptr_interrupt = 288 pointer callback: hw pointer: 576 (return value of pointer callback) Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 576, appl_ptr = 1152, avail_min = 288 avail_max = 576, hw_ptr_base = 0, hw_ptr_interrupt = 576 pointer callback: hw pointer: 864 (return value of pointer callback) Codec Do Copy: /* --------- Runtime Info ---------- */ state = 3 (SNDRV_PCM_STATE_RUNNING), hw_ptr = 864, appl_ptr = 1152, avail_min = 288 avail_max = 864, hw_ptr_base = 0, hw_ptr_interrupt = 864 pointer callback: hw pointer: 0 (return value of pointer callback) trigger callback: STOP prepare callback /* --------- Runtime Info ---------- */ state = 4 (SNDRV_PCM_STATE_XRUN), hw_ptr = 1152, appl_ptr = 1152, avail_min = 288 avail_max = 1152, hw_ptr_base = 1152, hw_ptr_interrupt = 1152 ----------------------------------------------------------------- The kernel hangs after prepare callback, from strace, I can see a endless ioctl(4, ..., ...) repeated. Does anyone have a few pointers on what is going wrong? Best Regards, Jonathan

2 1

Re: [alsa-devel] [PATCH] Fix request_irq error handling
by Ash Willis 16 Jul '07

16 Jul '07

Hah, um, ignore me. It seems that my c needs some work ;) In if(some_expression), I always though it is true if the expression is > 0, not != 0. If anything, that was a CodingStyle patch because it lets us see exactly what we're getting from request_irq but I can see that it is totally unnecessary :) > ----- Original Message ----- > From: "Ash Willis" <ashwillis(a)programmer.net> > To: alsa-devel(a)alsa-project.org > Subject: [alsa-devel] [PATCH] Fix request_irq error handling > Date: Sat, 14 Jul 2007 02:29:19 -0500 > > > request_irq returns 0 on success and a negative code on failure. > Many ALSA drivers incorrectly check for failure by testing for a > positive code instead of a negative one. > > Signed-Off-By: Ash Willis <ashwillis(a)programmer.net> > > > -- > We've Got Your Name at http://www.mail.com! > Get a FREE E-mail Account Today - Choose From 100+ Domains > << request_irq_error_handling.patch >> > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains

2 1

Re: [alsa-devel] No sound on Medion laptop (intel 82801G)
by albox＠free.fr 16 Jul '07

16 Jul '07

I have added a note for this bug on : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3160 I'm not really sure but it seems that the file /etc/modprobe.d/alsa-base is not loaded. http://ubuntuforums.org/showthread.php?p=3026300#post3026300 Any idea ? Thanks

1 0

[alsa-devel] Install of alsa 1.0.14.final on Fedora 7
by stan 15 Jul '07

15 Jul '07

On Fri, 13 Jul 2007 08:53:47 -0700 tdavis(a)dsl-only.net wrote: > For RPM based systems, I have a build environment for the drivers, > libs, and utils packages at > http://members.dsl-only.net/~tdavis/my-build.tar.bz2. You need to > be root to build them, but they won't interfere with your rpm > distribution. THe driver rpm will also make a backup of your > current drivers prior to installation. > > Tobin > Hi Tobin, I unpacked the archive and typed make. It created 4 RPMs. Two of them installed, alsa-lib and alsa-lib-devel. When I run a test on the drivers rpm, there are conflicts with the kernel headers already on the box. And I have no sound now, except in the audio configuration screen which is using the 14.rc3 packages. The packages are named with rh7 instead of fc7 as well. Do I have to run make install to get everything working, or is the header incompatability in the drivers a show stopper? Thanks.

2 5

[alsa-devel] MOTU PCI-424
by Per Eriksson 14 Jul '07

14 Jul '07

Hello, After talking to Jonathan Woithe about the development done to support MOTUs product families, Jonathan and I have reached the conclusion that ALSA support for the PCI-424 series of cards would mean a great benefit for Linux users to get high quality audio at a fair price. The PCI-424 card is used to drive the following products: MOTU 2408mk3 MOTU HD192 MOTU 24I/O The work will start this autumn, as soon as I get everything set-up, and the progress will be published on https://erinet.se/alsa/motu_pci_424/pci-424 specification.pdf The actual driver will for me be a later issue to solve, I hope to gather some understanding by reading the list and look through the releases. However, I would like some more information on how the Matrix is supposed to be used in the wiki system. Are the vendor pages only for supported devices, or also for devices concidered for being supported? Thanks, Per

1 0

[alsa-devel] [PATCH] Fix request_irq error handling
by Ash Willis 14 Jul '07

14 Jul '07

request_irq returns 0 on success and a negative code on failure. Many ALSA drivers incorrectly check for failure by testing for a positive code instead of a negative one. Signed-Off-By: Ash Willis <ashwillis(a)programmer.net> -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains

1 0

[alsa-devel] [PATCH] Add quirks for HP dx2200 and dx2250
by Claudio Matsuoka 14 Jul '07

14 Jul '07

This patch adds quirks for HP dx2200 and dx2250. They use the standard configuration for 3-stack ALC862.

1 0

[alsa-devel] Setting audio channels on ICH5 with SALSA
by Fausto Carvalho Marques SIlva 13 Jul '07

13 Jul '07

Setting audio channels on ICH5 with SALSA generates and: EINVAL Numbers of channels is 1. Does anyone have some ideias?? ( on ICH6 and ICH7 works well) thanks... Fausto Carvalho Marques Silva

3 6

[alsa-devel] [PATCH] Add quirk for HP Samba (ALC888)
by Claudio Matsuoka 13 Jul '07

13 Jul '07

Hi, This patch fixes 5.1 surround output for the HP Samba using the same quirk as the HP Lucknow. If HP machines are uniform in their wiring, I think we should rename ALC888_HP_NETTLE to ALC888_6ST_HP and ALC888_HP_LUCKNOW to ALC888_3ST_HP for generic HP 6stack and 3stack configurations. I'll probably receive more HP machines soon, so I'll see if this is the case. Best regards, Claudio

2 3

[alsa-devel] [PATCH] LG LW line-in capture
by Claudio Matsuoka 13 Jul '07

13 Jul '07

This patch adds line-in capture to LG LW (ALC880 codec). Tested with LW20.

2 2

[alsa-devel] hda-intel on latext macbook pro
by Steffen Schulz 13 Jul '07

13 Jul '07

Hi all, I wrote about a week ago about the problems with alsa on the latest macbook pro (rev3, with santa-rosa). As I described, model=w2jc sort of works for the macbooks. Front Mic and LineIn work. The LineIn can be configured to LineOut by setting 2-Channe to 6-Channel. I've seen where this is done in the code and it is also possible to have a working Mic in this configuration. As I have no idea how the driver or auto-detection or driver development/reverse engineering usually works, I tried to brute force the mixer controls. I found two more mute switches, one for the front speakers and one for the LineOut. But I was unable to activate the normal headphone output. Is there anyone here who can give me a hint what to search for? I noticed that when the Mic is disabled with the code that switches between 2ch and 6ch, the Mic control do not work anymore. Maybe it is the same for the headphone controls and some other settings? This is what I got so far: [ALC882_MBP] = { .mixers = { alc882_mbp_mixer, alc882_chmode_mixer }, .init_verbs = { alc882_init_verbs, alc882_eapd_verbs, alc880_gpio1_init_verbs }, .num_dacs = ARRAY_SIZE(alc882_dac_nids), .dac_nids = alc882_dac_nids, .num_channel_mode = ARRAY_SIZE(alc880_threestack_modes), .channel_mode = alc880_threestack_modes, .need_dac_fix = 1, .input_mux = &alc882_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, }, static struct snd_kcontrol_new alc882_mbp_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE ("Master Playback Switch", 0x0c, 2, HDA_INPUT), HDA_CODEC_MUTE ("Front Speaker Switch", 0x14, 0x00, HDA_OUTPUT), HDA_CODEC_MUTE ("Line Out Switch", 0x1a, 0x00, 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("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, 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("Mic Boost", 0x18, 0, HDA_INPUT), . . .[many controls that seem to do nothing] . { } }; I have no audio CD and no SPDIF-hardware, but the other controls do as they advertised. I would appreciate hints on finding the headphone controls. Hardware information can be seen in my last posts to the list. /Steffen -- Bildet Olsenbanden!

1 0

Re: [alsa-devel] RE : RE : M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 13 Jul '07

13 Jul '07

> With the help of Hakan I made a fix and updated the doc accordingly. > Hakan could you test this fix (to be applied after my last one) and > report if it is ok for you ? Yes, seem to work quite nicely now with AC3/DTS also in device_setup=0x00 mode. /Håkan

1 0

[alsa-devel] snd-hda-intel auto detect incorrect
by Matt Hirsch 13 Jul '07

13 Jul '07

I have an hp pavilion tx 1000 laptop running fedora 7. The auto detect feature of the snd-hda-intel dirver doesn't work on my sound card. As requested in Documentation/sound/alsa/ALSA-Configuration.txt I am forwarding the following to the developer mailing list: #uname -a Linux laptop 2.6.21-1.3228.fc7 #1 SMP Tue Jun 12 14:56:37 EDT 2007 x86_64 x86_64 x86_64 GNU/Linux # lspci -s 00:10.1 00:10.1 Audio device: nVidia Corporation MCP51 High Definition Audio (rev a2) # lspci -s 00:10.1 -nv 00:10.1 0403: 10de:026c (rev a2) Subsystem: 103c:30bf Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at c0000000 (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 # yum list alsa-\* Installed Packages alsa-lib.x86_64 1.0.14-0.4.rc3.fc7 installed alsa-lib.i386 1.0.14-0.4.rc3.fc7 installed alsa-utils.x86_64 1.0.14-0.7.rc2.fc7 installed With no options I get the following log message and the audio doesn't work: kernel: hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from BIOS... When loading the module as follows, the audio works: modprobe snd-hda-intel model=3stack I believe the card was detected as 3stack-dig, as the alsamixer options match when either no option or model=3stack-dig are given. Hope this helped. Thanks for all your hard work! Matt

2 2

[alsa-devel] Period Size in hw_params
by Paul Kavan 12 Jul '07

12 Jul '07

I finally have my basic audio applications working. I divided it into capture and playback and began with playback as I had a nice sinewave I could use to playback. Anyway, I think my hangup with getting the thing working was with the buffer and period information. My apps now do this: int periods = 2; /* Number of periods */ snd_pcm_uframes_t periodsize = 1000; /* Periodsize (bytes) */ /* Set number of periods */ if (snd_pcm_hw_params_set_periods(pcm_handle, hwparams, periods, 0) < 0) { fprintf(stderr, "Error setting periods.\n"); return(-1); } /* Set buffer size */ size = periodsize * periods; if (snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (size)>>2) < 0) { fprintf(stderr, "Error setting buffer size.\n"); return(-1); } I borrowed the period size from using aplay -v and seeing what it used. Setting number of periods to 2 was a little bit of guess work based on what others seem to use. I have looked at the alsa documentation and cannot see how I could have determined these numbers would have worked without trial and error. Can anyone either give me a explanation of how to determine these or a pointer to some decent documentation that talks about this? Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

1 0

[alsa-devel] [PATCH] hda: stac9202 mixer fix
by Matthew Ranostay 12 Jul '07

12 Jul '07

Added support for line out controls for STAC9202 cards, and fixed issue where master mixer control was being created twice for headphone and speaker outs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 273fac42ced9 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Wed Jul 04 15:17:38 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Wed Jul 11 22:13:06 2007 -0400 @@ -334,8 +334,6 @@ static struct snd_kcontrol_new stac9200_ }; static struct snd_kcontrol_new stac925x_mixer[] = { - HDA_CODEC_VOLUME("Master Playback Volume", 0xe, 0, HDA_OUTPUT), - HDA_CODEC_MUTE("Master Playback Switch", 0xe, 0, HDA_OUTPUT), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Input Source", @@ -1401,7 +1399,15 @@ static int stac92xx_auto_create_hp_ctls( continue; add_spec_dacs(spec, nid); } - + for (i = 0; i < cfg->line_outs; i++) { + nid = snd_hda_codec_read(codec, cfg->line_out_pins[i], 0, + AC_VERB_GET_CONNECT_LIST, 0) & 0xff; + if (check_in_dac_nids(spec, nid)) + nid = 0; + if (! nid) + continue; + add_spec_dacs(spec, nid); + } for (i = old_num_dacs; i < spec->multiout.num_dacs; i++) { static const char *pfxs[] = { "Speaker", "External Speaker", "Speaker2",

2 1

[alsa-devel] Can I mmap buffer allocated using SNDRV_DMA_TYPE_CONTINUOUS?
by Pharaoh . 12 Jul '07

12 Jul '07

Hi, I have allocated the DMA buffer using snd_pcm_lib_malloc_pages() and it was preallocated as: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data (GFP_KERNEL), 128 * 1024, 128 * 1024); Can I mmap such a dma buffer? I think yes... -pharaoh.

2 5

[alsa-devel] [RFC] SGI O2 MACE audio ALSA module
by TJ 12 Jul '07

12 Jul '07

Hi, It's been a while since I last posted something. Here is the latest version of my code. It is somewhat of a mess at the moment, but I plan to tidy it all up when it works better, any // comments are not permanent features. I have been using trial and error to find why it play too fast. I am glad to say that it does play sound correctly, bar one little issue that has me stuck at the moment, it plays too fast (for any sample frequency). Also how fast appears to depend on the player (tried aplay and ogg123). I have cc the ALSA dev too in the hope that between the two lists someone may spot something. Please can anyone who replies cc me. The patch was built against (applies to) linux-2.6.19.7 from linux-mips.org. It'll probably work on other versions. Notes on the module it self... There are two code 'paths' depending on sma_indirect=1|0, this var will be made a module load argument. When 0 ALSA will be writing directly to the MACE ring buffer, currently the state of this code is believed very broken I have not been working on it for a while. When 1 ALSA will be writing to a bounce buffer and the module will copy data to the ring buffer as per the original driver. Currently sma_indirect is hard coded 1 and it is this 'path' i'm currently interested in fixing. The module has some proc files for debugging, the contents of the ad1843 regs, and for values in in mace. There is a _spy version of the module (which wont exist in my final patch) that provides just these proc files without making any (init) changes to any of the hardware (ie read only). This allowed me to see what state the O2 jingle left the various bit in. The module has a limited mixer, which I want to do a lot more with in future, and the front volume buttons are supported. DAC1/ADC map to dev0 on the card, while DAC2 maps to dev1. I don't know what else to say, please have a listen and a look. It will play any samples 8k to 48k, and any bit format that ALSA knows about. Regards, Thorben

3 8

[alsa-devel] [PATCH] Make s3c24xx_i2s_set_clkdiv() change the correct bits
by Matt Reimer 12 Jul '07

12 Jul '07

Make s3c24xx_i2s_set_clkdiv() change the correct bits. Signed-off-by: Matt Reimer <mreimer(a)vpop.net> --- sound/soc/s3c24xx/s3c24xx-i2s.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index 8ca314d..39f0246 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -344,11 +344,11 @@ static int s3c24xx_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, DBG("Entered %s\n", __FUNCTION__); switch (div_id) { - case S3C24XX_DIV_MCLK: + case S3C24XX_DIV_BCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~S3C2410_IISMOD_FS_MASK; writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; - case S3C24XX_DIV_BCLK: + case S3C24XX_DIV_MCLK: reg = readl(s3c24xx_i2s.regs + S3C2410_IISMOD) & ~(S3C2410_IISMOD_384FS); writel(reg | div, s3c24xx_i2s.regs + S3C2410_IISMOD); break; -- 1.5.1.4

2 1

[alsa-devel] [PATCH] Add LG LW20 si3054 modem id
by Claudio Matsuoka 12 Jul '07

12 Jul '07

This patch adds the id of the LG LW20 laptop modem.

2 1

[alsa-devel] Using speaker-test for sound cards which dont support mmap
by Pharaoh . 12 Jul '07

12 Jul '07

Is it possible to use speaker-test for sound cards which do not support mmap? I straced speaker-test and it uses mmap call, can we bypass this?

3 2

[alsa-devel] Help with ALSA application development
by Paul Kavan 12 Jul '07

12 Jul '07

Hello: I am still trying to write my own basic application that captures and plays audio. I am working in an embedded linux environment and have the drivers working well. I can use arecord/aplay without any problem. However, I have been trying to follow some web examples for simple capture and playback without any success. Does anyone know of some sample code for S8 formatted, 8k rate capture and playback. I think my problem is two-fold. First is getting the hw-params set correctly. I have a function call that sets the interface up. I am trying to use the same params that aplay -v returns to me, but perhaps I have them wrong somehow. Secondly is buffering the samples. I am not sure I have that correct. Does anyone have some pointers on that? I have attached my code. If anyone has some time and insight into this, I would greatly appreciate it. Thanks. Paul

2 1

[alsa-devel] non-interleaved mode
by Steve Longerbeam 11 Jul '07

11 Jul '07

Hi, The driver for the sound engine I'm working on requires NONINTERLEAVED. Will ALSA core give my driver a full period of left-only samples, followed by a full period of right-only samples? Or are left/right samples split within a single period (ie. first half of period is left samples, second half is right samples)? Thanks, Steve

4 6

Re: [alsa-devel] Errors in application building
by Paul Kavan 11 Jul '07

11 Jul '07

On 7/3/07, Lee Revell <rlrevell(a)joe-job.com> wrote: > > Use gdb, strace is not useful for this > > On 7/3/07, Paul Kavan <pkavan(a)gmail.com> wrote: > > Hello all: > > > > Just tried to run with the warnings: > > > > > > > > > > > > > grhuser@pcL7:$ arm-linux-gcc -lasound -Wall -o grh_audio grh_audio.c > > > grh_audio.c: In function 'grh_capture': > > > grh_audio.c:112: warning: pointer targets in passing argument 3 of > > 'snd_pcm_hw_params_set_rate_near' differ in signedness > > > grh_audio.c: In function 'grh_playback': > > > grh_audio.c:207: warning: pointer targets in passing argument 3 of > > 'snd_pcm_hw_params_set_rate_near' differ in signedness > > > > > > Which comes from the code: > > > > > > 112: if ((err = snd_pcm_hw_params_set_rate_near (capture_handle, > > hw_params, &sample_rate, 0)) < 0) > > > > > > > > > 207: if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, > > hw_params, &sample_rate, 0)) < 0) > > > > However, I get a seg fault when I ran it. The strace of this is: > > > > /usr/sbin $ strace grh_audio > > execve("/usr/sbin/grh_audio", ["grh_audio"], [/* 7 vars */]) = 0 > > mmap2(NULL, 20, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) > = > > 0x40005000 > > stat("/etc/ld.so.cache", {st_mode=S_IFREG|0644, st_size=1150, ...}) = 0 > > open("/etc/ld.so.cache", O_RDONLY) = 3 > > mmap2(NULL, 1150, PROT_READ, MAP_SHARED, 3, 0) = 0x40006000 > > close(3) = 0 > > open("/usr/lib/libasound.so.2", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0755, st_size=762072, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0x\367\1\000"..., > > 4096) = 4096 > > mmap2(NULL, 794624, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > > 0x4000e000 > > mmap2(0x4000e000, 745564, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) > > = 0x4000e000 > > mmap2(0x400cc000, 15508, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > > 0xb6) = 0x400cc000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libgcc_s.so.1", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=44167, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0X\25\0\000"..., > > 4096) = 4096 > > mmap2(NULL, 65536, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > 0x400d0000 > > mmap2(0x400d0000, 28800, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x400d0000 > > mmap2(0x400df000, 548, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > 0x7) > > = 0x400df000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0000\253\0"..., > > 4096) = 4096 > > mmap2(NULL, 360448, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > > 0x400e0000 > > mmap2(0x400e0000, 302380, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) > > = 0x400e0000 > > mmap2(0x40132000, 5172, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > > 0x4a) = 0x40132000 > > mmap2(0x40134000, 16020, PROT_READ|PROT_WRITE, > > MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x40130 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libm.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=45812, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0t\22\0\000"..., > > 4096) = 4096 > > mmap2(NULL, 81920, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > 0x40138000 > > mmap2(0x40138000, 41428, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x40138000 > > mmap2(0x4014a000, 4100, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > 0xa) > > = 0x4014a000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libdl.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=8900, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0(\10\0\000"..., > > 4096) = 4096 > > mmap2(NULL, 40960, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > 0x4014c000 > > mmap2(0x4014c000, 5760, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x4014c000 > > mmap2(0x40155000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > 0x1) > > = 0x40155000 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libpthread.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=63852, ...}) = 0 > > mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, > 0) = > > 0x40007000 > > read(3, > > "\177ELF\1\1\1a\0\0\0\0\0\0\0\0\3\0(\0\1\0\0\0D<\0\0004"..., > > 4096) = 4096 > > mmap2(NULL, 106496, PROT_NONE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = > > 0x40156000 > > mmap2(0x40156000, 40952, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_FIXED, 3, > 0) = > > 0x40156000 > > mmap2(0x40168000, 21732, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, > > 0xa) = 0x40168000 > > mmap2(0x4016e000, 5788, PROT_READ|PROT_WRITE, > > MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x4016e0 > > close(3) = 0 > > munmap(0x40007000, 4096) = 0 > > open("/lib/libgcc_s.so.1", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=44167, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > open("/lib/libc.so.0", O_RDONLY) = 3 > > fstat(3, {st_mode=S_IFREG|0644, st_size=309840, ...}) = 0 > > close(3) = 0 > > munmap(0x40006000, 1150) = 0 > > stat("/lib/ld-uClibc.so.0", {st_mode=S_IFREG|0755, st_size=21096, ...}) > = 0 > > mprotect(0x40132000, 4096, PROT_READ) = 0 > > mprotect(0x4014a000, 4096, PROT_READ) = 0 > > mprotect(0x40155000, 4096, PROT_READ) = 0 > > mprotect(0x40168000, 4096, PROT_READ) = 0 > > mprotect(0x4000c000, 4096, PROT_READ) = 0 > > ioctl(0, SNDCTL_TMR_TIMEBASE or TCGETS, {B115200 opost isig icanon echo > > ...}) = 0 > > ioctl(1, SNDCTL_TMR_TIMEBASE or TCGETS, {B115200 opost isig icanon echo > > ...}) = 0 > > getpid() = 756 > > getrlimit(RLIMIT_STACK, {rlim_cur=8192*1024, rlim_max=RLIM_INFINITY}) = > 0 > > setrlimit(RLIMIT_STACK, {rlim_cur=2040*1024, rlim_max=RLIM_INFINITY}) = > 0 > > rt_sigaction(SIGRTMIN, {0x4015fd70, [], 0x4000000 /* SA_??? */}, NULL, > 8) = > > 0 > > rt_sigaction(SIGRT_1, {0x4015fc98, [RTMIN], 0x4000000 /* SA_??? */}, > NULL, > > 8) = 0 > > rt_sigaction(SIGRT_2, {0x4015f2d0, [], 0x4000000 /* SA_??? */}, NULL, 8) > = 0 > > rt_sigprocmask(SIG_BLOCK, [RTMIN], NULL, 8) = 0 > > rt_sigprocmask(SIG_UNBLOCK, [RT_1], NULL, 8) = 0 > > --- SIGSEGV (Segmentation fault) @ 0 (0) --- > > +++ killed by SIGSEGV +++ > > Process 756 detached > > > > > > I am still trying to locate why those warnings are popping up, but as > that > > part of the code is not called immediately, I am wondering why the seg > fault > > is occurring. Any clues? Lee: Have attempted to run this with gdb. However, I get: /usr/sbin $ gdb grh_audio dlopen failed on 'libthread_db.so.1' - File not found GDB will not be able to debug pthreads. GNU gdb 6.5 Copyright (C) 2006 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "arm-linux-uclibc"...(no debugging symbols found) (gdb) Trying to determine why...will let you know asap what I find. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 7

[alsa-devel] Query regarding sampling rates etc
by Pharaoh . 11 Jul '07

11 Jul '07

Hi, This is a very vague question, just thought of asking it after hours of debugging. I am testing my alsa driver, written for omap 730/850 based board. I am testing it by using aplay. Despite checking clocks, sampling rates etc many times and after doing many combinations thereof, I cant hear anything more than a noise which changes and becomes more like the original .wav file as I change various params like sampling rates/clocks etc. It becomes close to the actual thing, but not better. Any idea what are the params I should be tweaking? I know this is h/w dependent. -pharaoh

3 3

Re: [alsa-devel] RE : M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 11 Jul '07

11 Jul '07

> You'll find attached an equivalent fix that checks alt-settings > instead of device_setup param: Can you try it Hakan ? Ok, patch tested now. Seem to work. /Håkan

1 0

[alsa-devel] Alsa lib configure asoundlib.h
by Fausto Carvalho Marques SIlva 11 Jul '07

11 Jul '07

In salsa library(this library really saved my life, from am alsa-lib of 600K to 45K!!!), developed by Takashi Iwai, the creation of asoundlib.h is dynamic and based on the configure options ( --disable-mixer will remove mixer.h from asoundlib.h). Can alsa-lib do the same??I think it is a good "cleanup" way of doing client header files. Example from configure.ac from salsa-0.0.6: " echo "Creating asoundlib.h..." cp "$srcdir"/src/asoundlib-head.h src/asoundlib.h test "$mixer" = "yes" && echo "#include \"mixer.h\"" >> src/asoundlib.h test "$rawmidi" = "yes" && echo "#include \"rawmidi.h\"" >> src/asoundlib.h test "$hwdep" = "yes" && echo "#include \"hwdep.h\"" >> src/asoundlib.h test "$timer" = "yes" && echo "#include \"timer.h\"" >> src/asoundlib.h test "$sndconf" = "yes" && echo "#include \"conf.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seq_event.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seq.h\"" >> src/asoundlib.h test "$sndseq" = "yes" && echo "#include \"seqmid.h\"" >> src/asoundlib.h cat "$srcdir"/src/asoundlib-tail.h >> src/asoundlib.h " Fausto Carvalho Marques Silva

4 11

[alsa-devel] Mercurial repos
by Adrian McMenamin 11 Jul '07

11 Jul '07

Are the browsable repos broken - they don't seem to have been updated for days, though I've been told my patch has been applied.

3 2

Re: [alsa-devel] Writing a new plugin for ALSA
by Nobin Mathew 11 Jul '07

11 Jul '07

On 7/10/07, Takashi Iwai <tiwai(a)suse.de> wrote: > At Tue, 10 Jul 2007 17:45:03 +0530, > Sudhanshu Saxena wrote: > > > > Hi Takashi, > > > > I am working on this issue along with Nobin, for multiple multiple streams (at > > different rates) playback based on a priority. > > > > I attempted suspend and resume returning -ESTRPIPE, but it has two issues > > 1) All applications may need a change in Suspend logic specific to wait for higher > > priority stream to run to completion. > > Basically every app is _required_ to handle of suspend/resume > properly. > > > 2) I have to still find out a mechanism to reset the hardware settings are reconfigure > > it for second stream. > > snd_pcm_hw_reset has an access to reset using IOCTL "SNDRV_PCM_IOCTL_RESET" , I hope > > that can help, I am not > > very sure on this. > > Well, I don't think the plugin itself requires the reset. > > > As mentioned by Nobin we want to skip any upsampling caused when we use the dmix > > (rate) plugin. > > I was trying to find the way out, with a plugin with following changes: > > 1) All streams will open pcm with a priority. (Pre decided for the system) > > 2) Plug-in will have shared memory to hold the priority list of all applications which > > are accessing sound. > > 3) At any instance only one stream will be active. > > 4) All applications will hold the context of HW params and SW params (accessible > > through apis plugin will support). > > 5) If an application with a higher priority calls snd_pcm_open shared memory with the > > priority list will be updated and > > when the currently active stream attempts to re-write the data it will check for > > priority list. This will cause saving of params, drain of stream and reset of hw as > > mentioned using above IOCTL and the active stream would be changed to > > currently highest in the priority list. > > 6) The higher priority stream can then start writing data, on completion of access > > similar sequence of drain, hw reset will be caused and the active stream will be > > changed to next in the priority list. > > 7) On resume of lower priority stream the hw params will be set again. > > The question is how the application decides which stream to use. > If it can open a PCM with a different name (to snd_pcm_open call), and > if you have three individual PCM devices, the whole switching can > be done inside the kernel driver. That is, when an SCO stream is > opened during audio stream(s) is running, the driver triggers SUSPEND > mode for the audio stream. Then the app will stop and restart the > stream by itself. This means you don't need any change in user-space > side but needs some additions in the driver side. We thought about implementing this in the kernel driver, but after some investigation, we found that we are serializing the driver so much by providing three interfaces (audio, sco and Nav). say pcmC0D0p/c, pcmC0D1p/c and pcmC0D2p/c. Consider the case of two apps audio and sco. We have two interfaces for this pcmC0D0p/c and pcmC0D1p/c priority is audio < sco so interface priority is pcmC0D0p/c < pcmC0D1p/c. Assume that audio is already opened. When we open sco interface audio interface can be in any state, it i can be in paused state or it is about to close etc. So we will have so many states of application in the kernel driver. we need to maintain all these states of application in the kernel driver. Say tomorrow one more application came in to picture, then this complexity will increase. Locking of states of streams is also required in this case. correct me if I am going wrong > > BTW, any reason to drop Cc to alsa-devel ML? Such an open discussion > is very important for opensource development. I think Sudhanshu missed it, I am copying to alsa-devel > > > Takashi > thanks Nobin mathew

2 3

[alsa-devel] [PATCH] hda-codec - Add VIA HDA to si3054
by Claudio Matsuoka 10 Jul '07

10 Jul '07

The attached patch is needed to proper recognition of the si3054 modem in the Clevo m540 laptop. Thanks, Claudio

1 0

Re: [alsa-devel] default setting not fully working with new ALSA device
by Tobin Davis 10 Jul '07

10 Jul '07

Try this patch. It adds support for both ALC883 and HDMI ports on your system. Summary: HDA-Intel - Add support for MSI K9AGM2-FIH motherboard This patch adds support for the MSI K9AGM2-FIH on-board audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> On Thu, 2007-07-05 at 18:34 +0200, Peter Matthias wrote: > The Board is a MSI K9AGM2-FIH, Sockel AM2, ATI RS690G, mATX, PCIe > > The script gave following output: > "Your ALSA information is located at http://pastebin.ca/604529" > > Other drivers are statically linked in my kernel. > > Thank you for your work! In case of any questions, contact me. > > Peter > > > Am Mittwoch, 4. Juli 2007 schrieben Sie: > > Could you post the output from > > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh ? This > > script will give us more detailed information on what sound chip it is. > > We will also need to know the brand of board, for the driver file. > > > > Thanks, > > > > Tobin > > > > On Wed, 2007-07-04 at 16:48 +0200, Peter Matthias wrote: > > > Hello, > > > > > > I have got a 690G board with following lspic -nv output: > > > > > > 01:05.2 0403: 1002:7919 > > > Subsystem: 1002:7919 > > > Flags: bus master, fast devsel, latency 64, IRQ 19 > > > Memory at fe9e8000 (64-bit, non-prefetchable) [size=16K] > > > Capabilities: [50] Power Management version 2 > > > Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 > > > Enable- > > > > > > With the default setting, microphone does not work. With module > > > parameter "modprobe -i snd-hda-intel model=6stack-dig" microphone works > > > well. Please add to default setting. It was written in Linux Kernel > > > documentation to report this issue here. > > > > > > Thank you, > > > Peter > > > _______________________________________________ > > > Alsa-devel mailing list > > > Alsa-devel(a)alsa-project.org > > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Tobin Davis <tdavis(a)dsl-only.net>

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Re: [alsa-devel] Intel hda ICH7 not worked anymore after using wine (SOLVED!)
by Xavi de Blas 10 Jul '07

10 Jul '07

It worked!!! Thanks a lot I thought my hardware was broken! 2007/7/9, Tobin Davis <tdavis(a)dsl-only.net>: > > Patch needs to be applied as follows: > > cd alsa-driver-1.0.14/alsa-kernel > patch -p1 < LenovoX60S.patch > cd .. > > Then you can rerun configure && make && make install. > > Tobin > > On Mon, 2007-07-09 at 17:25 +0200, Xavi de Blas wrote: > > 2007/7/9, Tobin Davis <tdavis(a)dsl-only.net>:> Sorry for missing this thread. I just woke up.>> I looked at your first pastebin output, and the actual audio codec shows> up as 0x51d41981, when it should be 0x11d41981. This is why the Analog> driver patch isn't detecting it.>> Can you try this patch against alsa-driver-1.0.14? It should detect> this codec. Still not sure why it is misconfigured, unless there is a> bios issue.> > I'm afraid, probably when i tried the Linuxant codecs for the modem it broke thebios. Not sure, but sound worked, and continued working some rebootslater. But when i tried a few things with wine and rebooted, soundnever worked again > When i aply the patch (probably i don't do it good) there are problems: > > xavier@corall:/usr/local/src/alsa-src/alsa-driver-1.0.14$ catLenovoX60S.patch | patch -p1patching file pci/hda/patch_analog.cHunk #1 FAILED at 3402.1 out of 1 hunk FAILED -- saving rejects to file pci/hda/patch_analog.c.rej > > xavier@corall:/usr/local/src/alsa-src/alsa-driver-1.0.14/pci/hda$ catpatch_analog.c.rej*************** struct hda_codec_preset snd_hda_preset_a*** 3402,3407 **** { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a },--- 3402,3408 ---- { .id = 0x11d41882, .name = "AD1882", .patch = patch_ad1882 }, { .id = 0x11d41884, .name = "AD1884", .patch = patch_ad1884 }, { .id = 0x11d41981, .name = "AD1981", .patch = patch_ad1981 },+ { .id = 0x51d41981, .name = "AD1981", .patch = patch_ad1981 }, { .id = 0x11d41983, .name = "AD1983", .patch = patch_ad1983 }, { .id = 0x11d41984, .name = "AD1984", .patch = patch_ad1984 }, { .id = 0x11d41986, .name = "AD1986A", .patch = patch_ad1986a }, > > The patch is ok? > Thanks > (Tobin sorry for sending messge two times) > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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[alsa-devel] Writing a new plugin for ALSA
by Nobin Mathew 10 Jul '07

10 Jul '07

Hi, I am trying to write a new ALSA plugin, like dmix. My intend is to give priority to streams, i will not do up sample and down sample as we do in the case of dmix. If we do up sample and down sample then I think it will take lot of cpu. In my case I have three streams audio, bluetooth sco and Navigation sound data. priority is audio < sco < Nav audio rate is 44100hz, sco rate is 8000hz when sco is up audio should pause playback and audio should continue when sco is finished. I dont want to do software mixing. Thats why I am trying to write this new plugin. Assume that audio is active first, then sco comes to active state. So what i will do is in the library i will put audio app in sleep(), and re-configure the pcm for sco. Once sco is finished i will re-configure pcm for audio again like wise. so my question is there any api in the alsa-lib to take a copy of pcm structures, so when audio comes back gain i can use these saved pcm conf to that. Is there any api in the alsa-lib to reset the pcm structure to initialized state, means state just after open. Thanks Nobin Mathew

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[alsa-devel] [PATCH] snd-usb-audio: Add another QuickCam ID
by Ritesh Raj Sarraf 10 Jul '07

10 Jul '07

Hi Clemens, Can you push this for 2.6.23 ? Or is there anything else missing in the patch ? Thanks, Ritesh =================================================== On Friday 08 June 2007, Clemens Ladisch wrote: > Ritesh Raj Sarraf wrote: > > dell_xpsm1210_quickcam.patch > > Please provide a Signed-off-by line. > > I guess this patch is for the integrated webcam of the Dell XPS M1210? > > Hi Clemens, I've regenerate the patch properly this time. And yes, this patch is for the integrated webcam shipped with M1210. PS: Request you to CC me if there are any comments. I'm not subscribed to the list. Thanks, Ritesh -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention." -- Ritesh Raj Sarraf RESEARCHUT - http://www.researchut.com "Necessity is the mother of invention."

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[alsa-devel] ALC882 on Abit AW9D-MAX
by Dan Aloni 10 Jul '07

10 Jul '07

Hello, Yesterday I posted this bug in the bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3013 I have received no reponse yet. I'd like to cooperate on solving this. Thanks, -- Dan Aloni XIV LTD, http://www.xivstorage.com da-x (at) monatomic.org, dan (at) xiv.co.il

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[alsa-devel] Unable to find an usable access for 'default'
by Timur Tabi 10 Jul '07

10 Jul '07

I'm developing a new ASOC driver, and I get this error message: ALSA lib pcm_plug.c:729:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'default' Can someone tell me what this means? FYI, I'm in the early stages of testing my driver, and before I started I knew nothing about ALSA. This is on an embedded system, so I don't know if my ALSA installation is correct. Am I missing some kind of configuration file? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] Intel hda ICH7 not worked anymore after using wine
by Xavi de Blas 09 Jul '07

09 Jul '07

Hello, I installed Wine on my Ubuntu Feisty on IBM-lenovo X60s, and after reboot, sound never worked again. I tried to activate and to hide modem on the bios. I tried to stop machine, take off battery, wait 10s, put battery again, and boot without the power cable connected... and i tried lots of things from ubuntuforums. I reinstalled feisty and sound doesn't work. I used alsa-info.sh script, here is the result: http://pastebin.ca/610317 Should the new kernel 2.6.22 help me? http://kernelnewbies.org/Linux_2_6_22#head-0d133cda430e5d18f1c72dd3cfaa6a92… Thanks

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Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by Takashi Iwai 09 Jul '07

09 Jul '07

At Mon, 25 Jun 2007 15:16:41 -0500, Matt Mullins wrote: > > On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > Yes, single_cmd mode is used always. > > That's bad. > > I believe we have established that. What can I do about it? *Can* I > do anything about it? What functions should I start debugging? Has > this happened with other codecs before? What documentation will I > need or want other than the Intel HDA and STAC9205 specifications? Similar problems have been seen in some laptops, too. I vaguely remember that they need some ACPI-related boot options to fix it somehow. I think it's likely irrelevant with the codec chip, but rather a problem of the controller side. > I'm definitely willing to help work on this, as not many people have > this hardware. I'm just new to the sound system, so I'm still trying > to figure out what-calls-what. Takashi

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[alsa-devel] stac9200 speaker/hp not detected
by Brian Hinz 09 Jul '07

09 Jul '07

Hi, I'm trying to get the sound working on my Gateway S-7110M (a.k.a. NX100X?). Using the ini files from the windows driver, I created the following patch (applied to both the hg tree, and 1.0.14): --- a/alsa-kernel/pci/hda/patch_sigmatel.c 2007-07-03 21:59:21.000000000 -0400 +++ b/alsa-kernel/pci/hda/patch_sigmatel.c 2007-06-04 03:28:54.000000000 -0400 @@ -39,6 +39,7 @@ enum { STAC_REF, + STAC_M4, STAC_9200_MODELS }; @@ -453,12 +452,19 @@ 0x02a19020, 0x01a19021, 0x90100140, 0x01813122, }; +static unsigned int stac9200_M4_pin_configs[8] = { + 0x400000fe, 0x00451030, 0x400100f0, 0x90110010, + 0x400100f1, 0x02a1902e, 0x500000f2, 0x500000f3, +}; + static unsigned int *stac9200_brd_tbl[STAC_9200_MODELS] = { [STAC_REF] = ref9200_pin_configs, + [STAC_M4] = stac9200_M4_pin_configs, }; static const char *stac9200_models[STAC_9200_MODELS] = { [STAC_REF] = "ref", + [STAC_M4] = "m4", }; static struct snd_pci_quirk stac9200_cfg_tbl[] = { @@ -494,6 +486,9 @@ SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f5, "Dell Inspiron 1501", STAC_REF), + /* Gateway */ + SND_PCI_QUIRK(0x107b, 0x0205, "Gateway S-7110M", STAC_M4), + /* Panasonic */ SND_PCI_QUIRK(0x10f7, 0x8338, "Panasonic CF-74", STAC_REF), however it still does not work. I can see in dmesg that the speaker and headphones are not detected, but I can't seem to figure out why: [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:1810: hda_codec: model 'm4' is selected [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0xe/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [95006.300000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x0, fmic=0x10, line=0x0, fline=0x0, cd=0x0, aux=0x0 [95006.432000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [95006.432000] ALSA /home/bphinz/alsa/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) This laptop has only an internal speaker, a headphone out, and a mic in (unless there are other outs enabled when the dock is connected). The windows INI seems to be setting some gpio bits but I don't understand how to map the ini syntax into the alsa driver (or if it needs to be...) [Models] HDAUDIO\FUNC_01&VEN_8384&DEV_7690=this Default=this [CopyFiles.32] stacgui.cpl = 0x25 stlang.dll = 0x25 stacsv.exe = 0x25 [CopyFiles.64] stacui64.cpl = 0x25 stlang64.dll = 0x25 stacsv64.exe = 0x25 [LaunchProgramsAfter.32] STacSV -r [LaunchProgramsAfter.64] STacSV64 -r [HKLM\Software\Microsoft\Windows\CurrentVersion\Run.32] [HKLM\Software\Microsoft\Windows\CurrentVersion\Run.64] [DelRegKeys] HKR\Settings=Yes HKCU\Software\Sigmatel\Settings=Yes HKLM\Software\Sigmatel\Presets=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel Audio=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel C-Major Audio=Yes HKCU\Software\Microsoft\Windows\CurrentVersion\Applets\Volume Control\SigmaTel High Definition Audio=Yes [HKLM\Software\Sigmatel\GlobalState] [HKR] CodecName = "STAC9200" [HKR\Settings] TraceFlags = dword: 0x89B3 CPL_SaveExtraPower = hex: 0 CPL_UserControlsFixedDigitalOutRate = hex: 0 InitVerbs = hex: 0x03, 0xa0, 0x43, 0x01 9200MixerCompensation1PcmMode = hex: 2 9200MixerCompensation2PcmMode = hex: 3 CodecAccessAttempts = dword:00000352 [HKR\Settings\GPIO] 03 = hex: 0x85 AntiPopGpo = hex: 3,2 [HKR\Settings\pin] [HKR\Settings\Pin\08] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\09] CfgDflt = dword: 0x00451030 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400100F0 [HKR\Settings\Pin\0E] CfgDflt = dword: 0x90110010 Action = hex: 81,09 [HKR\Settings\Pin\0F] CfgDflt = dword: 0x400100F1 [HKR\Settings\Pin\10] CfgDflt = dword: 0x02A1902E [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000F2 [HKR\Settings\Pin\12] CfgDflt = dword: 0x500000F3 [HKR\Settings\Connselector] [HKR\settings\filter\Speaker] UniqueID = hex: 0 LegacyIoctlHere = hex: 1 Pins = hex: 0E RenderConverter = hex: 2 Associations = hex: 1 DigitalVolumeControl = hex: 0 EnableSetFormatSupport = hex: 1 EnableHwSpeakerConfigurationSupport = hex: 0 EnabledPcmRenderBitsAndRates = dword: 0x000FFFFE MasterControlsIndependentMixerInputs = hex: 0 EnableVendorPcBeep = hex: 1 PcBeepVolume = dword: 0 PcBeepMute = hex: 1 EnableInputMonitor = hex: 1 RecMonitorMute = hex: 1 EnabledNonPcmBitsAndRates = dword: 0x20040 BiosInterfaceAddress = dword: 0x000DF7E0 [HKR\settings\filter\SpdifOut] UniqueID = hex: 1 LegacyIoctlHere = hex: 0 Pins = hex: 09 SpdifOutConverter = hex: 5 SpdifMaster = hex: Associations = hex: 3 EnabledPcmRenderSpdifBitsAndRates = dword: 0x000FFFFF DisableSPDIFControl = hex: 0 [HKR\Settings\filter\MuxedIn] UniqueID = hex: 2 LegacyIoctlHere = hex: 0 Pins = hex: 10 CaptureConverter = hex: 3 EnableCompositeMuxAndAdcCtrl = hex: 0 Associations = hex: 2 MicBoost = hex: 1 DisableMicBoost = hex: 0 EnabledPcmCaptureBitsAndRates = dword: 0x000FFFFF MicInRecVolume = dword: 0x000a8000 [HKR\Settings\filter\MuxedIn\MaxLvlLimiters] MicBoostLevel = dword: 0x000c0000 [HKR\Settings\filter\MuxedIn\TopoNames] [HKLM\software\SigmaTel\Apo\GFX\Speaker\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{637c490d-eee3-4c0a-973f-371958802da2}" [HKLM\software\SigmaTel\Apo\LFX\Speaker\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{62dc1a93-ae24-464c-a43e-452f824c4250}" [HKLM\software\SigmaTel\Apo\GFX\SpdifOut\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{637c490d-eee3-4c0a-973f-371958802da2}" [HKLM\software\SigmaTel\Apo\LFX\SpdifOut\plugins\MSAPO] idEffect = dword: 50 idEffectType = dword: 50 load = dword: 1 order = dword: 5 ApoClsId = "{62dc1a93-ae24-464c-a43e-452f824c4250}" [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\SysEq] Load = dword: 1 IdEffect = dword: 101 IdEffectType = dword: 1 Order = dword: 1100 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq] NumBands = dword:12 RegVersion = dword: 1 Bypass = dword: 0 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band0] idxBand = dword: 0 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 1 GroupId = dword: 0 Freq =dword: 150 GainDbx1000 = dword: -3000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band1] idxBand = dword: 1 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 300 GainDbx1000 = dword: -3000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band2] idxBand = dword: 2 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 500 GainDbx1000 = dword: -2000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band3] idxBand = dword: 3 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 750 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band4] idxBand = dword: 4 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 1000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band5] idxBand = dword: 5 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 2000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band6] idxBand = dword: 6 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 8000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band7] idxBand = dword: 7 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 12000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band8] idxBand = dword: 8 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 15000 GainDbx1000 = dword: 0 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band9] idxBand = dword: 9 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band10] idxBand = dword: 10 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 0 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\SysEq\Band11] idxBand = dword: 11 Flags = dword: 0 ChannelMask = dword: 262143 Type = dword: 2 GroupId = dword: 0 Freq =dword: 17000 GainDbx1000 = dword: -1000 Bandwidthx1000 = dword: 1000 [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\Compressor] Load = dword: 1 IdEffect = dword: 5 Order = dword: 1150 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\Compressor] CompRatio= dword: 200 CompAttack= dword: 9910 CompRelease= dword: 9910 CompThreshDb= dword: -600 MakeupGainDb= dword: 200 GateRatio= dword: 600 GateAttack= dword: 9990 GateRelease= dword: 9990 GateThreshDb= dword: -9000 Flags= dword: 8 [HKLM\Software\Sigmatel\APO\GFX\Speaker\plugins\Limiter] Load = dword: 1 IdEffect = dword: 6 Order = dword: 1155 [HKLM\Software\Sigmatel\APO\GFX\Speaker\presets\_Initial\Limiter] CompRatio= dword: 3700 CompAttack= dword: 9900 CompRelease= dword: 9900 CompThreshDb= dword: 0 MakeupGainDb= dword: 0 GateRatio= dword: 100 GateAttack= dword: 9000 GateRelease= dword: 9000 GateThreshDb= dword: -9000 Flags= dword: 10 [HKLM\SOFTWARE\SigmaTel\STSysTray\1.0\EventExpressions] Version=dword:1 [HKLM\SOFTWARE\SigmaTel\STSysTray\1.0\EventExpressions\JackPresenseD] Action1="if(JackState(13) == 1,(BypassSystemEq(1)),(BypassSystemEq(0)))" Action2="if(JackState(13) == 1,(BypassSystemComp(1)),(BypassSystemComp(0)))" [HKLM\SOFTWARE\Sigmatel\STacGUI] Config_Tab = dword: 0x8 Config_Spk = dword: 0x0 Config_Adv = dword: 0x01 [HKLM\SOFTWARE\Sigmatel\STacGUI\XML] Nconfigs = dword: 0x0 here's the output of /proc/asound/card0/codec#0: Codec: SigmaTel STAC9200 Address: 0 Vendor Id: 0x83847690 Subsystem Id: 0x100 Revision Id: 0x102201 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: N/A Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0xd0401: Stereo Power: 0x0 Node 0x03 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x0a Node 0x04 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x08 Node 0x05 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x1e0]: 44100 48000 88200 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x06 [Vendor Defined Widget] wcaps 0xf30201: Stereo Digital Node 0x07 [Audio Selector] wcaps 0x300901: Stereo Connection: 3 0x02* 0x08 0x0a Node 0x08 [Pin Complex] wcaps 0x430681: Stereo Digital Pincap 0x0810024: IN EAPD Detect Pin Default 0x400000fe: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Power: 0x0 Node 0x09 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x00451030: [Jack] SPDIF Out at Ext N/A Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 2 0x05* 0x0a Node 0x0a [Audio Selector] wcaps 0x30090d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0d 0x0d] Connection: 1 0x0c Node 0x0b [Audio Selector] wcaps 0x300105: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x80 0x80] Connection: 1 0x07 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=0 Amp-Out vals: [0x04 0x04] Connection: 5 0x10* 0x0f 0x0e 0x0d 0x12 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x400100f0: [N/A] Line Out at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x0b Node 0x0e [Pin Complex] wcaps 0x400181: Stereo Pincap 0x083f: IN OUT HP Detect Pin Default 0x90110010: [Fixed] Speaker at Int N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0837: IN OUT Detect Pin Default 0x400100f1: [N/A] Line Out at Ext N/A Conn = 1/8, Color = Unknown Pin-ctls: 0x20: IN Connection: 1 0x0b Node 0x10 [Pin Complex] wcaps 0x400181: Stereo Pincap 0x081737: IN OUT Detect Pin Default 0x02a1902e: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x0b Node 0x11 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x00] Pincap 0x0810: OUT Pin Default 0x500000f2: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x13 Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x500000f3: [N/A] Line Out at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Node 0x13 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x14 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Thanks for any assistance! -brian

2 7

[alsa-devel] Question about write asoc driver
by Roy Huang 09 Jul '07

09 Jul '07

Hi guys, I am write a codec driver under asoc frame work for ad1980 by study the sample driver wm9712.c. I find there are some control item likes "out3 switch" to control 16h:15. There are also some widget item likes "out 3 PGA" to control 24:5. How can I determine a register bit should be a control item or widget item? Thanks ahead. -Roy

2 1

Re: [alsa-devel] [Pkg-alsa-devel] [PATCH] amixer and special 'front...' channels
by Elimar Riesebieter 09 Jul '07

09 Jul '07

This isn't Ubuntu/Debian specific and should be applied upstream? On Sat, 07 Jul 2007 the mental interface of David Decotigny told: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > > Hi, > > Imagine you have this kind of control: > > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Mic' > > ... and you want to set it to 'Front Mic': > > shell> amixer sset 'Input Source' 'Front Mic' > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Mic' > > This does not work ! Because amixer will think that 'Front' is a > modifier for 'Mic'. > > Attached is a patch against alsa-utils-1.0.13-1ubuntu5 that solves this. > With this patch, one can bypass the modifier parsing: one simply escapes > the parameter with one or more backslash(es), and that's all: > > shell> amixer sset 'Input Source' '\Front Mic' > Simple mixer control 'Input Source',0 > Capabilities: enum > Items: 'Mic' 'Front Mic' 'Line' > Item0: 'Front Mic' > > What this patch also does, is that it adds correct identification of > items. With the original version, given the following enumeration: > foo foobar > Then the parameter 'foobar' will be identified as 'foo', not 'foobar'. > The patch should solve this potential problem (not tested though). > > Bye, > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.6 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iD8DBQFGj/X0ld7vhusVrCERAjIZAJ9w9GLD+QBwDCaTs32hpW6i9nVMfgCfdS5w > jhipZ2jm87fq4zx4f5Q9Vsc= > =M2KP > -----END PGP SIGNATURE----- > diff -ru alsa-utils-1.0.13/amixer/amixer.1 alsa-utils-1.0.13-mine/amixer/amixer.1 > --- alsa-utils-1.0.13/amixer/amixer.1 2006-09-29 04:53:26.000000000 -0700 > +++ alsa-utils-1.0.13-mine/amixer/amixer.1 2007-07-07 13:09:04.000000000 -0700 > @@ -44,11 +44,14 @@ > The parameters \fIcap, nocap, mute, unmute, toggle\fP are used to > change capture (recording) and muting for the group specified. > > -The optional modifiers can be put as extra parameters to specify > -the stream direction or channels to apply. > -The modifiers \fIplayback\fP and \fIcapture\fP specify the stream, > -and the modifiers \fIfront, rear, center, woofer\fP are used to specify > -channels to be changed. > +The optional modifiers can be put as extra parameters to specify the > +stream direction or channels to apply. The modifiers \fIplayback\fP > +and \fIcapture\fP specify the stream, and the modifiers \fIfront, > +rear, center, woofer\fP are used to specify channels to be changed. If > +there is a confusion between the parameter and these modifiers, then > +any character in the parameter may be escaped with a backslash to > +disambiguate: '\\Front Mic' will be understood as "parameter 'Front > +Mic'", and not as "modifier 'Front' applied to 'Mic' parameter". > > A simple mixer control must be specified. Only one device can be controlled > at a time. > diff -ru alsa-utils-1.0.13/amixer/amixer.c alsa-utils-1.0.13-mine/amixer/amixer.c > --- alsa-utils-1.0.13/amixer/amixer.c 2007-07-07 13:03:39.000000000 -0700 > +++ alsa-utils-1.0.13-mine/amixer/amixer.c 2007-07-07 12:59:19.000000000 -0700 > @@ -1333,6 +1333,16 @@ > return def; > } > > +/** Transform '\a\b c \de' into 'ab c de' */ > +static void unescape(char * arg) > +{ > + char * c; > + for (c = arg ; arg && *arg ; arg ++) > + if (*arg != '\\') > + *(c++) = *arg; > + *c = '\0'; > +} > + > static int get_enum_item_index(snd_mixer_elem_t *elem, char **ptrp) > { > char *ptr = *ptrp; > @@ -1347,7 +1357,7 @@ > if (snd_mixer_selem_get_enum_item_name(elem, i, sizeof(name)-1, name) < 0) > continue; > len = strlen(name); > - if (! strncmp(name, ptr, len)) { > + if (!strncmp(name, ptr, len) && (ptr[len] == '\0')) { > if (! ptr[len] || ptr[len] == ',' || ptr[len] == '\n') { > ptr += len; > *ptrp = ptr; > @@ -1428,6 +1438,7 @@ > dir = dir_mask(&ptr, dir); > if (*ptr == '\0') > continue; > + unescape(ptr); > multi = (strchr(ptr, ',') != NULL); > optr = ptr; > for (chn = 0; chn <= SND_MIXER_SCHN_LAST; chn++) { > _______________________________________________ > Pkg-alsa-devel mailing list > Pkg-alsa-devel(a)lists.alioth.debian.org > http://lists.alioth.debian.org/mailman/listinfo/pkg-alsa-devel -- "Talking much about oneself can also be a means to conceal oneself." -Friedrich Nietzsche

2 1

[alsa-devel] [PATCH] usb-audio: another Logitech camera/microphone ID match
by Daniel Drake 09 Jul '07

09 Jul '07

From: George Shapovalov <george(a)gentoo.org> Another case of ALSA ticket #3040: we need an explicit USB ID match for another Logitech camera/microphone device. Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux/sound/usb/usbquirks.h =================================================================== --- linux.orig/sound/usb/usbquirks.h +++ linux/sound/usb/usbquirks.h @@ -57,6 +57,15 @@ USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, + .idProduct = 0x08ae, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, +{ + .match_flags = USB_DEVICE_ID_MATCH_DEVICE | + USB_DEVICE_ID_MATCH_INT_CLASS | + USB_DEVICE_ID_MATCH_INT_SUBCLASS, + .idVendor = 0x046d, .idProduct = 0x08f0, .bInterfaceClass = USB_CLASS_AUDIO, .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL

2 1

[alsa-devel] M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 09 Jul '07

09 Jul '07

Hi all ! I don't know if this is already known (?), but the M-Audio Audiophile USB seem to be possible to run in native S16_LE mode if initialized with device_setup=0x01 (possibly with 0x11 also, have not tested that). This is at least the case with my rather new box (bought a couple of weeks ago). Patch for usb/usbaudio.c (alsa-driver-1.0.14): static int is_big_endian_format(struct snd_usb_audio *chip, struct audioformat *fp) { switch (chip->usb_id) { case USB_ID(0x0763, 0x2001): /* M-Audio Quattro: captured data only */ if (fp->endpoint & USB_DIR_IN) return 1; break; case USB_ID(0x0763, 0x2003): /* M-Audio Audiophile USB */ + // Is small endian if setup with 0x01 ! + if (device_setup[chip->index] != 0x01) return 1; break; } return 0; } /Håkan

3 2

[alsa-devel] [PATCH] ALSA AICA sound on SEGA Dreamcast - fix behaviour in poor resource conditions
by Adrian McMenamin 09 Jul '07

09 Jul '07

Patch against code in alsa-kernel - tested with patched up 2.6.21-rc7 and working well Submitted by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff -ruN aicaold/aica.c aicanew/aica.c --- aicaold/aica.c 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.c 2007-07-05 23:02:02.000000000 +0100 @@ -65,10 +65,6 @@ /* Use workqueue */ -static struct spu_work_holder { - struct work_struct spu_dma_work; - void *sspointer; -} spu_working; static struct workqueue_struct *aica_queue; @@ -252,18 +248,15 @@ static void run_spu_dma(struct work_struct *work) { int buffer_size; - struct snd_pcm_substream *substream; struct snd_pcm_runtime *runtime; struct snd_card_aica *dreamcastcard; - struct spu_work_holder *holder = container_of(work, struct spu_work_holder, spu_dma_work); - substream = holder-> sspointer; - dreamcastcard = substream->pcm->private_data; - runtime = substream->runtime; + dreamcastcard = container_of(work, struct snd_card_aica, spu_dma_work); + runtime = dreamcastcard->substream->runtime; if (unlikely(dreamcastcard->dma_check == 0)) { buffer_size = frames_to_bytes(runtime, runtime->buffer_size); if (runtime->channels > 1) dreamcastcard->channel->flags |= 0x01; - aica_dma_transfer(runtime->channels, buffer_size, substream); + aica_dma_transfer(runtime->channels, buffer_size, dreamcastcard->substream); startup_aica(dreamcastcard); dreamcastcard->clicks = buffer_size / (AICA_PERIOD_SIZE * runtime->channels); @@ -271,7 +264,7 @@ } else { aica_dma_transfer(runtime->channels, AICA_PERIOD_SIZE * runtime->channels, - substream); + dreamcastcard->substream); snd_pcm_period_elapsed(dreamcastcard->substream); dreamcastcard->clicks++; if (unlikely(dreamcastcard->clicks >= AICA_PERIOD_NUMBER)) @@ -307,7 +300,8 @@ dreamcastcard->current_period = play_period; if (unlikely(dreamcastcard->dma_check == 0)) dreamcastcard->dma_check = 1; - queue_work(aica_queue, &(spu_working.spu_dma_work)); + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); + } static void spu_begin_dma(struct snd_pcm_substream *substream) @@ -317,10 +311,8 @@ struct snd_pcm_runtime *runtime; runtime = substream->runtime; dreamcastcard = substream->pcm->private_data; - /* Use queue to do the heavy lifting */ - spu_working.sspointer = substream; - INIT_WORK(&(spu_working.spu_dma_work), run_spu_dma); - queue_work(aica_queue, &(spu_working.spu_dma_work)); + //get the queue to do the work + queue_work(aica_queue, &(dreamcastcard->spu_dma_work)); /* Timer may already be running */ if (unlikely(dreamcastcard->timer.data)) { mod_timer(&dreamcastcard->timer, jiffies + 4); @@ -366,7 +358,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,17 +396,10 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; - case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); - break; default: return -EINVAL; } @@ -610,6 +597,8 @@ strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); strcpy(dreamcastcard->card->longname, "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + //start the worker thread + INIT_WORK(&(dreamcastcard->spu_dma_work), run_spu_dma); /* Load the PCM 'chip' */ err = snd_aicapcmchip(dreamcastcard, 0); if (unlikely(err < 0)) diff -ruN aicaold/aica.h aicanew/aica.h --- aicaold/aica.h 2007-07-05 23:01:39.000000000 +0100 +++ aicanew/aica.h 2007-07-05 23:02:02.000000000 +0100 @@ -69,6 +69,7 @@ }; struct snd_card_aica { + struct work_struct spu_dma_work; struct snd_card *card; struct aica_channel *channel; struct snd_pcm_substream *substream;

2 7

[alsa-devel] [HDA Intel/STAC92xx] mic not working (retasking?)...
by Daniel J Blueman 09 Jul '07

09 Jul '07

With both 2.6.20 and 2.6.22 kernels on Ubuntu 7.04 on my Sony Vaio SZ240, I'm unable to get my mic connector working at any cost. Well, almost...reading from /dev/dsp, I do see a minute change from 127 for 126 or 128 if I tap my external mic. This would imply the jack-retasking feature of this darn STAC chipset, right? How can I change the jack retasking with the Intel HDA drivers or get further insight into this problem? Thanks in advance, Daniel --- [1] $ lspci -s0:1b.0 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 $ amixer info Card default 'Intel'/'HDA Intel at 0xde400000 irq 22' Mixer name : 'SigmaTel CXD9872RD/K' Components : 'HDA:83847661 HDA:14f12bfa' Controls : 6 Simple ctrls : 5 $ amixer controls numid=2,iface=MIXER,name='Master Playback Switch' numid=1,iface=MIXER,name='Master Playback Volume' numid=6,iface=MIXER,name='PCM Playback Volume' numid=5,iface=MIXER,name='Capture Source' numid=4,iface=MIXER,name='Capture Switch' numid=3,iface=MIXER,name='Capture Volume' $ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'PCM',0 Simple mixer control 'Line',0 Simple mixer control 'Mic',0 Simple mixer control 'Capture',0 $ amixer contents numid=2,iface=MIXER,name='Master Playback Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=1,iface=MIXER,name='Master Playback Volume' ; type=INTEGER,access=rw------,values=2,min=0,max=127,step=0 : values=66,66 numid=6,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---RW-,values=2,min=0,max=255,step=0 : values=230,230 | dBscale-min=-51.00dB,step=0.20dB,mute=0 numid=5,iface=MIXER,name='Capture Source' ; type=ENUMERATED,access=rw------,values=1,items=2 ; Item #0 'Line' ; Item #1 'Mic' : values=1 numid=4,iface=MIXER,name='Capture Switch' ; type=BOOLEAN,access=rw------,values=2 : values=on,on numid=3,iface=MIXER,name='Capture Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=15,step=0 : values=15,15 | dBscale-min=0.00dB,step=1.50dB,mute=0 -- Daniel J Blueman

3 14

[alsa-devel] Remove dynamic configuration from alsa-lib
by Fausto Carvalho Marques SIlva 08 Jul '07

08 Jul '07

I have a hard job...to remove dynamic configuration from alsa-lib (conf.c,confmisc.c). I found it difficult because snd_conf_* functions are spread all over *.c files. Is there am easy way of doing it ? Does anyone have some "guidelines" of doing it? Any help is wellcome... Fausto Carvalho Marques Silva

5 5

[alsa-devel] hda-intel on macbook: unknown model for ACL882
by Steffen Schulz 07 Jul '07

07 Jul '07

Hi all, Since I didn't get much feedback on alsa-users, I figured I'd place a pointer here, too. The revision 3 macbook pro(santa-rosa) ships with a ALC882. Loading hda_intel with auto-detection results in very VERY quiet sound out of the headphones if 'front' is set to 100%, mic input seems to work too, but also very quiet. Providing an option 'model=w2jc' to modprobe however results in working(loud) front and mic. But this model does not seem to provide headphones output, so they don't work. This sounds to me like it may be easy to fix, so I hope you find this information useful. Would be great to have the sound working. Regards, Steffen (subscribed for the moment..) > The snd-hda-module loads with the following message: > | kernel: hda_codec: Unknown model for ALC882, trying auto-probe from BIOS... > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) > | kernel: ALSA /usr/src/alsa/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x18, fmic=0x0, line=0x1a, fline=0x0, cd=0x0, aux=0x0 > > aplay -l says: > > | **** List of PLAYBACK Hardware Devices **** > | card 0: Intel [HDA Intel], device 0: ALC882 Analog [ALC882 Analog] > | Subdevices: 1/1 > | Subdevice #0: subdevice #0 > | card 0: Intel [HDA Intel], device 1: ALC882 Digital [ALC882 Digital] > | Subdevices: 1/1 > | Subdevice #0: subdevice #0 > > lspci -vv -nn says: > > | 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) > | Subsystem: Apple Computer Inc. Unknown device [106b:00a0] > | 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: 256 bytes > | Interrupt: pin A routed to IRQ 20 > | Region 0: Memory at 9b500000 (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)

2 3

[alsa-devel] [HDA Intel/AD1984] Alsa 1.0.14 not working on ThinkPad T61
by joe+alsa＠nahmias.net 06 Jul '07

06 Jul '07

Hello, I'm having trouble getting sound on my ThinkPad T61 working. According to the changelog for alsa-driver 1.0.14, support was added in that version; however, no sound is coming out. I'm running Debian unstable and I'd be happy to test any patches, provide more info, etc... Any help would be appreciated. Thanks, --Joe PS - Here's some debugging info that may be of use: $ uname -srvmo Linux 2.6.21-2-686 #1 SMP Mon Jun 25 20:53:36 UTC 2007 i686 GNU/Linux $ dpkg -l alsa* | grep ^i ii alsa-base 1.0.14-1 ALSA driver configuration files ii alsa-modules-2.6.21-2-686 1.0.14-1+2.6.21-5 ALSA modules for kernel 2.6.21-2-686 ii alsa-source 1.0.14-1 ALSA driver sources ii alsa-utils 1.0.14-1 ALSA utilities $ aumix -q pcm 90, 90 mic 0, 0 pcm2 100, 100 igain 56, 56 $ lsmod | grep snd snd_hda_intel 244952 0 snd_pcm_oss 39680 0 snd_mixer_oss 15840 1 snd_pcm_oss snd_pcm 72292 2 snd_hda_intel,snd_pcm_oss snd_timer 20964 1 snd_pcm snd 49316 5 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer snd_page_alloc 9544 2 snd_hda_intel,snd_pcm soundcore 7424 1 snd $ sudo lspci -vv -s 00:1b.0 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Lenovo Unknown device 20ac 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 B routed to IRQ 21 Region 0: Memory at fe020000 (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) $ cat /proc/asound/card0/codec#0 Codec: Analog Devices AD1984 Address: 0 Vendor Id: 0x11d41984 Subsystem Id: 0x17aa20d7 Revision Id: 0x100400 Default PCM: rates [0x7ff]: 8000 11025 16000 22050 32000 44100 48000 88200 96000 176400 192000 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=0x27, nsteps=0x27, stepsize=0x05, mute=0 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 3 0x01* 0x08 0x09 Node 0x03 [Audio Output] wcaps 0x405: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0x80 0x80] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x405: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=0 Amp-Out vals: [0xa3 0xa3] Power: 0x0 Node 0x05 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Power: 0x0 Connection: 1 0x17 Node 0x06 [Audio Input] wcaps 0x10050b: Stereo Amp-In Amp-In caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-In vals: [0xa7 0xa7] Power: 0x0 Connection: 1 0x18 Node 0x07 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x22 0x21 Node 0x08 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x0c Node 0x09 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x0d Node 0x0a [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x04 0x21 Node 0x0b [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x0f 0x21 Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0x1e 0x1e] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0d [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x36, stepsize=0x05, mute=1 Amp-Out vals: [0xa7 0xa7] Connection: 5 0x14* 0x15 0x16 0x20 0x25 Node 0x0e [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03 0x04* Node 0x0f [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x00] Node 0x11 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] 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 0x07 Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 1 0x0a Node 0x13 [Pin Complex] wcaps 0x40050c: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x00] Pincap 0x0810010: OUT EAPD Pin Default 0x511301f0: [N/A] Speaker at Int Rear Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 1 0x1f Node 0x14 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x083727: IN Detect Pin Default 0x02a15021: [Jack] Mic at Ext Front Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: Pincap 0x083727: IN Detect Pin Default 0x90a7012e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x16 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0837: IN OUT Detect Pin Default 0x593301f0: [N/A] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Connection: 1 0x0b Node 0x17 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown Pin-ctls: 0x24: IN Node 0x18 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x55a601f0: [N/A] Mic at Int Top Conn = Digital, Color = Unknown Pin-ctls: 0x00: Node 0x19 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 2 0x20* 0x21 Node 0x1a [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x91f311f0: [Fixed] Other at Int Rear Conn = ATAPI, Color = Black Pin-ctls: 0x20: IN Node 0x1b [Pin Complex] wcaps 0x40030d: Stereo Digital Amp-Out Amp-Out caps: ofs=0x27, nsteps=0x27, stepsize=0x05, mute=1 Amp-Out vals: [0x27 0x27] Pincap 0x0810: OUT Pin Default 0x214411a0: [Jack] SPDIF Out at Sep Rear Conn = RCA, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x1c [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083737: IN OUT Detect Pin Default 0x21a15020: [Jack] Mic at Sep Rear Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Connection: 1 0x24 Node 0x1d [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 25 0x07* 0x19 0x0a 0x0b 0x0c 0x0d 0x0e 0x0f 0x1a 0x1c 0x11 0x12 0x13 0x14 0x15 0x16 0x1e 0x1f 0x20 0x21 0x22 0x23 0x24 0x25 0x26 Node 0x1e [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Connection: 2 0x0e 0x21 Node 0x1f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x1e Node 0x20 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x9c 0x9c] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 5 0x14 0x15 0x16 0x1a 0x25 Node 0x21 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1f 0x1f] Connection: 1 0x20 Node 0x22 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03 0x04* Node 0x23 [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x03* 0x04 Node 0x24 [Audio Mixer] wcaps 0x200103: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Connection: 2 0x23 0x21 Node 0x25 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 1 0x1c Node 0x26 [Vendor Defined Widget] wcaps 0xf00100: Mono Connection: 3 0x14* 0x15 0x1c

2 2

Re: [alsa-devel] [Fixed-Mostly] ALC268
by Steven Zakulec 06 Jul '07

06 Jul '07

I've found another issue: I've lost some mixer options now. I've lost 1 capture source at least. I've attached the result of alsactl -f somefile store from before sound worked and after. --------------------------------------- Original E-mail From: Steven Zakulec [spzakulec(a)mail.widener.edu] Date: 07/06/2007 10:04 AM To: alsa-devel(a)alsa-project.org Subject: [alsa-devel] [Fixed-Mostly] ALC268 I was finally able to get sound working, thanks to the bugreport here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 What I did was get the patch_realtek.c from the bugreport, put it in alsa-kernel/pci/hda, and then re-build the driver (plus I re-built the lib, but probably not necessary). I wasn't able to rmmod snd-hda-intel after the build finished, so I restarted the computer, and then it just worked. Unfortunately, the headphone jack is not working. The sound is only slightly, if at all, muted from the speakers when the headphones are plugged in. _______________________________________________ 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] [Fixed-Mostly] ALC268
by Steven Zakulec 06 Jul '07

06 Jul '07

I was finally able to get sound working, thanks to the bugreport here: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3104 What I did was get the patch_realtek.c from the bugreport, put it in alsa-kernel/pci/hda, and then re-build the driver (plus I re-built the lib, but probably not necessary). I wasn't able to rmmod snd-hda-intel after the build finished, so I restarted the computer, and then it just worked. Unfortunately, the headphone jack is not working. The sound is only slightly, if at all, muted from the speakers when the headphones are plugged in.

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[alsa-devel] PS3 - Surround sound on SPDIF set-up
by gt bradley 06 Jul '07

06 Jul '07

Crud :( I had asked them and they said that the hypervisor should allow for AC3 pass-through. In my inital review of the snd_ps3.c code, it looked like it was basically a data-pump, handling the DMA & interupts to move data from the passed in stream to the h/w (via the hypervisor). I also assumed that the iec bits (in headers in the SPDIF frames) were being set by higher levels in the software stack. If the bits are being set by the hardware and/or hypervisor then I'll have to fall back to plan "B". I've been programming for 20+ years, and have put kernel mods into production, but the last kernel hacking/device driver I did was in SYSV. I can forward changes/fixes to Geoff, for inclusion rolling things "upstream". I don't have direct access to inside information other than the source-code, but can forward the requests for more detailed info. let me know if I can help GT p.s. thanks for your time and patience. On 7/5/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Wed, 04 Jul 2007 18:23:09 -0700, > GT Bradley wrote: > > > > right, > > For my Fedora install, these files are at /etc/alsa/cards, (although > > the iec958.conf) is in /etc/alsa/pcm). > > > > The iec958.conf seems resonable, although I don't fully understand the > > syntax of the file, nor understand how the asoundrc file wires to > > /etc/alsa/... to the actual linux driver. It seems to pass through to > > > the default device (which is what I want) > > > > using the system-config-soundcard utility, I do get surround sound, and > > is displays the card as PS3 whcih matches the output of aplay -l I've > > > tryed createing a /etc/alsa/cards/PS3.conf file as follows: > > It seems that the current PS3 driver doesn't support the passthru > properly yet. For AC3, you'd have to set up the IEC958 status bits, > but the driver doesn't support it. Some receivers are robust that > work even without non-audio bit, but many receivers require the proper > status bits. I'll contact with SCE guys for fixing this issue. > > > Takashi >

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[alsa-devel] linked streams question
by Eliot Blennerhassett 06 Jul '07

06 Jul '07

Greetings, I'm adding linked stream support to my driver (asihpi), and having some trouble with the end of playback. What happens is that the slave streams never transition to SETUP state at the end of playback. (Eg playing 4 channel wav file to 2 linked stereo streams with aplay). The master stream reaches snd_pcm_drain_done() which then causes trigger(stop), resulting in all linked streams being stopped. But only the master state is set to SETUP However, now the slave streams are stopped, period_elapsed() is not called for a final time to (I assme) cause them to transition to the SETUP state. Eventually I get "playback drain error (DMA or IRQ trouble?)\n" My workaround is to set the state of each substream to SETUP in the trigger-stop operator, but no other driver needs to do this so I think it is probably the wrong solution - maybe misses some other cleanup code? Any suggestions on the proper way to handle this? regards Eliot

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[alsa-devel] [Fwd: Re: default setting not fully working with new ALSA device]
by Tobin Davis 05 Jul '07

05 Jul '07

-------- Forwarded Message -------- From: Peter Matthias <PeterMatthias(a)web.de> To: Tobin Davis <tdavis(a)dsl-only.net> Subject: Re: [alsa-devel] default setting not fully working with new ALSA device Date: Thu, 5 Jul 2007 18:34:08 +0200 The Board is a MSI K9AGM2-FIH, Sockel AM2, ATI RS690G, mATX, PCIe The script gave following output: "Your ALSA information is located at http://pastebin.ca/604529" Other drivers are statically linked in my kernel. Thank you for your work! In case of any questions, contact me. Peter Am Mittwoch, 4. Juli 2007 schrieben Sie: > Could you post the output from > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh ? This > script will give us more detailed information on what sound chip it is. > We will also need to know the brand of board, for the driver file. > > Thanks, > > Tobin > > On Wed, 2007-07-04 at 16:48 +0200, Peter Matthias wrote: > > Hello, > > > > I have got a 690G board with following lspic -nv output: > > > > 01:05.2 0403: 1002:7919 > > Subsystem: 1002:7919 > > Flags: bus master, fast devsel, latency 64, IRQ 19 > > Memory at fe9e8000 (64-bit, non-prefetchable) [size=16K] > > Capabilities: [50] Power Management version 2 > > Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 > > Enable- > > > > With the default setting, microphone does not work. With module > > parameter "modprobe -i snd-hda-intel model=6stack-dig" microphone works > > well. Please add to default setting. It was written in Linux Kernel > > documentation to report this issue here. > > > > Thank you, > > Peter > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel -- Tobin Davis <tdavis(a)dsl-only.net>

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[PATCH 1/1] ALSA sound driver for the AT73C213 DAC using Atmel SSC driver
by Hans-Christian Egtvedt 05 Jul '07

05 Jul '07

This patch adds support for the AT73C213 DAC using the misc Atmel SSC driver in I2S mode. The driver also requires a SPI to setup the registers and control volume. It has been tested with an AT32AP7000 on the ATSTK1000 development board. The driver should also work with any Atmel device with an SSC module supported by the Atmel SSC driver (atmel-ssc). The atmel-ssc driver is just submitted to the Linux kernel. Please see mail thread http://lkml.org/lkml/2007/7/16/32 Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- include/linux/spi/at73c213.h | 25 + sound/spi/Kconfig | 25 + sound/spi/Makefile | 3 + sound/spi/at73c213.c | 1121 ++++++++++++++++++++++++++++++++++++++++++ sound/spi/at73c213.h | 119 +++++ 5 files changed, 1293 insertions(+), 0 deletions(-) create mode 100644 include/linux/spi/at73c213.h create mode 100644 sound/spi/at73c213.c create mode 100644 sound/spi/at73c213.h diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h new file mode 100644 index 0000000..0f20a70 --- /dev/null +++ b/include/linux/spi/at73c213.h @@ -0,0 +1,25 @@ +/* + * Board-specific data used to set up AT73c213 audio DAC driver. + */ + +#ifndef __LINUX_SPI_AT73C213_H +#define __LINUX_SPI_AT73C213_H + +/** + * at73c213_board_info - how the external DAC is wired to the device. + * + * @ssc_id: SSC platform_driver id the DAC shall use to stream the audio. + * @dac_clk: the external clock used to provide master clock to the DAC. + * @shortname: a short discription for the DAC, seen by userspace tools. + * + * This struct contains the configuration of the hardware connection to the + * external DAC. The DAC needs a master clock and a I2S audio stream. It also + * provides a name which is used to identify it in userspace tools. + */ +struct at73c213_board_info { + int ssc_id; + struct clk *dac_clk; + char shortname[32]; +}; + +#endif /* __LINUX_SPI_AT73C213_H */ diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig index 60fbb2a..0d08c29 100644 --- a/sound/spi/Kconfig +++ b/sound/spi/Kconfig @@ -3,4 +3,29 @@ menu "SPI devices" depends on SND != n +config SND_AT73C213 + tristate "Atmel AT73C213 DAC driver" + depends on ATMEL_SSC + select SND_PCM + help + Say Y here if you want to use the Atmel AT73C213 external DAC. This + DAC can be found on Atmel development boards. + + This driver requires the Atmel SSC driver for sound sink, a + peripheral found on most AT91 and AVR32 microprocessors. + + To compile this driver as a module, choose M here: the module will be + called snd-at73c213. + +config SND_AT73C213_TARGET_BITRATE + int "Target bitrate for AT73C213" + depends on SND_AT73C213 + default "48000" + range 8000 50000 + help + Sets the target bitrate for the bitrate calculator in the driver. + Limited by hardware to be between 8000 Hz and 50000 Hz. + + Set to 48000 Hz by default. + endmenu diff --git a/sound/spi/Makefile b/sound/spi/Makefile index 30d8ec9..026fb73 100644 --- a/sound/spi/Makefile +++ b/sound/spi/Makefile @@ -1,2 +1,5 @@ # Makefile for SPI drivers +snd-at73c213-objs := at73c213.o + +obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c new file mode 100644 index 0000000..a466929 --- /dev/null +++ b/sound/spi/at73c213.c @@ -0,0 +1,1121 @@ +/* + * Driver for AT73C213 16-bit stereo DAC connected to Atmel SSC + * + * Copyright (C) 2006-2007 Atmel Norway + * + * 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. + */ + +/*#define DEBUG*/ + +#include <linux/clk.h> +#include <linux/err.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <sound/driver.h> +#include <sound/initval.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> + +#include <linux/atmel-ssc.h> + +#include <linux/spi/spi.h> +#include <linux/spi/at73c213.h> + +#include "at73c213.h" + +#define BITRATE_MIN 8000 /* Hardware limit? */ +#define BITRATE_TARGET CONFIG_SND_AT73C213_TARGET_BITRATE +#define BITRATE_MAX 50000 /* Hardware limit. */ + +/* Initial (hardware reset) AT73C213 register values. */ +static u8 snd_at73c213_original_image[18] = +{ + 0x00, /* 00 - CTRL */ + 0x05, /* 01 - LLIG */ + 0x05, /* 02 - RLIG */ + 0x08, /* 03 - LPMG */ + 0x08, /* 04 - RPMG */ + 0x00, /* 05 - LLOG */ + 0x00, /* 06 - RLOG */ + 0x22, /* 07 - OLC */ + 0x09, /* 08 - MC */ + 0x00, /* 09 - CSFC */ + 0x00, /* 0A - MISC */ + 0x00, /* 0B - */ + 0x00, /* 0C - PRECH */ + 0x05, /* 0D - AUXG */ + 0x00, /* 0E - */ + 0x00, /* 0F - */ + 0x00, /* 10 - RST */ + 0x00, /* 11 - PA_CTRL */ +}; + +struct snd_at73c213 { + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + struct at73c213_board_info *board; + int irq; + int period; + unsigned long bitrate; + struct clk *bitclk; + struct ssc_device *ssc; + struct spi_device *spi; + u8 spi_wbuffer[2]; + u8 spi_rbuffer[2]; + /* Image of the SPI registers in AT73C213. */ + u8 reg_image[18]; + /* Protect registers against concurrent access. */ + spinlock_t lock; +}; + +#define get_chip(card) ((struct snd_at73c213 *)card->private_data) + +static int +snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) +{ + struct spi_message msg; + struct spi_transfer msg_xfer = { + .len = 2, + .cs_change = 0, + }; + int retval; + + spi_message_init(&msg); + + chip->spi_wbuffer[0] = reg; + chip->spi_wbuffer[1] = val; + + msg_xfer.tx_buf = chip->spi_wbuffer; + msg_xfer.rx_buf = chip->spi_rbuffer; + spi_message_add_tail(&msg_xfer, &msg); + + retval = spi_sync(chip->spi, &msg); + + if (!retval) + chip->reg_image[reg] = val; + + return retval; +} + +static struct snd_pcm_hardware snd_at73c213_playback_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, /* Replaced by chip->bitrate later. */ + .rate_max = 50000, /* Replaced by chip->bitrate later. */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024 - 1, + .period_bytes_min = 512, + .period_bytes_max = 64 * 1024 - 1, + .periods_min = 4, + .periods_max = 1024, +}; + +/* + * Calculate and set bitrate and divisions. + */ +static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) +{ + unsigned long ssc_rate = clk_get_rate(chip->ssc->clk); + unsigned long dac_rate_new, ssc_div, status; + unsigned long ssc_div_max, ssc_div_min; + int max_tries; + + /* + * We connect two clocks here, picking divisors so the I2S clocks + * out data at the same rate the DAC clocks it in ... and as close + * as practical to the desired target rate. + * + * The DAC master clock (MCLK) is programmable, and is either 256 + * or (not here) 384 times the I2S output clock (BCLK). + */ + + /* SSC clock / (bitrate * stereo * 16-bit). */ + ssc_div = ssc_rate / (BITRATE_TARGET * 2 * 16); + ssc_div_min = ssc_rate / (BITRATE_MAX * 2 * 16); + ssc_div_max = ssc_rate / (BITRATE_MIN * 2 * 16); + max_tries = (ssc_div_max - ssc_div_min) / 2; + + if (max_tries < 1) + max_tries = 1; + + /* ssc_div must be a power of 2. */ + ssc_div = (ssc_div + 1) & ~1UL; + + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) { + ssc_div -= 2; + if ((ssc_rate / (ssc_div * 2 * 16)) > BITRATE_MAX) + return -ENXIO; + } + + /* Search for a possible bitrate. */ + do { + /* SSC clock / (ssc divider * 16-bit * stereo). */ + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) + return -ENXIO; + + /* 256 / (2 * 16) = 8 */ + dac_rate_new = 8 * (ssc_rate / ssc_div); + + status = clk_round_rate(chip->board->dac_clk, dac_rate_new); + if (status < 0) + return status; + + /* Ignore difference smaller than 256 Hz. */ + if ((status/256) == (dac_rate_new/256)) + goto set_rate; + + ssc_div += 2; + } while (--max_tries); + + /* Not able to find a valid bitrate. */ + return -ENXIO; + +set_rate: + status = clk_set_rate(chip->board->dac_clk, status); + if (status < 0) + return status; + + /* Set divider in SSC device. */ + ssc_writel(chip->ssc->regs, CMR, ssc_div/2); + + /* SSC clock / (ssc divider * 16-bit * stereo). */ + chip->bitrate = ssc_rate / (ssc_div * 16 * 2); + + dev_info(&chip->spi->dev, + "at73c213: supported bitrate is %lu (%lu divider)\n", + chip->bitrate, ssc_div); + + return 0; +} + +static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_at73c213_playback_hw.rate_min = chip->bitrate; + snd_at73c213_playback_hw.rate_max = chip->bitrate; + runtime->hw = snd_at73c213_playback_hw; + chip->substream = substream; + + return 0; +} + +static int snd_at73c213_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + chip->substream = NULL; + return 0; +} + +static int snd_at73c213_pcm_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_at73c213_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_at73c213_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int block_size; + + block_size = frames_to_bytes(runtime, runtime->period_size); + + chip->period = 0; + + ssc_writel(chip->ssc->regs, PDC_TPR, + (long)runtime->dma_addr); + ssc_writel(chip->ssc->regs, PDC_TCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + block_size); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + + return 0; +} + +static int snd_at73c213_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + int retval = 0; + + spin_lock(&chip->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ssc_writel(chip->ssc->regs, IER, SSC_BIT(IER_ENDTX)); + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTEN)); + break; + case SNDRV_PCM_TRIGGER_STOP: + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTDIS)); + ssc_writel(chip->ssc->regs, IDR, SSC_BIT(IDR_ENDTX)); + break; + default: + dev_dbg(&chip->spi->dev, "spurious command %x\n", cmd); + retval = -EINVAL; + break; + } + + spin_unlock(&chip->lock); + + return retval; +} + +static snd_pcm_uframes_t +snd_at73c213_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; + unsigned long bytes; + + bytes = ssc_readl(chip->ssc->regs, PDC_TPR) + - (unsigned long)runtime->dma_addr; + + pos = bytes_to_frames(runtime, bytes); + if (pos >= runtime->buffer_size) + pos -= runtime->buffer_size; + + return pos; +} + +static struct snd_pcm_ops at73c213_playback_ops = { + .open = snd_at73c213_pcm_open, + .close = snd_at73c213_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_at73c213_pcm_hw_params, + .hw_free = snd_at73c213_pcm_hw_free, + .prepare = snd_at73c213_pcm_prepare, + .trigger = snd_at73c213_pcm_trigger, + .pointer = snd_at73c213_pcm_pointer, +}; + +static void snd_at73c213_pcm_free(struct snd_pcm *pcm) +{ + struct snd_at73c213 *chip = snd_pcm_chip(pcm); + if (chip->pcm) { + snd_pcm_lib_preallocate_free_for_all(chip->pcm); + chip->pcm = NULL; + } +} + +static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device) +{ + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(chip->card, chip->card->shortname, + device, 1, 0, &pcm); + if (retval < 0) + goto out; + + pcm->private_data = chip; + pcm->private_free = snd_at73c213_pcm_free; + pcm->info_flags = SNDRV_PCM_INFO_BLOCK_TRANSFER; + strcpy(pcm->name, "at73c213"); + chip->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &at73c213_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(chip->pcm, + SNDRV_DMA_TYPE_DEV, &chip->spi->dev, + 64 * 1024, 64 * 1024); +out: + return retval; +} + +static irqreturn_t snd_at73c213_interrupt(int irq, void *dev_id) +{ + struct snd_at73c213 *chip = dev_id; + struct snd_pcm_runtime *runtime = chip->substream->runtime; + u32 status; + int offset; + int block_size; + int next_period; + int retval = IRQ_NONE; + + spin_lock(&chip->lock); + + block_size = frames_to_bytes(runtime, runtime->period_size); + status = ssc_readl(chip->ssc->regs, IMR); + + if (status & SSC_BIT(IMR_ENDTX)) { + chip->period++; + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + offset); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + retval = IRQ_HANDLED; + } + + ssc_readl(chip->ssc->regs, IMR); + spin_unlock(&chip->lock); + + if (status & SSC_BIT(IMR_ENDTX)) + snd_pcm_period_elapsed(chip->substream); + + return retval; +} + +/* + * Mixer functions. + */ +static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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, retval; + unsigned short val; + + val = (ucontrol->value.integer.value[0] & mask); + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val = (chip->reg_image[reg] & ~(mask << shift)) | val; + change = val != chip->reg_image[reg]; + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + if (mask == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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_irq(&chip->lock); + + ucontrol->value.integer.value[0] = + (chip->reg_image[left_reg] >> shift_left) & mask; + ucontrol->value.integer.value[1] = + (chip->reg_image[right_reg] >> shift_right) & mask; + + if (invert) { + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + ucontrol->value.integer.value[1] = + (mask - ucontrol->value.integer.value[1]); + } + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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, retval; + 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_irq(&chip->lock); + + val1 = (chip->reg_image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->reg_image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->reg_image[left_reg] + || val2 != chip->reg_image[right_reg]; + retval = snd_at73c213_write_reg(chip, left_reg, val1); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + retval = snd_at73c213_write_reg(chip, right_reg, val2); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + + spin_unlock_irq(&chip->lock); + + return change; + +out: + return retval; +} + +static int snd_at73c213_mono_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & 0x01; + + if (invert) + ucontrol->value.integer.value[0] = + (0x01 - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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, retval; + unsigned short val; + + if (ucontrol->value.integer.value[0]) + val = mask; + else + val = 0; + + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val |= (chip->reg_image[reg] & ~(mask << shift)); + change = val != chip->reg_image[reg]; + + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_pa_volume_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 = ((kcontrol->private_value >> 16) & 0xff) - 1; + + return 0; +} + +static int snd_at73c213_line_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +static int snd_at73c213_aux_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +#define AT73C213_MONO_SWITCH(xname, xindex, reg, shift, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_mono_switch_info, \ + .get = snd_at73c213_mono_switch_get, \ + .put = snd_at73c213_mono_switch_put, \ + .private_value = (reg | (shift << 8) | (mask << 16) | (invert << 24)) \ +} + +#define AT73C213_STEREO(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_stereo_info, \ + .get = snd_at73c213_stereo_get, \ + .put = snd_at73c213_stereo_put, \ + .private_value = (left_reg | (right_reg << 8) \ + | (shift_left << 16) | (shift_right << 19) \ + | (mask << 24) | (invert << 22)) \ +} + +static struct snd_kcontrol_new snd_at73c213_controls[] __devinitdata = { +AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1), +AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1), +AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1), +AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1), +AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, DAC_CTRL_ONPADRV, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PA Playback Volume", + .index = 0, + .info = snd_at73c213_pa_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | (0x0f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("PA High Gain Playback Switch", 0, PA_CTRL, PA_CTRL_APALP, 0x01, 1), +AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, PA_CTRL_APAON, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Capture Volume", + .index = 0, + .info = snd_at73c213_aux_capture_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = DAC_AUXG | (0 << 8) | (0x1f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Volume", + .index = 0, + .info = snd_at73c213_line_capture_volume_info, + .get = snd_at73c213_stereo_get, + .put = snd_at73c213_stereo_put, + .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19) + | (0x1f << 24) | (1 << 22), +}, +AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), +}; + +static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) +{ + struct snd_card *card; + int errval, idx; + + if (chip == NULL || chip->pcm == NULL) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) { + errval = snd_ctl_add(card, + snd_ctl_new1(&snd_at73c213_controls[idx], + chip)); + if (errval < 0) + goto cleanup; + } + + return 0; + +cleanup: + for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_find_numid(card, idx); + if (kctl) + snd_ctl_remove(card, kctl); + } + return errval; +} + +/* + * Device functions + */ +static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +{ + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Periode is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + /* + * Data length is 16 bit (16 - 1). + * Transmit MSB first. + * Transmit 2 words each transfer. + * Frame sync length is 16 bit (16 - 1). + * Frame starts on negative pulse. + */ + ssc_writel(chip->ssc->regs, TFMR, + SSC_BF(TFMR_DATLEN, 16 - 1) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATNB, 1) + | SSC_BF(TFMR_FSLEN, 16 - 1) + | SSC_BF(TFMR_FSOS, 1)); + + return 0; +} + +static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +{ + int retval; + unsigned char dac_ctrl = 0; + + retval = snd_at73c213_set_bitrate(chip); + if (retval) + goto out; + + /* Enable DAC master clock. */ + clk_enable(chip->board->dac_clk); + + /* Initialize at73c213 on SPI bus. */ + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); + if (retval) + goto out_clk; + msleep(1); + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x03); + if (retval) + goto out_clk; + + /* Precharge everything. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH)); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_CTRL, + (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR)); + if (retval) + goto out_clk; + + msleep(50); + + /* Stop precharging PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, + (1<<PA_CTRL_APALP) | 0x0f); + if (retval) + goto out_clk; + + msleep(450); + + /* Stop precharging DAC, turn on master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, (1<<DAC_PRECH_ONMSTR)); + if (retval) + goto out_clk; + + msleep(1); + + /* Turn on DAC. */ + dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR); + + retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl); + if (retval) + goto out_clk; + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out_clk; + + /* Enable I2S device, i.e. clock output. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + goto out; + +out_clk: + clk_disable(chip->board->dac_clk); +out: + return retval; +} + +static int snd_at73c213_dev_free(struct snd_device *device) +{ + struct snd_at73c213 *chip = device->device_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + chip->irq = -1; + } + + return 0; +} + +static int __devinit snd_at73c213_dev_init(struct snd_card *card, + struct spi_device *spi) +{ + static struct snd_device_ops ops = { + .dev_free = snd_at73c213_dev_free, + }; + struct snd_at73c213 *chip = get_chip(card); + int irq, retval; + + irq = chip->ssc->irq; + if (irq < 0) + return irq; + + spin_lock_init(&chip->lock); + chip->card = card; + chip->irq = -1; + + retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); + if (retval) { + dev_dbg(&chip->spi->dev, "unable to request irq %d\n", irq); + goto out; + } + chip->irq = irq; + + memcpy(&chip->reg_image, &snd_at73c213_original_image, + sizeof(snd_at73c213_original_image)); + + retval = snd_at73c213_ssc_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_chip_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_pcm_new(chip, 0); + if (retval) + goto out_irq; + + retval = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (retval) + goto out_irq; + + retval = snd_at73c213_mixer(chip); + if (retval) + goto out_snd_dev; + + snd_card_set_dev(card, &spi->dev); + + goto out; + +out_snd_dev: + snd_device_free(card, chip); +out_irq: + free_irq(chip->irq, chip); + chip->irq = -1; +out: + return retval; +} + +static int snd_at73c213_probe(struct spi_device *spi) +{ + struct snd_card *card; + struct snd_at73c213 *chip; + struct at73c213_board_info *board; + int retval; + char id[16]; + + board = spi->dev.platform_data; + if (!board) { + dev_dbg(&spi->dev, "no platform_data\n"); + return -ENXIO; + } + + if (!board->dac_clk) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return -ENXIO; + } + + if (IS_ERR(board->dac_clk)) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return PTR_ERR(board->dac_clk); + } + + retval = -ENOMEM; + + /* Allocate "card" using some unused identifiers. */ + snprintf(id, sizeof id, "at73c213_%d", board->ssc_id); + card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213)); + if (!card) + goto out; + + chip = card->private_data; + chip->spi = spi; + chip->board = board; + + chip->ssc = ssc_request(board->ssc_id); + if (IS_ERR(chip->ssc)) { + dev_dbg(&spi->dev, "could not get ssc%d device\n", + board->ssc_id); + retval = PTR_ERR(chip->ssc); + goto out_card; + } + + retval = snd_at73c213_dev_init(card, spi); + if (retval) + goto out_ssc; + + strcpy(card->driver, "at73c213"); + strcpy(card->shortname, board->shortname); + sprintf(card->longname, "%s on irq %d", card->shortname, chip->irq); + + retval = snd_card_register(card); + if (retval) + goto out_ssc; + + dev_set_drvdata(&spi->dev, card); + + goto out; + +out_ssc: + ssc_free(chip->ssc); +out_card: + snd_card_free(card); +out: + return retval; +} + +static int __devexit snd_at73c213_remove(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + int retval; + + /* Stop playback. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out; + + /* Turn off PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, (chip->reg_image[PA_CTRL]|0x0f)); + if (retval) + goto out; + msleep(10); + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APALP)|0x0f); + if (retval) + goto out; + + /* Turn off external DAC. */ + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c); + if (retval) + goto out; + msleep(2); + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x00); + if (retval) + goto out; + + /* Turn off master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0x00); + if (retval) + goto out; + +out: + /* Stop DAC master clock. */ + clk_disable(chip->board->dac_clk); + + ssc_free(chip->ssc); + snd_card_free(card); + dev_set_drvdata(&spi->dev, NULL); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + clk_disable(chip->board->dac_clk); + + return 0; +} + +static int snd_at73c213_resume(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + clk_enable(chip->board->dac_clk); + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + return 0; +} +#else +#define snd_at73c213_suspend NULL +#define snd_at73c213_resume NULL +#endif + +static struct spi_driver at73c213_driver = { + .driver = { + .name = "at73c213", + }, + .probe = snd_at73c213_probe, + .suspend = snd_at73c213_suspend, + .resume = snd_at73c213_resume, + .remove = __devexit_p(snd_at73c213_remove), +}; + +static int __init at73c213_init(void) +{ + return spi_register_driver(&at73c213_driver); +} +module_init(at73c213_init); + +static void __exit at73c213_exit(void) +{ + spi_unregister_driver(&at73c213_driver); +} +module_exit(at73c213_exit); + +MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt(a)atmel.com>"); +MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); +MODULE_LICENSE("GPL"); diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h new file mode 100644 index 0000000..fd8b372 --- /dev/null +++ b/sound/spi/at73c213.h @@ -0,0 +1,119 @@ +/* + * Driver for the AT73C213 16-bit stereo DAC on Atmel ATSTK1000 + * + * Copyright (C) 2006 - 2007 Atmel Corporation + * + * 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, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA + * 02111-1307, USA. + * + * The full GNU General Public License is included in this + * distribution in the file called COPYING. + */ + +#ifndef _SND_AT73C213_H +#define _SND_AT73C213_H + +/* DAC control register */ +#define DAC_CTRL 0x00 +#define DAC_CTRL_ONPADRV 7 +#define DAC_CTRL_ONAUXIN 6 +#define DAC_CTRL_ONDACR 5 +#define DAC_CTRL_ONDACL 4 +#define DAC_CTRL_ONLNOR 3 +#define DAC_CTRL_ONLNOL 2 +#define DAC_CTRL_ONLNIR 1 +#define DAC_CTRL_ONLNIL 0 + +/* DAC left line in gain register */ +#define DAC_LLIG 0x01 +#define DAC_LLIG_LLIG 0 + +/* DAC right line in gain register */ +#define DAC_RLIG 0x02 +#define DAC_RLIG_RLIG 0 + +/* DAC Left Master Playback Gain Register */ +#define DAC_LMPG 0x03 +#define DAC_LMPG_LMPG 0 + +/* DAC Right Master Playback Gain Register */ +#define DAC_RMPG 0x04 +#define DAC_RMPG_RMPG 0 + +/* DAC Left Line Out Gain Register */ +#define DAC_LLOG 0x05 +#define DAC_LLOG_LLOG 0 + +/* DAC Right Line Out Gain Register */ +#define DAC_RLOG 0x06 +#define DAC_RLOG_RLOG 0 + +/* DAC Output Level Control Register */ +#define DAC_OLC 0x07 +#define DAC_OLC_RSHORT 7 +#define DAC_OLC_ROLC 4 +#define DAC_OLC_LSHORT 3 +#define DAC_OLC_LOLC 0 + +/* DAC Mixer Control Register */ +#define DAC_MC 0x08 +#define DAC_MC_INVR 5 +#define DAC_MC_INVL 4 +#define DAC_MC_RMSMIN2 3 +#define DAC_MC_RMSMIN1 2 +#define DAC_MC_LMSMIN2 1 +#define DAC_MC_LMSMIN1 0 + +/* DAC Clock and Sampling Frequency Control Register */ +#define DAC_CSFC 0x09 +#define DAC_CSFC_OVRSEL 4 + +/* DAC Miscellaneous Register */ +#define DAC_MISC 0x0A +#define DAC_MISC_VCMCAPSEL 7 +#define DAC_MISC_DINTSEL 4 +#define DAC_MISC_DITHEN 3 +#define DAC_MISC_DEEMPEN 2 +#define DAC_MISC_NBITS 0 + +/* DAC Precharge Control Register */ +#define DAC_PRECH 0x0C +#define DAC_PRECH_PRCHGPDRV 7 +#define DAC_PRECH_PRCHGAUX1 6 +#define DAC_PRECH_PRCHGLNOR 5 +#define DAC_PRECH_PRCHGLNOL 4 +#define DAC_PRECH_PRCHGLNIR 3 +#define DAC_PRECH_PRCHGLNIL 2 +#define DAC_PRECH_PRCHG 1 +#define DAC_PRECH_ONMSTR 0 + +/* DAC Auxiliary Input Gain Control Register */ +#define DAC_AUXG 0x0D +#define DAC_AUXG_AUXG 0 + +/* DAC Reset Register */ +#define DAC_RST 0x10 +#define DAC_RST_RESMASK 2 +#define DAC_RST_RESFILZ 1 +#define DAC_RST_RSTZ 0 + +/* Power Amplifier Control Register */ +#define PA_CTRL 0x11 +#define PA_CTRL_APAON 6 +#define PA_CTRL_APAPRECH 5 +#define PA_CTRL_APALP 4 +#define PA_CTRL_APAGAIN 0 + +#endif /* _SND_AT73C213_H */ -- 1.5.2.3 --=-AUyv9PDBqTOzFdJ/BnKL--

1 0

[PATCH 1/1] ALSA sound driver for the AT73C213 DAC using Atmel SSC driver
by Hans-Christian Egtvedt 05 Jul '07

05 Jul '07

This patch adds support for the AT73C213 DAC using the misc Atmel SSC driver in I2S mode. The driver also requires a SPI to setup the registers and control volume. It has been tested with an AT32AP7000 on the ATSTK1000 development board. The driver should also work with any Atmel device with an SSC module supported by the Atmel SSC driver (atmel-ssc). The atmel-ssc driver is just submitted to the Linux kernel. Please see mail thread http://lkml.org/lkml/2007/7/16/32 Signed-off-by: Hans-Christian Egtvedt <hcegtvedt(a)atmel.com> --- include/linux/spi/at73c213.h | 25 + sound/spi/Kconfig | 25 + sound/spi/Makefile | 3 + sound/spi/at73c213.c | 1121 ++++++++++++++++++++++++++++++++++++++++++ sound/spi/at73c213.h | 119 +++++ 5 files changed, 1293 insertions(+), 0 deletions(-) create mode 100644 include/linux/spi/at73c213.h create mode 100644 sound/spi/at73c213.c create mode 100644 sound/spi/at73c213.h diff --git a/include/linux/spi/at73c213.h b/include/linux/spi/at73c213.h new file mode 100644 index 0000000..0f20a70 --- /dev/null +++ b/include/linux/spi/at73c213.h @@ -0,0 +1,25 @@ +/* + * Board-specific data used to set up AT73c213 audio DAC driver. + */ + +#ifndef __LINUX_SPI_AT73C213_H +#define __LINUX_SPI_AT73C213_H + +/** + * at73c213_board_info - how the external DAC is wired to the device. + * + * @ssc_id: SSC platform_driver id the DAC shall use to stream the audio. + * @dac_clk: the external clock used to provide master clock to the DAC. + * @shortname: a short discription for the DAC, seen by userspace tools. + * + * This struct contains the configuration of the hardware connection to the + * external DAC. The DAC needs a master clock and a I2S audio stream. It also + * provides a name which is used to identify it in userspace tools. + */ +struct at73c213_board_info { + int ssc_id; + struct clk *dac_clk; + char shortname[32]; +}; + +#endif /* __LINUX_SPI_AT73C213_H */ diff --git a/sound/spi/Kconfig b/sound/spi/Kconfig index 60fbb2a..0d08c29 100644 --- a/sound/spi/Kconfig +++ b/sound/spi/Kconfig @@ -3,4 +3,29 @@ menu "SPI devices" depends on SND != n +config SND_AT73C213 + tristate "Atmel AT73C213 DAC driver" + depends on ATMEL_SSC + select SND_PCM + help + Say Y here if you want to use the Atmel AT73C213 external DAC. This + DAC can be found on Atmel development boards. + + This driver requires the Atmel SSC driver for sound sink, a + peripheral found on most AT91 and AVR32 microprocessors. + + To compile this driver as a module, choose M here: the module will be + called snd-at73c213. + +config SND_AT73C213_TARGET_BITRATE + int "Target bitrate for AT73C213" + depends on SND_AT73C213 + default "48000" + range 8000 50000 + help + Sets the target bitrate for the bitrate calculator in the driver. + Limited by hardware to be between 8000 Hz and 50000 Hz. + + Set to 48000 Hz by default. + endmenu diff --git a/sound/spi/Makefile b/sound/spi/Makefile index 30d8ec9..026fb73 100644 --- a/sound/spi/Makefile +++ b/sound/spi/Makefile @@ -1,2 +1,5 @@ # Makefile for SPI drivers +snd-at73c213-objs := at73c213.o + +obj-$(CONFIG_SND_AT73C213) += snd-at73c213.o diff --git a/sound/spi/at73c213.c b/sound/spi/at73c213.c new file mode 100644 index 0000000..f514f47 --- /dev/null +++ b/sound/spi/at73c213.c @@ -0,0 +1,1121 @@ +/* + * Driver for AT73C213 16-bit stereo DAC connected to Atmel SSC + * + * Copyright (C) 2006-2007 Atmel Norway + * + * 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. + */ + +/*#define DEBUG*/ + +#include <linux/clk.h> +#include <linux/err.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/io.h> + +#include <sound/driver.h> +#include <sound/initval.h> +#include <sound/control.h> +#include <sound/core.h> +#include <sound/pcm.h> + +#include <linux/atmel-ssc.h> + +#include <linux/spi/spi.h> +#include <linux/spi/at73c213.h> + +#include "at73c213.h" + +#define BITRATE_MIN 8000 /* Hardware limit? */ +#define BITRATE_TARGET CONFIG_SND_AT73C213_TARGET_BITRATE +#define BITRATE_MAX 50000 /* Hardware limit. */ + +/* Initial (hardware reset) AT73C213 register values. */ +static u8 snd_at73c213_original_image[18] = +{ + 0x00, /* 00 - CTRL */ + 0x05, /* 01 - LLIG */ + 0x05, /* 02 - RLIG */ + 0x08, /* 03 - LPMG */ + 0x08, /* 04 - RPMG */ + 0x00, /* 05 - LLOG */ + 0x00, /* 06 - RLOG */ + 0x22, /* 07 - OLC */ + 0x09, /* 08 - MC */ + 0x00, /* 09 - CSFC */ + 0x00, /* 0A - MISC */ + 0x00, /* 0B - */ + 0x00, /* 0C - PRECH */ + 0x05, /* 0D - AUXG */ + 0x00, /* 0E - */ + 0x00, /* 0F - */ + 0x00, /* 10 - RST */ + 0x00, /* 11 - PA_CTRL */ +}; + +struct snd_at73c213 { + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *substream; + struct at73c213_board_info *board; + int irq; + int period; + unsigned long bitrate; + struct clk *bitclk; + struct ssc_device *ssc; + struct spi_device *spi; + u8 spi_wbuffer[2]; + u8 spi_rbuffer[2]; + /* Image of the SPI registers in AT73C213. */ + u8 reg_image[18]; + /* Protect registers against concurrent access. */ + spinlock_t lock; +}; + +#define get_chip(card) ((struct snd_at73c213 *)card->private_data) + +static int +snd_at73c213_write_reg(struct snd_at73c213 *chip, u8 reg, u8 val) +{ + struct spi_message msg; + struct spi_transfer msg_xfer = { + .len = 2, + .cs_change = 0, + }; + int retval; + + spi_message_init(&msg); + + chip->spi_wbuffer[0] = reg; + chip->spi_wbuffer[1] = val; + + msg_xfer.tx_buf = chip->spi_wbuffer; + msg_xfer.rx_buf = chip->spi_rbuffer; + spi_message_add_tail(&msg_xfer, &msg); + + retval = spi_sync(chip->spi, &msg); + + if (!retval) + chip->reg_image[reg] = val; + + return retval; +} + +static struct snd_pcm_hardware snd_at73c213_playback_hw = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER, + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 8000, /* Replaced by chip->bitrate later. */ + .rate_max = 50000, /* Replaced by chip->bitrate later. */ + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = 64 * 1024 - 1, + .period_bytes_min = 512, + .period_bytes_max = 64 * 1024 - 1, + .periods_min = 4, + .periods_max = 1024, +}; + +/* + * Calculate and set bitrate and divisions. + */ +static int snd_at73c213_set_bitrate(struct snd_at73c213 *chip) +{ + unsigned long ssc_rate = clk_get_rate(chip->ssc->clk); + unsigned long dac_rate_new, ssc_div, status; + unsigned long ssc_div_max, ssc_div_min; + int max_tries; + + /* + * We connect two clocks here, picking divisors so the I2S clocks + * out data at the same rate the DAC clocks it in ... and as close + * as practical to the desired target rate. + * + * The DAC master clock (MCLK) is programmable, and is either 256 + * or (not here) 384 times the I2S output clock (BCLK). + */ + + /* SSC clock / (bitrate * stereo * 16-bit). */ + ssc_div = ssc_rate / (BITRATE_TARGET * 2 * 16); + ssc_div_min = ssc_rate / (BITRATE_MAX * 2 * 16); + ssc_div_max = ssc_rate / (BITRATE_MIN * 2 * 16); + max_tries = (ssc_div_max - ssc_div_min) / 2; + + if (max_tries < 1) + max_tries = 1; + + /* ssc_div must be a power of 2. */ + ssc_div = (ssc_div + 1) & ~1UL; + + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) { + ssc_div -= 2; + if ((ssc_rate / (ssc_div * 2 * 16)) > BITRATE_MAX) + return -ENXIO; + } + + /* Search for a possible bitrate. */ + do { + /* SSC clock / (ssc divider * 16-bit * stereo). */ + if ((ssc_rate / (ssc_div * 2 * 16)) < BITRATE_MIN) + return -ENXIO; + + /* 256 / (2 * 16) = 8 */ + dac_rate_new = 8 * (ssc_rate / ssc_div); + + status = clk_round_rate(chip->board->dac_clk, dac_rate_new); + if (status < 0) + return status; + + /* Ignore difference smaller than 256 Hz. */ + if ((status/256) == (dac_rate_new/256)) + goto set_rate; + + ssc_div += 2; + } while (--max_tries); + + /* Not able to find a valid bitrate. */ + return -ENXIO; + +set_rate: + status = clk_set_rate(chip->board->dac_clk, status); + if (status < 0) + return status; + + /* Set divider in SSC device. */ + ssc_writel(chip->ssc->regs, CMR, ssc_div/2); + + /* SSC clock / (ssc divider * 16-bit * stereo). */ + chip->bitrate = ssc_rate / (ssc_div * 16 * 2); + + dev_info(&chip->spi->dev, + "at73c213: supported bitrate is %lu (%lu divider)\n", + chip->bitrate, ssc_div); + + return 0; +} + +static int snd_at73c213_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + + snd_at73c213_playback_hw.rate_min = chip->bitrate; + snd_at73c213_playback_hw.rate_max = chip->bitrate; + runtime->hw = snd_at73c213_playback_hw; + chip->substream = substream; + + return 0; +} + +static int snd_at73c213_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + chip->substream = NULL; + return 0; +} + +static int snd_at73c213_pcm_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_at73c213_pcm_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_at73c213_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + int block_size; + + block_size = frames_to_bytes(runtime, runtime->period_size); + + chip->period = 0; + + ssc_writel(chip->ssc->regs, PDC_TPR, + (long)runtime->dma_addr); + ssc_writel(chip->ssc->regs, PDC_TCR, runtime->period_size * 2); + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + block_size); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + + return 0; +} + +static int snd_at73c213_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + int retval = 0; + + spin_lock(&chip->lock); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + ssc_writel(chip->ssc->regs, IER, SSC_BIT(IER_ENDTX)); + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTEN)); + break; + case SNDRV_PCM_TRIGGER_STOP: + ssc_writel(chip->ssc->regs, PDC_PTCR, SSC_BIT(PDC_PTCR_TXTDIS)); + ssc_writel(chip->ssc->regs, IDR, SSC_BIT(IDR_ENDTX)); + break; + default: + dev_dbg(&chip->spi->dev, "spurious command %x\n", cmd); + retval = -EINVAL; + break; + } + + spin_unlock(&chip->lock); + + return retval; +} + +static snd_pcm_uframes_t +snd_at73c213_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_at73c213 *chip = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t pos; + unsigned long bytes; + + bytes = ssc_readl(chip->ssc->regs, PDC_TPR) + - (unsigned long)runtime->dma_addr; + + pos = bytes_to_frames(runtime, bytes); + if (pos >= runtime->buffer_size) + pos -= runtime->buffer_size; + + return pos; +} + +static struct snd_pcm_ops at73c213_playback_ops = { + .open = snd_at73c213_pcm_open, + .close = snd_at73c213_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_at73c213_pcm_hw_params, + .hw_free = snd_at73c213_pcm_hw_free, + .prepare = snd_at73c213_pcm_prepare, + .trigger = snd_at73c213_pcm_trigger, + .pointer = snd_at73c213_pcm_pointer, +}; + +static void snd_at73c213_pcm_free(struct snd_pcm *pcm) +{ + struct snd_at73c213 *chip = snd_pcm_chip(pcm); + if (chip->pcm) { + snd_pcm_lib_preallocate_free_for_all(chip->pcm); + chip->pcm = NULL; + } +} + +static int __devinit snd_at73c213_pcm_new(struct snd_at73c213 *chip, int device) +{ + struct snd_pcm *pcm; + int retval; + + retval = snd_pcm_new(chip->card, chip->card->shortname, + device, 1, 0, &pcm); + if (retval < 0) + goto out; + + pcm->private_data = chip; + pcm->private_free = snd_at73c213_pcm_free; + pcm->info_flags = SNDRV_PCM_INFO_BLOCK_TRANSFER; + strcpy(pcm->name, "at73c213"); + chip->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &at73c213_playback_ops); + + retval = snd_pcm_lib_preallocate_pages_for_all(chip->pcm, + SNDRV_DMA_TYPE_DEV, &chip->ssc->pdev->dev, + 64 * 1024, 64 * 1024); +out: + return retval; +} + +static irqreturn_t snd_at73c213_interrupt(int irq, void *dev_id) +{ + struct snd_at73c213 *chip = dev_id; + struct snd_pcm_runtime *runtime = chip->substream->runtime; + u32 status; + int offset; + int block_size; + int next_period; + int retval = IRQ_NONE; + + spin_lock(&chip->lock); + + block_size = frames_to_bytes(runtime, runtime->period_size); + status = ssc_readl(chip->ssc->regs, IMR); + + if (status & SSC_BIT(IMR_ENDTX)) { + chip->period++; + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + ssc_writel(chip->ssc->regs, PDC_TNPR, + (long)runtime->dma_addr + offset); + ssc_writel(chip->ssc->regs, PDC_TNCR, runtime->period_size * 2); + retval = IRQ_HANDLED; + } + + ssc_readl(chip->ssc->regs, IMR); + spin_unlock(&chip->lock); + + if (status & SSC_BIT(IMR_ENDTX)) + snd_pcm_period_elapsed(chip->substream); + + return retval; +} + +/* + * Mixer functions. + */ +static int snd_at73c213_mono_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & mask; + + if (invert) + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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, retval; + unsigned short val; + + val = (ucontrol->value.integer.value[0] & mask); + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val = (chip->reg_image[reg] & ~(mask << shift)) | val; + change = val != chip->reg_image[reg]; + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_stereo_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + int mask = (kcontrol->private_value >> 24) & 0xff; + + if (mask == 1) + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + else + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + + uinfo->count = 2; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = mask; + + return 0; +} + +static int snd_at73c213_stereo_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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_irq(&chip->lock); + + ucontrol->value.integer.value[0] = + (chip->reg_image[left_reg] >> shift_left) & mask; + ucontrol->value.integer.value[1] = + (chip->reg_image[right_reg] >> shift_right) & mask; + + if (invert) { + ucontrol->value.integer.value[0] = + (mask - ucontrol->value.integer.value[0]); + ucontrol->value.integer.value[1] = + (mask - ucontrol->value.integer.value[1]); + } + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_stereo_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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, retval; + 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_irq(&chip->lock); + + val1 = (chip->reg_image[left_reg] & ~(mask << shift_left)) | val1; + val2 = (chip->reg_image[right_reg] & ~(mask << shift_right)) | val2; + change = val1 != chip->reg_image[left_reg] + || val2 != chip->reg_image[right_reg]; + retval = snd_at73c213_write_reg(chip, left_reg, val1); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + retval = snd_at73c213_write_reg(chip, right_reg, val2); + if (retval) { + spin_unlock_irq(&chip->lock); + goto out; + } + + spin_unlock_irq(&chip->lock); + + return change; + +out: + return retval; +} + +static int snd_at73c213_mono_switch_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = 1; + uinfo->value.integer.min = 0; + uinfo->value.integer.max = 1; + + return 0; +} + +static int snd_at73c213_mono_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + int reg = kcontrol->private_value & 0xff; + int shift = (kcontrol->private_value >> 8) & 0xff; + int invert = (kcontrol->private_value >> 24) & 0xff; + + spin_lock_irq(&chip->lock); + + ucontrol->value.integer.value[0] = (chip->reg_image[reg] >> shift) & 0x01; + + if (invert) + ucontrol->value.integer.value[0] = + (0x01 - ucontrol->value.integer.value[0]); + + spin_unlock_irq(&chip->lock); + + return 0; +} + +static int snd_at73c213_mono_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_at73c213 *chip = snd_kcontrol_chip(kcontrol); + 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, retval; + unsigned short val; + + if (ucontrol->value.integer.value[0]) + val = mask; + else + val = 0; + + if (invert) + val = mask - val; + val <<= shift; + + spin_lock_irq(&chip->lock); + + val |= (chip->reg_image[reg] & ~(mask << shift)); + change = val != chip->reg_image[reg]; + + retval = snd_at73c213_write_reg(chip, reg, val); + + spin_unlock_irq(&chip->lock); + + if (retval) + return retval; + + return change; +} + +static int snd_at73c213_pa_volume_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 = ((kcontrol->private_value >> 16) & 0xff) - 1; + + return 0; +} + +static int snd_at73c213_line_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +static int snd_at73c213_aux_capture_volume_info( + struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + /* When inverted will give values 0x10001 => 0. */ + uinfo->value.integer.min = 14; + uinfo->value.integer.max = 31; + + return 0; +} + +#define AT73C213_MONO_SWITCH(xname, xindex, reg, shift, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_mono_switch_info, \ + .get = snd_at73c213_mono_switch_get, \ + .put = snd_at73c213_mono_switch_put, \ + .private_value = (reg | (shift << 8) | (mask << 16) | (invert << 24)) \ +} + +#define AT73C213_STEREO(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ +{ \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xindex, \ + .info = snd_at73c213_stereo_info, \ + .get = snd_at73c213_stereo_get, \ + .put = snd_at73c213_stereo_put, \ + .private_value = (left_reg | (right_reg << 8) \ + | (shift_left << 16) | (shift_right << 19) \ + | (mask << 24) | (invert << 22)) \ +} + +static struct snd_kcontrol_new snd_at73c213_controls[] __devinitdata = { +AT73C213_STEREO("Master Playback Volume", 0, DAC_LMPG, DAC_RMPG, 0, 0, 0x1f, 1), +AT73C213_STEREO("Master Playback Switch", 0, DAC_LMPG, DAC_RMPG, 5, 5, 1, 1), +AT73C213_STEREO("PCM Playback Volume", 0, DAC_LLOG, DAC_RLOG, 0, 0, 0x1f, 1), +AT73C213_STEREO("PCM Playback Switch", 0, DAC_LLOG, DAC_RLOG, 5, 5, 1, 1), +AT73C213_MONO_SWITCH("Mono PA Playback Switch", 0, DAC_CTRL, DAC_CTRL_ONPADRV, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PA Playback Volume", + .index = 0, + .info = snd_at73c213_pa_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = PA_CTRL | (PA_CTRL_APAGAIN << 8) | (0x0f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("PA High Gain Playback Switch", 0, PA_CTRL, PA_CTRL_APALP, 0x01, 1), +AT73C213_MONO_SWITCH("PA Playback Switch", 0, PA_CTRL, PA_CTRL_APAON, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Capture Volume", + .index = 0, + .info = snd_at73c213_aux_capture_volume_info, + .get = snd_at73c213_mono_get, + .put = snd_at73c213_mono_put, + .private_value = DAC_AUXG | (0 << 8) | (0x1f << 16) | (1 << 24), +}, +AT73C213_MONO_SWITCH("Aux Capture Switch", 0, DAC_CTRL, DAC_CTRL_ONAUXIN, 0x01, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Volume", + .index = 0, + .info = snd_at73c213_line_capture_volume_info, + .get = snd_at73c213_stereo_get, + .put = snd_at73c213_stereo_put, + .private_value = DAC_LLIG | (DAC_RLIG << 8) | (0 << 16) | (0 << 19) + | (0x1f << 24) | (1 << 22), +}, +AT73C213_MONO_SWITCH("Line Capture Switch", 0, DAC_CTRL, 0, 0x03, 0), +}; + +static int __devinit snd_at73c213_mixer(struct snd_at73c213 *chip) +{ + struct snd_card *card; + int errval, idx; + + if (chip == NULL || chip->pcm == NULL) + return -EINVAL; + + card = chip->card; + + strcpy(card->mixername, chip->pcm->name); + + for (idx = 0; idx < ARRAY_SIZE(snd_at73c213_controls); idx++) { + errval = snd_ctl_add(card, + snd_ctl_new1(&snd_at73c213_controls[idx], + chip)); + if (errval < 0) + goto cleanup; + } + + return 0; + +cleanup: + for (idx = 1; idx < ARRAY_SIZE(snd_at73c213_controls) + 1; idx++) { + struct snd_kcontrol *kctl; + kctl = snd_ctl_find_numid(card, idx); + if (kctl) + snd_ctl_remove(card, kctl); + } + return errval; +} + +/* + * Device functions + */ +static int snd_at73c213_ssc_init(struct snd_at73c213 *chip) +{ + /* + * Continuous clock output. + * Starts on falling TF. + * Delay 1 cycle (1 bit). + * Periode is 16 bit (16 - 1). + */ + ssc_writel(chip->ssc->regs, TCMR, + SSC_BF(TCMR_CKO, 1) + | SSC_BF(TCMR_START, 4) + | SSC_BF(TCMR_STTDLY, 1) + | SSC_BF(TCMR_PERIOD, 16 - 1)); + /* + * Data length is 16 bit (16 - 1). + * Transmit MSB first. + * Transmit 2 words each transfer. + * Frame sync length is 16 bit (16 - 1). + * Frame starts on negative pulse. + */ + ssc_writel(chip->ssc->regs, TFMR, + SSC_BF(TFMR_DATLEN, 16 - 1) + | SSC_BIT(TFMR_MSBF) + | SSC_BF(TFMR_DATNB, 1) + | SSC_BF(TFMR_FSLEN, 16 - 1) + | SSC_BF(TFMR_FSOS, 1)); + + return 0; +} + +static int snd_at73c213_chip_init(struct snd_at73c213 *chip) +{ + int retval; + unsigned char dac_ctrl = 0; + + retval = snd_at73c213_set_bitrate(chip); + if (retval) + goto out; + + /* Enable DAC master clock. */ + clk_enable(chip->board->dac_clk); + + /* Initialize at73c213 on SPI bus. */ + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x04); + if (retval) + goto out_clk; + msleep(1); + retval = snd_at73c213_write_reg(chip, DAC_RST, 0x03); + if (retval) + goto out_clk; + + /* Precharge everything. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0xff); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APAPRECH)); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_CTRL, + (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR)); + if (retval) + goto out_clk; + + msleep(50); + + /* Stop precharging PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, + (1<<PA_CTRL_APALP) | 0x0f); + if (retval) + goto out_clk; + + msleep(450); + + /* Stop precharging DAC, turn on master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, (1<<DAC_PRECH_ONMSTR)); + if (retval) + goto out_clk; + + msleep(1); + + /* Turn on DAC. */ + dac_ctrl = (1<<DAC_CTRL_ONDACL) | (1<<DAC_CTRL_ONDACR) + | (1<<DAC_CTRL_ONLNOL) | (1<<DAC_CTRL_ONLNOR); + + retval = snd_at73c213_write_reg(chip, DAC_CTRL, dac_ctrl); + if (retval) + goto out_clk; + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out_clk; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out_clk; + + /* Enable I2S device, i.e. clock output. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + goto out; + +out_clk: + clk_disable(chip->board->dac_clk); +out: + return retval; +} + +static int snd_at73c213_dev_free(struct snd_device *device) +{ + struct snd_at73c213 *chip = device->device_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + if (chip->irq >= 0) { + free_irq(chip->irq, chip); + chip->irq = -1; + } + + return 0; +} + +static int __devinit snd_at73c213_dev_init(struct snd_card *card, + struct spi_device *spi) +{ + static struct snd_device_ops ops = { + .dev_free = snd_at73c213_dev_free, + }; + struct snd_at73c213 *chip = get_chip(card); + int irq, retval; + + irq = chip->ssc->irq; + if (irq < 0) + return irq; + + spin_lock_init(&chip->lock); + chip->card = card; + chip->irq = -1; + + retval = request_irq(irq, snd_at73c213_interrupt, 0, "at73c213", chip); + if (retval) { + dev_dbg(&chip->spi->dev, "unable to request irq %d\n", irq); + goto out; + } + chip->irq = irq; + + memcpy(&chip->reg_image, &snd_at73c213_original_image, + sizeof(snd_at73c213_original_image)); + + retval = snd_at73c213_ssc_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_chip_init(chip); + if (retval) + goto out_irq; + + retval = snd_at73c213_pcm_new(chip, 0); + if (retval) + goto out_irq; + + retval = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); + if (retval) + goto out_irq; + + retval = snd_at73c213_mixer(chip); + if (retval) + goto out_snd_dev; + + snd_card_set_dev(card, &spi->dev); + + goto out; + +out_snd_dev: + snd_device_free(card, chip); +out_irq: + free_irq(chip->irq, chip); + chip->irq = -1; +out: + return retval; +} + +static int snd_at73c213_probe(struct spi_device *spi) +{ + struct snd_card *card; + struct snd_at73c213 *chip; + struct at73c213_board_info *board; + int retval; + char id[16]; + + board = spi->dev.platform_data; + if (!board) { + dev_dbg(&spi->dev, "no platform_data\n"); + return -ENXIO; + } + + if (!board->dac_clk) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return -ENXIO; + } + + if (IS_ERR(board->dac_clk)) { + dev_dbg(&spi->dev, "no DAC clk\n"); + return PTR_ERR(board->dac_clk); + } + + retval = -ENOMEM; + + /* Allocate "card" using some unused identifiers. */ + snprintf(id, sizeof id, "at73c213_%d", board->ssc_id); + card = snd_card_new(-1, id, THIS_MODULE, sizeof(struct snd_at73c213)); + if (!card) + goto out; + + chip = card->private_data; + chip->spi = spi; + chip->board = board; + + chip->ssc = ssc_request(board->ssc_id); + if (IS_ERR(chip->ssc)) { + dev_dbg(&spi->dev, "could not get ssc%d device\n", + board->ssc_id); + retval = PTR_ERR(chip->ssc); + goto out_card; + } + + retval = snd_at73c213_dev_init(card, spi); + if (retval) + goto out_ssc; + + strcpy(card->driver, "at73c213"); + strcpy(card->shortname, board->shortname); + sprintf(card->longname, "%s on irq %d", card->shortname, chip->irq); + + retval = snd_card_register(card); + if (retval) + goto out_ssc; + + dev_set_drvdata(&spi->dev, card); + + goto out; + +out_ssc: + ssc_free(chip->ssc); +out_card: + snd_card_free(card); +out: + return retval; +} + +static int __devexit snd_at73c213_remove(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + int retval; + + /* Stop playback. */ + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + + /* Mute sound. */ + retval = snd_at73c213_write_reg(chip, DAC_LMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RMPG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLOG, 0x3f); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_LLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_RLIG, 0x11); + if (retval) + goto out; + retval = snd_at73c213_write_reg(chip, DAC_AUXG, 0x11); + if (retval) + goto out; + + /* Turn off PA. */ + retval = snd_at73c213_write_reg(chip, PA_CTRL, (chip->reg_image[PA_CTRL]|0x0f)); + if (retval) + goto out; + msleep(10); + retval = snd_at73c213_write_reg(chip, PA_CTRL, (1<<PA_CTRL_APALP)|0x0f); + if (retval) + goto out; + + /* Turn off external DAC. */ + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x0c); + if (retval) + goto out; + msleep(2); + retval = snd_at73c213_write_reg(chip, DAC_CTRL, 0x00); + if (retval) + goto out; + + /* Turn off master power. */ + retval = snd_at73c213_write_reg(chip, DAC_PRECH, 0x00); + if (retval) + goto out; + +out: + /* Stop DAC master clock. */ + clk_disable(chip->board->dac_clk); + + ssc_free(chip->ssc); + snd_card_free(card); + dev_set_drvdata(&spi->dev, NULL); + + return 0; +} + +#ifdef CONFIG_PM +static int snd_at73c213_suspend(struct spi_device *spi, pm_message_t msg) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXDIS)); + clk_disable(chip->board->dac_clk); + + return 0; +} + +static int snd_at73c213_resume(struct spi_device *spi) +{ + struct snd_card *card = dev_get_drvdata(&spi->dev); + struct snd_at73c213 *chip = card->private_data; + + clk_enable(chip->board->dac_clk); + ssc_writel(chip->ssc->regs, CR, SSC_BIT(CR_TXEN)); + + return 0; +} +#else +#define snd_at73c213_suspend NULL +#define snd_at73c213_resume NULL +#endif + +static struct spi_driver at73c213_driver = { + .driver = { + .name = "at73c213", + }, + .probe = snd_at73c213_probe, + .suspend = snd_at73c213_suspend, + .resume = snd_at73c213_resume, + .remove = __devexit_p(snd_at73c213_remove), +}; + +static int __init at73c213_init(void) +{ + return spi_register_driver(&at73c213_driver); +} +module_init(at73c213_init); + +static void __exit at73c213_exit(void) +{ + spi_unregister_driver(&at73c213_driver); +} +module_exit(at73c213_exit); + +MODULE_AUTHOR("Hans-Christian Egtvedt <hcegtvedt(a)atmel.com>"); +MODULE_DESCRIPTION("Sound driver for AT73C213 with Atmel SSC"); +MODULE_LICENSE("GPL"); diff --git a/sound/spi/at73c213.h b/sound/spi/at73c213.h new file mode 100644 index 0000000..fd8b372 --- /dev/null +++ b/sound/spi/at73c213.h @@ -0,0 +1,119 @@ +/* + * Driver for the AT73C213 16-bit stereo DAC on Atmel ATSTK1000 + * + * Copyright (C) 2006 - 2007 Atmel Corporation + * + * 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, write to the Free Software + * Foundation, Inc., 59 Temple Place - Suite 330, Boston, MA + * 02111-1307, USA. + * + * The full GNU General Public License is included in this + * distribution in the file called COPYING. + */ + +#ifndef _SND_AT73C213_H +#define _SND_AT73C213_H + +/* DAC control register */ +#define DAC_CTRL 0x00 +#define DAC_CTRL_ONPADRV 7 +#define DAC_CTRL_ONAUXIN 6 +#define DAC_CTRL_ONDACR 5 +#define DAC_CTRL_ONDACL 4 +#define DAC_CTRL_ONLNOR 3 +#define DAC_CTRL_ONLNOL 2 +#define DAC_CTRL_ONLNIR 1 +#define DAC_CTRL_ONLNIL 0 + +/* DAC left line in gain register */ +#define DAC_LLIG 0x01 +#define DAC_LLIG_LLIG 0 + +/* DAC right line in gain register */ +#define DAC_RLIG 0x02 +#define DAC_RLIG_RLIG 0 + +/* DAC Left Master Playback Gain Register */ +#define DAC_LMPG 0x03 +#define DAC_LMPG_LMPG 0 + +/* DAC Right Master Playback Gain Register */ +#define DAC_RMPG 0x04 +#define DAC_RMPG_RMPG 0 + +/* DAC Left Line Out Gain Register */ +#define DAC_LLOG 0x05 +#define DAC_LLOG_LLOG 0 + +/* DAC Right Line Out Gain Register */ +#define DAC_RLOG 0x06 +#define DAC_RLOG_RLOG 0 + +/* DAC Output Level Control Register */ +#define DAC_OLC 0x07 +#define DAC_OLC_RSHORT 7 +#define DAC_OLC_ROLC 4 +#define DAC_OLC_LSHORT 3 +#define DAC_OLC_LOLC 0 + +/* DAC Mixer Control Register */ +#define DAC_MC 0x08 +#define DAC_MC_INVR 5 +#define DAC_MC_INVL 4 +#define DAC_MC_RMSMIN2 3 +#define DAC_MC_RMSMIN1 2 +#define DAC_MC_LMSMIN2 1 +#define DAC_MC_LMSMIN1 0 + +/* DAC Clock and Sampling Frequency Control Register */ +#define DAC_CSFC 0x09 +#define DAC_CSFC_OVRSEL 4 + +/* DAC Miscellaneous Register */ +#define DAC_MISC 0x0A +#define DAC_MISC_VCMCAPSEL 7 +#define DAC_MISC_DINTSEL 4 +#define DAC_MISC_DITHEN 3 +#define DAC_MISC_DEEMPEN 2 +#define DAC_MISC_NBITS 0 + +/* DAC Precharge Control Register */ +#define DAC_PRECH 0x0C +#define DAC_PRECH_PRCHGPDRV 7 +#define DAC_PRECH_PRCHGAUX1 6 +#define DAC_PRECH_PRCHGLNOR 5 +#define DAC_PRECH_PRCHGLNOL 4 +#define DAC_PRECH_PRCHGLNIR 3 +#define DAC_PRECH_PRCHGLNIL 2 +#define DAC_PRECH_PRCHG 1 +#define DAC_PRECH_ONMSTR 0 + +/* DAC Auxiliary Input Gain Control Register */ +#define DAC_AUXG 0x0D +#define DAC_AUXG_AUXG 0 + +/* DAC Reset Register */ +#define DAC_RST 0x10 +#define DAC_RST_RESMASK 2 +#define DAC_RST_RESFILZ 1 +#define DAC_RST_RSTZ 0 + +/* Power Amplifier Control Register */ +#define PA_CTRL 0x11 +#define PA_CTRL_APAON 6 +#define PA_CTRL_APAPRECH 5 +#define PA_CTRL_APALP 4 +#define PA_CTRL_APAGAIN 0 + +#endif /* _SND_AT73C213_H */ -- 1.5.2.3 --=-zhf1GaBSeEtL5hs3rxas--

1 0

[alsa-devel] error in snd_pcm_drain()
by Eliot Blennerhassett 05 Jul '07

05 Jul '07

Maybe an error in snd_pcm_drain - not sure what effect it might have... snd_pcm_group_for_each_entry(s, substream) { runtime = s->runtime; ... } up_read(&snd_pcm_link_rwsem); snd_pcm_stream_lock_irq(substream); /* pcm_native.c around line 1491: */ /* error? runtime has been changed by group_for_each should use substream->runtime instead? */ /* resume pause */ if (runtime->status->state == SNDRV_PCM_STATE_PAUSED) snd_pcm_pause(substream, 0); -- Eliot

2 1

[alsa-devel] [PATCH]fix mmap emulation bug of recording doesn't work
by Roy Huang 05 Jul '07

05 Jul '07

Record doesn't work if enabling mmap emulation and rate conversion needed, this patch fix this bug. diff -uNr alsa-lib-1.0.14a/src/pcm/pcm_hw.c alsa-lib-new/src/pcm/pcm_hw.c --- alsa-lib-1.0.14a/src/pcm/pcm_hw.c 2007-06-11 16:53:13.000000000 +0800 +++ alsa-lib-new/src/pcm/pcm_hw.c 2007-07-04 12:16:31.000000000 +0800 @@ -96,6 +96,7 @@ int shadow_appl_ptr: 1, avail_update_flag: 1, mmap_shm: 1; + snd_pcm_uframes_t hw_ptr; snd_pcm_uframes_t appl_ptr; /* restricted parameters */ snd_pcm_format_t format; @@ -373,7 +374,9 @@ if (pcm->stream == SND_PCM_STREAM_CAPTURE) { if (hw->mmap_shm) { hw->shadow_appl_ptr = 1; + hw->hw_ptr = 0; hw->appl_ptr = 0; + snd_pcm_set_hw_ptr(pcm, &hw->hw_ptr, -1, 0); snd_pcm_set_appl_ptr(pcm, &hw->appl_ptr, -1, 0); } else { hw->shadow_appl_ptr = 0; @@ -949,28 +952,91 @@ return size; } +static inline snd_pcm_uframes_t snd_pcm_hw_capture_avail(snd_pcm_t *pcm) +{ + snd_pcm_sframes_t avail; + snd_pcm_hw_t *hw = pcm->private_data; + + avail = hw->mmap_status->hw_ptr - hw->mmap_control->appl_ptr; + if (avail < 0) + avail += pcm->boundary; + return avail; +} + +static snd_pcm_sframes_t snd_pcm_hw_read_mmap(snd_pcm_t *pcm, snd_pcm_uframes_t size) +{ + snd_pcm_uframes_t xfer = 0; + snd_pcm_sframes_t err = 0; + snd_pcm_hw_t *hw = pcm->private_data; + if (! size) + return 0; + while (xfer < size) { + snd_pcm_uframes_t frames = size - xfer; + snd_pcm_uframes_t appl_offset = hw->mmap_control->appl_ptr % pcm->buffer_size; + snd_pcm_uframes_t cont = pcm->buffer_size - appl_offset; + if (cont < frames) + frames = cont; + switch (pcm->access) { + case SND_PCM_ACCESS_MMAP_INTERLEAVED: + { + const snd_pcm_channel_area_t *a = snd_pcm_mmap_areas(pcm); + char *buf = snd_pcm_channel_area_addr(a, appl_offset); + err = _snd_pcm_readi(pcm, buf, frames); + if (err >= 0) + frames = err; + break; + } + case SND_PCM_ACCESS_MMAP_NONINTERLEAVED: + { + snd_pcm_uframes_t channels = pcm->channels; + unsigned int c; + void *bufs[channels]; + const snd_pcm_channel_area_t *areas = snd_pcm_mmap_areas(pcm); + for (c = 0; c < channels; ++c) { + const snd_pcm_channel_area_t *a = &areas[c]; + bufs[c] = snd_pcm_channel_area_addr(a, appl_offset); + } + err = _snd_pcm_readn(pcm->fast_op_arg, bufs, frames); + if (err >= 0) + frames = err; + } + default: + SNDMSG("invalid access type %d", pcm->access); + return -EINVAL; + } + if (err < 0) + break; + xfer += frames; + } + if (xfer > 0) + return xfer; + return err; +} + static snd_pcm_sframes_t snd_pcm_hw_avail_update(snd_pcm_t *pcm) { snd_pcm_hw_t *hw = pcm->private_data; - snd_pcm_uframes_t avail; + snd_pcm_uframes_t avail, xfer_avail; sync_ptr(hw, 0); if (pcm->stream == SND_PCM_STREAM_PLAYBACK) { avail = snd_pcm_mmap_playback_avail(pcm); } else { avail = snd_pcm_mmap_capture_avail(pcm); - if (avail > 0 && hw->mmap_shm) { + if (avail < pcm->avail_min && hw->mmap_shm) { snd_pcm_sframes_t err; - snd_pcm_hw_t *hw = pcm->private_data; - hw->avail_update_flag = 1; - err = snd_pcm_read_mmap(pcm, avail); - hw->avail_update_flag = 0; - if (err < 0) - return err; - if ((snd_pcm_uframes_t)err != avail) - SNDMSG("short read %ld for avail %ld", err, avail); - return err; - } + xfer_avail = snd_pcm_hw_capture_avail(pcm); + xfer_avail -= xfer_avail % pcm->xfer_align; + if (xfer_avail > 0) { + hw->avail_update_flag = 1; + err = snd_pcm_hw_read_mmap(pcm, xfer_avail); + hw->avail_update_flag = 0; + if (err < 0) + return err; + hw->hw_ptr += err; + avail = snd_pcm_mmap_capture_avail(pcm); + } + } } switch (FAST_PCM_STATE(hw)) { case SNDRV_PCM_STATE_RUNNING:

2 4

[alsa-devel] default setting not fully working with new ALSA device
by Peter Matthias 05 Jul '07

05 Jul '07

Hello, I have got a 690G board with following lspic -nv output: 01:05.2 0403: 1002:7919 Subsystem: 1002:7919 Flags: bus master, fast devsel, latency 64, IRQ 19 Memory at fe9e8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- With the default setting, microphone does not work. With module parameter "modprobe -i snd-hda-intel model=6stack-dig" microphone works well. Please add to default setting. It was written in Linux Kernel documentation to report this issue here. Thank you, Peter

2 1

[alsa-devel] Very low headphones volume on HDA + Realtek ALC268
by Maurizio Monge 04 Jul '07

04 Jul '07

Hi, i've got an hda/alc268 card (pci id: [8086:284b], subsytem id: [1025:011f]) on my laptop, and the sound is ok for the loudspeakers but it is very very low for the headphones (the volume is at the top in the mixer, if i reduce the volume in the mixer the sound gets even lower). Under windows vista the headphones work well (ok, worked, before that i removed windows from my computer). I am using alsa hg, the most recent version at the moment i'm writing this e-mail. I there is any experiment that i can do or if i can help in any way (i know C very well) to fix this bug just let me know, and i'll be happy to contribute. More information about the hardware follows. Thanks! -- Ciao Maurizio http://stregatto.wordpress.com "Well we all shine on Like the moon and the stars and the sun" (John Lennon) lspci entry: 00:1b.0 Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: Acer Incorporated [ALI] Unknown device [1025:011f] 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 fc300000 (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 /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1025011f Revision Id: 0x100003 No Modem Function Group found 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 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2f 0x2f] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x05 [Vendor Defined Widget] wcaps 0xf00000: Mono 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 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x100111: Stereo PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0a [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0c [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0d [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x0e [Audio Mixer] wcaps 0x20010a: Mono Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00] Connection: 1 0x02 Node 0x0f [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Connection: 2 0x02 0x1d Node 0x10 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] [0x80 0x80] Connection: 3 0x03 0x1d 0x02 Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x13 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x14 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x02214020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0f Node 0x15 [Pin Complex] wcaps 0x40018d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a19830: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x19 [Pin Complex] wcaps 0x40008b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: Pincap 0x083724: IN Detect Pin Default 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281303e: [Jack] Line In at Ext Front Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 1 0x02 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4017952d: [N/A] Speaker at Ext N/A Conn = Analog, Color = Pink Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x23 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x90 0x90] Connection: 7 0x18 0x19 0x1a 0x1c 0x14 0x15* 0x12 Node 0x24 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x0b, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x01 0x01] Connection: 7 0x18 0x19 0x1a 0x1c 0x14* 0x15 0x13 /proc/asound/card0/codec#1: Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x10250093 Revision Id: 0x100000 Modem Function Group: 0x2

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[alsa-devel] Easiest way to test playback on an ASOC driver?
by Timur Tabi 04 Jul '07

04 Jul '07

I'm in the early stages of testing an ASOC driver, i.e. I'm trying to get playback to work. What is the easiest way to test playback? Is there a particular tool I should be using that does the most basic and direct playback of audio data through ALSA? -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] left/right bleed or crossover
by Howdy Pierce 04 Jul '07

04 Jul '07

I'm encountering the following problem on Fedora Core 6 running on a Dell Optiplex 745. This is a pretty vanilla computer with an ICH8 chipset and integrated AD1983 sound capabilities. I have wired up a funky line-input cable so that I have completely separate audio sources playing on the left and right inputs. When I either play these through to the speaker output directly (using alsamixer), or record the audio using arecord, the result is that there is a slight bit of bleeding or crossover, so I can faintly hear the left input on the right channel, and vice versa. If I record a wav file and then play it on another computer, the left/right bleeding follows the wav file, so it seems that the problem is in the capture process somehow. When I install Windows XP onto the same machine, the problem disappears, so I can rule out hardware errors. The application I'm working on relies on being able to reliably capture the right and left channels of audio independently. Since the problem can be seen using the standard arecord/aplay tools, though, I don't think it's my code. I've tried all the possible combinations inside alsamixer, and I don't think there's anything I've messed up there. I've put the relevant system details below. Thanks for any help you can provide. --Howdy ============================= Howdy Pierce Managing Partner Cardinal Peak, LLC work: (303) 665-3962 x101 cell: (303) 589-1645 email: howdy(a)cardinalpeak.com web: www.cardinalpeak.com ============================= [root@dundee ~]# cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xdfdfc000 irq 169 [root@dundee ~]# cat /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 1]: digital audio playback 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0] : control [root@dundee ~]# lspci 00:00.0 Host bridge: Intel Corporation 82Q963/Q965 Memory Controller Hub (rev 02) 00:01.0 PCI bridge: Intel Corporation 82Q963/Q965 PCI Express Root Port (rev 02) 00:02.0 Display controller: Intel Corporation 82Q963/Q965 Integrated Graphics Controller (rev 02) 00:02.1 Display controller: Intel Corporation 82Q963/Q965 Integrated Graphics Controller (rev 02) 00:1a.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #4 (rev 02) 00:1a.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #5 (rev 02) 00:1a.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI #2 (rev 02) 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 1 (rev 02) 00:1c.4 PCI bridge: Intel Corporation 82801H (ICH8 Family) PCI Express Port 5 (rev 02) 00:1d.0 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #1 (rev 02) 00:1d.1 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #2 (rev 02) 00:1d.2 USB Controller: Intel Corporation 82801H (ICH8 Family) USB UHCI #3 (rev 02) 00:1d.7 USB Controller: Intel Corporation 82801H (ICH8 Family) USB2 EHCI #1 (rev 02) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev f2) 00:1f.0 ISA bridge: Intel Corporation 82801HB/HR (ICH8/R) LPC Interface Controller (rev 02) 00:1f.2 IDE interface: Intel Corporation 82801H (ICH8 Family) 4 port SATA IDE Controller (rev 02) 00:1f.3 SMBus: Intel Corporation 82801H (ICH8 Family) SMBus Controller (rev 02) 00:1f.5 IDE interface: Intel Corporation 82801H (ICH8 Family) 2 port SATA IDE Controller (rev 02) 03:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5754 Gigabit Ethernet PCI Express (rev 02) 04:02.0 VGA compatible controller: nVidia Corporation NV18 [GeForce4 MX 4000 AGP 8x] (rev c1) [root@dundee ~]# rpm --query --all | grep alsa alsa-lib-1.0.12-2.fc6 alsa-lib-devel-1.0.12-2.fc6 alsa-utils-1.0.12-3.fc6 [root@dundee Intel]# cat /proc/asound/Intel/codec#0 Codec: Analog Devices AD1983 Address: 0 Vendor Id: 0x11d41983 Subsystem Id: 0x102801da Revision Id: 0x100400 Default PCM: rates 0x07f, bits 0x0e, types 0x1 Default Amp-In caps: N/A Default Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates 0x060, bits 0x02, types 0x5 Connection: 2 0x01* 0x04 Node 0x03 [Audio Output] wcaps 0x441: Stereo Power: 0x0 Node 0x04 [Audio Input] wcaps 0x100501: Stereo Power: 0x0 Connection: 1 0x14 Node 0x05 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x32 0x32] Pincap 0x0817: OUT Detect Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 2 0x03 0x0e* Node 0x06 [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x25 0x25] Pincap 0x081f: OUT HP Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 2 0x03 0x0e* Node 0x07 [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x3d, nsteps=0x3f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x991301f0: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0f Node 0x08 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x02a11020: [Jack] Mic at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x24: IN Node 0x09 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x0181302e: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x0a [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x414511f0: [N/A] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 4 0x03 0x0c 0x0d 0x0e* Node 0x0c [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x08* 0x09 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x09* 0x08 Node 0x0e [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x11 0x12 0x13 Node 0x0f [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0b Node 0x10 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x80] Node 0x11 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x97 0x17] Connection: 1 0x03 Node 0x12 [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 0x0c Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x9f 0x9f] Connection: 1 0x0d Node 0x14 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x0f 0x0f] Connection: 4 0x0c 0x0d* 0x0e 0x0f Node 0x15 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 15 0x05 0x06* 0x07 0x08 0x09 0x0b 0x0c 0x0d 0x0e 0x0f 0x10 0x11 0x12 0x13 0x14

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[alsa-devel] PCMCIA Audigy 2 broken in 2.6.21?
by Chuck Ebbert 04 Jul '07

04 Jul '07

Multiple users report the PCMCIA Audigy 2 adapter hangs the machine in 2.6.21 while it work OK in 2.6.20: https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=242208

3 3

[alsa-devel] [PATCH] Improve behaviour of Dreamcast aica ALSA driver in poor resource conditions
by Adrian McMenamin 04 Jul '07

04 Jul '07

This patch stops the driver from crashing in certain situations (eg if the network fails when NFS mounted), please apply. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> --- alsa-kernel/sh/aica.c 2007-06-23 15:25:55.000000000 +0100 +++ linux-2.6.21/sound/sh/aica.c 2007-07-03 22:28:16.000000000 +0100 @@ -366,7 +366,9 @@ *substream) { struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + flush_workqueue(aica_queue); del_timer(&dreamcastcard->timer); + aica_chn_halt(); kfree(dreamcastcard->channel); spu_disable(); return 0; @@ -402,16 +404,11 @@ static int snd_aicapcm_pcm_trigger(struct snd_pcm_substream *substream, int cmd) { - struct snd_card_aica *dreamcastcard; switch (cmd) { case SNDRV_PCM_TRIGGER_START: spu_begin_dma(substream); break; case SNDRV_PCM_TRIGGER_STOP: - dreamcastcard = substream->pcm->private_data; - if (dreamcastcard->timer.data) - del_timer(&dreamcastcard->timer); - aica_chn_halt(); break; default: return -EINVAL;

2 2

[alsa-devel] Errors in application building
by Paul Kavan 03 Jul '07

03 Jul '07

Hello all: I am working on my first bit of application code for ALSA and am not sure how to properly compile it against the libs. I have to cross compile the application on my host machine i686 PC for use on a ARM development board. I copied the alsa include directory into my staging include directory (this is where the cross-compiler works from), which allowed the program to find alsa/asoundlib.h However, I am still getting some errors that indicate to me it cannot find everything it needs. I am getting the following errors: grhuser@pcL7:$ arm-linux-gcc -Wall -o grh_audio grh_audio.c grh_audio.c: In function 'grh_capture': grh_audio.c:113: warning: pointer targets in passing argument 3 of 'snd_pcm_hw_params_set_rate_near' differ in signedness grh_audio.c: In function 'grh_playback': grh_audio.c:208: warning: pointer targets in passing argument 3 of 'snd_pcm_hw_params_set_rate_near' differ in signedness /tmp/ccMn48zN.o: In function `grh_capture': grh_audio.c:(.text+0x16c): undefined reference to `snd_pcm_open' grh_audio.c:(.text+0x198): undefined reference to `snd_strerror' grh_audio.c:(.text+0x1c0): undefined reference to `snd_pcm_hw_params_malloc' grh_audio.c:(.text+0x1e4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x214): undefined reference to `snd_pcm_hw_params_any' grh_audio.c:(.text+0x238): undefined reference to `snd_strerror' grh_audio.c:(.text+0x26c): undefined reference to `snd_pcm_hw_params_set_access' grh_audio.c:(.text+0x290): undefined reference to `snd_strerror' grh_audio.c:(.text+0x2c4): undefined reference to `snd_pcm_hw_params_set_format' grh_audio.c:(.text+0x2e8): undefined reference to `snd_strerror' grh_audio.c:(.text+0x324): undefined reference to `snd_pcm_hw_params_set_rate_near' grh_audio.c:(.text+0x348): undefined reference to `snd_strerror' grh_audio.c:(.text+0x380): undefined reference to `snd_pcm_hw_params_set_channels' grh_audio.c:(.text+0x3a4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x3d4): undefined reference to `snd_pcm_hw_params' grh_audio.c:(.text+0x3f8): undefined reference to `snd_strerror' grh_audio.c:(.text+0x420): undefined reference to `snd_pcm_hw_params_free' grh_audio.c:(.text+0x42c): undefined reference to `snd_pcm_prepare' grh_audio.c:(.text+0x450): undefined reference to `snd_strerror' grh_audio.c:(.text+0x494): undefined reference to `snd_pcm_readi' grh_audio.c:(.text+0x4c0): undefined reference to `snd_strerror' grh_audio.c:(.text+0x500): undefined reference to `snd_pcm_close' /tmp/ccMn48zN.o: In function `grh_playback': grh_audio.c:(.text+0x5b8): undefined reference to `snd_pcm_open' grh_audio.c:(.text+0x5e4): undefined reference to `snd_strerror' grh_audio.c:(.text+0x60c): undefined reference to `snd_pcm_hw_params_malloc' grh_audio.c:(.text+0x630): undefined reference to `snd_strerror' grh_audio.c:(.text+0x660): undefined reference to `snd_pcm_hw_params_any' grh_audio.c:(.text+0x684): undefined reference to `snd_strerror' grh_audio.c:(.text+0x6b8): undefined reference to `snd_pcm_hw_params_set_access' grh_audio.c:(.text+0x6dc): undefined reference to `snd_strerror' grh_audio.c:(.text+0x710): undefined reference to `snd_pcm_hw_params_set_format' grh_audio.c:(.text+0x734): undefined reference to `snd_strerror' grh_audio.c:(.text+0x770): undefined reference to `snd_pcm_hw_params_set_rate_near' grh_audio.c:(.text+0x794): undefined reference to `snd_strerror' grh_audio.c:(.text+0x7cc): undefined reference to `snd_pcm_hw_params_set_channels' grh_audio.c:(.text+0x7f0): undefined reference to `snd_strerror' grh_audio.c:(.text+0x820): undefined reference to `snd_pcm_hw_params' grh_audio.c:(.text+0x844): undefined reference to `snd_strerror' grh_audio.c:(.text+0x86c): undefined reference to `snd_pcm_hw_params_free' grh_audio.c:(.text+0x878): undefined reference to `snd_pcm_prepare' grh_audio.c:(.text+0x89c): undefined reference to `snd_strerror' grh_audio.c:(.text+0x8e0): undefined reference to `snd_pcm_writei' grh_audio.c:(.text+0x90c): undefined reference to `snd_strerror' grh_audio.c:(.text+0x94c): undefined reference to `snd_pcm_close' collect2: ld returned 1 exit status My application code is attached. If anyone could give me some pointers as to why these errors are occurring, I would appreciate it greatly. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

2 4

[alsa-devel] namehint problem...
by Fausto Carvalho Marques SIlva 03 Jul '07

03 Jul '07

Compiling alsa-lib-1.0.14rc4 with the above options: ./configure --disable-aload --disable-alisp --disable-instr --disable-seq --disable-hwdep --disable-rawmidi --with-debug=no --with-pcm-plugins=hw --with-ctl-plugins=hw After the build of asound and a linking against a simple program: #include <stdio.h> #include <stdlib.h> #include <alsa/asoundlib.h> main (int argc, char *argv[]) { int i; int err; short buf[128]; snd_pcm_t *playback_handle; snd_pcm_hw_params_t *hw_params; if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) { fprintf (stderr, "cannot open audio device %s (%s)\n", argv[1], snd_strerror (err)); exit (1); } snd_pcm_close (playback_handle); exit (0); } gcc -o teste teste.c -I./include src/.libs/libasound.so.2.0.0 -ldl src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_get_name' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_get_name' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_sizeof' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_set_device' src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_set_device' src/.libs/libasound.so.2.0.0: undefined reference to `snd_hwdep_info_sizeof' src/.libs/libasound.so.2.0.0: undefined reference to `snd_rawmidi_info_set_stream' collect2: ld returned 1 exit status After some verification discoverd that the problem is on src/control/namehint.c that references this functions but the configure removed rawmidi and hwdep. How can we resolve this problem? Fausto Carvalho Marques Silva

2 1

[alsa-devel] Application Coding
by Paul Kavan 03 Jul '07

03 Jul '07

Hello all: Beginning to write my application code now that I have alsa SoC working fairly well on my ARM development board. My audio application is fairly simple. I will be doing raw, S8, samples at an 8k sample rate. Essentially I need to do playback of a stream when signaled by an application to do so and capture when signaled by a button press. Capture takes precedent over playback. I am really new to the alsa api and I am planning on using aplay as a template. However, if anyone knows of an easier piece of code that may work as a template, please feel free to point it out to me. I am using the API documentation from the alsa-project website. If anyone knows of another good piece of documentation, please feel free to let me know. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] SALSA-Lib: snd_pcm_hw_params fails with MMAP
by J. Scott Merritt 03 Jul '07

03 Jul '07

Dear List, Using SALSA-Lib 0.0.5 ... After calling snd_pcm_hw_params_set_access with an access mode of SND_PCM_ACCESS_MMAP_INTERLEAVED, the subsequent call to snd_pcm_hw_params returns a snd_strerror error code of "Invalid argument". This same program operates properly in the target environment (ARM PXA270) when compiled and linked with the standard ALSA LIB 1.0.13. Thanks, Scott.

2 2

Re: [alsa-devel] ALSA - snd-hda-intel - Realtek ALC268
by Peter Poulsen 03 Jul '07

03 Jul '07

Citat af Chris Cheney <chris.cheney(a)canonical.com>: <snip> > I already have alc268 working fine on my box for sound, just tested it. > :) I am now working on making sure all the other functionality is > properly implemented. Sounds awesome :) looking forward to trying out your code /peter

1 0

[alsa-devel] roland synth sh-201
by carlo beccaria 03 Jul '07

03 Jul '07

hi all, with this change in usbquirks.h , i can use my synth roland sh-201 { /* Roland SH-201 */ USB_DEVICE(0x0582, 0x00ad), .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { .vendor_name = "Roland", .product_name = "SH-201", .ifnum = QUIRK_ANY_INTERFACE, .type = QUIRK_COMPOSITE, .data = (const struct snd_usb_audio_quirk[]) { { .ifnum = 0, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 0, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x01, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 1, .type = QUIRK_AUDIO_FIXED_ENDPOINT, .data = & (const struct audioformat) { .format = SNDRV_PCM_FORMAT_S24_3LE, .channels = 2, .iface = 1, .altsetting = 1, .altset_idx = 1, .attributes = 0, .endpoint = 0x81, .ep_attr = 0x01, .rates = SNDRV_PCM_RATE_CONTINUOUS, .rate_min = 44100, .rate_max = 44100, } }, { .ifnum = 2, .type = QUIRK_MIDI_FIXED_ENDPOINT, .data = & (const struct snd_usb_midi_endpoint_info) { .out_cables = 0x0007, .in_cables = 0x0007 } }, { .ifnum = -1 } } } }, my kernel is 2.6.21.5-rt17, (plus preempt patch) and i use debian testing the hardware seem to be very similar to Roland SC-D70 (windows .inf file are very similar) this is the device id Bus 001 Device 003: ID 0582:00ad Roland Corp. in attach you can see the lsusb (verbose) output becks@vaix:~$ amidi -l Dir Device Name IO hw:1,0,0 SH-201 MIDI 1 IO hw:1,0,1 SH-201 MIDI 2 IO hw:1,0,2 SH-201 MIDI 3 muse works fine with jack and midi output becks@vaix:~$ aseqdump -p 20:0 Waiting for data. Press Ctrl+C to end. Source_ Event_________________ Ch _Data__ 20:0 Note on 1 52 17 20:0 Note off 1 52 55 also midi-in works becks@vaix:~$ cat /proc/asound/cards 0 [A5451 ]: ALI5451 - ALI 5451 ALI 5451 at 0x8400, irq 9 1 [SH201 ]: USB-Audio - SH-201 Roland SH-201 at usb-0000:00:0c.0-1, full speed when i use my synth like a sound card, it don't work, i don't know exactly why (if is a patch problem, or rate/format) becks@vaix:~$ aplay -D hw:1,0 /usr/share/sounds/KDE_Window_UnSticky.wav Playing WAVE '/usr/share/sounds/KDE_Window_UnSticky.wav' : Signed 16 bit Little Endian, Rate 22050 Hz, Mono aplay: set_params:904: Sample format non available let me know if you need more info thanks -- carlo beccaria

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Re: [alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 03 Jul '07

03 Jul '07

I've attached the file. I tried using aplay and ffplay- both had no trouble opening and playing a file, there was just no sound. --------------------------------------- Original E-mail From: Takashi Iwai [tiwai(a)suse.de] Date: 07/02/2007 01:25 PM To: Steven Zakulec [spzakulec(a)mail.widener.edu] Subject: Re: [alsa-devel] [HDA-Intel/ALC268] no sound At Wed, 27 Jun 2007 17:34:15 -0400 (EDT), Steven Zakulec wrote: > > (Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) > > I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. And the contents of /proc/asound/card0/codec#* files, please. If you change the mixer status, these files also changed, too. Also, please inform what you tested more exactly. In which hardware configuration and which application with what parameters... Takashi > > state.Intel { > control.1 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 64' > iface MIXER > name 'Front Playback Volume' > value.0 57 > value.1 57 > } > control.2 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Front Playback Switch' > value.0 true > value.1 true > } > control.3 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 64' > iface MIXER > name 'Headphone Playback Volume' > value.0 63 > value.1 63 > } > control.4 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Headphone Playback Switch' > value.0 true > value.1 true > } > control.5 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 31' > iface MIXER > name 'Capture Volume' > value.0 1 > value.1 1 > } > control.6 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > value.0 false > value.1 false > } > control.7 { > comment.access 'read write' > comment.type INTEGER > comment.count 2 > comment.range '0 - 31' > iface MIXER > name 'Capture Volume' > index 1 > value.0 1 > value.1 1 > } > control.8 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 2 > iface MIXER > name 'Capture Switch' > index 1 > value.0 false > value.1 false > } > control.9 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > value Mic > } > control.10 { > comment.access 'read write' > comment.type ENUMERATED > comment.count 1 > comment.item.0 Mic > comment.item.1 'Front Mic' > iface MIXER > name 'Input Source' > index 1 > value Mic > } > control.11 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Off-hook Switch' > value false > } > control.12 { > comment.access 'read write' > comment.type BOOLEAN > comment.count 1 > iface MIXER > name 'Caller ID Switch' > value false > } > control.13 { > comment.access 'read write user' > comment.type INTEGER > comment.count 2 > comment.range '0 - 255' > comment.tlv '0000000100000008ffffec1400000014' > iface MIXER > name 'PCM Playback Volume' > value.0 255 > value.1 255 > } > } > hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) > [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 > [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 > [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) > > > > comment.count 2 > : > > _______________________________________________ > 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] HDA Intel/ALC268
by Peter Poulsen 02 Jul '07

02 Jul '07

Is there anybody that has had any luck with this, or I'm a fighting a loosing battle? /peter

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[alsa-devel] [HDA-Intel/ALC268] no sound
by Steven Zakulec 02 Jul '07

02 Jul '07

(Continued from this thread: http://mailman.alsa-project.org/pipermail/alsa-devel/2007-June/001459.html) I've included the result of alsactl -f somefile store. I thought it got sent to the list earlier, but apparently not. Below this I also added the alsa things from dmesg. state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Front Playback Volume' value.0 57 value.1 57 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' iface MIXER name 'Headphone Playback Volume' value.0 63 value.1 63 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' value.0 1 value.1 1 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 false value.1 false } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Capture Volume' index 1 value.0 1 value.1 1 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.9 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' value Mic } control.10 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 'Front Mic' iface MIXER name 'Input Source' index 1 value Mic } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Off-hook Switch' value false } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Caller ID Switch' value false } control.13 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } hda_codec: Unknown model for ALC268, trying auto-probe from BIOS... [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2339: autoconfig: line_outs=1 (0x14/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2343: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2347: hp_outs=1 (0x15/0x0/0x0/0x0/0x0) [ 18.512000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/hda_codec.c:2355: inputs: mic=0x19, fmic=0x18, line=0x0, fline=0x0, cd=0x0, aux=0x0 [ 19.236000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 19.244000] ALSA /home/szakulec/alsa-hg/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:256: Link Frame Detect(FDT) is not ready (line status: 0000) comment.count 2 :

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[alsa-devel] [HDA Intel/STAC92xx] Alsa 1.0.14: error in stac92xx_auto_fill_dac_nid
by Jan de Mooij 02 Jul '07

02 Jul '07

I have a strange problem with alsa on Ubuntu Gutsy. If I compile and install the Alsa 1.0.14 driver, my sound card is not detected, and sound doesn't work. I also get the following message in dmesg: [ 33.360190] stac92xx_auto_fill_dac_nids: No available DAC for pin 0x15 So the problem seems to be in the stac92xx_auto_fill_dac_nids function. When I copy that function from alsa 1.0.13 into the 1.0.14 source, everything works okay... It looks like the function is rewritten for version 1.0.14, but for me it does not work anymore. I have to do the above steps with each new kernel update in Ubuntu Gutsy... Therefore I would like to fix it, but I don't know where to start. Thanks in advance. Jan

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[alsa-devel] Acer Aspire 9810 hda-intel RealTek ALC883 no sound thru speakers only via SPDIF/Headphones.
by Adrian Grassi 02 Jul '07

02 Jul '07

Hi guys, I have a Acer Aspire 9810 with hda-intel realtek ALC883 chipset and no matter what version of alsa and ditribution I try I only get sound thru the shared SPDIF/Headphone Jack. This laptop has a built in 2.1 power speaker system which works fine under windows but not in any linux distro's I have tried: Ubuntu 7.04 - Kubuntu 7.04 - Suse 10.2 - Ubuntu Gutsy 7.10 Beta - Mandriva Spring 2007.1 I think this is a configuration issue however I can't find any info anywhere on how to fix it, I did find a few posts where several people are having the same issue. Can someone please shed some light on how to fix this issue, if any additional information is needed please let me know, I have been trying to solve this for over 5 days now and is driving me bananas. Here is the aadebug from alsa-wiki: Debug INFO ALSA Audio Debug v0.1.0 - Sat Jun 30 02:01:40 EDT 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux XTTG-Mob1 2.6.22-7-generic #1 SMP Mon Jun 25 17:33:14 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 256024 1 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,saa7134_alsa,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 2 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 12 snd_hda_intel,saa7134_alsa,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 10888 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 0xd2300000 irq 23 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xd2005000 irq 16 2: : timer 3: : sequencer 4: [ 1- 0]: digital audio capture 5: [ 1] : control 6: [ 0- 2]: digital audio capture 7: [ 0- 0]: digital audio playback 8: [ 0- 0]: digital audio capture 9: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 01-00: SAA7134 PCM : SAA7134 PCM : capture 1 00-02: ALC883 Analog : ALC883 Analog : capture 2 00-00: ALC883 Analog : ALC883 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 controlC1 pcmC0D0c pcmC0D0p pcmC0D2c pcmC1D0c seq timer CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 CPU T7200 @ 2.00GHz cpu MHz : 2000.000 model name : Intel(R) Core(TM)2 CPU T7200 @ 2.00GHz cpu MHz : 2000.000 RAM ------------------------------------------------------- MemTotal: 2074528 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS/940GML and 945GT Express Memory Controller Hub (rev 03) 09:04.0 Multimedia controller: Philips Semiconductors SAA7133/SAA7135 Video Broadcast Decoder (rev d1) 09:06.2 Mass storage controller: Texas Instruments 5-in-1 Multimedia Card Reader (SD/MMC/MS/MS PRO/xD)

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[alsa-devel] High-pitched noise on output from cmi9739 (snd_intel8x0)
by j t 02 Jul '07

02 Jul '07

Hi all. I have a cmi9739 (sis7012) built into an asus k7s8x motherboard which produces a quiet but noticeable noise on top of the regular output (the noise only occurs during playback). I logged a bug to alsa's bugtrack: (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2288) about a year ago, but unfortunately I still have the problem with alsa release 1.0.14 To try to find out what's going on, I tried using seven (7) different drivers for this soundcard (both Linux and Windows XP) to find out when the problem occurs. Here's the results: a) Alsa (snd_intel8x0) version 1.0.14 under Ubuntu 7.10 alpha 2. Problem EXISTS. b) OSS 4Front Opensound v4.1 (v4.1test0-070624). Problem EXISTS c) CMedia Linux OSS driver cmaudio-070 (from http://www.cmedia.com.tw). Problem EXISTS. d) Windows XP "built-in" driver for SiS 7012 version 5.10.0.6120. Problem EXISTS. e) Windows XP drivers from Asus website for SiS 7012 version 5.12.1.38. Problem DOES NOT EXIST. f) Windows XP drivers from CMedia website (http://www.cmedia.com.tw) version 5.12.1.39. Problem DOES NOT EXIST. g) Windows XP drivers from CMedia forum (http://www.cmedia.com.tw/forums) version 5.12.1.51. Problem DOES NOT EXIST. Given that the problem does not happen with three out of the seven drivers, I assume it's a driver problem. Is there anything that I can do debug this further? Thank you.

2 1

[alsa-devel] SALSA-Lib versus C++
by J. Scott Merritt 02 Jul '07

02 Jul '07

Dear List, It -appears- to me that the source file headers generated by SALSA-Lib-0.0.5 (for example SALSA:asoundlib.h) are not generally usable from within a C++ program compiled with g++. The first problem that I encountered was a failure in IOC_TYPECHECK from asm-generic/ioctl.h. After much searching, I concluded that this is a component of continuing struggles related to "sanitized linux-kernel-headers". Some of the Debian folks have supplied patches to disable this particular check when the ioctl.h file is included from userland, but these patched headers are not present in my cross-compile environment. This behavior apparently is not a problem with gcc or g++ < 2.3.3 - but shows up in later versions of g++. ... a real mess as best I can tell ... The next problem is that when warnings are enabled, g++ complains vigorously about void pointers that are implicitly cast to typed pointers through-out all of the SALSA-Lib supplied header files. Perhaps SALSA-Lib was never intended from usable from C++ programs - and indeed it could argued that its intended application (embedded systems) should avoid the use of C++. If this is the case, perhaps a mention of this restriction might be appropriate on the SALSA-Lib Website and/or in its README file ? Thanks again for all of your help in trying to get this working. Best regards, Scott.

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[alsa-devel] NVidia AC97 / CK804: arecord hangs in system call poll()
by Rainer Menzner 02 Jul '07

02 Jul '07

Hi, my soundcard is an onboard AC97 CK804 NVidia. I use snd-intel8x0 module. Everything works fine except recording sound. When calling arecord -f cd buffer.wav the program infinitely hangs, and an strace shows that it is a poll() call that does not return. After interrupting, a wave file of 44 bytes which is the header only is left. My setup is Kernel 2.6.21.3, all ALSA-software is updated to the latest version, that is 1.0.14(a). In a previous setup with an older ALSA version and kernel 2.6.16.11 recording did work with the same hardware. Now that I updated all ALSA software, even with that older kernel version the same recording problem came up. From that I concluded that it is somehow related to the ALSA-lib version. I have tried to downgrade ALSA-lib down to 1.0.0 to no avail. I have also made tests appropriate if this would be an interrupt problem (using pci=noroute et al) which also did not solve the problem. I have scrolled though the archives of alsa-user and there have been a few posts regarding the same problem (but with other chip sets). However, in those threads no definitive solution of the problem has been discussed. So I tried it again in alsa-user but did not get any feedback ... Now, I finally try this developer list. Your comments and help are much appreciated. Let me know if you need further info. I'm not a Linux system programmer or specialist, but I would be willing to do some debugging if somebody explains what to check. Thanks for any support, -Rainer

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[alsa-devel] hda-Intel Realtek ALC883 chipset support
by Adriano Grassi 30 Jun '07

30 Jun '07

Hi guys I have sound on different linux distro's however only the spdif/headphone share output, no sound from the internal speakers I have been trying to get this work for the last 5 days no success. So i was wondering if the support for this chipset is in place. I even tried the lastest candidate release alsa files 14rc4 with no luck, this is a high definition audio card with multiple outputs analog and digital and i have tried everything i can think of with no success, I tried multiple distro's with the same result, I don't this it is a driver issue is probably more like a configuration issue. Can anyone help me figure this one out. Thanks. 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd2300000 irq 98 2: : timer 3: [ 0- 2]: digital audio playback 4: [ 0- 1]: digital audio capture 5: [ 0- 0]: digital audio playback 6: [ 0- 0]: digital audio capture 7: [ 0] : control 8: : sequencer cat: /proc/asound/Intel: Is a directory 0 snd_hda_intel cat: /proc/asound/oss: Is a directory 00-02: ALC883 Digital : ALC883 Digital : playback 1 00-01: ALC883 Analog : ALC883 Analog : capture 2 00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 2 cat: /proc/asound/seq: Is a directory G0: system timer : 4000.000us (10000000 ticks) P0-0-0: PCM playback 0-0-0 : SLAVE P0-0-1: PCM capture 0-0-1 : SLAVE P0-0-3: PCM capture 0-0-3 : SLAVE P0-1-1: PCM capture 0-1-1 : SLAVE P0-1-3: PCM capture 0-1-3 : SLAVE P0-2-0: PCM playback 0-2-0 : SLAVE Advanced Linux Sound Architecture Driver Version 1.0.12rc1 (Thu Jun 22 13:55:50 2006 UTC).

1 0

[alsa-devel] Digigram VX source code
by Colin Stewart Campbell 30 Jun '07

30 Jun '07

About 5 years ago Digigram released their VX soundcard source for OSX through the alsa project. Is there somewhere I can still get a hold of it? I have in mind a legacy card coreaudio driver for Intel macs. Colin

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[alsa-devel] helping debugging
by Thierry Vignaud 30 Jun '07

30 Jun '07

Hi Here's a suggestion from Dave Airlie: http://airlied.livejournal.com/46087.html "If you are going to have lots of unsigned int sized free running pointers called things like hw_ptr or hw.ptr or anything, for debugging it would be really cool if you started those counters at INT_MAX-100000 or something instead of at zero, 24 hours waiting for a bug to trip in the pcm rate conversion is not fun and I'm guessing it'll be some stupid interger overflow or dumb boundary condition hitting..."

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Re: [alsa-devel] Fwd: Problems with hda_intel, Santa Rosa, and suspend
by albox＠free.fr 30 Jun '07

30 Jun '07

Could this issue be related to the other one with Medion laptops : https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3160 Takashi, you said you were going to check the older patch that was working on ALSA 1.0.13, did you have a chance to do that ? Takashi Iwai wrote: > At Mon, 25 Jun 2007 15:16:41 -0500, > Matt Mullins wrote: >> On 6/25/07, Takashi Iwai <tiwai(a)suse.de> wrote: >>>> Yes, single_cmd mode is used always. >>> That's bad. >> I believe we have established that. What can I do about it? *Can* I >> do anything about it? What functions should I start debugging? Has >> this happened with other codecs before? What documentation will I >> need or want other than the Intel HDA and STAC9205 specifications? > > Similar problems have been seen in some laptops, too. I vaguely > remember that they need some ACPI-related boot options to fix it > somehow. I think it's likely irrelevant with the codec chip, but > rather a problem of the controller side. > >> I'm definitely willing to help work on this, as not many people have >> this hardware. I'm just new to the sound system, so I'm still trying >> to figure out what-calls-what. > > > Takashi > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > >

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Re: [alsa-devel] Creative Soundblaster X-Fi
by Ash Willis 30 Jun '07

30 Jun '07

Hi all, Sorry for straying a little off topic, but talk of these couple of chips got me thinking about unsupported hardware in general. Does anyone know the legal implications of reverse engineering with regards to producing new drivers? I myself wouldn't mind reversing a certain chip or two but I'm not too aware of the legality of it all, particularly in the case where the vendor is hostile or unwilling to release chip information. In general, is it legally acceptable to reverse engineer for the purposes of interoperability? Are vendors likely to make a noise if their hardware gains support under Linux without or against their consent? or is it more a matter of protecting the original product specification than hiding details of the product's interface? Takashi, may I ask your thoughts on this? What's your position on accepting reverse engineered code? (Let's assume that the driver works as well as one written in a more correct manner ;)) I'm fully aware that uncooperative vendors don't deserve to have their hardware supported, but I think that Linux users and Linux itself deserves the support. Ash -- We've Got Your Name at http://www.mail.com! Get a FREE E-mail Account Today - Choose From 100+ Domains

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[alsa-devel] [1/2] 2.6.22-rc6: known regressions v2
by Michal Piotrowski 30 Jun '07

30 Jun '07

Hi all, Here is a list of some known regressions in 2.6.22-rc6. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions List of Aces Name Regressions fixed since 21-Jun-2007 Hugh Dickins 2 Andi Kleen 1 Björn Steinbrink 1 Bjorn Helgaas 1 Jean Delvare 1 Olaf Hering 1 Siddha, Suresh B 1 Trent Piepho 1 Ville Syrjälä 1 Unclassified Subject : Kernel hang on CMOS_READ References : http://lkml.org/lkml/2007/6/3/138 Submitter : Rodrigo Luiz <troops.of.doom(a)gmail.com> Status : Unknown ALSA Subject : snd-aoa causes badness in lib/kref.c:33 References : http://bugzilla.kernel.org/show_bug.cgi?id=8513 Submitter : Ben Collins <bcollins(a)ubuntu.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Status : Unknown FS Subject : 2.6.22-rc4-git5 reiserfs: null ptr deref. References : http://lkml.org/lkml/2007/6/13/322 Submitter : Randy Dunlap <randy.dunlap(a)oracle.com> Handled-By : Vladimir V. Saveliev <vs(a)namesys.com> Status : problem is being debugged IDE Subject : 2.6.22-rcX: hda: lost interrupt References : http://lkml.org/lkml/2007/6/29/121 Submitter : David Chinner <dgc(a)sgi.com> Status : unknown SATA/PATA Subject : libata IT821X driver still fails! Hard-freezes system References : http://lkml.org/lkml/2007/6/14/82 Submitter : Rodney Gordon II <meff(a)spherevision.org> Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

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[alsa-devel] Creative Soundblaster X-Fi
by Tim Laszlo 29 Jun '07

29 Jun '07

Has anyone worked on developing an open source driver for the sound blaster x-fi cards? I've been sitting on one for about a year (mental note check support *before* buying..) and would love to use it already... Creative has promised drivers but they seem to be in no rush to support Linux. I'd like to try my hand at working on a driver for this, but without any technical specs from creative I wouldn't really know where to begin. Anyone have any suggestions on how to start? Anyone worked on this yet? Am I nuts for even considering this?? Timl

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[alsa-devel] SALSA-Lib: Playback overrun on initial PCM start
by J. Scott Merritt 29 Jun '07

29 Jun '07

Dear List, Using SALSA-Lib 0.0.3, I call: snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0); snd_pcm_hw_params_malloc (&hw_params); snd_pcm_hw_params_any (playback_handle, hw_params); snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED); snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)); snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, 44100, 0); snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2); snd_pcm_hw_params_set_buffer_size_last (playback_handle, hw_params, &bfrsize); snd_pcm_hw_params_get_buffer_size (hw_params, &bfrsize); snd_pcm_hw_params (playback_handle, hw_params); snd_pcm_hw_params_free (hw_params); snd_pcm_sw_params_malloc (&sw_params); snd_pcm_sw_params_current (playback_handle, sw_params); snd_pcm_sw_params_set_avail_min (playback_handle, sw_params, 4096); snd_pcm_sw_params_set_start_threshold (playback_handle, sw_params, 10000U); snd_pcm_sw_params_set_xfer_align (playback_handle, sw_params, 1); snd_pcm_sw_params (playback_handle, sw_params); state = snd_pcm_state (playback_handle); frames_to_deliver = snd_pcm_avail_update (playback_handle); while ((frames_to_deliver = snd_pcm_avail_update (playback_handle)) > 4096) { snd_pcm_writei (playback_handle, &buf, 4096) } As soon as the buffer start threshold is reached, or alternatively if I manually start PCM stream, the PCM stream reports an overrun (i.e. "Broken Pipe"). If I then examine the PCM state, it is in the overrun (XRUN) state. The code is being cross-compiled and tested on an ARM processor (PXA270) which is running linux kernel 2.6.21. This same code sequence -does- operate properly (in the same ARM test environment) when compiled against AlsaLib 1.0.13. Another apparent (and perhaps relevent) difference between AlsaLib and SALSA-Lib is that if I open the PCM stream with O_NONBLOCK and then call snd_pcm_poll_descriptors, AlsaLib gives me a FD = 4, whereas SALSA-Lib reports an FD = 3; Thanks, Scott.

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Re: [alsa-devel] MOTU development
by Jonathan Woithe 29 Jun '07

29 Jun '07

Hi Per > I'd like to know more about the health on the connections to MOTU, if > they give any information about their products and how the development > has proceeded over the years. It kind of depends which MOTU devices you're asking about. There is, for example, an alsa driver for the MOTU Midi Timepiece AV (MTPAV) which presumedly makes the midi side of this available via the ALSA sequencer system. I can't speak for any of MOTU's PCI/PCIe cards. However, in connection with their more recent outboard interfaces MOTU have been, to put it mildly, extremely unhelpful. Some on the net go so far as to say they are hostile to Linux. The facts are that despite multiple people requesting information from MOTU on how to drive their firewire interfaces MOTU have responded every time with a very terse non-negotiable "no, never". I don't know if anyone's ever asked about their USB-equipped devices but I can't imagine the answer would be any different. Despite this, some progress is being made in supporting MOTU firewire devices under linux. For various reasons this support is not through ALSA but instead as part of the FFADO project. FFADO provides a userspace library for driving firewire audio devices in general and a JACK backend allowing such devices to be used by any jack-enabled software. An ALSA backend is planned but no work has been done towards this yet. Check out www.ffado.org for information on this project. Device support in FFADO varies. Some interfaces for which we have vendor support are doing very well, while supporting an interface such as the MOTUs without any documentation is challenging. However, the MOTU control protocol is pretty well understood at this point in time and it is possible to actually use the MOTU firewire devices for real work under Linux (as I have been for about 9 months now). The trap is that one needs to use an earlier revision of ffado at present. Back in February some key FFADO infrastructure changed. From before that until the last week or so I've been busy with production projects and so haven't had a chance to update the MOTU FFADO driver and restore its functionality. I'm back on the case now though and hope to make progress in the next month or so. If you want further details about all this please contact me off-list and I'll fill you in. Best regards jonathan

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[alsa-devel] ALSA Application Programming Questions
by Paul Kavan 28 Jun '07

28 Jun '07

Hello all: I have been working on getting ALSA to work on my at91sam9260 with a simple voice codec for a few weeks. For the most part, I have my ARM board producing nice audio with aplay/arecord. Now I need a custom application that will allow me to create an 'intercom' like device with my ARM board. I am planning on writing the application as I do not think there is something out there that will accomplish what I want. I am really new to ALSA--as some of you know--and, as such, have never written an application for ALSA. I have found the ALSA driver api page on the alsa site. Wondering if there is any other good documentation on writing alsa applications? Any advice is greatly appreciated as well. One of my concerns is building the applications against the libs. I have the libs on my target machine. As my host machine has a version of my rfs on it, the libs are there as well. Just wondering how to accomplish cross-compiling new apps. I suppose I could copy the libs into the directory with my cross-compiling tools. Again, any advice would be great. Thanks. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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[alsa-devel] sound on SMDK2410 working
by Jose Henrique Spahn Torres 28 Jun '07

28 Jun '07

Hi all, I was able to get sound on smdk2410 to work. I can play .wav files with the following asound.conf file: /etc $ more asound.conf pcm.!default { type hw card 0 } ctl.!default { type hw card 0 } However the sound is distorted, in the sense that it sounds as played with a lower rate. I tried force a rate with -r 44100 or other rates, with no success. Can anyone point me to the right direction? Thanks, Henrique

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[alsa-devel] SALSA-Lib: snd_pcm_hw_params_any clobbers memory
by J. Scott Merritt 28 Jun '07

28 Jun '07

Dear list, Using SALSA-Lib 0.0.2, I call snd_pcm_open snd_pcm_hw_params_malloc snd_pcm_hw_params_any The call to snd_pcm_hw_params_any appears to clobber memory beyond the end of the hw_params allocated by snd_pcm_hw_params_malloc (as reported by linking in -lmcheck). The code is being cross-compiled and tested on an ARM processor (PXA270) which is running linux kernel 2.6.21. This same code sequence -does- operate properly (in the same ARM test environment) when compiled against AlsaLib 1.0.13. I got "lost" somewhere around the hw_refine IOCTL .... Any suggestions appreciated :) Thanks, Scott.

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[alsa-devel] MOTU development
by Per Eriksson 28 Jun '07

28 Jun '07

Dear list, My name is Per Eriksson, working as a musician with music and interactive content in Sweden. I'd like to know more about the health on the connections to MOTU, if they give any information about their products and how the development has proceeded over the years. I am thankful for any input. Best Regards Per Eriksson per(a)erinet.se

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#430833: libasound2: PCM plugin `lfloat' not built
by Elimar Riesebieter 28 Jun '07

28 Jun '07

forwarded 430833 alsa-devel(a)alsa-project.org -- On Wed, 27 Jun 2007 the mental interface of Mark Wooding told: > Package: libasound2 > Version: 1.0.14a-1 > Severity: normal > Tags: patch > > The PCM plugin `lfloat' isn't compiled into libasound. In particular, > this prevents the `plug' plugin from doing linear<->float conversion, > and in turn this means that hardly any ALSA clients can use the JACK > plugin from libasound2-plugins, because JACK uses FLOAT_LE format > throughout. > > The reason for this is quite simple: the configure script is broken! In > particular, the sense of the `softfloat' check used to decide whether to > disable the plugin is wrong! (That's aside from the question of whether > it's sensible to fail to build the thing just because it might be slow.) > We should solve this in general. It's not a Debian specific wish so forwarded upstream. > > Index: alsa-lib-1.0.14a/configure.in > =================================================================== > --- alsa-lib-1.0.14a.orig/configure.in 2007-06-27 16:24:58.000000000 > +0100 > +++ alsa-lib-1.0.14a/configure.in 2007-06-27 16:25:19.000000000 > +0100 > @@ -416,7 +416,7 @@ > build_pcm_share="no" > fi > > -if test "$softfloat" != "yes"; then > +if test "$softfloat" = "yes"; then > build_pcm_lfloat="no" > fi > > Index: alsa-lib-1.0.14a/configure > =================================================================== > --- alsa-lib-1.0.14a.orig/configure 2007-06-27 16:25:23.000000000 > +0100 > +++ alsa-lib-1.0.14a/configure 2007-06-27 16:25:38.000000000 +0100 > @@ -20946,7 +20946,7 @@ > build_pcm_share="no" > fi > > -if test "$softfloat" != "yes"; then > +if test "$softfloat" = "yes"; then > build_pcm_lfloat="no" > fi > > Elimar -- Experience is something you don't get until just after you need it!

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[alsa-devel] Sound disappears after a long run, pcm_rate plugin maybe
by Dave Airlie 28 Jun '07

28 Jun '07

Hi, I've been debugging this bug which I first noticed close to a year ago but worked around by disabling dmix. Now I've had to enable dmix and try and fix it. My system is SDL on alsa-lib-1.0.13 on 2.6.21 on intel 810 with a Realtek ALC655. I'm using SDL to open a 22050 16-bit stereo stream, and dmix is set to 48000. When I run sound in my single app (only one app running for testing purposes) around 20-24 hours later (I haven't got exact timings as I can't sit in front of the pc for so long) the sound stop s happening just in that app. I can run aplay separately and it plays fine. If I stop my app (ctrl-C and cont in gdb) or attach/detach strace, I get an XRUN and my SDL code calls snd_pcm_prepare and the sounds comes back. Now I can't tell you how much it sucks to only get a single debugging run every 24 hours for 1 break point and if gdb stuffs it up you lose big time. I think I've tracked it down to pcm_rate.c as in if I open the sound card at 48000 instead of 22050 it doesn't happen and I've added a lot of printfs to the rate code in 1.0.13 and gotten huge log file traces of it.. In a normal rate trace when things are working I get traces like this: snd_pcm_rate_sync_playback_area: 442023720 442023720 525 986185728 snd_pcm_rate_sync_playback_area: 442023720 442023720 1549 986185728 snd_pcm_rate_sync_playback_area: 442023720 442023720 1549 986185728 snd_pcm_rate_sync_playback_area: 442024660 442023720 1549 986185728 snd_pcm_rate_commit_area: 6 940 1024 1881 snd_pcm_rate_sync_playback_area: 940 442023720 1881 snd_pcm_rate_sync_playback_area: 442024660 442024660 525 986185728 snd_pcm_rate_sync_playback_area: 442024660 442024660 1549 986185728 snd_pcm_rate_sync_playback_area: 442024660 442024660 1549 986185728 snd_pcm_rate_sync_playback_area: 442025595 442024660 1549 986185728 snd_pcm_rate_sync_playback_area: 442025600 442024660 1549 986185728 snd_pcm_rate_commit_area: 946 940 1024 1881 snd_pcm_rate_sync_playback_area: 940 442024660 1881 snd_pcm_rate_sync_playback_area: 442025600 442025600 525 986185728 snd_pcm_rate_sync_playback_area: 442025600 442025600 1549 986185728 snd_pcm_rate_sync_playback_area: 442025600 442025600 1549 986185728 snd_pcm_rate_sync_playback_area: 442026540 442025600 1549 986185728 snd_pcm_rate_commit_area: 5 940 1024 1881 snd_pcm_rate_sync_playback_area: 940 442025600 1881 snd_pcm_rate_sync_playback_area: 442026540 442026540 525 986185728 snd_pcm_rate_sync_playback_area: 442026540 442026540 1549 986185728 snd_pcm_rate_sync_playback_area: 442026540 442026540 1549 986185728 snd_pcm_rate_sync_playback_area: 442027476 442026540 1549 986185728 snd_pcm_rate_sync_playback_area: 442027480 442026540 1549 986185728 playback are debug code slave_size = snd_pcm_avail_update(slave); if (slave_size < 0) return slave_size; fprintf(stderr,"%s: %u %d %u %u\n", __func__, appl_ptr, rate->last_commit_ptr, slave_size, pcm->boundary); if (appl_ptr < rate->last_commit_ptr) xfer = appl_ptr - rate->last_commit_ptr + pcm->boundary; else xfer = appl_ptr - rate->last_commit_ptr; while (xfer >= pcm->period_size && (snd_pcm_uframes_t)slave_size >= rate->gen.slave->period_size) { err = snd_pcm_rate_commit_next_period(pcm, rate->last_commit_ptr % pcm->buffer_size); fprintf(stderr, "%s: %u %u %u\n", __func__, xfer, rate->last_commit_ptr, pcm->buffer_size); if (err == 0) commit area debug fprintf(stderr,"%s: %u %u %u %u\n", __func__, appl_offset, size, slave_size, pcm->buffer_size); areas = snd_pcm_mmap_areas(pcm); if (cont >= size) { In the not so good case (with different debug - two sites and I hadn't access to my debug code from one so I wrote it again) you get large numbers of snd_pcm_rate_sync_playback_area: 152605942 912 152385750 snd_pcm_rate_sync_playback_area: 152605942 912 152385750 snd_pcm_rate_sync_playback_area: 152606412 912 152385750 snd_pcm_rate_sync_playback_area: 152606412 912 152385750 snd_pcm_rate_sync_playback_area: 152606705 912 152385750 snd_pcm_rate_sync_playback_area: 152606882 912 152385750 snd_pcm_rate_sync_playback_area: 152606882 912 152385750 snd_pcm_rate_sync_playback_area: 152607352 912 152385750 snd_pcm_rate_sync_playback_area: 152607352 912 152385750 ...... snd_pcm_rate_sync_playback_area: 152769972 912 152385750 snd_pcm_rate_sync_playback_area: 152770381 1936 152385750 snd_pcm_rate_commit_area: 1411 572 470 snd_pcm_rate_sync_playback_area: 384631 1 1936 152385750 739770368 snd_pcm_rate_sync_playback_area: 152770442 912 152386220 .... more of these.... snd_pcm_rate_sync_playback_area: 152910502 912 152386220 snd_pcm_rate_sync_playback_area: 152910502 912 152386220 snd_pcm_rate_sync_playback_area: 152910972 912 152386220 snd_pcm_rate_sync_playback_area: 152910972 912 152386220 appl ptr is 152910972 recalc: 3022344724 1411 3072 470 1024 snd_pcm_rate_sync_playback_area: 152910972 1936 152386220 snd_pcm_rate_commit_area: 1411 1042 470 snd_pcm_rate_sync_playback_area: 524752 1 1936 152386220 739770368 snd_pcm_rate_sync_playback_area: 152910972 1936 152386690 snd_pcm_rate_commit_area: 1411 101 470 snd_pcm_rate_sync_playback_area: 524282 1 1936 152386690 739770368 snd_pcm_rate_sync_playback_area: 152910972 1936 152387160 ... this pattern repeats one commit per two playbacks until it goes back in the top... the debug code for these are recalc: fprintf(stderr,"%s: %lu %lu %lu %lu %lu\n", __func__, val, pcm->buffer_size, slave->buffer_size, pcm->period_size, slave->period_size); commit area areas = snd_pcm_mmap_areas(pcm); fprintf(stderr,"%s: %u %u %u\n", __func__, pcm->buffer_size, appl_offset, size); if (cont >= size) { playback area: fprintf(stderr,"%s: %u %d %d\n", __func__, appl_ptr, slave_size, rate->last_commit_ptr); if (appl_ptr < rate->last_commit_ptr) xfer = appl_ptr - rate->last_commit_ptr + pcm->boundary; else xfer = appl_ptr - rate->last_commit_ptr; while (xfer >= pcm->period_size && (snd_pcm_uframes_t)slave_size >= rate->gen.slave->period_size) { err = snd_pcm_rate_commit_next_period(pcm, rate->last_commit_ptr % pcm->buffer_size); fprintf(stderr,"%s: %u %d %d %u %u\n", __func__, xfer, err, slave_size, rate->last_commit_ptr, pcm->boundary); I'm hoping it is just some calculation in the rate plugin overflowing and I was blaming SDL but I've fixed up the codepaths to look like aplay and it still happened.. any hints would be appreciated on a) what that rate code is actually doing in the fail case, b) what the rate code is doing in the non-fail case :-) Dave.

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[alsa-devel] SoC multiple streams
by Steve Longerbeam 28 Jun '07

28 Jun '07

Hi all, I'm developing an SoC driver for a sound engine that supports playing up to 64 simultaneous substreams, all mixed in h/w to a stereo DAC. But it doesn't appear that the SoC framework supports multiple substreams, according to soc_new_pcm(). Am I missing something, or is this on the asoc development roadmap? Thanks, Steve

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[alsa-devel] SoC Machine Code Suggestions
by Paul Kavan 27 Jun '07

27 Jun '07

First, thanks to all for helping me get the at91 soc stuff working with my board and codec. It now plays telecom quality audio using aplay and records it as well using arecord. For those who were following my trials and did not hear what the solution was, essentially, when the at91-ssc code goes into playback mode, it only activates the TX side of the SSC (PCM) bus. Similarly, when it is in capture mode, it only activates the RX side of the bus. This certainly makes sense and works in most situations I am sure. However, I needed both the RX and TX frames operating in both playback and capture. As such, I had to modify at91-ssc slightly to accomplish this. Now that I have decent audio, I want to clean up around the edges of the audio and determine the best way to proceed with my application code. First. when the pcm interface starts/stops, I get a pop that comes from the codec analog output transitioning some. Now, there may be some hardware fixes to this, but what I would like to do it mute the amp until the last possible second before audio is played. I can accomplish this with a simple gpio call, but am not sure where best to turn the mute on and off. Any suggestions with that would be appreciated. Also, my next phase of development is to determine whether I need to write my own application to accomplish what I want to do, or if there is an application out there that will do the trick--and that I can cross-compile to an ARM. What I would like is an application I will stream audio (half-duplex) between two ARM boards connected via ethernet. I would need to be able to control the start/stop of the audio. Sometimes this control will depend on the state of the machine and other times it will be activate with a button press by the user. I hope that makes sense. Obviously it would have to seamlessly stream to the codec via ALSA. Also, there may be times when it streams to a pc. Now, the codec uses 8-bit ulaw, so this code if it were on a pc, would need to be able to play that on the pc in its native state or some type of conversion on the fly. I guess you can think of the application as similar to voip, but half duplex instead of full duplex. Does anyone know if something like this exists that I might use. If not, how hard would it be to code this for ALSA? Thanks again and I look forward to your responses. Paul -- *************************************** Paul David Kavan Project Engineer GRH Electronics, Inc. 402-734-4900 pkavan(a)gmail.com ***************************************

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Re: [alsa-devel] SALSA-Lib version 0.0.1 release
by Takashi Iwai 27 Jun '07

27 Jun '07

At Mon, 25 Jun 2007 23:32:39 +0100, Adrian McMenamin wrote: > > > You can check ldd to the app which shlib it's linked with. > > It should be linked with libsalsa, not libasound. > > > > > > Takashi > > > > I expect I am being a bit thick here but what app? I use mpg123 and > it's not linked with any alsa libs. > > / # ldd ./mpg123 > libm.so.0 => /lib/libm.so.0 (0x2956d000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x2958b000) > libc.so.0 => /lib/libc.so.0 (0x295aa000) > ld-uClibc.so.0 => /lib/ld-uClibc.so.0 (0x29558000) mpg123 is an OSS app. > / # ldd aplay > libasound.so.2 => /usr/lib/libasound.so.2 (0x2956d000) > libm.so.0 => /lib/libm.so.0 (0x29652000) > libdl.so.0 => /lib/libdl.so.0 (0x29670000) > libpthread.so.0 => /lib/libpthread.so.0 (0x29683000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x296a4000) > libc.so.0 => /lib/libc.so.0 (0x296c3000) > ld-uClibc.so.0 => /lib/ld-uClibc.so.0 (0x29558000) It's liekly linked with the old libasound. SALSA-lib is a source-level compatible library and it's _not_ binary-level compatible. Thus, it's no drop-in replacement. After installing salsa-lib (better use --enable-everything option to make a libasound symlink), then you'll have to rebuild the other apps linked with alsa-lib again. Takashi

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