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Re: [alsa-devel] [Alsa-user] Sound card recording support @ 96kHz or above
by Vedran Miletić 26 Aug '08

26 Aug '08

Terrasoniq TS22 (newer revision of TerraTec PHASE 22) might do it for you. Almost anything Envy24-series based will offer what you want. Not sure how you would check that it actually works, though. 2008/8/26 Ben Gould <ben.gould(a)gmail.com>: > These cards don't seem to be in production any more. > > Any suggestions for more recent off-the-shelf hardware that will capture > a single stereo input at 96kHz (using ALSA)? > > -- ben > > Sergei Steshenko wrote: >> >> -----Original Message----- >> From: Ben Gould <ben.gould(a)gmail.com> >> To: alsa-user(a)lists.sourceforge.net >> Date: Thu, 21 Aug 2008 16:57:43 +0059 >> Subject: [Alsa-user] Sound card recording support @ 96kHz or above >> >>> Hi all, >>> >>> A project I'm working on using linux & alsa requires a higher than >>> normal sampling rate, not just up-sampling from 48kHz. >>> >>> I wonder whether anyone here can recommend a sound card that's capable >>> of recording a single stereo input at 96kHz sample rate (16 bits would >>> be sufficient). >>> >>> Obviously it needs to be properly supported by alsa... >>> >>> Thanks in advance >>> >>> -- ben >>> >> >> M-Audio Revolution 7.1, 5.1. >> >> Regards, >> Sergei. >> >> > > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > _______________________________________________ > Alsa-user mailing list > Alsa-user(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-user > -- Vedran Miletić

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[alsa-devel] Commit 2a3fdf2e7bbeae557b4a1ee64c071ae4c3c561e5 makes unable to compile ca106 module
by Vedran Miletić 26 Aug '08

26 Aug '08

Here's a proper patch: --- a/pci/ca0106/ca0106_main.c +++ b/pci/ca0106/ca0106_main.c @@ -249,11 +249,12 @@ static struct snd_ca0106_details ca0106_chip_details[] = { .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , - /* Another MSI K8N Diamond MB, which has apprently a different SSID */ + /* MSI K8N Diamond PLUS MB */ { .serial = 0x10091102, .name = "MSI K8N Diamond MB", .gpio_type = 2, - .i2c_adc = 1 } , + .i2c_adc = 1, + .spi_dac = 2 } , /* Shuttle XPC SD31P which has an onboard Creative Labs * Sound Blaster Live! 24-bit EAX * high-definition 7.1 audio processor". The comma was missing after .spi_dac line. -- Vedran Miletić

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Re: [alsa-devel] Alsa-driver compile fail on Ubuntu powerpc
by nguyen tritai 26 Aug '08

26 Aug '08

Takashi, Now I can build and install driver for ca0106 sb Audigy but I got another issue when play an example file. >> aplay -vv child.wav Playing WAVE '/root/child.wav' : Unsigned 8 bit, Rate 22050 Hz, Mono Plug PCM: Rate conversion PCM (48000, sformat=U8) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 22050 exact rate : 22050 (22050/1) msbits : 8 buffer_size : 3763 period_size : 470 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 470 period_event : 0 start_threshold : 3763 stop_threshold : 3763 silence_threshold: 0 silence_size : 0 boundary : 493223936 Slave: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : U8 subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 8 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : NONE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 8192 stop_threshold : 8192 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Hardware PCM card 0 'CA0106' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 2 rate : 48000 exact rate : 48000 (48000/1) msbits : 32 buffer_size : 8192 period_size : 1024 period_time : 21333 tstamp_mode : ENABLE period_step : 1 avail_min : 1024 period_event : 0 start_threshold : 1 stop_threshold : 1073741824 silence_threshold: 0 silence_size : 1073741824 boundary : 1073741824 ######### + | 35%Segmentation fault Kernel dump message >> dmesg Unable to handle kernel paging request for data at address 0x00000000 Faulting instruction address: 0xc0033fd4 Oops: Kernel access of bad area, sig: 11 [#7] Canyonlands Modules linked in: snd_seq_midi snd_ca0106 snd_rawmidi snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device snd_ac97_codec snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd_page_alloc snd ac97_bus NIP: c0033fd4 LR: c004ac94 CTR: e901a6c8 REGS: dfbf3d40 TRAP: 0300 Tainted: G D (2.6.26-00015-gc0b03c1-dirty) MSR: 00029000 <EE,ME> CR: 48022444 XER: 20000000 DEAR: 00000000, ESR: 00000000 TASK = dfa57000[15245] 'aplay' THREAD: dfbf2000 GPR00: c004ac94 dfbf3df0 dfa57000 00000000 c0b2c180 00000000 3f10c149 00000000 GPR08: 00000000 00000000 9e370001 c030b10c 00000149 10025478 1000bd14 bff7ab68 GPR16: c0320000 c030ae74 c0320000 00000000 00000000 00000000 00000000 de14c340 GPR24: 4814b000 00000000 dfbf2000 de00c2c0 c0b2c180 00000000 00000000 c0b2c180 NIP [c0033fd4] __wake_up_bit+0x10/0x40 LR [c004ac94] unlock_page+0x48/0x5c Call Trace: [dfbf3e10] [c004ac94] unlock_page+0x48/0x5c [dfbf3e20] [c005a2d8] __do_fault+0x254/0x618 [dfbf3e80] [c000faf0] do_page_fault+0x2d8/0x518 [dfbf3f40] [c000d134] handle_page_fault+0xc/0x80 Instruction dump: 7d6b4a14 800b0280 812b0278 20000020 7c630430 54631838 7c634a14 4e800020 7c0802a6 9421ffe0 90810008 90010024 <80030000> 90a1000c 7f801800 419e0014 ---[ end trace a0b9d3bfbd44991b ]--- I tried with another example file.wav but the problem is the same. The issue seems from accessing memory. I'm using AMCC 460EX board with distribution Ubuntu-7.04 Feel free to help me. Thanks, TaiNguyen. I tried with another wav On Mon, Aug 18, 2008 at 9:52 AM, nguyen tritai <nguyentritai(a)gmail.com>wrote: > Hi Takashi, > > Thank for your help. I checked in the config.log as you told me and see > that the issue is from the linux-header . > I'm now working on it. > > Thanks, > > TaiNguyen > > > On Thu, Aug 14, 2008 at 8:49 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> At Thu, 14 Aug 2008 16:25:24 +0700, >> nguyen tritai wrote: >> > >> > Hi all, >> > >> > I'm trying to bring up audio on powerpc AMCC 440EPX board with the >> soundcard >> > Sound Blaster Audigy CA0106. >> > The board is runing Ubuntu 7.04. >> > I configured: ./configure --with-cards=ca0106 --with-sequencer=yes >> > --with-kernel=/usr/src/linux >> > Then I make and get issue: >> > >> > make[2]: Leaving directory `/root/alsa-driver-1.0.17/usb' >> > make[2]: Entering directory `/root/alsa-driver-1.0.17/misc' >> > make[2]: Leaving directory `/root/alsa-driver-1.0.17/misc' >> > make[1]: Leaving directory `/root/alsa-driver-1.0.17' >> > make -C /usr/src/linux SUBDIRS=/root/alsa-driver-1.0.17 CPP="gcc -E" >> > CC="gcc" modules >> > make[1]: Entering directory `/usr/src/linux' >> > CC [M] /root/alsa-driver-1.0.17/acore/memory_wrapper.o >> > make[1]: Leaving directory `/usr/src/linux' >> > root@sequoia:~/alsa-driver-1.0.17# vim log >> > In file included from >> /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: >> > /root/alsa-driver-1.0.17/include/adriver.h:278:1: warning: "GFP_DMA32" >> > redefined >> > In file included from include/linux/slab.h:14, >> > from include/linux/percpu.h:5, >> > from include/linux/rcupdate.h:41, >> > from include/linux/pid.h:4, >> > from include/linux/sched.h:76, >> > from /root/alsa-driver-1.0.17/include/asm/elf.h:5, >> > from include/linux/elf.h:7, >> > from include/linux/module.h:14, >> > from /root/alsa-driver-1.0.17/include/adriver.h:26, >> > from >> /root/alsa-driver-1.0.17/acore/memory_wrapper.c:25: >> > include/linux/gfp.h:105:1: warning: this is the location of the previous >> > definition >> >> This means that configure script didn't detect the things correctly. >> Doesn't the snapshot version work, too? >> If no, check config.log about the relevant part checking DMA32. >> >> >> Takashi >> > >

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Re: [alsa-devel] Online code snapshots on ALSA server
by Jaroslav Kysela 26 Aug '08

26 Aug '08

On Mon, 25 Aug 2008, Sean McNamara wrote: > Glad to hear it, Jaroslav! That's much easier than checking out from GIT. > > If these are automatically updated, any idea how frequently the > packages will be regenerated? Is it after every commit, or only at > particular time intervals? You can grab the latest code immediately after commit to GIT repository (but generation of new package takes awhile as mentioned on page when it's not built on server already). The package generation is triggered from the web server by users requesting latest package immediately. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] Compilation of Alsa 1.0.17 on Suse10 crashes
by Takashi Iwai 26 Aug '08

26 Aug '08

At Fri, 22 Aug 2008 19:08:29 +0200 (CEST), Vaclav Peroutka wrote: > > Hello Mr. Takashi, > > I tried to compile alsa-driver-snapshot from the page you advised me. But the problem was the same. I am sending you configure.log and make.log files. Hopefully they will clarify the problem. > > I still have problem with mts64.c. OK, I see the problem, too. As a temporary workaround, you can change alsa-driver/kconfig-vers. This is a list of the least kernel version for some drivers, and mts64 is there, too. Change the version to 2.6.26 or such, so that it won't be built for your kernel. Meanwhile, I'll try to fix the problem in the upstream. thanks, Takashi > > Can you help me please ? > > Thank you in advance, > Vaclav > > > ------------ Původní zpráva ------------ > > Od: Takashi Iwai <tiwai(a)suse.de> > > Předmět: Re: [alsa-devel] Compilation of Alsa 1.0.17 on Suse10 crashes > > Datum: 22.8.2008 08:27:13 > > ---------------------------------------- > > At Thu, 21 Aug 2008 23:59:14 +0200 (CEST), > > Vaclav Peroutka wrote: > > > > > > Hello all, > > > > > > I downloaded alsa-driver-1.0.17 and tried to compile it. I started with > > "./configure --with-suse=auto --with-oss=yes --with-cards=all,mts64=n" and then > > "make". > > > > > > But I always get following: > > > ... > > > CC [M] /tmp/alsa-driver-1.0.17/drivers/mts64.o > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c: In function ‘snd_mts64_attach’: > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:896: error: implicit declaration of > > function ‘platform_device_alloc’ > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:896: warning: assignment makes pointer > > from integer without a cast > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:903: error: implicit declaration of > > function ‘platform_device_add’ > > > /tmp/alsa-driver-1.0.17/drivers/mts64.c:904: error: implicit declaration of > > function ‘platform_device_put’ > > > make[4]: *** [/tmp/alsa-driver-1.0.17/drivers/mts64.o] Error 1 > > > make[3]: *** [/tmp/alsa-driver-1.0.17/drivers] Error 2 > > > make[2]: *** [_module_/tmp/alsa-driver-1.0.17] Error 2 > > > make[1]: *** [modules] Error 2 > > > make[1]: Leaving directory `/usr/src/linux-2.6.13-15' > > > > > > My GCC: > > > > gcc --version > > > gcc (GCC) 4.0.2 20050901 (prerelease) (SUSE Linux) > > > > > > Is there something missing ? What should I do now ? > > > > Use the later version, at best, the daily snapshot tarball > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > > > Or, you can try alsa-driver-kmp on standard SUSE systems. > > http://download.opensuse.org/repositories/multimedia:/audio:/KMP/ > > > > Check openSUSE Wiki SDB:AudioTroubleshooting for details. > > > > > > 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] [ALSA PATCH] alsa-git merge request
by Takashi Iwai 26 Aug '08

26 Aug '08

At Tue, 26 Aug 2008 00:43:59 +0200, Guillaume Chazarain wrote: > > On Mon, Aug 25, 2008 at 11:36 PM, Guillaume Chazarain <guichaz(a)gmail.com> wrote: > > On Mon, Aug 25, 2008 at 7:50 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> I guess it's fixed as well by passing model=auto? > > > > Indeed that works too. > > I spoke too soon, with model=auto I get back the "Front" mixer that I > used during the git bisection to find the culprit, but I get no sound > at all in the plugged in headphones or the internal speakers. This is odd. Are you sure both are the same kernel but just one commit reverted? Takashi

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[alsa-devel] The state of Linux audio
by Fred . 26 Aug '08

26 Aug '08

What is the state of audio in Linux? How does it compare against other operating systems? Is the Linux audio functionality good or bad? What could be better? What is planned for the future?

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[alsa-devel] Sound Blaster Audigy (snd_ca0106 Creative Labs|SB Audigy LS): How to use the mixer?
by Colin Guthrie 25 Aug '08

25 Aug '08

Hi, Just trying to help a user with a problem and it seems to be down to the fact that his SB Audigy LS does not have a "Master" or "PCM" mixer control. I asked Lennart about this and he told that Takashi said he wanted to make sure that all devices have at least a Master vol control. This one doesn't! Please let me know what you need in order to fix this. This is with 1.0.17... I can't see anything in the changelog that would suggest this is fixed in the 1.0.18rc1. Please let me know if you need e.g. PCI ids etc. as I can ask the user to provide these. Col -- Colin Guthrie gmane(at)colin.guthr.ie http://colin.guthr.ie/ Day Job: Tribalogic Limited [http://www.tribalogic.net/] Open Source: Mandriva Linux Contributor [http://www.mandriva.com/] PulseAudio Hacker [http://www.pulseaudio.org/] Trac Hacker [http://trac.edgewall.org/] Simple mixer control 'Line in',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 255 [100%] [24.00dB] Front Right: Capture 255 [100%] [24.00dB] Simple mixer control 'Mic',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 255 [100%] [24.00dB] Front Right: Capture 255 [100%] [24.00dB] Simple mixer control 'Phone',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 207 [81%] [0.00dB] Front Right: Capture 207 [81%] [0.00dB] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Center/LFE',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'IEC958 Front',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'IEC958 Rear',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'IEC958 Unknown',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'Aux',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 255 Front Left: Capture 207 [81%] [0.00dB] Front Right: Capture 207 [81%] [0.00dB] Simple mixer control 'Analog Center/LFE',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Front',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Rear',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Side',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] [on] Front Right: Playback 255 [100%] [12.00dB] [on] Simple mixer control 'Analog Source',0 Capabilities: cenum Items: 'Phone' 'Mic' 'Line in' 'Aux' Item0: 'Line in' Simple mixer control 'CAPTURE feedback',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [12.00dB] Front Right: Playback 255 [100%] [12.00dB] Simple mixer control 'Digital Source',0 Capabilities: cenum Items: 'IEC958 out' 'i2s mixer out' 'IEC958 in' 'i2s in' 'AC97 in' 'SRC out' Item0: 'i2s in' Simple mixer control 'Shared Mic/Line in',0 Capabilities: cenum Items: 'Line in' 'Mic in' Item0: 'Line in'

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[alsa-devel] Online code snapshots on ALSA server
by Jaroslav Kysela 25 Aug '08

25 Aug '08

Hi all, finally, I made possible to download a very fresh ALSA code in tarballs from our repository - http://www.alsa-project.org/snapshot/ . Users can get the recent code immediately now. There is no reason to wait a day or so after commit to the GIT repository now. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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[alsa-devel] PATCH - via82xx vt1618 7.1 Audio Support
by John L. Utz III 25 Aug '08

25 Aug '08

Makes Audio > 2.0 work on the VIA vt1618 7.1 Channel AC97 Codec Patched against ALSA-GIT Fri Aug 22 16:24:52 PDT 2008 Authored-by: John L. Utz III john.utz(a)dmx.com Signed-off-by: John L. Utz III john.utz(a)dmx.com diff --git a/pci/ac97/ac97_codec.c b/pci/ac97/ac97_codec.c index d0023e9..89f3050 100644 --- a/pci/ac97/ac97_codec.c +++ b/pci/ac97/ac97_codec.c @@ -168,7 +168,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x54584e20, 0xffffffff, "TLC320AD9xC", NULL, NULL }, { 0x56494161, 0xffffffff, "VIA1612A", NULL, NULL }, // modified ICE1232 with S/PDIF { 0x56494170, 0xffffffff, "VIA1617A", patch_vt1617a, NULL }, // modified VT1616 with S/PDIF -{ 0x56494182, 0xffffffff, "VIA1618", NULL, NULL }, +{ 0x56494182, 0xffffffff, "VIA1618", patch_vt1618, NULL }, { 0x57454301, 0xffffffff, "W83971D", NULL, NULL }, { 0x574d4c00, 0xffffffff, "WM9701,WM9701A", NULL, NULL }, { 0x574d4C03, 0xffffffff, "WM9703,WM9707,WM9708,WM9717", patch_wolfson03, NULL}, @@ -609,7 +609,6 @@ AC97_SINGLE("PC Speaker Playback Volume", AC97_PC_BEEP, 1, 15, 1) static const struct snd_kcontrol_new snd_ac97_controls_mic_boost = AC97_SINGLE("Mic Boost (+20dB)", AC97_MIC, 6, 1, 0); - static const char* std_rec_sel[] = {"Mic", "CD", "Video", "Aux", "Line", "Mix", "Mix Mono", "Phone"}; static const char* std_3d_path[] = {"pre 3D", "post 3D"}; static const char* std_mix[] = {"Mix", "Mic"}; diff --git a/pci/ac97/ac97_patch.c b/pci/ac97/ac97_patch.c index bb028f8..24f1e98 100644 --- a/pci/ac97/ac97_patch.c +++ b/pci/ac97/ac97_patch.c @@ -3465,7 +3465,7 @@ static int patch_vt1616(struct snd_ac97 * ac97) /* * unfortunately, the vt1617a stashes the twiddlers required for - * nooding the i/o jacks on 2 different regs. * thameans that we cant + * noodling the i/o jacks on 2 different regs. that means that we cant * use the easy way provided by AC97_ENUM_DOUBLE() we have to write * are own funcs. * @@ -3498,7 +3498,7 @@ static int snd_ac97_vt1617a_smart51_get(struct snd_kcontrol *kcontrol, pac97 = snd_kcontrol_chip(kcontrol); /* grab codec handle */ - /* grab our desirec bits, then mash them together in a manner + /* grab our desired bits, then mash them together in a manner * consistent with Table 6 on page 17 in the 1617a docs */ usSM51 = snd_ac97_read(pac97, 0x7a) >> 14; @@ -3576,6 +3576,200 @@ int patch_vt1617a(struct snd_ac97 * ac97) return err; } +/* VIA VT1618 8 CHANNEL AC97 CODEC */ + +/* config aux in jack - not found on 3 jack motherboards or soundcards */ + +static int snd_ac97_vt1618_aux_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char *acTxtAux[] = {"Aux In", "Back Surr Out"}; + + return ac97_enum_text_info(kcontrol, uinfo, acTxtAux, 2); +} + +static int snd_ac97_vt1618_aux_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.enumerated.item[0] = + (snd_ac97_read(snd_kcontrol_chip(kcontrol), 0x5c) & 0x0008)>>3; + return 0; +} + +static int snd_ac97_vt1618_aux_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* toggle surround rear dac power */ + + snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x5c, 0x0008, + ucontrol->value.enumerated.item[0] << 3); + + /* toggle aux in surround rear out jack */ + + return snd_ac97_update_bits(snd_kcontrol_chip(kcontrol), 0x76, 0x0008, + ucontrol->value.enumerated.item[0] << 3); +} + + +/* + * VIA implements 'Smart 5.1' completely differently on the 1618 than + * it does on the 1617a. awesome! They seem to have sourced this + * particular revision of the technology from somebody else, it's + * called Universal Audio Jack and it shows up on some other folk's chips + * as well. + * + * ordering in this list reflects vt1618 docs for Reg 60h and + * the block diagram, DACs are as follows: + * + * OUT_O -> Front, + * OUT_1 -> Surround, + * OUT_2 -> C/LFE + * + * Unlike the 1617a, each OUT has a consistent set of mappings + * for all bitpatterns other than 00: + * + * 01 Unmixed Output + * 10 Line In + * 11 Mic In + * + * Special Case of 00: + * + * OUT_0 Mixed Output + * OUT_1 Reserved + * OUT_2 Reserved + * + * I have no idea what the hell Reserved does, but on an MSI + * CN700T, i have to set it to get surround output - YMMV, bad + * shit may happen. + * + * If other chips use Universal Audio Jack, then this code might be + * applicable to them. + */ + +static int snd_ac97_vt1618_UAJ_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + /* cribbed from ac97_surround_jack_mode_info() ordering in these lists + * reflects vt1618 docs for Vendor Defined Register 0x60 */ + + static const char *acTxtUAJ0[] = + { "Speaker Out", "DAC Unmixed Out", "Line In", "Mic In"}; + + static const char *acTxtUAJ1[] = + { "Surround Out", "DAC Unmixed Out", "Line In", "Mic In"}; + + static const char *acTxtUAJ2[] = + { "Center LFE Out", "DAC Unmixed Out", "Line In", "Mic In"}; + + const char **ppacTxt = acTxtUAJ0; + + if (0x000C == kcontrol->private_value) + ppacTxt = acTxtUAJ1; + + if (0x0030 == kcontrol->private_value) + ppacTxt = acTxtUAJ2; + + return ac97_enum_text_info(kcontrol, uinfo, ppacTxt, 4); +} + +/* All of the vt1618 Universal Audio Jack twiddlers are on + Vendor Defined Register 0x60, page 0. The bits, and thus + the mask, are the only thing that changes */ + +static int snd_ac97_vt1618_UAJ_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned short usDatPag, usUAJ, usShift = 0; + struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); + + if (0x000C == kcontrol->private_value) + usShift = 2; + + if (0x0030 == kcontrol->private_value) + usShift = 4; + + mutex_lock(&pac97->page_mutex); + + usDatPag = snd_ac97_read(pac97, AC97_INT_PAGING) & AC97_PAGE_MASK; + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, 0); + + usUAJ = snd_ac97_read(pac97, 0x60) & kcontrol->private_value; + + snd_ac97_update_bits(pac97, AC97_INT_PAGING, AC97_PAGE_MASK, usDatPag); + mutex_unlock(&pac97->page_mutex); + + ucontrol->value.enumerated.item[0] = usUAJ >> usShift; + + return 0; +} + +static int snd_ac97_vt1618_UAJ_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned short usSft = 0; + + if (0x000C == kcontrol->private_value) + usSft = 2; + + if (0x0030 == kcontrol->private_value) + usSft = 4; + + /* UAJ1 and UAJ2 are not supposed to have 00 written to them?? i + * dunno, because thats something that i have to do to get 5.1 out to + * work. */ + + return ac97_update_bits_page(snd_kcontrol_chip(kcontrol), + 0x60, kcontrol->private_value, + ucontrol->value.enumerated.item[0]<<usSft, + 0); +} + +static const struct snd_kcontrol_new snd_ac97_controls_vt1618[] = { + AC97_SINGLE("Exchange Center/LFE", 0x5a, 8, 1, 0), + AC97_SINGLE("DC Offset", 0x5a, 10, 1, 0), + AC97_SINGLE("Soft Mute", 0x5c, 0, 1, 1), + AC97_SINGLE("Headphone Amp", 0x5c, 5, 1, 1), + AC97_DOUBLE("Back Surr Volume", 0x5e, 8, 0, 31, 1), + AC97_SINGLE("Back Surr Switch", 0x5e, 15, 1, 1), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Aux Jack", + .info = snd_ac97_vt1618_aux_info, + .get = snd_ac97_vt1618_aux_get, + .put = snd_ac97_vt1618_aux_put, + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Speaker Jack", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0x0003 + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Jack", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0x000C + }, + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic Jack", + .info = snd_ac97_vt1618_UAJ_info, + .get = snd_ac97_vt1618_UAJ_get, + .put = snd_ac97_vt1618_UAJ_put, + .private_value = 0x0030 + }, +}; + +int patch_vt1618(struct snd_ac97 *ac97) +{ + return patch_build_controls(ac97, snd_ac97_controls_vt1618, + ARRAY_SIZE(snd_ac97_controls_vt1618)); +} + /* */ static void it2646_update_jacks(struct snd_ac97 *ac97)

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