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[alsa-devel] Multichannel capture support in E-MU 1212
by dustin＠seznam.cz 15 May '07

15 May '07

Hello, I am working together with Ctirad on supporting more capture channels in E-MU1212. So far we have managed to increase the number of channels from the default of 8 to 16 (A_FXWC2 = 0xffffffff, added new snd_emu10k1_audigy_dsp_convert_32_to_2x16 calls in emufx.c), there is still a number of issues to solve. I would like to get your opinion on the form of adding this support to the module. The options are either hard-coding the 16 capture channels into the code, assuming that users would prefer as many channels as possible and do the routing in jack, or introducing a module parameter emu1010-capture-channels (range e.g. 1 to 16 - untested yet) and leaving the default number at the existing 8. Furthermore, the PCMs other than efx seem to be of no use for the EMU-1x1x models, probably fairly confusing for new users. Would you find it OK, if they were removed for this specific card (if (emu->card_capabilities->emu1010)), keeping only the playback and capture PCM id 2? Thanks a lot for your suggestions. Based on them, I will prepare the patch (with comments on what we found out about the card - it is a pitty that the driver has almost no big-picture comments on the supported cards' specifics, making new developers spend a number of long nights going through the nontrivial code and guessing). Pavel.

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[alsa-devel] intel8x0 / ALC655 no-capture solved
by Vegard Nossum 15 May '07

15 May '07

Hello, I've successfully debugged a problem I've had with being unable to record anything from microphones or line-in. Detailed information about the bug along with a temporary work-around can be found here: http://bugtrack.alsa-project.org/alsa-bug/view.php?id=3066 Now I'd just like to inform about the presence of the bug (at least I'm pretty sure it's a bug), and that it's solved by the means of a work-around. I'd be very happy to see a fix in the *code* for this, however. I'd even like to try it myself (heck, it's just a register write), but I'm not certain WHERE/WHEN to make the write, under which conditions, and for which range of cards (ie. is it specific to the ALC655 or ALC*, or perhaps Realtek chips, or should the fix be made for the Intel driver, or all AC'97 cards?). Guidance would be appreciated :) Maybe it's even as easy as a card config option, but I wouldn't know. Thanks. Kinds regards, Vegard Nossum PS: I'm not on the list, so please CC any replies.

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Re: [alsa-devel] [PATCH] Coping with short replies in usbmixer
by Thomas Reitmayr 15 May '07

15 May '07

Hi, here is it again and signed off. This patch makes sure that short USB replies are treated as an error when requesting the value of a certain mixer control. Signed-off-by: Thomas Reitmayr <thomas(a)devbase.at> --- sound/usb/usbmixer.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff -rup a/sound/usb/usbmixer.c b/sound/usb/usbmixer.c --- a/sound/usb/usbmixer.c 2007-05-11 21:58:33.000000000 +0200 +++ b/sound/usb/usbmixer.c 2007-05-11 21:59:14.000000000 +0200 @@ -360,7 +360,7 @@ static int get_ctl_value(struct usb_mixe request, USB_RECIP_INTERFACE | USB_TYPE_CLASS | USB_DIR_IN, validx, cval->mixer->ctrlif | (cval->id << 8), - buf, val_len, 100) >= 0) { + buf, val_len, 100) >= val_len) { *value_ret = convert_signed_value(cval, snd_usb_combine_bytes(buf, val_len)); return 0; }

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[alsa-devel] [PATCH] Include quirks from Ubuntu Dapper/Edgy/Feisty
by crimsun＠fungus.sh.nu 15 May '07

15 May '07

# HG changeset patch # User crimsun(a)Box.ncat.edu # Date 1179219408 14400 # Node ID f38d824e5df6fb7d9677cce13f91ad868e617fe4 # Parent 38fe50050f000c3a9660134a86f713dc23f7ec79 [PATCH] Include quirks from Ubuntu Dapper/Edgy/Feisty Included is a patch against current alsa-kernel hg tip adding quirks that Ubuntu Dapper/Edgy/Feisty contains. Signed-off-by: Daniel T Chen <crimsun(a)ubuntu.com> diff -r 38fe50050f00 -r f38d824e5df6 pci/ac97/ac97_patch.c --- a/pci/ac97/ac97_patch.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/ac97/ac97_patch.c Tue May 15 04:56:48 2007 -0400 @@ -1782,6 +1782,11 @@ static unsigned int ad1981_jacks_blackli 0x10140534, /* Thinkpad X31 */ 0x10140537, /* Thinkpad T41p */ 0x10140554, /* Thinkpad T42p/R50p */ + 0x10140567, /* Thinkpad T43p 2668-G7U */ + 0x10140581, /* Thinkpad X41-2527 */ + 0x104380b0, /* Asus A7V8X-MX */ + 0x11790241, /* Toshiba Satellite A-15 S127 */ + 0x144dc01a, /* Samsung NP-X20C004/SEG */ 0 /* end */ }; diff -r 38fe50050f00 -r f38d824e5df6 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Tue May 15 04:56:48 2007 -0400 @@ -467,6 +467,8 @@ static struct snd_pci_quirk stac9200_cfg "Dell XPS M1710", STAC_REF), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01cf, "Dell Precision M90", STAC_REF), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01d6, + "unknown Dell", STAC_REF), {} /* terminator */ }; diff -r 38fe50050f00 -r f38d824e5df6 pci/intel8x0.c --- a/pci/intel8x0.c Mon May 14 18:40:07 2007 +0200 +++ b/pci/intel8x0.c Tue May 15 04:56:48 2007 -0400 @@ -1799,6 +1799,18 @@ static struct ac97_quirk ac97_quirks[] _ }, { .subvendor = 0x1028, + .subdevice = 0x0186, + .name = "Dell Latitude D810", /* cf. Malone #41015 */ + .type = AC97_TUNE_HP_MUTE_LED + }, + { + .subvendor = 0x1028, + .subdevice = 0x0188, + .name = "Dell Inspiron 6000", + .type = AC97_TUNE_HP_MUTE_LED /* cf. Malone #41015 */ + }, + { + .subvendor = 0x1028, .subdevice = 0x0191, .name = "Dell Inspiron 8600", .type = AC97_TUNE_HP_ONLY @@ -1819,7 +1831,7 @@ static struct ac97_quirk ac97_quirks[] _ .subvendor = 0x103c, .subdevice = 0x088c, .name = "HP nc8000", - .type = AC97_TUNE_MUTE_LED + .type = AC97_TUNE_HP_MUTE_LED }, { .subvendor = 0x103c, @@ -1909,6 +1921,12 @@ static struct ac97_quirk ac97_quirks[] _ .subvendor = 0x10cf, .subdevice = 0x1253, .name = "Fujitsu S6210", /* STAC9750/51 */ + .type = AC97_TUNE_HP_ONLY + }, + { + .subvendor = 0x10cf, + .subdevice = 0x127e, + .name = "Fujitsu Lifebook C1211D", .type = AC97_TUNE_HP_ONLY }, { -- Daniel T. Chen crimsun(a)ubuntu.com GPG key: 0xC88ABDA3

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[alsa-devel] soft vol
by Michel Benoit 15 May '07

15 May '07

Hello Now that I have my pcm playback working with aplay with our at91 aSoc sound driver I would like to control the volume with softvol. I followed the instructions from http://alsa.opensrc.org/How_to_use_softvol_to_control_the_master_volume and I have the following asound.con file in /etc pcm.!default { type plug slave.pcm "softvol" } pcm.softvol { type softvol slave { pcm "plughw:0,0" } control { name "PCM volume" card 0 } } When I play a wav file directly to the device it works fine: # aplay -Dplughw:0,0 Front_Center.wav Playing WAVE 'Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono but when I try to use the default softvol device I get the following output (and no sound): # aplay Front_Center.wav Playing WAVE 'Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ALSA sound/core/control.c:1244: unknown ioctl = 0xc2c45512 ... Does anyone have any ideas as to why this is happening? Michel

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[alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric 15 May '07

15 May '07

I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio chipset eric@gondor:~$ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) When modprobing the modules I get : eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout while /var/log/messages says : May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: azx_get_response timeout, switching to polling mode... Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the results of aplay : eric@gondor:~$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:~$ Here are some other infos : eric@gondor:~$ tail -2 /proc/asound/oss/sndstat 0: Realtek ALC880 1: SAA7134 Mixer amixer shows this : eric@gondor:~$ amixer Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 228 [89%] Front Right: Playback 228 [89%] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 55 [43%] [-2304.00dB] [on] Front Right: Playback 55 [43%] [-2304.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 64 [50%] [-2016.00dB] [on] Front Right: Playback 64 [50%] [-2016.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 56 [44%] [-2272.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 58 [46%] [-2208.00dB] [on] Front Right: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 57 [45%] [-2240.00dB] [on] Front Right: Playback 57 [45%] [-2240.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 53 [42%] [-2368.00dB] [on] Front Right: Playback 53 [42%] [-2368.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 31 [24%] [-3072.00dB] [on] Front Right: Capture 31 [24%] [-3072.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 27 [21%] [-3200.00dB] [on] Front Right: Capture 27 [21%] [-3200.00dB] [on] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 24 [19%] [-3296.00dB] [on] Front Right: Capture 24 [19%] [-3296.00dB] [on] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '6ch' Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] aplay -l seems to work : eric@gondor:~$ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 Analog] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 eric@gondor:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xb0000000 irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xb3006000 irq 22 if I try to do a cat /proc/asound/card0/codec#0 it hangs while /var/log/messages gives a lot of May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid dep_range_val 0:7fff May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid dep_range_val 0:7fff ........................ eric@gondor:~$ sudo lspci -nv 00:00.0 0600: 8086:2580 (rev 04) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: b1000000-b2ffffff Prefetchable memory behind bridge: c0000000-cfffffff Capabilities: [88] Subsystem: 8086:2582 Capabilities: [80] Power Management version 2 Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [a0] Express Root Port (Slot+) IRQ 0 00:1b.0 0403: 8086:2668 (rev 03) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 20 Memory at b0000000 (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 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1800 [size=32] 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 3000 [size=32] 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 3020 [size=32] 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 3040 [size=32] 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 Memory at b0004000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 I/O behind bridge: 00004000-00004fff Memory behind bridge: b3000000-b30fffff Prefetchable memory behind bridge: 0000000050000000-0000000055ffffff Capabilities: [50] Subsystem: 1558:0900 00:1f.0 0601: 8086:2640 (rev 03) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 3080 [size=16] 00:1f.3 0c05: 8086:266a (rev 03) Subsystem: 1558:0900 Flags: medium devsel, IRQ 10 I/O ports at 3060 [size=32] 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at b2000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at b1000000 (64-bit, non-prefetchable) [size=16M] Capabilities: [60] Power Management version 2 Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [78] Express Endpoint IRQ 0 0a:00.0 0607: 104c:ac50 (rev 02) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at b3007000 (32-bit, non-prefetchable) [size=4K] Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00004800-000048ff I/O window 1: 00004c00-00004cff 16-bit legacy interface ports at 0001 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at b3004000 (32-bit, non-prefetchable) [size=2K] Memory at b3000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 0a:02.0 0104: 105a:3373 (rev 02) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 I/O ports at 4480 [size=64] I/O ports at 44c0 [size=16] I/O ports at 4400 [size=128] Memory at b3005000 (32-bit, non-prefetchable) [size=4K] Memory at b3020000 (32-bit, non-prefetchable) [size=128K] Capabilities: [60] Power Management version 2 0a:03.0 0200: 10ec:8169 (rev 10) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 I/O ports at 4000 [size=256] Memory at b3004800 (32-bit, non-prefetchable) [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [dc] Power Management version 2 0a:04.0 0480: 1131:7133 (rev f0) Subsystem: 1461:f31e Flags: bus master, medium devsel, latency 64, IRQ 23 Memory at b3006000 (32-bit, non-prefetchable) [size=2K] Capabilities: [40] Power Management version 2 0a:05.0 0280: 1814:0201 (rev 01) Subsystem: 1462:6833 Flags: bus master, slow devsel, latency 64, IRQ 20 Memory at b3008000 (32-bit, non-prefetchable) [size=8K] Capabilities: [40] Power Management version 2 asoundconf output : eric@gondor:~$ asoundconf list Names of available sound cards: Intel SAA7134 I tried a lot of workarounds but none worked : boot options (irqpoll, pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... I must add that I already had this problem with previous distributions of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, it did worked ONLY when I self-compiled the driver with debug option enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it does not work. I cannot have one sound :-(. Please help me .... Eric

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[alsa-devel] default PCM volume
by Michel Benoit 15 May '07

15 May '07

Hello again, I am running alsa on a custom AT91SAM9260 arm based SOC board together with a a simple external PCM1725 codec. When using alsa-lib and the at91 soc driver (pcm+i2s) I can playback PCM samples using both aplay and test/pcm. Using softvol I can also reduce the volume of the playback. The problem I have is that the playback volume (with softvol=100%) is not very loud. The volume is the same even without the softvol plugin. The PCM codec has no hw volume control. According to our hw team the sound level should be much higher. Is there a default PCM attenuation (volume reduction) built into alsa lib? If so, how can it be adjusted? Thanks, Michel

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[alsa-devel] [PATCH] ALSA sound driver for SEGA Dreamcast AICA (pcm)
by Adrian McMenamin 15 May '07

15 May '07

ALSA support for the SEGA Dreamcast Yamaha AICA sound device (pcm) This patch adds ALSA sound support for pcm playback on two channels on the SEGA Dreamcast built-in sound device (the Yamaha AICA) Add driver for the AICA sound device built into the SEGA Dreamcast Hook it all up with the build system. Signed-off-by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk>

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[alsa-devel] [PATCH] SH7760 ASoC support
by Manuel Lauss 14 May '07

14 May '07

Hello, here goes attempt #3 for the SH7760 ASoC patch. Changes v2 -> v3 - added a generic demo board for AC97 Changes v1 -> v2 - incorporate Paul Mundt's suggestions - work around an oops when aplay is used in mmap mode ('aplay -M') by changing the allocation type to SNDRV_DMA_TYPE_CONTINUOUS As soon as I get Linux working on the EDOSK7780 board, I'll complete SH7780 support as well. Thanks, Manuel Lauss --- ALSA ASoC support for SH7760 - SH7760 DMA engine (dmabrg) - AC97 driver for HAC unit(s) found on SH7760/SH7780 - I2S driver for SSI unit(s) found on SH7760/SH7780 - generic SH7760-AC97 "demoboard" sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/sh/Kconfig | 39 +++++ sound/soc/sh/Makefile | 14 ++ sound/soc/sh/dma-sh7760.c | 354 +++++++++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 322 +++++++++++++++++++++++++++++++++++ sound/soc/sh/sh7760-ac97.c | 92 ++++++++++ sound/soc/sh/ssi.c | 400 ++++++++++++++++++++++++++++++++++++++++++++ 8 files changed, 1224 insertions(+), 1 deletions(-) Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dccaa4b..3e3df26 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,6 +26,7 @@ menu "SoC Platforms" depends on SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/sh/Kconfig" endmenu # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 98e6f49..e1c9a70 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ sh/ + diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 0000000..a332e51 --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,39 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +config SND_SH7760_AC97 + tristate "SH7760 AC97 sound support" + depends on CPU_SUBTYPE_SH7760 && SND_SOC_PCM_SH7760 + select SND_SOC_SH4_HAC + select SND_SOC_AC97_CODEC + help + This option enables generic sound support for the first + AC97 unit of the SH7760. + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 0000000..a8e8ab8 --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,14 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards +snd-soc-sh7760-ac97-objs := sh7760-ac97.o + +obj-$(CONFIG_SND_SH7760_AC97) += snd-soc-sh7760-ac97.o diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 0000000..cdee374 --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,354 @@ +/* + * SH7760 ("camelot") DMABRG audio DMA unit support + * + * Copyright (C) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which + * trigger an interrupt when one half of the programmed transfer size + * has been xmitted. + * + * FIXME: little-endian only for now + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dmabrg.h> + + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of a + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + pr_debug("PCM data: addr 0x%08ulx len %d\n", + (u32)runtime->dma_addr, runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = (unsigned long)runtime->dma_area; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = (unsigned long)runtime->dma_area; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * position at the time the IRQ was triggered, which results in + * fast-playback sound in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel + * in MMAP mode (i.e. aplay -M) + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-pcm", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 0000000..8e3f039 --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,322 @@ +/* + * Hitachi Audio Controller (AC97) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable HAC output pins! + */ + +/* BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only + * the FIRST can be used since ASoC does not pass any information to the + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/wait.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "hac: reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + pr_debug("hac: invalid depth %d bit\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/sh7760-ac97.c b/sound/soc/sh/sh7760-ac97.c new file mode 100644 index 0000000..5563f14 --- /dev/null +++ b/sound/soc/sh/sh7760-ac97.c @@ -0,0 +1,92 @@ +/* + * Generic AC97 sound support for SH7760 + * + * (c) 2007 Manuel Lauss + * + * Licensed under the GPLv2. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <asm/io.h> + +#include "../codecs/ac97.h" + +#define IPSEL 0xFE400034 + +/* platform specific structs can be declared here */ +extern struct snd_soc_cpu_dai sh4_hac_dai[2]; +extern struct snd_soc_platform sh7760_soc_platform; + +static int machine_init(struct snd_soc_codec *codec) +{ + snd_soc_dapm_sync_endpoints(codec); + return 0; +} + +static struct snd_soc_dai_link sh7760_ac97_dai = { + .name = "AC97", + .stream_name = "AC97 HiFi", + .cpu_dai = &sh4_hac_dai[0], /* HAC0 */ + .codec_dai = &ac97_dai, + .init = machine_init, + .ops = NULL, +}; + +static struct snd_soc_machine sh7760_ac97_soc_machine = { + .name = "SH7760 AC97", + .dai_link = &sh7760_ac97_dai, + .num_links = 1, +}; + +static struct snd_soc_device sh7760_ac97_snd_devdata = { + .machine = &sh7760_ac97_soc_machine, + .platform = &sh7760_soc_platform, + .codec_dev = &soc_codec_dev_ac97, +}; + +static struct platform_device *sh7760_ac97_snd_device; + +static int __init sh7760_ac97_init(void) +{ + int ret; + unsigned short ipsel; + + /* enable both AC97 controllers in pinmux reg */ + ipsel = ctrl_inw(IPSEL); + ctrl_outw(ipsel | (3 << 10), IPSEL); + + ret = -ENOMEM; + sh7760_ac97_snd_device = platform_device_alloc("soc-audio", -1); + if (!sh7760_ac97_snd_device) + goto out; + + platform_set_drvdata(sh7760_ac97_snd_device, + &sh7760_ac97_snd_devdata); + sh7760_ac97_snd_devdata.dev = &sh7760_ac97_snd_device->dev; + ret = platform_device_add(sh7760_ac97_snd_device); + + if (ret) + platform_device_put(sh7760_ac97_snd_device); + +out: + return ret; +} + +static void __exit sh7760_ac97_exit(void) +{ + platform_device_unregister(sh7760_ac97_snd_device); +} + +module_init(sh7760_ac97_init); +module_exit(sh7760_ac97_exit); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("Generic SH7760 AC97 sound machine"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 0000000..b72bc31 --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,400 @@ +/* + * Serial Sound Interface (I2S) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable SSI output pins! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT) +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK (7 << CR_DWL_SHIFT) +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK (7 << CR_SWL_SHIFT) +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_SHIFT 4 +#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */ +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + pr_debug("ssi: already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); + pr_debug("bits: %d channels: %d\n", bits, channels); + + ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | + CR_SWL_MASK); + + /* direction (send/receive) */ + if (!recv) + ssicr |= CR_TRMD; /* transmit */ + + /* channels */ + if ((channels < 2) || (channels > 8) || (channels & 1)) { + pr_debug("ssi: invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK; + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + pr_debug("ssi: invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP | + CR_SWS_MASTER | CR_SCK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + pr_debug("ssi: unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + pr_debug("ssi: invalid inversion\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + pr_debug("ssi: invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>");

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 14 May '07

14 May '07

Since the only successful modprobe is done with the jumper on SOFTWARE instead of EPPROM, I'll only tested the following: Jumper: Software: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 without an error, dmesg - gives no hint, verbose, mention... cat /proc/asound/cards: 0 [Pro ]: SB Pro - Sound Blaster Pro Sound Blaster Pro at 0x220, irq 5, dma 1 So I did the following: cd /proc/asound/ cat * cat: Pro: Is a directory cat: card0: Is a directory 0 [Pro ]: SB Pro - Sound Blaster Pro Sound Blaster Pro at 0x220, irq 5, dma 1 2: : timer 3: : sequencer 4: [ 0- 0]: raw midi 5: [ 0- 0]: hardware dependent 6: [ 0- 0]: digital audio playback 7: [ 0- 0]: digital audio capture 8: [ 0] : control 00-00: OPL3 FM 0 snd_sb8 cat: oss: Is a directory 00-00: SB8 DSP : DSP v3.1 : playback 1 : capture 1 cat: 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 Advanced Linux Sound Architecture Driver Version 1.0.14rc1 (Tue Jan 09 09:56:17 2007 UTC). Thanks for your help, Ram Kromberg, ramkromberg(a)mail.com

2 1

[alsa-devel] Alsa 1.0.13 (and others) does not work on ALC880
by Eric 14 May '07

14 May '07

I installed Ubuntu Feisty on a D900t laptop : it contains a ALC880 audio chipset eric@gondor:~$ lspci |grep Audio 00:1b.0 Audio device: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) High Definition Audio Controller (rev 03) When modprobing the modules I get : eric@gondor:~$ sudo modprobe -v snd-hda-intel model=3stack-digout insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-codec.ko insmod /lib/modules/2.6.20-15-generic/kernel/sound/pci/hda/snd-hda-intel.ko model=3stack-digout while /var/log/messages says : May 14 14:42:45 gondor kernel: [ 3699.070904] hda_intel: azx_get_response timeout, switching to polling mode... Feisty dist. is provided with alsa-1.0.13. I get no sound, here are the results of aplay : eric@gondor:~$ aplay ALSA lib pcm_direct.c:867:(snd_pcm_direct_initialize_slave) snd_pcm_hw_params_any failed ALSA lib pcm_dmix.c:876:(snd_pcm_dmix_open) unable to initialize slave aplay: main:550: Erreur d'ouverture audio: Argument invalide eric@gondor:~$ Here are some other infos : eric@gondor:~$ tail -2 /proc/asound/oss/sndstat 0: Realtek ALC880 1: SAA7134 Mixer amixer shows this : eric@gondor:~$ amixer Simple mixer control 'Headphone',0 Capabilities: pswitch Playback channels: Front Left - Front Right Mono: Front Left: Playback [on] Front Right: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 228 [89%] Front Right: Playback 228 [89%] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 55 [43%] [-2304.00dB] [on] Front Right: Playback 55 [43%] [-2304.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 64 [50%] [-2016.00dB] [on] Front Right: Playback 64 [50%] [-2016.00dB] [on] Simple mixer control 'Center',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 56 [44%] [-2272.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 127 Mono: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 60 [47%] [-2144.00dB] [on] Front Right: Playback 60 [47%] [-2144.00dB] [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 58 [46%] [-2208.00dB] [on] Front Right: Playback 58 [46%] [-2208.00dB] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 57 [45%] [-2240.00dB] [on] Front Right: Playback 57 [45%] [-2240.00dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'PC Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 127 Mono: Front Left: Playback 53 [42%] [-2368.00dB] [on] Front Right: Playback 53 [42%] [-2368.00dB] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 31 [24%] [-3072.00dB] [on] Front Right: Capture 31 [24%] [-3072.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 27 [21%] [-3200.00dB] [on] Front Right: Capture 27 [21%] [-3200.00dB] [on] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 127 Front Left: Capture 24 [19%] [-3296.00dB] [on] Front Right: Capture 24 [19%] [-3296.00dB] [on] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '6ch' Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] aplay -l seems to work : eric@gondor:~$ aplay -l **** Liste des PLAYBACK périphériques **** carte 0: Intel [HDA Intel], périphérique 0 : ALC880 Analog [ALC880 Analog] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 1 : ALC880 Digital [ALC880 Digital] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 carte 0: Intel [HDA Intel], périphérique 6 : Si3054 Modem [Si3054 Modem] Sous-périphériques: 1/1 Sous-périphérique: #0: subdevice #0 eric@gondor:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xb0000000 irq 17 1 [SAA7134 ]: SAA7134 - SAA7134 saa7133[0] at 0xb3006000 irq 22 if I try to do a cat /proc/asound/card0/codec#0 it hangs while /var/log/messages gives a lot of May 14 14:44:52 gondor kernel: [ 3825.457412] hda_codec: invalid dep_range_val 0:7fff May 14 14:44:52 gondor kernel: [ 3825.457696] hda_codec: invalid dep_range_val 0:7fff ........................ eric@gondor:~$ sudo lspci -nv 00:00.0 0600: 8086:2580 (rev 04) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0 Capabilities: [e0] Vendor Specific Information 00:01.0 0604: 8086:2581 (rev 04) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: b1000000-b2ffffff Prefetchable memory behind bridge: c0000000-cfffffff Capabilities: [88] Subsystem: 8086:2582 Capabilities: [80] Power Management version 2 Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Capabilities: [a0] Express Root Port (Slot+) IRQ 0 00:1b.0 0403: 8086:2668 (rev 03) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 20 Memory at b0000000 (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 00:1d.0 0c03: 8086:2658 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 I/O ports at 1800 [size=32] 00:1d.1 0c03: 8086:2659 (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 19 I/O ports at 3000 [size=32] 00:1d.2 0c03: 8086:265a (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 3020 [size=32] 00:1d.3 0c03: 8086:265b (rev 03) (prog-if 00 [UHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 20 I/O ports at 3040 [size=32] 00:1d.7 0c03: 8086:265c (rev 03) (prog-if 20 [EHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 18 Memory at b0004000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Capabilities: [58] Debug port 00:1e.0 0604: 8086:244e (rev d3) (prog-if 01 [Subtractive decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=0a, subordinate=0e, sec-latency=32 I/O behind bridge: 00004000-00004fff Memory behind bridge: b3000000-b30fffff Prefetchable memory behind bridge: 0000000050000000-0000000055ffffff Capabilities: [50] Subsystem: 1558:0900 00:1f.0 0601: 8086:2640 (rev 03) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0 00:1f.1 0101: 8086:266f (rev 03) (prog-if 8a [Master SecP PriP]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 0, IRQ 17 I/O ports at 01f0 [size=8] I/O ports at 03f4 [size=1] I/O ports at 0170 [size=8] I/O ports at 0374 [size=1] I/O ports at 3080 [size=16] 00:1f.3 0c05: 8086:266a (rev 03) Subsystem: 1558:0900 Flags: medium devsel, IRQ 10 I/O ports at 3060 [size=32] 01:00.0 0300: 10de:00c8 (rev a2) (prog-if 00 [VGA]) Subsystem: 1558:0900 Flags: bus master, fast devsel, latency 0, IRQ 11 Memory at b2000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at b1000000 (64-bit, non-prefetchable) [size=16M] Capabilities: [60] Power Management version 2 Capabilities: [68] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [78] Express Endpoint IRQ 0 0a:00.0 0607: 104c:ac50 (rev 02) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at b3007000 (32-bit, non-prefetchable) [size=4K] Bus: primary=0a, secondary=0b, subordinate=0e, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 58000000-5bfff000 I/O window 0: 00004800-000048ff I/O window 1: 00004c00-00004cff 16-bit legacy interface ports at 0001 0a:01.0 0c00: 104c:8023 (prog-if 10 [OHCI]) Subsystem: 1558:0900 Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at b3004000 (32-bit, non-prefetchable) [size=2K] Memory at b3000000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 0a:02.0 0104: 105a:3373 (rev 02) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 96, IRQ 19 I/O ports at 4480 [size=64] I/O ports at 44c0 [size=16] I/O ports at 4400 [size=128] Memory at b3005000 (32-bit, non-prefetchable) [size=4K] Memory at b3020000 (32-bit, non-prefetchable) [size=128K] Capabilities: [60] Power Management version 2 0a:03.0 0200: 10ec:8169 (rev 10) Subsystem: 1558:0900 Flags: bus master, 66MHz, medium devsel, latency 64, IRQ 22 I/O ports at 4000 [size=256] Memory at b3004800 (32-bit, non-prefetchable) [size=256] [virtual] Expansion ROM at 54000000 [disabled] [size=128K] Capabilities: [dc] Power Management version 2 0a:04.0 0480: 1131:7133 (rev f0) Subsystem: 1461:f31e Flags: bus master, medium devsel, latency 64, IRQ 23 Memory at b3006000 (32-bit, non-prefetchable) [size=2K] Capabilities: [40] Power Management version 2 0a:05.0 0280: 1814:0201 (rev 01) Subsystem: 1462:6833 Flags: bus master, slow devsel, latency 64, IRQ 20 Memory at b3008000 (32-bit, non-prefetchable) [size=8K] Capabilities: [40] Power Management version 2 asoundconf output : eric@gondor:~$ asoundconf list Names of available sound cards: Intel SAA7134 I tried a lot of workarounds but none worked : boot options (irqpoll, pci=noacpi), single_cmd=1, position_fix=1, several models of hda-intel driver, I tried 1.0.13 and 1.0.14rc4 driver, with debug on... I must add that I already had this problem with previous distributions of Ubuntu. It worked with an old one (Breezy) then in Dapper and Edgy, it did worked ONLY when I self-compiled the driver with debug option enabled I never understood why. And now with Feisty/alsa 1.0.13/ubuntu kernel 2.6.20, or an ubuntu kernel with self-compiled alsa 1.0.14rc4, it does not work. I cannot have one sound :-(. Please help me .... Eric

1 0

[alsa-devel] alsa-lib functions reporting conflicting information with two sound cards
by nick smethurst 14 May '07

14 May '07

Hey guys I am rather confused that I am getting crossed information when calling snd_names_list and snd_pcm_info in order to gather information about pcm devices. I have two sound cards in my development pc: RME Digi 96/8 PCI and an onboard VIA 8237. I'm trying to develop a piece of code that generates a list of objects containing pcm device information, including information on virtual devices. My current idea is to use snd_names_list() to get a full list of devices, and then get information about them by opening each device and querying. The bizzare thing is that snd_names_list() is returning comments that do not agree with the associated device. For example, given the following code: #include <alsa/asoundlib.h> int main() { snd_devname_t * list; snd_config_update(); snd_names_list("pcm", &list); while (list) { if (!strcmp(list->name, "hw:0,0")) { printf("hw:0,0 comment: %s\n", list->comment); break; } list = list->next; } snd_names_list_free(list); snd_pcm_t * pcm; snd_pcm_info_t * pcm_info; snd_pcm_info_malloc(&pcm_info); snd_pcm_open(&pcm, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0); snd_pcm_info(pcm, pcm_info); printf("id: %s\n", snd_pcm_info_get_id(pcm_info)); snd_pcm_close(pcm); snd_pcm_info_free(pcm_info); return 0; } The result on my pc is: hw:0,0 comment: Physical Device - VIA 8237 (Duplex) id: Digi96 IEC958 So I'm apparently doing something very wrong here. I would like to ask if there is an easier and more reliable way of getting a list of pcm devices and querying them, and if not, what I am doing wrong here? Regards Nicholas

6 16

[alsa-devel] [PATCH1/5] ASoC S3C24xx machine drivers - s3c2443-AC97
by Liam Girdwood 14 May '07

14 May '07

Sorry for the lateness in the current merge window of this patch series. From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds AC97 support to the Samsung S3C2443 CPU. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

3 4

[alsa-devel] Undefined Symbols: snd-aoa-codec-tas.c:944 warning: implicit declaration of function 'of_get_property'
by Elimar Riesebieter 14 May '07

14 May '07

Hi all, the aoa drivers in 1.0.14rc4 are not usable. Compiling the modules there are much warnings "implicit declaration of function 'of_get_property'" Loading the module dmesg tells: snd_aoa: Unknown symbol of_get_property. rc3 out of 2.6.21 works fine. Any hints? Thanks Elimar -- You cannot propel yourself forward by patting yourself on the back.

3 4

[alsa-devel] Dreamcast AICA
by Adrian McMenamin 14 May '07

14 May '07

I haven't touched this for a year, but after some small changes (to run queues) I can compile this and it runs well. Anyone care to comment on the prospects of getting it committed? I'll obviously be happy to be named as the maintainer in that case... diff -Naur ./linux-2.6.21-orig/sound/Kconfig ./linux-2.6.21/sound/Kconfig --- ./linux-2.6.21-orig/sound/Kconfig 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Kconfig 2007-05-07 21:20:54.000000000 +0100 @@ -64,6 +64,8 @@ source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff -Naur ./linux-2.6.21-orig/sound/Makefile ./linux-2.6.21/sound/Makefile --- ./linux-2.6.21-orig/sound/Makefile 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Makefile 2007-05-07 21:37:05.000000000 +0100 @@ -5,7 +5,7 @@ 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-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff -Naur ./linux-2.6.21-orig/sound/sh/aica.c ./linux-2.6.21/sound/sh/aica.c --- ./linux-2.6.21-orig/sound/sh/aica.c 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.c 2007-05-12 15:27:54.000000000 +0100 @@ -0,0 +1,677 @@ +/* +* This code is licenced under +* the General Public Licence +* version 2 +* +* Copyright Adrian McMenamin 2005, 2006 +* <adrian(a)mcmen.demon.co.uk> +* See also http://newgolddream.dyndns.info/cgi-bin/cvsweb +* +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of version 2 of the GNU General Public License as published by +* the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#include <linux/init.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <linux/wait.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/timer.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <asm/io.h> +#include <asm/dma.h> +#include <asm/dreamcast/sysasic.h> +#include "aica.h" + +MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); +MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); + +/* module parameters */ +#define CARD_NAME "AICA" +static int index = -1; +static char *id; +static int enable = 1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param(enable, bool, 0644); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +/* Use workqueue */ + +static struct spu_work_holder { + struct work_struct spu_dma_work; + void *sspointer; +} spu_working; + +static struct workqueue_struct *aica_queue; + +/* Simple platform device */ +static struct platform_device *pd; +static struct resource aica_memory_space[2] = { + { + .name = "AICA ARM CONTROL", + .start = ARM_RESET_REGISTER, + .flags = IORESOURCE_MEM, + .end = ARM_RESET_REGISTER + 3, + }, + { + .name = "AICA Sound RAM", + .start = SPU_MEMORY_BASE, + .flags = IORESOURCE_MEM, + .end = SPU_MEMORY_BASE + 0x200000 - 1, + }, +}; + +/* SPU specific functions */ +/* spu_write_wait - wait for G2-SH FIFO to clear */ +static inline void spu_write_wait(void) +{ + int time_count; + time_count = 0; + while (1) { + if (!(readl(G2_FIFO) & 0x11)) + break; + /* To ensure hardware failure doesn't wedge kernel */ + time_count++; + if (time_count > 0x10000) + break; + } +} + +/* spu_memset - write to memory in SPU address space */ +static void spu_memset(uint32_t toi, uint32_t what, int length) +{ + int i; + snd_assert(length % 4 == 0, return); + spu_write_wait(); + for (i = 0; i < length; i++) { + writel(what, toi + SPU_MEMORY_BASE); + toi++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_memload - write to SPU address space */ +static void spu_memload(uint32_t toi, void __iomem * from, int length) +{ + uint32_t __iomem *froml = from; + uint32_t __iomem *to = (uint32_t __iomem *) (SPU_MEMORY_BASE + toi); + int i, val; + if (length % 4) + length = (length / 4) + 1; + else + length = length / 4; + spu_write_wait(); + for (i = 0; i < length; i++) { + val = *froml; + writel(val, to); + froml++; + to++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_disable - set spu registers to stop sound output */ +static void spu_disable(void) +{ + int i; + uint32_t regval; + spu_write_wait(); + regval = readl(ARM_RESET_REGISTER); + regval |= 1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); + for (i = 0; i < 64; i++) { + spu_write_wait(); + regval = readl(SPU_REGISTER_BASE + (i * 0x80)); + regval = (regval & ~0x4000) | 0x8000; + spu_write_wait(); + writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + } +} + +/* spu_enable - set spu registers to enable sound output */ +static void spu_enable(void) +{ + uint32_t regval = readl(ARM_RESET_REGISTER); + regval &= ~1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); +} + +/* + * Halt the sound processor, clear the memory, + * load some default ARM7 code, and then restart ARM7 +*/ +static void spu_reset(void) +{ + spu_disable(); + spu_memset(0, 0, 0x200000 / 4); + /* Put ARM7 in endless loop */ + ctrl_outl(0xea000002, SPU_MEMORY_BASE); + spu_enable(); +} + +/* aica_chn_start - write to spu to start playback */ +static void aica_chn_start(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_START, (uint32_t *) AICA_CONTROL_POINT); +} + +/* aica_chn_halt - write to spu to halt playback */ +static void aica_chn_halt(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_STOP, (uint32_t *) AICA_CONTROL_POINT); +} + +/* ALSA code below */ +static struct snd_pcm_hardware snd_pcm_aica_playback_hw = { + .info = (SNDRV_PCM_INFO_NONINTERLEAVED), + .formats = + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IMA_ADPCM), + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = AICA_BUFFER_SIZE, + .period_bytes_min = AICA_PERIOD_SIZE, + .period_bytes_max = AICA_PERIOD_SIZE, + .periods_min = AICA_PERIOD_NUMBER, + .periods_max = AICA_PERIOD_NUMBER, +}; + +static int aica_dma_transfer(int channels, int buffer_size, + struct snd_pcm_substream *substream) +{ + int q, err, period_offset; + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + err = 0; + dreamcastcard = substream->pcm->private_data; + period_offset = dreamcastcard->clicks; + period_offset %= (AICA_PERIOD_NUMBER / channels); + runtime = substream->runtime; + for (q = 0; q < channels; q++) { + err = dma_xfer(AICA_DMA_CHANNEL, + (unsigned long)(runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), + AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + + AICA_PERIOD_SIZE * period_offset, + buffer_size / channels, AICA_DMA_MODE); + if (unlikely(err < 0)) + break; + dma_wait_for_completion(AICA_DMA_CHANNEL); + } + return err; +} + +static void startup_aica(struct snd_card_aica *dreamcastcard) +{ + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + aica_chn_start(); +} + +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; + 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); + startup_aica(dreamcastcard); + dreamcastcard->clicks = + buffer_size / (AICA_PERIOD_SIZE * runtime->channels); + return; + } else { + aica_dma_transfer(runtime->channels, + AICA_PERIOD_SIZE * runtime->channels, + substream); + snd_pcm_period_elapsed(dreamcastcard->substream); + dreamcastcard->clicks++; + dreamcastcard->clicks %= AICA_PERIOD_NUMBER; + mod_timer(&dreamcastcard->timer, jiffies + 1); + } +} + +static void aica_period_elapsed(unsigned long timer_var) +{ + /*timer fuction - so cannot sleep */ + int play_period; + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream; + struct snd_card_aica *dreamcastcard; + substream = (struct snd_pcm_substream *)timer_var; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Have we played out an additional period? */ + play_period = + frames_to_bytes(runtime, + readl + (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) / + AICA_PERIOD_SIZE; + if (play_period == dreamcastcard->current_period) { + /* reschedule the timer */ + dreamcastcard->timer.expires = jiffies + 1; + add_timer(&(dreamcastcard->timer)); + return; + } + if (runtime->channels > 1) + dreamcastcard->current_period = play_period; + if (unlikely(dreamcastcard->dma_check == 0)) + dreamcastcard->dma_check = 1; + queue_work(aica_queue, &(spu_working.spu_dma_work)); +} + +static void spu_begin_dma(struct snd_pcm_substream *substream) +{ + /* Must be atomic */ + struct snd_card_aica *dreamcastcard; + 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)); + /* Timer may already be running */ + if (unlikely(dreamcastcard->timer.data)) { + mod_timer(&dreamcastcard->timer, jiffies + 4); + return; + } + init_timer(&(dreamcastcard->timer)); + dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.function = aica_period_elapsed; + dreamcastcard->timer.expires = jiffies + 4; + add_timer(&(dreamcastcard->timer)); +} + +static int snd_aicapcm_pcm_open(struct snd_pcm_substream + *substream) +{ + struct snd_pcm_runtime *runtime; + struct aica_channel *channel; + struct snd_card_aica *dreamcastcard; + if (!enable) + return -ENOENT; + dreamcastcard = substream->pcm->private_data; + channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL); + if (!channel) + return -ENOMEM; + /* set defaults for channel */ + channel->sfmt = SM_8BIT; + channel->cmd = AICA_CMD_START; + channel->vol = dreamcastcard->master_volume; + channel->pan = 0x80; + channel->pos = 0; + channel->flags = 0; /* default to mono */ + dreamcastcard->channel = channel; + runtime = substream->runtime; + runtime->hw = snd_pcm_aica_playback_hw; + spu_enable(); + dreamcastcard->clicks = 0; + dreamcastcard->current_period = 0; + dreamcastcard->dma_check = 0; + return 0; +} + +static int snd_aicapcm_pcm_close(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + del_timer(&dreamcastcard->timer); + kfree(dreamcastcard->channel); + spu_disable(); + return 0; +} + +static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream + *substream) +{ + /* Free the DMA buffer */ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream + *substream, struct snd_pcm_hw_params + *hw_params) +{ + /* Allocate a DMA buffer using ALSA built-ins */ + return + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE) + dreamcastcard->channel->sfmt = SM_16BIT; + dreamcastcard->channel->freq = substream->runtime->rate; + dreamcastcard->substream = substream; + return 0; +} + +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; + } + return 0; +} + +static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream + *substream) +{ + return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER); +} + +static struct snd_pcm_ops snd_aicapcm_playback_ops = { + .open = snd_aicapcm_pcm_open, + .close = snd_aicapcm_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aicapcm_pcm_hw_params, + .hw_free = snd_aicapcm_pcm_hw_free, + .prepare = snd_aicapcm_pcm_prepare, + .trigger = snd_aicapcm_pcm_trigger, + .pointer = snd_aicapcm_pcm_pointer, +}; + +/* TO DO: set up to handle more than one pcm instance */ +static int __init snd_aicapcmchip(struct snd_card_aica + *dreamcastcard, int pcm_index) +{ + struct snd_pcm *pcm; + int err; + /* AICA has no capture ability */ + err = + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + if (unlikely(err < 0)) + return err; + pcm->private_data = dreamcastcard; + strcpy(pcm->name, "AICA PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aicapcm_playback_ops); + /* Allocate the DMA buffers */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), + AICA_BUFFER_SIZE, + AICA_BUFFER_SIZE); + return err; +} + +/* Mixer controls */ +static int aica_pcmswitch_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 aica_pcmswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */ + return 0; +} + +static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (ucontrol->value.integer.value[0] == 1) + return 0; /* TO DO: Fix me */ + else + aica_chn_halt(); + return 0; +} + +static int aica_pcmvolume_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 = 0xFF; + return 0; +} + +static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; /* we've not yet been set up */ + ucontrol->value.integer.value[0] = dreamcastcard->channel->vol; + return 0; +} + +static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; + if (unlikely(dreamcastcard->channel->vol == + ucontrol->value.integer.value[0])) + return 0; + dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; + dreamcastcard->master_volume = ucontrol->value.integer.value[0]; + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + + return 1; +} + +static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .info = aica_pcmswitch_info, + .get = aica_pcmswitch_get, + .put = aica_pcmswitch_put +}; + +static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = aica_pcmvolume_info, + .get = aica_pcmvolume_get, + .put = aica_pcmvolume_put +}; + +static int load_aica_firmware(void) +{ + int err; + const struct firmware *fw_entry; + err = 0; + spu_reset(); + err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); + if (unlikely(err)) + return err; + /* write firware into memory */ + spu_disable(); + spu_memload(0, fw_entry->data, fw_entry->size); + spu_enable(); + release_firmware(fw_entry); + return err; +} + +static int __devinit add_aicamixer_controls(struct snd_card_aica + *dreamcastcard) +{ + int err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + return 0; +} + +static int snd_aica_remove(struct platform_device *devptr) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = platform_get_drvdata(devptr); + if (unlikely(!dreamcastcard)) + return -ENODEV; + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + platform_set_drvdata(devptr, NULL); + return 0; +} + +static int __init snd_aica_probe(struct platform_device *devptr) +{ + int err; + struct snd_card_aica *dreamcastcard; + dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + if (unlikely(!dreamcastcard)) + return -ENOMEM; + dreamcastcard->card = + snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); + if (unlikely(!dreamcastcard->card)) { + kfree(dreamcastcard); + return -ENODEV; + } + strcpy(dreamcastcard->card->driver, "snd_aica"); + strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); + strcpy(dreamcastcard->card->longname, + "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Load the PCM 'chip' */ + err = snd_aicapcmchip(dreamcastcard, 0); + if (unlikely(err < 0)) + goto freedreamcast; + snd_card_set_dev(dreamcastcard->card, &devptr->dev); + dreamcastcard->timer.data = 0; + dreamcastcard->channel = NULL; + /* Add basic controls */ + err = add_aicamixer_controls(dreamcastcard); + if (unlikely(err < 0)) + goto freedreamcast; + /* Register the card with ALSA subsystem */ + err = snd_card_register(dreamcastcard->card); + if (unlikely(err < 0)) + goto freedreamcast; + platform_set_drvdata(devptr, dreamcastcard); + aica_queue = create_workqueue("aica"); + if (unlikely(!aica_queue)) + goto freedreamcast; + snd_printk + ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + return 0; + freedreamcast: + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + return err; +} + +static struct platform_driver snd_aica_driver = { + .probe = snd_aica_probe, + .remove = snd_aica_remove, + .driver = { + .name = SND_AICA_DRIVER}, +}; + +static int __init aica_init(void) +{ + int err; + err = platform_driver_register(&snd_aica_driver); + if (unlikely(err < 0)) + return err; + pd = platform_device_register_simple(SND_AICA_DRIVER, -1, + aica_memory_space, 2); + if (unlikely(IS_ERR(pd))) { + platform_driver_unregister(&snd_aica_driver); + return PTR_ERR(pd); + } + /* Load the firmware */ + err = load_aica_firmware(); + if (unlikely(err < 0)) + return err; + + return 0; +} + +static void __exit aica_exit(void) +{ + /* Destroy the aica kernel thread */ + destroy_workqueue(aica_queue); + platform_device_unregister(pd); + platform_driver_unregister(&snd_aica_driver); + /* Kill any sound still playing and reset ARM7 to safe state */ + spu_reset(); +} + +module_init(aica_init); +module_exit(aica_exit); diff -Naur ./linux-2.6.21-orig/sound/sh/aica.h ./linux-2.6.21/sound/sh/aica.h --- ./linux-2.6.21-orig/sound/sh/aica.h 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.h 2007-05-07 21:47:31.000000000 +0100 @@ -0,0 +1,80 @@ +/* aica.h + * Header file for ALSA driver for + * Sega Dreamcast Yamaha AICA sound + * Copyright Adrian McMenamin + * <adrian(a)mcmen.demon.co.uk> + * 2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License as published by + * the Free Software Foundation. + * + * 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 + * + */ + +/* SPU memory and register constants etc */ +#define G2_FIFO 0xa05f688c +#define SPU_MEMORY_BASE 0xA0800000 +#define ARM_RESET_REGISTER 0xA0702C00 +#define SPU_REGISTER_BASE 0xA0700000 + +/* AICA channels stuff */ +#define AICA_CONTROL_POINT 0xA0810000 +#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008 +#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004 + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +/* Buffer and period size */ +#define AICA_BUFFER_SIZE 0x8000 +#define AICA_PERIOD_SIZE 0x800 +#define AICA_PERIOD_NUMBER 16 + +#define AICA_CHANNEL0_OFFSET 0x11000 +#define AICA_CHANNEL1_OFFSET 0x21000 +#define CHANNEL_OFFSET 0x10000 + +#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_MODE 5 + +#define SND_AICA_DRIVER "AICA" + +struct aica_channel { + uint32_t cmd; /* Command ID */ + uint32_t pos; /* Sample position */ + uint32_t length; /* Sample length */ + uint32_t freq; /* Frequency */ + uint32_t vol; /* Volume 0-255 */ + uint32_t pan; /* Pan 0-255 */ + uint32_t sfmt; /* Sound format */ + uint32_t flags; /* Bit flags */ +}; + +struct snd_card_aica { + struct snd_card *card; + struct aica_channel *channel; + struct snd_pcm_substream *substream; + int clicks; + int current_period; + struct timer_list timer; + int master_volume; + int dma_check; +}; diff -Naur ./linux-2.6.21-orig/sound/sh/Kconfig ./linux-2.6.21/sound/sh/Kconfig --- ./linux-2.6.21-orig/sound/sh/Kconfig 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Kconfig 2007-05-07 21:28:30.000000000 +0100 @@ -0,0 +1,14 @@ +# ALSA SH drivers + +menu "SUPERH devices" + depends on SND!=n && SUPERH + +config SND_AICA + tristate "Dreamcast Yamaha AICA sound" + depends on SH_DREAMCAST && SND + select SND_PCM + help + ALSA Sound driver for the SEGA Dreamcast console. + +endmenu + diff -Naur ./linux-2.6.21-orig/sound/sh/Makefile ./linux-2.6.21/sound/sh/Makefile --- ./linux-2.6.21-orig/sound/sh/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Makefile 2007-05-07 21:34:10.000000000 +0100 @@ -0,0 +1,8 @@ +# +# Makefile for ALSA +# + +snd-aica-objs := aica.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_AICA) += snd-aica.o Signed-off by: Adrian McMenamin

2 1

[alsa-devel] [PATCH] Coping with short replies in usbmixer
by thomas＠devbase.at 14 May '07

14 May '07

Hi, while trying to find out why a USB handset (Yealink P1K) frequently gets muted when changing volume, I found that short USB responses cause the problem. The function get_ctl_value(..) in usbmixer.c requests the state of a certain control and retries up to 10 times if an error occurs. However a short USB response is not treated as an error as neither the transfer flag URB_SHORT_NOT_OK is set nor is the return value of snd_usb_ctl_msg(..) compared with the requested amount of data. As a result the function get_ctl_value(..) might not retry the USB request but returns an unpredictable result without reporting an error. The attached patch forces the number of returned bytes to be at least val_len. It was generated from kernel version 2.6.21. If there are any issues with the patch please let me know. Otherwise I would be happy if it could be applied to the ALSA sources. Regards, -Thomas

2 1

[alsa-devel] Meaning of this Plug PCM Transformation table?
by Nobin Mathew 14 May '07

14 May '07

Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 3840 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'IDENTITY' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 3840 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 This is capture Handle logs Plug PCM: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0*0.5 + 1*0.5 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Hardware PCM card 0 'IDENTITY' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 256 period_time : 32000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 256 xfer_align : 256 start_threshold : 1 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 I am having an application which does capture and playback simultaneously and is single threaded? What is the meaning of this Transformation table?

1 0

[alsa-devel] [2/2] 2.6.22-rc1: known regressions
by Michal Piotrowski 13 May '07

13 May '07

Hi all, Here is a list of some known regressions in 2.6.22-rc. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions (http://bugzilla.kernel.org/ is not available, information are not up-to-date) Networking Subject : panic with e1000 driver on HP Integrity servers References : http://bugzilla.kernel.org/show_bug.cgi?id=8455 Submitter : Doug Chapman <doug.chapman(a)hp.com> Caused-By : Auke Kok <auke-jan.h.kok(a)intel.com> commit e0aac5a289b1dacbc94bd9ae8c449bcdf9ab508c Status : problem is being debugged Timers/NOHZ Subject : 2.6.21-git4 BUG: soft lockup detected on CPU#1! References : http://lkml.org/lkml/2007/5/2/511 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Thomas Gleixner <tglx(a)linutronix.de> Status : problem is being debugged ALSA Subject : lost snd_4236 device References : http://lkml.org/lkml/2007/5/13/87 Submitter : Pete Clements <clem(a)clem.clem-digital.net> Status : Unknown Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

1 0

[alsa-devel] e-mu 1212m and 1820(m) patch
by Ctirad Fertr 13 May '07

13 May '07

Hello, here is a little patch which enables ADAT and S/PDIF functionality. It adds possibility to sync with either S/PDIF or ADAT clock as well. 44.1/48kHz only, though it works pretty well for me. Regards, Ctirad

2 6

[alsa-devel] embedded sound architecture question
by Joachim Förster 13 May '07

13 May '07

Hi ALSA devs, I'm going to write an ALSA driver for a not yet existing AC97 controller, which is going to be "written" (VHDL), too (at the same time). Platform/Board is a Xilinx ML403 with Virtex-4 FPGA, PowerPC 405 architecture, OPB/OCP bus, AC97 Codec LM4550. Before presenting my question, I have to say, that I'm a beginner with ALSA/Linux driver development. My question is: Does the architecture described below make sense/is reasonable with ALSA and Linux? The problem is, that there is no DMA controller and implementing an OPB master device which would be able to do DMA itself is not an option at this time. So our thoughts were: Integrate the "DMA ring buffer" (which is usually somewhere in main memory/RAM) into the AC97 controller. Make ALSA access this HW buffer as if it was in main memory. This way, the device (AC97 controller) has "direct access" to its buffer "memory" - this could be called "fake DMA". The AC97 controller would have tell us where it is while playing and firing interrupts after one period, so that we don't write to values which are in the current period, instead update the area where of the past played periods etc. ... The buffer should be a "ring buffer", right? Mapping this "IO memory" into kernel space should be possible with io_remap_page_range(), right? I would have to implement the mmap() callback in my driver, to setup the given VMA (with the above function), right? So, ALSA library/applications will be able to use MMAP mode, which is what we want to achieve? [We don't want copy()/silence(). An intermediate buffer with ack()/tasklet/workqueue + FIFO in HW would be an alternative.] I read, that many applications don't work, if MMAP mode is not supported and classic read/write (copy()/silence()) is used, only. Is there a black list of apps, which don't work? Thanks for reading & your time, Joachim

4 4

[alsa-devel] apologies for this email
by Adrian McMenamin 13 May '07

13 May '07

My last two made it in to the archive, but didn't get delivered to me. Did anyone else see them?

3 2

[alsa-devel] Sound driver for Sega Dreamcast
by Adrian McMenamin 12 May '07

12 May '07

My last email is in the archive, but seem to make it back to me - in any case this code is better... but in case my last email didn't get delivered anywhere, I am asking for views on what needs to be done to get this committed. diff -Naur ./linux-2.6.21-orig/sound/Kconfig ./linux-2.6.21/sound/Kconfig --- ./linux-2.6.21-orig/sound/Kconfig 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Kconfig 2007-05-07 21:20:54.000000000 +0100 @@ -64,6 +64,8 @@ source "sound/mips/Kconfig" +source "sound/sh/Kconfig" + # the following will depend on the order of config. # here assuming USB is defined before ALSA source "sound/usb/Kconfig" diff -Naur ./linux-2.6.21-orig/sound/Makefile ./linux-2.6.21/sound/Makefile --- ./linux-2.6.21-orig/sound/Makefile 2007-04-26 04:08:32.000000000 +0100 +++ ./linux-2.6.21/sound/Makefile 2007-05-07 21:37:05.000000000 +0100 @@ -5,7 +5,7 @@ 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-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ soc/ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out diff -Naur ./linux-2.6.21-orig/sound/sh/aica.c ./linux-2.6.21/sound/sh/aica.c --- ./linux-2.6.21-orig/sound/sh/aica.c 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.c 2007-05-12 19:10:32.000000000 +0100 @@ -0,0 +1,676 @@ +/* +* This code is licenced under +* the General Public Licence +* version 2 +* +* Copyright Adrian McMenamin 2005, 2006 +* <adrian(a)mcmen.demon.co.uk> +* See also http://newgolddream.dyndns.info/cgi-bin/cvsweb +* +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of version 2 of the GNU General Public License as published by +* the Free Software Foundation. +* +* This program is distributed in the hope that it will be useful, +* but WITHOUT ANY WARRANTY; without even the implied warranty of +* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the +* GNU General Public License for more details. +* +* You should have received a copy of the GNU General Public License +* along with this program; if not, write to the Free Software +* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA +* +*/ + +#include <linux/init.h> +#include <linux/jiffies.h> +#include <linux/slab.h> +#include <linux/time.h> +#include <linux/wait.h> +#include <linux/moduleparam.h> +#include <linux/platform_device.h> +#include <linux/firmware.h> +#include <linux/timer.h> +#include <linux/delay.h> +#include <linux/workqueue.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/info.h> +#include <asm/io.h> +#include <asm/dma.h> +#include <asm/dreamcast/sysasic.h> +#include "aica.h" + +MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); +MODULE_DESCRIPTION("Dreamcast AICA sound (pcm) driver"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Yamaha/SEGA, AICA}}"); + +/* module parameters */ +#define CARD_NAME "AICA" +static int index = -1; +static char *id; +static int enable = 1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard."); +module_param(enable, bool, 0644); +MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard."); + +/* Use workqueue */ + +static struct spu_work_holder { + struct work_struct spu_dma_work; + void *sspointer; +} spu_working; + +static struct workqueue_struct *aica_queue; + +/* Simple platform device */ +static struct platform_device *pd; +static struct resource aica_memory_space[2] = { + { + .name = "AICA ARM CONTROL", + .start = ARM_RESET_REGISTER, + .flags = IORESOURCE_MEM, + .end = ARM_RESET_REGISTER + 3, + }, + { + .name = "AICA Sound RAM", + .start = SPU_MEMORY_BASE, + .flags = IORESOURCE_MEM, + .end = SPU_MEMORY_BASE + 0x200000 - 1, + }, +}; + +/* SPU specific functions */ +/* spu_write_wait - wait for G2-SH FIFO to clear */ +static inline void spu_write_wait(void) +{ + int time_count; + time_count = 0; + while (1) { + if (!(readl(G2_FIFO) & 0x11)) + break; + /* To ensure hardware failure doesn't wedge kernel */ + time_count++; + if (time_count > 0x10000) + break; + } +} + +/* spu_memset - write to memory in SPU address space */ +static void spu_memset(uint32_t toi, uint32_t what, int length) +{ + int i; + snd_assert(length % 4 == 0, return); + spu_write_wait(); + for (i = 0; i < length; i++) { + writel(what, toi + SPU_MEMORY_BASE); + toi++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_memload - write to SPU address space */ +static void spu_memload(uint32_t toi, void __iomem * from, int length) +{ + uint32_t __iomem *froml = from; + uint32_t __iomem *to = (uint32_t __iomem *) (SPU_MEMORY_BASE + toi); + int i, val; + if (length % 4) + length = (length / 4) + 1; + else + length = length / 4; + spu_write_wait(); + for (i = 0; i < length; i++) { + val = *froml; + writel(val, to); + froml++; + to++; + if (i && !(i % 8)) + spu_write_wait(); + } +} + +/* spu_disable - set spu registers to stop sound output */ +static void spu_disable(void) +{ + int i; + uint32_t regval; + spu_write_wait(); + regval = readl(ARM_RESET_REGISTER); + regval |= 1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); + for (i = 0; i < 64; i++) { + spu_write_wait(); + regval = readl(SPU_REGISTER_BASE + (i * 0x80)); + regval = (regval & ~0x4000) | 0x8000; + spu_write_wait(); + writel(regval, SPU_REGISTER_BASE + (i * 0x80)); + } +} + +/* spu_enable - set spu registers to enable sound output */ +static void spu_enable(void) +{ + uint32_t regval = readl(ARM_RESET_REGISTER); + regval &= ~1; + spu_write_wait(); + writel(regval, ARM_RESET_REGISTER); +} + +/* + * Halt the sound processor, clear the memory, + * load some default ARM7 code, and then restart ARM7 +*/ +static void spu_reset(void) +{ + spu_disable(); + spu_memset(0, 0, 0x200000 / 4); + /* Put ARM7 in endless loop */ + ctrl_outl(0xea000002, SPU_MEMORY_BASE); + spu_enable(); +} + +/* aica_chn_start - write to spu to start playback */ +static void aica_chn_start(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_START, (uint32_t *) AICA_CONTROL_POINT); +} + +/* aica_chn_halt - write to spu to halt playback */ +static void aica_chn_halt(void) +{ + spu_write_wait(); + writel(AICA_CMD_KICK | AICA_CMD_STOP, (uint32_t *) AICA_CONTROL_POINT); +} + +/* ALSA code below */ +static struct snd_pcm_hardware snd_pcm_aica_playback_hw = { + .info = (SNDRV_PCM_INFO_NONINTERLEAVED), + .formats = + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_IMA_ADPCM), + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = AICA_BUFFER_SIZE, + .period_bytes_min = AICA_PERIOD_SIZE, + .period_bytes_max = AICA_PERIOD_SIZE, + .periods_min = AICA_PERIOD_NUMBER, + .periods_max = AICA_PERIOD_NUMBER, +}; + +static int aica_dma_transfer(int channels, int buffer_size, + struct snd_pcm_substream *substream) +{ + int q, err, period_offset; + struct snd_card_aica *dreamcastcard; + struct snd_pcm_runtime *runtime; + err = 0; + dreamcastcard = substream->pcm->private_data; + period_offset = dreamcastcard->clicks; + period_offset %= (AICA_PERIOD_NUMBER / channels); + runtime = substream->runtime; + for (q = 0; q < channels; q++) { + err = dma_xfer(AICA_DMA_CHANNEL, + (unsigned long)(runtime->dma_area + + (AICA_BUFFER_SIZE * q) / + channels + + AICA_PERIOD_SIZE * + period_offset), + AICA_CHANNEL0_OFFSET + q * CHANNEL_OFFSET + + AICA_PERIOD_SIZE * period_offset, + buffer_size / channels, AICA_DMA_MODE); + if (unlikely(err < 0)) + break; + dma_wait_for_completion(AICA_DMA_CHANNEL); + } + return err; +} + +static void startup_aica(struct snd_card_aica *dreamcastcard) +{ + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + aica_chn_start(); +} + +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; + 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); + startup_aica(dreamcastcard); + dreamcastcard->clicks = + buffer_size / (AICA_PERIOD_SIZE * runtime->channels); + return; + } else { + aica_dma_transfer(runtime->channels, + AICA_PERIOD_SIZE * runtime->channels, + substream); + snd_pcm_period_elapsed(dreamcastcard->substream); + dreamcastcard->clicks++; + dreamcastcard->clicks %= AICA_PERIOD_NUMBER; + mod_timer(&dreamcastcard->timer, jiffies + 1); + } +} + +static void aica_period_elapsed(unsigned long timer_var) +{ + /*timer fuction - so cannot sleep */ + int play_period; + struct snd_pcm_runtime *runtime; + struct snd_pcm_substream *substream; + struct snd_card_aica *dreamcastcard; + substream = (struct snd_pcm_substream *)timer_var; + runtime = substream->runtime; + dreamcastcard = substream->pcm->private_data; + /* Have we played out an additional period? */ + play_period = + frames_to_bytes(runtime, + readl + (AICA_CONTROL_CHANNEL_SAMPLE_NUMBER)) / + AICA_PERIOD_SIZE; + if (play_period == dreamcastcard->current_period) { + /* reschedule the timer */ + mod_timer(&(dreamcastcard->timer), jiffies + 1); + return; + } + if (runtime->channels > 1) + dreamcastcard->current_period = play_period; + if (unlikely(dreamcastcard->dma_check == 0)) + dreamcastcard->dma_check = 1; + queue_work(aica_queue, &(spu_working.spu_dma_work)); +} + +static void spu_begin_dma(struct snd_pcm_substream *substream) +{ + /* Must be atomic */ + struct snd_card_aica *dreamcastcard; + 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)); + /* Timer may already be running */ + if (unlikely(dreamcastcard->timer.data)) { + mod_timer(&dreamcastcard->timer, jiffies + 4); + return; + } + init_timer(&(dreamcastcard->timer)); + dreamcastcard->timer.data = (unsigned long)substream; + dreamcastcard->timer.function = aica_period_elapsed; + dreamcastcard->timer.expires = jiffies + 4; + add_timer(&(dreamcastcard->timer)); +} + +static int snd_aicapcm_pcm_open(struct snd_pcm_substream + *substream) +{ + struct snd_pcm_runtime *runtime; + struct aica_channel *channel; + struct snd_card_aica *dreamcastcard; + if (!enable) + return -ENOENT; + dreamcastcard = substream->pcm->private_data; + channel = kmalloc(sizeof(struct aica_channel), GFP_KERNEL); + if (!channel) + return -ENOMEM; + /* set defaults for channel */ + channel->sfmt = SM_8BIT; + channel->cmd = AICA_CMD_START; + channel->vol = dreamcastcard->master_volume; + channel->pan = 0x80; + channel->pos = 0; + channel->flags = 0; /* default to mono */ + dreamcastcard->channel = channel; + runtime = substream->runtime; + runtime->hw = snd_pcm_aica_playback_hw; + spu_enable(); + dreamcastcard->clicks = 0; + dreamcastcard->current_period = 0; + dreamcastcard->dma_check = 0; + return 0; +} + +static int snd_aicapcm_pcm_close(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + del_timer(&dreamcastcard->timer); + kfree(dreamcastcard->channel); + spu_disable(); + return 0; +} + +static int snd_aicapcm_pcm_hw_free(struct snd_pcm_substream + *substream) +{ + /* Free the DMA buffer */ + return snd_pcm_lib_free_pages(substream); +} + +static int snd_aicapcm_pcm_hw_params(struct snd_pcm_substream + *substream, struct snd_pcm_hw_params + *hw_params) +{ + /* Allocate a DMA buffer using ALSA built-ins */ + return + snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); +} + +static int snd_aicapcm_pcm_prepare(struct snd_pcm_substream + *substream) +{ + struct snd_card_aica *dreamcastcard = substream->pcm->private_data; + if ((substream->runtime)->format == SNDRV_PCM_FORMAT_S16_LE) + dreamcastcard->channel->sfmt = SM_16BIT; + dreamcastcard->channel->freq = substream->runtime->rate; + dreamcastcard->substream = substream; + return 0; +} + +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; + } + return 0; +} + +static unsigned long snd_aicapcm_pcm_pointer(struct snd_pcm_substream + *substream) +{ + return readl(AICA_CONTROL_CHANNEL_SAMPLE_NUMBER); +} + +static struct snd_pcm_ops snd_aicapcm_playback_ops = { + .open = snd_aicapcm_pcm_open, + .close = snd_aicapcm_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_aicapcm_pcm_hw_params, + .hw_free = snd_aicapcm_pcm_hw_free, + .prepare = snd_aicapcm_pcm_prepare, + .trigger = snd_aicapcm_pcm_trigger, + .pointer = snd_aicapcm_pcm_pointer, +}; + +/* TO DO: set up to handle more than one pcm instance */ +static int __init snd_aicapcmchip(struct snd_card_aica + *dreamcastcard, int pcm_index) +{ + struct snd_pcm *pcm; + int err; + /* AICA has no capture ability */ + err = + snd_pcm_new(dreamcastcard->card, "AICA PCM", pcm_index, 1, 0, &pcm); + if (unlikely(err < 0)) + return err; + pcm->private_data = dreamcastcard; + strcpy(pcm->name, "AICA PCM"); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_aicapcm_playback_ops); + /* Allocate the DMA buffers */ + err = + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data + (GFP_KERNEL), + AICA_BUFFER_SIZE, + AICA_BUFFER_SIZE); + return err; +} + +/* Mixer controls */ +static int aica_pcmswitch_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 aica_pcmswitch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0] = 1; /* TO DO: Fix me */ + return 0; +} + +static int aica_pcmswitch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + if (ucontrol->value.integer.value[0] == 1) + return 0; /* TO DO: Fix me */ + else + aica_chn_halt(); + return 0; +} + +static int aica_pcmvolume_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 = 0xFF; + return 0; +} + +static int aica_pcmvolume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; /* we've not yet been set up */ + ucontrol->value.integer.value[0] = dreamcastcard->channel->vol; + return 0; +} + +static int aica_pcmvolume_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = kcontrol->private_data; + if (unlikely(!dreamcastcard->channel)) + return -ETXTBSY; + if (unlikely(dreamcastcard->channel->vol == + ucontrol->value.integer.value[0])) + return 0; + dreamcastcard->channel->vol = ucontrol->value.integer.value[0]; + dreamcastcard->master_volume = ucontrol->value.integer.value[0]; + spu_memload(AICA_CHANNEL0_CONTROL_OFFSET, + (uint8_t *) dreamcastcard->channel, + sizeof(struct aica_channel)); + + return 1; +} + +static struct snd_kcontrol_new snd_aica_pcmswitch_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Switch", + .index = 0, + .info = aica_pcmswitch_info, + .get = aica_pcmswitch_get, + .put = aica_pcmswitch_put +}; + +static struct snd_kcontrol_new snd_aica_pcmvolume_control __devinitdata = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "PCM Playback Volume", + .index = 0, + .info = aica_pcmvolume_info, + .get = aica_pcmvolume_get, + .put = aica_pcmvolume_put +}; + +static int load_aica_firmware(void) +{ + int err; + const struct firmware *fw_entry; + err = 0; + spu_reset(); + err = request_firmware(&fw_entry, "aica_firmware.bin", &pd->dev); + if (unlikely(err)) + return err; + /* write firware into memory */ + spu_disable(); + spu_memload(0, fw_entry->data, fw_entry->size); + spu_enable(); + release_firmware(fw_entry); + return err; +} + +static int __devinit add_aicamixer_controls(struct snd_card_aica + *dreamcastcard) +{ + int err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmvolume_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + err = snd_ctl_add + (dreamcastcard->card, + snd_ctl_new1(&snd_aica_pcmswitch_control, dreamcastcard)); + if (unlikely(err < 0)) + return err; + return 0; +} + +static int snd_aica_remove(struct platform_device *devptr) +{ + struct snd_card_aica *dreamcastcard; + dreamcastcard = platform_get_drvdata(devptr); + if (unlikely(!dreamcastcard)) + return -ENODEV; + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + platform_set_drvdata(devptr, NULL); + return 0; +} + +static int __init snd_aica_probe(struct platform_device *devptr) +{ + int err; + struct snd_card_aica *dreamcastcard; + dreamcastcard = kmalloc(sizeof(struct snd_card_aica), GFP_KERNEL); + if (unlikely(!dreamcastcard)) + return -ENOMEM; + dreamcastcard->card = + snd_card_new(index, SND_AICA_DRIVER, THIS_MODULE, 0); + if (unlikely(!dreamcastcard->card)) { + kfree(dreamcastcard); + return -ENODEV; + } + strcpy(dreamcastcard->card->driver, "snd_aica"); + strcpy(dreamcastcard->card->shortname, SND_AICA_DRIVER); + strcpy(dreamcastcard->card->longname, + "Yamaha AICA Super Intelligent Sound Processor for SEGA Dreamcast"); + /* Load the PCM 'chip' */ + err = snd_aicapcmchip(dreamcastcard, 0); + if (unlikely(err < 0)) + goto freedreamcast; + snd_card_set_dev(dreamcastcard->card, &devptr->dev); + dreamcastcard->timer.data = 0; + dreamcastcard->channel = NULL; + /* Add basic controls */ + err = add_aicamixer_controls(dreamcastcard); + if (unlikely(err < 0)) + goto freedreamcast; + /* Register the card with ALSA subsystem */ + err = snd_card_register(dreamcastcard->card); + if (unlikely(err < 0)) + goto freedreamcast; + platform_set_drvdata(devptr, dreamcastcard); + aica_queue = create_workqueue("aica"); + if (unlikely(!aica_queue)) + goto freedreamcast; + snd_printk + ("ALSA Driver for Yamaha AICA Super Intelligent Sound Processor\n"); + return 0; + freedreamcast: + snd_card_free(dreamcastcard->card); + kfree(dreamcastcard); + return err; +} + +static struct platform_driver snd_aica_driver = { + .probe = snd_aica_probe, + .remove = snd_aica_remove, + .driver = { + .name = SND_AICA_DRIVER}, +}; + +static int __init aica_init(void) +{ + int err; + err = platform_driver_register(&snd_aica_driver); + if (unlikely(err < 0)) + return err; + pd = platform_device_register_simple(SND_AICA_DRIVER, -1, + aica_memory_space, 2); + if (unlikely(IS_ERR(pd))) { + platform_driver_unregister(&snd_aica_driver); + return PTR_ERR(pd); + } + /* Load the firmware */ + err = load_aica_firmware(); + if (unlikely(err < 0)) + return err; + + return 0; +} + +static void __exit aica_exit(void) +{ + /* Destroy the aica kernel thread */ + destroy_workqueue(aica_queue); + platform_device_unregister(pd); + platform_driver_unregister(&snd_aica_driver); + /* Kill any sound still playing and reset ARM7 to safe state */ + spu_reset(); +} + +module_init(aica_init); +module_exit(aica_exit); diff -Naur ./linux-2.6.21-orig/sound/sh/aica.h ./linux-2.6.21/sound/sh/aica.h --- ./linux-2.6.21-orig/sound/sh/aica.h 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/aica.h 2007-05-07 21:47:31.000000000 +0100 @@ -0,0 +1,80 @@ +/* aica.h + * Header file for ALSA driver for + * Sega Dreamcast Yamaha AICA sound + * Copyright Adrian McMenamin + * <adrian(a)mcmen.demon.co.uk> + * 2006 + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of version 2 of the GNU General Public License as published by + * the Free Software Foundation. + * + * 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 + * + */ + +/* SPU memory and register constants etc */ +#define G2_FIFO 0xa05f688c +#define SPU_MEMORY_BASE 0xA0800000 +#define ARM_RESET_REGISTER 0xA0702C00 +#define SPU_REGISTER_BASE 0xA0700000 + +/* AICA channels stuff */ +#define AICA_CONTROL_POINT 0xA0810000 +#define AICA_CONTROL_CHANNEL_SAMPLE_NUMBER 0xA0810008 +#define AICA_CHANNEL0_CONTROL_OFFSET 0x10004 + +/* Command values */ +#define AICA_CMD_KICK 0x80000000 +#define AICA_CMD_NONE 0 +#define AICA_CMD_START 1 +#define AICA_CMD_STOP 2 +#define AICA_CMD_VOL 3 + +/* Sound modes */ +#define SM_8BIT 1 +#define SM_16BIT 0 +#define SM_ADPCM 2 + +/* Buffer and period size */ +#define AICA_BUFFER_SIZE 0x8000 +#define AICA_PERIOD_SIZE 0x800 +#define AICA_PERIOD_NUMBER 16 + +#define AICA_CHANNEL0_OFFSET 0x11000 +#define AICA_CHANNEL1_OFFSET 0x21000 +#define CHANNEL_OFFSET 0x10000 + +#define AICA_DMA_CHANNEL 0 +#define AICA_DMA_MODE 5 + +#define SND_AICA_DRIVER "AICA" + +struct aica_channel { + uint32_t cmd; /* Command ID */ + uint32_t pos; /* Sample position */ + uint32_t length; /* Sample length */ + uint32_t freq; /* Frequency */ + uint32_t vol; /* Volume 0-255 */ + uint32_t pan; /* Pan 0-255 */ + uint32_t sfmt; /* Sound format */ + uint32_t flags; /* Bit flags */ +}; + +struct snd_card_aica { + struct snd_card *card; + struct aica_channel *channel; + struct snd_pcm_substream *substream; + int clicks; + int current_period; + struct timer_list timer; + int master_volume; + int dma_check; +}; diff -Naur ./linux-2.6.21-orig/sound/sh/Kconfig ./linux-2.6.21/sound/sh/Kconfig --- ./linux-2.6.21-orig/sound/sh/Kconfig 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Kconfig 2007-05-07 21:28:30.000000000 +0100 @@ -0,0 +1,14 @@ +# ALSA SH drivers + +menu "SUPERH devices" + depends on SND!=n && SUPERH + +config SND_AICA + tristate "Dreamcast Yamaha AICA sound" + depends on SH_DREAMCAST && SND + select SND_PCM + help + ALSA Sound driver for the SEGA Dreamcast console. + +endmenu + diff -Naur ./linux-2.6.21-orig/sound/sh/Makefile ./linux-2.6.21/sound/sh/Makefile --- ./linux-2.6.21-orig/sound/sh/Makefile 1970-01-01 01:00:00.000000000 +0100 +++ ./linux-2.6.21/sound/sh/Makefile 2007-05-07 21:34:10.000000000 +0100 @@ -0,0 +1,8 @@ +# +# Makefile for ALSA +# + +snd-aica-objs := aica.o + +# Toplevel Module Dependency +obj-$(CONFIG_SND_AICA) += snd-aica.o

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 12 May '07

12 May '07

o, I thought the epprom settings are default until the driver kicks in... So I set the jumpers accordingly and still no luck: IDESEL: 1-2 (Disabled - Its the CD IDE ON\OFF) JMPCFG: 1-2 (EPPROM ON) JMPB0: 1-2 (22XH) IDEIRQ: 1 (IRQ 15 - shouldn't make a different) JX1: 2-3 (Carbon MIC - again just letting you know though it shouldn't make a different) This should make sense with the picture I sent earlier... I also removed another card I had connected - ISA Modem, Just making sure their are no conflicts... Here's the grim results: sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1 FATAL: Error inserting snd_sb8 (/lib/modules/2.6.20-15-generic/kernel/sound/isa/sb/snd-sb8.ko): No such device P.S: Also an interesting note is that I now installed Win98 on top of that Dos and it worked too - It did also mention to use IRQ 11 so I also reserved in the bios, but again to no avail. Maybe I'm missing something ? Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] Mackie XD-2 Spike USB Drivers for Linux
by Sava Tatic 11 May '07

11 May '07

Hello, I have been trying to get the Mackie XD-2 Spike (http://www.mackie.com/products/spike/) to work on my Kubuntu Feisty machine, and so far I have only gotten it to play out (I had to use plughw instead of hw). I am using the generic USB drivers that came with the distro. I am having trouble recording anything on the card. Alsamixer (or Kmix) do not see the card. When I do alsamixer -c 1 (0 is my onboard card), it returns "no mixer elems found". I am aware that this card is not in the ALSA Matrix of supported cards, but I am wondering what would it take for it to become supported? All the best, Sava

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[alsa-devel] Capturing MIDI
by Steve Merrony 11 May '07

11 May '07

I hope this isn't too off-topic. I run an open source project (GENPO) which can be thought of as a kind of MIDI switchboard and filter. It uses ALSA throughout. I'd like to add a new feature to the software - a facility to simply record to a standard MIDI file anything that the program produces. I've searched though the API and I can't find anything documented which will help much, so here's my question... Is there a recommended library for C++ ALSA programs to use for handling (writing) standard MIDI files? Regards, Steve

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[alsa-devel] [PATCH2/5] ASoC S3C24xx machine drivers - Openmoko Neo1973
by Liam Girdwood 11 May '07

11 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds ALSA support for the Openmoko Neo1973 phone. Features:- * HiFi Playback and capture. * Phone calls supported. * Support for BT PCM in WM8753 voice interface. * Support for LM4857 audio amp. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH] ASoC AC97 device reg bugfix
by Liam Girdwood 11 May '07

11 May '07

This patch fixes a bug whereby AC97 bus device data was being clobbered when AC97 codecs using the generic ac97_codec.c driver were being registered. Codecs that didn't use the generic driver were unaffected (e.g. WM9712, WM9713). Changes:- o Add new AC97 codec class for custom (or need bus dev registration) AC97 codecs. o Only register/deregister this custom codec device with the AC97 bus. The generic AC97 driver already does this for generic codec devices. This may be related to bug #3038 :- https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3038 Manuel, can you try this on SH & Au1x00. Thanks Liam

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 11 May '07

11 May '07

Well, it didn't take long so here's the results : "sudo modprobe snd-sb8 port=0x220 irq=5 dma8=1" gave no errors, and "lsmod" later gave: Module Size Used by snd_sb8 9216 0 snd_sb8_dsp 12288 1 snd_sb8 snd_sb_common 17792 2 snd_sb8,snd_sb8_dsp snd_opl3_lib 11520 1 snd_sb8 snd_hwdep 9988 1 snd_opl3_lib snd_pcm_oss 44544 0 snd_pcm 79876 3 snd_sb8,snd_sb8_dsp,snd_pcm_oss snd_mixer_oss 17408 1 snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 32896 0 snd_seq_midi 9600 0 snd_rawmidi 25472 2 snd_sb8_dsp,snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 52592 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 23684 3 snd_opl3_lib,snd_pcm,snd_seq snd_seq_device 9100 6 snd_opl3_lib,snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54020 13 snd_sb8,snd_sb8_dsp,snd_sb_common,snd_opl3_lib,snd_hwdep,snd_pcm_oss,snd_pcm,snd_mixer_oss,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8672 1 snd snd_page_alloc 10888 1 snd_pcm nfs 240876 0 nfsd 218992 17 exportfs 6912 1 nfsd lockd 64904 3 nfs,nfsd sunrpc 161340 12 nfs,nfsd,lockd apm 22752 1 ppdev 10116 0 cpufreq_stats 7360 0 cpufreq_ondemand 9228 0 cpufreq_powersave 2688 0 freq_table 5792 2 cpufreq_stats,cpufreq_ondemand cpufreq_conservative 8200 0 cpufreq_userspace 5408 0 nls_iso8859_1 5120 1 nls_cp437 6784 1 vfat 14208 1 fat 53916 1 vfat ext2 66824 1 lp 12452 0 parport_pc 36388 1 parport 36936 3 ppdev,lp,parport_pc pcspkr 4224 0 ipv6 268704 8 tsdev 8768 0 evdev 11008 1 ext3 133128 1 jbd 59816 1 ext3 mbcache 9604 2 ext2,ext3 8139too 27648 0 ide_cd 32672 0 cdrom 37664 1 ide_cd ide_disk 17024 5 floppy 59524 0 ehci_hcd 34188 0 8139cp 25088 0 mii 6528 2 8139too,8139cp uhci_hcd 25360 0 usbcore 134280 3 ehci_hcd,uhci_hcd piix 10756 0 [permanent] generic 5124 0 [permanent] ata_generic 9092 0 libata 125720 1 ata_generic scsi_mod 142348 1 libata fbcon 42656 0 tileblit 3584 1 fbcon font 9216 1 fbcon bitblit 6912 1 fbcon softcursor 3200 1 bitblit vesafb 9220 0 capability 5896 0 commoncap 8192 1 capability But when I try to play an ogg gxine crushes an xmms ignores, when trying an mp3 gxine again crushes and xmms gives the soundcard no configured error... Just a recap: I verifyed using dos the settings (hwset and volset - default and save), and used the bios to set "IRQ 5" as "ISA Legacy". I didn't use the jumper as I see no need since the epprom already does the job. So I'm stuck, What do you think I should do now ? Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] detect extern/internel mic
by xinyun liu 10 May '07

10 May '07

It's interesting that when I plug in a headphone or an external microphone, the build in headphone and microphone will not work. Is it a driver feature or the hardware do it? Can anyone give some clue? I'd like to see the code but don't know how to start. Please give me help. Thx!

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Re: [alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 10 May '07

10 May '07

I have test all modes for alc883. The result is here: 0. 3stack-dig. both extern/internal can play, but can't record sound 1. 3stack-6ch-dig same with 3stack-dig 2. 3stack-6ch can play, but can't record sound. Can hear noise from the Mic. 3. 6stack-dig both can play sound, can't record sound 4. tarta-dig With alsamixer, can set to 2ch or 6ch, both can play sound, can't 5. tarta-2ch-dig both can play sound, can't record sound. 6. 6stack-dig-demo both can play sound, can't record sound. 7. aser both can play sound, can't record sound 8. medion can't play sound 9. laptop-eapd can play sound, but can't record sound 10. lenovo-101e both can play sound, can't record sound 11. auto both can play sound, and can record sound. But the sound quality is too bad to use from the internel Mic. After test it a while on switching the extern/internal Mic, the quality both droped. So I use lenovo-101e model, ( rebuild the driver, rmmod/insmod, use alsamixer to adjust the limits). Then I use auto mode, and the sound record work again from external Mic. After I switch to internel Mic, the quality droped again. Do you have any idea about it? Thx! 2007/4/30, Ustun ERGENOGLU <ustun.ergenoglu(a)gmail.com>: > I have a Fujitsu Siemens Amilo pa1510, with the same sound hardware, > and the same problem. I have posted some messages here, made a bug > report but no answer came :( i'm going hopeless. > > 2007/4/29, xinyun liu <xinyunliu.osc(a)gmail.com>: > > I know it's the system bios does not set the pin configuration right. > > But I don't know how to try out a correct model. > > > > Can anyone help to give some hints to fix it? > > > > > > Regards, > > Xinyun > > > > > > 2007/4/29, xinyun liu <xinyunliu.osc(a)gmail.com>: > > > hi, > > > I have an Acer Aspire 5100. I can play sound but with the build in > > > mic, I can't record sound. The driver is the latest HG code. > > > http://pastebin.ca/464406 > > > > > > [root@localhost ~]# lspci -nnv -d1002:437b > > > 00:14.2 0403: 1002:437b (rev 01) > > > Subsystem: 1025:009f > > > Flags: bus master, slow devsel, latency 64, IRQ 19 > > > Memory at c0000000 (64-bit, non-prefetchable) [size=16K] > > > Capabilities: [50] Power Management version 2 > > > Capabilities: [60] Message Signalled Interrupts: 64bit+ > > > Queue=0/0 Enable- > > > > > > > > > [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 > > > [root@localhost ~]# vi /tmp/1 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > > > codec_mask = 0xb > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' > > > is selected for config 1025:0 (Acer laptop) > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: > > > codec_mask = 0xb > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' > > > is selected for config 1025:0 (Acer laptop) > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x8, stream=0x1, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x8, stream=0x0, channel=0, format=0x0 > > > ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: > > > azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x4, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x3, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x5, stream=0x5, channel=0, format=0x11 > > > ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: > > > NID=0x2, stream=0x0, channel=0, format=0x0 > > > > > _______________________________________________ > > 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] alsa-driver-1.0.14rc3 won't compile with kernel 2.6.20.3 when Processor family is Core2
by Chris Pemberton 10 May '07

10 May '07

Laptop has Pentium Dual Core and Intel HD Audio Kernel is 2.6.20.3 compiled for Processor family Core2. In-kernel alsa drivers are 1.0.14rc3 (cat /proc/asound/version) and compile ok. In-kernel drivers do not work for this sound card. Downloaded 1.0.14rc3 and applied patch_realtek.c ./configure --prefix=/usr --with-cards=hda-intel "make" gives the following error: make[1]: Entering directory `/usr/src/linux-2.6.20.3' > /usr/src/linux-2.6.20.3/Makefile:491: > /usr/src/linux-2.6.20.3/arch//Makefile: No such file or directory If the kernel is compiled with Processor family = Pentium M, the external alsa drivers compile clean and the soundcard works. So, Processor family = Core2 cannot be selected when using external alsa-driver-1.0.14rc3. Chris Pemberton

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[alsa-devel] Problem with ALSA Capture and Playback : start_delay
by Nobin Mathew 10 May '07

10 May '07

Why start_delay=1 for capture and start_delay=0 for playback what this means I have one app which does both capture and playback, but when i tried to read ALSA buffer(capture) periodically it gives ALSa buffer overflow error. what my app does is read from ALSA (capture mic) and write to ALSA(playback). In this case for both Capture and Playback setting i am using start_delay = 0. But if i am giving some delay between Read and Write no overflow error happens Any idea why this happens Thanks

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Re: [alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 10 May '07

10 May '07

I already installed dos + the drivers and ubuntu, So I'll try working with that. As far as I can tell from the win95 driver name, the "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" is full-duplex : In : http://www.aztech.com/support_download_soundcard01.html : 2316FD95.ZIP 90,526 Windows 95 Full Duplex Audio Driver Version 4.8.0 Build 1128 I will try working with the software first, then probably either the epprom or the jumper (since I have dos + drivers working I'm tempted to try the epprom first ;-) ) I can't at the moment compile in that machine (My poor little xubuntu is barely working as it is, I don't even dare installing samba let alone a dev-env of any sorts..., maybe when I'll get everything working I'll install slackware, I decided gentoo is not a good idea ;-) ) so I won't be testing your drivers any time soon :-( , I do however hope you'll submit your drivers cause a non PnP driver can't hurt and waiting for confirms on such old hw will be point-less... When I'll have results I'll let you know... Also, I'm CCing the mailing list now as pair request... Thanks Rene, Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] [PATCH] snd-aoa-i2sbus: use MODULE_DEVICE_TABLE instead of plain MODULE_ALIAS
by Johannes Berg 10 May '07

10 May '07

This patch changes snd-aoa-i2sbus to use MODULE_DEVICE_TABLE instead of a hardcoded MODULE_ALIAS. Thanks to Sylvain Munaut for pointing this out. Signed-off-by: Johannes Berg <johannes(a)sipsolutions.net> --- Please apply to the alsa tree. sound/aoa/soundbus/i2sbus/i2sbus-core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) --- wireless-dev.orig/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:08:25.903457795 +0200 +++ wireless-dev/sound/aoa/soundbus/i2sbus/i2sbus-core.c 2007-03-28 13:09:21.253457795 +0200 @@ -23,9 +23,6 @@ MODULE_LICENSE("GPL"); MODULE_AUTHOR("Johannes Berg <johannes(a)sipsolutions.net>"); MODULE_DESCRIPTION("Apple Soundbus: I2S support"); -/* for auto-loading, declare that we handle this weird - * string that macio puts into the relevant device */ -MODULE_ALIAS("of:Ni2sTi2sC"); static int force; module_param(force, int, 0444); @@ -37,6 +34,8 @@ static struct of_device_id i2sbus_match[ { } }; +MODULE_DEVICE_TABLE(of, i2sbus_match); + static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, struct dbdma_command_mem *r, int numcmds)

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[alsa-devel] [PATCH] ASoC AC97 static GPL symbol fix
by Liam Girdwood 10 May '07

10 May '07

This patch fixes a bug whereby the GPL'ed symbol ac97_dai was being declared as static. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] Can someone verify support for "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" ?
by ramkromberg＠mail.com 10 May '07

10 May '07

Hello alsa devs, I have an old "AzTech Sound Galaxy Nova 16 Extra II-3D - using AZT-2316/R Chipset - FCC-ID:I38-MMSN846" which I know to work on windows 2000 natively and on other windows using drivers from http://www.aztech.com/support_download_soundcard01.html, however, I was unsuccessful in using it in linux ( I tried many dists, but my most recent failure was with xubuntu 7.04 which have the most recent alsa). I believe the card is unsupported in alsa and I was hoping someone might be free to help me with this legacy card ? Since the card is unsupported I figured this should go the devs. * Their is this piece of code which might be relevant: http://www.gelato.unsw.edu.au/lxr/source/sound/isa/azt2320.c 21 This driver should provide support for most Aztech AZT2320 based cards. 22 Several AZT2316 chips are also supported/tested, but autoprobe doesn't 23 work: all module option have to be set. * I'm unsure of what it means to configure manually or how to do so... I do however have the card working currently with windows 2000 so if its a question of IRQ and DMA, I can find those with relative ease. * I obviously tried modprobeing the snd-sgalaxy driver/module, with no success - possibly related ? Thanks Ram Kromberg ramkromberg(a)mail.com

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[alsa-devel] Sound on SMDK2410
by Jose Henrique Spahn Torres 10 May '07

10 May '07

Hi all, I am trying to get sound working on a SMDK-2410 evaluation board. I compiled kernel 2.6.20 with alsa-driver-1.0.14rc4. However, I could not find the specific uda-1341TS driver for this board. I would like to know the status of this driver, if there will be one some time soon. Thanks, Henrique

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Re: [alsa-devel] [PATCH] SH7760 ASoC support
by Manuel Lauss 10 May '07

10 May '07

On Thu, May 10, 2007 at 09:58:17PM +0900, Paul Mundt wrote: > On Thu, May 10, 2007 at 02:08:33PM +0200, Manuel Lauss wrote: > > +## > > +## Audio unit modules > > +## > > + > > +config SND_SOC_SH4_HAC > > + select AC97_BUS > > + select SND_SOC_AC97_BUS > > + select SND_AC97_CODEC > > + tristate > > + > > +config SND_SOC_SH4_SSI > > + tristate > > + > This might seem like a silly question, but how are these two supposed to > be enabled? At the very least you should have some dependencies on the > CPU subtypes that support these blocks. > > > + > > + > > +## > > +## Boards > > +## > > + > Or are you expecting that the board-specific code will simply select > these? That was the idea; the other ASoC boards do similar things. > Are you planning on submitting a stub that enables these? Or is this just > to be a placeholder until some board wants them? I suppose I can submit my AC97 board code, it does not depend on anything that is not already submitted. I'll add the ac97 board code to the patch and re-submit it. Thanks, Manuel Lauss

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[alsa-devel] [PATCH5/5] ASoC S3C24xx machine drivers - Kconfig
by Liam Girdwood 10 May '07

10 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds Kconfig and build support for the Neo1973, SMDK2443 and S3C2443 AC97 ALSA audio drivers. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH4/5] ASoC S3C24xx machine drivers - I2C ID for LM4857
by Liam Girdwood 10 May '07

10 May '07

This patch adds I2C ID for the LM4857 audio amp and corrects the spacing of the WM8731, WM8750 and WM8753 ID's. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH3/5] ASoC S3C24xx machine drivers - SMDK 2443
by Liam Girdwood 10 May '07

10 May '07

From: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> This patch adds ALSA support for the SMDK2443 reference board. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH] SH7760 ASoC support
by Manuel Lauss 10 May '07

10 May '07

Hello, here goes round 2 of the SH7760 ASoC patch. Changes since last submission: - incorporate Paul Mundt's suggestions - work around an oops when aplay is used in mmap mode ('aplay -M') by changing the allocation type to SNDRV_DMA_TYPE_CONTINUOUS Thanks, Manuel Lauss --- ALSA ASoC support for SH7760 - AC97 driver for HAC unit(s) found on SH7760/SH7780 - I2S driver for SSI unit(s) found on SH7760/SH7780 - SH7760 DMA engine (dmabrg) sound/soc/Kconfig | 1 + sound/soc/Makefile | 3 +- sound/soc/sh/Kconfig | 30 ++++ sound/soc/sh/Makefile | 11 ++ sound/soc/sh/dma-sh7760.c | 350 +++++++++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 323 ++++++++++++++++++++++++++++++++++++ sound/soc/sh/ssi.c | 400 +++++++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1117 insertions(+), 1 deletions(-) Signed-off-by: Manuel Lauss <mano(a)roarinelk.homelinux.net> diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dccaa4b..3e3df26 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,6 +26,7 @@ menu "SoC Platforms" depends on SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/sh/Kconfig" endmenu # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 98e6f49..e1c9a70 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ sh/ + diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 0000000..6c90c2c --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,30 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 0000000..32bf713 --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,11 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 0000000..d495cf5 --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,350 @@ +/* + * SH7760 ("camelot") DMABRG audio DMA unit support + * + * Copyright (C) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which + * trigger an interrupt when one half of the programmed transfer size + * has been xmitted. + * + * FIXME: little-endian only for now + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dmabrg.h> + + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in e.g. MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + pr_debug("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + pr_debug("PCM data: addr 0x%08lx len %d\n", + (u32)runtime->dma_addr, runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = runtime->dma_area; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = runtime->dma_area; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * position at the time the IRQ was triggered, which results in + * fast-playback sound in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, GFP_KERNEL, + DMABRG_PREALLOC_BUFFER, DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-pcm", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 0000000..696ecb8 --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,323 @@ +/* + * Hitachi Audio Controller (AC97) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable HAC output pins! + */ + +/* + * BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only + * the FIRST can be used since ASoC does not pass any information to the + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/wait.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "hac: reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + pr_debug("hac: invalid depth %d bit\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 0000000..d69bf63 --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,400 @@ +/* + * Serial Sound Interface (I2S) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable SSI output pins! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK (3 << CR_CHNL_SHIFT) +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK (7 << CR_DWL_SHIFT) +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK (7 << CR_SWL_SHIFT) +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_SHIFT 4 +#define CR_CKDIV_MASK (7 << CR_CKDIV_SHIFT) /* bitclock divider */ +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + pr_debug("ssi: already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + pr_debug("ssi_hw_params() enter\nssicr was %08lx\n", ssicr); + pr_debug("bits: %d channels: %d\n", bits, channels); + + ssicr &= ~(CR_TRMD | CR_CHNL_MASK | CR_DWL_MASK | CR_PDTA | + CR_SWL_MASK); + + /* direction (send/receive) */ + if (!recv) + ssicr |= CR_TRMD; /* transmit */ + + /* channels */ + if ((channels < 2) || (channels > 8) || (channels & 1)) { + pr_debug("ssi: invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + pr_debug("ssi: invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + pr_debug("ssi_hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~CR_CKDIV_MASK; + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + pr_debug("ssi: invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + pr_debug("ssi_set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA | CR_BREN | CR_SWSP | CR_SCKP | + CR_SWS_MASTER | CR_SCK_MASTER); + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + pr_debug("ssi: unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + pr_debug("ssi: invalid inversion\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + pr_debug("ssi: invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + pr_debug("ssi_set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>");

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[alsa-devel] Mic and Line In Capture AC97 slots issue
by Nobin Mathew 10 May '07

10 May '07

http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/000835.html http://mailman.alsa-project.org/pipermail/alsa-devel/2007-May/000837.html the overrun issue was because my application was not fast enough. I resolved that problem by disabling capture of AC-Link slot4 in my ac97 controller. So capture happens only in slot3. According to ac97 spec Mic comes in Slot6 of AC-Link. I am using wolfson wm9713 codec. In this LineR and LineL are routed to LeftADC and RightADC. And Mic1 (Mic A) is also coonected to LeftADC and RightADC. So i am assuming that Mic 1 will come in Slot3 and Slot4. But my application is taking only mono data, so XRUN will happen. What is the correct configuration for this? What should be the configuration if i want to capture both Line IN and Mic? means AC-Link settings Thanks

2 1

[alsa-devel] [PATCH] snd-usb-audio: Another Logitech QuickCam ID
by Daniel Drake 10 May '07

10 May '07

This patch adds the ID for another quickcam microphone, reported by freqmod on ALSA ticket #0003040 I'm going to submit a USB patch separately to provide a macro to simplify these entries, as suggested by Alan Stern. We could switch to using that in future. 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 @@ -49,6 +49,15 @@ USB_DEVICE_ID_MATCH_INT_CLASS | USB_DEVICE_ID_MATCH_INT_SUBCLASS, .idVendor = 0x046d, + .idProduct = 0x0850, + .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] Recording stops after a short while on ATI SB450
by Kjeld Flarup 10 May '07

10 May '07

Hello This list first got to my attention after I filed a bug https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3064 The short story is that then using arecord the input stops after a short while. The only bug report I get is this when starting to capture. hda-intel: Invalid position buffer, using LPIB read method instead. I'm actually on the edge of buying a USB headset for this computer. I unfortunately only have it for 2½ weeks, then my mother in law is going to have it. Meanwhile I can test almost anything if anybody has an idea. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Forssavej 49, 7600 Struer, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 1

[alsa-devel] VIA 8237 - random switch to 8kHz sample rate?
by dustin＠seznam.cz 09 May '07

09 May '07

Hello, Occasionally my integrated VIA 8237 + AD1888 stops transmitting SPDIF AC3 (non-audio) streams. When troubleshoting, aplay -v -Dplughw:0 44.1kHz.wav shows in the actual card info section sample rate of 8kHz. The sound via DAC reflects this - no trebles. Logically, my AVR cannot detect 8kHz SPDIF a simply mutes. Since there is no sample rate mixer control, the only remedy is to reload corresponding alsa modules. The source via82xx.c uses 48kHz only. Please would you point me at some direction to watch, check, debug, etc.? It happens very rarely, I do not know how to reproduce it yet. Thanks a lot, Pavel.

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[alsa-devel] Fwd: ALSA problems with simultaneous Capture and playback
by Nobin Mathew 09 May '07

09 May '07

---------- Forwarded message ---------- From: Nobin Mathew <nobin.mathew(a)gmail.com> Date: May 9, 2007 11:57 PM Subject: Re: [alsa-devel] ALSA problems with simultaneous Capture and playback To: Lee Revell <rlrevell(a)joe-job.com> when I do capture and Playback independently for any channel configurations and any sampling rate configuration it works perfectly. Following configuration of simutaneous playback and capture work. loop "line out" and "line in" externaly using stereo jack. start arecord and then start aplay. Then it records properly. The application which is giving erorr is custom application which sets params properly. if i run "aplay" and "arecord" first (after boot) and then i run my application then it works properly. reverse is not working (i.e. running my app first destroys whole ALSA, then even arecord and aplay is not working). On 5/9/07, Lee Revell <rlrevell(a)joe-job.com> wrote: > On 5/9/07, Nobin Mathew <nobin.mathew(a)gmail.com> wrote: > > One read and write handle 256 frames. > > > > Write seems to be OK, but read fails > > > > State 4 is SND_PCM_STATE_XRUN > > > > why this XRUN is happening? > > If your hardware buffer is small you may need to run your application > with realtime priority (SCHED_FIFO) to avoid xruns. > > I guess this is a custom driver? Please post a link to the source code. > > Lee >

1 0

[alsa-devel] ALSA problems with simultaneous Capture and playback
by Nobin Mathew 09 May '07

09 May '07

I am facing a problem with simultaneous playback and Capture. I am doing Mic Capture and Playing it back. this is format i am using SND_PCM_FORMAT_S16_LE. I am recording only one channel "Mono". this is my driver pcm_hardware configuration static struct snd_pcm_hardware pcm_hw_info = { .info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_RESUME, /* * ALSA doesn't support 18-bit or 20-bit packed into 32-bit * words. It also doesn't support 12-bit at all. */ .formats = SNDRV_PCM_FMTBIT_S16_LE, /* should this be continuous or knot? */ .buffer_bytes_max = 64 * 1024, .period_bytes_min = 1024, .period_bytes_max = 1024, .periods_min = 4, .periods_max = 64, }; I am getting following errors when i do simultaneous capture and playback. Start the while looop MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsaTrying to free free IRQ27 512 256 ReadFromAlsa():State:4, iiLen=2048 ReadFromAlsa():State:2 Overflow state MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsaTrying to free free IRQ27 512 256 ReadFromAlsa():State:4, iiLen=1280 ReadFromAlsa():State:2 Overflow state MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsa 512 256 MY WriteToAlsa 512 256 MY ReadFromAlsaTrying to free free IRQ27 512 256 ReadFromAlsa():State:4, iiLen=1280 ReadFromAlsa():State:2 Overflow state One read and write handle 256 frames. Write seems to be OK, but read fails State 4 is SND_PCM_STATE_XRUN why this XRUN is happening? any idea why this is coming? where i am going wrong? Thanks

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[alsa-devel] FIXME for Rear Channels of CMI8738, how do i write a softvol to fix this defect?
by John L. Utz III 09 May '07

09 May '07

Hello; the CMI8738 allows for 4 channel output by providing a second DAC, but sadly the hardware has no volume control for said second DAC. A review of /usr/share/alsa/cards/CMI8738-MC6.conf shows that the implementor of said file was aware of the need to fix the problem: # 2nd DAC # FIXME: we need a volume attenuator for rear channel. CMI8738-MC6.pcm.rear.0 { @args [ CARD ] @args.CARD { type string } type hw card $CARD device 1 } I'd really like to fix this! Is it fixable by creating a softvol plugin? http://www.alsa-project.org/alsa-doc/alsa-lib/pcm_plugins.html#pcm_plugins if so, can i have a hint? :-) so, if i have to invoke speaker test with -Dhw:0,1 to get 4 channel out, what does that tell me about how i need to configure a softvol for this item? Or am i missing the point and it's actually the *driver* that needs to be doctored up to provide a softvol implementation. If so, how? tnx! johnu

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[alsa-devel] Realtek: ALC265/268 Support?
by Henry Walpool 09 May '07

09 May '07

Thought I checked my specs but think I missed the mark. I have a new Toshiba Laptop A205-S4577 Laptop and the laptop has a realtek high definition card. The hda-intel driver detects the card but has no sound and only two output controls in alsamixer nothing else. I have tried the latest release candidate ALSA 1.0.14rc4 without any luck. Is this card supported. Details below? Distribution: Kubantu Feisty 7.04, Kernel 2.6.20-15-generic, and used a custom 2.6.20.1 kernel and tried the lastest release drivers. Realtek ID 268 $ lspci -v -v 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 0 2) Subsystem: Toshiba America Info Systems Unknown device ff10 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PER R- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 21 Region 0: Memory at f0b40000 (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 $ lspci -v -n 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1179:ff10 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at f0b40000 (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 $ cat /proc/asound/card0/codec#0 Codec: Realtek ID 268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff10 Revision Id: 0x100003 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: [0x01 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: [0x40 0x40] 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 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown 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 0x02211020: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0x20: IN 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: [0x02 0x02] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x02a11830: [Jack] Mic at Ext Front Conn = 1/8, Color = Black 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 0x99a30940: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: 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 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black 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 0x4015812d: [N/A] Speaker at Ext N/A Conn = Optical, Color = Purple 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: [0x01 0x01] 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: [0x14 0x14] Connection: 7 0x18* 0x19 0x1a 0x1c 0x14 0x15 0x13 $ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf0b40000 irq 21 Thanks!

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[alsa-devel] Fwd: Fwd: [ASoC] Mic record is not working with wm9713 ASoC driver
by Nobin Mathew 09 May '07

09 May '07

Forwarded Conversation Subject: [ASoC] Mic record is not working with wm9713 ASoC driver ------------------------ From: Nobin Mathew <nobin.mathew(a)gmail.com> To: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 2:48 PM I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". My settings 'Mic A Source' -> "Mic 1" 'Mic Boost (+20dB)' [on] 'Capture' [on] 'Left Capture Source' "Mic 1" 'Right Capture Source' "Mic 1" i am issuing following command to record "arecord -M -f S16_LE sound.wav" But what i found is driver is not enabling "Mic bias" and "Mic A PGA" this dump of /sys/devices/platform/soc-audio/dapm_widget while "arecord" is running is given below Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: D0 In idea why this is happening? what is going wrong with my settings? Nobin Mathew -------- From: Andrew Morton <akpm(a)linux-foundation.org> To: Nobin Mathew <nobin.mathew(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org, Jaroslav Kysela <perex(a)suse.cz>, Takashi Iwai <tiwai(a)suse.de> Date: Wed, May 9, 2007 at 3:51 PM [Quoted text hidden](maintainers cc'ed) -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 3:55 PM On Wed, 2007-05-09 at 14:48 +0530, Nobin Mathew wrote: > I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". I've replied to this duplicate post on alsa-dev. Liam -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Cc: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 4:14 PM Liam Sorry for disturbing you again, I could not find that reply on alsa devel mailing list Can u send that to me Thanks [Quoted text hidden] -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Date: Wed, May 9, 2007 at 4:22 PM The mail server is probably a little behind..... give it about 30 mins... Liam [Quoted text hidden]> - > To unsubscribe from this list: send the line "unsubscribe linux-kernel" in > the body of a message to majordomo(a)vger.kernel.org > More majordomo info at http://vger.kernel.org/majordomo-info.html > Please read the FAQ at http://www.tux.org/lkml/ -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Date: Wed, May 9, 2007 at 4:25 PM Attachments: identity_wm9713.c Liam i got the mail. I am not setting any audio paths in my machine driver [Quoted text hidden] -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 4:28 PM Attachments: identity_wm9713.c [Quoted text hidden] -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Date: Wed, May 9, 2007 at 5:00 PM Liam i have few more doubts i have only Mic1, i am not connecting to Mic2A and Mic 2B So can i remove audio path to Mic2A and Mic2B means my audio path will be like static const char* audio_map[][3] = { /* mic is connected to mic1 - with bias */ {"MIC1", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 1"}, #if 0 /* mic is connected to mic2A - with bias */ {"MIC2A", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 2"}, /* mic is connected to mic2B - with bias */ {"MIC2B", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 3"}, #endif {NULL, NULL, NULL}, }; instead of static const char* audio_map[][3] = { /* mic is connected to mic1 - with bias */ {"MIC1", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 1"}, /* mic is connected to mic2A - with bias */ {"MIC2A", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 2"}, /* mic is connected to mic2B - with bias */ {"MIC2B", NULL, "Mic Bias"}, {"Mic Bias", NULL, "Mic 3"}, {NULL, NULL, NULL}, }; what is the meaning of these lines, what is RXP and RXN /* set up mainstone codec pins */ // snd_soc_dapm_set_endpoint(codec, "RXP", 0); // snd_soc_dapm_set_endpoint(codec, "RXN", 0); snd_soc_dapm_set_endpoint(codec, "MIC1", 0); [Quoted text hidden] -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Date: Wed, May 9, 2007 at 5:30 PM On Wed, 2007-05-09 at 17:00 +0530, Nobin Mathew wrote: > Liam i have few more doubts > > i have only Mic1, i am not connecting to Mic2A and Mic 2B > So can i remove audio path to Mic2A and Mic2B > means > my audio path will be like > static const char* audio_map[][3] = { > > /* mic is connected to mic1 - with bias */ > {"MIC1", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 1"}, > #if 0 > /* mic is connected to mic2A - with bias */ > {"MIC2A", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 2"}, > /* mic is connected to mic2B - with bias */ > {"MIC2B", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 3"}, > #endif > > {NULL, NULL, NULL}, > }; > This is fine, it's safe to remove unused Mic's. > instead of > > static const char* audio_map[][3] = { > > /* mic is connected to mic1 - with bias */ > {"MIC1", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 1"}, > /* mic is connected to mic2A - with bias */ > {"MIC2A", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 2"}, > /* mic is connected to mic2B - with bias */ > {"MIC2B", NULL, "Mic Bias"}, > {"Mic Bias", NULL, "Mic 3"}, > > {NULL, NULL, NULL}, > }; > > > what is the meaning of these lines, what is RXP and RXN > > /* set up mainstone codec pins */ > // snd_soc_dapm_set_endpoint(codec, "RXP", 0); > // snd_soc_dapm_set_endpoint(codec, "RXN", 0); > snd_soc_dapm_set_endpoint(codec, "MIC1", 0); RXP and RXN are from a different codec and should be removed. Also remove the MIC1 endpoint setting to 0 as this will disconnect it. Comments from driver :- static int identity_wm9713_ac97_init(struct snd_soc_codec *codec) { struct platform_controller *controller = codec->ac97->bus->private_data; struct aaci *aaci = (struct aaci *)controller->bus_controller; snd_ac97_pcm_assign(codec->ac97->bus, 3, ac97_defs); aaci->ac97_bus = codec->ac97->bus; /* * * Disable AC97 PC Beep input on audio codecs. * */ if (ac97_is_audio(codec->ac97)) snd_ac97_write_cache(codec->ac97, AC97_PC_BEEP, 0x801e); return 0; } AC97 beep should be disabled in userspace. e.g. alsa configuration file or by changing mixers settings (alsactl can save and restore settings). unsigned int identity_wm9713_config_sysclk(struct snd_soc_pcm_runtime *rtd, struct snd_soc_clock_info *info) { return rtd->codec_dai->config_sysclk(rtd->codec_dai, info, 12187500); } This function has been removed in current ASoC versions. It's best to upgrade from kernel.org or from here :- http://opensource.wolfsonmicro.com/cgi-bin/gitweb/gitweb.cgi?p=linux-2.6-as… Liam -------- From: Nobin Mathew <nobin.mathew(a)gmail.com> To: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Date: Wed, May 9, 2007 at 5:30 PM Liam Thanks of lot i got it working Cheers [Quoted text hidden] -------- From: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> To: Nobin Mathew <nobin.mathew(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org Date: Wed, May 9, 2007 at 5:38 PM It's probably best to keep this thread on alsa-dev otherwise things start to get confusing when cross posting occurs. Liam --------

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[alsa-devel] [ASoC] Mic record is not working with wm9713 ASoC driver
by Nobin Mathew 09 May '07

09 May '07

I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". My settings 'Mic A Source' -> "Mic 1" 'Mic Boost (+20dB)' [on] 'Capture' [on] 'Left Capture Source' "Mic 1" 'Right Capture Source' "Mic 1" i am issuing following command to record "arecord -M -f S16_LE sound.wav" But what i found is driver is not enabling "Mic bias" and "Mic A PGA" this dump of /sys/devices/platform/soc-audio/dapm_widget while "arecord" is running is given below Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: D0 In idea why this is happening? what is going wrong with my settings? Nobin Mathew

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[alsa-devel] [RFC PATCH] ASoC AC97/I2S for SH7760
by Manuel Lauss 09 May '07

09 May '07

ALSA ASoC support for SH7760 - SH7760 DMA (requires DMABRG patch sent to linux-sh list) - AC97 support for HAC unit found on SH7760/SH7780 - I2S support for SSI unit found on SH7760/SH7780 sound/soc/Kconfig | 1 sound/soc/Makefile | 3 sound/soc/sh/Kconfig | 30 +++ sound/soc/sh/Makefile | 11 + sound/soc/sh/dma-sh7760.c | 361 ++++++++++++++++++++++++++++++++++++++++ sound/soc/sh/hac.c | 335 +++++++++++++++++++++++++++++++++++++ sound/soc/sh/ssi.c | 413 ++++++++++++++++++++++++++++++++++++++++++++++ 7 files changed, 1153 insertions(+), 1 deletion(-) Comments welcome! Thanks, Manuel Lauss diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index dccaa4b..3e3df26 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -26,6 +26,7 @@ menu "SoC Platforms" depends on SND_SOC source "sound/soc/at91/Kconfig" source "sound/soc/pxa/Kconfig" +source "sound/soc/sh/Kconfig" endmenu # Supported codecs diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 98e6f49..e1c9a70 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,4 +1,5 @@ snd-soc-core-objs := soc-core.o soc-dapm.o obj-$(CONFIG_SND_SOC) += snd-soc-core.o -obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ +obj-$(CONFIG_SND_SOC) += codecs/ at91/ pxa/ sh/ + diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig new file mode 100644 index 0000000..6c90c2c --- /dev/null +++ b/sound/soc/sh/Kconfig @@ -0,0 +1,30 @@ +menu "SoC Audio support for SuperH" + +config SND_SOC_PCM_SH7760 + tristate "SoC Audio support for Renesas SH7760" + depends on CPU_SUBTYPE_SH7760 && SND_SOC + select SH_DMABRG + help + Enable this option for SH7760 AC97/I2S audio support. + + +## +## Audio unit modules +## + +config SND_SOC_SH4_HAC + select AC97_BUS + select SND_SOC_AC97_BUS + select SND_AC97_CODEC + tristate + +config SND_SOC_SH4_SSI + tristate + + + +## +## Boards +## + +endmenu diff --git a/sound/soc/sh/Makefile b/sound/soc/sh/Makefile new file mode 100644 index 0000000..32bf713 --- /dev/null +++ b/sound/soc/sh/Makefile @@ -0,0 +1,11 @@ +## DMA engines +snd-soc-dma-sh7760-objs := dma-sh7760.o +obj-$(CONFIG_SND_SOC_PCM_SH7760) += snd-soc-dma-sh7760.o + +## audio units found on some SH-4 +snd-soc-hac-objs := hac.o +snd-soc-ssi-objs := ssi.o +obj-$(CONFIG_SND_SOC_SH4_HAC) += snd-soc-hac.o +obj-$(CONFIG_SND_SOC_SH4_SSI) += snd-soc-ssi.o + +## boards diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c new file mode 100644 index 0000000..a2aa1a8 --- /dev/null +++ b/sound/soc/sh/dma-sh7760.c @@ -0,0 +1,361 @@ +/* + * SH7760 ("camelot") DMABRG audio DMA unit support + * + * Copyright (C) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * The SH7760 DMABRG provides 4 dma channels (2x rec, 2x play), which + * trigger an interrupt when one half of the programmed transfer size + * has been xmitted. + * + * FIXME: little-endian only for now + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/platform_device.h> +#include <linux/dma-mapping.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <asm/dmabrg.h> +#include <asm/io.h> + +/* #define PCM_DEBUG */ + +#define MSG(x...) printk(KERN_INFO "sh7760-pcm: " x) + +#ifdef PCM_DEBUG +#define DBG(x...) MSG(x) +#else +#define DBG(x...) do {} while (0) +#endif + +/* registers and bits */ +#define BRGATXSAR 0x00 +#define BRGARXDAR 0x04 +#define BRGATXTCR 0x08 +#define BRGARXTCR 0x0C +#define BRGACR 0x10 +#define BRGATXTCNT 0x14 +#define BRGARXTCNT 0x18 + +#define ACR_RAR (1 << 18) +#define ACR_RDS (1 << 17) +#define ACR_RDE (1 << 16) +#define ACR_TAR (1 << 2) +#define ACR_TDS (1 << 1) +#define ACR_TDE (1 << 0) + +/* receiver/transmitter data alignment */ +#define ACR_RAM_NONE (0 << 24) +#define ACR_RAM_4BYTE (1 << 24) +#define ACR_RAM_2WORD (2 << 24) +#define ACR_TAM_NONE (0 << 8) +#define ACR_TAM_4BYTE (1 << 8) +#define ACR_TAM_2WORD (2 << 8) + + +struct camelot_pcm { + unsigned long mmio; /* DMABRG audio channel control reg MMIO */ + unsigned int txid; /* ID of first DMABRG IRQ for this unit */ + + struct snd_pcm_substream *tx_ss; + unsigned long tx_period_size; + unsigned int tx_period; + + struct snd_pcm_substream *rx_ss; + unsigned long rx_period_size; + unsigned int rx_period; + +} cam_pcm_data[2] = { + { + .mmio = 0xFE3C0040, + .txid = DMABRGIRQ_A0TXF, + }, + { + .mmio = 0xFE3C0060, + .txid = DMABRGIRQ_A1TXF, + }, +}; + +#define BRGREG(x) (*(unsigned long *)(cam->mmio + (x))) + +/* + * set a minimum of 16kb per period, to avoid interrupt-"storm" and + * resulting skipping. In general, the bigger the minimum size, the + * better for overall system performance. (The SH7760 is a puny CPU + * with a slow SDRAM interface and poor internal bus bandwidth, + * *especially* when the LCDC is active). The minimum for the DMAC + * is 8 bytes; 16kbytes are enough to get skip-free playback of + * 44kHz/16bit/stereo MP3 on a lightly loaded system, and maintain + * reasonable responsiveness in e.g. MPlayer. + */ +#define DMABRG_PERIOD_MIN 16 * 1024 +#define DMABRG_PERIOD_MAX 0x03fffffc +#define DMABRG_PREALLOC_BUFFER 32 * 1024 +#define DMABRG_PREALLOC_BUFFER_MAX 32 * 1024 + +/* support everything the SSI supports */ +#define DMABRG_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define DMABRG_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +static struct snd_pcm_hardware camelot_pcm_hardware = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = DMABRG_FMTS, + .rates = DMABRG_RATES, + .rate_min = 8000, + .rate_max = 192000, + .channels_min = 2, + .channels_max = 8, /* max of the SSI */ + .buffer_bytes_max = DMABRG_PERIOD_MAX, + .period_bytes_min = DMABRG_PERIOD_MIN, + .period_bytes_max = DMABRG_PERIOD_MAX / 2, + .periods_min = 2, + .periods_max = 2, + .fifo_size = 128, +}; + +static void camelot_txdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->tx_period ^= 1; + snd_pcm_period_elapsed(cam->tx_ss); +} + +static void camelot_rxdma(void *data) +{ + struct camelot_pcm *cam = data; + cam->rx_period ^= 1; + snd_pcm_period_elapsed(cam->rx_ss); +} + +static int camelot_pcm_open(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret, dmairq; + + snd_soc_set_runtime_hwparams(substream, &camelot_pcm_hardware); + + /* DMABRG buffer half/full events */ + dmairq = (recv) ? cam->txid + 2 : cam->txid; + if (recv) { + cam->rx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_rxdma, cam); + if (unlikely(ret)) { + MSG("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1,camelot_rxdma, cam); + } else { + cam->tx_ss = substream; + ret = dmabrg_request_irq(dmairq, camelot_txdma, cam); + if (unlikely(ret)) { + MSG("audio unit %d irqs already taken!\n", + rtd->dai->cpu_dai->id); + return -EBUSY; + } + (void)dmabrg_request_irq(dmairq + 1, camelot_txdma, cam); + } + return 0; +} + +static int camelot_pcm_close(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int dmairq; + + dmairq = (recv) ? cam->txid + 2 : cam->txid; + + if (recv) + cam->rx_ss = NULL; + else + cam->tx_ss = NULL; + + dmabrg_free_irq(dmairq + 1); + dmabrg_free_irq(dmairq); + + return 0; +} + +static int camelot_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + int ret; + + ret = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (ret < 0) + return ret; + + if (recv) { + cam->rx_period_size = params_period_bytes(hw_params); + cam->rx_period = 0; + } else { + cam->tx_period_size = params_period_bytes(hw_params); + cam->tx_period = 0; + } + return 0; +} + +static int camelot_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int camelot_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + + DBG("PCM data: addr 0x%08lx len %d\n", (u32)runtime->dma_addr, + runtime->dma_bytes); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + BRGREG(BRGATXSAR) = runtime->dma_addr; + BRGREG(BRGATXTCR) = runtime->dma_bytes; + } else { + BRGREG(BRGARXDAR) = runtime->dma_addr; + BRGREG(BRGARXTCR) = runtime->dma_bytes; + } + + return 0; +} + +static inline void dmabrg_play_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: XFER start, auto-addr-reload */ + BRGREG(BRGACR) = acr | ACR_TDE | ACR_TAR | ACR_TAM_2WORD; +} + +static inline void dmabrg_play_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data transmission */ + BRGREG(BRGACR) = acr | ACR_TDS; +} + +static inline void dmabrg_rec_dma_start(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* start DMABRG engine: recv start, auto-reload */ + BRGREG(BRGACR) = acr | ACR_RDE | ACR_RAR | ACR_RAM_2WORD; +} + +static inline void dmabrg_rec_dma_stop(struct camelot_pcm *cam) +{ + unsigned long acr = BRGREG(BRGACR) & ~(ACR_TDS | ACR_RDS); + /* forcibly terminate data receiver */ + BRGREG(BRGACR) = acr | ACR_RDS; +} + +static int camelot_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (recv) + dmabrg_rec_dma_start(cam); + else + dmabrg_play_dma_start(cam); + break; + case SNDRV_PCM_TRIGGER_STOP: + if (recv) + dmabrg_rec_dma_stop(cam); + else + dmabrg_play_dma_stop(cam); + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t camelot_pos(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct camelot_pcm *cam = &cam_pcm_data[rtd->dai->cpu_dai->id]; + int recv = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0:1; + unsigned long pos; + + /* cannot use the DMABRG pointer register: under load, by the + * time ALSA comes around to read the register, it is already + * far ahead (or worse, already done with the fragment) of the + * time the IRQ was triggered, which gives fast-playback sound + * in my test application (ScummVM) + */ + if (recv) + pos = cam->rx_period ? cam->rx_period_size : 0; + else + pos = cam->tx_period ? cam->tx_period_size : 0; + + return bytes_to_frames(runtime, pos); +} + +static struct snd_pcm_ops camelot_pcm_ops = { + .open = camelot_pcm_open, + .close = camelot_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = camelot_hw_params, + .hw_free = camelot_hw_free, + .prepare = camelot_prepare, + .trigger = camelot_trigger, + .pointer = camelot_pos, +}; + +static void camelot_pcm_free(struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_free_for_all(pcm); +} + +static int camelot_pcm_new(struct snd_card *card, + struct snd_soc_codec_dai *dai, + struct snd_pcm *pcm) +{ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + card->dev, DMABRG_PREALLOC_BUFFER, + DMABRG_PREALLOC_BUFFER_MAX); + + return 0; +} + +/* au1xpsc audio platform */ +struct snd_soc_platform sh7760_soc_platform = { + .name = "sh7760-dmabrg", + .pcm_ops = &camelot_pcm_ops, + .pcm_new = camelot_pcm_new, + .pcm_free = camelot_pcm_free, +}; +EXPORT_SYMBOL_GPL(sh7760_soc_platform); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SH7760 Audio DMA (DMABRG) driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/hac.c b/sound/soc/sh/hac.c new file mode 100644 index 0000000..157268c --- /dev/null +++ b/sound/soc/sh/hac.c @@ -0,0 +1,335 @@ +/* + * Hitachi Audio Controller (AC97) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable HAC output pins! + */ + +/* + * BIG FAT FIXME: although the SH7760 has 2 independent AC97 units, only + * the FIRST can be used since ASoC does not pass any information to the + * ac97_read/write() functions regarding WHICH unit to use. You'll have + * to edit the code a bit to use the other AC97 unit. --mlau + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/interrupt.h> +#include <linux/wait.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +/* #define AC97_DEBUG */ + +#ifdef AC97_DEBUG +#define MSG(x...) printk(KERN_INFO "sh4-hac: " x) +#else +#define MSG(x...) do {} while (0) +#endif + +/* regs and bits */ +#define HACCR 0x08 +#define HACCSAR 0x20 +#define HACCSDR 0x24 +#define HACPCML 0x28 +#define HACPCMR 0x2C +#define HACTIER 0x50 +#define HACTSR 0x54 +#define HACRIER 0x58 +#define HACRSR 0x5C +#define HACACR 0x60 + +#define CR_CR (1 << 15) /* "codec-ready" indicator */ +#define CR_CDRT (1 << 11) /* cold reset */ +#define CR_WMRT (1 << 10) /* warm reset */ +#define CR_B9 (1 << 9) /* the mysterious "bit 9" */ +#define CR_ST (1 << 5) /* AC97 link start bit */ + +#define CSAR_RD (1 << 19) /* AC97 data read bit */ +#define CSAR_WR (0) + +#define TSR_CMDAMT (1 << 31) +#define TSR_CMDDMT (1 << 30) + +#define RIER_STARYIE (1 << 22) +#define RIER_STDRYIE (1 << 21) + +#define RSR_STARY (1 << 22) +#define RSR_STDRY (1 << 21) + +#define ACR_DMARX16 (1 << 30) +#define ACR_DMATX16 (1 << 29) +#define ACR_TX12ATOM (1 << 26) +#define ACR_DMARX20 ((1 << 24) | (1 << 22)) +#define ACR_DMATX20 ((1 << 23) | (1 << 21)) + +#define CSDR_SHIFT 4 +#define CSDR_MASK (0xffff << CSDR_SHIFT) +#define CSAR_SHIFT 12 +#define CSAR_MASK (0x7f << CSAR_SHIFT) + +#define AC97_WRITE_RETRY 1 +#define AC97_READ_RETRY 5 + +/* manual-suggested AC97 codec access timeouts (us) */ +#define TMO_E1 500 /* 21 < E1 < 1000 */ +#define TMO_E2 13 /* 13 < E2 */ +#define TMO_E3 21 /* 21 < E3 */ +#define TMO_E4 500 /* 21 < E4 < 1000 */ + +struct hac_priv { + unsigned long mmio; /* HAC base address */ +} hac_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE240000, + }, + { + .mmio = 0xFE250000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE40000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +#define HACREG(reg) (*(unsigned long *)(hac->mmio + (reg))) + +/* + * AC97 read/write flow as outlined in the SH7760 manual (pages 903-906) + */ +static int hac_get_codec_data(struct hac_priv *hac, unsigned short r, + unsigned short *v) +{ + unsigned int to1, to2, i; + unsigned short adr; + + for (i = 0; i < AC97_READ_RETRY; ++i) { + *v = 0; + /* wait for HAC to receive something from the codec */ + for (to1 = TMO_E4; + to1 && !(HACREG(HACRSR) & RSR_STARY); + --to1) + udelay(1); + for (to2 = TMO_E4; + to2 && !(HACREG(HACRSR) & RSR_STDRY); + --to2) + udelay(1); + + if (!to1 && !to2) + return 0; /* codec comm is down */ + + adr = ((HACREG(HACCSAR) & CSAR_MASK) >> CSAR_SHIFT); + *v = ((HACREG(HACCSDR) & CSDR_MASK) >> CSDR_SHIFT); + + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + + if (r == adr) + break; + + /* manual says: wait at least 21 usec before retrying */ + udelay(21); + } + HACREG(HACRSR) &= ~(RSR_STDRY | RSR_STARY); + return (i < AC97_READ_RETRY); +} + +static unsigned short hac_read_codec_aux(struct hac_priv *hac, + unsigned short reg) +{ + unsigned short val; + unsigned int i, to; + + for (i = 0; i < AC97_READ_RETRY; i++) { + /* send_read_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDAMT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) | CSAR_RD; + local_irq_enable(); + + for (to = TMO_E3; + to && !(HACREG(HACTSR) & TSR_CMDAMT); + --to) + udelay(1); + + HACREG(HACTSR) &= ~TSR_CMDAMT; + val = 0; + if (hac_get_codec_data(hac, reg, &val) != 0) + break; + } + + if (i == AC97_READ_RETRY) + return ~0; + + return val; +} + +static void hac_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int i, to; + /* write_codec_aux */ + for (i = 0; i < AC97_WRITE_RETRY; i++) { + /* send_write_request */ + local_irq_disable(); + HACREG(HACTSR) &= ~(TSR_CMDDMT | TSR_CMDAMT); + HACREG(HACCSDR) = (val << CSDR_SHIFT); + HACREG(HACCSAR) = (reg << CSAR_SHIFT) & (~CSAR_RD); + local_irq_enable(); + + /* poll-wait for CMDAMT and CMDDMT */ + for (to = TMO_E1; + to && !(HACREG(HACTSR) & (TSR_CMDAMT|TSR_CMDDMT)); + --to) + udelay(1); + + HACREG(HACTSR) &= ~(TSR_CMDAMT | TSR_CMDDMT); + if (to) + break; + /* timeout, try again */ + } +} + +static unsigned short hac_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + return hac_read_codec_aux(hac, reg); +} + +static void hac_ac97_warmrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac = &hac_cpu_data[unit_id]; + unsigned int tmo; + + HACREG(HACCR) = CR_WMRT | CR_ST | CR_B9; + msleep(10); + HACREG(HACCR) = CR_ST | CR_B9; + for (tmo = 1000; (tmo > 0) && !(HACREG(HACCR) & CR_CR); tmo--) + udelay(1); + + if (!tmo) + printk(KERN_INFO "HAC reset: AC97 link down!\n"); + /* settings this bit lets us have a conversation with codec */ + HACREG(HACACR) |= ACR_TX12ATOM; +} + +static void hac_ac97_coldrst(struct snd_ac97 *ac97) +{ + int unit_id = 0 /* ac97->private_data */; + struct hac_priv *hac; + hac = &hac_cpu_data[unit_id]; + + HACREG(HACCR) = 0; + HACREG(HACCR) = CR_CDRT | CR_ST | CR_B9; + msleep(10); + hac_ac97_warmrst(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = hac_ac97_read, + .write = hac_ac97_write, + .reset = hac_ac97_coldrst, + .warm_reset = hac_ac97_warmrst, +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +static int hac_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct hac_priv *hac = &hac_cpu_data[rtd->dai->cpu_dai->id]; + int d = substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? 0 : 1; + + switch (params->msbits) { + case 16: + HACREG(HACACR) |= d ? ACR_DMARX16 : ACR_DMATX16; + HACREG(HACACR) &= d ? ~ACR_DMARX20 : ~ACR_DMATX20; + break; + case 20: + HACREG(HACACR) &= d ? ~ACR_DMARX16 : ~ACR_DMATX16; + HACREG(HACACR) |= d ? ACR_DMARX20 : ACR_DMATX20; + break; + default: + MSG("unsupported depth %d bits\n", params->msbits); + return -EINVAL; + break; + } + + return 0; +} + +#define AC97_RATES \ + SNDRV_PCM_RATE_8000_192000 + +#define AC97_FMTS \ + SNDRV_PCM_FMTBIT_S16_LE + +struct snd_soc_cpu_dai sh4_hac_dai[] = { +{ + .name = "HAC0", + .id = 0, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "HAC1", + .id = 1, + .type = SND_SOC_DAI_AC97, + .playback = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .rates = AC97_RATES, + .formats = AC97_FMTS, + .channels_min = 2, + .channels_max = 2, + }, + .ops = { + .hw_params = hac_hw_params, + }, + +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_hac_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip HAC (AC97) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); diff --git a/sound/soc/sh/ssi.c b/sound/soc/sh/ssi.c new file mode 100644 index 0000000..79642e0 --- /dev/null +++ b/sound/soc/sh/ssi.c @@ -0,0 +1,413 @@ +/* + * Serial Sound Interface (I2S) support for SH7760/SH7780 + * + * Copyright (c) 2007 Manuel Lauss <mano(a)roarinelk.homelinux.net> + * + * licensed under the terms outlined in the file COPYING at the root + * of the linux kernel sources. + * + * dont forget to set IPSEL/OMSEL register bits (in your board code) to + * enable SSI output pins! + */ + +/* + * LIMITATIONS: + * The SSI unit has only one physical data line, so full duplex is + * impossible. This can be remedied on the SH7760 by using the + * other SSI unit for recording; however the SH7780 has only 1 SSI + * unit, and its pins are shared with the AC97 unit, among others. + * + * FEATURES: + * The SSI features "compressed mode": in this mode it continuously + * streams PCM data over the I2S lines and uses LRCK as a handshake + * signal. Can be used to send compressed data (AC3/DTS) to a DSP. + * The number of bits sent over the wire in a frame can be adjusted + * and can be independent from the actual sample bit depth. This is + * useful to support TDM mode codecs like the AD1939 which have a + * fixed TDM slot size, regardless of sample resolution. + * + * Don't forget to set IPSEL register to enable SSI output! + */ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/io.h> + +/* #define SSI_DEBUG */ + +#ifdef SSI_DEBUG +#define MSG(x...) printk(KERN_INFO "sh4-i2s: " x) +#else +#define MSG(x...) do {} while (0) +#endif + +#define SSICR 0x00 +#define SSISR 0x04 + +#define CR_DMAEN (1 << 28) +#define CR_CHNL_SHIFT 22 +#define CR_CHNL_MASK 3 +#define CR_DWL_SHIFT 19 +#define CR_DWL_MASK 7 +#define CR_SWL_SHIFT 16 +#define CR_SWL_MASK 7 +#define CR_SCK_MASTER (1 << 15) /* bitclock master bit */ +#define CR_SWS_MASTER (1 << 14) /* wordselect master bit */ +#define CR_SCKP (1 << 13) /* I2Sclock polarity */ +#define CR_SWSP (1 << 12) /* LRCK polarity */ +#define CR_SPDP (1 << 11) +#define CR_SDTA (1 << 10) /* i2s alignment (msb/lsb) */ +#define CR_PDTA (1 << 9) /* fifo data alignment */ +#define CR_DEL (1 << 8) /* delay data by 1 i2sclk */ +#define CR_BREN (1 << 7) /* clock gating in burst mode */ +#define CR_CKDIV_MASK 7 /* bitclock divider */ +#define CR_CKDIV_SHIFT 4 +#define CR_MUTE (1 << 3) /* SSI mute */ +#define CR_CPEN (1 << 2) /* compressed mode */ +#define CR_TRMD (1 << 1) /* transmit/receive select */ +#define CR_EN (1 << 0) /* enable SSI */ + +#define SSIREG(reg) (*(unsigned long *)(ssi->mmio + (reg))) + +struct ssi_priv { + unsigned long mmio; + unsigned long sysclk; + int inuse; +} ssi_cpu_data[] = { +#if defined(CONFIG_CPU_SUBTYPE_SH7760) + { + .mmio = 0xFE680000, + }, + { + .mmio = 0xFE690000, + }, +#elif defined(CONFIG_CPU_SUBTYPE_SH7780) + { + .mmio = 0xFFE70000, + }, +#else +#error "Unsupported SuperH SoC" +#endif +}; + +/* + * track usage of the SSI; it is simplex-only so prevent attempts of + * concurrent playback + capture. FIXME: any locking required? + */ +static int ssi_startup(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + if (ssi->inuse) { + MSG("SSI already in use!\n"); + return -EBUSY; + } else + ssi->inuse = 1; + return 0; +} + +static void ssi_shutdown(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + ssi->inuse = 0; +} + +static int ssi_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + SSIREG(SSICR) |= CR_DMAEN | CR_EN; + break; + case SNDRV_PCM_TRIGGER_STOP: + SSIREG(SSICR) &= ~(CR_DMAEN | CR_EN); + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct ssi_priv *ssi = &ssi_cpu_data[rtd->dai->cpu_dai->id]; + unsigned long ssicr = SSIREG(SSICR); + unsigned int bits, channels, swl, recv, i; + + channels = params_channels(params); + bits = params->msbits; + recv = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) ? 0 : 1; + + MSG("hw_params() enter\nssicr was %08lx\n", ssicr); + MSG("bits: %d channels: %d\n", bits, channels); + + /* direction (send/receive) */ + if (recv) + ssicr &= ~CR_TRMD; + else + ssicr |= CR_TRMD; + + /* channels */ + ssicr &= ~(CR_CHNL_MASK << CR_CHNL_SHIFT); + if ((channels < 2) || (channels > 8) || (channels & 1)) { + MSG("invalid number of channels\n"); + return -EINVAL; + } + ssicr |= ((channels >> 1) - 1) << CR_CHNL_SHIFT; + + /* DATA WORD LENGTH (DWL): databits in audio sample */ + i = 0; + ssicr &= ~(CR_DWL_MASK << CR_DWL_SHIFT); + switch (bits) { + case 32: ++i; + case 24: ++i; + case 22: ++i; + case 20: ++i; + case 18: ++i; + case 16: ++i; + ssicr |= i << CR_DWL_SHIFT; + case 8: break; + default: + MSG("invalid sample width\n"); + return -EINVAL; + } + + /* + * SYSTEM WORD LENGTH: size in bits of half a frame over the I2S + * wires. This is usually bits_per_sample x channels/2; i.e. in + * Stereo mode the SWL equals DWL. SWL can be bigger than the + * product of (channels_per_slot x samplebits), e.g. for codecs + * like the AD1939 which only accept 32bit wide TDM slots. For + * "standard" I2S operation we set SWL = chans / 2 * DWL here. + * Waiting for ASoC to get TDM support ;-) + */ + ssicr &= ~((CR_SWL_MASK << CR_SWL_SHIFT) | CR_PDTA); + if ((bits > 16) && (bits <= 24)) { + bits = 24; /* these are padded by the SSI */ + /*ssicr |= CR_PDTA;*/ /* cpu/data endianness ? */ + } + i = 0; + swl = (bits * channels) / 2; + switch (swl) { + case 256: ++i; + case 128: ++i; + case 64: ++i; + case 48: ++i; + case 32: ++i; + case 16: ++i; + ssicr |= i << CR_SWL_SHIFT; + case 8: break; + default: + MSG("invalid system word length computed\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + + MSG("hw_params() leave\nssicr is now %08lx\n", ssicr); + return 0; +} + +static int ssi_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, int clk_id, + unsigned int freq, int dir) +{ + struct ssi_priv *ssi = &ssi_cpu_data[cpu_dai->id]; + + ssi->sysclk = freq; + + return 0; +} + +/* + * This divider is used to generate the SSI_SCK (I2S bitclock) from the + * clock at the HAC_BIT_CLK ("oversampling clock") pin. + */ +static int ssi_set_clkdiv(struct snd_soc_cpu_dai *dai, int did, int div) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr; + int i; + + i = 0; + ssicr = SSIREG(SSICR) & ~(CR_CKDIV_MASK << CR_CKDIV_SHIFT); + switch (div) { + case 16: ++i; + case 8: ++i; + case 4: ++i; + case 2: ++i; + SSIREG(SSICR) = ssicr | (i << CR_CKDIV_SHIFT); + case 1: break; + default: + MSG("invalid sck divider %d\n", div); + return -EINVAL; + } + + return 0; +} + +static int ssi_set_fmt(struct snd_soc_cpu_dai *dai, unsigned int fmt) +{ + struct ssi_priv *ssi = &ssi_cpu_data[dai->id]; + unsigned long ssicr = SSIREG(SSICR); + + MSG("set_fmt()\nssicr was 0x%08lx\n", ssicr); + + ssicr &= ~(CR_DEL | CR_PDTA); /* I2S format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_RIGHT_J: + ssicr |= CR_DEL | CR_PDTA; + break; + case SND_SOC_DAIFMT_LEFT_J: + ssicr |= CR_DEL; + break; + default: + MSG("unsupported format\n"); + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_CLOCK_MASK) { + case SND_SOC_DAIFMT_CONT: + ssicr &= ~CR_BREN; + break; + case SND_SOC_DAIFMT_GATED: + ssicr |= CR_BREN; + break; + } + + ssicr &= ~(CR_SWSP | CR_SCKP); + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + ssicr |= CR_SCKP; /* sample data at low clkedge */ + break; + case SND_SOC_DAIFMT_NB_IF: + ssicr |= CR_SCKP | CR_SWSP; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + case SND_SOC_DAIFMT_IB_IF: + ssicr |= CR_SWSP; /* word select starts low */ + break; + default: + MSG("invalid inversion\n"); + return -EINVAL; + } + + ssicr &= ~(CR_SWS_MASTER | CR_SCK_MASTER); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + break; + case SND_SOC_DAIFMT_CBS_CFM: + ssicr |= CR_SCK_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFS: + ssicr |= CR_SWS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + ssicr |= CR_SWS_MASTER | CR_SCK_MASTER; + break; + default: + MSG("invalid master/slave configuration\n"); + return -EINVAL; + } + + SSIREG(SSICR) = ssicr; + MSG("set_fmt() leave\nssicr is now 0x%08lx\n", ssicr); + + return 0; +} + +/* the SSI depends on an external clocksource (at HAC_BIT_CLK) even in + * Master mode, so really this is board specific; the SSI can do any + * rate with the right bitclk and divider settings. + */ +#define SSI_RATES \ + SNDRV_PCM_RATE_8000_192000 + +/* the SSI can do 8-32 bit samples, with 8 possible channels */ +#define SSI_FMTS \ + (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_U20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_U24_3LE | \ + SNDRV_PCM_FMTBIT_S32_LE | SNDRV_PCM_FMTBIT_U32_LE) + +struct snd_soc_cpu_dai sh4_ssi_dai[] = { +{ + .name = "SSI0", + .id = 0, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#ifdef CONFIG_CPU_SUBTYPE_SH7760 +{ + .name = "SSI1", + .id = 1, + .type = SND_SOC_DAI_I2S, + .playback = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .capture = { + .rates = SSI_RATES, + .formats = SSI_FMTS, + .channels_min = 2, + .channels_max = 8, + }, + .ops = { + .startup = ssi_startup, + .shutdown = ssi_shutdown, + .trigger = ssi_trigger, + .hw_params = ssi_hw_params, + }, + .dai_ops = { + .set_sysclk = ssi_set_sysclk, + .set_clkdiv = ssi_set_clkdiv, + .set_fmt = ssi_set_fmt, + }, +}, +#endif +}; +EXPORT_SYMBOL_GPL(sh4_ssi_dai); + +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("SuperH onchip SSI (I2S) audio driver"); +MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>");

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[alsa-devel] [ASoC] Mic record is not working with wm9713 ASoC driver
by Nobin Mathew 09 May '07

09 May '07

I am trying to record mic with AsoC wm9713 driver. My Mic is connected "Mic 1". My settings 'Mic A Source' -> "Mic 1" 'Mic Boost (+20dB)' [on] 'Capture' [on] 'Left Capture Source' "Mic 1" 'Right Capture Source' "Mic 1" i am issuing following command to record "arecord -M -f S16_LE sound.wav" But what i found is driver is not enabling "Mic bias" and "Mic A PGA" this dump of /sys/devices/platform/soc-audio/dapm_widget while "arecord" is running is given below Mic Bias: Off Mic B Pre Amp: Off Mic A Pre Amp: On Mic B PGA: Off Mic A PGA: Off Mono In: Off Right Line In: Off Left Line In: Off Mono Out: Off Out 4: Off Out 3: Off Right Speaker: Off Left Speaker: Off Right Headphone: Off Left Headphone: Off Right ADC: On Left ADC: On Aux DAC: Off Voice DAC: Off Capture Mixer: Off HP Mixer: Off AC97 Mixer: Off Right DAC: Off Left DAC: Off Speaker Mixer: Off Mono Mixer: Off Right HP Mixer: Off Left HP Mixer: Off PM State: D0 In idea why this is happening? what is going wrong with my settings? Nobin Mathew

1 0

[alsa-devel] snd-serial-u16550 overflow problems?
by r10 kindsofpeople 08 May '07

08 May '07

{reposting (with apologies) from alsa-user at another user's suggestion} I'm having what appears to be overflow problems within the snd-serial-u16550 driver. I'm using 1.0.14rc1. When calling "amidi -p hw:0,0,0 -s FILE.SYX" where hw:0,0,0 is the snd-serial-u16550 port, the data coming from the serial port appears to be a small fraction (900 bytes) of the total file size (13000 bytes). If I repeat the experiment, but send to a USB MIDI port, all is well. If I send the data to the serial port using calls to snd_seq_event_output_direct(), and I add a 1 second pause between every sysex message, all is well (aside from it taking too long). I'm logging data directly at the PC serial port using another PC, so I've eliminated the possibility that the serial-to-MIDI hardware is losing the data. My code is using blocking mode, and besides, amidi behaves the same way as my code. I'm hesitant to blame the driver code (especially given that last time I did this, it was an interrupt configuration problem), but I could use some help figuring out where to look next. Please? Are there calls I can use to ensure that the previous message has gone out before writing the next one? Given the amount of data I'm sending, should I be using snd_seq_event_output and scheduling the events on the queue? John _________________________________________________________________ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-us&ocid=TXT_TAGHM_migrati…

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[alsa-devel] [PATCH] ppc-beep - switch to using input_dev->dev.parent
by Dmitry Torokhov 08 May '07

08 May '07

In preparation for struct class_device -> struct device input core conversion, switch to using input_dev->dev.parent when specifying device position in sysfs tree. Also, do not access input_dev->private directly, use helpers. Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> --- sound/ppc/beep.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) Index: work/sound/ppc/beep.c =================================================================== --- work.orig/sound/ppc/beep.c +++ work/sound/ppc/beep.c @@ -118,7 +118,7 @@ static int snd_pmac_beep_event(struct in default: return -1; } - chip = dev->private; + chip = input_get_drvdata(dev); if (! chip || (beep = chip->beep) == NULL) return -1; @@ -239,8 +239,8 @@ int __init snd_pmac_attach_beep(struct s input_dev->evbit[0] = BIT(EV_SND); input_dev->sndbit[0] = BIT(SND_BELL) | BIT(SND_TONE); input_dev->event = snd_pmac_beep_event; - input_dev->private = chip; - input_dev->cdev.dev = &chip->pdev->dev; + input_dev->dev.parent = &chip->pdev->dev; + input_set_drvdata(input_dev, chip); beep->dev = input_dev; beep->buf = dmabuf; @@ -251,8 +251,8 @@ int __init snd_pmac_attach_beep(struct s err = snd_ctl_add(chip->card, beep_ctl); if (err < 0) goto fail1; - - chip->beep = beep; + + chip->beep = beep; err = input_register_device(beep->dev); if (err)

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[alsa-devel] Fwd: [Alsa-devel] Read the input of a soundcard with ALSA
by Nyx 07 May '07

07 May '07

Hi, I have some problems to read the input of my soundcard using Alsa and I don't really understand how to have access to the input of my soundcard. I follow : http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht… # So, in my program, I have created two handles for capture and playback : char *device = "hw:0,0"; snd_output_t *output; snd_input_t *input; snd_pcm_t *phandle, *chandle; # then I connect the phandle to the out err = snd_output_stdio_attach(&output, stdout, 0); if (..) # Do I have to connect the chandle with the input ? # err = snd_input_stdio_attach(&input, stdin, 0); # I open the access to PCM err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) if (..) err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) if(..) # Hardware and software Parameters ## Hardware Parameters snd_pcm_hw_params; int err; err = snd_pcm_hw_params_any(phandle, params); if(..); err = snd_pcm_hw_params_set_access(phandle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); if(..); err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); if(..); err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), if(..); err = snd_pcm_hw_params(phandle, params); ##Software parameters err = snd_pcm_sw_params(phandle, params); if(..); err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); if(..); err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); # same configuration for phandle # Link output to input and start err = snd_pcm_link(chandle, phandle); if(..); err = snd_pcm_start(chandle); if(..); #Then I want to read my input frames_in = 0; in_max = 0; latency = 28; buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); while (1) { r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); } # description of the function long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t *max) { long r; do { r = snd_pcm_readi(handle, buf, len); } while (r == -EAGAIN); if (r > 0) { *frames += r; if ((long)*max < r) *max = r; } printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, buf[i]); return r; } So I compile it and I run it, and I have $ read = Broken Pipe, len = 28 and buf[..] = 0.000 Anyone can help me to read the input of my sound card ? Thank you -- K PS : if I sent this message twice, excuse-me. I haven't confirm my subscription the first time

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[alsa-devel] [PATCH] Add sequence parsing for speaker-outs in snd_hda_parse_pin_def_config()
by Steve Longerbeam 07 May '07

07 May '07

2 2

Re: [alsa-devel] Getting Started on Alsa
by Ash Willis 07 May '07

07 May '07

Hi Chris, > I'd like to take action on bug #305, since it's my own itch. I've done very > low-level programming, but not on Linux device drivers. I'm very familiar > with command line development tools. > If you're not familiar with drivers or audio drivers in particular then reading Writing an ALSA Driver by Takashi Iwai might be a good place to start. You're referring to the suspend/resume bug in the cs46xx driver, right? Maybe try dumping all the registers while it's working and dumping them while it's broken and comparing the two sets of data? I guess the suspend function is doing something that it shouldn't be, or the resume function isn't doing something that it should be...good luck on finding out what that is. Ash =

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Re: [alsa-devel] Getting Started on Alsa
by Ash Willis 07 May '07

07 May '07

> Thanks for the tips, Ash. I've found the Iwai document. Now there's some > light reading. I take it that it's still relevant, even though it hasn't > been updated since 2005. > I haven't actually read it since 2005 I'm afraid :) but I don't think anything's changed that drastically. The fundamentals are all there. > Yes, the cs46xx driver. That chip is in a lot of machines (at least ten > ThinkPad models), and with this bug nobody running Alsa likes Linux very > much. > Yeah, it's a pretty poor show that this bug has existed for so long. Let's hope you can zap it! > Is there some (handy) way to dump the registers from DOS or Windows as a > comparison? > Well, that depends. I'm not familiar with the hardware. There's a Windows program called PCItree. It allows you to view the current status of the ioport of a chosen device. Now, that's pretty handy if the regs for the card are mapped directly onto the io port. It's usually the case though, I think, that the registers aren't just mapped on and you'll need to query the card. If they're not mapped onto the io port (and forgive me again - not familiar with this chip) then you could use a debugger such as Numega SoftIce to manually issue queries to find out the contents of the registers. Sounds like a happy fun time. Writing the code to do a reg dump might be fun too. You may also want to use a debugger/disassembler with Windows (or other) drivers in order to find out what those drivers do to the card on resume. > For development: Since the drivers go into the kernel, I should be able to > find a page on getting a development environment set up... right? I'm You don't need to mess with the kernel as ALSA is available as a package. All you really need is the latest copy of alsa-driver and a compiler and you're good to go. Ash =

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[alsa-devel] 1.0.14~rc4 prepared for upload.
by Elimar Riesebieter 06 May '07

06 May '07

As the compile errors in alsa-lib fixed I've preparded 1.0.14~rc4 (svn r1849) for upload: alsa-driver alsa-lib alsa-oss alsa-plugins alsa-utils alsa-tools stays still as 1.0.13~rc3. Jordi, could you please upload? Now I go for a beer ;) Elimar -- You cannot propel yourself forward by patting yourself on the back.

1 1

[alsa-devel] Getting Started on Alsa
by Chris Schumann 06 May '07

06 May '07

Hello, I'd like to take action on bug #305, since it's my own itch. I've done very low-level programming, but not on Linux device drivers. I'm very familiar with command line development tools. Is there any documentation on how to get started doing bugfixes or development for Alsa, or do I just look for source and start hacking? Thanks, Chrs

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[alsa-devel] [PATCH] hda_codec: restore control element values during resume
by Daniel Drake 06 May '07

06 May '07

After resume from suspend-to-RAM, there is no sound output from my Intel HDA hardware: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) Found in a Dell Inspiron 640m, SigmaTel STAC9200 chip. After resuming, sound immediately returns when you adjust the volume one notch in either direction. I reported this on the ALSA bug tracker (#0002989). Further investigation shows that the AC_VERB_SET_AMP_GAIN_MUTE verb is not being executed during resume, which seems strange. Does other hardware really store this value during a suspend/resume cycle? Or maybe the volume is generally controlled through other means on other setups? I'm not entirely sure what the difference is between kcontrol and kcontrol_new structures, but I note that there is already some code to restore the values for kcontrol_new elements. I solved the issue with the following patch. Is it correct? Signed-off-by: Daniel Drake <dsd(a)gentoo.org> Index: linux-2.6/sound/pci/hda/hda_codec.c =================================================================== --- linux-2.6.orig/sound/pci/hda/hda_codec.c +++ linux-2.6/sound/pci/hda/hda_codec.c @@ -2248,15 +2248,39 @@ EXPORT_SYMBOL(snd_hda_suspend); int snd_hda_resume(struct hda_bus *bus) { struct list_head *p; + struct snd_ctl_elem_value *val = kmalloc(sizeof(*val), GFP_KERNEL); + + if (!val) + return -ENOMEM; list_for_each(p, &bus->codec_list) { struct hda_codec *codec = list_entry(p, struct hda_codec, list); + struct snd_kcontrol *kctl; + + codec->in_resume = 1; + hda_set_power_state(codec, codec->afg ? codec->afg : codec->mfg, AC_PWRST_D0); + + + list_for_each_entry(kctl, &bus->card->controls, list) { + memset(val, 0, sizeof(*val)); + val->id = kctl->id; + /* Assume that get callback reads only from cache, + * not accessing the real hardware + */ + if (snd_ctl_elem_read(bus->card, val) < 0) + continue; + snd_ctl_elem_write(bus->card, NULL, val); + } + if (codec->patch_ops.resume) codec->patch_ops.resume(codec); + + codec->in_resume = 0; } + kfree(val); return 0; }

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[alsa-devel] New Soundcard Matrix
by Jaroslav Kysela 05 May '07

05 May '07

Hello all, I moved the contents of matrix from XML files to Wiki - see http://bugtrack.alsa-project.org/main/index.php/Matrix:Main Please, no more updates to old matrix now. I left the information structure (we can update it later): Vendor -> Device -> ALSA Module If any volunteers are watching this list, we have many "wanted" pages now: http://bugtrack.alsa-project.org/main/index.php/Special:Wantedpages Example page for Matrix:Tag-* : http://bugtrack.alsa-project.org/main/index.php?title=Matrix:Tag-PCI&action… Example page for Matrix:Module-* : http://bugtrack.alsa-project.org/main/index.php?title=Matrix:Module-ali5451… Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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Re: [alsa-devel] Wiki Site
by Jaroslav Kysela 05 May '07

05 May '07

On Fri, 4 May 2007, cameron porter wrote: > Greetings, > > I noticed the notice on your front page needing people to help transfer the > site contents over to your wiki that you set up. I did a little bit of work, > and managed to get a navigation, downloads page, application page, and some > other links. Your work is greatly appreciated here. > I would be glad to continue, if you think I am doing things right. The next > thing I had planned to make some changes to was the documentation section, > add a changes section, add a structure whereby you have an article for each > package, and then the folder for that package contains all the different > versions that have been or will be released. Please give me any thoughts on > any of these, and tell me if you think they are a good idea, and whether I > should continue or not. We need to move the contents at first, but I agree that a good page structure should be created during this process. Please, go ahead - the nice thing on wiki is that we can rearrange information later (if we find a problem with the structure). I just started to move probably most problematic part - soundcard matrix to wiki. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Call for Explanation of ALSA Config Behavior
by Jeff Rush 05 May '07

05 May '07

I'm trying to get ALSA to support all the little features of my laptop, which is based on the Intel HDA chipset with the Realtek codec. I'm also playing with the softvol plug to add a volume controls over certain virtual devices. A few questions I can't figure out: 1. Isn't it the standard convention that there be a control named "Master" that adjusts the overall volume of a device? So why does the Intel HDA ALSA driver define one named "Headphone" that does the same thing instead of following that convention? Is there some reason, or perhaps a difference in opinion within the ALSA community on the naming of controls? Shouldn't every ALSA driver define a "Master" control? 2. When I add a control using the softvol plugin, the mixer GUI shows it in both the playback and the capture set of controls. Why and how can I tag a control to belong to only one set or the other? Being a volume control it seems it should only be in the playback collection. 3. Once I add such a control, it -never- goes away. What I mean is that if I rename it, or remove it, and restart ALSA or even restart the entire computer, the old name for the control always shows up in the mixer GUI. I've grepped *all* of /etc, looking in the /etc/asound.conf, /etc/asound.state and ~/.asoundrc files but the old name isn't anywhere -- where is it coming from? Thanks very much for any help you can provide, For years I've just tried to "use" ALSA but now I'm trying to actually understand its subtle nuances. -Jeff

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[alsa-devel] DMix period_size should always be half of buffer size?
by Prabhu Sivaraja 05 May '07

05 May '07

This is a followup to my earlier question about dmix not being able to mix small files. I investigated it more and here is what I found. Again background info. I have a 64KB as kernel buffer and a 64KB circular buffer for my device. In a poll thread I copy the data obtained from ALSA lib from the kernel buffer to my device circular buffer by keeping track of appl_ptr and hw_ptr(dmix only). Everything was working fine except when I was trying to mix files smaller than 64KB(my buffer size), the small file was not getting mixed through the dmix plugin. Anything over 64KB was getting mixed just fine. So I set about reducing my period size from 32KB to 4KB while maintaining my buffer size at 64KB. When I tried to do this my hw_ptr never gets incremented from 0. Hence my driver is stuck in a loop waiting for bytes to copy. This is only observed when I am using dmix plugin and NOT when using hw or plughw. When my period size is half the size of my buffer everything works fine with dmix (I verified this by reducing my buffer size to 8KB and making my period size 4KB. hw_ptr gets incremented then but I get buffer problems). It seems like dmix is forcing me to keep my period size half the size of my buffer. hw and plughw devices do not seem to do that. Is this a limitation of dmix. Could anyone comment on the dmix limitation that I am experiencing. Thanks a lot of all your help. Prabhu.

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[alsa-devel] Aureon 7.1 Firewire supported by ALSA ?
by Michael Gerdau 04 May '07

04 May '07

Hi list, looking for an external (USB or Firewire) interface to record at least a single stereo 44.1kHz/24bit S/P-DIF signal I've come across the Aureon 7.1 Firewire but have not been able to find out whether it is already supported by ALSA. Another option would be the Aureon 5.1 USB MKII but according to the specs it is 16bit only. However when feeding S/P-DIF with a stereo 44.1kHz/24bit would the driver be recognize that and give me that signal unaltered or would it be trimmed to 16bit ? [Or is my understanding of S/P-DIF just plain wrong and such a signal would not go through the box at all ?] Any advice welcome, best, Michael -- Technosis GmbH, Geschäftsführer: Michael Gerdau, Tobias Dittmar Sitz Hamburg; HRB 89145 Amtsgericht Hamburg Vote against SPAM - see http://www.politik-digital.de/spam/ Michael Gerdau email: mgd(a)technosis.de GPG-keys available on request or at public keyserver

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[alsa-devel] [PATCH] multi-channel playback and capture fixes for stac92xx codecs
by Steve Longerbeam 04 May '07

04 May '07

Hi, I have picked up where Matt Porter left off on stac92xx HDA codec devel. I've been working with the 9200, 9221, 9202, 9250, and 9205 codecs, and I found some problems with muti-channel playback on 922x and 9205. Also a problem with input mux selection for codecs with more than one configured input pin. I've tested the attached patch on all the above listed codecs. Steve

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[alsa-devel] [PATCH] snd-usb-audio: Add another QuickCam ID
by Daniel Drake 04 May '07

04 May '07

This adds another Logitech QuickCam USB ID. These devices have vendor-specific device class values, but standards-compliant audio interfaces. 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 @@ -53,6 +53,16 @@ .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 = 0x08f6, + .bInterfaceClass = USB_CLASS_AUDIO, + .bInterfaceSubClass = USB_SUBCLASS_AUDIO_CONTROL +}, + /* * Yamaha devices

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[alsa-devel] support for Medion laptops
by albox＠free.fr 04 May '07

04 May '07

Hi, I installed Ubuntu Feisty a few days ago and my sound card driver doesn't work anymore. It was working perfectly before updating to Feisty but I'm not sure the problem comes from the update since I was not using the last ALSA version but an older one patched with a patch released by Tobin Davis a few months ago (something else could have been changed in the ALSA source code since then) : --- alsa-driver/alsa-kernel/pci/hda/patch_realtek.c.orig 2006-10-08 10:58:28.000000000 -0700 +++ alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2006-10-09 10:50:03.000000000 -0700 @@ -112,6 +112,7 @@ ALC883_6ST_DIG, ALC888_DEMO_BOARD, ALC883_ACER, + ALC883_MEDION, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -5080,6 +5081,8 @@ .config = ALC883_ACER }, { .pci_subvendor = 0x1025, .pci_subdevice = 0x009f, .config = ALC883_ACER }, + { .pci_subvendor = 0x161f, .pci_subdevice = 0xd82b, + .modelname = "medion", .config = ALC883_MEDION } { .modelname = "auto", .config = ALC883_AUTO }, {} }; @@ -5167,6 +5170,20 @@ .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_MEDION] = { + .mixers = { alc883_base_mixer, + alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, + alc882_eapd_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_sixstack_modes), + .channel_mode = alc883_sixstack_modes, + .input_mux = &alc883_capture_source, + } + }; You can find some parts of the emails about this subject on this web page : http://www.spinics.net/lists/alsa-devel/msg04396.html The driver (Developement release 1.0.14rc3) can detect my sound card but I can't figure out what's wrong. Can anybody help me ? $ dmesg | grep alsa [ 20.468000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:696: codec_mask = 0x3 [ 20.648000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:1768: hda_codec: model 'medion' is selected for config 161f:2054 (Medion laptop) [ 21.376000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:245: si3054: cannot initialize. EXT MID = 0000 [ 21.384000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/patch_si3054.c:257: Link Frame Detect(FDT) is not ready (line status: 0000) [ 76.256000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 76.256000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 76.264000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 76.272000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 76.280000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 76.292000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 76.300000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 76.308000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 76.320000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 76.332000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 76.332000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 76.344000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 76.356000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 76.364000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 76.376000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 76.384000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 76.392000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 76.408000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 76.592000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 76.592000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 76.600000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 76.608000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 76.616000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 153.564000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 153.572000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 153.580000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 153.588000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 216.896000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 216.896000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 216.904000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 216.912000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 216.920000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 216.932000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 216.940000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 216.948000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 216.956000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 [ 216.972000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1158: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 [ 216.972000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 [ 216.980000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 [ 216.988000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 [ 216.996000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 [ 218.824000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 [ 218.832000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 [ 218.840000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 [ 218.848000] ALSA /home/alboot/Sources/alsa/alsa-driver-1.0.14rc3/pci/hda/hda_codec.c:627: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 Thanks

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[alsa-devel] dmix does not mix small files??
by Prabhu Sivaraja 03 May '07

03 May '07

Hi, I've been stuck on this dmix problem for the past week and have tried everything under the sun. Please any help or comment is much appreciated. Thanks a lot. Background info. ----------------------- In my ALSA driver I have thread that keeps track of hw_ptr(dmix) appl_ptr(for all other cases) and copies the incoming data from kernal buffer into device circular buffer(which is monitored by the device). I allocate 64KB kernal buffer in my module_init function . (See allocation below) snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), 64*1024 , 64*1024); So I monitor this kernal buffer and copy data from this to my device circular buffer which is also 64KB. My period size is defined as 32KB. (See below) substream->runtime->hw.period_bytes_min = 1024*32; substream->runtime->hw.period_bytes_max = 1024*32; substream->runtime->hw.periods_min = 2; substream->runtime->hw.periods_max = 2; My asound.conf file pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } Everything was working fine until I noticed something weird. When I am playing a long wav file by invoking aplay and then try and play another small wav file simultaneously, the second file never gets mixed if the file is less than 64KB(my buffer size). If the second file is larger than 64KB then the second file gets mixed with the first one as it should since I am using dmix plugin. So in order to debug this I tried to decrease my period_size = 4KB but when I do this my hw_ptr never moves. My question. 1. Why do you think a file smaller than 64KB does not get mixed. (Obviously it has something to do with my buffer size but I dont get why?) 2. When I decrease my period_size to 4KB while keeping my buffer size at 64KB, why doesnt my hw_ptr move? 3. How else would I debug this? Thanks in advance. I would really appreciate any help. Regards Prabhu.

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[alsa-devel] alsa test function?
by Ashlesha Shintre 03 May '07

03 May '07

Hi, Is there a test program for alsa, that gives a report on whether alsa is working correct with the underlying hardware or just does elementary testing of alsa on a system? Thanks, Ashlesha.

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[alsa-devel] 1.0.14rc4 release
by Jaroslav Kysela 03 May '07

03 May '07

Hello all, 1.0.14rc4 packages are available for download on http://www.alsa-project.org . The full changelog can be obtained here: http://www.alsa-project.org/changes/v1-0-14rc3--v1-0-14rc4.txt . Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Add support for new HP DV series laptops
by Tobin Davis 03 May '07

03 May '07

This patch adds support for 3 new HP laptops to the Conexant "Venice" driver. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] ALSA TImers
by Nobin Mathew 03 May '07

03 May '07

Hi, I am new to ALSA. Can any body explain me the use of timers in ALSA, timers like the one in pcm_timer.c Thanks in advance Nobin Mathew

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Re: [alsa-devel] Support for SAM9407?
by Ash Willis 03 May '07

03 May '07

Hi Pieter, > Are you interested in joining the FFADO project? (short into: FFADO = > new FreeBoB = drivers for FireWire Audio Interfaces / www.ffado.org). > I have absolutely zero experience in firewire audio and firewire in general to be honest, but the project does sound interesting. > To be frank/direct here, I think that ALSA is better 'completed' with > FFADO support than with driver support for obsolete cards. But of course > I'm a little biased here :). Be as Frank as you want, I totally agree with you :) The thing is, I have limited time and experience (I have written one ALSA driver). Although supporting older hardware doesn't bring a whole lot to the project, it does provide me with a training ground so I can be of greater use in the future. I can work at my own pace on an area that nobody else will be working on. I can improve my skills without stepping on other people's toes. Your project sounds interesting but I don't feel I have the knowledge to keep up on a project like this. This is why I'm messing around with this stuff, so that I can do some 'real work' some day soon ;) regards, Ash =

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[alsa-devel] alsa driver api
by Michel Benoit 03 May '07

03 May '07

Hi Does anyone have some code examples that show how to send pcm samples directly to an alsa driver. I'm (still) having trouble building/using alsa-lib and I want to make sure that my driver works correctly. If I understand correctly it is possible to send sound samples using the alsa driver api only (no alsa-lib). All I want to do for now is send a fixed sine wave to the driver. I saw some code snippets that seemed to do just that in a recent email on this list but it would help a lot if I could have a look at the complete application. Thanks, Michel

2 1

[alsa-devel] Newb wants to fix snd_cs46xx driver
by chris＠idlelion.net 03 May '07

03 May '07

There's been a bug on this chip's driver for at least four years. https://bugtrack.alsa-project.org/alsa-bug/view.php?id=305 There are two ThinkPad 600X's in my home that have the problem. The chip is also in the ThinkPad A20m, A20p, A21m, A21p, A22m, A22p, T20, T21 and T22... and probably other brands as well. In the 600X, it happens to be paired with a CS4297A... hey, should I just try to get Linux to use that? (I use Fedora BTW.) How would I try to do that? Or... better still... how would I get either the '97A driver installed automatically by soundcard detection, or failing that, actually fix the driver? (I am a software developer who has written machine code for fun.) Thanks, Chris

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[alsa-devel] Support for SAM9407?
by Ash Willis 02 May '07

02 May '07

Hi, I've just been digging around for information on the old Dream SAM9407 chip. What's the current support status in ALSA for this chip? I know some work was done on producing drivers for cards based on this chip and I know there were plans to port the efforts to ALSA. Did that ever happen? It doesn't seem so. Was there a specific reason why it wasn't ported in the end? In the interest of making ALSA that bit more complete as well as trying to improve my skills, I've been looking around for some older hardware that I can write drivers for. This chip seems like a good candidate considering the cards haven't quite disappeared into oblivion :) I was looking into the possiblity of porting the remaining kernel OSS drivers to ALSA...but I can't even find any test hardware because they're so rare. Ash Willis =

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Re: [alsa-devel] alsa driver api
by Ciaccia 02 May '07

02 May '07

Hi Michel, Maybe you are referring to my post. I always use alsa-lib to send data to the driver, I think it is the easiest way and preferred way. Months ago I wanted to do the same (i.e. send data to the alsa-driver without alsa-lib) but then I understood this was not the right way. I'm not saying you cannot or should not try that, but from my experience (I spent a lot of time) it's not going to work. The problem I had was that my system did not support shared objects, nor -ldl, so I could not user the "standard" alsa-lib (which usually require pthread, dl, an *.so). Then Takashi gave me a patch for that, and now I'm an happy "static" alsa-lib user. Maybe one of the gurus (Takashi?) will give you some tips, but I think you really need a good reason for that. Good luck, sorry for not helping you Andrea ----- Original Message ---- From: Michel Benoit <murpme(a)gmail.com> To: ciacciax(a)yahoo.com; alsa-devel(a)alsa-project.org Sent: Wednesday, May 2, 2007 7:40:29 PM Subject: alsa driver api Hi Does anyone have some code examples that show how to send pcm samples directly to an alsa driver. I'm (still) having trouble building/using alsa-lib and I want to make sure that my driver works correctly. If I understand correctly it is possible to send sound samples using the alsa driver api only (no alsa-lib). All I want to do for now is send a fixed sine wave to the driver. I saw some code snippets that seemed to do just that in a recent email on this list but it would help a lot if I could have a look at the complete application. Thanks, Michel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] how does a buffer underrun / broken pipe error take place?
by Ashlesha Shintre 02 May '07

02 May '07

Hi, Following up from my previous email, is there a website that explains in greater detail exactly how a buffer underrun occurs? I think that might help greatly in debugging my program, attached with the previous email. Thanks, Ashlesha.

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[alsa-devel] error in recovery from buffer under-run.
by Ashlesha Shintre 02 May '07

02 May '07

Hi, Thanks for the link you emailed earlier. I tried writing my own code, based on the how to in the alsa wiki. I have attached that code with this email. The problem it faces is a buffer underrun every alternate time, i try writing to the device. In my code, to take care of underruns, I do the following: period_size=64; buffer is a short int array which is 16000 in length. every alternate call to snd_pcm_writei is successful but only write in the vicinity of 1350 frames, instead of size which is initialised to length of buffer. Now, in the code below, the snd_pcm_recover function prints a message indicating a buffer under-run has occured, however, it returns a 0, indicating success in recovery from the error! I am not sure how to deal with this -- is there a problem with my ALSA libraries? Sorry about the length of this email and thanks again, Ashlesha. while(size>(2*period_size)) > { > frames = snd_pcm_writei (pcm_handle, buffer, size); > printf("frames returned is %d\n",frames); > if (frames>0) > temp=frames; > if (frames == -32){ > frames = snd_pcm_recover(pcm_handle, frames, 0); > printf("recover returned %d\n",frames); > } > if (frames < 0){ > printf("snd_pcm_writei failed: %s\n", snd_strerror(frames)); > return -1; > } > printf("%d frames written\n",frames); > if(temp>0) > size=size-temp; > } >

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Re: [alsa-devel] [ALSA][hda-codec] fix-connexant-headphone-mute
by Takashi Iwai 02 May '07

02 May '07

[keep cc to alsa-devel please] At Wed, 02 May 2007 13:47:53 +0300, Andrei Popa wrote: > > On Wed, 2007-05-02 at 12:16 +0200, Takashi Iwai wrote: > > At Sun, 29 Apr 2007 05:19:50 +0300, > > Andrei Popa wrote: > > > > > > On Sun, 2007-04-29 at 04:43 +0300, Andrei Popa wrote: > > > > Hi, > > > > > > > > The sound works ok with the built-in speakers on my Fujitsu laptop(amilo > > > > pro v3205). But if I plug in headphones in the audio out jack there is > > > > no sound in the headphones. With kernel 2.6.20 it worked ok. With 2.6.21 > > > > it doesn't. > > > > This BUG is reported in ALSA bugtracking system with ID 0003039. > > > > > > > > Patch that fixes this bug: > > > > > > > > diff --git a/sound/pci/hda/patch_conexant.c > > > > b/sound/pci/hda/patch_conexant.c > > > > index 46e93c6..5edd330 100644 > > > > --- a/sound/pci/hda/patch_conexant.c > > > > +++ b/sound/pci/hda/patch_conexant.c > > > > @@ -636,7 +636,7 @@ static void cxt5045_hp_automute(struct hda_codec > > > > *codec) > > > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > > snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, > > > > bits); > > > > } > > > > > > > > /* unsolicited event for HP jack sensing */ > > > > @@ -988,10 +988,10 @@ static void cxt5047_hp_automute(struct hda_codec > > > > *codec) > > > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > > snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, > > > > bits); > > > > /* Mute/Unmute PCM 2 for good measure - some systems need this > > > > */ > > > > snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, > > > > bits); > > > > + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, > > > > bits); > > > > } > > > > > > > > /* toggle input of built-in and mic jack appropriately */ > > > > > > > > > > first patch was broken. it only worked for the right channel. > > > this fixes this: > > > > > > diff --git a/sound/pci/hda/patch_conexant.c > > > b/sound/pci/hda/patch_conexant.c > > > index 46e93c6..0822df7 100644 > > > --- a/sound/pci/hda/patch_conexant.c > > > +++ b/sound/pci/hda/patch_conexant.c > > > @@ -635,8 +635,8 @@ static void cxt5045_hp_automute(struct hda_codec > > > *codec) > > > AC_VERB_GET_PIN_SENSE, 0) & > > > 0x80000000; > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > - snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, > > > bits); > > > - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, > > > bits); > > > } > > > > > > /* unsolicited event for HP jack sensing */ > > > @@ -987,11 +987,11 @@ static void cxt5047_hp_automute(struct hda_codec > > > *codec) > > > AC_VERB_GET_PIN_SENSE, 0) & > > > 0x80000000; > > > > > > bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; > > > - snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, > > > bits); > > > - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, > > > bits); > > > /* Mute/Unmute PCM 2 for good measure - some systems need this > > > */ > > > - snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, > > > bits); > > > - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0, > > > bits); > > > + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, > > > bits); > > > } > > > > > > /* toggle input of built-in and mic jack appropriately */ > > > > But this means to _unmute_ the speaker (0x1d) when you plug HP. > > So, something else should be wrong. Is hp_present detected properly > > per plugging or unplugging? > > > > when I debugged the problem I think hp_present was "0" when the HP were > plugged in and "-28953535"(a big negative number) when the HP were > unplugged. Weird, then this returns exactly the inversed bit. Could you double-check? Anyway, having both this auto-mute and manual switches is confusing. If they exist, we'd need to add notifier to the control at unsol event handler. Takashi

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[alsa-devel] Intel ICH6 with AD1986: Mic not working
by Joerg Platte 02 May '07

02 May '07

Hi! On my computer both microphones are not working but playback and all other audio stuff seems to work. I'm not sure if the microphones ever worked with alsa because I never tried it before. On windows they are working as expected, hence the hardware seems to be OK. Is there anything I can do to further debug this? Is it a known problem? Below you can find more information about the hardware: root@ds19:~> uname -a Linux ds19 2.6.20-1-686 #1 SMP Tue Apr 24 21:52:11 UTC 2007 i686 GNU/Linux (Debian kernel) Doesn't work with 2.6.18 as well. root@ds19:~> cat /proc/asound/card0/codec97#0/ac97#0-0 0-0/0: Analog Devices AD1986 PCI Subsys Vendor: 0x1734 PCI Subsys Device: 0x105a Revision : 0x02 Compat. Class : 0x00 Subsys. Vendor ID: 0x0000 Subsys. ID : 0x0000 Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 16-bit 3D enhancement : No 3D Stereo Enhancement Current setup Mic gain : +20dB [+20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Double rate slots: 10/11 Extended ID : codec=0 rev=2 AMAP LDAC SDAC CDAC DSA=0 SPDIF DRA VRA Extended status : SPCV LDAC SDAC CDAC SPDIF=10/11 VRA PCM front DAC : 48000Hz PCM Surr DAC : 48000Hz PCM LFE DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Copyright Category=0x22 Generation=1 Rate=48kHz Gain Inverted Buffer delay Location Master Out : 0.0 dBV - 16/fs Rear I/O Panel AUX Out : 0.0 dBV - 16/fs Rear I/O Panel Center/LFE Out : 0.0 dBV - 16/fs Rear I/O Panel Mic 1 : 0.0 dBV - 16/fs Rear I/O Panel Mic 2 : 0.0 dBV - 16/fs Rear I/O Panel Line In : 0.0 dBV - 16/fs Rear I/O Panel AD18XX configuration Unchained : 0x1000,0x0000,0x0000 Chained : 0x0000,0x0000,0x0000 root@ds19:~> cat /proc/asound/card0/codec97#0/ac97#0-0+regs 0:00 = 0090 0:02 = 0404 0:04 = 0606 0:06 = 0006 0:08 = 0000 0:0a = 0000 0:0c = 801f 0:0e = 0040 0:10 = 9f9f 0:12 = 0606 0:14 = 0000 0:16 = 9f9f 0:18 = 0606 0:1a = 0000 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 800f 0:28 = 0bc7 0:2a = 05f1 0:2c = bb80 0:2e = bb80 0:30 = bb80 0:32 = bb80 0:34 = 0000 0:36 = 0300 0:38 = 0202 0:3a = 2a20 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 1000 0:5e = 0000 0:60 = 8080 0:62 = 0000 0:64 = 0000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 8080 0:6e = 8080 0:70 = 0000 0:72 = 0000 0:74 = 1001 0:76 = 7c7c 0:78 = 0000 0:7a = 0000 0:7c = 4144 0:7e = 5378 root@ds19:~> cat /proc/asound/cards 0 [ICH6 ]: ICH4 - Intel ICH6 Intel ICH6 with AD1986 at 0xf0140800, irq 16 root@ds19:~> lspci 00:00.0 Host bridge: Intel Corporation 82915G/P/GV/GL/PL/910GL Memory Controller Hub (rev 04) 00:02.0 VGA compatible controller: Intel Corporation 82915G/GV/910GL Integrated Graphics Controller (rev 04) 00:02.1 Display controller: Intel Corporation 82915G Integrated Graphics Controller (rev 04) 00:1c.0 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 1 (rev 03) 00:1c.1 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 2 (rev 03) 00:1c.2 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 3 (rev 03) 00:1c.3 PCI bridge: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) PCI Express Port 4 (rev 03) 00:1d.0 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #1 (rev 03) 00:1d.1 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #2 (rev 03) 00:1d.2 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #3 (rev 03) 00:1d.3 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB UHCI #4 (rev 03) 00:1d.7 USB Controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) USB2 EHCI Controller (rev 03) 00:1e.0 PCI bridge: Intel Corporation 82801 PCI Bridge (rev d3) 00:1e.2 Multimedia audio controller: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) AC'97 Audio Controller (rev 03) 00:1f.0 ISA bridge: Intel Corporation 82801FB/FR (ICH6/ICH6R) LPC Interface Bridge (rev 03) 00:1f.1 IDE interface: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) IDE Controller (rev 03) 00:1f.2 IDE interface: Intel Corporation 82801FB/FW (ICH6/ICH6W) SATA Controller (rev 03) 00:1f.3 SMBus: Intel Corporation 82801FB/FBM/FR/FW/FRW (ICH6 Family) SMBus Controller (rev 03) 07:00.0 Ethernet controller: Broadcom Corporation NetXtreme BCM5751 Gigabit Ethernet PCI Express (rev 01) root@ds19:~> lsmod | grep snd snd_intel8x0 31420 1 snd_ac97_codec 89412 1 snd_intel8x0 ac97_bus 2432 1 snd_ac97_codec snd_pcm_oss 39232 0 snd_mixer_oss 15488 1 snd_pcm_oss snd_pcm 72036 3 snd_intel8x0,snd_ac97_codec,snd_pcm_oss snd_timer 20964 1 snd_pcm snd 48196 8 snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 7616 1 snd snd_page_alloc 9736 2 snd_intel8x0,snd_pcm regards, Jörg Grüße, Jörg -- PGP Key: send mail with subject 'SEND PGP-KEY' PGP Key-ID: FD 4E 21 1D PGP Fingerprint: 388A872AFC5649D3 BCEC65778BE0C605

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[alsa-devel] 2.6.21: known regressions with patches (for -stable team)
by Michal Piotrowski 02 May '07

02 May '07

Hi all, Here is a list of known regressions with patches available. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions Subject : 2.6.21: longhaul freq-scaling broken References : http://lkml.org/lkml/2007/5/1/260 Submitter : Jan Kiszka <jan.kiszka(a)web.de> Patch : http://git.kernel.org/?p=linux/kernel/git/davej/cpufreq.git;a=commitdiff_pl… Status : patch available Subject : Kernel hang at "kernel: switched to high resolution mode on cpu 0" when either/both of CONFIG_NO_HZ, CONFIG_HIGH_RES_TIMERS is used References : http://lkml.org/lkml/2007/4/30/464 Submitter : Mark Lord <lkml(a)rtr.ca> Handled-By : Daniel Walker <dwalker(a)mvista.com> Thomas Gleixner <tglx(a)linutronix.de> Patch : http://lkml.org/lkml/2007/4/13/190 Status : patch was suggested Subject : ACPI related kernel panic References : http://lkml.org/lkml/2007/4/30/193 http://bugzilla.kernel.org/show_bug.cgi?id=8385 Submitter : Sebastian Kemper <sebastian_ml(a)gmx.net> Patches : http://bugzilla.kernel.org/attachment.cgi?id=11331&action=view http://bugzilla.kernel.org/attachment.cgi?id=11332&action=view Status : patches was suggested Subject : sis900 oops in 2.6.21.1 after bringing up the network References : http://lkml.org/lkml/2007/4/29/138 http://lkml.org/lkml/2007/4/29/151 Patch : http://git.kernel.org/?p=linux/kernel/git/torvalds/linux-2.6.git;a=commitdi… Submitter : Brett Ryland <nappers(a)gmail.com> Status : patch available Subject : reiserfs -- circular locking References : http://lkml.org/lkml/2007/4/26/648 Submitter : David Brownell < david-b(a)pacbell.net > Handled-By : Jeff Mahoney < jeffm(a)suse.com > Patch : http://lkml.org/lkml/2007/4/27/332 Status : patch available Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger < markus.rechberger(a)amd.com > Status : patches available Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski < michal.k.k.piotrowski(a)gmail.com > Handled-By : Takashi Iwai < tiwai(a)suse.de > Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available Subject : qla2xxx driver no longer works References : http://lkml.org/lkml/2007/4/16/49 http://lkml.org/lkml/2007/4/20/224 Submitter : David Miller < davem(a)davemloft.net > Robert Peterson < rpeterso(a)redhat.com > Handled-By : David Miller < davem(a)davemloft.net > Patch : http://lkml.org/lkml/2007/4/16/234 Status : patch was suggested Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] [PATCH] snd-usb-audio: explicitly match Logitech QuickCam
by Daniel Drake 02 May '07

02 May '07

Commit 93c8bf45e083b89dffe3a708363c15c1b220c723 modified the USB device matching behaviour to ignore interface class matches if the device class is vendor-specific. This patch adds an explicit ID match for the Logitech QuickCam Messenger, which has a vendor specific device class (but standards-compliant audio interfaces). This fixes a 2.6.20 regression where the audio component of this device was no longer usable. http://bugs.gentoo.org/show_bug.cgi?id=175715 https://bugs.launchpad.net/ubuntu/+source/linux-source-2.6.20/+bug/93822 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3040 Based on a patch from sergiom 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 @@ -40,6 +40,21 @@ .bInterfaceClass = USB_CLASS_VENDOR_SPEC /* + * Logitech QuickCam: bDeviceClass is vendor-specific, so generic interface + * class matches do not take effect without an explicit ID match. + */ + +{ + .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 +}, + +/* * Yamaha devices */

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[alsa-devel] ALC260 patch for Packard Bell Easy Note v7900
by Andrea Gozzi 02 May '07

02 May '07

Please see if you can accept this patch that fixes sound/headphones/mic on Packard Bell EN v7900 notebook (ALC260 chipset). The code is not mine, I diff'ed using pshou's (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2189) but since it works perfectly it would be nice to see it in the next alsa release. Andrea

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[alsa-devel] [ALSA][hda-codec] fix-connexant-headphone-mute
by Andrei Popa 02 May '07

02 May '07

Hi, The sound works ok with the built-in speakers on my Fujitsu laptop(amilo pro v3205). But if I plug in headphones in the audio out jack there is no sound in the headphones. With kernel 2.6.20 it worked ok. With 2.6.21 it doesn't. This BUG is reported in ALSA bugtracking system with ID 0003039. Patch that fixes this bug: diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 46e93c6..5edd330 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -636,7 +636,7 @@ static void cxt5045_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; snd_hda_codec_amp_update(codec, 0x10, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x10, 1, HDA_OUTPUT, 0, 0, bits); } /* unsolicited event for HP jack sensing */ @@ -988,10 +988,10 @@ static void cxt5047_hp_automute(struct hda_codec *codec) bits = (spec->hp_present || !spec->cur_eapd) ? 0x80 : 0; snd_hda_codec_amp_update(codec, 0x1d, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x1d, 1, HDA_OUTPUT, 0, 0, bits); /* Mute/Unmute PCM 2 for good measure - some systems need this */ snd_hda_codec_amp_update(codec, 0x1c, 0, HDA_OUTPUT, 0, 0x80, bits); - snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0x80, bits); + snd_hda_codec_amp_update(codec, 0x1c, 1, HDA_OUTPUT, 0, 0, bits); } /* toggle input of built-in and mic jack appropriately */

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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Giridhar Pemmasani 02 May '07

02 May '07

--- Adrian Bunk <bunk(a)stusta.de> wrote: > On Sun, Apr 29, 2007 at 09:55:18AM -0700, Giridhar Pemmasani wrote: > > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails > to > > boot - it seems to stop at various points depending on what other modules > are > > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with > everything > > else being same, using maestro3.c from 2.6.20 kernel and using it in > 2.6.21 > > kernel also works - i.e., system boots fine. I will try to isolate the > issue > > and follow up. > > There were only the two attached patches between 2.6.20 and 2.6.21 to > this driver: > - maestro3: add request_firmware() > - maestro3 - Use quirk list helper function > > 2.6.21 + 2.6.20 maestro3.c works for you. > > Does 2.6.21 + 2.6.20 maestro3.c + patch-maestro3-1 work? 2.6.21 with 2.6.20's maestro3.c with patch-maestro3-1 exhibited the same issue. However, I identified the cause: maestro3 driver in 2.6.21 (even just patch-maestro3-1) tries to load firmware file from user space. But each time request_firmware is called it takes about 1 minute to timeout (because I don't have the firmware installed?). If I let the system take its time, even with maestro3.c from 2.6.21, it boots after timing out twice. So after about 2 minutes of no activity, system does boot. Following are trace messages I added to identify the issue: [ 17.004000] input: TPPS/2 IBM TrackPoint as /class/input/input5 [ 70.904000] sound/pci/maestro3.c (snd_m3_create:2763): -2 [ 70.920000] sound/pci/maestro3.c (snd_m3_create:2774): -2 [ 130.940000] sound/pci/maestro3.c (snd_m3_create:2777): -2 [ 130.956000] sound/pci/maestro3.c (snd_m3_create:2788): -2 [ 130.972000] sound/pci/maestro3.c (snd_m3_create:2794): 0 [ 130.992000] sound/pci/maestro3.c (snd_m3_create:2797): 0 [ 131.012000] sound/pci/maestro3.c (snd_m3_create:2802): 0 [ 131.044000] sound/pci/maestro3.c (snd_m3_create:2804): 0 [ 131.580000] sound/pci/maestro3.c (snd_m3_create:2807): 0 [ 131.596000] sound/pci/maestro3.c (snd_m3_create:2810): 0 [ 131.612000] sound/pci/maestro3.c (snd_m3_create:2814): 0 [ 131.628000] sound/pci/maestro3.c (snd_m3_create:2821): 0 [ 132.364000] device-mapper: ioctl: 4.11.0-ioctl (2006-10-12) initialised: dm-d evel(a)redhat.com Giri __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Audio DMA buffer layout and setup
by Markus Korber 02 May '07

02 May '07

Hi, I'm working with a custom sound chip with its own DMA hardware. The figure shows how the I2S DMA expects data to be prepared: IRQ1 +----------------+ | page_3 (right) |\ +----------------+ > double buffer 1 | page_2 (left) |/ IRQ0 +----------------+ | page_1 (right) |\ +----------------+ > double buffer 0 | page_0 (left) |/ +----------------+ The hardware has 2 double buffers, each containing left and right sample regions, i.e. it alternately transmits samples from page_0 and page_1, i.e. double buffer 0, then generates an interrupt and starts transmitting page_2 and page_3. After that another interrupt is generated and the process begins again with double buffer 0. So my question is, how to tell ALSA about this memory layout? Is it possible to operate in MMAP mode with SNDRV_PCM_INFO_NONINTERLEAVED? But then, how to tell the application that it should either fill double buffer 0 or 1? Or should I use RW mode? Or something else? Furthermore, since the hardware doesn't provide a 'current' DMA pointer containing the DMA address being transmitted at the moment (I only have interrupts generated after each double buffer has been transmitted), what should the pcm_pointer() callback return? (I guess it should return 0 when called before IRQ0 and sizeof(double buffer 0) between IRQ0 and IRQ1?) Best regards, Markus Korber

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[alsa-devel] driver mixer control questions
by Tim Harvey 02 May '07

02 May '07

I'm troubleshooting an issue with an ALSA driver I wrote for a very unique hardware platform. In this case I have an audio codec which only has a usable 'line-in' capture source. Currently the only mixer control I've created is a 'Master Playback Volume'. Do I need to create some kind of control for 'line-in'? I'm unclear how to 'register' capture inputs so that they will be returned by the OSS mixer emulation's ioctl SOUND_MIXER_READ_RECMASK. Currently that ioctl returns 0 for my mixer device. Also, is a control 'switch' such as 'Master Playback Switch' simply a mute control or is it intended to be any type of control which could have enumerated values? Thanks, Tim

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Re: [alsa-devel] Support for SAM9407?
by Ash Willis 02 May '07

02 May '07

Thanks for your detailed response. That chip definitely does seem like a beast that I really don't want to tame... > > How about a Media Vision Pro Audio Spectrum 16? Those can still be found > relatively easily and it's one the cards for which an OSS/Free driver is > still kept in the tree. I have a pair of those as well so I'll happily test > if you want to write a driver for it... > Yeah, we have a winner. I just ordered one. I'm busy right now but I will get this started in 4 weeks. Thanks for the offer of testing. I'll hunt you down when I have something to show you :) Ash =

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[alsa-devel] porter wanted: AMD Geode SCx200 driver OSS to ALSA
by Martin-Éric Racine 02 May '07

02 May '07

Greetings, I work for a company that develops hardware using the AMD Geode chipset. While our current product line is based on the Geode LX model (which is very well supported in ALSA), our legacy products are based on the earlier Geode SC1200 and SC2200, for which no ALSA support exists. However, AMD has OSS driver source code available at: http://www.amd.com/us-en/ConnectivitySolutions/TechnicalResources/0,,50_233… We basically need this ported to an ALSA snd-scx200 module to replace it. (Would now also be a good time to ask for snd-sc5535audio to be renamed snd-sc5535 for consistence with other driver names?) We could provide a sample SC2200-based unit to whoever volunteers for porting AMD's old OSS code to ALSA, Please contact me for details. The resulting code will be directly submitted to the ALSA project for inclusion in the mainstream kernel, to make sure that everyone can benefit from it. Best Regards, -- Martin-Éric Racine http://q-funk.iki.fi

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Re: [alsa-devel] Endiannes and alsa (in an embedded device)
by Ciaccia 02 May '07

02 May '07

Forget about that, it took 4 hours but I understood why it was not working... Since I'm working with 24bit resolution (and not 16 as I usually do), the samples were to little... Now I replaced the sin line with: sample = (int)(4000000.0f * sin((float)i * 2.0f * M_PI / (512.0f / 50.0f))); and it's working fine. I really thought it was too strange to be true... Sorry for the spam :-| Andrea ----- Original Message ---- From: Ciaccia <ciacciax(a)yahoo.com> To: alsa-devel(a)alsa-project.org Sent: Tuesday, May 1, 2007 8:49:18 PM Subject: [alsa-devel] Endiannes and alsa (in an embedded device) Dear all, I have a question that really puzzles me, I wonder if someone can explain to me what I'm doing wrong... :-| I have programmed an ALSA driver for an embedded device, everything works fine and all the application use my driver without any glitches. The snd_pcm_hardware for this device is defined as following: static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info= SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .formats = SNDRV_PCM_FMTBIT_S24_LE, .buffer_bytes_max = 131072, .period_bytes_min = 512, .period_bytes_max = 32768, .periods_min = 1, .periods_max = MAX_PERIODS, .fifo_size = 32, }; Then I tried to program a test application that uses the hardware device directly (without using alsa-lib to convert the format and so on). I open the device as following: if ((err = snd_pcm_open (&playback_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0)) < 0) { Everything works fine until I try to write data to the stream. I tried both SND_PCM_ACCESS_RW_INTERLEAVED and SND_PCM_ACCESS_MMAP_INTERLEAVED, but I always have the following problem. When I want to write data to the stream, I have to "reverse" the byte order, otherwise I hear noise on the speaker. The system I am working on is little-endian (it has cirrus ep93xx arm core), so I assume I could write data directly to the buffer. Here is what I have to do: // fill the buffer for (i = 0; i < 512; i++) { int sample = (int)(20000.0f * sin((float)i * 2.0f * M_PI / 512.0f)); int reversed; char *src = (char *)(&sample); char *dst = (char *)(&reversed); dst[0] = src[3]; dst[1] = src[2]; dst[2] = src[1]; dst[3] = src[0]; *(buffer++) = reversed; *(buffer++) = reversed; } It this normal? I thought that since the system is little-endian and the format of the driver is S24_LE, I could copy the data directly to the stream (either using snd_pcm_writei or snd_pcm_mmap_begin). I think the driver is ok, since all the applications run with no problem at all. Can someone explain to me how/why the applications such as aplay/arecord work ok? Every help is greatly appreciated Thanks in advance Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Endiannes and alsa (in an embedded device)
by Ciaccia 01 May '07

01 May '07

Dear all, I have a question that really puzzles me, I wonder if someone can explain to me what I'm doing wrong... :-| I have programmed an ALSA driver for an embedded device, everything works fine and all the application use my driver without any glitches. The snd_pcm_hardware for this device is defined as following: static const struct snd_pcm_hardware ep93xx_pcm_hardware = { .info= SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER, .rates = SNDRV_PCM_RATE_8000_48000, .rate_min = 8000, .rate_max = 48000, .channels_min = 2, .channels_max = 2, .formats = SNDRV_PCM_FMTBIT_S24_LE, .buffer_bytes_max = 131072, .period_bytes_min = 512, .period_bytes_max = 32768, .periods_min = 1, .periods_max = MAX_PERIODS, .fifo_size = 32, }; Then I tried to program a test application that uses the hardware device directly (without using alsa-lib to convert the format and so on). I open the device as following: if ((err = snd_pcm_open (&playback_handle, "hw:0,0", SND_PCM_STREAM_PLAYBACK, 0)) < 0) { Everything works fine until I try to write data to the stream. I tried both SND_PCM_ACCESS_RW_INTERLEAVED and SND_PCM_ACCESS_MMAP_INTERLEAVED, but I always have the following problem. When I want to write data to the stream, I have to "reverse" the byte order, otherwise I hear noise on the speaker. The system I am working on is little-endian (it has cirrus ep93xx arm core), so I assume I could write data directly to the buffer. Here is what I have to do: // fill the buffer for (i = 0; i < 512; i++) { int sample = (int)(20000.0f * sin((float)i * 2.0f * M_PI / 512.0f)); int reversed; char *src = (char *)(&sample); char *dst = (char *)(&reversed); dst[0] = src[3]; dst[1] = src[2]; dst[2] = src[1]; dst[3] = src[0]; *(buffer++) = reversed; *(buffer++) = reversed; } It this normal? I thought that since the system is little-endian and the format of the driver is S24_LE, I could copy the data directly to the stream (either using snd_pcm_writei or snd_pcm_mmap_begin). I think the driver is ok, since all the applications run with no problem at all. Can someone explain to me how/why the applications such as aplay/arecord work ok? Every help is greatly appreciated Thanks in advance Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Implementing Priority Based Device ALSA device interface. ALSA drivers
by Nobin Mathew 01 May '07

01 May '07

Hi Friends, I am a new to ALSA, I am developing a ALSA ASoC driver. Now driver can support both audio playback and capture. Now the problem is handling of application with different priority. We have three applications say app1, app2 and app3; they are in increasing order of priority. One scenario of app1 and app2 is below If app2 is accessing the device while app1 playback is going on, then app1 should stop (clearly do pause) playback and allow app2 to playback, and app1 should continue playback when app2 exit. The requirement is that there should be no changes to apps, except the devices each app will access. There will be no daemons in user space which will handle priority. For that i implemented three interfaces like pcmC0p0, pcmC0p1 and pcmC0p2 there are in increaing order of priority. All these interfaces will access the same audio hardware. Only thing is each will have separate ALSA kernel buffer. In this implementation app1 will always access pcmC0p0, and app2 pcmC0p1 like that. What i am doing is, in driver i will pause the pcmC0p0 substream using snd_pcm_pause() when pcmC0p1 is accessed. Same for pcmc0p2. This call will change the state of pcmc0p0 substream to SNDRV_PCM_STATE_PAUSED, and this call should notify the user app also. When app2 playback is over i will change the state of pcmC0p0 substream to SNDRV_PCM_STATE_RUNNING by another call to snd_pcm_pause(). This implementation requires some minor modifications in the ALSA lib and ALSA app (handling of PAUSE from kernel). But what i found is kernel is not notifying the user using snd_timer_notify() (user call back) function because i am using the substream timer snd_pcm_timer which is there in pcm_timer.c. For this timer SNDRV_TIMER_HW_SLAVE flag is set so it will not notify the user. What is the use of these timers in ALSA? Am i going wrong? Is my design correct? Can u suggest some alternate design? Sorry for my poor English. Thanks Nobin Mathew

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[alsa-devel] Set fragment size for USB sound card
by Jason Heeris 01 May '07

01 May '07

System: NEC Versa M540 laptop Dist: Debian Etch/Lenny Kernel: 2.6.18 Alsa: using drivers in kernel, although no difference when compiled from alsa-source package. Card in question: C-Media USB sound card (more info below) Hi there, I realise that this may not be strictly a developer list question, but someone on the alsa-user list suggested I try here. Any help would be appreciated. The question is a continuation of: http://sourceforge.net/mailarchive/forum.php?thread_name=1176015858.4497.9.… and http://www.linuxquestions.org/questions/showthread.php?t=540205 Essentially, I have a C-Media USB audio adaptor with which to use my headphones, and it only works with a fragment size of 2048 or less (discovered using 'alsaplayer -f 2048'). I would like to be able to plug it in, have ALSA recognise it and use this fragment size for it. Without specifying this fragment size, I get: -- $ alsaplayer -i text -o alsa -d hw:2 /usr/share/sounds/generic.wav Unable to install hw params: ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 44100 PERIOD_TIME: (23219 23220) PERIOD_SIZE: 1024 PERIOD_BYTES: 4096 PERIODS: 8 BUFFER_TIME: (185759 185760) BUFFER_SIZE: 8192 BUFFER_BYTES: 32768 TICK_TIME: 1000 failed to configure output device...trying OSS ...done playing -- ...and the sound comes through the laptop speakers. I tried defining a slave PCM: -- pcm.usbtest { type dmix ipc_key 192837465 ipc_key_add_uid true slave { pcm "hw:2,0" # period_size 2048 buffer_size 2048 } } -- But trying 'alsaplayer -i text -o alsa -d usbtest /usr/share/sounds/generic.wav' gave exactly the same error. >From my previous post: -- ...I also tried forcing the index of the card by adding the following to /etc/modprobe.d/alsa-base, but even this only works half of the time: -- # Prevent abnormal drivers from grabbing index 0 options snd-bt87x index=-2 options snd-cx88_alsa index=-2 options snd-atiixp-modem index=-2 # options snd-intel8x0m index=-2 options snd-via82xx-modem index=-2 options snd-intel8x0 index=0 options snd-intel8x0m index=1 options snd-usb-audio index=2 vid=0d8c pid=000c options snd-usb-audio index=3 -- The contents of '/proc/asound/cards' are: -- 0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with ALC655 at 0xf0000400, irq 5 1 [Modem ]: ICH-MODEM - Intel 82801DB-ICH4 Modem Intel 82801DB-ICH4 Modem at 0xe200, irq 5 2 [default ]: USB-Audio - C-Media USB Headphone Set C-Media USB Headphone Set at usb-0000:00:1d.1-1, full speed -- ...although sometimes the ICH-MODEM is missing and the C-Media card is at #1. Card information from 'lsusb': Bus 002 Device 003: ID 0d8c:000c C-Media Electronics, Inc. Audio Adapter Usually I boot with the card plugged in, but if I plug it in after booting, the new modules loaded are: snd_hwdep, snd_rawmidi, snd_usb_audio and snd_usb_lib. -- Anyway, as I said, any help would be appreciated. Cheers, Jason

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[alsa-devel] Possible bug in snd_mixer_selem_set_playback_switch_all
by Tim Burrell 30 Apr '07

30 Apr '07

Hi there, I've encountered a situation where if I very quickly, and repeatedly make calls to snd_mixer_selem_set_playback_switch_all I get some strange undefined behavior. The situation is I'm looping over all mixer elements (on all sound cards) to do a system wide mute / unmute. If I make numerous calls to this function it ends up that some mixers only toggle switches on specific channels (right or left), and sometimes the volume on certain (random) channels gets set to some also random (usually 0, but can be anywhere between 0 and max) level. I've been able to remedy the situation by inserting a 10ms delay in between calls to snd_mixer_selem_set_playback_switch_all. Clearly though this is undesirable. I don't have more than one sound card to test on so I can't be sure if it's a problem with all drivers or just mine (hda-intel). I can certainly provide more information if requested, just let me know what I can do. Or if I'm doing something wrong I'd love to be made aware of something I can do to fix this behavior. Thanks, Tim.

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[alsa-devel] List of all Apple Mac's hda-intel pin configurations
by Ivan N. Zlatev 30 Apr '07

30 Apr '07

I have attached all windows drivers' .ini files (taken from latest windows drivers cd of Boot Camp), which include all pin configuration as well as probably some more useful information (GPIO and stuff). I hope this will help you to make a nice update. [Subsystem ID - configuration] 106B0800 - Board1.ini 106B0600 - Board2.ini 106B0700 - Board2.ini 106B0E00 - Board3.ini 106B0F00 - Board3.ini 106B1600 - Board3.ini 106B1700 - Board3.ini 106B0200 - Board3.ini 106B1E00 - Board3.ini 106B0400 - Board4.ini 106B1A00 - Board4.ini 106B0A00 - Board5.ini 106B2200 - Board5.ini 00000100 - Board4.ini Regards. -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work? [Solved]
by Cody Jung 30 Apr '07

30 Apr '07

On 4/29/07, Rene Herman <rene.herman(a)gmail.com> wrote: > On 04/29/2007 09:31 AM, Rene Herman wrote: > > > I've attached (to try and avoid linewrap) the lines that you will need in > > /etc/modprobe.conf. > > No I haven't. > > Rene. > > alias snd-card-0 snd-cs4231 > alias sound-slot-0 snd-cs4231 > > options snd-cs4231 index=0 port=0x534 irq=5 dma1=1 dma2=0 > > install snd-cs4231 echo activate >/sys/bus/pnp/devices/01\:01.00/resources &&\ > sleep 1 &&\ > /sbin/modprobe --ignore-install snd-cs4231 > > remove snd-cs4231 /sbin/modprobe -r --ignore-remove snd-cs4231 &&\ > echo disable >/sys/bus/pnp/devices/01\:01.00/resources > > Wahoo! Thank you, thank you! I ended up not going the /etc/modules way, partially because it wasn't working and partially because I didn't know what I was doing. Instead I added the following 4 lines to a file and included it in my init scripts. echo activate > /sys/bus/pnp/devices/01\:01.00/resources sleep 1 modprobe snd modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 And it works great! Thanks again for putting up with my incompetence. ~Cody Jung

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Re: [alsa-devel] Boot failure with 2.6.21 with snd-maestro3
by Michal Piotrowski 29 Apr '07

29 Apr '07

Hi, On 29/04/07, Giridhar Pemmasani <pgiri(a)yahoo.com> wrote: > When snd-maestro3 (ESS Maestro 3 driver) is enabled, kernel 2.6.21 fails to > boot - it seems to stop at various points depending on what other modules are > enabled. Disabling snd-maestro3 fixes all issues. Moreover, with everything > else being same, using maestro3.c from 2.6.20 kernel and using it in 2.6.21 > kernel also works - i.e., system boots fine. I will try to isolate the issue > and follow up. Please revert this patches http://www.stardust.webpages.pl/files/patches/alsa_revert/ and test kernel after each revert. Takashi, I'll add this to known regressions for -stable team. > > Giri Regards, Michal -- Michal K. K. Piotrowski Kernel Monkeys (http://kernel.wikidot.com/start)

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[alsa-devel] Can't record sound on Aser Aspire 5100. ATI SB450, RelTek ALC 883
by xinyun liu 29 Apr '07

29 Apr '07

hi, I have an Acer Aspire 5100. I can play sound but with the build in mic, I can't record sound. The driver is the latest HG code. http://pastebin.ca/464406 [root@localhost ~]# lspci -nnv -d1002:437b 00:14.2 0403: 1002:437b (rev 01) Subsystem: 1025:009f Flags: bus master, slow devsel, latency 64, IRQ 19 Memory at c0000000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- [root@localhost ~]# dmesg | fgrep ALSA > /tmp/1 [root@localhost ~]# vi /tmp/1 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:698: codec_mask = 0xb ALSA ~/alsa-driver/pci/hda/hda_codec.c:1820: hda_codec: model 'acer' is selected for config 1025:0 (Acer laptop) ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 ALSA ~/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:1188: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x4, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x5, stream=0x5, channel=0, format=0x11 ALSA ~/alsa-driver/pci/hda/hda_codec.c:630: hda_codec_setup_stream: NID=0x2, stream=0x0, channel=0, format=0x0

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/29/2007 04:18 AM, Cody Jung wrote: > Er...that appears to be a .diff file. Do I have to do something with it > for it to be workable, or...? Yes. You'd have to apply the patch (as it's called) to a recent kernel source tree, reconfigure the kernel adding this new VIVO driver to your config and then recompile and reinstall that kernel. Oh well, you posting to alsa-user and not alsa-devel should've been a hint. Rest assured the driver would've worked nicely :-/ There is another thing you can do. What was posted was just a wrapper around cs4231-lib in much the same way that the existing snd-cs4231 module is and that module you _will_ already have installed. That one doesn't do anything automatic though, so you will have to know how to manually enable your card first. First, open a root shell and look through the /sys/bus/pnp/devices directory for your VIVO. Under the "xx:yy.zz" sub-directories there you will find "id" files, and one of them will say "ENS1011". Example: # cat /sys/bus/pnp/devices/01\:01.00/id ENS1011 After having found it in there, activate the card: # echo activate >/sys/bus/pnp/devices/01\:01.00/resources and look at what resources were assigned to it by the PnP layer: # cat /sys/bus/pnp/devices/01\:01.00/resources state = active io 0x330-0x33f io 0x530-0x537 io 0x2100-0x213f irq 5 irq 9 dma 1 dma 0 You can now load the snd-cs4231 module specifying those card resources manually: # modprobe snd-cs4231 port=0x534 irq=5 dma1=1 dma2=0 Yes, 0x534. The second io resource + 4. This should've loaded without problem. Start up alsamixer: # alsamixer and up and unmute the PCM volume. With -> and <- you can walk through the sliders; ^ ups the one you're on, and "m" toggles the mute (the "MM" at the bottom should turn to a highlighted "OO"). PCM is the first slider. 90 seems to be an okay value. Quit alsamixer again with Alt-Q and now you should be able to hear things. If you have a plain "wav" file around somewhere, that makes for the quickest test: # aplay foo.wav No, I'm not going to tell you to plug in your speakers. If anything in the above doesn't work as expected, your system is not as expected and there's isn't much I can help with. If everything does work, you can automate it for the next time. I've attached (to try and avoid linewrap) the lines that you will need in /etc/modprobe.conf. Ubuntu has no doubt found a way to make that dark and muddy again so you might need to add it to an /etc/modprobe/whatever file instead. Please ask a fellow ubuntu victim for advice about translating this generic linux advice to your install. Once you do have that all setup, sound should "Just Work". You probably need to adjust the volume as specified above once only after which ubuntu will save it for the next boot, but if you don't hear anything ater rebooting, please check that first. Rene.

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Re: [alsa-devel] [Alsa-user] ALSA Newbie - Getting Ensoniq SoundscapeVivo to work?
by Rene Herman 29 Apr '07

29 Apr '07

On 04/28/2007 07:06 AM, Cody Jung wrote: > As it suggests in the subject, I'm both a Linux-newbie and an > ALSA-newbie. I have an old clunker of a machine running Ubuntu Feisty > that I built out of parts left on curbs during spring cleanup, and that's > how I came across my sound card. It's an Ensoniq SoundscapeVivo (Not a > Vivo90, AFAIK). As far as I'm aware, there are no other VIVO's than the VIVO90. > I've gone through install and troubleshooting guides galore, but I can't > seem to get ALSA to recognize my card, and I'm confused and rather tired > of it all. > > Do you think you can help, and if so, what information do you need from > me? I probably can; I have both a Soundscape VIVO90 and a much older Soundscape S-2000. My VIVO90 is an ISA-PnP card and has a Ensoniq MARK5-RO, an OTTOR2C and a Analog Devices AD1845 codec: http://en.wikipedia.org/wiki/Ensoniq_Soundscape_VIVO90 I tested that when I acquired it a while ago and if I remember correctly, it just worked... mmm. come to think of it, it might be the case that I needed to add its PnP ID to the driver. Not sure anymore. Could you confirm whether or not your Soundscape is an ISA-PnP card and if so (that's also a way to find out if you don't know if it is) run "pnpdump" with it installed and look for a PnP-ID for the card in it? Something of the form "ENSxxxx". If your card is one of the older non-PnP soundscapes it's going to be a bit harder; they also need firmware and all that and I haven't tested that card myself yet. (please use the @alsa-project.org list addresses by the way; in the case of alsa-devel, the sourceforge one is gone even). Rene.

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[alsa-devel] Apple iMac pin configs
by Ivan N. Zlatev 28 Apr '07

28 Apr '07

Hello, I have a 20'' iMac machine with a hda-intel (SigmaTel 9221 A1 codec) and I have successfully been using the mac mini pin config, but there is one problem - the microphone doesn't work. The machine has a built-in microphone, built-in speakers, 1 digitial out jack and 1 for headphones. With the mac mini pin config I get 2 mics: "Front Mic" and "Mic" but none of those works. (alsamixer-macmini.gz for info). Today I spent some time on the search for the the correct pin configuration and it should be this [1] according to the windows drivers. This configuration causes amixer.gz, basically 1 "Mic" (which again doesn't record), but also "Mic as Output" and "Line in as output" (I don't have a line in as far as i know). I have also tried the macbook-pro and macbook introduced with 1.0.14rc3, but the mic doesn't work. I have attached * cat /proc/asound/card0/codec#0 * Mac OS X's complete dump of information of the audio driver - imac20-ioreg-audio.gz * amixer with the below config and the one with the macmini pin config Can you possible assist me to get the below pin config in tune, so that everything works? Regards. [1] ========== [HKR\Settings\pin] [HKR\Settings\Pin\0A] CfgDflt = dword: 0x0121E230 Action = hex: 01,0A,81,0C,81,10 [HKR\Settings\Pin\0B] CfgDflt = dword: 0x90A70120 [HKR\Settings\Pin\0C] CfgDflt = dword: 0x9017E110 [HKR\Settings\Pin\0D] CfgDflt = dword: 0x400000FE [HKR\Settings\Pin\0E] CfgDflt = dword: 0x400000FD Action = hex: 01,10,81,0A,81,0C [HKR\Settings\Pin\0F] CfgDflt = dword: 0x0181E02E Action = hex: 09,0F [HKR\Settings\Pin\10] CfgDflt = dword: 0x1145E240 [HKR\Settings\Pin\11] CfgDflt = dword: 0x500000FA [HKR\Settings\Pin\15] CfgDflt = dword: 0x400000FC [HKR\Settings\Pin\1B] CfgDflt = dword: 0x400000FB -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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Re: [alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 27 Apr '07

27 Apr '07

Thanks to the help I've gotten on this list my alsa driver is working great. I got a little hung up not realizing that the alsa core basically provides everything you need to implement your own circular buffer if you need to use an intermediate buffer. The only question I still have is that if alsa allocates your DMA buffer for you based on snd_pcm_hardware_t buffer_bytes_max what exactly does snd_pcm_lib_preallocate_pages_for_all allocate buffers for? Does this allocate the buffers that would be passed to the copy callback? Thanks, Tim --- Tim Harvey <tim_harvey(a)yahoo.com> wrote: > Greetings, > > I'm setting out to write an ALSA driver for a very limited device thats > behind > a PLD. The actual codec is a uda1380 and does 16bit MSB stereo > encode/decode. > The way the device is driven the sampling rate is limited to 8KHz and data > transfer to/from the device I have to handle manually. > > I've read over the 'writing an ALSA driver' tutorial and have started the > driver but have run into a few questions: > > - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB > samples does the following snd_pcm_hardware_t look right?: > > static snd_pcm_hardware_t my_playback_hw = { > .info = (SNDRV_PCM_INFO_MMAP | > SNDRV_PCM_INFO_INTERLEAVED | > SNDRV_PCM_INFO_BLOCK_TRANSFER | > SNDRV_PCM_INFO_MMAP_VALID), > .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? > .rates = SNDRV_PCM_RATE_8000, // seems redundant? > .rate_min = 8000, > .rate_max = 8000, > .channels_min = 2, > .channels_max = 2, > .buffer_bytes_max = 32768, > .period_bytes_min = 4096, > .period_bytes_max = 32768, > .periods_min = 1, > .periods_max = 1024, > }; > > - I'm a little confused at how to deal with copying data from/to the device. > > It looks like I would implement the copy callback and copy the passed buffer > to > the device in the device specific manner I need to (playback), but I'm not > clear where to put data that I receive from the device (capture). I notice > that when the hw_params callback is called a dma buffer has been created by > ALSA in the substream->runtime struct. I assume this is created based on hte > buffer_bytes_max field above and I'm to copy data to this buffer? How do I > know where to copy it within the buffer and what data ALSA has already > consumed > from the buffer? > > - I might need to create an intermediate buffer between ALSA's rather large > buffers and my hardware buffer. Is there any driver that I could look at > that > does this? The tutorial talks about the vxpocket driver but that code looks > very foriegn from the rest of the ALSA drivers. > > - I'm not sure I understand what the 'period' is about. Its clear to me > that > I need to call snd_pcm_period_elapsed() periodically from either an IRQ or > other timed callback, but I'm not clear what the hardware pointer is or how > that converts to 'frames' and 'periods'. > > - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA > provides a mechanism for converting data from other formats? > > Thanks for any pointers, > > Tim >

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[alsa-devel] [ASoC] list corruption during AC97 registration
by Manuel Lauss 27 Apr '07

27 Apr '07

Hi Folks, I hit the following oops (on 2.6.21) when ASoC registers an AC97 codec. Happens both on MIPS and SH, for example the mips oops: asoc: AC97 HiFi <-> au1xpsc-ac97-1 mapping ok list_add corruption. prev->next should be next (80488f60), but was 865ffad8. (prev=865ffad8). <snip> Call Trace: [<80271290>] __list_add+0x68/0x94 [<802ac0e4>] bus_attach_device+0x48/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<c00882a8>] snd_soc_register_card+0x1d0/0x2c8 [snd_soc_core] [<c00b3268>] ac97_soc_probe+0x17c/0x1e4 [snd_soc_ac97] [<c0088f48>] soc_probe+0xb0/0x1ac [snd_soc_core] [<802ad410>] really_probe+0x148/0x268 [<802ac1cc>] bus_for_each_drv+0x50/0xa0 [<802ad758>] device_attach+0xe4/0x194 [<802ac0cc>] bus_attach_device+0x30/0x78 [<802aa4c4>] device_add+0x3d8/0x844 [<802aff24>] platform_device_add+0x13c/0x198 [<c00b71c8>] exm1200_mobo_ac97_init+0x1c8/0x1fc [snd_soc_exmmb_ac97] [<801478d0>] sys_init_module+0x1448/0x15b8 [<8010bcc0>] stack_done+0x20/0x3c System.map says: 80488e60 D ac97_bus_type As I said, it happens on both MIPS and SH (with machine and PCM drivers written by myself). With CONFIG_DEBUG_LIST disabled the code works and plays sound, but still I'd like to know what causes this list corruption. Thanks for any hints! -- Manuel Lauss

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[alsa-devel] dsnoop issues
by Gordon Miller 27 Apr '07

27 Apr '07

Hi All my snd_pcm_readi calls return too few frames. Also when I switch to stereo, the sound is muted. I cannot change this by adjusting period & buffer sizes. dsnoop is enabled by default. The documentation for snd_pcm_readi says that the only way that the frame count can be short is because of a signal or xrun. Well this clearly isn't the case. I think that direct snoop is the cause of this. Are there any remedies ? because I am roundly SICK of being told that my application must be wrong "because everyone else's works". I have spent the last 3 weeks debugging an app - that always worked properly - because of the faulty dsnoop implementation. It is abundantly clear that dsnoop is far from ready for use, but I am prepared to put some work into this. Whose idea was it to enable dsnoop by default, when it doesn't even work ? I was quite happy with OSS. Changing everything to be alsa entailed a lot of work, and I am beginning to think it was a complete waste of my time. ICE1712, 2.6.20.6 kernel, alsa 1.0.14rc1 Gordon

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[alsa-devel] New bugtracker categories
by Liam Girdwood 27 Apr '07

27 Apr '07

I'd like to request the following new categories for the ALSA bugtracker :- ASoC - platform - s3c24xx ASoC - platform - pxa2xx ASoC - platform - i.MX ASoC - platform - AT91xxxx ASoC - codecs ASoC - core Thanks Liam

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[alsa-devel] [ASoC] Sample format non available
by Markus Korber 27 Apr '07

27 Apr '07

Dear list, I'm currently implementing a ASoC solution for a custom chip and I've written a codec, I2S, PCM, and machine driver. However, when using aplay I get the following error (requested format was: 2): root:~# aplay -M -D hw:0,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo aplay: set_params:906: Sample format non available However, I've told each piece involved, that it is capable of playing S16_LE. So I don't know what I am missing here? Should I provide more debugging output like RULES_DEBUG output from sound/core/pcm_native.c? ,----[ CODEC DAI ] | #define CS4265_RATES (SNDRV_PCM_RATE_8000_192000) | #define CS4265_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE) | | struct snd_soc_codec_dai cs4265_dai = { | .name = "CS4265", | .playback = { | .stream_name = "Playback", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | .capture = { | .stream_name = "Capture", | .channels_min = 1, | .channels_max = 2, | .rates = CS4265_RATES, | .formats = CS4265_FORMATS,}, | [...] | }; `---- ,----[ CPU DAI ] | #define CHIP_I2S_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ | SNDRV_PCM_RATE_48000) | #define CHIP_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE) | | struct snd_soc_cpu_dai chip_i2s_dai = { | /* DAI description */ | [...] | /* DAI callbacks */ | [...] | /* DAI capabilities */ | .playback = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | .capture = { | .channels_min = 2, | .channels_max = 2, | .rates = CHIP_I2S_RATES, | .formats = CHIP_I2S_FORMATS,}, | /* ops */ | [...] `---- ,----[ HW DAI ] | static const struct snd_pcm_hardware chip_pcm_hardware = { | .info = SNDRV_PCM_INFO_MMAP | | SNDRV_PCM_INFO_MMAP_VALID | /* For OSS emulation */ | SNDRV_PCM_INFO_BLOCK_TRANSFER | /* For OSS emulation */ | SNDRV_PCM_INFO_NONINTERLEAVED, | .formats = SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S8 | | SNDRV_PCM_FMTBIT_U16_LE | SNDRV_PCM_FMTBIT_S16_LE | | .channels_min = 2, /* Stereo */ | .channels_max = 2, | .period_bytes_min = 32, | .period_bytes_max = 8192, | .periods_min = 2, | .periods_max = 1024, /* PAGE_SIZE/sizeof(chip_dma_desc), */ | .buffer_bytes_max = 32 * 1024, | .fifo_size = 0, /* Still unused in ALSA? */ | }; `---- root:~# uname -a Linux 2.6.21 root:~# cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). root:~# aplay --version aplay: version 1.0.14rc2 by Jaroslav Kysela <perex(a)suse.cz> root:~# aplay -M test.wav chip-pcm: Entered chip_pcm_open board_cs4265: Entered board_cs4265_startup Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo ALSA lib pcm_plug.c:840:(snd_pcm_plug_hw_refine_cchange) Unable to find an usable client format chip-pcm: Entered chip_pcm_hw_free chip-cs4265: Entered cs4265_shutdown chip-cs4265: Entered cs4265_write board_cs4265: Entered board_cs4265_shutdown chip-pcm: Entered chip_pcm_close ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U8 ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U16_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U32_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: FLOAT64_BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IEC958_SUBFRAME_LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: MU_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: A_LAW ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: IMA_ADPCM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: GSM ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: SPECIAL ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U24_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U20_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: S18_3BE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3LE ALSA lib pcm_plug.c:844:(snd_pcm_plug_hw_refine_cchange) Format: U18_3BE aplay: set_params:878: Broken configuration for this PCM: no configurations available Regards, Markus Korber

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[alsa-devel] How to play a pcm file
by Flávio Yuiti 26 Apr '07

26 Apr '07

Greetings, I'm a new user of alsa api and I need to play pcm, adpcm and wav files using Alsa api. My code is below. I did a test with pcm file and the sound is playing very fast, the properties of the pcm file are 128kbps, 16 bits little endian, 1 channel, 8 khz, but I don't know how to set correctly the parameters in api. Please help me with this problem. Point me my errors. Thanks, Flavio int Sound::PlayFile(char *pszFileName, ULONG initialPoint, UINT qtdTimesToPlay, BOOL mix) { int dir; int result; int size; unsigned int val; snd_pcm_hw_params_t *params; snd_pcm_uframes_t frames; char* buffer; long tamanhoBuffer; FILE* file; int bytesLidos; // Open PCM device for playback. if((result = snd_pcm_open(&handleSound,"hw:0,0",SND_PCM_STREAM_PLAYBACK,0)) < 0) { fprintf(stderr,"\nUnable to open pcm device: %s",snd_strerror(result)); exit(1); } // Allocate a hardware parameters object. snd_pcm_hw_params_alloca(&hardwareParam); // Fill it in with default values. snd_pcm_hw_params_any(handleSound,hardwareParam); // Set the desired hardware parameters. // Interleaved mode. snd_pcm_hw_params_set_access(handleSound,hardwareParam,SND_PCM_ACCESS_RW_INTERLEAVED); // Signed 16-bit little-endian format. snd_pcm_hw_params_set_format(handleSound,hardwareParam,SND_PCM_FORMAT_S16_LE); // Two channels (stereo), one channel mono. snd_pcm_hw_params_set_channels(handleSound,hardwareParam,1); // Set rate. //snd_pcm_hw_params_set_rate(handleSound,hardwareParam,8000,0); // Set period size to 16384 frames. //snd_pcm_hw_params_set_period_size(handleSound,hardwareParam,16384,0); // Write the parameters to the driver. if((result = snd_pcm_hw_params(handleSound,hardwareParam)) < 0) { fprintf(stderr,"\nUnable to set hw parameters: %s",snd_strerror(result)); exit(1); } tamanhoBuffer = 16384; if((result = snd_pcm_prepare(handleSound)) < 0) { fprintf(stderr,"Cannot prepare audio interface for use: %s\n",snd_strerror(result)); exit(1); } file = fopen(pszFileName,"rb"); buffer = (char *)malloc(tamanhoBuffer); while((bytesLidos = fread(buffer,1,tamanhoBuffer,file)) > 0) { if((result = snd_pcm_writei(handleSound,buffer,tamanhoBuffer)) != tamanhoBuffer) { fprintf(stderr,"Write to audio interface failed: %s\n",snd_strerror(result)); exit(1); } } snd_pcm_drain(handleSound); snd_pcm_close(handleSound); free(buffer); fclose(file); return OK; } __________________________________________________ Fale com seus amigos de graça com o novo Yahoo! Messenger http://br.messenger.yahoo.com/

2 2

[alsa-devel] [PATCH] PnP: Protocol for Media Vision Jazz16 sound cards
by Rask Ingemann Lambertsen 26 Apr '07

26 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add code to detect and PnP protocol to manage Media Vision Jazz16 sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20.6. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch depends on the patch "ALSA: Support Media Vision Jazz16 chips" because the patched include/sound/sb.h is needed. drivers/pnp/Kconfig | 12 drivers/pnp/Makefile | 1 drivers/pnp/jazz16pnp.c | 591 ++++++++++++++++++++++++++++++++++++++ 3 files changed, 604 insertions(+) Main changes since the last patch: Request just one DMA channel (1, 3, 5 or 7). Request port 0x388-0x38b for the OPL3 alias to avoid conflicts. Use the recommended resources from the manual for the preferred option. More SB port bases: 0x210, 0x230 and 0x250. diff -rup linux-2.6.20.6-clean/drivers/pnp/Kconfig linux-2.6.20.6-ril/drivers/pnp/Kconfig --- linux-2.6.20.6-clean/drivers/pnp/Kconfig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/drivers/pnp/Kconfig 2007-04-07 20:10:35.000000000 +0200 @@ -37,5 +37,17 @@ source "drivers/pnp/pnpbios/Kconfig" source "drivers/pnp/pnpacpi/Kconfig" +# +# Jazz16 Plug and Play configuration +# +config JAZZ16PNP + bool "Jazz16 sound card Plug and Play support" + depends on PNP && ISA + help + Say Y here if you would like support for automatic configuration + of I/O ports, DMA channels and IRQ lines of Jazz16 sound cards. + + Say Y if you have such a card. Otherwise, say N. + endmenu diff -rup linux-2.6.20.6-clean/drivers/pnp/Makefile linux-2.6.20.6-ril/drivers/pnp/Makefile --- linux-2.6.20.6-clean/drivers/pnp/Makefile 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/drivers/pnp/Makefile 2007-04-07 20:10:35.000000000 +0200 @@ -7,3 +7,4 @@ obj-y := core.o card.o driver.o resourc obj-$(CONFIG_PNPACPI) += pnpacpi/ obj-$(CONFIG_PNPBIOS) += pnpbios/ obj-$(CONFIG_ISAPNP) += isapnp/ +obj-$(CONFIG_JAZZ16PNP) += jazz16pnp.o --- /dev/null 2007-04-11 19:30:17.320748640 +0200 +++ linux-2.6.20.6-ril/drivers/pnp/jazz16pnp.c 2007-04-11 21:41:39.000000000 +0200 @@ -0,0 +1,591 @@ +/* + * Media Vision Jazz16 Plug & Play support. + * + * Copyright (c) 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * + * 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 + * + * This driver manages two devices: + * + * Device 0: The game port. + * Device 1: The SoundBlaster look-alike part. + * + * Sources: + * sound/oss/sb_common.c + * sys/dev/isa/sbdsp.c (OpenBSD) + */ + +#include <linux/types.h> +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16 Plug & Play support"); +MODULE_LICENSE("GPL"); + +int jazz16pnp_disable = false; /* Disable Jazz16 PnP. */ +module_param_named(disable, jazz16pnp_disable, bool, false); +MODULE_PARM_DESC(disable, "Jazz16 Plug & Play disable switch"); + +int jazz16pnp_verbose = false; /* Verbosity level. */ +module_param_named(verbose, jazz16pnp_verbose, bool, false); +MODULE_PARM_DESC(verbose, "Jazz16 Plug & Play verbosity level"); + +#define PFX "jazz16pnp: " +#define PFX_DEBUG KERN_DEBUG PFX +#define PFX_INFO KERN_INFO PFX +#define PFX_WARN KERN_WARNING PFX + +/* Jazz16 configuration. */ +#define SB_JAZZ16_CONFIG_PORT 0x201 +#define SB_JAZZ16_WAKEUP 0xaf +#define SB_JAZZ16_SET_PORTS 0x50 +#define SB_JAZZ16_SET_DMAINTR 0xFB + +extern void *pnp_alloc(long size); +extern void pnp_free_option(struct pnp_option *option); + +static struct pnp_dev *jazz16_gameport_dev = NULL; +static struct pnp_dev *jazz16_sb_dev = NULL; +static struct resource *jazz16pnp_fm_res = NULL; + +static uint jazz16_sb_port = 0, jazz16_sb_irq = 0; +static uint jazz16_sb_dma1 = 0, jazz16_sb_dma2 = 0; +static uint jazz16_mpu401_port = 0, jazz16_mpu401_irq = 0; + +static uint jazz16pnp_count = 0; + +static struct pnp_protocol jazz16pnp_protocol; + +static bool __init jazz16pnp_easy_option(struct pnp_dev *dev, uint pri, + struct pnp_port *port, uint nports, + struct pnp_irq *irq, uint nirqs, + struct pnp_dma *dma, uint ndmas) +{ + struct pnp_option *opt; + struct pnp_port *new_port; + struct pnp_irq *new_irq; + struct pnp_dma *new_dma; + uint i; + + switch (pri) { + case PNP_RES_PRIORITY_PREFERRED: + case PNP_RES_PRIORITY_ACCEPTABLE: + case PNP_RES_PRIORITY_FUNCTIONAL: + opt = pnp_register_dependent_option(dev, pri); + break; + + default: + opt = pnp_register_independent_option(dev); + break; + } + if (!opt) + return false; + + for (i = 0; i < nports; i ++) { + new_port = pnp_alloc(sizeof(*new_port)); + if (!new_port) + goto out_no_mem; + memcpy(new_port, &port[i], sizeof(*new_port)); + pnp_register_port_resource(opt, new_port); + } + for (i = 0; i < nirqs; i ++) { + new_irq = pnp_alloc(sizeof(*new_irq)); + if (!new_irq) + goto out_no_mem; + memcpy(new_irq, &irq[i], sizeof(*new_irq)); + pnp_register_irq_resource(opt, new_irq); + } + for (i = 0; i < ndmas; i ++) { + new_dma = pnp_alloc(sizeof(*new_dma)); + if (!new_dma) + goto out_no_mem; + memcpy(new_dma, &dma[i], sizeof(*new_dma)); + pnp_register_dma_resource(opt, new_dma); + } + return true; + +out_no_mem: + return false; +} + +static struct pnp_port jazz16pnp_game_ports[] __initdata = { + { .min = SB_JAZZ16_CONFIG_PORT, + .max = SB_JAZZ16_CONFIG_PORT, + .size = 1, .align = 0, + .flags = PNP_PORT_FLAG_FIXED, }, +}; + +static int __init jazz16pnp_add_gameport(void) +{ + struct pnp_dev *dev; + struct pnp_id *dev_id; + + if (!(dev = pnp_alloc(sizeof(*dev)))) + goto out_no_dev; + if (!(dev_id = pnp_alloc(sizeof(*dev_id)))) + goto out_no_id; + if (!jazz16pnp_easy_option(dev, PNP_RES_PRIORITY_INVALID, + jazz16pnp_game_ports, 1, + NULL, 0, NULL, 0)) + goto out_no_option; + + dev->protocol = &jazz16pnp_protocol; + dev->capabilities = PNP_READ; + dev->active = 1; + dev->number = 0; + + pnp_init_resource_table(&dev->res); + dev->res.port_resource[0].start = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + dev->res.port_resource[0].flags = IORESOURCE_IO; + + memcpy(dev_id->id, "PNPb02f", PNP_ID_LEN); + pnp_add_id(dev_id, dev); + + jazz16_gameport_dev = dev; + pnp_add_device(dev); + return 0; + +out_no_option: + if (dev->independent) + pnp_free_option(dev->independent); + kfree(dev_id); +out_no_id: + kfree(dev); +out_no_dev: + return -ENOMEM; +}; + +static struct pnp_port jazz16pnp_pref_ports[] __initdata = { + { .min = 0x220, .max = 0x220, .align = 0x00, .size = 0x10 }, + { .min = 0x330, .max = 0x330, .align = 0x00, .size = 0x02 }, + { .min = 0x388, .max = 0x388, .align = 0x00, .size = 0x04 }, +}; + +static struct pnp_port jazz16pnp_ports[] __initdata = { + { .min = 0x210, .max = 0x260, .align = 0x10, .size = 0x10 }, + { .min = 0x300, .max = 0x330, .align = 0x10, .size = 0x02 }, + { .min = 0x388, .max = 0x388, .align = 0x00, .size = 0x04 }, +}; + +static struct pnp_irq jazz16pnp_pref_irqs[] __initdata = { + { .map[0] = (1 << 5), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 9), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_irq jazz16pnp_irqs[] __initdata = { + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, + { .map[0] = (1 << 3) | (1 << 5) | (1 << 7) | + (1 << 9) | (1 << 10) | (1 << 15), + .flags = IORESOURCE_IRQ_HIGHEDGE }, +}; + +static struct pnp_dma jazz16pnp_pref_dmas[] __initdata = { + { .map = (1 << 1), .flags = IORESOURCE_DMA_8BIT, }, +}; + +static struct pnp_dma jazz16pnp_dmas[] __initdata = { + { .map = (1 << 1) | (1 << 3) | (1 << 5) | (1 << 7), + .flags = IORESOURCE_DMA_8AND16BIT, }, +}; + +static int __init jazz16pnp_add_sb(void) +{ + struct pnp_dev *sbdev; + struct pnp_id *sbid; + + if (!(sbdev = pnp_alloc(sizeof(*sbdev)))) + goto out_no_sbdev; + if (!(sbid = pnp_alloc(sizeof(*sbid)))) + goto out_no_sbid; + + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_PREFERRED, + jazz16pnp_pref_ports, 3, + jazz16pnp_pref_irqs, 2, + jazz16pnp_pref_dmas, 1)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 3, + jazz16pnp_irqs, 2, + jazz16pnp_dmas, 1)) + goto out_no_sboption; + if (!jazz16pnp_easy_option(sbdev, PNP_RES_PRIORITY_ACCEPTABLE, + jazz16pnp_ports, 3, + jazz16pnp_irqs, 1, + jazz16pnp_dmas, 1)) + goto out_no_sboption; + memcpy(sbid->id, "PNPb00f", PNP_ID_LEN); + pnp_add_id(sbid, sbdev); + + pnp_init_resource_table(&sbdev->res); + sbdev->capabilities = PNP_READ | PNP_WRITE | PNP_CONFIGURABLE | PNP_DISABLE; + sbdev->protocol = &jazz16pnp_protocol; + sbdev->number = 1; + + jazz16_sb_dev = sbdev; + pnp_add_device(sbdev); + return 0; + +out_no_sboption: + if (sbdev->dependent) + pnp_free_option(sbdev->dependent); + kfree(sbid); +out_no_sbid: + kfree(sbdev); +out_no_sbdev: + return -ENOMEM; +} + +/* Configure the SB and MPU ports. Passing 0 for sbport disables the Jazz16. */ +static void jazz16_port_setup(const uint sbport, const uint mpuport) +{ + u8 config; + + switch (sbport) { + case 0x210: + case 0x220: + case 0x230: + case 0x240: + case 0x250: + case 0x260: + config = sbport & 0x70; + jazz16_sb_port = sbport; + break; + + default: + config = 0; + jazz16_sb_port = 0; + break; + } + + switch (mpuport) { + case 0x300: + case 0x310: + case 0x320: + case 0x330: + config |= (mpuport >> 4) & 0x07; + jazz16_mpu401_port = mpuport; + break; + + default: + if (jazz16_sb_port) + BUG(); + } + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB port = %#x, MPU port = %#x.\n", + jazz16_sb_port, jazz16_mpu401_port); + outb(SB_JAZZ16_WAKEUP, SB_JAZZ16_CONFIG_PORT); + outb(SB_JAZZ16_SET_PORTS, SB_JAZZ16_CONFIG_PORT); + outb(config, SB_JAZZ16_CONFIG_PORT); +} + +static u8 jazz16_irq_config(uint irq) +{ + switch (irq) { + case 3: return 3; + case 5: return 1; + case 7: return 4; + case 9: return 2; + case 10: return 5; + case 15: return 6; + default: return 0; + } +} + +static u8 jazz16_dma_config(uint dma) +{ + switch (dma) { + case 1: return 1; + case 3: return 2; + case 5: return 3; + case 7: return 4; + default: return 0; + } +} + +/* The my_sbdsp_xxx functions were ripped from sound/isa/sb/sb_common.c. */ +#define BUSY_LOOPS 100000 + +static int my_sbdsp_command(unsigned int base, unsigned char val) +{ + int i; + for (i = BUSY_LOOPS; i; i--) + if ((inb(SBP1(base, STATUS)) & 0x80) == 0) { + outb(val, SBP1(base, COMMAND)); + return 1; + } + return 0; +} + +static int my_sbdsp_get_byte(unsigned int base) +{ + int val; + int i; + for (i = BUSY_LOOPS; i; i--) { + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + val = inb(SBP1(base, READ)); + return val; + } + } + return -ENODEV; +} + +static int my_sbdsp_reset(unsigned int base) +{ + int i; + + outb(1, SBP1(base, RESET)); + udelay(10); + outb(0, SBP1(base, RESET)); + udelay(30); + for (i = BUSY_LOOPS; i; i--) + if (inb(SBP1(base, DATA_AVAIL)) & 0x80) { + if (inb(SBP1(base, READ)) == 0xaa) + return 0; + else + break; + } + return -ENODEV; +} + +static int jazz16_irq_dma_setup(unsigned int base, uint sbirq, uint mpuirq, uint dma8, uint dma16) +{ + u8 dma_config, irq_config; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Setting SB irq %u, dma %u&%u, MPU irq %u.\n", + sbirq, dma8, dma16, mpuirq); + irq_config = jazz16_irq_config(sbirq) | jazz16_irq_config(mpuirq) << 4; + dma_config = jazz16_dma_config(dma8) | jazz16_dma_config(dma16) << 4; + if (!my_sbdsp_command(base, SB_JAZZ16_SET_DMAINTR)) + return 0; + + if (!my_sbdsp_command(base, dma_config)) + return 0; + jazz16_sb_dma1 = dma8; + jazz16_sb_dma2 = dma16; + + if (!my_sbdsp_command(base, irq_config)) + return 0; + jazz16_sb_irq = sbirq; + jazz16_mpu401_irq = mpuirq; + return 1; +} + +static int jazz16pnp_get_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + pnp_init_resource_table(res); + if (dev == jazz16_gameport_dev) { + res->port_resource[0].start = res->port_resource[0].end = SB_JAZZ16_CONFIG_PORT; + res->port_resource[0].flags = IORESOURCE_IO; + } else if (dev == jazz16_sb_dev) { + if (!jazz16_sb_port) + return 0; + res->port_resource[0].start = jazz16_sb_port; + res->port_resource[0].end = jazz16_sb_port + 0x10 - 1; + res->port_resource[0].flags = IORESOURCE_IO; + + res->irq_resource[0].start = jazz16_sb_irq; + res->irq_resource[0].end = jazz16_sb_irq; + res->irq_resource[0].flags = IORESOURCE_IRQ; + if (jazz16_mpu401_irq) { + res->irq_resource[1].start = jazz16_mpu401_irq; + res->irq_resource[1].end = jazz16_mpu401_irq; + res->irq_resource[1].flags = IORESOURCE_IRQ; + } + + res->dma_resource[0].start = jazz16_sb_dma1; + res->dma_resource[0].end = jazz16_sb_dma1; + res->dma_resource[0].flags = IORESOURCE_DMA; + if (jazz16_sb_dma2 != jazz16_sb_dma1) { + res->dma_resource[1].start = jazz16_sb_dma2; + res->dma_resource[1].end = jazz16_sb_dma2; + res->dma_resource[1].flags = IORESOURCE_DMA; + } + res->port_resource[1].start = jazz16_mpu401_port; + res->port_resource[1].end = jazz16_mpu401_port + 2 - 1; + res->port_resource[1].flags = IORESOURCE_IO; + } else + return -EINVAL; + return 0; +} + +/* Disables the device if supported. */ +static int jazz16pnp_disable_resources(struct pnp_dev *dev) +{ + if (dev == jazz16_gameport_dev) { + return -ENOSYS; + } else if (dev == jazz16_sb_dev) { + if (jazz16_sb_port) + jazz16_irq_dma_setup(jazz16_sb_port, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + if (jazz16pnp_fm_res) { + release_resource(jazz16pnp_fm_res); + jazz16pnp_fm_res = NULL; + } + dev->active = false; + return 0; + } else { + return -EINVAL; + } +} + +#define PNP_RES_IS_SET(res,type,TYPE,i) \ + (((res)->type##_resource[i].flags & (IORESOURCE_##TYPE | IORESOURCE_UNSET)) \ + == IORESOURCE_##TYPE) + +static int jazz16pnp_set_resources(struct pnp_dev *dev, struct pnp_resource_table *res) +{ + unsigned int sbirq, mpuirq, dma1, dma2; + + if (dev == jazz16_gameport_dev) + return -ENOSYS; + + else if (dev == jazz16_sb_dev) { + if (PNP_RES_IS_SET(res, dma, DMA, 0)) + dma1 = res->dma_resource[0].start; + else + return -EINVAL; + if (PNP_RES_IS_SET(res, dma, DMA, 1)) + dma2 = res->dma_resource[1].start; + else + dma2 = dma1; /* "half duplex" operation. */ + if (PNP_RES_IS_SET(res, irq, IRQ, 0)) + sbirq = res->irq_resource[0].start; + else + return -EINVAL; + if (PNP_RES_IS_SET(res, irq, IRQ, 1)) + mpuirq = res->irq_resource[1].start; + else + mpuirq = 0; /* MPU-401 driver supports this. */ + if (! PNP_RES_IS_SET(res, port, IO, 0)) + return -EINVAL; + if (! PNP_RES_IS_SET(res, port, IO, 1)) + return -EINVAL; + + /* Block the OPL3 alias to avoid conflicts. */ + jazz16pnp_fm_res = request_region(0x388, 4, "SoundBlaster FM"); + /* Avoid setting the mpu port after setting the sb port + because doing so will glitch the mpu port from the + previous one to the new one. */ + jazz16_port_setup(res->port_resource[0].start, + res->port_resource[1].start); + jazz16_irq_dma_setup(jazz16_sb_port, sbirq, mpuirq, dma1, dma2); + dev->active = true; + return 0; + } else + return -EINVAL; +} +#undef PNP_RES_IS_SET + +static struct pnp_protocol jazz16pnp_protocol = { + .name = "Jazz16 PnP", + .get = jazz16pnp_get_resources, + .set = jazz16pnp_set_resources, + .disable = jazz16pnp_disable_resources, +}; + +static uint __init jazz16_check(unsigned int base) +{ + if (my_sbdsp_reset(base)) + return 0; + if (!(my_sbdsp_command(base, SB_DSP_GET_JAZZ_VERSION))) + return 0; + return SB_VERSION_IS_JAZZ16(my_sbdsp_get_byte(base)); +} + +static int __init jazz16pnp_init(void) +{ + int err = 0; + unsigned int base, mpubase; + + if (jazz16pnp_disable == 1) { + printk(PFX_INFO "Jazz16 Plug & Play support disabled.\n"); + return 0; + } + if (!request_region(SB_JAZZ16_CONFIG_PORT, 1, "Jazz16 PnP probe")) + return -EBUSY; + + for (mpubase = 0x300; mpubase <= 0x330; mpubase += 0x10) + if (request_region(mpubase, 2, "Jazz16 PnP probe")) + break; + if (mpubase > 0x330) + goto out_no_mpubase; + + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Scanning for Jazz16 cards...\n"); + + for (base = 0x260; base >= 0x210; base -= 0x10) { + if (!request_region(base, 16, "Jazz16 PnP probe")) + continue; + if (jazz16pnp_verbose) + printk(PFX_DEBUG "Probing at %#x (with MPU at %#x).\n", + base, mpubase); + jazz16_port_setup(base, mpubase); + if (jazz16_check(base)) + jazz16pnp_count++; + jazz16_irq_dma_setup(base, 0, 0, 0, 0); + jazz16_port_setup(0, 0); + release_region(base, 16); + if (jazz16pnp_count) + break; + } + if (base < 0x210) + goto out_no_sbbase; + if (!jazz16pnp_count) { + err = -ENODEV; + goto out_no_devices; + } + if (pnp_register_protocol(&jazz16pnp_protocol) < 0) + err = -ENOMEM; + else { + if (jazz16pnp_add_gameport()) + printk(PFX_WARN "Failed to add game port.\n"); + if (jazz16pnp_add_sb()) + printk(PFX_WARN "Failed to add SB port.\n"); + err = 0; + } +out_no_devices: + printk(PFX_INFO "%u Jazz16 card%s detected.\n", jazz16pnp_count, + jazz16pnp_count == 1 ? "" : "s"); +out_no_sbbase: + release_region(mpubase, 2); +out_no_mpubase: + release_region(SB_JAZZ16_CONFIG_PORT, 1); + return err; +} +module_init(jazz16pnp_init); + +static void __exit jazz16pnp_exit(void) +{ + pnp_unregister_protocol(&jazz16pnp_protocol); +} +module_exit(jazz16pnp_exit); -- Rask Ingemann Lambertsen

3 8

[alsa-devel] 2.6.21-rc7: known regressions with patches
by Adrian Bunk 26 Apr '07

26 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20 with patches available. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available Subject : qla2xxx driver no longer works References : http://lkml.org/lkml/2007/4/16/49 http://lkml.org/lkml/2007/4/20/224 Submitter : David Miller <davem(a)davemloft.net> Robert Peterson <rpeterso(a)redhat.com> Handled-By : David Miller <davem(a)davemloft.net> Patch : http://lkml.org/lkml/2007/4/16/234 Status : patch was suggested

1 0

[alsa-devel] playback open error:
by Ashlesha Shintre 25 Apr '07

25 Apr '07

Hi, I m trying to run the simple playback program that I have attached with this email -- and I get the following error: ALSA lib pcm_dmix.c:862:(snd_pcm_dmix_open) unable to open slave > Playback open error: Device or resource busy I am running this program as a function called wavplay (attached) -- however, if i run the same program, as part of the main function, then there is no problem -- What i mean is that if i basically just paste the code for the wavplay function in main and provide the input values (the pcm data to be played) -- then it can very well open the default device and play the pcm data that is being passed to it! Any pointers? Thanks, Ashlesha.

2 2

[alsa-devel] daily snapshot & version ID.
by Tobin Davis 25 Apr '07

25 Apr '07

Would it be possible to add a version ID update to the daily snapshots that you generate? It would make tracking a little easier, and we could get more testing on recent snapshots. That or spin another RC soon. -- Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] crash/reboot with rawmidi on ice1712 dual opteron
by Florian 25 Apr '07

25 Apr '07

Hi, on one particular machine, an IBM workstation (Intellistation A Pro with dual opteron 250, 4GB RAM), opening a rawmidi input port on an M-Audio 2496 card will cause an immediate reboot of the machine. I've tried with different SMP kernels and different ALSA versions (including 1.0.14rc3), but always the same. Syslog does not show anything. Using a USB MIDI interface works fine. Also, MIDI input on a similar workstation (Intellistation Z Pro, dual xeon, 1GB RAM) with M-Audio 2496 works without problems. Any ideas? Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

3 19

[alsa-devel] Digigram ISA cards
by Ash Willis 25 Apr '07

25 Apr '07

Hi, I'm thinking about trying to write a driver for some old Digigram cards soon, namely the PCX11 and PCX9. I might be wrong but it seems these cards natively support MPEG layers I and II and only actually support PCM by re-encoding in-driver. Does alsa-lib support MPEG conversions? Are there examples of drivers that already do this type of thing? Please forgive my ignorance. I grepped the sources and didn't find too much info. Ash =

2 1

[alsa-devel] [1/3] 2.6.21-rc7: known regressions (v2)
by Adrian Bunk 25 Apr '07

25 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : gammu no longer works References : http://lkml.org/lkml/2007/4/20/84 Submitter : Wolfgang Erig <Wolfgang.Erig(a)gmx.de> Status : unknown Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Handled-By : Takashi Iwai <tiwai(a)suse.de> Patch : http://lkml.org/lkml/2007/4/20/117 Status : patch available

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. Signed off by Tobin Davis <tdavis(a)dsl-only.net> Wrong "From" email address selected earlier. Oops.

2 2

[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

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[alsa-devel] HDA-Intel: Fix headphone squeal on Conexant audio
by Tobin Davis 24 Apr '07

24 Apr '07

This patch fixes the headphone squeal and noise on Conexant CX20551 (Waikiki) audio. -- Tobin Davis <gruemaster(a)email.uophx.edu>

2 1

[alsa-devel] ICE 1724 MIDI support still broken?
by Daniel James 24 Apr '07

24 Apr '07

Hello ALSA team, We have a user with an ESI Juli@ whose Linux box hangs when any MIDI application starts up. I read that MIDI support is broken generally on ICE 1724 cards; is this still the case in recent driver releases? If so, is there any interest in fixing this long-standing issue? The user has indicated that his box would be available by ssh for any tests required. Cheers! Daniel

2 3

[alsa-devel] Standard config and plugin paths
by Takashi Iwai 24 Apr '07

24 Apr '07

Hi, while working on openSUSE 10.3 ALSA packaging, I had a discussion regarding the config and plugin paths. Basically, the current configs (typically in /usr/share/alsa or /etc/alsa) and binary plugins (/usr/lib/alsa-lib) are fairly strictly bound to the specific library, libasound.so.2. That is, if we publish a newer, incompatible version, libasound.so.3, these files will be very unlikely unusable, too. The suggestion was that we should keep these config and plugin files in the version-dependent paths so that no conflictions would occur at update. For example, instead of /usr/share/alsa, /usr/share/alsa2 or whatever. The fix would be easy -- just add some options to configure script of alsa-lib and alsa-plugins. So, the question is what path names are most suitable as the standard. For example, I find the following OK: - $DATADIR/alsa-1.0/ - $LIBDIR/alsa-lib-1.0/ Or, the deeper paths: - $DATADIR/alsa/1.0/ - $LIBDIR/alsa-lib/1.0/ For keeping the backward compatibility, we may implement fallback to read the old default path in addition, if the new path doesn't exist. In that case, the former suggestion would be cleaner. Any comments or suggestions? Takashi

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[alsa-devel] Questions about writing a new ALSA driver for a very limitted device
by Tim Harvey 24 Apr '07

24 Apr '07

Greetings, I'm setting out to write an ALSA driver for a very limited device thats behind a PLD. The actual codec is a uda1380 and does 16bit MSB stereo encode/decode. The way the device is driven the sampling rate is limited to 8KHz and data transfer to/from the device I have to handle manually. I've read over the 'writing an ALSA driver' tutorial and have started the driver but have run into a few questions: - for the specs I've given above: 8000Hz sampling rate, 16bit stereo MSB samples does the following snd_pcm_hardware_t look right?: static snd_pcm_hardware_t my_playback_hw = { .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID), .formats = SNDRV_PCM_FMTBIT_U16_BE, // is this 16bit MSB? .rates = SNDRV_PCM_RATE_8000, // seems redundant? .rate_min = 8000, .rate_max = 8000, .channels_min = 2, .channels_max = 2, .buffer_bytes_max = 32768, .period_bytes_min = 4096, .period_bytes_max = 32768, .periods_min = 1, .periods_max = 1024, }; - I'm a little confused at how to deal with copying data from/to the device. It looks like I would implement the copy callback and copy the passed buffer to the device in the device specific manner I need to (playback), but I'm not clear where to put data that I receive from the device (capture). I notice that when the hw_params callback is called a dma buffer has been created by ALSA in the substream->runtime struct. I assume this is created based on hte buffer_bytes_max field above and I'm to copy data to this buffer? How do I know where to copy it within the buffer and what data ALSA has already consumed from the buffer? - I might need to create an intermediate buffer between ALSA's rather large buffers and my hardware buffer. Is there any driver that I could look at that does this? The tutorial talks about the vxpocket driver but that code looks very foriegn from the rest of the ALSA drivers. - I'm not sure I understand what the 'period' is about. Its clear to me that I need to call snd_pcm_period_elapsed() periodically from either an IRQ or other timed callback, but I'm not clear what the hardware pointer is or how that converts to 'frames' and 'periods'. - in general, if my hardware device only allows 16bit MSB 8KHz, does ALSA provides a mechanism for converting data from other formats? Thanks for any pointers, Tim

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[alsa-devel] What does start_threshold mean, and what value should it contain?
by j t 24 Apr '07

24 Apr '07

Hi. I'm trying to debug a problem with xruns which happen when playing a wav file with vlc, but I can't reproduce the problems when using "aplay -M" to play the same wav file. (logged as https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3031). I've checked the contents of /proc/asound/card0/pcm0p/sub0/ in both cases - they appear to be very similar except for the contents of sw_params: when I use vlc, start_threshold says 1, but when I use "aplay -M", start_threshold says 16384. So, what does start_threshold mean, is there a particular value that start_threshold "should" contain, and could this difference be the cause of the xruns? (The aplay man page doesn't seem to have any flags for controlling the value of this parameter, so it doesn't appear to be "user-configurable"). (Apologies if this is the wrong mailing list - I though that -devel would be better than -user, but I could be wrong). Thanks, Jaime

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Re: [alsa-devel] Fwd: No sound with ATI SB450
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 22:45:00 +1000, Ben DiDonc wrote: > > Tobin, Takashi, > Thank you very much for your help. > Takashi's last suggestion did it and I finally got a working ATI SB450 > sound card with Ubtuntu Feisty (kernel 2.6.20 and > alsa-driver-hg20070421). I did a (last) alsa-info.sh in case it helps > others, see http://pastebin.ca/454115 > Here's what it took: > modprobe snd_hda_intel probe_mask=8 model=auto > Without probe_mask, the ALC861 is not properly detected. Without > model=auto, playback doesnt work. > Let me know If I can be of any help or if you want me to try something else. As I requested, get /proc/asound/card0/codec#* files for both working and non-working states. Then we can compare what is not set up properly. probe_mask is a bit hard to fix right now. A least, we need to know which modem codec is harmful. Please check the codec#* file again with probe_mask=1. Takashi > Thanks again. > Ben > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 23 Apr 2007 21:58:17 +1000, > > Ben DiDonc wrote: > > > > > > Tobin, Takashi, > > > upon Takashi's advice, I tried loading hda-intel > > > (alsa-driver-hg20070421) using the probe_mask argument under 2.6.20. > > > With probe_mask=1, the codec isnt detected and only the modem shows up > > > (just like before). However, with probe_mask=8, the codec IS correctly > > > detected (alsa-info.sh at http://pastebin.ca/454059 ). syslog shows: > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 19 > > > ALSA .../alsa-driver-hg20070421/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:696: > > > codec_mask = 0x9 > > > ALSA .../alsa/alsa-driver-hg20070421/pci/hda/hda_codec.c:1819: > > > hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) > > > > > > However, there is still no playback (no muted control, all volumes at > > > 100% as shown by amixer in alsa-info.sh). Here again, capture works, I > > > successfully captured from the mic with arecord. > > > > Now you can try what I suggested, model=auto. And compare with > > codec#* files. If you have /proc/asound/card0/codec#* files on the > > working system, it'd be better, in addition... > > > > > > Takashi > > > > > > > > Can I draw your attention on something ? I m not sure about what i m > > > going to say (specially the part on NIDs) but i ll try. I know Tobin > > > already explained that this was normal, but if you have a look at the > > > post where I loaded edgy (2 posts ago), where the playback > > > works(2.6.17 with 1.0.12), you ll see that when I aplay something, > > > syslog shows only a single call to hda_codec_setup_stream with > > > NID=0x3. In all cases where playback doesnt work (1.0.14rcX under > > > 2.6.20), running aplay makes 4 calls to hda_codec_setup_stream with > > > NID=3, 6 ,5 and 4 in turn. Why does it try 6 5 and 4 ? Is it because 3 > > > fails ? We know that 3 should be used as its the one that works (under > > > 2.6.17). Now arent NIDs node IDs of codecs connected to the HDA > > > controller ? Is it possible that 6, 5 and 4 are mistakenly detected > > > and cannot be used for playback ? > > > Let me know if I m way off track. > > > > > Ben > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > At Mon, 23 Apr 2007 11:05:19 +0000, > > > > Ben DiDonc wrote: > > > > > > > > > > Takashi, > > > > > I did try passing model=auto to all 14 release candidates, > > > > > > > > Did you already try the HG version (or daily snapshot), right? > > > > > > > > > but without > > > > > success. the codec is never detected (see my previous posts for > > > > > alsa-info.sh outputs with both 2.6.20 and 2.6.17). > > > > > My problem is I cant get my sound card to work with kernel 2.6.20, > > > > > when it used to work with 2.6.17. > > > > > > > > The codec mask probed is 0x09, so you can pass either probe_mask=8 or > > > > probe_mask=1 to skip the modem detection (if it matters). > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > Ben > > > > > > > > > > On 4/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > > > > At Mon, 23 Apr 2007 10:50:18 +0000, > > > > > > Ben DiDonc wrote: > > > > > > > > > > > > > > Tobin, > > > > > > > Unfortunately, ACPI was on at all times. My laptop hardly ever boots > > > > > > > without ACPI (randomly, the SATA driver fails before it can even mount > > > > > > > the root partition). > > > > > > > I ran some more tests today: > > > > > > > I loaded Ubuntu Edgy from the live CD where the sound chip used to > > > > > > > work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works > > > > > > > straight away. From there , I ran the same series of test: download > > > > > > > and build (--with-debug=detect) the next version of alsa-driver, run a > > > > > > > capture from mic and try playing it back, check syslog and run > > > > > > > alsa-info.sh. I did this for successive versions of alsa-driver > > > > > > > (starting with 1.0.12) until it fails. > > > > > > > Here are the results: > > > > > > > > > > > > > > Alsa driver 1.0.12rc1(shipped with edgy): > > > > > > > capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 > > > > > > > > > > > > > > Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) > > > > > > > syslog lines after modprobe: > > > > > > > ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: > > > > > > > codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: > > > > > > > line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 > > > > > > > (0x0/0x0/0x0/0x0/0x0) > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, > > > > > > > dig_out=0x0, din_in=0x0 > > > > > > > ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, > > > > > > > fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 > > > > > > > > > > > > > > capture: working > > > > > > > //capture starting > > > > > > > hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, > > > > > > > channel=0, format=0x11 > > > > > > > //capture stopping > > > > > > > hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > playback: working > > > > > > > //starting > > > > > > > ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, > > > > > > > fragsize=0x1000, format=0x11 > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, > > > > > > > channel=0, format=0x11 > > > > > > > //stopping > > > > > > > ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, > > > > > > > channel=0, format=0x0 > > > > > > > > > > > > > > alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) > > > > > > > modprobe snd-hda-intel: > > > > > > > hda_intel.c:680: codec_mask = 0x9 > > > > > > > hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... > > > > > > > hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) > > > > > > > hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, > > > > > > > cd=0x11, aux=0x0 > > > > > > > capture works fine, syslog output similar to 1.0.12 plus an extra > > > > > > > "hda-intel: Invalid position buffer, using LPIB read method instead" > > > > > > > in the middle of the capture. Playback works fine too, with the same > > > > > > > syslog entries as 1.0.12 > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) > > > > > > > modprobe: > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works (output similar to 1.0.12) > > > > > > > Playback does NOT work this time: > > > > > > > syslog shows a call to azx_pcm_prepare: and 4 calls to > > > > > > > hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out > > > > > > > my previous post) > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) > > > > > > > modprobe: the codec is correctly detected > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for > > > > > > > config 1179:ff10 (Toshiba) > > > > > > > Capture works. Playback does NOT work, same syslog entries as > > > > > > > 1.0.14rc1 with multiple calls to hda_codec_setup_stream > > > > > > > > > > > > > > > > > > > > > alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) > > > > > > > modprobe: the codec isnt detected. > > > > > > > ALSA hda_intel.c:696: codec_mask = 0x9 > > > > > > > capture and playback dont work. > > > > > > > > > > > > > > Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy > > > > > > > (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but > > > > > > > not playback, 14rc3 doesnt detect the codec. > > > > > > > What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel > > > > > > > between releases 13 and 14 ... > > > > > > > > > > > > Try to pass model=auto to snd-hda-intel module, and check whether it > > > > > > works (make sure that you have unmuted/adjusted mixer elements before > > > > > > that). Then get /proc/asound/card0/codec#* files, and compare with > > > > > > the files without model option. > > > > > > > > > > > > > > > > > > Takashi > > > > > > > > > > > > > > > > > > > > >

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[alsa-devel] Compilation of latest alsa-hg
by dustin＠seznam.cz 24 Apr '07

24 Apr '07

Hello, please could you point me at a functioning description of latest complete hg tree compilation? The guides e.g. http://www.suseforums.net/index.php?showtopic=30156&pid=170707&st=0&#entry1… end up with /home/pavel/tmp/alsa-hg/alsa/alsa-driver/soc/codecs/../../alsa-kernel/soc/codecs/wm8753.c:1707: error: ‘I2C_DRIVERID_WM8753’ undeclared here (not in a function) Thanks a lot. Pavel.

2 1

Re: [alsa-devel] Functioning support for Prodigy 192
by Takashi Iwai 24 Apr '07

24 Apr '07

At Mon, 23 Apr 2007 14:00:22 +0200 (CEST), dustin(a)seznam.cz wrote: > > Hello, > > Back here again :) > > While comparing media reading precision of various CD players I hit the following issues: > > 1. De-emphasis filter in AK4114 is turned on, distorting incoming data. A patch to disable: > > diff -r aed6a15c9398 pci/ice1712/prodigy192.c > --- a/pci/ice1712/prodigy192.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/prodigy192.c Mon Apr 23 12:47:01 2007 +0200 > @@ -714,7 +714,7 @@ static int prodigy192_ak4114_init(struct > { > static const unsigned char ak4114_init_vals[] = { > AK4114_RST | AK4114_PWN | AK4114_OCKS0 | AK4114_OCKS1, > - AK4114_DIF_I24I2S, /* ice1724 expects I2S and provides clock */ > + AK4114_DIF_I24I2S | AK4114_DEM0 , /* ice1724 expects I2S and provides clock, DEM0 disables deemphasis filter */ > AK4114_TX1E, > AK4114_EFH_1024 | AK4114_DIT, /* default input RX0 */ > 0, Looks OK. > 2. All data recorded when master clock was provided by AK4114 (Multi Track Internal Clock= IEC958 Input) had double the sampling rate. For each input sample there were two identical output samples in the file. > > Here is where the FPGA on MIODIO gets introduced. Both MCKO1 and MCKO2 of ak4114 are fed to FPGA, which outputs master clock to SPMCLKIN of ice1724. Unfortunately my oscilloscope has just died, so experimentally I found out that only a combination of OCKS0=1, OCKS1 = 1 (128fs, 64fs output) and ice1724- VT1724_MT_I2S_MCLK_128X=0 (256fs input) yields correct sampling rate. In order for this to work the FPGA must double the MCK01 rate (I cannot check the frequency now). > > It may actually make sense as it is probably the only way to support input sampling frequencies all the way up to 192kHz (128fs is max ak4114 can output at 192kHz) without having to switch the VT1724_MT_I2S_MCLK_128X bit in ice1724. In consumer mode the chips have no information about the sampling frequency (ak4114 does no detection) and the driver would not know when to set the bit. > > The problem is that due to switching sequence of Multi Track Internal Clock enum the VT1724_MT_I2S_MCLK_128X stays set (from the previous 192kHz option). I think it should not be set for ice1724 slave mode at all, here is the patch which fixes the sampling rate. Looks fine, too. Could you post a patch with a short patch summary and sign-off to apply to the upstream tree? thanks, Takashi > > Thanks, > > Pavel. > > diff -r aed6a15c9398 pci/ice1712/ice1724.c > --- a/pci/ice1712/ice1724.c Mon Apr 23 11:54:41 2007 +0200 > +++ b/pci/ice1712/ice1724.c Mon Apr 23 13:57:19 2007 +0200 > @@ -1666,7 +1666,11 @@ static int snd_vt1724_pro_internal_clock > spin_lock_irq(&ice->reg_lock); > oval = inb(ICEMT1724(ice, RATE)); > if (ucontrol->value.enumerated.item[0] == spdif) { > + unsigned char i2s_oval; > outb(oval | VT1724_SPDIF_MASTER, ICEMT1724(ice, RATE)); > + /* setting 256fs */ > + i2s_oval = inb(ICEMT1724(ice, I2S_FORMAT)); > + outb( i2s_oval & ~VT1724_MT_I2S_MCLK_128X, ICEMT1724(ice, I2S_FORMAT)); > } else { > rate = rates[ucontrol->value.integer.value[0] % 15]; > if (rate <= get_max_rate(ice)) { > >

2 2

[alsa-devel] Buffer Overrun Messages
by John Dey 24 Apr '07

24 Apr '07

Hi, I have been driving fluidsynth with with aplaymidi and get the following: fluidsynth: ALSAsequencer buffer overrun, lost event. I have played around with buffer size and buffer number and also run with and without jackd. The sound out doesn't seem to be affect. Can anyone give me some guidance on how to eliminate the error. Could the error be caused by fluidsynth? I have also run the program on PS2/Yellowdog linux and CCRMA fc5 (x86) both with similar results. Thanks. John

2 1

[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Add support for Asus A8JN Laptop
by Tobin Davis 23 Apr '07

23 Apr '07

This patch adds support for the Asus A8JN Laptop. Other modes were tested, this one worked best. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

Re: [alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, Unfortunately, ACPI was on at all times. My laptop hardly ever boots without ACPI (randomly, the SATA driver fails before it can even mount the root partition). I ran some more tests today: I loaded Ubuntu Edgy from the live CD where the sound chip used to work (kernel 2.6.17, alsa driver 1.0.12rc1). Obviously the sound works straight away. From there , I ran the same series of test: download and build (--with-debug=detect) the next version of alsa-driver, run a capture from mic and try playing it back, check syslog and run alsa-info.sh. I did this for successive versions of alsa-driver (starting with 1.0.12) until it fails. Here are the results: Alsa driver 1.0.12rc1(shipped with edgy): capture: working, playback working, alsa-info.sh at http://pastebin.ca/453947 Alsa driver 1.0.12: (alsa-info.sh at http://pastebin.ca/453971 ) syslog lines after modprobe: ACPI: PCI Interrupt 0000:00:14.2[A] -> GSI 16 (level, low) -> IRQ 233 ALSA alsa-driver-1.0.12/pci/hda/../../alsa-kernel/pci/hda/hda_intel.c:663: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2150: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2154: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2156: hp=0x0, dig_out=0x0, din_in=0x0 ALSA alsa-driver-1.0.12/pci/hda/hda_codec.c:2164: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture: working //capture starting hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x1, channel=0, format=0x11 //capture stopping hda_codec.c:628: hda_codec_setup_stream: NID=0x8, stream=0x0, channel=0, format=0x0 playback: working //starting ALSA hda_intel.c:1124: azx_pcm_prepare: bufsize=0x10000, fragsize=0x1000, format=0x11 ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x5, channel=0, format=0x11 //stopping ALSA hda_codec.c:628: hda_codec_setup_stream: NID=0x3, stream=0x0, channel=0, format=0x0 alsa-driver-1.0.13 (alsa-info.sh at http://pastebin.ca/453975 ) modprobe snd-hda-intel: hda_intel.c:680: codec_mask = 0x9 hda_codec: Unknown model for ALC861, trying auto-probe from BIOS... hda_codec.c:2164: autoconfig: line_outs=1 (0xb/0x0/0x0/0x0/0x0) hda_codec.c:2168: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2172: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) hda_codec.c:2180: inputs: mic=0xd, fmic=0x0, line=0x0, fline=0x0, cd=0x11, aux=0x0 capture works fine, syslog output similar to 1.0.12 plus an extra "hda-intel: Invalid position buffer, using LPIB read method instead" in the middle of the capture. Playback works fine too, with the same syslog entries as 1.0.12 alsa-driver-1.0.14rc1 (alsa-info.sh at http://pastebin.ca/453991 ) modprobe: ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1768: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works (output similar to 1.0.12) Playback does NOT work this time: syslog shows a call to azx_pcm_prepare: and 4 calls to hda_codec_setup_stream with different nids (3, 6 ,5, 4 as pointed out my previous post) alsa-driver-1.0.14rc2 (alsa-info.sh at http://pastebin.ca/453799 ) modprobe: the codec is correctly detected ALSA hda_intel.c:696: codec_mask = 0x9 ALSA hda_codec.c:1764: hda_codec: model 'toshiba' is selected for config 1179:ff10 (Toshiba) Capture works. Playback does NOT work, same syslog entries as 1.0.14rc1 with multiple calls to hda_codec_setup_stream alsa-driver-1.0.14rc3 (alsa-info.sh at http://pastebin.ca/453785 ) modprobe: the codec isnt detected. ALSA hda_intel.c:696: codec_mask = 0x9 capture and playback dont work. Summary: I tried alsa-driver 1.012 through to 1.0.14rc3 on ubuntu edgy (kernel 2.6.17). 12 and 13 work fine, 14rc1 and 2 capture works but not playback, 14rc3 doesnt detect the codec. What next ? not sure, maybe make a diff of alsa-kernel/pci/hda-intel between releases 13 and 14 ...

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[alsa-devel] ad1986a snd_hda_intel patch
by Dmitry Ursegov 23 Apr '07

23 Apr '07

Hello! I have an asus a8jn laptop and a integrated audio with Analog Devices AD1986A chip. $ lspci -vnn 0:1b.0 Audio device [0403]: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller [8086:27d8] (rev 02) Subsystem: ASUSTeK Computer Inc. Unknown device [1043:1447] Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at febfc000 (64-bit, non-prefetchable) [size=16K] Capabilities: <access denied> The audio seems to work properly on the right channel, but not on the left. All I get on the left channel is a loud screeching noise followed by a pop when the audio turns off. The microphon jack don't work also. I make a patch from the solution described in this http://mailman.alsa-project.org/pipermail/alsa-devel/2007-March/000093.html message: diff -Nur alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c --- alsa-driver-1.0.14rc3/alsa-kernel/pci/hda/patch_analog.c 2007-03-06 15:26:32.000000000 +0300 +++ alsa-driver-1.0.14rc3_ad1986a/alsa-kernel/pci/hda/patch_analog.c 2007-04-20 23:32:19.000000000 +0400 @@ -873,7 +873,7 @@ board_config = snd_hda_check_board_config(codec, AD1986A_MODELS, ad1986a_models, ad1986a_cfg_tbl); - switch (board_config) { +/* switch (board_config) { case AD1986A_3STACK: spec->num_mixers = 2; spec->mixers[1] = ad1986a_3st_mixers; @@ -912,6 +912,18 @@ spec->multiout.dig_out_nid = 0; break; } +*/ + + spec->num_mixers = 2; + spec->mixers[1] = ad1986a_3st_mixers; + spec->num_init_verbs = 3; + spec->init_verbs[1] = ad1986a_3st_init_verbs; + spec->init_verbs[2] = ad1986a_ch2_init; + spec->channel_mode = ad1986a_modes; + spec->num_channel_mode = ARRAY_SIZE(ad1986a_modes); + spec->need_dac_fix = 1; + spec->multiout.max_channels = 2; + spec->multiout.num_dacs = 1; return 0; } I don't know, why snd_hda_check_board_config() return wrong value, but that workaround works. Thanks! Dmitry Ursegov

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[alsa-devel] Strange code in
by Pete Zaitcev 23 Apr '07

23 Apr '07

Dear Takashi: Hans de Goede (on cc:) hits a problem when snd-usb-audio breaks with: "USB device is in the shutdown state, cannot create a card instance". This is how the place looks: chip = NULL; for (i = 0; i < SNDRV_CARDS; i++) { if (usb_chip[i] && usb_chip[i]->dev == dev) { if (usb_chip[i]->shutdown) { snd_printk(KERN_ERR "USB device is in the shutdown state, cannot create a card instance\n"); goto __error; } chip = usb_chip[i]; break; } } The code looks broken to me, because dev at this time is a new device structure, freshly allocated. You cannot compare it with an old instance, and I don't know why you would want to. Unfortunately (for fortunately), I have zero clue about the way ALSA works, specifically rules for snd_device_ops.dev_free, so I am asking you to look at this problem. Yours, -- Pete

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[alsa-devel] [PATCH] sound: fix incorrect use of platform_device_register()
by Akinobu Mita 23 Apr '07

23 Apr '07

The platform_device allocated by platform_device_alloc() should be added to the device hierarchy by platform_device_add() instead of platform_device_register(). Otherwise it will hit WARN_ON() in platform_device_register(). by illegal refcount. This patch fixes such incorrect usages in portman2x4 and mts64 drivers. Also it removes unnecessary trailing whitespaces. Cc: Jaroslav Kysela <perex(a)suse.cz> Signed-off-by: Akinobu Mita <akinobu.mita(a)gmail.com> Index: 2.6-mm/sound/drivers/portman2x4.c =================================================================== --- 2.6-mm.orig/sound/drivers/portman2x4.c +++ 2.6-mm/sound/drivers/portman2x4.c @@ -676,13 +676,13 @@ static void __devinit snd_portman_attach struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; } Index: 2.6-mm/sound/drivers/mts64.c =================================================================== --- 2.6-mm.orig/sound/drivers/mts64.c +++ 2.6-mm/sound/drivers/mts64.c @@ -892,13 +892,13 @@ static void __devinit snd_mts64_attach(s struct platform_device *device; device = platform_device_alloc(PLATFORM_DRIVER, device_count); - if (!device) + if (!device) return; /* Temporary assignment to forward the parport */ platform_set_drvdata(device, p); - if (platform_device_register(device) < 0) { + if (platform_device_add(device) < 0) { platform_device_put(device); return; }

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Re: [alsa-devel] Problem with Conexant
by Tobin Davis 23 Apr '07

23 Apr '07

Did you try adjusting the PCM volume levels? They may be set too low. That particular sound chip required a lot of manipulating to get functional, and is still very problematic. Unfortunately, I have no way of testing it, so users (like yourself) have to be my ears. The last time I worked with that particular configuration, the user that was testing said that it took a bit of tweaking with the PCM and other volume levels to get good sound. Tobin On Mon, 2007-04-23 at 08:11 +0307, Black X wrote: > after I install the last alsa-driver from the link you give me it > ( alsa-driver-hg20070421.tar.bz2 ) I still have the same problem. now > what I most to do to inform you about this issue? > > > Thank you. > > On Sun, 2007-04-22 at 19:27 -0700, Tobin Davis wrote: > > Could you try the latest alsa-driver snapshot from > > ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver? > > > > I can't tell what revision of Conexant code you are using based on > > your email. Anything after 20070413 should work better, and it will > > also be the most current. I haven't made any changes to the code > > since ~April 6. If you are still having audio issues, let me know. > > > > Tobin > > > > > > On Sun, 2007-04-22 at 23:31 +0307, Black X wrote: > > > Hello > > > > > > First at all I wanna thanks Tobin Davis for his support to get Conexant > > > audio devices working under Linux, and I was testing his last patchs, > > > but I had a little problems with my sound card, the first one is about > > > headphone, it is works but with noise, and when I'm maximize the volume > > > of headphone the sound quality starting so weak. > > > > > > The Second problem I had it is when I run the 3D Desktop Effects (XGL > > > with Compiz), that when I play any sound clip and moving any window or > > > rotate desktop cube that's making a noise with the sound, and the > > > quality of the sound clip it is so weak with slow playing in the sound. > > > > > > I make a little screen cast about this problem and you can see it at > > > this link: > > > http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html > > > > > > any way this is my alsa information: http://pastebin.ca/452453 > > > finaly my device information : > > > http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… > > > > > > I 'm sorry, but the English it's not my mother language. > > > > > > Thank you. > > > > > > _______________________________________________ > > > 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> > -- Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Ok, I try that before and try it now, still no change, and still have the same problem. When I highest the sound volume of headphone this makes sound starting noisy (and that's the same with PCM). About speaker it's work fine, but IF I moving any window or rotate the desktop cube this makes the sound very slow with noise.

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[alsa-devel] partial fix for bug #2708: [ALSA - driver] PCI - hda-intel
by Jeffrey Arbuckle 23 Apr '07

23 Apr '07

Using alsa-driver-1.0.14rc3 with an iMac Core2 Duo 24" results in audio only available through the headphones. Additionally the sound is very low, even at the highest mixer setting. This problem is very similar to the bug recently fixed in 1.0.14rc2 with the audio on the MacPro. I was able to make a patch that applies the same fix for the MacPro to my iMac. The patch fixes the audio so that the built in speakers work normally, but now the headphones do not work at all. Obviously some more work is needed on this. The patch I've attached just adds my device's' PCI SSID to an existing switch statement. I am willing to provide some more testing if necessary. Here is my systems info: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Realtek Semiconductor Co., Ltd. Unknown device 0885 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 92400000 (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 # cat /proc/asound/card0/codec#0 Codec: Realtek ALC885 Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x106b1000 Revision Id: 0x100103 # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14rc3. Compiled on Apr 22 2007 for kernel 2.6.20-15-generic (SMP).

2 1

[alsa-devel] Problem with Conexant
by Black X 23 Apr '07

23 Apr '07

Hello First at all I wanna thanks Tobin Davis for his support to get Conexant audio devices working under Linux, and I was testing his last patchs, but I had a little problems with my sound card, the first one is about headphone, it is works but with noise, and when I'm maximize the volume of headphone the sound quality starting so weak. The Second problem I had it is when I run the 3D Desktop Effects (XGL with Compiz), that when I play any sound clip and moving any window or rotate desktop cube that's making a noise with the sound, and the quality of the sound clip it is so weak with slow playing in the sound. I make a little screen cast about this problem and you can see it at this link: http://www.zshare.net/download/conexantproblem-ogg-tar-gz.html any way this is my alsa information: http://pastebin.ca/452453 finaly my device information : http://h10025.www1.hp.com/ewfrf/wc/document?docname=c00715998&dlc=en&lc=en&… I 'm sorry, but the English it's not my mother language. Thank you.

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[alsa-devel] Qustions on dmix - aplay - hw_ptr - appl_ptr
by Matthew Foster 23 Apr '07

23 Apr '07

I am a driver developer that is new to ALSA. I have read the tutorial as an example on how to write an ALSA driver. I appreciate the documentation you have provided online. I was wondering if you might have time to possibly answer a question. I am having trouble understanding things in the memory/buffer management area. I am not writing a driver for a PCI device, I am writing a driver that acts as an interface from the ALSA to a render device. So I essentially have a circular buffer I am transferring the ALSA data to. I set up the ALSA buffer like this: snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), ALSA_BUF_SIZE, ALSA_BUF_SIZE)) < 0); And later on in my driver after open is called I have a buffer polling loop that does the period notification and constantly monitors my render (ouput) buffer. I have my asound.conf setup like this: pcm.mix { type dmix ipc_key 1024 slave { pcm "hw:0,0" rate 48000 } } pcm.!default { type plug slave.pcm "mix" } This is to enable the dmix plugin so I can run aplay more than once and have the two streams mixed in. The problem I am having is that it seems that when I run aplay like this: "aplay close.wav" I only see this pointer moving: substream->runtime->status->hw_ptrbir And if I run aplay like this: "aplay -D plughw:0,0 close.wav" I only see this pointer moving: substream->runtime->control->appl_ptr This seems odd to me and currently in my code I just check which pointer moves and use that to calculate how much data I need to write to my render device. This is so I can support both use cases. Another major problem is that when I play a short wav file 25KB or so and use the dmix "aplay close.wav" it randomly doesn't play at sometimes. Also if I run "aplay close.wav" and then very quickly run "aplay close.wav" again at the same prompt I get corruption of the data which produces screeching. In the case where I run aplay one right after the other I notice that the hw_ptr moves differently (less) than if I run it, wait, run again. I am also noticing also that when I do the aplay over and over quickly that these values change (get lower) in my snd_pcm_prepare callback: chip_data->buf_size = snd_pcm_lib_buffer_bytes(substream); chip_data->period_buf_size = snd_pcm_lib_period_bytes(substream); It seems like those value should remain constant, and I use those values to calculate later in my polling thread to notify of a period lapse. It's almost like the runtime is telling me the period has changed and the buffer size has changed. But this only happens in this case with a short wav file and running aplay over and over quickly. Is there a way to make sure the hw_ptr doesn't get affected by multiple instances of aplay trying to render out small clips. It seems like maybe I am setting up something incorrectly or maybe not blocking/ waiting correctly. I am at a loss in trying to figure this out, wondering if you would be so kind as to comment on my situation. I really appreciate your time in reading this email. Matt

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[alsa-devel] Fwd: No sound with ATI SB450
by Ben DiDonc 23 Apr '07

23 Apr '07

Tobin, I downloaded and installed alsa-driver, alsa-lib and alsa-utils, but still no sound coming out of those speakers... I ran alsa-info.sh and the results are at http://pastebin.ca/451128 I am clueless as to what to do next, but I m ready to try anything. Any help would be greatly appreciated. Ben On 4/21/07, Tobin Davis < tdavis(a)dsl-only.net> wrote: > > Try downloading a current daily snapshot from ftp://ftp.suse.com/pub/projects/alsa/snapshot/ > . > > There was an issue with the ATI HD Audio detection that was fixed after > 14rc3 was released. > > Tobin > > On Sat, 2007-04-21 at 16:00 +1000, Ben DiDonc wrote: > > The output of alsa-info.sh is available at http://pastebin.ca/451007 > > > On 4/21/07, *Tobin Davis* < tdavis(a)dsl-only.net> wrote: > > Could you download an run http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > , then post the resulting link? This will help determine what setup you > have. > > Tobin Davis > > > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. > Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I > just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) > and the only mixer settings available are those of the modem, named off-hook > and caller-id (see output of amixer, aplay -l , and the rest in attached > file output_feisty) I decided to upgrade alsa-driver and alsa-lib to > 1.0.14rc2. after downloading the package, I unloaded all sound related > modules (everything with "snd_" in it) in the right order and did: > ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; > sudo modprobe snd_hda_intel A minor improvement: the mixer now shows > controls for both the modem and the sound card (see > output_after_upgrade_to_rc2), but unfortunately, there is still no sound > coming out of those speakers (even though alsamixer shows control for PCM > and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it > a shot. This time again, only modem related settings appeared in alsamixer, > and no sound card stuff (see aplay and amixer in > output_after_upgrade_to_rc3), and no sound either. Does anyone know how to > get some sound out of these speakers (given that it used to work with > alsa1.0.12rc1...) ? Thanks for your help Ben > > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> > > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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Re: [alsa-devel] iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 22 Apr '07

22 Apr '07

[Nice analysis cut out..] At Tue, 03 Apr 2007 10:17:16 +0200, Dominique Dumont wrote: > > If the stream format is not valid when the digi converter is set up, > there's a chance that the digi converter goes belly up. > > I reckon that setting the stream *then* the digi converter would be safer. OK, now question is whether we need to turn off DIGI_CONV* once before setting it up again. Anyway, a test patch is below. It forces to set DIGI_CONV* verbs in prepare. Give it a try. thanks, Takashi diff -r 05ecca0fba92 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_codec.c Tue Apr 03 15:28:47 2007 +0200 @@ -1083,6 +1083,19 @@ static int snd_hda_spdif_default_put(str return change; } +static void setup_digi_convert(struct hda_codec *codec, hda_nid_t nid) +{ + unsigned short val = codec->spdif_ctls; + + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, + val & 0xff); + if (get_wcaps(codec, nid) & AC_WCAP_OUT_AMP) + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, + AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | + AC_AMP_SET_OUTPUT | + ((val & 1) ? 0 : 0x80)); +} + static int snd_hda_spdif_out_switch_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; @@ -1114,10 +1127,7 @@ static int snd_hda_spdif_out_switch_put( change = codec->spdif_ctls != val; if (change || codec->in_resume) { codec->spdif_ctls = val; - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, val & 0xff); - snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_AMP_GAIN_MUTE, - AC_AMP_SET_RIGHT | AC_AMP_SET_LEFT | - AC_AMP_SET_OUTPUT | ((val & 1) ? 0 : 0x80)); + setup_digi_convert(codec, nid); } mutex_unlock(&codec->spdif_mutex); return change; @@ -1901,6 +1911,21 @@ int snd_hda_multi_out_dig_open(struct hd return 0; } +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + mutex_lock(&codec->spdif_mutex); + snd_hda_codec_setup_stream(codec, mout->dig_out_nid, + stream_tag, 0, format); + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); + mutex_unlock(&codec->spdif_mutex); + return 0; +} + /* * release the digital out */ @@ -1949,6 +1974,8 @@ int snd_hda_multi_out_analog_prepare(str mout->dig_out_used = 0; snd_hda_codec_setup_stream(codec, mout->dig_out_nid, 0, 0, 0); } + /* set up DIGI_* verbs again */ + setup_digi_convert(codec, mout->dig_out_nid); } mutex_unlock(&codec->spdif_mutex); diff -r 05ecca0fba92 pci/hda/hda_local.h --- a/pci/hda/hda_local.h Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/hda_local.h Tue Apr 03 15:31:29 2007 +0200 @@ -148,6 +148,11 @@ struct hda_multi_out { int snd_hda_multi_out_dig_open(struct hda_codec *codec, struct hda_multi_out *mout); int snd_hda_multi_out_dig_close(struct hda_codec *codec, struct hda_multi_out *mout); +int snd_hda_multi_out_dig_prepare(struct hda_codec *codec, + struct hda_multi_out *mout, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_open(struct hda_codec *codec, struct hda_multi_out *mout, struct snd_pcm_substream *substream); int snd_hda_multi_out_analog_prepare(struct hda_codec *codec, struct hda_multi_out *mout, diff -r 05ecca0fba92 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Apr 03 13:20:49 2007 +0200 +++ b/pci/hda/patch_realtek.c Tue Apr 03 15:32:23 2007 +0200 @@ -1916,6 +1916,17 @@ static int alc880_dig_playback_pcm_open( return snd_hda_multi_out_dig_open(codec, &spec->multiout); } +static int alc880_dig_playback_pcm_prepare(struct hda_pcm_stream *hinfo, + struct hda_codec *codec, + unsigned int stream_tag, + unsigned int format, + struct snd_pcm_substream *substream) +{ + struct alc_spec *spec = codec->spec; + return snd_hda_multi_out_dig_prepare(codec, &spec->multiout, + stream_tag, format, substream); +} + static int alc880_dig_playback_pcm_close(struct hda_pcm_stream *hinfo, struct hda_codec *codec, struct snd_pcm_substream *substream) @@ -1984,7 +1995,8 @@ static struct hda_pcm_stream alc880_pcm_ /* NID is set in alc_build_pcms */ .ops = { .open = alc880_dig_playback_pcm_open, - .close = alc880_dig_playback_pcm_close + .close = alc880_dig_playback_pcm_close, + .prepare = alc880_dig_playback_pcm_prepare }, };

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[alsa-devel] Please ignore, test.
by Rene Herman 22 Apr '07

22 Apr '07

Body Attachment

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Re: [alsa-devel] No sound with ATI SB450
by Ben DiDonc 21 Apr '07

21 Apr '07

The output of alsa-info.sh is available at http://pastebin.ca/451007 On 4/21/07, Tobin Davis <tdavis(a)dsl-only.net> wrote: > > Could you download an run > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh , then post the > resulting link? This will help determine what setup you have. > > Tobin Davis > > On Sat, 2007-04-21 at 14:56 +1000, Ben DiDonc wrote: > > HiI have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450.Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy).I just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa1.0.14rc1) and the only mixer settings available are those of themodem, named off-hook and caller-id (see output of amixer, aplay -l ,and the rest in attached file output_feisty) > I decided to upgrade alsa-driver and alsa-lib to 1.0.14rc2. afterdownloading the package, I unloaded all sound related modules(everything with "snd_" in it) in the right order and did:./configure --with-oss=yes --with-cards=hda-intel;make; sudo makeinstall; sudo modprobe snd_hda_intelA minor improvement: the mixer now shows controls for both the modemand the sound card (see output_after_upgrade_to_rc2), butunfortunately, there is still no sound coming out of those speakers(even though alsamixer shows control for PCM and the rest...). > Finally, I downloaded alsa-driver-1.0.14rc3 and gave it a shot. Thistime again, only modem related settings appeared in alsamixer, and nosound card stuff (see aplay and amixer inoutput_after_upgrade_to_rc3), and no sound either. > Does anyone know how to get some sound out of these speakers (giventhat it used to work with alsa1.0.12rc1...) ? > Thanks for your helpBen > _______________________________________________Alsa-devel mailing listAlsa-devel@alsa-project.orghttp://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- > Tobin Davis <tdavis(a)dsl-only.net> >

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[alsa-devel] No sound with ATI SB450
by Ben DiDonc 21 Apr '07

21 Apr '07

Hi I have a toshiba A100 (see attached file lspci-nnv) with a ATI SB450. Sound worked fine with kernel 2.6.17 and alsa 1.0.12rc1 (ubuntu edgy). I just did a fresh install of ubuntu feisty (kernel 2.6.20, alsa 1.0.14rc1) and the only mixer settings available are those of the modem, named off-hook and caller-id (see output of amixer, aplay -l , and the rest in attached file output_feisty) I decided to upgrade alsa-driver and alsa-lib to 1.0.14rc2. after downloading the package, I unloaded all sound related modules (everything with "snd_" in it) in the right order and did: ./configure --with-oss=yes --with-cards=hda-intel;make; sudo make install; sudo modprobe snd_hda_intel A minor improvement: the mixer now shows controls for both the modem and the sound card (see output_after_upgrade_to_rc2), but unfortunately, there is still no sound coming out of those speakers (even though alsamixer shows control for PCM and the rest...). Finally, I downloaded alsa-driver-1.0.14rc3 and gave it a shot. This time again, only modem related settings appeared in alsamixer, and no sound card stuff (see aplay and amixer in output_after_upgrade_to_rc3), and no sound either. Does anyone know how to get some sound out of these speakers (given that it used to work with alsa1.0.12rc1...) ? Thanks for your help Ben

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[alsa-devel] [PATCHv2] ALSA: Support Media Vision Jazz16 chips
by Rask Ingemann Lambertsen 20 Apr '07

20 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add support for Jazz16 features to the common SoundBlaster code. - Add a front end for Jazz16 based sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20.6. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch is a prerequisite for the Jazz PnP patch, which needs the patched include/sound/sb.h. Documentation/sound/alsa/ALSA-Configuration.txt | 10 include/sound/sb.h | 5 sound/isa/Kconfig | 15 + sound/isa/sb/Makefile | 2 sound/isa/sb/jazz16.c | 183 ++++++++++++++ sound/isa/sb/sb8_main.c | 98 +++++++ sound/isa/sb/sb_common.c | 12 sound/isa/sb/sb_mixer.c | 3 8 files changed, 323 insertions(+), 5 deletions(-) Changes since the last patch: The same DMA channel is now used for both 8-bit and 16-bit sound. DSP high-speed mode isn't used at all now for the Jazz16. A 16-bit DMA buffer can now be 128 kB instead of 64 kB. diff -rup linux-2.6.20.6-clean/Documentation/sound/alsa/ALSA-Configuration.txt linux-2.6.20.6-ril/Documentation/sound/alsa/ALSA-Configuration.txt --- linux-2.6.20.6-clean/Documentation/sound/alsa/ALSA-Configuration.txt 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/Documentation/sound/alsa/ALSA-Configuration.txt 2007-04-07 17:31:38.000000000 +0200 @@ -1120,6 +1120,16 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based on Media Vision Jazz16 chip (PnP only) + + This module supports multiple cards and PnP. + + Note: You will most likely want to enable Jazz16 PnP support + (CONFIG_JAZZ16PNP) in the kernel. + Module snd-korg1212 ------------------- diff -rup linux-2.6.20.6-clean/include/sound/sb.h linux-2.6.20.6-ril/include/sound/sb.h --- linux-2.6.20.6-clean/include/sound/sb.h 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/include/sound/sb.h 2007-04-07 17:31:38.000000000 +0200 @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ @@ -142,6 +143,7 @@ struct snd_sb { #define SB_DSP_HI_INPUT_AUTO 0x98 #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +267,9 @@ struct snd_sb { #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ diff -rup linux-2.6.20.6-clean/sound/isa/Kconfig linux-2.6.20.6-ril/sound/isa/Kconfig --- linux-2.6.20.6-clean/sound/isa/Kconfig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/Kconfig 2007-04-07 17:31:38.000000000 +0200 @@ -248,6 +248,21 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Media Vision Jazz16 based + soundcards. + + You probably want to enable Jazz16 Plug and Play support + (CONFIG_JAZZ16PNP) also. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND diff -rup linux-2.6.20.6-clean/sound/isa/sb/Makefile linux-2.6.20.6-ril/sound/isa/sb/Makefile --- linux-2.6.20.6-clean/sound/isa/sb/Makefile 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/Makefile 2007-04-07 17:31:38.000000000 +0200 @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb8_main.c linux-2.6.20.6-ril/sound/isa/sb/sb8_main.c --- linux-2.6.20.6-clean/sound/isa/sb/sb8_main.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb8_main.c 2007-04-12 00:03:35.000000000 +0200 @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Wed Apr 11 23:56:18 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements (mostly ported from sound/oss/sb_common.c). */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,13 +113,52 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + if (c->max > 1) { + unsigned int num = 0, den = 0; + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char stereo, format; + stereo = runtime->channels > 1; rate = runtime->rate; switch (chip->hardware) { case SB_HW_PRO: @@ -123,6 +174,7 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_20: chip->playback_format = SB_DSP_LO_OUTPUT_AUTO; break; @@ -134,8 +186,14 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + else + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -167,6 +225,8 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->playback_format != SB_DSP_OUTPUT) { + if (chip->dma8 > 3) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); @@ -233,7 +293,9 @@ static int snd_sb8_capture_prepare(struc struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; unsigned int mixreg, rate, size, count; + unsigned char stereo, format; + stereo = runtime->channels > 1; rate = runtime->rate; switch (chip->hardware) { case SB_HW_PRO: @@ -250,6 +312,7 @@ static int snd_sb8_capture_prepare(struc break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_20: chip->capture_format = SB_DSP_LO_INPUT_AUTO; break; @@ -261,10 +324,14 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + else + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (stereo || chip->hardware == SB_HW_JAZZ16) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -279,6 +346,8 @@ static int snd_sb8_capture_prepare(struc snd_sbdsp_command(chip, 256 - runtime->rate_den); } if (chip->capture_format != SB_DSP_OUTPUT) { + if (chip->dma8 > 3) + count /= 2; count--; snd_sbdsp_command(chip, SB_DSP_BLOCK_SIZE); snd_sbdsp_command(chip, count & 0xff); @@ -443,6 +512,19 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; @@ -465,6 +547,14 @@ static int snd_sb8_open(struct snd_pcm_s } snd_pcm_hw_constraint_ratnums(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_clock); + if (chip->dma8 > 3) { + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 2); + snd_pcm_hw_constraint_step(runtime, 0, + SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 2); + runtime->hw.buffer_bytes_max = 128 * 1024 * 1024; + runtime->hw.period_bytes_max = 128 * 1024 * 1024; + } return 0; } @@ -512,6 +602,7 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, struct snd_card *card = chip->card; struct snd_pcm *pcm; int err; + size_t max_prealloc; if (rpcm) *rpcm = NULL; @@ -524,9 +615,10 @@ int snd_sb8dsp_pcm(struct snd_sb *chip, snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_sb8_playback_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_sb8_capture_ops); + max_prealloc = (chip->dma8 > 3) ? 128 * 1024 : 64 * 1024; snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, snd_dma_isa_data(), - 64*1024, 64*1024); + 64*1024, max_prealloc); if (rpcm) *rpcm = pcm; diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb_common.c linux-2.6.20.6-ril/sound/isa/sb/sb_common.c --- linux-2.6.20.6-clean/sound/isa/sb/sb_common.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb_common.c 2007-04-11 20:23:55.000000000 +0200 @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; diff -rup linux-2.6.20.6-clean/sound/isa/sb/sb_mixer.c linux-2.6.20.6-ril/sound/isa/sb/sb_mixer.c --- linux-2.6.20.6-clean/sound/isa/sb/sb_mixer.c 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20.6-ril/sound/isa/sb/sb_mixer.c 2007-04-07 17:31:38.000000000 +0200 @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: --- /dev/null 2007-04-11 19:30:17.320748640 +0200 +++ linux-2.6.20.6-ril/sound/isa/sb/jazz16.c 2007-04-05 23:30:27.000000000 +0200 @@ -0,0 +1,183 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * 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 Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * There is also an MPU-401 interface. + * + * AFAIK, the following systems have an on-board Jazz16 chip: + * IBM PPS Model 6015 + * Texas Intstruments/Acer Travelmate TM5000 series. + * Please tell me if this driver works with these boards or not. + */ + +#include <sound/driver.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +MODULE_AUTHOR("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION("Jazz16"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Jazz16 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Jazz16 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Jazz16 soundcard."); + +#define PFX "jazz16: " +#define PFX_WARN KERN_WARNING PFX +#define PFX_ERR KERN_ERR PFX + +static struct pnp_driver snd_jazz16_pnp_driver; + +static irqreturn_t jazz16_interrupt(int irq, void *card) +{ + return snd_sb8dsp_interrupt(card); +} + +static struct pnp_device_id snd_jazz16_pnpids[] = { + { .id = "PNPb00f" }, + { .id = "" } +}; +MODULE_DEVICE_TABLE(pnp, snd_jazz16_pnpids); + +static int __devinit snd_jazz16_pnp_probe(struct pnp_dev *pnp_dev, + const struct pnp_device_id *pnp_id) +{ struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int err; + uint sbport, sbirq, sbdma, mpuport, mpuirq; + static uint dev_num = 0; + + if (enable[dev_num]) + card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); + else + card = NULL; + dev_num++; + if (card == NULL) + return enable[dev_num] ? -ENOMEM : -ENODEV; + snd_card_set_dev(card, &pnp_dev->dev); + pnp_set_drvdata(pnp_dev, card); + /* TODO use pnp_port_valid(), pnp_port_flags(), pnp_port_length()... */ + sbport = pnp_port_start(pnp_dev, 0); + sbirq = pnp_irq(pnp_dev, 0); + sbdma = pnp_dma(pnp_dev, 0); + err = snd_sbdsp_create(card, sbport, sbirq, jazz16_interrupt, + sbdma, -1, SB_HW_AUTO, &chip); + if (err < 0) + goto out_err; + + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at %#x, irq %u, dma %u", + chip->name, sbport, sbirq, chip->dma8); + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(PFX_ERR "Not a Jazz16 chip at %#x.\n", sbport); + snd_card_free(card); + return -ENODEV; + } + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) + goto out_err; + + err = snd_sbmixer_new(chip); + if (err < 0) + goto out_err; + + err = snd_opl3_create(card, sbport, sbport + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) + snd_printk(PFX_WARN "No OPL device found, skipping.\n"); + else { + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto out_err; + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto out_err; + } + if (pnp_port_valid(pnp_dev, 1) + && !(pnp_port_flags(pnp_dev, 1) & IORESOURCE_DISABLED)) { + int mpuirqflags; + if (!pnp_irq_valid(pnp_dev, 1) + || pnp_irq_flags(pnp_dev, 1) & IORESOURCE_DISABLED) { + mpuirq = -1; + mpuirqflags = 0; + } else { + mpuirq = pnp_irq(pnp_dev, 1); + mpuirqflags = IRQF_DISABLED; + } + mpuport = pnp_port_start(pnp_dev, 1); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport, 0, mpuirq, + mpuirqflags, NULL); + if (err < 0) + snd_printk(PFX_WARN "MPU-401 device not configured, skipping.\n"); + } + err = snd_card_register(card); + if (err >= 0) + return 0; +out_err: + snd_card_free(card); + return err; +} + +static void __devexit snd_jazz16_pnp_remove(struct pnp_dev *dev) +{ + snd_card_free(pnp_get_drvdata(dev)); + pnp_set_drvdata(dev, NULL); +} + +static struct pnp_driver snd_jazz16_pnp_driver = { + .name = "Jazz16", + .id_table = snd_jazz16_pnpids, + .probe = snd_jazz16_pnp_probe, + .remove = __devexit_p(snd_jazz16_pnp_remove), +}; + +static int __devinit alsa_card_jazz16_init(void) +{ + return pnp_register_driver(&snd_jazz16_pnp_driver); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + pnp_unregister_driver(&snd_jazz16_pnp_driver); +} + +module_init(alsa_card_jazz16_init); +module_exit(alsa_card_jazz16_exit); -- Rask Ingemann Lambertsen

3 3

[alsa-devel] [Alsa-devel] Read the input of a soundcard with ALSA
by Nyx 20 Apr '07

20 Apr '07

Hi, I have some problems to read the input of my soundcard using Alsa and I don't really understand how to have access to the input of my soundcard. I follow : http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html http://www.alsa-project.org/alsa-doc/alsa-lib/_2test_2latency_8c-example.ht… # So, in my program, I have created two handles for capture and playback : char *device = "hw:0,0"; snd_output_t *output; snd_input_t *input; snd_pcm_t *phandle, *chandle; # then I connect the phandle to the out err = snd_output_stdio_attach(&output, stdout, 0); if (..) # Do I have to connect the chandle with the input ? # err = snd_input_stdio_attach(&input, stdin, 0); # I open the access to PCM err = snd_pcm_open(&phandle, pdevice, SND_PCM_STREAM_PLAYBACK, 0) if (..) err = snd_pcm_open(&chandle, pdevice, SND_PCM_STREAM_CAPTURE, 0) if(..) # Hardware and software Parameters ## Hardware Parameters snd_pcm_hw_params; int err; err = snd_pcm_hw_params_any(phandle, params); if(..); err = snd_pcm_hw_params_set_access(phandle, params,SND_PCM_ACCESS_RW_INTERLEAVED); if(..); err = snd_pcm_hw_params_set_format(phandle, params,SND_PCM_FORMAT_S16); if(..); err = snd_pcm_hw_params_set_channels(phandle, params, 1); if(..); err = snd_pcm_hw_params_set_rate_near(phandle, params, 44100, 0); if(..); err = snd_pcm_hw_params_set_buffer_time_near(phandle, params, 500000, &dir); if(..); err = snd_pcm_hw_params_set_time_near(phandle, params, 100000, &dir), if(..); err = snd_pcm_hw_params(phandle, params); ##Software parameters err = snd_pcm_sw_params(phandle, params); if(..); err = snd_pcm_sw_params_set_start_threshold(phandle, swparams, 0x7fffffff); if(..); err = snd_pcm_sw_params_set_avail_min(phandle, swparams, 4); # same configuration for phandle # Link output to input and start err = snd_pcm_link(chandle, phandle); if(..); err = snd_pcm_start(chandle); if(..); #Then I want to read my input frames_in = 0; in_max = 0; latency = 28; buffer = malloc((latency_max * snd_pcm_format_width(format) / 8) * 2); while (1) { r = readbuf(chandle, buffer, latency, &frames_in, &in_max)); if(..); } # description of the function long readbuf(snd_pcm_t *handle, char *buf, long len, size_t *frames, size_t *max) { long r; do { r = snd_pcm_readi(handle, buf, len); } while (r == -EAGAIN); if (r > 0) { *frames += r; if ((long)*max < r) *max = r; } printf ("r = %s, len %li and buf[%d] %f \r", snd_strerror(r), len,i, buf[i]); return r; } So I compile it and I run it, and I have $ read = Broken Pipe, len = 28 and buf[..] = 0.000 Anyone can help me to read the input of my sound card ? Thank you -- K

1 0

Re: [alsa-devel] [PATCH 16/16] alsa sound support for mips
by Ralf Baechle 20 Apr '07

20 Apr '07

On Sun, Apr 15, 2007 at 11:26:05PM +0800, tiansm(a)lemote.com wrote: (Adding a few more people to the cc'list) > sound/core/pcm_native.c | 10 ++++++++++ > sound/core/sgbuf.c | 9 +++++++++ > 2 files changed, 19 insertions(+), 0 deletions(-) > > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 3e276fc..9005bac 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -3145,7 +3145,11 @@ 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) Please use CONFIG_MIPS instead of __mips__ in #if / #ifdefs. The question if #ifdefing is the right approach to solve this problem is something else but I think no, .... > + page = virt_to_page(CAC_ADDR(vaddr)); > +#else > page = virt_to_page(vaddr); > +#endif So this is needed because the MIPS virt_to_page is returning a unsuitable value if vaddress is not a KSEG0 (64-bit: cached XKPHYS) address which is what GFP allocations and the slab will return. So now we have to deciede if a) the MIPS __pa() should be changed to handle uncached addresses. b) the sound code here is simply broken. Some drivers seem to allocate runtime->dma_area from vmalloc, so this whole area in the sound code is looking like built on quicksand ... > } > get_page(page); > if (type) > @@ -3261,6 +3265,12 @@ 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); > + area->vm_flags |= ( VM_RESERVED | VM_IO); VM_RESERVED will prevent the buffer from being freed. I assume that is another workaround for some kernel subsystem blowing up when being fed a pointer to an uncached RAM address? This smells like a memory leak. > +#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..535f0bc 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)); VM_RESERVED will prevent the buffer from being freed. I assume that is another workaround for some kernel subsystem blowing up when being fed a pointer to an uncached RAM address? This smells like a memory leak. > +#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 | VM_IO, pgprot_noncached(PAGE_KERNEL)); > +#else > dmab->area = vmap(sgbuf->page_table, sgbuf->pages, VM_MAP, PAGE_KERNEL); > +#endif I would suggest to get rid of this ifdef with a new arch-specific function like vmap_io_buffer which will do whatever a platform seems fit for this case? > if (! dmab->area) > goto _failed; > return dmab->area; Ralf

3 2

[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by Pablo N. Hess 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

2 1

[alsa-devel] [KJ][PATCH] sound: SPIN_LOCK_UNLOCKED cleanup
by Milind Arun Choudhary 19 Apr '07

19 Apr '07

SPIN_LOCK_UNLOCKED cleanup,use __SPIN_LOCK_UNLOCKED instead Signed-off-by: Milind Arun Choudhary <milindchoudhary(a)gmail.com> --- soc/at91/at91-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/at91/at91-i2s.c b/sound/soc/at91/at91-i2s.c index 9fc0c03..d65a47d 100644 --- a/sound/soc/at91/at91-i2s.c +++ b/sound/soc/at91/at91-i2s.c @@ -151,20 +151,20 @@ static struct at91_ssc_info { } ssc_info[NUM_SSC_DEVICES] = { { .name = "ssc0", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[0].lock), .dir_mask = 0, .initialized = 0, }, #if NUM_SSC_DEVICES == 3 { .name = "ssc1", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[1].lock), .dir_mask = 0, .initialized = 0, }, { .name = "ssc2", - .lock = SPIN_LOCK_UNLOCKED, + .lock = __SPIN_LOCK_UNLOCKED(ssc_info[2].lock), .dir_mask = 0, .initialized = 0, }, -- Milind Arun Choudhary

1 0

[alsa-devel] CS4270 supported?
by Timur Tabi 19 Apr '07

19 Apr '07

I've never worked with ALSA before, but I'm about to start developing on an ALSA driver for our hardware, which has a CS4270 on it attached via an I2S interface. After scanning the source code and mailing lists, I see no mention of the CS4270. So does that mean I would need to write an ASoC codec driver for the CS4270? If so, does the data sheet (http://www.cirrus.com/en/pubs/proDatasheet/CS4270_PP1.pdf) contain enough information? -- Timur Tabi Linux Kernel Developer @ Freescale

4 7

[alsa-devel] Trying to support hda-intel ALC883_CLEVO
by natunobilis＠gentoobr.org 19 Apr '07

19 Apr '07

I have a Clevo M600 notebook, which comes with Realtek's HDA Intel audio chip. 'lspci -vn' says: 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 1558:0660 Flags: bus master, fast devsel, latency 0, IRQ 16 Memory at 6c340000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) Linux 2.6.20 (Gentoo's gentoo-sources-2.6.20r4) recognizes it as an ALC883 chip, but no parameter combination would make the sound work properly (and that includes 'model=clevo'). As this module ships, no output would come from the built-in speakers, and the headphone volume was miserably low. patch_realtek.c supports vendor=1558 device=0660, but only as a ALC880 chip. However, mine *is* an ALC883 vendor=1558 device=0660. After copying the ALC880_CLEVO support code lines into a new ALC883_CLEVO (and changing every occurence of ALC880 to ALC883 in that new code block), recompiling the kernel and loading the snd-hda-intel with model=clevo, almost everything started working fine. I mean, sound comes as perfectly as it can through the built-in speakers, and the built-in mic also outputs its sound through these speakers. However, I can't capture anything. Alsamixer shows no capture device at all (although all playback devices appear to be present). This is very strange to me, as the built-in mic really *does* output its sound through the speakers. As you can probably see, I don't know anything about sound chips. I just did what seemed understandable (to me) to add proper support for my ALC883_CLEVO device. Thank you very much if you can help me solve this issue, as I really need to speak with my family and girlfriend far, far away. Cheers, Pablo Hess

1 0

[alsa-devel] Surround problems on ALC883
by Pavel Patz 19 Apr '07

19 Apr '07

Hi all, I have problems with chip Realtek ALC883 (lspci: 00:06.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2)) and surround sound. Stereo sound is working (games, music) fine - all 6 speaker are playing. But when I tried to run "aplay chan-id.wav", i've heard only "front left" (front & rear left + center were playing) and "front right" (right front & rear were playing), then was silence until the end of the file. Running "aplay -Dsurround51 chan-id.wav" causes that aplay hangs-up without any output. Running it with -v parameter produces this output: Playing WAVE 'Desktop/Surround-SDL-testfiles/chan-id.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Channels 6 Soft volume PCM Control: PCM Playback Volume min_dB: -51 resolution: 256 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 Slave: Hardware PCM card 0 'HDA NVidia' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 6 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 5460 period_size : 1365 period_time : 30952 tick_time : 1000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1365 xfer_align : 1365 start_threshold : 5460 stop_threshold : 5460 silence_threshold: 0 silence_size : 0 boundary : 6147413491360727040 and then hang up until i press ctrl+c. Same results i get with speaker-test. I'm using alsa-lib-1.0.14-0.4.rc3.fc7 I've found similar problem in archive (about half a year old) but it still seems it isn't working. Is there any possibility, that this will be fixed in future? Thanks for all your help (and please leave me in CC, i'm not on mailing-list). Following lsmod & /proc/asound/card0/codec#0 [cdome@andromeda ~]$ /sbin/lsmod |grep snd snd_hda_intel 27936 2 snd_hda_codec 262400 1 snd_hda_intel snd_seq_dummy 11908 0 snd_seq_oss 40448 0 snd_seq_midi_event 16128 1 snd_seq_oss snd_seq 60704 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi_event snd_seq_device 16404 3 snd_seq_dummy,snd_seq_oss,snd_seq snd_pcm_oss 50944 0 snd_mixer_oss 24320 2 snd_pcm_oss snd_pcm 90888 3 snd_hda_intel,snd_hda_codec,snd_pcm_oss snd_timer 30088 2 snd_seq,snd_pcm snd 67624 12 snd_hda_intel,snd_hda_codec,snd_seq_dummy,snd_seq_oss,snd_seq,snd_seq_device,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer soundcore 17505 2 snd snd_page_alloc 16912 2 snd_hda_intel,snd_pcm [cdome@andromeda ~]$ cat /proc/asound/card0/codec#0 Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x1458e601 Revision Id: 0x100002 Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x10 0x10] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x08, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x17 0x17] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x1a] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x1a 0x04] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01014410: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01011412: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01016411: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x083e: IN OUT HP Detect Pin Default 0x01012414: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey Pin-ctls: 0x40: OUT Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01a19c40: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02a19c50: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x01813c41: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x02214c20: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x9933014f: [Fixed] CD at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x014b6130: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = Orange Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x01cb7160: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = Yellow Pin-ctls: 0x00: Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x22 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Amp-Out caps: N/A Amp-Out vals: [0x00 0x00] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x25 [Audio Output] wcaps 0x11: Stereo PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b

3 8

[alsa-devel] snd_pcm_hardware_t.info question
by Ciaccia 18 Apr '07

18 Apr '07

Hi all, I am trying to fix (improve?) a driver for an embedded ARM device I bought months ago, and for some reasons some ALSA applications work fine while other ones don't... Looking at the driver I got, I noticed that the info field in the snd_pcm_hardware_t struct just defines: SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_PAUSE while the info in the tutorial "writing an alsa driver" defines: SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP_VALID Furthermore, the driver defines a copy callback (actually 2, one for playback and one for capture), where it copies the data from the use space to the dma buffer. Since the dma buffer was preallocated with snd_pcm_lib_preallocate_pages_for_all and is accessible from the "outside", I wonder why the original author did this... I have some questions for you gurus: - what is the difference between SNDRV_PCM_INFO_MMAP and SNDRV_PCM_INFO_MMAP_VALID? What is SNDRV_PCM_INFO_BLOCK_TRANSFER? - The code calls snd_pcm_lib_preallocate_pages_for_all function as following: /* allocate the pcm(DMA) memory */ ret = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,0, 4*128*1024, 4*128*1024); is the DMA_TYPE right for an ARM device? Is NULL a correct value for data in this case? - In case I get rid of the copy callback, how do I specify the format of the stream? How does Where can I find an example for that? Thanks a lot Andrea __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 5

[alsa-devel] [PATCH] sound: convert "sound" subdirectory to UTF-8
by John Anthony Kazos Jr. 17 Apr '07

17 Apr '07

From: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> Convert the "sound" subdirectory to UTF-8. The following files are changed. sound/oss/{trident.c,es1371.c,pas2_pcm.c} sound/oss/emu10k1/{passthrough.h,passthrough.c} sound/pci/mixart/mixart.c Signed-off-by: John Anthony Kazos Jr. <jakj(a)j-a-k-j.com> --- I can't get my mail client to send in ISO-8859-1 explicitly instead of whatever it decides I want, so the actual patch is attached in octet-stream format. The below patch is just for reference and will very likely NOT work, unless this time it picks ISO-8859-1. Also, since the patch includes both encodings, whichever encoding you select while viewing it will display one correct and one garbled line. However, the bytecodes -are- correct. diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.c 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.c -- Emu10k1 digital passthrough - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h --- linux-2.6.21-rc7.orig/sound/oss/emu10k1/passthrough.h 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/emu10k1/passthrough.h 2007-04-17 13:55:45.000000000 -0400 @@ -1,13 +1,13 @@ /* ********************************************************************** * passthrough.h -- Emu10k1 digital passthrough header file - * Copyright (C) 2001 Juha Yrjölä <jyrjola(a)cc.hut.fi> + * Copyright (C) 2001 Juha YrjÃ¶lÃ¤ <jyrjola(a)cc.hut.fi> * ********************************************************************** * * Date Author Summary of changes * ---- ------ ------------------ - * May 15, 2001 Juha Yrjölä base code release + * May 15, 2001 Juha YrjÃ¶lÃ¤ base code release * ********************************************************************** * diff -uprN linux-2.6.21-rc7.orig/sound/oss/es1371.c linux-2.6.21-rc7.mod/sound/oss/es1371.c --- linux-2.6.21-rc7.orig/sound/oss/es1371.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/es1371.c 2007-04-17 13:55:45.000000000 -0400 @@ -100,7 +100,7 @@ * Tjeerd Mulder <tjeerd.mulder(a)fujitsu-siemens.com> * 05.01.2001 0.29 Hopefully updates will not be required anymore when Creative bumps * the CT5880 revision. - * suggested by Stephan Müller <smueller(a)chronox.de> + * suggested by Stephan MÃ¼ller <smueller(a)chronox.de> * 31.01.2001 0.30 Register/Unregister gameport * Fix SETTRIGGER non OSS API conformity * 14.07.2001 0.31 Add list of laptops needing amplifier control diff -uprN linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c --- linux-2.6.21-rc7.orig/sound/oss/pas2_pcm.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/oss/pas2_pcm.c 2007-04-17 13:55:45.000000000 -0400 @@ -85,7 +85,7 @@ static int pcm_set_speed(int arg) * come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/. * * I cleared bit 5 of these values, since that bit controls the master - * mute flag. (Olav Wölfelschneider) + * mute flag. (Olav WÃ¶lfelschneider) * */ #if !defined NO_AUTO_FILTER_SET diff -uprN linux-2.6.21-rc7.orig/sound/oss/trident.c linux-2.6.21-rc7.mod/sound/oss/trident.c --- linux-2.6.21-rc7.orig/sound/oss/trident.c 2007-04-17 11:42:18.000000000 -0400 +++ linux-2.6.21-rc7.mod/sound/oss/trident.c 2007-04-17 13:55:45.000000000 -0400 @@ -18,7 +18,7 @@ * Ollie Lho <ollie(a)sis.com.tw> SiS 7018 Audio Core Support * Ching-Ling Lee <cling-li(a)ali.com.tw> ALi 5451 Audio Core Support * Matt Wu <mattwu(a)acersoftech.com.cn> ALi 5451 Audio Core Support - * Peter Wächtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support + * Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> CyberPro5050 support * Muli Ben-Yehuda <mulix(a)mulix.org> * * @@ -89,7 +89,7 @@ * use set_current_state, properly release resources on failure in * trident_probe, get rid of check_region * v0.14.9c - * August 10 2001 Peter Wächtler <pwaechtler(a)loewe-komp.de> + * August 10 2001 Peter WÃ¤chtler <pwaechtler(a)loewe-komp.de> * added support for Tvia (formerly Integraphics/IGST) CyberPro5050 * this chip is often found in settop boxes (combined video+audio) * v0.14.9b diff -uprN linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c --- linux-2.6.21-rc7.orig/sound/pci/mixart/mixart.c 2007-02-04 13:44:54.000000000 -0500 +++ linux-2.6.21-rc7.mod/sound/pci/mixart/mixart.c 2007-04-17 13:55:45.000000000 -0400 @@ -472,7 +472,7 @@ static int snd_mixart_prepare(struct snd struct snd_mixart *chip = snd_pcm_substream_chip(subs); struct mixart_stream *stream = subs->runtime->private_data; - /* TODO de façon non bloquante, réappliquer les hw_params (rate, bits, codec) */ + /* TODO de faÃ§on non bloquante, rÃ©appliquer les hw_params (rate, bits, codec) */ snd_printdd("snd_mixart_prepare\n");

1 0

[alsa-devel] Request for CS4280 Manual
by Amar singh 17 Apr '07

17 Apr '07

Hello Sir, I'm a newbie in Linux Deivce Drivers and Kernel Development. We're writing our own driver for sound card - Cirrus Logic, Sound Fusion CS4280. in Fedora Core 6 Linux flavour. For that I've been looking for the programming manual of the above said sound card, for a long time now, but didn't succeed in getting one. Since nothing being fruitful for me in this case, we're stuck up with our project, at the start. Therefore I'd kindly like to request you to plz, send me, if you can, the programming manual in the pdf/doc format or a link to the place from where we can get that document. We'd really appreciate your help for the matter. Looking ahead for positive reply from you. Thanks and Regards, Amar singh Emertxe Information Technologies (P) Ltd., Bangalore - 56. India.

3 3

[alsa-devel] [PATCH] ASoC Kconfig description.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch makes the ASoC Kconfig descriptions a little more meaningful. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

2 1

[alsa-devel] Why do I not see patches in the emails?
by Timur Tabi 17 Apr '07

17 Apr '07

Can someone tell me why I can't see that actual patches in the posts that include them? I'm guessing it's some odd configuration option, but I went to http://mailman.alsa-project.org/mailman/options/alsa-devel and I didn't see anything applicable. Here's an example. As you can see, there is a text/plain attachment where the patch would be, but it's empty. Sender: alsa-devel-bounces(a)alsa-project.org Errors-To: alsa-devel-bounces(a)alsa-project.org Return-Path: alsa-devel-bounces(a)alsa-project.org X-OriginalArrivalTime: 16 Apr 2007 13:52:51.0309 (UTC) FILETIME=[864861D0:01C7802E] --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain Content-Transfer-Encoding: 7bit This patch by Frank Madarino updates the AT91xxxx Makefile and Kconfig to build the renamed SSC files. Changes:- o Rename various i2s labels to ssc Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> --=-Uhr6lvaXHMfZKXRJXsBj Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel --=-Uhr6lvaXHMfZKXRJXsBj-- -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] FW: Individual PCM stream control
by Nagalla, Hari 17 Apr '07

17 Apr '07

Thanks for the reply.. We followed your suggestions and we see the PCM stream controls from the user space after we have created the controls at the driver using IFACE_PCM tags.. But when I try to set the 'index' in 'snd_ctl_elem_id' to reflect the PCM substream number, it seems to always get reset to '0'. Can you please suggest, how to set the index. --- snapshot of the code -------- snd_ctl_elem_info_alloca(&info); snd_ctl_elem_id_alloca(&id); snd_ctl_elem_value_alloca(&control); snd_ctl_elem_id_set_numid(id, ctl_num); snd_ctl_elem_id_set_interface(id, SND_CTL_ELEM_IFACE_PCM); fprintf(stdout,"setting the mute/unmute for stream index 0x%x \n", (unsigned int) stream_id); snd_ctl_elem_id_set_index(id,(unsigned int) stream_id); if (handle == NULL && (err = snd_ctl_open(&handle, card, 0)) < 0) { fprintf(stderr,"Control %s open error: %s\n", card, snd_strerror(err)); return err; } snd_ctl_elem_info_set_id(info, id); if ((err = snd_ctl_elem_info(handle, info)) < 0) { fprintf(stderr,"Cannot find the given element from control %s\n", card); return err; } snd_ctl_elem_info_get_id(info, id); /* FIXME: Remove it when hctl find works ok !!! */ type = snd_ctl_elem_info_get_type(info); count = snd_ctl_elem_info_get_count(info); snd_ctl_elem_value_set_id(control, id); show_control_id(id); printf("\n"); fprintf(stdout, "Retrieving the stream index 0x%x \n",snd_ctl_elem_id_get_index(id)); ---------------------------------------------------------------- Thanks a bunch !! Hari > -----Original Message----- > From: Takashi Iwai [mailto:tiwai@suse.de] > Sent: Wednesday, April 04, 2007 11:21 AM > To: Nagalla, Hari > Cc: alsa-devel(a)alsa-project.org > Subject: Re: [alsa-devel] Individual PCM stream control > > At Fri, 30 Mar 2007 05:47:11 -0500, > Nagalla, Hari wrote: > > > > Thanks for the advice.. Couple of inline questions below.. > > > > > > > -----Original Message----- > > > > I am new to ALSA, so some help from experienced folks would help.. > > > > > > > > We are trying to develop ALSA support for our DSP based mixer. In > > our > > > > architecture DSP performs the mixing, sample rate conversion etc. My > > > > question is how, can does the user space applications exercise > > > > controlling individual PCM stream to the mixer? Ex: individual > > stream > > > > gain control, individual stream mute, etc.. Any user space example > > code > > > > would be useful too.. > > > > > > Most drivers create PCM volume controls with IFACE_PCM tag, and set up > > > the volume appropriately at each PCM open, or let alsa-lib configs to > > > set up the initial volumes. Usually, the control element index > > > corresponds to the PCM substream number. > > > > Which alsa library API lets us to set up the PCM stream volume? And can > > it be used dynamically when the stream is active? > > It is controlled via the normal control API. Open the snd_ctl_t > handle for the same card (e.g. "hw:0" or "default"), and change the > value of the element. However, there is no standard name for the > individual PCM volume controls, and currrently depending on the driver > implementation... > > > I could n't figure out 'index' of the control element. > > It's the index in snd_ctl_elem_id. You can access via > snd_ctl_elem_id_{get|set}_index() in alsa-lib. > > > Takashi

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Re: [alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 17 Apr '07

17 Apr '07

I applied the first hint from Jaroslav (at the end of this post the new asound.conf), but the problem seems not gone: one time each 256 seconds I get a very strong cpu load spike, that is maybe related (my guess) to the usage of plugin (maybe periodic cleaning each 256 secs?). When there is no spike, the cpu load is 0.0% with occasional 0.1% any 60-90 secs. I ran the latency test code with plughw:0 (I can't use latency.c with my virtual cards because they can't link the in and out togheter) using the blocking mode, real time, no poll. Any 256 secs, spike! (a cpu load for latency.c of 45.00%, for roughly a sec, with the cumulated cpu time of the latency.c app reported from top -d 1.0 passing from: 0:00.01 to: 0:00.64 to: 0:01.26 etc, any 256 seconds. latency.c output: ./latency -P plughw:0 -C plughw:0 -f S16_LE -c 1 -r 8000 -m 320 -M 960 -s 600 -b Playback device is plughw:0 Capture device is plughw:0 Parameters are 8000Hz, S16_LE, 1 channels, blocking mode Wanted tick time: 0us, poll mode: no Loop limit is 4800000 frames, minimum latency = 320, maximum latency = 960 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0 1 <- 0 2 <- 0 3 <- 0 4 <- 0 5 <- 0 6 <- 0 7 <- 0 8 <- 0 9 <- 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 10 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Plug PCM: Route conversion PCM (sformat=S32_LE) Transformation table: 0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 + 5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 + 10*0.0833333 + 11*0.0833333 Its setup is: stream : CAPTURE access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Slave: Hardware PCM card 0 'M Audio Delta 1010LT' device 0 subdevice 0 Its setup is: stream : CAPTURE access : MMAP_INTERLEAVED format : S32_LE subformat : STD channels : 12 rate : 8000 exact rate : 8000 (8000/1) msbits : 24 buffer_size : 320 period_size : 160 period_time : 20000 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 160 xfer_align : 1 start_threshold : 2147483647 stop_threshold : 320 silence_threshold: 0 silence_size : 0 boundary : 1342177280 Trying latency 320 frames, 40000.000us, 40.000000ms (25.0000Hz) Success Playback: *** frames = 4800320 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482713139 delay : 317 avail : 3 avail_max : 192 Capture: *** frames = 4800000 *** state : RUNNING trigger_time: 1176471431.639255951 tstamp : 1176472031.482757926 delay : 1 avail : 1 avail_max : 189 Maximum read: 160 frames Maximum read latency: 20000.000us, 20.000000ms (50.0000Hz) Hardware sync Playback time = 1176471431.639255, Record time = 1176471431.639255, diff = 0 new asound.conf: pcm.c3 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 2 } } pcm.capt3 { type plug slave.pcm c3 } pcm.c4 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 3 } } pcm.capt4 { type plug slave.pcm c4 } pcm.c5 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 4 } } pcm.capt5 { type plug slave.pcm c5 } pcm.c6 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 5 } } pcm.capt6 { type plug slave.pcm c6 } pcm.c7 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 6 } } pcm.capt7 { type plug slave.pcm c7 } pcm.c8 { type dsnoop ipc_key 56789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } bindings { 0 7 } } pcm.capt8 { type plug slave.pcm c8 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 } On 4/13/07, Giovanni Maruzzelli <gmaruzz.lists(a)gmail.com> wrote: > Thanks a lot Jaroslav, I'll apply your suggestions and I'll report here > > On 4/13/07, Jaroslav Kysela <perex(a)suse.cz> wrote: > > On Fri, 13 Apr 2007, Giovanni Maruzzelli wrote: > > > > > Dear Alsa developers, > > > > > > I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. > > > > > > I want to use the /etc/asound.conf (.asoundrc) to split the card (that > > > has 8 in and 8 out - but I use just 6 in/out), with minimal latency. > > > > > > I mean, I want to record and play (duplex) from/to each virtual card > > > concurrently and from separate application instances. > > > > > > My app works with S16_LE, 8000, mono. > > > > > > Is the following asound.conf (.asoundrc) correct and efficient for this? > > > > Use dsnoop bindings rather than routing (ttable) for capture too, > > otherwise alsa-lib works with all channels internally. > > > > > I got spikes of cpu load (at 5-6%) on each virtual card any roughly > > > 256 seconds when using a(play|record) to record or play from/to any > > > > aplay|arecord is not very good test. Use latency test code - available in > > alsa-lib/test directory. > > > > Jaroslav > > > > ----- > > Jaroslav Kysela <perex(a)suse.cz> > > Linux Kernel Sound Maintainer > > ALSA Project, SUSE Labs > > >

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[alsa-devel] [PATCH 1/2] ASoC Samsung S3c24xx updates - audio DMA cleanup.
by Liam Girdwood 17 Apr '07

17 Apr '07

This patch cleans up the audio DMA for the Samsung S3C24xx platform. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] [ALSA] Add first generation macbook subsystem ID
by bainonline＠gmail.com 17 Apr '07

17 Apr '07

From: Abhijit Bhopatkar <bainonline(a)gmail.com> First generation MacBooks were getting ignored by sigmatel drivers and wrongly being identified as MACMINI. This patch makes them identify as MACBOOK. Signed-off-by: Abhijit Bhopatkar <bainonline(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 3 +++ 1 files changed, 3 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index c94291b..cb99df5 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1905,6 +1905,9 @@ static int patch_stac922x(struct hda_codec *codec) */ printk(KERN_INFO "hda_codec: STAC922x, Apple subsys_id=%x\n", codec->subsystem_id); switch (codec->subsystem_id) { + case 0x106b0a00: /* MacBook First generatoin */ + spec->board_config = STAC_MACBOOK; + break; case 0x106b0200: /* MacBook Pro first generation */ spec->board_config = STAC_MACBOOK_PRO_V1; break; -- 1.4.4.2

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[alsa-devel] [1/2] 2.6.21-rc7: known regressions
by Adrian Bunk 17 Apr '07

17 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : ali_pata: boot from CD fails References : http://lkml.org/lkml/2007/3/31/160 Submitter : Stephen Clark <Stephen.Clark(a)seclark.us> Status : unknown Subject : kernels fail to boot with drives on ATIIXP controller (ACPI/IRQ related) References : https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=229621 http://lkml.org/lkml/2007/3/4/257 Submitter : Michal Jaegermann <michal(a)ellpspace.math.ualberta.ca> Status : unknown Subject : boot failure: rtl8139: exception in interrupt routine References : http://lkml.org/lkml/2007/3/31/160 Submitter : Stephen Clark <Stephen.Clark(a)seclark.us> Status : unknown Subject : laptops with e1000: lockups References : https://bugzilla.redhat.com/bugzilla/show_bug.cgi?id=229603 Submitter : Dave Jones <davej(a)redhat.com> Handled-By : Jesse Brandeburg <jesse.brandeburg(a)intel.com> Status : problem is being debugged Subject : forcedeth: interface hangs under load References : http://lkml.org/lkml/2007/4/3/39 http://www.spinics.net/lists/netdev/msg28981.html Submitter : Ingo Molnar <mingo(a)elte.hu> Handled-By : Ingo Molnar <mingo(a)elte.hu> Ayaz Abdulla <aabdulla(a)nvidia.com> David Miller <davem(a)davemloft.net> Status : problem is being debugged Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : Oops when changing USB DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 http://lkml.org/lkml/2007/4/5/154 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Status : patches available

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[alsa-devel] [PATCH] ASoC export AC97 DAI
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch exports the ASoC AC97 Digital Audio Interface as a GPL'ed symbol. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 1/2] ASoC WM8753 codec support.
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch series adds support for the WM8753 codec as found on the OpenMoko Neo 1973 (other Neo 1973 and Samsung S3C24xx patches to follow today) as well other new devices. Features:- o HiFi and Voice DAI supported (inc runtime switching of DAI mode) o DAPM o All mixers o PLL calculator o 16,20 and 24bit samples. o WM8753 I2C ID added to include/linux/i2c-id.h Signed-off-by: Harald Welte <laforge(a)openmoko.org> Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Seth Forshee <seth.forshee(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com> Privacy & Confidentiality Notice ------------------------------------------------- This message and any attachments contain privileged and confidential information that is intended solely for the person(s) to whom it is addressed. If you are not an intended recipient you must not: read; copy; distribute; discuss; take any action in or make any reliance upon the contents of this message; nor open or read any attachment. If you have received this message in error, please notify us as soon as possible on the following telephone number and destroy this message including any attachments. Thank you. ------------------------------------------------- Wolfson Microelectronics plc Tel: +44 (0)131 272 7000 Fax: +44 (0)131 272 7001 Web: www.wolfsonmicro.com Registered in Scotland Company number SC089839 Registered office: Westfield House, 26 Westfield Road, Edinburgh, EH11 2QB, UK

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[alsa-devel] [PATCH 2/2] ASoC Samsung S3C24xx updates - i2s
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch adds DMA size fields to the S3C24xx audio DMA params and exports the S3C24xx I2S digital audio interface. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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Re: [alsa-devel] [PATCH] Re: iec958 switch uneffective while playing ac3 stream
by Takashi Iwai 16 Apr '07

16 Apr '07

At Thu, 05 Apr 2007 10:47:52 +0200, Dominique Dumont wrote: > > Takashi Iwai <tiwai(a)suse.de> writes: > > > The patch is a bit dangerous because it unconditionally resets the > > DIGI_CONVERT verb for each widget. > > Agreed. We could also test the NID before switching on and off DigEn > in the DIGI_CONVERT. > > OTOH the switch would occur for *all* HDA chips which may not be what > you want. I'm not 100% sure that this particular procedure is necessary for all codecs. So, I prefer having it in the hda-codec part rather in hda-intel part. > > The below is my modified version. > > It doesn't turn off before the controller side reset as you did. So, > > if my version doesn't work, it implies that the SPDIF reset has to be > > done before the controller reset. Please give it a try. > > It works also ! I've tested AC3 and 44KHz. I cannot test 96KHz or > 196Khz (my amp is too old). Good to hear. > > + /* turn off SPDIF once; otherwise the IEC958 bits won't be updated */ > > I had more the impression that the SPDIF stream is messed up if you > change the stream format while the digital converter is working... > > > > + if (codec->spdif_ctls & 1) > > Shouldn't you use AC_DIG1_ENABLE for a proper patch ? > > if (codec->spdif_ctls & AC_DIG1_ENABLE) > > > + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_DIGI_CONVERT_1, > > + codec->spdif_ctls & 0xfe); > > likewise: codec->spdif_ctls & 0xff & ~AC_DIG1_ENABLE ? > > > + snd_hda_codec_setup_stream(codec, nid, stream_tag, 0, format); > > + /* turn on again (if needed) */ > > + if (codec->spdif_ctls & 1) > > if (codec->spdif_ctls & AC_DIG1_ENABLE) ? Yeah, that'd be better. I'll fix and commit to the HG tree later. thanks, Takashi

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[alsa-devel] [PATCH 1/3] ASoC AT91xxxx - SSC port DSP support
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch series by Frank Madarino updates the AT91xxxx core to add DSP/PCM audio hardware formats. Changes:- o Rename at19-i2s.c -> at91-ssc.c o Rename at91-i2s.h -> at91-ssc.h o Add DSP hardware formats. o Rename various I2S labels to SSC Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 2/2] ASoC WM8753 codec - build changes
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch adds the WM8753 codec driver to the kernel build system. Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 3/3] ASoC AT91xxxx build
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch by Frank Madarino updates the AT91xxxx Makefile and Kconfig to build the renamed SSC files. Changes:- o Rename various i2s labels to ssc Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH 2/3] ASoC AT91xxxx eti B1 machine SSC changes
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch by Frank Madarino updates the eti B1 machine to use the newer AT91xxxx SSC core with the DSP/PCM audio hardware changes. Changes:- o #include "at91-ssc.h" instead of "at91-i2s.h" o Rename various I2S labels to SSC Signed-off-by: Frank Mandarino <fmandarino(a)endrelia.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] ASoC DAPM switching for reentrant codec paths.
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch fixes an issue whereby power was applied to any inactive analog path that would leave and reenter a codec (e.g. ACOP -> ACIN on WM8753). This change now checks for such paths and DAPM will power them down when not in use. Signed-off-by: Seth Forshee <seth.forshee(a)gmail.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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[alsa-devel] [PATCH] ASoC WM9712 kmemdup
by Liam Girdwood 16 Apr '07

16 Apr '07

This patch creates the WM9712 codec register cache using kmemdup instead of doing a kzalloc followed by a memcpy. Signed-off-by: Graeme Gregory <gg(a)opensource.wolfsonmicro.com> Signed-off-by: Liam Girdwood <lg(a)opensource.wolfsonmicro.com>

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Re: [alsa-devel] 2.6.21-rc6, ASUS M2V 1.01G, ALSA detection/non-support
by Richard A Nelson 16 Apr '07

16 Apr '07

On Sun, 15 Apr 2007, Jiri Slaby wrote: > Date: Sun, 15 Apr 2007 23:45:29 +0200 > From: Jiri Slaby <jirislaby(a)gmail.com> > To: Richard A Nelson <cowboy(a)cavein.org> > Cc: linux-kernel <linux-kernel(a)vger.kernel.org> > Subject: Re: 2.6.21-rc6, ASUS M2V 1.01G, ALSA detection/non-support > > Cc: alsa-devel(a)alsa-project.org > > Richard A Nelson napsal(a): >> My son has an earlier version of the M2V & BIOS (0901) and his gets full >> 6ch >> sound. >> >> The 1.01G, and related BIOS (1304 or 1701) do not properly handle the >> onboard sound: >> hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from >> BIOS... > > What does your /proc/asound/version say? Both machines say: Advanced Linux Sound Architecture Driver Version 1.0.14rc3 (Wed Mar 14 07:25:50 2007 UTC). The machines are running Debian Sid -- Rick Nelson Life'll kill ya -- Warren Zevon Then you'll be dead -- Life'll kill ya

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Re: [alsa-devel] 2.6.21-rc6, ASUS M2V 1.01G, ALSA detection/non-support
by Jiri Slaby 16 Apr '07

16 Apr '07

Really Cc: alsa-devel(a)alsa-project.org Richard A Nelson napsal(a): > My son has an earlier version of the M2V & BIOS (0901) and his gets full > 6ch > sound. > > The 1.01G, and related BIOS (1304 or 1701) do not properly handle the > onboard sound: > hda_codec: Unknown model for ALC660VD/ALC861VD, trying auto-probe from > BIOS... What does your /proc/asound/version say? > it seems to truely be a BIOS issue, as I don't see a difference in lspci > -vvxxx between our systems: > > 80:01.0 Audio device: VIA Technologies, Inc. VIA High Definition Audio > Controller (rev 10) > Subsystem: ASUSTeK Computer Inc. Unknown device 81e7 > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- > ParErr- Step ping- 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 17 > Region 0: Memory at fbffc000 (64-bit, non-prefetchable) [size=16K] > Capabilities: [50] Power Management version 2 > Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA > PME(D0+,D1-,D2-,D3ho t+,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 > 00: 06 11 88 32 06 00 10 00 10 00 03 04 10 00 00 00 > 10: 04 c0 ff fb 00 00 00 00 00 00 00 00 00 00 00 00 > 20: 00 00 00 00 00 00 00 00 00 00 00 00 43 10 e7 81 > 30: 00 00 00 00 50 00 00 00 00 00 00 00 0a 01 00 00 > 40: 00 30 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > 50: 01 60 42 c8 00 00 00 00 00 00 00 00 00 00 00 00 > 60: 05 70 80 00 00 00 00 00 00 00 00 00 00 00 00 00 > 70: 10 00 91 00 00 00 00 00 00 00 30 00 00 00 00 00 > 80: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > 90: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > a0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > b0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > c0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > d0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > e0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > f0: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 > > ALSA, instead of seeing ALC660, on my box sees ALC660-VD, with very > little in the way of devices (only front output) - which does not work > either :( > > Stumped on where to go from here... regards, -- http://www.fi.muni.cz/~xslaby/ Jiri Slaby faculty of informatics, masaryk university, brno, cz e-mail: jirislaby gmail com, gpg pubkey fingerprint: B674 9967 0407 CE62 ACC8 22A0 32CC 55C3 39D4 7A7E

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[alsa-devel] HDA Intel - Conexant CX20549 loaded but not working
by S Boucher 15 Apr '07

15 Apr '07

Using alsa-driver-1.0.14rc3 I have the following problems: - cannot record from internal mic (of an hp dv6245) - spdif not working - using jack I get lots of xruns (I don't get any xrun on the exact same system if I use an M-Audio Transit, so I'm assuming something ain't kosher in the conexant driver) - I was able to record cleanly from ext mic (although I had to select int mic) I'm with Ubuntu Feisty. N E Waze, if I can be of assistance let me know soon. The nice HP DV6245 laptop is heading back to the store in 1-2 days (nice system, except I don't have time to learn sign language :-)). __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[alsa-devel] Kernel crash with 2.6.20.x : snd-usb-audio implicated...
by Chris Rankin 15 Apr '07

15 Apr '07

Hi, The 2.6.20.x kernels all crash on my SMP-HT machine, and the oops report always shows that udev was trying to access an endpoint belonging to the snd-usb-audio driver. The frustrating part is that this particular endpoint shouldn't even exist! (It doesn't exist whenever I check the /sys/class/usb_endpoint directory.) Is there any way that userspace ("winecfg" in my case) can cause the snd-usb-audio driver to switch from (If# = 2, Alt = 0) to (If# = 2, Alt > 0) please? Because that is currently the only way that I can see this endpoint ever being created. Thanks, Chris --- bugme-daemon(a)bugzilla.kernel.org wrote: > Date: Sat, 14 Apr 2007 05:22:23 -0700 > From: bugme-daemon(a)bugzilla.kernel.org > To: rankincj(a)yahoo.com > Subject: [Bug 8198] udevd triggers oops when walking through /sys > > http://bugzilla.kernel.org/show_bug.cgi?id=8198 > > ------- Additional Comments From rankincj(a)yahoo.com 2007-04-14 05:22 ------- > Created an attachment (id=11144) > --> (http://bugzilla.kernel.org/attachment.cgi?id=11144&action=view) > dmesg output from 2.6.20.7 oops > > I don't know if this output is any more useful than any of the others. And I > can't see where the "single bit error" comes from when the output is full of > "use after free" poison. However, this much at least is different: > > EIP: [<6b6b6b6b>] 0x6b6b6b6b SS:ESP 0068:ec9eaf48 > <3>Slab corruption: start=eceea92c, len=512 > Redzone: 0x5a2cf071/0x5a2cf071. > Last user: [<c0226642>](kfree_skbmem+0xb/0x9c) > 080: 6b 6b 6b 6b 6a 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > Single bit error detected. Probably bad RAM. > Run memtest86+ or a similar memory test tool. > Prev obj: start=eceea720, len=512 > Redzone: 0x170fc2a5/0x170fc2a5. > Last user: [<f886eda5>](usb_create_ep_files+0xf1/0x2c2 [usbcore]) > 000: f8 fd 1b c2 e0 ee 4e f7 01 00 00 00 00 00 00 00 > 010: 30 a7 ee ec 30 a7 ee ec f4 91 1e c0 96 90 1e c0 > Next obj: start=eceeab38, len=512 > Redzone: 0x5a2cf071/0x5a2cf071. > Last user: [<c0226642>](kfree_skbmem+0xb/0x9c) > 000: 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > 010: 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b 6b > > Again, it's claiming that the last sysfs file accessed was the non-existent > /class/usb_endpoint/usbdev1.3_ep84/dev, which I simply cannot explain because > this endpoint belongs to If# = 2, Alt > 0: > > I: If#= 2 Alt= 0 #EPs= 0 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > I: If#= 2 Alt= 1 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 90 Ivl=1ms > I: If#= 2 Alt= 2 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 46 Ivl=1ms > I: If#= 2 Alt= 3 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 25 Ivl=1ms > I: If#= 2 Alt= 4 #EPs= 1 Cls=01(audio) Sub=02 Prot=00 Driver=snd-usb-audio > E: Ad=84(I) Atr=01(Isoc) MxPS= 16 Ivl=1ms > > Could the ALSA USB audio driver be changing the "Alt" on the fly somehow, > causing usbcore to need to create the extra endpoint in /sys? ___________________________________________________________ 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] [2/3] 2.6.21-rc6: known regressions
by Adrian Bunk 14 Apr '07

14 Apr '07

This email lists some known regressions in Linus' tree compared to 2.6.20. If you find your name in the Cc header, you are either submitter of one of the bugs, maintainer of an affectected subsystem or driver, a patch of you caused a breakage or I'm considering you in any other way possibly involved with one or more of these issues. Due to the huge amount of recipients, please trim the Cc when answering. Subject : snd_hda_intel doesn't work with ASUS M2V mainboard References : http://bugzilla.kernel.org/show_bug.cgi?id=8273 Submitter : Hans-Georg Rist <hg.rist(a)web.de> Status : unknown Subject : snd_intel8x0: divide error: 0000 References : http://lkml.org/lkml/2007/3/5/252 Submitter : Michal Piotrowski <michal.k.k.piotrowski(a)gmail.com> Status : unknown Subject : hal daemon crashes after pulling a USB serial device References : http://www.opensubscriber.com/message/linux-usb-devel@lists.sourceforge.net… Submitter : Andi Kleen <ak(a)suse.de> Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debugged Subject : USB: iPod doesn't work (CONFIG_USB_SUSPEND) References : http://lkml.org/lkml/2007/3/21/320 Submitter : Tino Keitel <tino.keitel(a)gmx.de> Caused-By : Marcelo Tosatti <marcelo(a)kvack.org> commit 1d619f128ba911cd3e6d6ad3475f146eb92f5c27 Handled-By : Oliver Neukum <oneukum(a)suse.de> Status : problem is being debuggged Subject : USB: Oops when changing DVB-T adapter References : http://lkml.org/lkml/2007/3/9/212 Submitter : CIJOML <cijoml(a)volny.cz> Handled-By : Markus Rechberger <markus.rechberger(a)amd.com> Patch : http://lkml.org/lkml/2007/4/5/154 Status : patches available

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[alsa-devel] function location?
by Ashlesha Shintre 13 Apr '07

13 Apr '07

Hi, I am trying to write an application to just play a wave file and record sound using ALSA -- I looked at the pcm_min.c file and tried to run it - I am using UBUNTU and 1.0.11-7ubuntu3 sound library called libasound2. I cant seem to find the pcm.c file which has all the snd_pcm_open and other functions used in the pcm_min.c file! I am using : /usr/include/alsa/asoundlib.h header -- but i m not sure where the functions defined in this header are -- and, find / -name pcm.c yields nothing --nor does find / -name snd_pcm_open Any pointers please! Thanks, Ashlesha.

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[alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 13 Apr '07

13 Apr '07

Dear Alsa developers, I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. I want to use the /etc/asound.conf (.asoundrc) to split the card (that has 8 in and 8 out - but I use just 6 in/out), with minimal latency. I mean, I want to record and play (duplex) from/to each virtual card concurrently and from separate application instances. My app works with S16_LE, 8000, mono. Is the following asound.conf (.asoundrc) correct and efficient for this? I got spikes of cpu load (at 5-6%) on each virtual card any roughly 256 seconds when using a(play|record) to record or play from/to any channel, and the spikes cumulate with the number of virtual cards concurrently in use (eg. if I use 6 virtual cards concurrently I got spikes of 30% cpu load each 256 seconds). I need to create ctls? It would behave better if I use an asym to link toghether each couple of input and output? Any hints? Thanks a lot in advance, Giovanni that is the /etc/asound.conf I'm using: pcm.capt { type dsnoop ipc_key 223456 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.capt3 { type plug slave.pcm capt ttable.0.2 1 } pcm.capt4 { #used type plug slave.pcm capt ttable.0.3 1 } pcm.capt5 { type plug slave.pcm capt ttable.0.4 1 } pcm.capt6 { type plug slave.pcm capt ttable.0.5 1 } pcm.capt7 { type plug slave.pcm capt ttable.0.6 1 } pcm.capt8 { type plug slave.pcm capt ttable.0.7 1 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 }

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[alsa-devel] 1010lt (ice1712) asoundrc for 6 virtual mono cards question
by Giovanni Maruzzelli 13 Apr '07

13 Apr '07

Dear Alsa developers, I want to use a 1010lt (ice1712 driver) as 6 analog mono duplex cards. I want to use the /etc/asound.conf (.asoundrc) to split the card (that has 8 in and 8 out - but I use just 6 in/out), with minimal latency. I mean, I want to record and play (duplex) from/to each virtual card concurrently and from separate application instances. My app works with S16_LE, 8000, mono. Is the following asound.conf (.asoundrc) correct and efficient for this? I got spikes of cpu load (at 5-6%) on each virtual card any roughly 256 seconds when using a(play|record) to record or play from/to any channel, and the spikes cumulate with the number of virtual cards concurrently in use (eg. if I use 6 virtual cards concurrently I got spikes of 30% cpu load each 256 seconds). I need to create ctls? It would behave better if I use an asym to link toghether each couple of input and output? Any hints? Thanks a lot in advance, Giovanni that is the /etc/asound.conf I'm using: pcm.capt { type dsnoop ipc_key 223456 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 12 format S32_LE } } pcm.capt3 { type plug slave.pcm capt ttable.0.2 1 } pcm.capt4 { #used type plug slave.pcm capt ttable.0.3 1 } pcm.capt5 { type plug slave.pcm capt ttable.0.4 1 } pcm.capt6 { type plug slave.pcm capt ttable.0.5 1 } pcm.capt7 { type plug slave.pcm capt ttable.0.6 1 } pcm.capt8 { type plug slave.pcm capt ttable.0.7 1 } ############################# ############################# pcm.s3 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 2 } } pcm.play3 { type plug slave.pcm s3 } pcm.s4 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 3 } } pcm.play4 { type plug slave.pcm s4 } pcm.s5 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 4 } } pcm.play5 { type plug slave.pcm s5 } pcm.s6 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 5 } } pcm.play6 { type plug slave.pcm s6 } pcm.s7 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 6 } } pcm.play7 { type plug slave.pcm s7 } pcm.s8 { type dshare ipc_key 123456789 slave { pcm "hw:0,0" rate 8000 period_time 0 period_size 320 channels 10 format S32_LE } bindings { 0 7 } } pcm.play8 { type plug slave.pcm s8 }

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[alsa-devel] Playback Devices in PREPARED State Throw Off POLLOUT Events?
by Jeff Rush 13 Apr '07

13 Apr '07

A question re normal internal operation of ALSA. I'm using the poll() interface to wait for a audio playback device to ask for more samples. The device is in the PREPARED state, not RUNNING state. At the beginning of my program, _before_ I write any samples to it, I'm getting repeated POLLOUT events, when I do not yet have data available to send. They seem to come frequent enough to load down my CPU, rather than every N ticks, so it seems. I thought a device was idle in the PREPARED state, and you had to either set the start conditions and write data, or explicitly invoke snd_pcm_start(). In my case, I wanted to use snd_pcm_start() once I had data to send, and also used the start conditions, set at 50%, to cause the audio output to them proceed to the speaker. The output snd_pcm_status_dump() right after exiting from the very first poll() ever in the program shows the following, and that I have a POLLOUT event pending. --- Status of Headphones Player --- state : PREPARED trigger_time: 0.000000 tstamp : 1176119509.658821000 delay : 0 avail : 2400 avail_max : 0 POLLOUT fired, calling ready_for_write() I suppose I could force it back to an earlier state like SND_PCM_STATE_OPEN or SND_PCM_STATE_SETUP, but that seems a bit harsh, since the snd_pcm_hw_params() takes it to the SND_PCM_STATE_PREPARED itself. Thanks for any tips, -Jeff

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 13 Apr '07

13 Apr '07

On Fri, 13 Apr 2007, Johannes Berg wrote: > On Thu, 2007-04-12 at 13:15 +0200, Rene Herman wrote: > > > > With Johannes don't know what kind of sound the chip produces using > > > OSS I'm stuck again. I have attached the samples to this e-mail. What > > > would be a reasonable debug step here? > > > > This message got stuck in the moderation queue due to being too big to > > pass through. The list limit is set at 60 kB and while that may be too > > low a limit, the 1.5 MB that was attached to this one was rather big. > > Too big for public lists. > > > > I approved it anyway, and then the attachment got scrubbed none the less > > by the list software it seems. > > I copied the files: > http://johannes.sipsolutions.net/files/stefan-noise.wav > http://johannes.sipsolutions.net/files/stefan-aplay.wav > > The interesting thing to note is that aoss and snd-pcm-oss have the > exact same effect, which is totally strange unless... hey! does aoss > maybe fall back to real oss when it is available? Point is the OSS modules aren't loaded :\ Stefan

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[alsa-devel] rawmidi timestamps
by Florian 13 Apr '07

13 Apr '07

Hi, thanks for replying the "blocking read" question - it works fine with poll(). I'm using its timeout mechanism to bail out every 200msecs of inactivity. I wondered what snd_rawmidi_status_get_tstamp() is meant to return? Briefly checking out the driver midlevel, the timestamp is just ignored and setmem'ed to 0, or did I miss anything? Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

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Re: [alsa-devel] Alsa OSS - who is maintaining it?
by Stefan de Konink 12 Apr '07

12 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA512 Hi, With Johannes don't know what kind of sound the chip produces using OSS I'm stuck again. I have attached the samples to this e-mail. What would be a reasonable debug step here? Stefan -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGHgiuYH1+F2Rqwn0RCidgAJ44R70RyxNLoWOh681UGFbP+rrVFACfROLN bVFdMfHcIP4nLUDfaqrsbbQ= =MRRf -----END PGP SIGNATURE-----

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[alsa-devel] Functioning support for Prodigy 192
by dustin＠seznam.cz 12 Apr '07

12 Apr '07

Hi Takashi, I am finishing changes to prodigy192.c to make the card functional. I modified eeprom values (e.g. 49MHz crystal), fixed digital output of the card (requires GPIO), added support for MI/ODI/O and added input switch mic/line. So far I have not done any MIDI tests. Before posting corresponding patches I would like to resolve a few remaining issues. The ak4114 based miodio card is a separately sold add-on and provides SPDIF out (directly connected to SPDIFOUT pin of ice1724), SPDIF IN (TOSLINK to input 0, coax to input 1 of ak4114) + MIDI IN/OUT. SPDIF OUT required no code, SPDIF IN is fully working with my last patches, switching between toslink/coax works in my prodigy192.c. Windows drivers detect miodio automatically, offering corresponding controls when connected. What would be a proper way in the driver to activate the miodio support? There is one Xilinx FPGA chip onboard, perhaps it is used for MIDI (although ice1724 has a built-in support, right?), perhaps it informs original windows drivers about version and capabilities of the addon card. Another option would be to test communication with ak4114, whether we can read contents of a particular register. Perhaps a proper place to do so would be in snd_ak4114_create of ak4114.c which could return error when no communication could be established with ak4114 (I do not think this is the case now). What solution would you suggest? Another question concerns capture input switches, both analog and digital. For now I have used a single switch SNDRV_CTL_ELEM_TYPE_BOOLEAN for each. On the capture tab of alsamixer that is not intuitive (capture off = mice, capture on = line etc.). I would prefer standard "capture input radio buttons" like e.g. in ac97 codec (ich5 + AD1985). I would need two independent groups of two "radio buttons". Please could you point me to some self-explaining code implementing similar feature in another driver which a newbie could understand? Thanks a lot. Pavel.

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[alsa-devel] ALSA Wiki
by Jaroslav Kysela 12 Apr '07

12 Apr '07

Hello all, new Wiki running well known MediaWiki (wikipedia) which will replace ALSA WWW contents is prepared at URL http://bugtrack.alsa-project.org/main/ . I would like ask possible volunteers to start moving the current contents of ALSA WWW server to this Wiki. Also, if you like to be a WikiSysop to change at least some Wiki layout - you have a chance now. Also, we need sysops to rollback wrong changes. The edit policy is simple. You have to be logged in and you MUST acknowledge your e-mail address. I'm sorry, but we cannot take contents from anonymous users. Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Unblock MIDI IN in blocking read mode
by Florian 12 Apr '07

12 Apr '07

Hi, for best performance, I use a blocking snd_rawmidi_read() on an ALSA MIDI input device, byte by byte. It works fine, except: 1) How do I close the device from a different thread? I.e. how can I unblock the blocking read() call when it's time to close the device and no MIDI data is currently coming in? Just calling snd_rawmidi_close() will cause a crash. Using snd_rawmidi_drop() did not release the blocking read() in the other thread. 2) How is a blocking read() typically implemented in the ALSA MIDI driver? Maybe I can resort to non-blocking mode and avoid the situation above. Thanks, Florian -- Florian Bomers bome.com ------------------------------------------------------- Music Software, Development Tools: http://www.bome.com Java Sound extensions, plugins: http://www.tritonus.org The Java Sound Resources: http://www.jsresources.org ------------------------------------------------------- Please quote this email in your reply. Thanks!

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[alsa-devel] [PATCH] ALSA: Support Media Vision Jazz16 chips
by Rask Ingemann Lambertsen 11 Apr '07

11 Apr '07

From: Rask Ingemann Lambertsen <rask(a)sygehus.dk> - Add support for Jazz16 features to the common SoundBlaster code. - Add a front end for Jazz16 based sound cards. Tested with a Jazz16 based sound card and Linux 2.6.20. Signed-off-by: Rask Ingemann Lambertsen <rask(a)sygehus.dk> --- This patch is a prerequisite for the Jazz PnP patch, which needs the patched include/sound/sb.h. Documentation/sound/alsa/ALSA-Configuration.txt | 10 include/sound/sb.h | 5 sound/isa/Kconfig | 15 + sound/isa/sb/Makefile | 2 sound/isa/sb/jazz16.c | 193 ++++++++++++++ sound/isa/sb/sb8_main.c | 105 +++++++ sound/isa/sb/sb_common.c | 12 sound/isa/sb/sb_mixer.c | 3 8 files changed, 335 insertions(+), 10 deletions(-) --- linux-2.6.20/sound/isa/sb/Makefile.orig 2007-02-19 21:00:09.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/Makefile 2007-03-02 20:14:48.000000000 +0100 @@ -12,6 +12,7 @@ snd-sb16-objs := sb16.o snd-sbawe-objs := sbawe.o emu8000.o snd-emu8000-synth-objs := emu8000_synth.o emu8000_callback.o emu8000_patch.o emu8000_pcm.o snd-es968-objs := es968.o +snd-jazz16-objs := jazz16.o # # this function returns: @@ -30,6 +31,7 @@ obj-$(CONFIG_SND_SB16) += snd-sb16.o snd obj-$(CONFIG_SND_SBAWE) += snd-sbawe.o snd-sb16-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ES968) += snd-es968.o snd-sb8-dsp.o snd-sb-common.o obj-$(CONFIG_SND_ALS4000) += snd-sb-common.o +obj-$(CONFIG_SND_JAZZ16) += snd-jazz16.o snd-sb8-dsp.o snd-sb-common.o ifeq ($(CONFIG_SND_SB16_CSP),y) obj-$(CONFIG_SND_SB16) += snd-sb16-csp.o obj-$(CONFIG_SND_SBAWE) += snd-sb16-csp.o --- linux-2.6.20/sound/isa/sb/jazz16.c-orig 2007-02-17 20:01:40.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/jazz16.c 2007-03-18 13:43:35.000000000 +0100 @@ -0,0 +1,193 @@ +/* Driver for Media Vision Jazz16 based boards. + * + * Copyright (c) 2007 by Rask Ingemann Lambertsen + * + * 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 Jazz16 is an SB Pro compatible chip with a 16-bit mode and higher + * playback and capture rates added to it. It is not SB 16 compatible. + * There is also an MPU-401 interface. + * + * The IBM PPS Model 6015 has a Jazz16 chip on board. Please tell me if it + * works with this driver or not. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/kernel.h> +#include <linux/errno.h> +#include <linux/slab.h> +#include <linux/init.h> +#include <linux/pnp.h> +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/delay.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/sb.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> +#include <sound/initval.h> + +MODULE_AUTHOR ("Rask Ingemann Lambertsen <rask(a)sygehus.dk>"); +MODULE_DESCRIPTION ("Jazz16"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable switches */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for Jazz16 soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for Jazz16 soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable Jazz16 soundcard."); + +#define PFX "jazz16: " +#define PFX_WARN KERN_WARNING PFX +#define PFX_ERR KERN_ERR PFX + +static struct pnp_driver snd_jazz16_pnp_driver; + +static irqreturn_t jazz16_interrupt(int irq, void *dev_id) +{ + struct snd_sb *chip = (struct snd_sb *) dev_id; + return snd_sb8dsp_interrupt(chip); +} + +static struct pnp_device_id snd_jazz16_pnpids[] = { + { .id = "PNPb00f" }, + { .id = "" } +}; +MODULE_DEVICE_TABLE(pnp, snd_jazz16_pnpids); + +static int __devinit snd_jazz16_pnp_probe(struct pnp_dev *pnp_dev, + const struct pnp_device_id *pnp_id) +{ struct snd_sb *chip; + struct snd_card *card; + struct snd_opl3 *opl3; + int err; + uint sbport, sbirq, sbdma8, sbdma16, mpuport, mpuirq; + static uint dev_num = 0; + + if (enable[dev_num]) + card = snd_card_new(index[dev_num], id[dev_num], THIS_MODULE, 0); + else + card = NULL; + dev_num ++; + if (card == NULL) + return -ENOMEM; + pnp_set_drvdata(pnp_dev, card); + /* TODO use pnp_port_valid(), pnp_port_flags(), pnp_port_length()... */ + sbport = pnp_port_start(pnp_dev, 0); + sbirq = pnp_irq(pnp_dev, 0); + sbdma8 = pnp_dma(pnp_dev, 0); + sbdma16 = pnp_dma(pnp_dev, 1); + err = snd_sbdsp_create(card, sbport, sbirq, jazz16_interrupt, + sbdma8, sbdma16, SB_HW_AUTO, &chip); + if (err < 0) { + snd_card_free(card); + return err; + } + /* TODO length limits - strncpy()/snprintf() and friends. */ + strcpy(card->driver, "Jazz16"); + strcpy(card->shortname, "Media Vision Jazz16"); + sprintf(card->longname, "%s at 0x%x, irq %u, dma8 %u, dma16 %u", + chip->name, sbport, sbirq, sbdma8, sbdma16); + + if (chip->hardware != SB_HW_JAZZ16) { + snd_printk(PFX_ERR "Not a Jazz16 chip at 0x%x.\n", sbport); + snd_card_free(card); + return -ENODEV; + } + err = snd_sb8dsp_pcm(chip, 0, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_sbmixer_new(chip); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_opl3_create(card, sbport, sbport + 2, + OPL3_HW_AUTO, 1, &opl3); + if (err < 0) { + snd_printk(PFX_WARN "No OPL device found, skipping.\n"); + } else { + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) { + snd_card_free(card); + return err; + } + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) { + snd_card_free(card); + return err; + } + } + if (pnp_port_valid(pnp_dev, 1) + && !(pnp_port_flags(pnp_dev, 1) & IORESOURCE_DISABLED)) { + int mpuirqflags; + if (!pnp_irq_valid(pnp_dev, 1) + || pnp_irq_flags(pnp_dev, 1) & IORESOURCE_DISABLED) { + mpuirq = -1; + mpuirqflags = 0; + } else { + mpuirq = pnp_irq(pnp_dev, 1); + mpuirqflags = IRQF_DISABLED; + } + mpuport = pnp_port_start(pnp_dev, 1); + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpuport, 0, mpuirq, + mpuirqflags, NULL); + if (err < 0) { + snd_printk(PFX_WARN "MPU-401 device not configured, skipping.\n"); + } + } + err = snd_card_register(card); + if (err < 0) { + snd_card_free(card); + return err; + } + return 0; +} + +static void __devexit snd_jazz16_pnp_remove(struct pnp_dev *dev) +{ + snd_card_free(pnp_get_drvdata(dev)); + pnp_set_drvdata(dev, NULL); +} + +static struct pnp_driver snd_jazz16_pnp_driver = { + .name = "Jazz16", + .id_table = snd_jazz16_pnpids, + .probe = snd_jazz16_pnp_probe, + .remove = __devexit_p(snd_jazz16_pnp_remove), +}; + +static int __devinit alsa_card_jazz16_init (void) +{ + return pnp_register_driver(&snd_jazz16_pnp_driver); +} + +static void __devexit alsa_card_jazz16_exit(void) +{ + pnp_unregister_driver(&snd_jazz16_pnp_driver); +} + +module_init (alsa_card_jazz16_init); +module_exit (alsa_card_jazz16_exit); --- linux-2.6.20/sound/isa/sb/sb_mixer.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb_mixer.c 2007-02-25 01:09:05.000000000 +0100 @@ -811,6 +811,7 @@ int snd_sbmixer_new(struct snd_sb *chip) return err; break; case SB_HW_PRO: + case SB_HW_JAZZ16: if ((err = snd_sbmixer_init(chip, snd_sbpro_controls, ARRAY_SIZE(snd_sbpro_controls), @@ -946,6 +947,7 @@ void snd_sbmixer_suspend(struct snd_sb * save_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: save_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: @@ -971,6 +973,7 @@ void snd_sbmixer_resume(struct snd_sb *c restore_mixer(chip, sb20_saved_regs, ARRAY_SIZE(sb20_saved_regs)); break; case SB_HW_PRO: + case SB_HW_JAZZ16: restore_mixer(chip, sbpro_saved_regs, ARRAY_SIZE(sbpro_saved_regs)); break; case SB_HW_16: --- linux-2.6.20/sound/isa/sb/sb_common.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb_common.c 2007-03-03 12:50:32.000000000 +0100 @@ -146,8 +146,16 @@ static int snd_sbdsp_probe(struct snd_sb } break; case 3: - chip->hardware = SB_HW_PRO; - str = "Pro"; + spin_lock_irqsave(&chip->reg_lock, flags); + if (snd_sbdsp_command(chip, SB_DSP_GET_JAZZ_VERSION) && + SB_VERSION_IS_JAZZ16(snd_sbdsp_get_byte (chip))) { + chip->hardware = SB_HW_JAZZ16; + str = "Pro (Jazz16)"; + } else { + chip->hardware = SB_HW_PRO; + str = "Pro"; + } + spin_unlock_irqrestore (&chip->reg_lock, flags); break; case 4: chip->hardware = SB_HW_16; --- linux-2.6.20/sound/isa/sb/sb8_main.c.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/sb/sb8_main.c 2007-03-24 23:30:22.000000000 +0100 @@ -28,6 +28,9 @@ * * Wed Jul 12 22:02:55 CEST 2000 Uros Bizjak <uros(a)kss-loka.si> * Cleaned up and rewrote lowlevel routines. + * + * Sat Mar 24 23:29:20 CEST 2007 Rask Ingemann Lambertsen <rask(a)sygehus.dk> + * Added Jazz16 enhancements (ported from sound/oss/sb_common.c). */ #include <sound/driver.h> @@ -73,6 +76,15 @@ static struct snd_ratnum stereo_clocks[] } }; +/* For stereo playback and capture, the denominator is divided by two, so it + * must be even to get the intended sample rate. */ +static struct snd_ratnum jazz16_stereo_clock = { + .num = SB8_CLOCK, + .den_min = 2, + .den_max = 512, + .den_step = 2, +}; + static int snd_sb8_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) { @@ -101,14 +113,53 @@ static int snd_sb8_hw_constraint_channel return 0; } +static int snd_jazz16_hw_constraint_rate_channels(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + if (c->max > 1) { + unsigned int num = 0, den = 0; + err = snd_interval_ratnum(r, 1, &jazz16_stereo_clock, &num, &den); + if (err >= 0 && den) { + params->rate_num = num; + params->rate_den = den; + } + } + return err; +} + +static int snd_jazz16_hw_constraint_channels_rate(struct snd_pcm_hw_params *params, + struct snd_pcm_hw_rule *rule) +{ + struct snd_interval *r = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + int err = 0; + + /* Force mono mode if the sample rate interval doesn't allow stereo. */ + if (SB8_DEN(r->min) == SB8_DEN(r->max) + && SB8_DEN(r->min) & 1) + { + struct snd_interval t = { .min = 1, .max = 1 }; + struct snd_interval *c; + + c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + err = snd_interval_refine(c, &t); + } + return err; +} + static int snd_sb8_playback_prepare(struct snd_pcm_substream *substream) { unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -117,6 +168,7 @@ static int snd_sb8_playback_prepare(stru break; } /* fallthru */ + case SB_HW_JAZZ16: case SB_HW_201: if (rate > 23000) { chip->playback_format = SB_DSP_HI_OUTPUT_AUTO; @@ -134,8 +186,17 @@ static int snd_sb8_playback_prepare(stru } size = chip->p_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->p_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + } else { + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_ON); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); if (runtime->channels > 1) { /* set playback stereo mode */ spin_lock(&chip->mixer_lock); @@ -173,7 +234,7 @@ static int snd_sb8_playback_prepare(stru snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_WRITE | DMA_AUTOINIT); return 0; } @@ -232,9 +293,11 @@ static int snd_sb8_capture_prepare(struc unsigned long flags; struct snd_sb *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int mixreg, rate, size, count; + unsigned int mixreg, rate, size, count, dma; + unsigned char stereo, format; rate = runtime->rate; + stereo = runtime->channels > 1; switch (chip->hardware) { case SB_HW_PRO: if (runtime->channels > 1) { @@ -242,6 +305,8 @@ static int snd_sb8_capture_prepare(struc chip->capture_format = SB_DSP_HI_INPUT_AUTO; break; } + /* fall through */ + case SB_HW_JAZZ16: chip->capture_format = (rate > 23000) ? SB_DSP_HI_INPUT_AUTO : SB_DSP_LO_INPUT_AUTO; break; case SB_HW_201: @@ -261,10 +326,17 @@ static int snd_sb8_capture_prepare(struc } size = chip->c_dma_size = snd_pcm_lib_buffer_bytes(substream); count = chip->c_period_size = snd_pcm_lib_period_bytes(substream); + if (runtime->format == SNDRV_PCM_FORMAT_S16_LE) { + dma = chip->dma16; + format = stereo ? SB_DSP_STEREO_16BIT : SB_DSP_MONO_16BIT; + } else { + dma = chip->dma8; + format = stereo ? SB_DSP_STEREO_8BIT : SB_DSP_MONO_8BIT; + } spin_lock_irqsave(&chip->reg_lock, flags); snd_sbdsp_command(chip, SB_DSP_SPEAKER_OFF); - if (runtime->channels > 1) - snd_sbdsp_command(chip, SB_DSP_STEREO_8BIT); + if (chip->hardware >= SB_HW_PRO) + snd_sbdsp_command(chip, format); snd_sbdsp_command(chip, SB_DSP_SAMPLE_RATE); if (runtime->channels > 1) { snd_sbdsp_command(chip, 256 - runtime->rate_den / 2); @@ -285,7 +357,7 @@ static int snd_sb8_capture_prepare(struc snd_sbdsp_command(chip, count >> 8); } spin_unlock_irqrestore(&chip->reg_lock, flags); - snd_dma_program(chip->dma8, runtime->dma_addr, + snd_dma_program(dma, runtime->dma_addr, size, DMA_MODE_READ | DMA_AUTOINIT); return 0; } @@ -358,10 +430,12 @@ static snd_pcm_uframes_t snd_sb8_playbac { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_PLAYBACK_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->p_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->p_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -369,10 +443,12 @@ static snd_pcm_uframes_t snd_sb8_capture { struct snd_sb *chip = snd_pcm_substream_chip(substream); size_t ptr; + int dma; if (chip->mode != SB_MODE_CAPTURE_8) return 0; - ptr = snd_dma_pointer(chip->dma8, chip->c_dma_size); + dma = (substream->runtime->format == SNDRV_PCM_FORMAT_S16_LE) ? chip->dma16 : chip->dma8; + ptr = snd_dma_pointer(dma, chip->c_dma_size); return bytes_to_frames(substream->runtime, ptr); } @@ -443,6 +519,19 @@ static int snd_sb8_open(struct snd_pcm_s runtime->hw = snd_sb8_capture; } switch (chip->hardware) { + case SB_HW_JAZZ16: + runtime->hw.formats |= SNDRV_PCM_FMTBIT_S16_LE; + runtime->hw.rates |= SNDRV_PCM_RATE_8000_44100; + runtime->hw.rate_max = 45455; + runtime->hw.channels_max = 2; + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + snd_jazz16_hw_constraint_rate_channels, NULL, + SNDRV_PCM_HW_PARAM_CHANNELS, + SNDRV_PCM_HW_PARAM_RATE, -1); + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + snd_jazz16_hw_constraint_channels_rate, NULL, + SNDRV_PCM_HW_PARAM_RATE, -1); + break; case SB_HW_PRO: runtime->hw.rate_max = 44100; runtime->hw.channels_max = 2; --- linux-2.6.20/sound/isa/Kconfig.orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/sound/isa/Kconfig 2007-03-25 21:49:44.000000000 +0200 @@ -248,6 +248,21 @@ config SND_INTERWAVE_STB To compile this driver as a module, choose M here: the module will be called snd-interwave-stb. +config SND_JAZZ16 + tristate "Media Vision Jazz16" + depends on SND + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Media Vision Jazz16 based + soundcards. + + You probably want to enable Jazz16 Plug and Play support + (CONFIG_JAZZ16PNP) also. + + To compile this driver as a module, choose M here: the module + will be called snd-jazz16. + config SND_OPL3SA2 tristate "Yamaha OPL3-SA2/SA3" depends on SND --- linux-2.6.20/Documentation/sound/alsa/ALSA-Configuration.txt-orig 2007-02-04 19:44:54.000000000 +0100 +++ linux-2.6.20/Documentation/sound/alsa/ALSA-Configuration.txt 2007-03-25 20:35:31.000000000 +0200 @@ -1120,6 +1120,16 @@ Prior to version 0.9.0rc4 options had a This module supports multiple cards, autoprobe and ISA PnP. + Module snd-jazz16 + ----------------- + + Module for sound cards based on Media Vision Jazz16 chip (PnP only) + + This module supports multiple cards and PnP. + + Note: You will most likely want to enable Jazz16 PnP support + (CONFIG_JAZZ16PNP) in the kernel. + Module snd-korg1212 ------------------- --- linux-2.6.20/include/sound/sb.h.orig 2007-02-19 20:57:51.000000000 +0100 +++ linux-2.6.20/include/sound/sb.h 2007-03-25 21:37:01.000000000 +0200 @@ -33,6 +33,7 @@ enum sb_hw_type { SB_HW_20, SB_HW_201, SB_HW_PRO, + SB_HW_JAZZ16, /* Media Vision Jazz16 chip */ SB_HW_16, SB_HW_16CSP, /* SB16 with CSP chip */ SB_HW_ALS100, /* Avance Logic ALS100 chip */ @@ -142,6 +143,7 @@ struct snd_sb { #define SB_DSP_HI_INPUT_AUTO 0x98 #define SB_DSP_IMMED_INT 0xf2 #define SB_DSP_GET_VERSION 0xe1 +#define SB_DSP_GET_JAZZ_VERSION 0xfa #define SB_DSP_SPEAKER_ON 0xd1 #define SB_DSP_SPEAKER_OFF 0xd3 #define SB_DSP_DMA8_OFF 0xd0 @@ -265,6 +267,9 @@ struct snd_sb { #define SB_DMASETUP_DMA6 0x40 #define SB_DMASETUP_DMA7 0x80 +/* Check the output of SB_DSP_GET_JAZZ_VERSION. */ +#define SB_VERSION_IS_JAZZ16(x) ((x) == 0x12) + /* * */ -- Rask Ingemann Lambertsen

3 15

[alsa-devel] snd_mixer_selem_channel_id -- should i be using this? if not, what should i use?
by John L. Utz III 11 Apr '07

11 Apr '07

Hello Everybody; mixer.h indicates that snd_mixer_selem_channel_id is a legacy api. so, what is the non legacy api for setting volumes on individual channels, ie Master L&R, PCM L&R and Surround L&R ? tnx! johnu

1 0

[alsa-devel] Interrupts and kmalloc in alsa modules
by Ciaccia 11 Apr '07

11 Apr '07

Hi all, I am working with an ARM embedded system and I porting an alsa application to that system. Cirrus (the manufacturer of the board) wrote a generic alsa driver for "their" kernel (a kernel based on the 2.6.8 with some patches) and I ported the module to the vanilla kernel 2.6.19. The module works fine (play and capture both work), but the function that allocates the memory prints a strange message in the kernel log when the buffers are allocated (the module calls kmalloc, which prints "sleeping function called from invalid context" to the log). The buffers are allocated when the streams are started, not when the module is loaded, I think this is standard... The buffers are allocated with the following line: s->buffers = (audio_buf_t *) kmalloc(sizeof(audio_buf_t) * s->nbfrags, GFP_KERNEL); When the line here above is executed, slab.c prints the error the kernel log. The error comes from the "might_sleep_if" line: static inline void cache_alloc_debugcheck_before(struct kmem_cache *cachep, gfp_t flags) { might_sleep_if(flags & __GFP_WAIT); #if DEBUG kmem_flagcheck(cachep, flags); #endif } Playing with the module, I discovered that if I enable the interrupts before calling kmalloc, the error disappears. I wonder if this is correct, or if something is wrong... Could someone of you explain to me why I have to enable the interrupts before calling kmalloc? The strange thing is that the module never explicitly turns them off (I attached the file to the mail) Any help is greatly appreciated ;-) Bye Andrea ____________________________________________________________________________________ Food fight? Enjoy some healthy debate in the Yahoo! Answers Food & Drink Q&A. http://answers.yahoo.com/dir/?link=list&sid=396545367

3 5

[alsa-devel] sound on asus p5ld2-se
by Andrius 10 Apr '07

10 Apr '07

Hi, It says "If the default configuration doesn't work and one of the above matches with your device, report it together with the PCI subsystem ID (output of "lspci -nv") to ALSA BTS or alsa-devel ML" in ALSA-configuration.txt so, here it is: i'm using asus p5ld2-se mobo with intel ich7 chipset, and just have installed new debian stable (4.0). And encountered a problem: onboard audio sound from one of the speakers was just static noise, and if i adjusted volume or anything in alsamixer, the sound kept dissappearing. I tried module=3stack option in modprobe, added line options snd-hda-intel index=0 model=3stack in /etc/modprobe.d/sound and everything is ok now. i attached lspci -nv output to this letter p.s. on ubuntu live cd and other distros i've tried there were same problem

2 1

[alsa-devel] Driver for user-space sound source/sink(s)
by Alex Makarov 10 Apr '07

10 Apr '07

Hi We're considering implementing the driver for user-space sound input/output. For example: "Line in" is provided by some application by streaming data to a file in /dev/..., "Line out" can be read by any application from /dev/... It would be also nice to have arbitrary number of streams, for example to have different applications to "capture" audio from different sources. I wonder if we reinvent the wheel, or maybe there's an ongoing project for the similar driver. If not, could anyone advise how this can be achieved. We had no experience with ALSA before, but pretty strong in C/C++ development. Thanks! Alex

1 0

[alsa-devel] Volume control channels mixed
by Simon van der Linden 10 Apr '07

10 Apr '07

Hello, I use Alsa in the latest Ubuntu bêta, with my MacBook. lspci gives the audio card as: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Sound is working pretty good. However, volume control channels (in alsamixer) are mixed: with laptop speakers: Master controls treble volume Front controls bass volume with headphones: Master has no effect Front controls the volume I'm unfamiliar with Alsa, but I guess it shouldn't be too hard to correct that. Browsing the web didn't give me more informations. Perhaps some of you could help me? -- Simon van der Linden jid: simon(a)vanderlinden.eu.org

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Re: [alsa-devel] [Audacity-devel] distortion with alsa I/O - no distortion with oss emulation
by Simon Lewis 10 Apr '07

10 Apr '07

Dear Alsa developers Audacity seems to work perfectly with AOSS and Jack but not directly with ALSA on some linux PCs. Trying to use ALSA directly leads to severe distortion due to lack of sync with Alsa. (I hope that is the correct terminology.) I received this reply from the Audacity developers, what tools are available from ALSA to investigate this problem? How can I determine what the real "Alsa configuration" is that portaudio has used setting up communication with Alsa. Best Regards, Simon > Markus Meyer schrieb: > > > As for Simon's problems with "synchronisation", I'm not so sure that it > is a Portaudio issue, or a software issue at all. Software issues tend > to be of the "on/off" type, and Portaudio does work well with ALSA in > the general case. Apart from being a hardware issue (defective sound > card, main board or something) this could also well be a race condition > in the ALSA driver (kernel module) which is only triggered by certain > buffer sizes, order of driver requests etc. Because Audacity is > multi-threaded it certainly stresses the driver/kernel subsystem more > than just, say, "arecord", which just gets blocks from the driver and > writes it out to disk. > > > Markus > > >> Simon Lewis schrieb: >> >> The main issue is that on some linux machines, mine included, there is >> severe distortion when Audacity is used with ALSA drivers. >> >> You will be pleased to hear that I have found the root cause of the >> problem. It is nothing to do with the options with which Audacity is >> compiled. The root cause is that portaudio loses synchronisation with >> ALSA's data stream. >> >> I was able to verify this yesterday, as I could start Audacity with >> Jack, and through Jack set up a number of simulation conditions and >> monitor the results. >> >> With my standard Jack settings of 96kHz / 256 Frames per period / 2 >> periods per buffer (latency 5.33 msecs.), ALSA initialises with sample >> format 32bit little-endian, 2 periods for capture and 2 periods for >> playback. >> >> Under these conditions Audacity cannot synchronise with the data stream >> and severe distortion is the result. >> >> However, increasing the frames per period to 1024 (latency 21.3 msec) >> and Audacity (1.3.2 with resampling disabled) runs perfectly and I can >> record for a long time. (CPU load is 9.6%) >> >> Reducing the frames per period to 512 and the same version of Audacity >> starts OK but after a few seconds, sometimes longer losses sync with the >> data stream. >> >> The standard 1.3.2 version of Audacity with resampling enabled does not >> sync on start with 512 frames per period but runs OK with Jack set to >> 1024 frames per period. >> >> The 1.3.3 version of Audacity complied with libsamplerate without >> portmixer runs OK with 512 frames per period under jack. >> >> NONE of the Audacity versions will run directly with the ALSA drivers. >> The Version with resampling disabled often starts OK, but after 30 sec >> to a few minutes of recording sync is lost and naturally distortion is >> the result. >> >> The conclusion is that portaudio is very poor at syncing with the >> alsadata stream, something which Jack can do with very little buffering. >> >> >> Ciao, Simon

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[alsa-devel] Correct use of ak4114.c?
by dustin＠seznam.cz 10 Apr '07

10 Apr '07

Hello, while working on support for AK4114 in Audiotrak MI/ODI/O card, I used corresponding code from revo.c (and similarly from juli.c), calling function snd_ak4114_create. However, this function does not initialize array ak4114->kctls[idx], leading to consequent kernel oops after the periodically called snd_ak4114_check_rate_and_errors, specifically in function snd_ctl_notify called by snd_ctl_notify(ak4114->card, SNDRV_CTL_EVENT_MASK_VALUE, &ak4114->kctls[0]->id) Temporarily I resolved the problem by commenting out the snd_ctl_notify calls. However, it seems author of ak4114.c intended to create the device with snd_ak4114_build instead which defines all the controls. I did not hit the problem until fixing the 4-wire communication in revo.c - that explaing why nobody had this problem with revo cards. I do not understand how the code can work in juli.c - perhaps I have overlooked something there. My question: Is the correct way to add null pointer checks before snd_ctl_notify calls only, or to use snd_ak4114_build insted of snd_ak4114_create? If using snd_ak4114_build, where do I get the parameters ply_substream and cap_substream? int snd_ak4114_build(struct ak4114 *ak4114, struct snd_pcm_substream *playback_substream, struct snd_pcm_substream *capture_substream); Thanks for any suggestions. Pavel.

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[alsa-devel] seq: timing unstable on slow computer when busy
by Gerald Grabner 09 Apr '07

09 Apr '07

Hi there, I'm trying to use a simple (self written) sequencer application on an old laptop with a 350 MHz CPU. It works, but the timing of output events becomes unstable when my program is busy with things like scheduling events for the subsequent loop or an unmuted track. I'm wondering about the best approach to tackle this problem. Depending on how the priority of the kernel's event scheduling is related to the priority of an application, I could imagine some of the following measures: * Running my process with lower or higher priority * Using (priorized) threads for the event scheduling * Setting Linux kernel config options * Setting ALSA related options (driver or lib) * ... While playing midi, I'm scheduling an echo event a few ticks before the end of every sequence, upon which a function to schedule the events of the next sequence is called. What is the typical length of this scheduling span: Just one bar? Or longer? Many thanks in advance, Gerald BTW: My operation system is Gentoo's Linux 2.6.19 with alsa-driver 1.0.14_rc1.

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[alsa-devel] Popping noise on HP dv6000z (CX20549) running x86 kernel
by enigma_0Z 08 Apr '07

08 Apr '07

Here's my situation: If I play a game (any game, in particular, however, neverball/putt or sauerbraten), the sound has these little soft "pop" noises happening so quick per second I cannot count them. This is when I'm running on an x86 kernel, however, when I run an x86_64 kernel, these are not present. I've got a 64-bit processor, specifically an AMD Sempron 3500+ Does Anyone know why? I've got a conexant CX20549 in my laptop. Thanks, enigma

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[alsa-devel] dont knw what this error means
by Ashlesha Shintre 06 Apr '07

06 Apr '07

Hi, I am getting the following error when I try to playback a raw sample file: "ALSA lib pcm.c:7216snd_pcm_set_params) Unable to get period size for PLAYBACK: Invalid argument" Does anyone know what this means? Thanks Ashlesha.

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