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[alsa-devel] [PATCH] hda: 92HD71BXX Mono Mute Support
by Matthew Ranostay 09 Jan '08

09 Jan '08

Added a mono output mute mixer for the 92hd71bxx family of codecs, this also removes the need for the mono out node to explicitly unmuted in the core init. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r b3915b3a8a22 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Jan 08 12:33:19 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Jan 08 17:58:12 2008 -0500 @@ -534,8 +534,6 @@ static struct hda_verb stac92hd71bxx_cor { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* unmute mono out node */ - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, }; static struct hda_verb stac92hd71bxx_analog_core_init[] = { @@ -553,8 +551,6 @@ static struct hda_verb stac92hd71bxx_ana { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, - /* unmute mono out node */ - { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -714,6 +710,8 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -728,6 +726,8 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ };

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[alsa-devel] No sound over HDMI on HDA-Intel - HDA ATI HDMI
by Georgi Chorbadzhiyski 09 Jan '08

09 Jan '08

Hi, I have an Asus MB with integrated ATI X1200 audio controller connected over HDMI to an LCD TV. intel-hda driver correctly identifies and drives the card, but I get no sound over HDMI only over the standard stereo output jack. I've unmuted iec958 control on the HDMI card. I'm testing the card like this: root@browser:~# aplay -D plughw:1,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Stereo No sound. root@browser:~# aplay -D plughw:0,0 test.wav Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 16000 Hz, Stereo Sound is ok. Is there some patches that I could try to get the HDMI output working? Similar reports: http://www.mail-archive.com/ubuntu-bugs@lists.ubuntu.com/msg590477.html http://www.redhat.com/archives/rhl-list/2007-November/msg00372.html aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: HDA Generic [HDA Generic] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 0: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 /proc/asound/cards 0 [SB ]: HDA-Intel - HDA ATI SB HDA ATI SB at 0xfe8f4000 irq 16 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xfeae8000 irq 19 lspci 00:00.0 Host bridge: ATI Technologies Inc RS690 Host Bridge 00:01.0 PCI bridge: ATI Technologies Inc RS690 PCI to PCI Bridge (Internal gfx) 00:06.0 PCI bridge: ATI Technologies Inc RS690 PCI to PCI Bridge (PCI Express Port 2) 00:07.0 PCI bridge: ATI Technologies Inc RS690 PCI to PCI Bridge (PCI Express Port 3) 00:12.0 SATA controller: ATI Technologies Inc SB600 Non-Raid-5 SATA 00:13.0 USB Controller: ATI Technologies Inc SB600 USB (OHCI0) 00:13.1 USB Controller: ATI Technologies Inc SB600 USB (OHCI1) 00:13.2 USB Controller: ATI Technologies Inc SB600 USB (OHCI2) 00:13.3 USB Controller: ATI Technologies Inc SB600 USB (OHCI3) 00:13.4 USB Controller: ATI Technologies Inc SB600 USB (OHCI4) 00:13.5 USB Controller: ATI Technologies Inc SB600 USB Controller (EHCI) 00:14.0 SMBus: ATI Technologies Inc SBx00 SMBus Controller (rev 14) 00:14.1 IDE interface: ATI Technologies Inc SB600 IDE 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia 00:14.3 ISA bridge: ATI Technologies Inc SB600 PCI to LPC Bridge 00:14.4 PCI bridge: ATI Technologies Inc SBx00 PCI to PCI Bridge 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:05.0 VGA compatible controller: ATI Technologies Inc RS690 [Radeon X1200 Series] 01:05.2 Audio device: ATI Technologies Inc Radeon X1200 Series Audio Controller 03:00.0 Ethernet controller: Attansic Technology Corp. L1 Gigabit Ethernet Adapter (rev b0) lspci -vv 01:05.2 Audio device: ATI Technologies Inc Radeon X1200 Series Audio Controller Subsystem: ASUSTeK Computer Inc. Unknown device 8287 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 64, Cache Line Size: 64 bytes Interrupt: pin B routed to IRQ 19 Region 0: Memory at feae8000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA 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 dmesg: Linux version 2.6.24-rc6 (gf@gf) (gcc version 4.1.2) #32 SMP PREEMPT Tue Jan 8 19:08:44 EET 2008 BIOS-provided physical RAM map: BIOS-e820: 0000000000000000 - 000000000009fc00 (usable) BIOS-e820: 000000000009fc00 - 00000000000a0000 (reserved) BIOS-e820: 00000000000e4000 - 0000000000100000 (reserved) BIOS-e820: 0000000000100000 - 0000000037fb0000 (usable) BIOS-e820: 0000000037fb0000 - 0000000037fbe000 (ACPI data) BIOS-e820: 0000000037fbe000 - 0000000037fe0000 (ACPI NVS) BIOS-e820: 0000000037fe0000 - 0000000038000000 (reserved) BIOS-e820: 00000000fff00000 - 0000000100000000 (reserved) 895MB LOWMEM available. found SMP MP-table at 000ff780 Entering add_active_range(0, 0, 229296) 0 entries of 256 used Zone PFN ranges: DMA 0 -> 4096 Normal 4096 -> 229296 Movable zone start PFN for each node early_node_map[1] active PFN ranges 0: 0 -> 229296 On node 0 totalpages: 229296 DMA zone: 32 pages used for memmap DMA zone: 0 pages reserved DMA zone: 4064 pages, LIFO batch:0 Normal zone: 1759 pages used for memmap Normal zone: 223441 pages, LIFO batch:31 Movable zone: 0 pages used for memmap DMI present. ATI board detected. Disabling timer routing over 8254. Intel MultiProcessor Specification v1.4 Virtual Wire compatibility mode. OEM ID: ASUS Product ID: APIC at: 0xFEE00000 Processor #0 15:15 APIC version 16 I/O APIC #1 Version 33 at 0xFEC00000. Enabling APIC mode: Flat. Using 1 I/O APICs Processors: 1 Allocating PCI resources starting at 40000000 (gap: 38000000:c7f00000) Built 1 zonelists in Zone order, mobility grouping on. Total pages: 227505 Kernel command line: auto BOOT_IMAGE=Linux ro root=801 quiet mapped APIC to ffffb000 (fee00000) mapped IOAPIC to ffffa000 (fec00000) Enabling fast FPU save and restore... done. Enabling unmasked SIMD FPU exception support... done. Initializing CPU#0 PID hash table entries: 4096 (order: 12, 16384 bytes) Detected 2009.066 MHz processor. Console: colour dummy device 80x25 console [tty0] enabled Dentry cache hash table entries: 131072 (order: 7, 524288 bytes) Inode-cache hash table entries: 65536 (order: 6, 262144 bytes) Memory: 906216k/917184k available (1460k kernel code, 10468k reserved, 560k data, 172k init, 0k highmem) virtual kernel memory layout: fixmap : 0xfffba000 - 0xfffff000 ( 276 kB) vmalloc : 0xf8800000 - 0xfffb8000 ( 119 MB) lowmem : 0xc0000000 - 0xf7fb0000 ( 895 MB) .init : 0xc02ff000 - 0xc032a000 ( 172 kB) .data : 0xc026d14d - 0xc02f9154 ( 560 kB) .text : 0xc0100000 - 0xc026d14d (1460 kB) Checking if this processor honours the WP bit even in supervisor mode... Ok. SLUB: Genslabs=11, HWalign=64, Order=0-1, MinObjects=4, CPUs=1, Nodes=1 Calibrating delay using timer specific routine.. 4109.60 BogoMIPS (lpj=2054803) Mount-cache hash table entries: 512 CPU: After generic identify, caps: 078bfbff ebd3fbff 00000000 00000000 00002001 00000000 0000001d 00000000 CPU: L1 I Cache: 64K (64 bytes/line), D cache 64K (64 bytes/line) CPU: L2 Cache: 512K (64 bytes/line) CPU: After all inits, caps: 078bfbff ebd3fbff 00000000 00000410 00002001 00000000 0000001d 00000000 Compat vDSO mapped to ffffe000. Checking 'hlt' instruction... OK. SMP alternatives: switching to UP code Freeing SMP alternatives: 8k freed CPU0: AMD Athlon(tm) 64 Processor 3200+ stepping 02 Total of 1 processors activated (4109.60 BogoMIPS). ExtINT not setup in hardware but reported by MP table ENABLING IO-APIC IRQs ..TIMER: vector=0x31 apic1=0 pin1=2 apic2=0 pin2=0 Brought up 1 CPUs net_namespace: 64 bytes NET: Registered protocol family 16 PCI: PCI BIOS revision 3.00 entry at 0xf0031, last bus=4 Setting up standard PCI resources SCSI subsystem initialized libata version 3.00 loaded. usbcore: registered new interface driver usbfs usbcore: registered new interface driver hub usbcore: registered new device driver usb PCI: Probing PCI hardware PCI: Probing PCI hardware (bus 00) PCI: Transparent bridge - 0000:00:14.4 PCI: Using IRQ router default [1002/438d] at 0000:00:14.3 PCI->APIC IRQ transform: 0000:00:12.0[A] -> IRQ 22 PCI->APIC IRQ transform: 0000:00:13.0[A] -> IRQ 16 PCI->APIC IRQ transform: 0000:00:13.1[B] -> IRQ 17 PCI->APIC IRQ transform: 0000:00:13.2[C] -> IRQ 18 PCI->APIC IRQ transform: 0000:00:13.3[B] -> IRQ 17 PCI->APIC IRQ transform: 0000:00:13.4[C] -> IRQ 18 PCI->APIC IRQ transform: 0000:00:13.5[D] -> IRQ 19 PCI->APIC IRQ transform: 0000:00:14.1[A] -> IRQ 16 PCI->APIC IRQ transform: 0000:00:14.2[A] -> IRQ 16 PCI->APIC IRQ transform: 0000:01:05.0[A] -> IRQ 18 PCI->APIC IRQ transform: 0000:01:05.2[B] -> IRQ 19 PCI->APIC IRQ transform: 0000:03:00.0[A] -> IRQ 19 PCI: Bridge: 0000:00:01.0 IO window: e000-efff MEM window: fe900000-feafffff PREFETCH window: f0000000-f7ffffff PCI: Bridge: 0000:00:06.0 IO window: disabled. MEM window: disabled. PREFETCH window: disabled. PCI: Bridge: 0000:00:07.0 IO window: disabled. MEM window: feb00000-febfffff PREFETCH window: disabled. PCI: Bridge: 0000:00:14.4 IO window: disabled. MEM window: disabled. PREFETCH window: disabled. PCI: Setting latency timer of device 0000:00:06.0 to 64 PCI: Setting latency timer of device 0000:00:07.0 to 64 NET: Registered protocol family 2 Time: tsc clocksource has been installed. IP route cache hash table entries: 32768 (order: 5, 131072 bytes) TCP established hash table entries: 131072 (order: 8, 1048576 bytes) TCP bind hash table entries: 65536 (order: 7, 786432 bytes) TCP: Hash tables configured (established 131072 bind 65536) TCP reno registered io scheduler noop registered (default) Boot video device is 0000:01:05.0 vesafb: framebuffer at 0xf0000000, mapped to 0xf8800000, using 2000k, total 16384k vesafb: mode is 640x400x32, linelength=2560, pages=15 vesafb: protected mode interface info at c000:a034 vesafb: pmi: set display start = c00ca0be, set palette = c00ca17c vesafb: scrolling: redraw vesafb: Truecolor: size=0:8:8:8, shift=0:16:8:0 Console: switching to colour frame buffer device 80x25 fb0: VESA VGA frame buffer device Real Time Clock Driver v1.12ac Linux agpgart interface v0.102 Serial: 8250/16550 driver $Revision: 1.90 $ 4 ports, IRQ sharing disabled PCI: Setting latency timer of device 0000:03:00.0 to 64 atl1 0000:03:00.0: version 2.0.7 ahci 0000:00:12.0: version 3.0 ahci 0000:00:12.0: controller can't do 64bit DMA, forcing 32bit ahci 0000:00:12.0: controller can't do PMP, turning off CAP_PMP ahci 0000:00:12.0: AHCI 0001.0100 32 slots 4 ports 3 Gbps 0xf impl SATA mode ahci 0000:00:12.0: flags: ncq sntf ilck pm led clo pio slum part scsi0 : ahci scsi1 : ahci scsi2 : ahci scsi3 : ahci ata1: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ff900 irq 22 ata2: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ff980 irq 22 ata3: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ffa00 irq 22 ata4: SATA max UDMA/133 abar m1024@0xfe8ff800 port 0xfe8ffa80 irq 22 ata1: SATA link down (SStatus 0 SControl 300) ata2: SATA link up 1.5 Gbps (SStatus 113 SControl 300) ata2.00: ATA-7: ST380815AS, 3.AAD, max UDMA/133 ata2.00: 156301488 sectors, multi 16: LBA48 NCQ (depth 31/32) ata2.00: configured for UDMA/133 ata3: SATA link down (SStatus 0 SControl 300) ata4: SATA link down (SStatus 0 SControl 300) scsi 1:0:0:0: Direct-Access ATA ST380815AS 3.AA PQ: 0 ANSI: 5 sd 1:0:0:0: [sda] 156301488 512-byte hardware sectors (80026 MB) sd 1:0:0:0: [sda] Write Protect is off sd 1:0:0:0: [sda] Mode Sense: 00 3a 00 00 sd 1:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sd 1:0:0:0: [sda] 156301488 512-byte hardware sectors (80026 MB) sd 1:0:0:0: [sda] Write Protect is off sd 1:0:0:0: [sda] Mode Sense: 00 3a 00 00 sd 1:0:0:0: [sda] Write cache: enabled, read cache: enabled, doesn't support DPO or FUA sda: sda1 sd 1:0:0:0: [sda] Attached SCSI disk ehci_hcd 0000:00:13.5: EHCI Host Controller ehci_hcd 0000:00:13.5: new USB bus registered, assigned bus number 1 ehci_hcd 0000:00:13.5: debug port 1 ehci_hcd 0000:00:13.5: irq 19, io mem 0xfe8ff000 ehci_hcd 0000:00:13.5: USB 2.0 started, EHCI 1.00, driver 10 Dec 2004 usb usb1: configuration #1 chosen from 1 choice hub 1-0:1.0: USB hub found hub 1-0:1.0: 10 ports detected ohci_hcd: 2006 August 04 USB 1.1 'Open' Host Controller (OHCI) Driver ohci_hcd 0000:00:13.0: OHCI Host Controller ohci_hcd 0000:00:13.0: new USB bus registered, assigned bus number 2 ohci_hcd 0000:00:13.0: irq 16, io mem 0xfe8fe000 usb usb2: configuration #1 chosen from 1 choice hub 2-0:1.0: USB hub found hub 2-0:1.0: 2 ports detected ohci_hcd 0000:00:13.1: OHCI Host Controller ohci_hcd 0000:00:13.1: new USB bus registered, assigned bus number 3 ohci_hcd 0000:00:13.1: irq 17, io mem 0xfe8fd000 usb usb3: configuration #1 chosen from 1 choice hub 3-0:1.0: USB hub found hub 3-0:1.0: 2 ports detected ohci_hcd 0000:00:13.2: OHCI Host Controller ohci_hcd 0000:00:13.2: new USB bus registered, assigned bus number 4 ohci_hcd 0000:00:13.2: irq 18, io mem 0xfe8fc000 usb usb4: configuration #1 chosen from 1 choice hub 4-0:1.0: USB hub found hub 4-0:1.0: 2 ports detected ohci_hcd 0000:00:13.3: OHCI Host Controller ohci_hcd 0000:00:13.3: new USB bus registered, assigned bus number 5 ohci_hcd 0000:00:13.3: irq 17, io mem 0xfe8fb000 usb usb5: configuration #1 chosen from 1 choice hub 5-0:1.0: USB hub found hub 5-0:1.0: 2 ports detected usb 3-2: new low speed USB device using ohci_hcd and address 2 ohci_hcd 0000:00:13.4: OHCI Host Controller ohci_hcd 0000:00:13.4: new USB bus registered, assigned bus number 6 ohci_hcd 0000:00:13.4: irq 18, io mem 0xfe8fa000 usb 3-2: configuration #1 chosen from 1 choice usb usb6: configuration #1 chosen from 1 choice hub 6-0:1.0: USB hub found hub 6-0:1.0: 2 ports detected USB Universal Host Controller Interface driver v3.0 serio: i8042 KBD port at 0x60,0x64 irq 1 serio: i8042 AUX port at 0x60,0x64 irq 12 mice: PS/2 mouse device common for all mice input: Logitech USB-PS/2 Optical Mouse as /devices/pci0000:00/0000:00:13.1/usb3/3-2/3-2:1.0/input/input0 input: USB HID v1.10 Mouse [Logitech USB-PS/2 Optical Mouse] on usb-0000:00:13.1-2 usbcore: registered new interface driver usbhid drivers/hid/usbhid/hid-core.c: v2.6:USB HID core driver Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Nov 20 19:16:42 2007 UTC). input: AT Translated Set 2 keyboard as /devices/platform/i8042/serio0/input/input1 ALSA device list: #0: HDA ATI SB at 0xfe8f4000 irq 16 #1: HDA ATI HDMI at 0xfeae8000 irq 19 TCP cubic registered NET: Registered protocol family 1 NET: Registered protocol family 17 Using IPI Shortcut mode VFS: Mounted root (ext2 filesystem) readonly. Freeing unused kernel memory: 172k freed atl1 0000:03:00.0: eth0 link is up 100 Mbps full duplex -- Georgi Chorbadzhiyski http://georgi.unixsol.org/ - To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 09 Jan '08

09 Jan '08

Ok, here's the current state of affairs... both the compilation and driver seem to work well now, though I haven't tested SPDIF yet. I don't know why both John Stile and I were getting ioctl errors, perhaps if it is still happening to him he can elucidate, but it isn't happening to me any more with the latest build. I obtained and built a clean mercurial clone this morning (07 Jan 08): %> cd /usr/src %> rm -rf alsa-driver %> hg clone http://hg-mirror.alsa-project.org/alsa-driver alsa-driver %> cd alsa-driver %> hg clone http://hg-mirror.alsa-project.org/alsa-kernel alsa-kernel %> hgcompile [...] compilation proceeds cleanly... %> make install %> alsaconf [...] select card and configure it via alsamixer Lo and belold, it all worked! I have a clean compilation and good response from aplay and other user space tools. I haven't applied any patches, and I haven't been able to test SPDIF input yet but I'm betting John Stile should be able to report that faster than I can get to it... Thank you very much for your help with this and for all of the work you and the ALSA team do! -- Darryl -------------- Original message ---------------------- From: Takashi Iwai <tiwai(a)suse.de> > At Wed, 02 Jan 2008 22:31:19 +0000, > Darryl House wrote: > > > > > > -------------- Original message ---------------------- > > [...] > > > > I'd like to get this card's spdif working. What else can I do to help? > > > > Although the driver loads, the user space tools can't use the card any > > > > more. Thank you. > > > > > > > Should I assume at this point no more work will be done on the > > > snd-hda-intel driver to fixing SPDIF-in for the vt1708 sound card? > > > Thank you for all your work so far. > > > > > > > I think it is a bit early to make that assumption. There isn't a lot > > of demand for this particular card, but it doesn't seem that it > > will take too much work to get it going again... after all, it did > > basically work before the last patch, and I am hoping that Takashi > > will find some time soon to address the issues. > > Hm, I see no obvious error in the patch. Are you sure that the latest > HG driver works properly if you don't apply the patch? > > Let's begin with that. > > > Takashi > _______________________________________________ > 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] Problem using arecord and aplay
by conn intel 08 Jan '08

08 Jan '08

Dear Friends, I am using kernel 2.6.22.2, and following alsa library and utilities : alsa-lib-1.0.14a & alsa-utils-1.0.14. I am having problem executing arecord and aplay alsa utilities. After executing below command it returns immeaditely to command prompt with following messages. The command and output:: # arecord -c 2 -d 10 -f S16_LE -r 44100 -t wav /home/test.wav Recording WAVE '/home/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, St ereo ALSA sound/arm/as352x-audio.c:1153: len:15888 arecord: set_params:1012: unable to install sw params: start_mode: DATA xrun_mode: STOP tstamp_mode: NONE period_step: 1 sleep_min: 0 avail_min: 0 xfer_align: 10 silence_threshold: 0 silence_size: 0 boundary: 1445068800 # waiting for any comments or suggestions. Thank you, Ankur.

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Re: [alsa-devel] no input/output analog channels on rme multiface I
by rmgls＠free.fr 08 Jan '08

08 Jan '08

On Fri, 21 Dec 2007 13:41:42 +0100 Takashi Iwai <tiwai(a)suse.de> wrote >> on Sun, 09 Dec 2007 23:39:46 +0100, rmgls(a)free.fr wrote: >> Hi all, >> >> the multiface 1, does not work for me. >> it is recognized as hammerfal DSP, but the amixer shows no analog in/out >> (there are 8 analog I/O in the box) and of course no volume control for >> any channel. > Did you try hdspmixer? > Takashi yes i tried it, without success, hdspmixer does not start with an (inaccesible) message. as i mentionned earlier, the firmware is correctly loaded, and the aslamixer doesn't show any analog I/O. of course one can't play any analog sound, and the midi part is not detected: (is it the same problem?° Note: the kernel is 22.3, running on debian, with the hg version from 20080107: drivers, libs, and all. thanks for your help. Raoul rmgls(a)free.Fr

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[alsa-devel] [RFC] alsa-kernel clean-ups
by Takashi Iwai 08 Jan '08

08 Jan '08

Hi, ALSA kernel tree has lots of unused and rotten codes. I'd like to clean up this. Here we go... 1. Indirect control element access This is what I already suggested. This hasn't been used (we have no proper API in alsa-lib), and 32/64bit wrapper doesn't work at all. Also, the idea to copy such a big data area via ioctl is bad. If we need a big matrix mixer, let make each matrix element accessible individually. Copying the whole matrix at each time isn't efficient. 2. PCM xfer_align parameter This sw_params parameter has never been used in a sane manner, and no one understands what this does exactly. The current implementation looks also buggy because it allows write of shorter size than xfer_align. So, if you do partial writes, the write isn't actually aligned at all. Removing this parameter will make some pcm_lib_* code more readable (and less buggy). 3. PCM sleep_min and tick The "tick" in PCM is set (again) via sw_params. And, nobody uses this feature at all except for a command line option of aplay. (This is literally "nobody", as I checked alsa-lib API calls in all programs in some major distros.) Above all, if we need finer wake-ups for the position update, it's basically an issue that the driver should solve, not tuned by each application. 4. sound/driver.h This header file exists only for some hacks to adapt alsa-driver tree. It's useless for building in the kernel. Let's move a few lines in it to sound/core.h and remove it. For building the modules on alsa-driver external tree, we can simply add #include "adriver.h" to each build stub (alsa-driver/*/*.c) before inclusion of alsa-kernel codes. I'll post some patches to clean the above thing up in the following posts. Please speak up if you have objections. Takashi

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[alsa-devel] Driver - PCM MMAP mode timestamps
by Jaroslav Kysela 08 Jan '08

08 Jan '08

Hi, it seems that we have another inconsistency with PCM MMAP mode timestamps. If I read code correctly, snd_pcm_status() returns timestamp from last interrupt BUT updates hw_ptr to current (calling snd_pcm_update_hw_ptr() again). It means, that application will get unrelated timestamp to hw position returned in snd_pcm_status() in case of mmap mode. I don't see any way to use this timestamp now. To fix this issue, I would like to create a next status ioctl which will return only mmap timestamp plus last related interrupt ring buffer position and remove 'if (runtime->tstamp_mode == SNDRV_PCM_TSTAMP_MMAP)' condition from snd_pcm_status() in driver completely. Any opinion? Other idea? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

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Re: [alsa-devel] ALC268 "test" model
by Jonathan Woithe 08 Jan '08

08 Jan '08

Hi Tobin > > I try to use a patch from J. Woithe to put a test model into the > > ALC268 driver [1] (enabling all pins etc.); unfortunately it gives a > > lot of compiling errors ([2]) which I am unable to understand as I > > don't have sufficient knowledge of the internal API of HDA codec. > > > > Can someone please look over it and correct the mistakes? > > > > Marco Schuster > > > > [1]: http://harddisk.is-a-geek.org/patch.txt > > [2]: http://harddisk.is-a-geek.org/compiling-errors.txt > > Hi, and thanks for the effort. I'll fix this up a bit and send out an > updated patch this weekend. I have an ALC268 test board to work with > now, so I should be able to get it figured out. I'm not entirely sure what's going on here with this patch. As is indicated by the patch itself it's against ALSA in the 2.6.24-rc6 kernel and it applies and compiles fine for me. The last few error messages at the second URL seems to suggest that the patch was in some way corrupted because I'm seeing non-ASCII characters in the error messages. I should note that the patch as referenced by Marco was not believed by me to be ready for release yet - that is why I haven't forwarded it on to the list for consideration yet. I sent it to Marco privately and I had hoped to utilise Marco's system to verify basic functionality of the patch before posting it for consideration. It was prepared in a hurry at the end of last year just before I went on leave and have not had a chance to revisit the patch or the issues that Marco saw until now. Since you have an ALC268 board it's probably sensible for you to take over the patch at this point - I don't have an ALC268 system. However, understand that by making this initial patch available to Marco privately it was not my intention to create a lot of extra work for yourself (or any other developer for that matter). I had intended to see it through to operational status before bothering the list with it. Are you happy to take over management of this patch or would you like me to resume dialog with Marco? If the patch at the above URL is in some way corrupted drop me a line and I'll forward you the original. Regards jonathan

3 5

[alsa-devel] [PATCH] hda: Dynamically create digital gain mixers
by Matthew Ranostay 08 Jan '08

08 Jan '08

Dynamically create digital gain mixers for dmics that have out-amp support. Also some 92HD73xx's codecs don't have DMIC gains, so this also prevents creating dead mixers. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 37956a45462d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 07 15:16:37 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 07 15:04:42 2008 -0500 @@ -463,9 +463,6 @@ static struct hda_verb stac92hd73xx_6ch_ /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -490,9 +487,6 @@ static struct hda_verb stac92hd73xx_8ch_ /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -519,9 +513,6 @@ static struct hda_verb stac92hd73xx_10ch /* setup adcs to point to mixer */ { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, - { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ - { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ - { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, @@ -635,10 +626,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -665,10 +652,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -695,10 +678,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -725,10 +704,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), @@ -745,10 +720,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - - /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -2313,15 +2284,18 @@ static int stac92xx_auto_create_dmic_inp struct sigmatel_spec *spec = codec->spec; struct hda_input_mux *dimux = &spec->private_dimux; hda_nid_t con_lst[HDA_MAX_NUM_INPUTS]; - int i, j; + int err, i, j; + char name[32]; dimux->items[dimux->num_items].label = stac92xx_dmic_labels[0]; dimux->items[dimux->num_items].index = 0; dimux->num_items++; for (i = 0; i < spec->num_dmics; i++) { + hda_nid_t nid; int index; int num_cons; + unsigned int wcaps; unsigned int def_conf; def_conf = snd_hda_codec_read(codec, @@ -2332,17 +2306,32 @@ static int stac92xx_auto_create_dmic_inp if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; + nid = spec->dmic_nids[i]; num_cons = snd_hda_get_connections(codec, spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) - if (con_lst[j] == spec->dmic_nids[i]) { + if (con_lst[j] == nid) { index = j; goto found; } continue; found: + wcaps = get_wcaps(codec, nid); + + if (wcaps & AC_WCAP_OUT_AMP) { + sprintf(name, "%s Capture Volume", + stac92xx_dmic_labels[dimux->num_items]); + + err = stac92xx_add_control(spec, + STAC_CTL_WIDGET_VOL, + name, + HDA_COMPOSE_AMP_VAL(nid, 3, 0, HDA_OUTPUT)); + if (err < 0) + return err; + } + dimux->items[dimux->num_items].label = stac92xx_dmic_labels[dimux->num_items]; dimux->items[dimux->num_items].index = index;

2 1

[alsa-devel] HDA: Enable chipset gcap usage
by Tobin Davis 08 Jan '08

08 Jan '08

Summary: hda_intel.c use gcap register to determine number of streams supported by southbridge. This patch removes hardcoded values for the number of streams supported by the southbridge in most chipsets, and reads these values from the chipset directly. Most systems are hardwired for 4 streams in each direction, but newer chipsets change that capability. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 5

Re: [alsa-devel] fujitsu siemens amilo la 1703 vt1807 hda intel card problem
by Tobin Davis 07 Jan '08

07 Jan '08

The information you provided is good, we're just a bit behind. I'm trying to get some programming specs for this codec now. In the mean time, could you try using a tip build of the alsa driver? You can download it here: ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver/ This may have some newer code that will enable your audio. Also, try the drivers on http://viaarena.com. THey may have an updated version that hasn't hit the main tree yet. Tobin On Mon, 2008-01-07 at 21:59 +0100, Norayr Chilingaryan wrote: > Excuse me for bothering you. > May be I need to provide other info? > Did you receive the link? > Di you have any ideas? > > Thank you > > Norayr > > > On Jan 5, 2008 5:53 PM, Norayr Chilingaryan < gnorayr(a)gmail.com> > wrote: > > http://pastebin.ca/842745 > > Thank you very much! > > > > > > On Jan 5, 2008 3:56 AM, Tobin Davis <tdavis(a)dsl-only.net> > wrote: > > Could you download and run > http://bulletproof.servebeer.com/alsa/scripts/alsa-info.sh > > This script will provide a lot of useful information. > After running it, please reply to this email with the > link that the script generates, so we can retrieve the > data, thanks. > > Tobin > > > > > On Sat, 2008-01-05 at 02:48 +0100, Norayr Chilingaryan > wrote: > > > > Hello I decided to write here, firtly, because my > > messages in aksa-user have been ingored, and > > secondly, because I understood that my card most > > probably just unsupported and I must contact > > developers. I have tried to pass options > > model=fujitsu, because I have a fujitsu siements > > notebook etc I have also tried to pass amny other > > options in /etc/modules.d/alsa but withnot effect I > > use gentoo, and the only thing I noticed with latest > > kernels and alsa drivers 1.0.15 is that I can hear > > sound through the headphones, but with low volume. > > It is not possible to get volume higher or speaker > > to make sounds now. So, please, this is information > > about the card 04:01.0 Audio device: VIA > > Technologies, Inc. VIA High Definition Audio > > Controller (rev 10) kakraphoon ~ # > > cat /proc/asound/card0/codec#* | grep Codec Codec: > > VIA VIA VT1708 Codec: Motorola Si3054 Thank you very > > much Norayr > > > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > -- Tobin Davis > > > Hain't we got all the fools in town on our side? And hain't that a big > enough majority in any town? > -- Mark Twain, "Huckleberry Finn" > > > > > -- Tobin Davis Anything anybody can say about America is true. -- Emmett Grogan

1 0

[alsa-devel] Cross compiling alsa-lib (again)
by Adrian McMenamin 07 Jan '08

07 Jan '08

Well, I searched through 96 weeks of archived alsa-devel postings to get to this one: http://article.gmane.org/gmane.linux.alsa.devel/37120 (Which shows I am doing the right thing in any case.) This post recommends this: CFLAGS="-Os -m4 -ml" ./configure --prefix=/usr --host=sh4-unknown-linux-gnu --target=sh4-unknown-linux-gnu --build=i686-pc-linux-gnu I admit to not using the additional cflags stuff before, but even if I use this: CFLAGS="-Os -m4 -ml" ./configure --host=sh4-linux --prefix=/usr CC=/home/adrian/dreamy-linux/buildroot/build_sh4/staging_dir/bin/sh4-linux-gcc --disable-python --disable-rawmidi --enable-static --with-confdir=/home/adrian/dreamy-linux/buildroot/project_build_sh4/dreamcast/root/usr/share/alsa --disable-mixer --enable-static=yes --enable-shared=no --target=sh4-linux --build=i686-linux Followed by: make and make DESTDIR=/home/adrian/dreamy-linux/buildroot/project_build_sh4/dreamcast/root install I still get this: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AICA [AICA], device 0: AICA PCM [AICA PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 Followed by: $ aplay -L null Discard all samples (playback) or generate zero samples (capture) $ aplay /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' aplay: set_params:895: Access type not available $ aplay -D hw:0,0 /aine-email.wav Playing WAVE '/aine-email.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: set_params:895: Access type not available Just as before. Any clues?

4 8

Re: [alsa-devel] [PATCH] hda: Advanced power management for codecs.
by Takashi Iwai 07 Jan '08

07 Jan '08

At Mon, 07 Jan 2008 12:31:49 -0500, Matthew Ranostay wrote: > > >> @@ -2830,7 +2883,12 @@ static void stac92xx_hp_detect(struct hd > >> for (i = 0; i < cfg->speaker_outs; i++) > >> stac92xx_set_pinctl(codec, cfg->speaker_pins[i], > >> AC_PINCTL_OUT_EN); > >> - } > >> + for (i = 0; i < cfg->hp_outs; i++) > >> + pwr_state |= 1 << (cfg->hp_pins[i] - 0xa); > >> + } > >> + > >> + /* power down unused ports */ > >> + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, pwr_state); > >> > > > > Is this verb available on every STAC codec? I took a look at some > > specs but they don't show this verb at all. > > > Ah no this a new one for recent codecs, probably need to some sort of > check for this. Or is it equivalent to power down each unused pin widget? If so, we can do it somehow better in the common code. Takashi

1 0

[alsa-devel] [PATCH] hda: Advanced power management for codecs.
by Matthew Ranostay 07 Jan '08

07 Jan '08

Added support for advanced power management features in new IDT codecs. Inactive ADCs and DACs are changed to D3 mode when not in playback or capture, also inactive line-out/headphone ports are disabled when jack detect does not sense a connection. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 5b03c0176aa5 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Jan 07 13:33:45 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Mon Jan 07 11:22:56 2008 -0500 @@ -160,6 +160,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; + unsigned int io_state[2]; unsigned int clfe_swap; unsigned int aloopback; @@ -438,6 +439,52 @@ static int stac92xx_aloopback_put(struct kcontrol->private_value >> 16, dac_mode); return 1; +} + +static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) +{ + int i; + + for (i = 0; i < spec->multiout.num_dacs; i++) { + if (spec->multiout.dac_nids[i] == nid) + return 1; + } + + return 0; +} + +static int stac92xx_mixer_amp_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct hda_codec *codec = snd_kcontrol_chip(kcontrol); + struct sigmatel_spec *spec = codec->spec; + hda_nid_t nid = kcontrol->private_value & 0xffff; + int idx = (kcontrol->private_value >> 19) & 0xf; + int chs = (kcontrol->private_value >> 16) & 0x3; + long *valp = ucontrol->value.integer.value; + unsigned int wid_caps, val, change; + + change = snd_hda_mixer_amp_switch_put(kcontrol, ucontrol); +#ifdef CONFIG_SND_HDA_POWER_SAVE + if (!is_in_dac_nids(spec, nid)) + nid = spec->adc_nids[idx]; + + val = *valp++; + wid_caps = get_wcaps(codec, nid); + if (wid_caps & AC_WCAP_POWER) { + if (chs == 3) + val |= *valp; + else { /* Center/LFE mixers workaround */ + spec->io_state[--chs] = val; + val |= spec->io_state[chs ? 0 : 1]; + } + + /* set to power state D3 mode if DAC/ADC isn't being used */ + snd_hda_codec_write(codec, nid, 0, AC_VERB_SET_POWER_STATE, + val ? 0: AC_PWRST_D3); + } +#endif + return change; } static struct hda_verb stac9200_core_init[] = { @@ -622,6 +669,20 @@ static struct hda_verb stac9205_core_ini .private_value = verb_read | (verb_write << 16), \ } +#define STAC_CODEC_MUTE_IDX(xname, xcidx, nid, xindex, direction) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = xname, \ + .index = xcidx, \ + .info = snd_hda_mixer_amp_switch_info, \ + .get = snd_hda_mixer_amp_switch_get, \ + .put = stac92xx_mixer_amp_switch_put, \ + .private_value = HDA_COMPOSE_AMP_VAL(nid, 3, xindex, direction), \ + } + +#define STAC_CODEC_MUTE(xname, nid, xindex, direction) \ + STAC_CODEC_MUTE_IDX(xname, 0, nid, xindex, direction) + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), HDA_CODEC_MUTE("Master Playback Switch", 0xb, 0, HDA_OUTPUT), @@ -640,10 +701,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -670,10 +731,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -700,10 +761,10 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), - - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x1, HDA_OUTPUT), HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), @@ -730,12 +791,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x1, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x1, HDA_OUTPUT), HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), @@ -751,12 +812,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x1, HDA_OUTPUT), + STAC_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x1, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x1, HDA_OUTPUT), { } /* end */ }; @@ -857,7 +918,7 @@ static int stac92xx_build_controls(struc if (err < 0) return err; } - return 0; + return 0; } static unsigned int ref9200_pin_configs[8] = { @@ -1962,7 +2023,7 @@ enum { static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), - HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_CODEC_MUTE(NULL, 0, 0, 0), STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2059,19 +2120,6 @@ static int stac92xx_add_dyn_out_pins(str return 0; } - -static int is_in_dac_nids(struct sigmatel_spec *spec, hda_nid_t nid) -{ - int i; - - for (i = 0; i < spec->multiout.num_dacs; i++) { - if (spec->multiout.dac_nids[i] == nid) - return 1; - } - - return 0; -} - /* * Fill in the dac_nids table from the parsed pin configuration * This function only works when every pin in line_out_pins[] @@ -2806,6 +2854,7 @@ static void stac92xx_hp_detect(struct hd struct sigmatel_spec *spec = codec->spec; struct auto_pin_cfg *cfg = &spec->autocfg; int i, presence; + unsigned char pwr_state = 0; presence = 0; for (i = 0; i < cfg->hp_outs; i++) { @@ -2816,12 +2865,16 @@ static void stac92xx_hp_detect(struct hd if (presence) { /* disable lineouts, enable hp */ - for (i = 0; i < cfg->line_outs; i++) + for (i = 0; i < cfg->line_outs; i++) { stac92xx_reset_pinctl(codec, cfg->line_out_pins[i], AC_PINCTL_OUT_EN); - for (i = 0; i < cfg->speaker_outs; i++) + pwr_state |= 1 << (cfg->line_out_pins[i] - 0xa); + } + for (i = 0; i < cfg->speaker_outs; i++) { stac92xx_reset_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); + pwr_state |= 1 << (cfg->speaker_pins[i] - 0xa); + } } else { /* enable lineouts, disable hp */ for (i = 0; i < cfg->line_outs; i++) @@ -2830,7 +2883,12 @@ static void stac92xx_hp_detect(struct hd for (i = 0; i < cfg->speaker_outs; i++) stac92xx_set_pinctl(codec, cfg->speaker_pins[i], AC_PINCTL_OUT_EN); - } + for (i = 0; i < cfg->hp_outs; i++) + pwr_state |= 1 << (cfg->hp_pins[i] - 0xa); + } + + /* power down unused ports */ + snd_hda_codec_write(codec, codec->afg, 0, 0x7ec, pwr_state); } static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res)

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[alsa-devel] snd_pcm_status_get_tstamp()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! The documentation for snd_pcm_status_get_tstamp() is not really enlightening. After a call to snd_pcm_status() what data does the tstamp field actually carry? Some kind of system clock timestamp when the data in the status struct was valid, or some kind of sound card clock timestamp that is relative to the point in time where the device was opened/started? Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Status of snd_pcm_rewind()
by Lennart Poettering 07 Jan '08

07 Jan '08

Hi! What's the status of snd_pcm_rewind()? Is it available on all devices? I assume not (at least dmix doesn't...). But I guess I can safely assume it is available when mmap is used? Is there any API function to test if it is available? I am working on adding "glitch-free" support to PulseAudio. I.e. have a huge playback buffer (> 2s), and schedule playback based on system timers instead of sound card interrupts, thus allowing us to dynamically adjust the inetrrupt frequency to the clients that are connected, and thus save power. I can enable this technique only if I can rewrite the playback buffer whenever I need to, thus I need to in advance if snd_pcm_rewind() will work or not. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Hello and providing ALSA support to sdl
by M-.-n 07 Jan '08

07 Jan '08

Hi there, My name is Marc and I'm writing a tracker software originally geared towards gameconsole (first GP32 but now GP2X, PC, Mac) and since I got this lovely eee, I'm planning on porting it to debian-derived platforms. Actually it works already but the SDL version I have uses oss and it seems to hickup sometimes although the cpu power consumption seems fine. Other programs (vlc/audacity) seems no to reflect this problem so I guess it's due to the fact they use ALSA while the SDL I got doesn't. When configuring SDL, it seems not to include ALSA (by not finding alsa/soundlib.h) and I'd like to add it but I'm a little confused on the step to follow. ALSA seems to be installed but I guess I miss the includes and compiling libs. Does it mean I need to configure/compile alsa-lib ? Do I risk corrupting my system if the installed libraries are not compatible ? Sorry for the rambling, but I'm only a beginner in terms of developping audio apps under linux. Regards, Marc.

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 07 Jan '08

07 Jan '08

-------------- Original message ---------------------- [...] > > I'd like to get this card's spdif working. What else can I do to help? > > Although the driver loads, the user space tools can't use the card any > > more. Thank you. > > > Should I assume at this point no more work will be done on the > snd-hda-intel driver to fixing SPDIF-in for the vt1708 sound card? > Thank you for all your work so far. > I think it is a bit early to make that assumption. There isn't a lot of demand for this particular card, but it doesn't seem that it will take too much work to get it going again... after all, it did basically work before the last patch, and I am hoping that Takashi will find some time soon to address the issues. As of this moment, I assume you're in the same boat I'm in, where the user space tools report an inappropriate ioctl for the device. -- Darryl

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[alsa-devel] Conexant CX2051 on HP Spartan
by Claudio Matsuoka 07 Jan '08

07 Jan '08

Hi, I'm working with Herton Krzesinski on a HP Spartan laptop which has a Conexant CX2051 configured with internal and external microphones connected to two different ADC nodes (0x14 and 0x15 in http://helllabs.org/tmp/hp-spartan-ng.png). We were able to get input from one or other ADC, but we weren't able to switch from one capture source to another (switching between 0x17 and 0x1d would be simple, but HP chose to route the external and internal mics to two different ADCs). What is the recommended way to handle this kind of setup?

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[alsa-devel] [PATCH] Another EMU0404 Board ID
by Veli-Matti Valtonen 07 Jan '08

07 Jan '08

Hey, This is based on pseudo-random playing around with the capabilities. With ca0102 this card gives no output atall, ca0108 appears to work fine, so it rather looks similar to the EMU1010b/EMU1010 changes. Some other people seem to have succeeded in using this aswell: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3496 diff -u -r alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c --- alsa-driver-hg20071222/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-22 03:00:09.000000000 +0200 +++ alsa-driver-hg20071222-0404b/alsa-kernel/pci/emu10k1/emu10k1_main.c 2007-12-23 23:01:12.807047391 +0200 @@ -1419,6 +1419,14 @@ .ca0102_chip = 1, .spk71 = 1, .emu_model = EMU_MODEL_EMU1010}, /* Emu 1010 */ + /* EMU0404b */ + {.vendor = 0x1102, .device = 0x0008, .subsystem = 0x40021102, + .driver = "Audigy2", .name = "E-mu 0404b [4002]", + .id = "EMU0404", + .emu10k2_chip = 1, + .ca0108_chip = 1, + .spk71 = 1, + .emu_model = EMU_MODEL_EMU0404}, /* EMU 0404 */ /* Tested by James(a)superbug.co.uk 20-3-2007. */ {.vendor = 0x1102, .device = 0x0004, .subsystem = 0x40021102, .driver = "Audigy2", .name = "E-mu 0404 [4002]",

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[alsa-devel] es18xx: Enable wavetable input from ESS chips
by Krzysztof Helt 07 Jan '08

07 Jan '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch enables wavetable chips ES689/ES69X connected to ESS ES18xx chips. The wavetable chip uses FM DAC if the clock signal from the wavetable is active. It has no effect if there is no ESS wavetable chip present. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch has a gotcha. In theory, the OPL synthesis or wavetable input should use FM DAC depending on the active clock signal from the wavetable chip. Unfortunately, the default state of the wavetable chip from ESS (ES689/ES690/ES692) is to send clock signal all the time. This blocks OPL3 signal if the wavetable chip is present. This problem can be solved with sysex command 4 send to the wavetable chip. It sets the wavetable chip to auto-deactivate mode which switches the wavetable off if there are 5 seconds without midi input (according to the es1879 datasheet). I tested this patch on the es1868 chip without the wavetable. The OPL3 synthesis worked (no clock signal from the wavetable). I tested it on the card with es1868 chip and es690 wavetable. The OPL3 synthesis did not work, the wavetable worked. Both the OPL3 and wavetable worked if the sysex command 4 was sent to the wavetable chip (e.g. by amidi tool). Is it ok to add this sysex command sending to the driver? How? The wavetable clock signal can be detected (there is status bit which says if wavetable is active). Another question is about ES938 chip (bass, treble and 3D control) on the card. This chip is controlled by sysex midi commands. It can be detected (by reading settings with sysex commands). Is it possible to add this to the alsamixer or external tool for setting these controls is recommended? Regards, Krzysztof diff -urp linux-2.6.24/sound/isa/es18xx.c linux-new/sound/isa/es18xx.c --- linux-2.6.24/sound/isa/es18xx.c 2007-12-25 23:23:03.000000000 +0100 +++ linux-new/sound/isa/es18xx.c 2007-12-26 16:30:08.000000000 +0100 @@ -1442,6 +1442,8 @@ static int __devinit snd_es18xx_initiali snd_es18xx_write(chip, 0xB2, 0x50); /* Enable MPU and hardware volume interrupt */ snd_es18xx_mixer_write(chip, 0x64, 0x42); + /* Enable ESS wavetable input */ + snd_es18xx_mixer_bits(chip, 0x48, 0x10, 0x10); } else { int irqmask, dma1mask, dma2mask; ---------------------------------------------------------------------- Rozdajemy nagrody! Wygraj samochod! Sprawdz >> http://link.interia.pl/f1cb8

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[alsa-devel] [PATCH] [ALSA-KERNEL] Add IEC958 digital out support to AD1984 for Lenovo Thinkpads T61/X61
by Jerone Young 07 Jan '08

07 Jan '08

This patch adds IEC958 digital out support for the AD1984 sound card. This card can be found in Lenovo Thinkapds T61/X61. The digital out is not located on the Thinkpad, but optional docking station (it's coxial digital out). I've add this support as it is done the exact same way for the AD1983 & AD1884. I have tested this patch with my Lenovo Thinkpad T61 hooked up to a docking station (that has the digital coxial) and then run to my home theater reciever. Works like a charm :-) Patch is attached and also in this email. Signed-off-by: Jerone Young <jerone(a)gmail.com> diff -r 1227a1c12325 pci/hda/patch_analog.c --- a/pci/hda/patch_analog.c Mon Dec 24 14:40:56 2007 +0100 +++ b/pci/hda/patch_analog.c Thu Dec 27 17:41:13 2007 -0600 @@ -3095,6 +3095,16 @@ static struct snd_kcontrol_new ad1984_th .get = ad198x_mux_enum_get, .put = ad198x_mux_enum_put, }, + /* SPDIF controls */ + HDA_CODEC_VOLUME("IEC958 Playback Volume", 0x1b, 0x0, HDA_OUTPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,NONE) "Source", + /* identical with ad1983 */ + .info = ad1983_spdif_route_info, + .get = ad1983_spdif_route_get, + .put = ad1983_spdif_route_put, + }, { } /* end */ }; @@ -3197,7 +3207,7 @@ static int patch_ad1984(struct hda_codec codec->patch_ops.build_pcms = ad1984_build_pcms; break; case AD1984_THINKPAD: - spec->multiout.dig_out_nid = 0; + spec->multiout.dig_out_nid = AD1884_SPDIF_OUT; spec->input_mux = &ad1984_thinkpad_capture_source; spec->mixers[0] = ad1984_thinkpad_mixers; spec->init_verbs[spec->num_init_verbs++] = ad1984_thinkpad_init_verbs;

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[alsa-devel] [PATCH] hda: STAC927x VREF fix
by Matthew Ranostay 07 Jan '08

07 Jan '08

Some laptops incorrectly assume the front input jack as a line in instead of a microphone in. Which in turn disables the voltage reference, in which non-amplified input is not possible. This patch enables VREF80 for the input jack. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r eeed28ed6b7d pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Jan 04 09:22:20 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Jan 04 19:41:06 2008 -0700 @@ -3344,6 +3344,7 @@ static int patch_stac927x(struct hda_cod spec->mixer = stac927x_mixer; break; case STAC_DELL_BIOS: + stac92xx_set_config_reg(codec, 0x0e, 0x02a79130); case STAC_DELL_3ST: /* GPIO2 High = Enable EAPD */ spec->gpio_mask = spec->gpio_data = 0x00000004;

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[alsa-devel] ICE1724 chipset and envy24control
by Norval Watson 07 Jan '08

07 Jan '08

Hi, I just got an M-Audio Audiophile 192 soundcard with ICE1724 chipset. Works out of the box on debian sid. Is there any way to control this card with envy24control utility from alsa-tools? envy24control is written for ICE1712 chipset. Is there any project to develop control utility for ICE1724 card? Seasons greetings, Norval Watson Make the switch to the world's best email. Get the new Yahoo!7 Mail now. www.yahoo7.com.au/worldsbestemail

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[alsa-devel] python midi wrapper
by Aldrin Martoq 07 Jan '08

07 Jan '08

Hello, I need a midi/seq wrapper for Python. I've already created some code for the sequencer API and works great! Here is my python test code: -------------------------------------------------------------------- #!/usr/bin/env python import sys sys.path.append('build/lib.linux-i686-2.5') import alsaaudio print "Creating Sequencer..." seq = alsaaudio.Seq("hw", "Python seqtest.py", alsaaudio.SEQ_OPEN_DUPLEX, alsaaudio.SEQ_NONBLOCK) myid = seq.get_client_id() print "... seq:", myid, seq print "Creating Port..." myport = seq.create_simple_port("Processor port", alsaaudio.SEQ_PORT_CAP_WRITE | alsaaudio.SEQ_PORT_CAP_SUBS_WRITE, alsaaudio.SEQ_PORT_TYPE_APPLICATION) print "... port:", myport #queue = seq.create_queue() #print "... queue:", queue # dump clients and ports def dump_connection(): print print "Client/ports available:" for client in seq.get_client_ports(): clientname, clientid, portlist = client print " client: %3d %s" % (clientid, clientname) for port in portlist: portname, portid, connections = port print " port: %3d:%-2d +-%s" % (clientid, portid, portname) readconn, writeconn = connections for conn in readconn: print " connection to: %s" % conn for conn in writeconn: print " connection from: %s" % conn dump_connection() Is the front seq.connect_ports("0:1", "%d:%d" % (myid, myport)); seq.connect_ports("24:0", "%d:%d" % (myid, myport)); dump_connection() while True: events = seq.receive_events(60000) for ev in events: print ev print "\t", ev.data() -------------------------------------------------------------------- I have tried to contact the mantainer of PyAlsaAudio (cwi(a)unispeed.dk from the README file) for merging my code in his project, but have not received a response so far and the date of files seems old. Is anybody here interested in the merge? How can I put this effor in python-alsaaudio or may I create a new project for this specific task? Is the pyalsa project related to pyalsaaudio? Thanks, -- Aldrin Martoq

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[alsa-devel] function snd_mixer_load failed: Inappropriate ioctl for device
by Salatiel Filho 07 Jan '08

07 Jan '08

*** please let me in the to or cc fields. I am not on the list *** I am having a big trouble to make sound work on my ARM machine. I am running debian 4. # uname -a Linux LS-GL7D6 2.6.16.57-arm2 #7 Wed Dec 12 18:14:51 BRT 2007 armv5tejl GNU/Linux # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. Modules load successfully # lsmod Module Size Used by snd_pcm_oss 47680 0 snd_mixer_oss 15040 1 snd_pcm_oss snd_usb_audio 79904 0 snd_pcm 75877 2 snd_pcm_oss,snd_usb_audio snd_timer 20164 1 snd_pcm snd_page_alloc 8264 1 snd_pcm snd_usb_lib 15808 1 snd_usb_audio snd_rawmidi 21376 1 snd_usb_lib snd_seq_device 7116 1 snd_rawmidi snd_hwdep 7460 1 snd_usb_audio snd 54368 9 snd_pcm_oss,snd_mixer_oss,snd_usb_audio,snd_pcm,snd_timer,snd_usb_lib,snd_rawmidi,snd_seq_device,snd_hwdep soundcore 7684 1 snd nfs 102044 0 lockd 61780 1 nfs sunrpc 142808 3 nfs,lockd dm_mod 48176 0 usbhid 38628 0 # cat /proc/asound/cards 0 [default ]: USB-Audio - C-Media USB Headphone Set C-Media USB Headphone Set at usb-ehci_platform.4523-1, full speed # cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.15. Compiled on Dec 12 2007 for kernel 2.6.16.57-arm2. but everytime i try to use alsamixer i get: # alsamixer alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device # aplay test.wav Playing WAVE 'test.wav' : Unsigned 8 bit, Rate 8000 Hz, Mono aplay: xrun:1090: read/write error, state = PREPARED # strace alsamixer ... stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/usr//share/alsa/alsa.conf", O_RDONLY) = 3 fstat64(3, {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 read(3, "#\n# ALSA library configuration "..., 4096) = 4096 read(3, "r\n\t\t\t\tname defaults.pcm.device\n\t"..., 4096) = 4096 read(3, "{\n\t\t\t@func refer\n\t\t\tname default"..., 4096) = 419 read(3, "", 4096) = 0 read(3, "", 4096) = 0 close(3) = 0 munmap(0x40017000, 4096) = 0 access("/etc/asound.conf", R_OK) = -1 ENOENT (No such file or directory) access("/root/.asoundrc", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da64) = 0 ioctl(3, UI_DEV_CREATE, 0xbec2db2c) = 0 close(3) = 0 stat64("/usr//share/alsa/alsa.conf", {st_mode=S_IFREG|0644, st_size=8611, ...}) = 0 open("/dev/snd/controlC0", O_RDONLY) = 3 close(3) = 0 open("/dev/snd/controlC0", O_RDWR) = 3 ioctl(3, USBDEVFS_CONTROL, 0xbec2da44) = 0 fcntl64(3, F_GETFL) = 0x2 (flags O_RDWR) fcntl64(3, F_SETFL, O_RDWR|O_NONBLOCK) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_RELEASEINTERFACE, 0xbec2dac0) = 0 ioctl(3, USBDEVFS_CONNECTINFO, 0xbec2d86c) = 0 ioctl(3, USBDEVFS_IOCTL, 0xbec2d42c) = -1 ENOTTY (Inappropriate ioctl for device) fstat64(1, {st_mode=S_IFCHR|0620, st_rdev=makedev(136, 2), ...}) = 0 mmap2(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40017000 write(1, "\n", 1 ) = 1 write(2, "alsamixer: function snd_mixer_lo"..., 74alsamixer: function snd_mixer_load failed: Inappropriate ioctl for device ) = 74 munmap(0x40017000, 4096) = 0 io_submit(0x2, 0, 0x8 <unfinished ... exit status 2> any help? The same card works ok on my x86 machine using . So the problem is not the card but something else. -- []'s Salatiel "O maior prazer do inteligente é bancar o idiota diante de um idiota que banca o inteligente".

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[alsa-devel] Test music sampled at 192K
by Jon Smirl 06 Jan '08

06 Jan '08

Can anyone point me to some uncompressed test music/tones sampled at 192K/sec? -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] 5.1 surround--no front center speaker sound--[NVidia CK804]
by Steve Thornley 05 Jan '08

05 Jan '08

Nforce 4 MSI motherboard integrated sound----since my Creative X-Fi isn't supported, YET can not get the front center speaker to work. jlt:/home/steven# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: CK804 [NVidia CK804], device 0: Intel ICH [NVidia CK804] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: CK804 [NVidia CK804], device 2: Intel ICH - IEC958 [NVidia CK804 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 Here is my present asound.conf file that I created in /etc for my Debian etch 4.0----amd64 system X2-4200-cpu # Set default sound card # Useful so that all settings can be changed to a different card here. pcm.snd_card { type hw card 0 } pcm.dmix6 { type dmix ipc_key 1024 ipc_key_add_uid false # let multiple users share ipc_perm 0660 # IPC permissions (octal, default 0600) slave { pcm snd_card # see below rate 48000 channels 6 period_time 0 period_size 1024 buffer_time 0 buffer_size 4096 } } #route for stereo only pcm.dmix20 { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 } #route for stereo sound+center+subwoofer pcm.ch31dup { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } pcm.ch51dup { type route slave.pcm dmix6 slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 0.5 ttable.1.4 0.5 ttable.0.5 0.5 ttable.1.5 0.5 } pcm.duplex { type asym playback.pcm "ch51dup" # playback.pcm "dmix2" # capture.pcm "mydsnoop" capture.pcm "snd_card" } ################### # CONVERSION PLUG # ################### # Setting the default pcm device allows the conversion # rate to be selected on the fly. # duplex mode allows any alsa enabled app to read/write # to the dmix plug (Fixes a problem with wine). pcm.!default { type plug slave.pcm "duplex" } pcm.dsp "duplex" pcm.dsp1 "duplex" pcm.fr { type route slave.pcm "dmix6" slave.channels 6 ttable.0.0 1 ttable.1.1 1 } pcm.rr { type route slave.pcm "dmix6" slave.channels 6 ttable.0.2 1 ttable.1.3 1 } pcm.lfe { type route slave.pcm "dmix6" slave.channels 6 ttable.0.4 1 ttable.1.5 1 } Here is my back up with other configurations for the file that created the same 4 speakers and subwoofer sounds but no front center speaker sounds pcm.ch51dup { type route slave.pcm dmix6 slave.channels 6 ttable.0.0 1 ttable.1.1 1 ttable.0.2 1 ttable.1.3 1 ttable.0.4 1 ttable.1.4 1 ttable.0.5 1 ttable.1.5 1 } ########Old Setup # pcm.51to40 # { # type route # slave.pcm surround51 # slave.channels 6 # # # Front and rear, at 75% of original signal strength # ttable.0.0 0.75 # ttable.1.1 0.75 # ttable.2.2 0.75 # ttable.3.3 0.75 # # Center channel routing (routed to front-left and front-right), # 6dB gaindrop (gain half of main channels) per channel # ttable.4.0 0.375 # ttable.4.1 0.375 # LFE channel routing (routed to front-left and front-right), # 6dB gaindrop (gain half of main channels) per channel # ttable.5.0 0.375 # ttable.5.1 0.375 # } When I changed a setting from Autodetect to ALSA I could not get any sound--so I compiled everything from source with the latests tars and still nothing, and once I turned the Autodetect in Multimedia Systems Selector<Preferences<Desktop for the Default output plugin, I still had no sound in my center front speaker. Whew. I'd love it for my Creative X-Fi card to be supported, but right now, I really like my front center speaker back. I'm at the end of my list of possible fixes, maybe someone who understands can take a look at this an have a go ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

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[alsa-devel] fujitsu siemens amilo la 1703 vt1807 hda intel card problem
by Norayr Chilingaryan 05 Jan '08

05 Jan '08

Hello I decided to write here, firtly, because my messages in aksa-user have been ingored, and secondly, because I understood that my card most probably just unsupported and I must contact developers. I have tried to pass options model=fujitsu, because I have a fujitsu siements notebook etc I have also tried to pass amny other options in /etc/modules.d/alsa but withnot effect I use gentoo, and the only thing I noticed with latest kernels and alsa drivers 1.0.15 is that I can hear sound through the headphones, but with low volume. It is not possible to get volume higher or speaker to make sounds now. So, please, this is information about the card 04:01.0 Audio device: VIA Technologies, Inc. VIA High Definition Audio Controller (rev 10) kakraphoon ~ # cat /proc/asound/card0/codec#* | grep Codec Codec: VIA VIA VT1708 Codec: Motorola Si3054 Thank you very much Norayr

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Re: [alsa-devel] [PlanetCCRMA] drop outs with FC8 and latest CCRMA kernel
by John Dey 04 Jan '08

04 Jan '08

Hector, I have read your response and can only comment that I have been a user of CCRMA's version of Fedora for some time. I am not responding to place my recommendation for that distribution over Ubuntu since I only tried Ubuntu early on when it was not stable. I would like to offer my observation on what I have felt for a long time regarding to multimedia (sound) on linux. What has made linux OS so successful is that it has a legion of volunteers willing to contribute to the cause of advancing the state of the OS for free but even more importantly, in my opinion, is the fact that Linus Torvalds has been the leader to keep those volunteers motivated while controlling/coordinating the development of the OS. He and others (what were very productive volunteers) have even figured ways to make a living at what they have done so well for free. I am unaware of any such individual (or organization) functioning similarly for multimedia-sound). My reason for choosing CCRMA and staying with it is because Fernando Lopez-Lezcano has been available, almost at moments notice, to answer and solve most of problems that I have encountered. (I don't know how he has the patients and stamina to do what he does.) I do get frustrated that I can not get my ATI graphics card to work and have to settle for a less then optimal driver for video but my major concern recently, and maybe I am wrong, is the state of soundcard drivers available for linux (particularly notebooks). I have had good luck with desktops and m-audio soundcards but with notebooks I don't have any comfort that there are cards that fully work. (This comment maybe too harsh.) Without reliable soundcard drivers, I worry that the whole elegant alsa sound system and sound applications my never achieve their full potential and may just wither and die. I have heard some of the arguments that vendors have been non-cooperative. I do believe that there should be ways to structure an approach that would give the linux/sound community leverage over the vendor in order to get their cooperation, e.g. a non-profit organization with support from colleges and university music departments. I would like to hear if I am all wet. However, before we get too far into the new year, I had to get these thoughts off my chest. Happy New Year? And thanks to all who contribute. John Dey On Jan 4, 2008, at 10:12 AM, Hector Centeno wrote: > Hi again, > > > On Jan 3, 2008 9:04 PM, Lamar Owen <lowen(a)pari.edu> wrote: >> On Thursday 03 January 2008, bH wrote: >>> Lamar Owen wrote: >> >> The point is, once built for Ubuntu 7.10, it has to be rebuilt for >> 8.04, then >> 8.10 (or whatever), and so on; you don't gain anything going to >> Ubuntu, as it >> is just as fast moving as Fedora is in terms of support; the repo >> would still >> have to be rebuilt every six months, as packages for 7.04 may or may >> not work >> for 7.10 (and packages for 6.06 will definitely not work for 7.10 or >> the >> upcoming 8.04). >> > > I think I disagree about this. You would gain a lot going to Ubuntu > instead of Fedora. It is true that Ubuntu gets upgraded every 6 months > (as Fedora does) but in Ubuntu you have the hard work of many people > trying to make Linux usable and friendly and at the same time to keep > it up to date, while in Fedora you get a bunch of experimental and > broken things thrown to the users for beta testing. For instance, > something like having and ATI graphics card working, with full 3D > rendering, under Fedora 8 is almost impossible, while in Ubuntu works > out of the box, or if you want to install the Nvidia drivers in Ubuntu > you just get them from the repository or better yet, just click on the > proprietary drivers icon that pops up at the panel bar when it detects > your Nvidia hardware. I know I could just download the sources and > compile myself... but well, it saves time to have things planed ahead > by someone who cares about usability. > > Regarding the upgrading issue, I do mind having the latest version of > Gnome, for example, as it usually means less bugs and better > usability. As I said before, I think Linux-Audio and > Linux-as-a-desktop is quite young and in development, you can't really > get stuck with a buggy and underdeveloped application for too long as > you might be missing a stabler and more usable system. > > Just my thoughts. > > Cheers! > > Hector > > > >> But all of that is moot; PlanetCCRMA is currently an RPM repository, >> and the >> effort required for changing to be a Debian repository is significant; >> probably a lot harder than keeping up with Fedora (this is something >> I know a >> little bit about; I packaged PostgreSQL RPM's for Red Hat Linux and >> Fedora >> from 1999 up through 2004; I investigated Debian packaging, and it is >> very >> very different from RPM packaging. Now, nando is a very bright >> fellow; I >> have no doubts he is capable to doing that. The bigger question is >> whether >> he wants to do so. >> > > > > >> Further, there are already Debian packagers for some of this; you >> should >> really talk to them about updating their versions, rather than muddy >> the >> currently well-unified repository waters for Ubuntu. That is really >> the >> major advantage over Fedora that Ubuntu has; Fedora's repositories >> have >> historically been somewhat fractured. >> >> However, there is work ongoing to merge PlanetCCRMA RPM's into Fedora >> itself, >> which would lessen nando's work load. This I'm sure he looks forward >> to! >> >> Now, the latency could be due to other things set up in the >> PlanetCCRMA kernel >> and other libs versus the Ubuntu linux-rt kernel; don't know. It >> would be >> interesting to see what the .config for the Ubuntu rt kernel looks >> like >> versus the PlanetCCRMA one. >> >> I myself am having issues with the F8 PlanetCCRMA rt kernel; I'm >> running F7 on >> my main DAW desktop for now, but will be upgrading probably next >> week, as >> things are stable enough at this point for my uses. I'll probably >> forgo the >> rt kernel for now, as my uses currently don't require the low latency. >> >> I for one am thrilled with what nando has been doing for the last >> several >> years; many thanks, nando! >> -- >> >> Lamar Owen >> Chief Information Officer >> Pisgah Astronomical Research Institute >> 1 PARI Drive >> Rosman, NC 28772 >> (828)862-5554 >> www.pari.edu >> >> _______________________________________________ >> PlanetCCRMA mailing list >> PlanetCCRMA(a)ccrma.stanford.edu >> http://ccrma-mail.stanford.edu/mailman/listinfo/planetccrma >> > > _______________________________________________ > PlanetCCRMA mailing list > PlanetCCRMA(a)ccrma.stanford.edu > http://ccrma-mail.stanford.edu/mailman/listinfo/planetccrma >

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[alsa-devel] posting
by Steve Thornley 04 Jan '08

04 Jan '08

What is appropriate. Especially, right now I'm trying to get my 5.1 system to work as something other than a 2 speaker set. looking for help, compiled from source last night ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[alsa-devel] Cross compiling alsa-lib
by Adrian McMenamin 04 Jan '08

04 Jan '08

Is there anyone here able to give a canonical statement on how to configure alsa-lib for a cross compiled build linked to uclibc? I think the issue may be getting the path right for the libraries on the host machine.

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by Takashi Iwai 02 Jan '08

02 Jan '08

At Thu, 20 Dec 2007 10:13:29 -0800, John Stile wrote: > > On Thu, 2007-12-20 at 12:09 +0100, Takashi Iwai wrote: > > At Thu, 20 Dec 2007 00:37:46 +0000, > > forumjunk(a)comcast.net wrote: > > > > > > > > > For VIA platforms, to avoid the driver compile error in pci.h, reconfigure your kernel to remove SMP support, and set the architecture type to PC-compatible. > > > > > > (you already know this, I'm sure): > > > Recompile the kernel (as a paranoid, I always trash /lib/modules/<kernel_version> before making modules_install). Install the kernel, of course, and reconfigure your boot loader and/or initrd and reboot. > > > > > > There are still several fixes that need to be made to get all of this to compile cleanly, but I haven't got the permanent fixes nailed down yet. I'm almost positive that these are alsa-driver configure errors or omissions. I can't (yet) find a kernel config that will avoid them. The fixes below are harmless, and they will allow you to get the driver compiled. > > > > My guess is that the error of configure script is the only point we > > need to fix. > > > > > Add this definition to the top section of acore/memory_wrapper.c after the line that reads #include <linux/version.h> > > > > > > #define VMALLOC_VMADDR(x) ((unsigned long)(x)) > > > > > > Change these lines (45 and 46 or close to that) of acore/memory_wrapper.c from: > > > > > > pmd = pmd_offset(pgd, lpage); > > > pte = pte_offset(pmd, lpage); > > > > > > to > > > > > > pmd = pmd_offset((pud_t *)pgd, lpage); > > > pte = pte_offset_kernel(pmd, lpage); > > > > > > The driver will then compile cleanly (at least with no errors or warnings relating to any of these changes) > > > > Don't do this. This part shouldn't be compiled at all for recent > > kernels. This is, again, the problem of configure script that > > detected the codes wrongly. > > > > > > So, could you guys show config.log? Otherwise it's hard to know > > what's really wrong. > > > > > > thanks, > I recompiled the kernel without SMP support, and as PC-compatible (per > forumjunk(a)comcast.net). In config.log, vmalloc_to_page shows a warning > starting at line 540. "function cpuid_count causes a warning: pointer > targets in passing argument 4 of 'native_cpuid' differ in signedness". > --------------------------------------- > The config.log Thanks. Does the patch below fix the probelm? Run hgcompile after applying it. Takashi diff -r 414b1156a8b3 configure.in --- a/configure.in Tue Dec 18 16:09:20 2007 +0100 +++ b/configure.in Fri Dec 21 12:10:53 2007 +0100 @@ -1230,7 +1230,7 @@ esac dnl set ia32 (X86) case "$processor" in - i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*) + i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*|x86_32) if $KCC -mpreferred-stack-boundary=2 -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then c_opts="-mpreferred-stack-boundary=2 $c_opts" CONFIG_X86=y

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[alsa-devel] ALSA HDA problem with 64-bit kernels
by Timo Jantunen 01 Jan '08

01 Jan '08

Heip! I recently moved to 64-bit kernel, but mixed sounds (ie. mpegs played with a media player) don't work with optical S/PDIF connection. AC3/DTS passthrough still works, and I can hear the mixed sounds on analog connector. And yes, I have unmuted the IEC958 mixer setting. Tested kernels: ALSA 1.0.14: 2.6.23.1 32bit: works 2.6.23.1 64bit: broken 2.6.23.12 64bit: broken 2.6.23.12 64bit kernel, 32bit userspace: broken ALSA 1.0.15: 2.6.24-rc6 64bit: broken (and half of the mixer settings are missing) The soundcard is integrated to the motherboard (Abit IP35 Pro) and I can't find any more information on it than it is "7.1 CH HD Audio CODEC". Relevant part of dmesg (it is the same for 32/64 bit versions, expect that the interrput goes to IRQ 18): ===cut dmesg Advanced Linux Sound Architecture Driver Version 1.0.14 (Fri Jul 20 09:12:58 2007 UTC). ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 22 PCI: Setting latency timer of device 0000:00:1b.0 to 64 hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... usbcore: registered new interface driver snd-usb-audio ALSA device list: #0: HDA Intel at 0xfdff4000 irq 22 ===cut lspci 00:1b.0 Audio device [0403]: Intel Corporation 82801I (ICH9 Family) HD Audio Controller [8086:293e] (rev 02) Subsystem: ABIT Computer Corp. Unknown device [147b:1083] Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at fdff4000 (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 (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset+ DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel Capabilities: [130] Root Complex Link ===cut asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfdff4000 irq 22 ===cut There are no differences in /proc/asound directory entries expect the cards file (different IRQ). Any ideas how to fix this? //T

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[alsa-devel] Getting the AICA device to work properly
by Adrian McMenamin 31 Dec '07

31 Dec '07

I have built mpg123 for SH4 (using uclibc for a buildroot/busybox environment): ./configure --host=sh4-linux CC="/home/adrian/buildroot/build_sh4/staging_dir/bin/sh4-linux-uclibc-gcc" --with-cpu=generic_fpu --prefix=/tftpboot/192.168.61.55/usr LDFLAGS="-L/tftpboot/192.168.61.55/lib" --oldincludedir=/tftpboot/192.168.61.55/usr/include -with-audio=alsa --sysconfdir=/tftpboot/192.168.61.55/etc --enable-modules=no --enable-static --enable-debug When I try to play sound I get this: $ mpg123 -v ms.mp3 [getlopt.c:115] debug: getsingleopt: -v [getlopt.c:115] debug: getsingleopt: ms.mp3 [audio.c:501] debug: opening normal audio/file [libmpg123.c:190] debug: set flags to 0x40 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layers 1, 2 and 3 version 1.0.1; written and copyright by Michael Hipp and others free software (LGPL/GPL) without any warranty but with best wishes [frame.c:464] debug: frame_gapless_init: from 0 to 0 samples [libmpg123.c:99] debug: cpu opt setting Decoder: generic [frame.c:141] debug: frame 0x435650 buffer [frame.c:242] debug: frame 0x435650 buffer done [tabinit.c:85] debug: MMX decode tables [tabinit.c:107] debug: MMX decode tables done [audio.c:279] debug: audio_capabilities ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for '(null)' [mpg123.c:813] debug: Going to play ms.mp3 [mpg123.c:483] debug: OK... going to finally open. [frame.c:464] debug: frame_gapless_init: from 0 to 0 samples [readers.c:620] error: Cannot file ms.mp3: No such file or directory [mpg123.c:495] error: Cannot open ms.mp3: File access error. (code 22) [playlist.c:116] debug: going to free() the playlist [playlist.c:120] debug: free()ing entry 0 [playlist.c:126] debug: free()d the playlist [audio.c:441] debug: exit output [audio.c:524] debug: closing output [audio.c:148] debug: closing output module The main problem appears to be that ALSA lib has no knowledge of the installed (snd-aica) sound module - eg $ aplay -L null Discard all samples (playback) or generate zero samples (capture) How do I fix this? I don't think I had this problem before because my alsalib set up was somewhat banjaxed - things like aplay segfaulted - but the oss smulation layer worked well. When I use the emulation layer now (rebuilding mpg123) I now get: $ mpg123 ./ms.mp3 [getlopt.c:115] debug: getsingleopt: ./ms.mp3 [audio.c:501] debug: opening normal audio/file [oss.c:180] error: Can't open default sound device! [audio.c:505] error: failed to open audio device [mpg123.c:666] error: Failed to initialize output, goodbye.

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[alsa-devel] Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)
by Michael Kühn 29 Dec '07

29 Dec '07

Hey folks, i'm the owner of a Acer 6492 Notebook. Details of this soundchip are at the bottom of this e-mail. I want to support you for developing a version of ALSA that support this type of intel HDA audio device. So tell me, how i can help you. (Maybe with testing developer version?). My OS is Gentoo Linux, and i used the latest ALSA driver (just unmask the alsa-driver package). Hope it's ok. :) Thanks for reply. P.S.: Details of the audiochip can found here: http://nopaste.com/p/aS3FUn5pqb

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[alsa-devel] ALC268 "test" model
by Marco Schuster 28 Dec '07

28 Dec '07

Hi all, I try to use a patch from J. Woithe to put a test model into the ALC268 driver [1] (enabling all pins etc.); unfortunately it gives a lot of compiling errors ([2]) which I am unable to understand as I don't have sufficient knowledge of the internal API of HDA codec. Can someone please look over it and correct the mistakes? Marco Schuster [1]: http://harddisk.is-a-geek.org/patch.txt [2]: http://harddisk.is-a-geek.org/compiling-errors.txt

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[alsa-devel] snd-hda-intel driver on sony VGN TZ11MN
by Jonathan Coe 26 Dec '07

26 Dec '07

The sound appears to play using audio players but no sound comes out of the speakers. Plugging in headphones and removing them leads to sound coming out of the speakers. This behaviour can be corrected by passing suitable parameters to the driver module. _From lspci: Subsystem: 104d:900e Arguments added to /etc/modprobe.d/sound: options snd-hda-intel model=hippo I hope this is all the information you need to get this working. If not please email and I'll be happy to help if I can. kind regards, Jon

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[alsa-devel] doc update for http://bugtrack.alsa-project.org/main/index.php/Matrix:Vendor-Intel
by Paul Winkler 24 Dec '07

24 Dec '07

Hi, The page at http://bugtrack.alsa-project.org/main/index.php/Matrix:Vendor-Intel makes it seem that cards identified as ICH7 must use the hda_intel driver. This is apparently not always true. I have a dell desktop where snd_hda_intel refuses to load. I fired up alsaconf and it suggested using the intel8x0. I tried that and it works fine. So maybe ICH7 should be added to "chipsets" in the first row on that page. Thanks, - Paul Winkler p.s. here's some system info: pw@pw ~ $ lspci | grep audio 00:1e.2 Multimedia audio controller: Intel Corporation 82801G (ICH7 Family) AC'97 Audio Controller (rev 01) pw@pw ~ $ cat /proc/asound/cards 0 [ICH7 ]: ICH4 - Intel ICH7 Intel ICH7 with AD1981B at irq 23 pw@pw ~ $ lsmod | grep snd snd_intel8x0 27176 0 snd_ac97_codec 90456 1 snd_intel8x0 ac97_bus 4224 1 snd_ac97_codec snd_pcm 60808 2 snd_intel8x0,snd_ac97_codec snd_timer 18440 1 snd_pcm snd 38192 4 snd_intel8x0,snd_ac97_codec,snd_pcm,snd_timer snd_page_alloc 8976 2 snd_intel8x0,snd_pcm

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Re: [alsa-devel] [PATCH]Device ID for Toshiba laptop which uses AD1986A
by Jiang zhe 24 Dec '07

24 Dec '07

On Fri, 2007-12-21 at 21:39 +0200, Andy Shevchenko wrote: > Hi Jiang zhe! > > On Thu, Dec 20, 2007 at 09:44:14AM +0800, Jiang zhe wrote next: > > > @@ -859,6 +859,7 @@ static struct snd_pci_quirk ad1986a_cfg_ > > SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), > > SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), > > SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), > > + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), > > {} > > }; > Please, keep list sorted by PCI SSID. > Thank you very much! ^_^ --- alsa-driver-hg20071224/alsa-kernel/pci/hda/patch_analog.c 2007-12-21 09:00:06.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_analog.c 2007-12-24 10:07:54.000000000 +0800 @@ -848,6 +848,7 @@ static struct snd_pci_quirk ad1986a_cfg_ SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x1043, 0x8234, "ASUS M2N", AD1986A_3STACK), SND_PCI_QUIRK(0x10de, 0xcb84, "ASUS A8N-VM", AD1986A_3STACK), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), SND_PCI_QUIRK(0x144d, 0xb03c, "Samsung R55", AD1986A_3STACK), SND_PCI_QUIRK(0x144d, 0xc01e, "FSC V2060", AD1986A_LAPTOP), SND_PCI_QUIRK(0x144d, 0xc023, "Samsung X60", AD1986A_LAPTOP_EAPD), @@ -859,7 +860,6 @@ static struct snd_pci_quirk ad1986a_cfg_ SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), - SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), {} };

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[alsa-devel] Gadget Labs Linux Alsa driver is coming!! (and you are invited)
by Mike Mazarick 24 Dec '07

24 Dec '07

There is a recently revitalized project to create an alsa linux driver for the Gadget Labs series of cards. http://sourceforge.net/projects/wavepro-driver/ There is an active mailing list of the developers, testers, and interested parties who are helping located here: wavepro-driver-developers(a)lists.sourceforge.net It is recommended that if you are interested in reading the progress of this effort, that you subscribe soon because the mailing list will go 'private' around the middle of January (to keep out spam and other unwanted intrusions). The end goal of this project is to have the driver included in the alsa distribution and eventually into the linux distributions so that owners of this card can simply have the card installed in their system and all the necessary steps will be done automatically by the installation, as is the case with SoundBlasters, onboard sound cards, Delta 1010s, RME 9652s, etc. Having a working driver is several months away from now, and the developers who are helping write this are quite good, however, we are all inexperienced at creating drivers for linux and are unsure of the needed steps to get it included in the Alsa tree. We are also very receptive to any advice or help with the creation of this driver; feel free to sign up and be a part of this project. We have a major advantage because there is already a Windows driver available with source (including new FPGA firmware), which was developed after the Gadget Labs company went under. Best wishes (and happy holidays!!) Mike Mazarick

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[alsa-devel] [lennart@poettering.net: Status of ALSA "simple" mixer interface]
by Lennart Poettering 23 Dec '07

23 Dec '07

Hi! A few days ago I sent the attached emails to Takashi and filed a couple of bugs on the ALSA BTS. Takashi suggested to post these issues on the ML as well, to allow public discussion. So, here we go. The relevant bugs are: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3570 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3569 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3568 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3572 https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3571 Lennart

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[alsa-devel] [PATCH] PulseAudio plugin: report XRUN state back to application
by Lennart Poettering 23 Dec '07

23 Dec '07

Hi! Please consider merging this patch we ship in Fedora: http://cvs.fedoraproject.org/viewcvs/*checkout*/devel/alsa-plugins/1.0.14-s… It adds support to report back XRUN to the application if one happens. This is required to make some applications work on top of the pulse plugin. One being XMMS, which checks if a song finished to play by waiting for an XRUN (yes, I don't argue that XMMS shouldn't do that, but nonetheless it is a good thing if XRUNs are reported properly.) Now, this patch is a bit problematic, since it touches pcm->io.state which is not supposed to be touched by ioplug-based plugins -- according to the comments in ioplug. However, I saw no other way to fix this, and from judging the ioplug code overwriting the variable should be safe. Thus I decided to simply touch the variable but documented it in a comment. Alternatively, ioplug could be beefed up to properly handle XRUNs, but I thought this fix is simpler. Also reported on the ALSA BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3579 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] write/read system calls and ioctl
by Michael Voigt 22 Dec '07

22 Dec '07

Dear ALSA developers, At the beginning, I would like to give a short introdution of myself, then my question follows. My project: =============== Currently, I am interning at the TU Dresden operating systems group's [1] and it is my task to port ALSA to DROPS [2, 3], a L4-microkernel-based real-time system. The long-term goal of this project should be to establish a truely real-time capable FOSS platform for audio recording and processing. The current state of the project: By using DDELinux [4] and writing a small kernel-user inferface emulation layer, we now have an unchanged version of ALSA and an unchanged version of salsa-lib [5] running on our system. The next steps will be to test how stable this port is and fix bugs. Then, during my diploma thesis, it is my task to port Jack (or an equivalent) to our system and to evaluate how to make the whole thing to work in real-time. But now my question: =============== In the file pcm_native in alsa-core read (snd_pcm_read) and write (snd_pcm_write) functions are attached to the snd_pcm_f_ops. But as far as I can see, these write- und read- system calls aren't used either in salsa-lib or in alsa-lib. I can see that in former versions of alsa-lib (for instance alsalib-0.0.3) the write system call was used to implement snd_pcm_write (LIB); but now we have snd_pcm_writei (LIB) and snd_pcm_writen (LIB) which use ioctl. I guess this was done to realize both interleaved and noninterleaved audio playback (The old snd_pcm_write (LIB) was based on bytes not on frames and did not have a notion of interleaved/noninterleaved playback?). If I have a look at snd_pcm_write (KERNEL, pcm_native.c), I can see it also ends up in snd_pcm_lib_write(...). So does ioctl with SNDRV_PCM_IOCTL_WRITEI_FRAMES as argument, but snd_pcm_write (KERNEL) does not copy the frames to kernel memory. My question: what is snd_pcm_write in pcm_native.c good for? Is it only there to keep ALSA compatible to older versions of alsa-lib? Kindest regards, Michael Voigt ================ [1] http://www.inf.tu-dresden.de/index.php?node_id=1141&ln=en [2] http://www.osnews.com/story.php/15814/Introduction-to-TUD-OS/ [3] http://www.realtimelinuxfoundation.org/variants/variants.html#VARIANTS_DROPS [4] http://demo.tudos.org/dsweeper_tutorial.html [5] http://ftp.isr.ist.utl.pt/pub/MIRRORS/ftp.suse.com/people/tiwai/salsa-lib/ ================ -- Today is Pungenday, the 61st day of The Aftermath in the YOLD 3173

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 22 Dec '07

22 Dec '07

Well, I spoke too soon... with or without the via patch, I now get an error from the driver about an inappropriate ioctl for the device while KDE is trying to start the sound system. On a command line, madplay and aplay return this: #> aplay /usr/share/sounds/KDE_Glass_Break.ogg #> ALSA lib pcm_dmix.c:866:(snd_pcm_dmix_open) unable to open slave #> aplay: main:546: audio open error: Device or resource busy #> aplay /usr/share/sounds/pop.wav #> ALSA lib pcm_dmix.c:866:(snd_pcm_dmix_open) unable to open slave #> aplay: main:546: audio open error: Device or resource busy #> madplay /usr/share/sounds/KDE_Glass_Break.ogg #> MPEG Audio Decoder 0.15.2 (beta) - Copyright © 2000-2004 Robert Leslie et al. #> audio: /dev/dsp: Device or resource busy #> madplay /usr/share/sounds/pop.wav #> MPEG Audio Decoder 0.15.2 (beta) - Copyright © 2000-2004 Robert Leslie et al. #> audio: /dev/dsp: Device or resource busy alsaconf correctly detects the VT1708, and alsamixer and alsactl all seem to work correctly. I've tried obtaining and installing the latest mercurial clone of alsa-lib, and I've tried this with or without the patch_via.c patch below. Am I doing something wrong? -- Darryl -------------- Original message ---------------------- From: forumjunk(a)comcast.net (Darryl House) > The patch fixed all compilation errors for me, thank you. I did have to apply it > manually, but that was rather simple. > > I can confirm that SPDIF input doesn't work for the card, but I haven't applied > the patch that you suggested to John Stile for that, yet. I'll do that a bit > later today if you still recommend it > > Is this patch still recommended? > > > diff -r 2d3893cf030c pci/hda/patch_via.c > > --- a/pci/hda/patch_via.c Tue Dec 18 15:42:53 2007 +0100 > > +++ b/pci/hda/patch_via.c Tue Dec 18 17:48:00 2007 +0100 > > @@ -568,25 +568,27 @@ static int via_init(struct hda_codec *co > > if (IS_VT1708_VENDORID(codec->vendor_id)) { > > snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, > > AC_VERB_SET_PIN_WIDGET_CONTROL, > > - 0x40); > > + PIN_OUT); > > snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, > > AC_VERB_SET_EAPD_BTLENABLE, 0x02); > > } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) || > > IS_VT1709_6CH_VENDORID(codec->vendor_id)) { > > snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, > > AC_VERB_SET_PIN_WIDGET_CONTROL, > > - 0x40); > > + PIN_OUT); > > snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, > > AC_VERB_SET_EAPD_BTLENABLE, 0x02); > > } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) || > > IS_VT1708B_4CH_VENDORID(codec->vendor_id)) { > > snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, > > AC_VERB_SET_PIN_WIDGET_CONTROL, > > - 0x40); > > + PIN_OUT); > > snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, > > AC_VERB_SET_EAPD_BTLENABLE, 0x02); > > } > > - } > > + } else /* enable SPDIF-input pin */ > > + snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, > > + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); > > > > return 0; > > } > > Thanks again! > > -- Darryl > > > -------------- Original message ---------------------- > From: Takashi Iwai <tiwai(a)suse.de> > [...] > > Thanks. Does the patch below fix the probelm? > > Run hgcompile after applying it. > > > > > > Takashi > > > > diff -r 414b1156a8b3 configure.in > > --- a/configure.in Tue Dec 18 16:09:20 2007 +0100 > > +++ b/configure.in Fri Dec 21 12:10:53 2007 +0100 > > @@ -1230,7 +1230,7 @@ esac > > > > dnl set ia32 (X86) > > case "$processor" in > > - i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*) > > + i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*|x86_32) > > if $KCC -mpreferred-stack-boundary=2 -S -o /dev/null -xc /dev/null > > >/dev/null 2>&1; then > > c_opts="-mpreferred-stack-boundary=2 $c_opts" > > CONFIG_X86=y > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 21 Dec '07

21 Dec '07

The patch fixed all compilation errors for me, thank you. I did have to apply it manually, but that was rather simple. I can confirm that SPDIF input doesn't work for the card, but I haven't applied the patch that you suggested to John Stile for that, yet. I'll do that a bit later today if you still recommend it Is this patch still recommended? > diff -r 2d3893cf030c pci/hda/patch_via.c > --- a/pci/hda/patch_via.c Tue Dec 18 15:42:53 2007 +0100 > +++ b/pci/hda/patch_via.c Tue Dec 18 17:48:00 2007 +0100 > @@ -568,25 +568,27 @@ static int via_init(struct hda_codec *co > if (IS_VT1708_VENDORID(codec->vendor_id)) { > snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, > AC_VERB_SET_PIN_WIDGET_CONTROL, > - 0x40); > + PIN_OUT); > snd_hda_codec_write(codec, VT1708_DIGIN_PIN, 0, > AC_VERB_SET_EAPD_BTLENABLE, 0x02); > } else if (IS_VT1709_10CH_VENDORID(codec->vendor_id) || > IS_VT1709_6CH_VENDORID(codec->vendor_id)) { > snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, > AC_VERB_SET_PIN_WIDGET_CONTROL, > - 0x40); > + PIN_OUT); > snd_hda_codec_write(codec, VT1709_DIGIN_PIN, 0, > AC_VERB_SET_EAPD_BTLENABLE, 0x02); > } else if (IS_VT1708B_8CH_VENDORID(codec->vendor_id) || > IS_VT1708B_4CH_VENDORID(codec->vendor_id)) { > snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, > AC_VERB_SET_PIN_WIDGET_CONTROL, > - 0x40); > + PIN_OUT); > snd_hda_codec_write(codec, VT1708B_DIGIN_PIN, 0, > AC_VERB_SET_EAPD_BTLENABLE, 0x02); > } > - } > + } else /* enable SPDIF-input pin */ > + snd_hda_codec_write(codec, spec->autocfg.dig_in_pin, 0, > + AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN); > > return 0; > } Thanks again! -- Darryl -------------- Original message ---------------------- From: Takashi Iwai <tiwai(a)suse.de> [...] > Thanks. Does the patch below fix the probelm? > Run hgcompile after applying it. > > > Takashi > > diff -r 414b1156a8b3 configure.in > --- a/configure.in Tue Dec 18 16:09:20 2007 +0100 > +++ b/configure.in Fri Dec 21 12:10:53 2007 +0100 > @@ -1230,7 +1230,7 @@ esac > > dnl set ia32 (X86) > case "$processor" in > - i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*) > + i?86*|k?|crusoe|mcyrixiii|mwinchip|viac3_2|mcore2|geodegx*|x86_32) > if $KCC -mpreferred-stack-boundary=2 -S -o /dev/null -xc /dev/null > >/dev/null 2>&1; then > c_opts="-mpreferred-stack-boundary=2 $c_opts" > CONFIG_X86=y > _______________________________________________ > 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] no input/output analog channels on rme multiface I
by rmgls＠free.fr 21 Dec '07

21 Dec '07

Hi all, the multiface 1, does not work for me. it is recognized as hammerfal DSP, but the amixer shows no analog in/out (there are 8 analog I/O in the box) and of course no volume control for any channel. the driver: hg: 20071205, the last lib, and firmware, and utils. unpluging the box (power off), and loading the firmware on windows gives: in the debian (2.6.24-rc3) dmesg: e9652/../../alsa-kernel/pci/rme9652/hdsp.c:5037: Hammerfall-DSP: Firmware already present, initializing card. ... pcmcia: Detected deprecated PCMCIA ioctl usage from process: discover. pcmcia: This interface will soon be removed from the kernel; please expect breakage unless you upgrade to new tools. pcmcia: see http://www.kernel.org/pub/linux/utils/kernel/pcmcia/pcmcia.html for details. sndstat reports: Xilinx FPGA ??? and now amixer -c2: Simple mixer control 'Line Out',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Emphasis Bit',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Input Connector',0 Capabilities: enum Items: 'Optical' 'Coaxial' 'Internal' Item0: 'Internal' Simple mixer control 'IEC958 Non-audio Bit',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Output also on ADAT1',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Professional Bit',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'ADAT Lock Status',1 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' Item0: 'No Lock' Simple mixer control 'ADAT Sync Lock Status',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' Item0: 'No Lock' Simple mixer control 'AutoSync Reference',0 Capabilities: enum Items: 'Word' 'ADAT Sync' 'IEC958' 'None' 'ADAT1' 'ADAT2' 'ADAT3' Item0: 'None' Simple mixer control 'External Rate',0 Capabilities: enum Items: '32000' '44100' '48000' '64000' '88200' '96000' 'None' Item0: 'None' Simple mixer control 'Preferred Sync Reference',0 Capabilities: enum Items: 'Word' 'IEC958' 'ADAT1' 'ADAT Sync' Item0: 'IEC958' Simple mixer control 'SPDIF Lock Status',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' Item0: 'No Lock' Simple mixer control 'SPDIF Sample Rate',0 Capabilities: enum Items: '32000' '44100' '48000' '64000' '88200' '96000' 'None' Item0: 'None' Simple mixer control 'Sample Clock Source',0 Capabilities: enum Items: 'AutoSync' 'Internal 32.0 kHz' 'Internal 44.1 kHz' 'Internal 48.0 kHz' 'Internal 64.0 kHz' 'Internal 88.2 kHz' 'Internal 96.0 kHz' Item0: 'Internal 44.1 kHz' Simple mixer control 'Sample Clock Source Locking',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'System Clock Mode',0 Capabilities: enum Items: 'Master' 'Slave' Item0: 'Master' Simple mixer control 'System Sample Rate',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 0 Mono: 0 [0%] Simple mixer control 'Word Clock Lock Status',0 Capabilities: enum Items: 'No Lock' 'Lock' 'Sync' Item0: 'No Lock' please do i miss something? thanks rmgls(a)free.fr

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[alsa-devel] Userspace control of HDA port connections
by Andrew Paprocki 21 Dec '07

21 Dec '07

Takashi, I'm wondering if there is any interest in providing a userspace way of modifying the HDA port/mixer input/output mappings? The driver currently uses hard-coded models to define the standard configuration for a given piece of hardware, but this doesn't allow a user to redefine a line-in port as an output port if the chipset has an available unmapped mixer. I'm bringing this up because from what I understand of the ALC883 is that it has 4 mixers, yet on my motherboard there are only 3 ports configured as output by default (6stack-dig). Correct me if I'm wrong, but I should be able to map all of the physical jacks as outputs and then assign each of the mixers to a jack or more than one jack. That would require adding a new model to the driver and/or unloading/reloading the module with a new model string. In a system which has the driver built-in and isn't using modules, this requires a reboot. If there was a userspace way of reconfiguring the jacks (via /proc/asound?) then perhaps even all of the hardware -> model mappings could be moved into a little tool run at boot time which doesn't require a kernel change every time new hardware is introduced. A nice little ncurses tool could even be made to modify port mappings. What do you think? If I'm misunderstanding the capabilities of the chips please let me know. Thanks, -Andrew p.s. I still have yet to test out the multi-card patch. I'll reply to that once I have more info.

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[alsa-devel] [PATCH] fix entries for U24_BE and U24_LE in pcm_formats[] array
by Timur Tabi 21 Dec '07

21 Dec '07

SNDRV_PCM_FORMAT_U24_BE and SNDRV_PCM_FORMAT_U24_LE represent 24-bit samples that are aligned into a 4-byte word, so the most significant byte is always zero. The entries for these two sample formats in the pcm_formats[] array only specified 3 bytes. For SNDRV_PCM_FORMAT_U24_LE, the compiler automatically adds the 4th byte, but for SNDRV_PCM_FORMAT_U24_BE the entry was just wrong. This patch fixes both entries. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- I do not have hardware that supports U24_BE, so I can't test this change. sound/core/pcm_misc.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index dd9aa51..077fc0b 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -71,11 +71,11 @@ static struct pcm_format_data pcm_formats[SNDRV_PCM_FORMAT_LAST+1] = { }, [SNDRV_PCM_FORMAT_U24_LE] = { .width = 24, .phys = 32, .le = 1, .signd = 0, - .silence = { 0x00, 0x00, 0x80 }, + .silence = { 0x00, 0x00, 0x80, 0x00 }, }, [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, - .silence = { 0x80, 0x00, 0x00 }, + .silence = { 0x00, 0x80, 0x00, 0x00 }, }, [SNDRV_PCM_FORMAT_S32_LE] = { .width = 32, .phys = 32, .le = 1, .signd = 1, -- 1.5.2.4

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[alsa-devel] Using dmix/dsnoop/dshare to access individual channels
by Steve Strobel 21 Dec '07

21 Dec '07

I am using a 4-input, 8-output codec (AD1938) in an application similar to a live sound mixing board, where a combination of the input signals get mixed for each output. I can mix four inputs to four outputs with a command like: arecord --file-type raw --channels=4 --format=S32_LE --rate=16000 \ | crosspoint \ | aplay --file-type raw --channels=4 --format=S32_LE --rate=16000 where crosspoint is a simple program that reads from stdin, does some mixing, and outputs to stdout. My next step is to play and record wave files from the various channels, while the crosspoint is running. For example, I might need to play message1.wav to channel 2 and message2.wav to channel 3 while recording channel 1 to message3.wav. I am looking for advice about the best way to accomplish that. It seems like it should be possible using a combination of dmix, snoop, and dshare, as follows: - Use dsnoop to split the 4-channel input to the crosspoint and another virtual sound input, call it "A". - Use dshare to split "A" into four mono sources. Then can use arecord normally to capture wave files. - Use dmix and possibly dshare to similarly let the crosspoint continue to use the multi-channel output while also providing a mono sound device for each channel that I can use with aplay. Am I on the right track, or would I be better off adding functionality to the crosspoint application to read and write wave files? Thanks for any suggestions. Steve --- Steve Strobel Link Communications, Inc. 1035 Cerise Rd Billings, MT 59101-7378 (406) 245-5002 ext 102 (406) 245-4889 (fax) WWW: http://www.link-comm.com MailTo:steve.strobel@link-comm.com

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 20 Dec '07

20 Dec '07

For VIA platforms, to avoid the driver compile error in pci.h, reconfigure your kernel to remove SMP support, and set the architecture type to PC-compatible. (you already know this, I'm sure): Recompile the kernel (as a paranoid, I always trash /lib/modules/<kernel_version> before making modules_install). Install the kernel, of course, and reconfigure your boot loader and/or initrd and reboot. There are still several fixes that need to be made to get all of this to compile cleanly, but I haven't got the permanent fixes nailed down yet. I'm almost positive that these are alsa-driver configure errors or omissions. I can't (yet) find a kernel config that will avoid them. The fixes below are harmless, and they will allow you to get the driver compiled. Add this definition to the top section of acore/memory_wrapper.c after the line that reads #include <linux/version.h> #define VMALLOC_VMADDR(x) ((unsigned long)(x)) Change these lines (45 and 46 or close to that) of acore/memory_wrapper.c from: pmd = pmd_offset(pgd, lpage); pte = pte_offset(pmd, lpage); to pmd = pmd_offset((pud_t *)pgd, lpage); pte = pte_offset_kernel(pmd, lpage); The driver will then compile cleanly (at least with no errors or warnings relating to any of these changes) I will work more as time allows to find or create configure tests to make sure that the correct versions of these functions get included or patched.

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[alsa-devel] [PATCH 0/2] ASoC drivers for the Freescale MPC8610 SoC
by Timur Tabi 20 Dec '07

20 Dec '07

These patches add ALSA SoC device drivers for the Freescale MPC8610 SoC and the MPC8610-HPCD reference board. The first patch updates some files in arch/powerpc and booting-without-of.txt. The second patch adds the new audio driver files to sound/soc. Both must be applied in order to avoid a build break.

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[alsa-devel] write/read system calls and ioctl
by Michael Voigt 20 Dec '07

20 Dec '07

Dear ALSA developers, At the beginning, I would like to give a short introdution of myself, then my question follows. My project: =============== Currently, I am interning at the TU Dresden operating systems group's [1] and it is my task to port ALSA to DROPS [2, 3], a L4-microkernel-based real-time system. The long-term goal of this project should be to establish a truely real-time capable FOSS platform for audio recording and processing. The current state of the project: By using DDELinux [4] and writing a small kernel-user inferface emulation layer, we now have an unchanged version of ALSA and an unchanged version of salsa-lib [5] running on our system. The next steps will be to test how stable this port is and fix bugs. Then, during my diploma thesis, it is my task to port Jack (or an equivalent) to our system and to evaluate how to make the whole thing to work in real-time. But now my question: =============== In the file pcm_native in alsa-core read (snd_pcm_read) and write (snd_pcm_write) functions are attached to the snd_pcm_f_ops. But as far as I can see, these write- und read- system calls aren't used either in salsa-lib or in alsa-lib. I can see that in former versions of alsa-lib (for instance alsalib-0.0.3) the write system call was used to implement snd_pcm_write (LIB); but now we have snd_pcm_writei (LIB) and snd_pcm_writen (LIB) which use ioctl. I guess this was done to realize both interleaved and noninterleaved audio playback (The old snd_pcm_write (LIB) was based on bytes not on frames and did not have a notion of interleaved/noninterleaved playback?). If I have a look at snd_pcm_write (KERNEL, pcm_native.c), I can see it also ends up in snd_pcm_lib_write(...). So does ioctl with SNDRV_PCM_IOCTL_WRITEI_FRAMES as argument, but snd_pcm_write (KERNEL) does not copy the frames to kernel memory. My question: what is snd_pcm_write in pcm_native.c good for? Is it only there to keep ALSA compatible to older versions of alsa-lib? Kindest regards, Michael Voigt ================ [1] http://www.inf.tu-dresden.de/index.php?node_id=1141&ln=en [2] http://www.osnews.com/story.php/15814/Introduction-to-TUD-OS/ [3] http://www.realtimelinuxfoundation.org/variants/variants.html#VARIANTS_DROPS [4] http://demo.tudos.org/dsweeper_tutorial.html [5] http://ftp.isr.ist.utl.pt/pub/MIRRORS/ftp.suse.com/people/tiwai/salsa-lib/ ================ -- Today is Pungenday, the 61st day of The Aftermath in the YOLD 3173

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Re: [alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by forumjunk＠comcast.net 20 Dec '07

20 Dec '07

> Don't do this. This part shouldn't be compiled at all for recent > kernels. This is, again, the problem of configure script that > detected the codes wrongly. I apologize if I've muddied the water with any of this, I've discovered that the C7 and VIA in general are the forgotten children of the IT world, and I've had to do many things by trial and error. The target platform is an embedded digital signage application and I needed to get audio working quickly. And it does work, but I certainly recognize that my code changes are hacks and need to be addressed properly. I've tried various kernel config changes dealing with memory management and that hasn't helped, so I've reverted to a generic configuration that matches John Stiles' as far as the platform, ALSA and OSS are concerned, as you can see from the log below. Make, with this configuration, fails as John Stiles has reported above. This is the config.log from a clean mercurial clone this morning (its huge, I hope this is ok to do in a mailing list...) --- This file contains any messages produced by compilers while running configure, to aid debugging if configure makes a mistake. It was created by configure, which was generated by GNU Autoconf 2.61. Invocation command line was $ ./configure --with-debug=full --with-isapnp=yes --with-sequencer=yes ## --------- ## ## Platform. ## ## --------- ## hostname = viziframe uname -m = i686 uname -r = 2.6.23.11 uname -s = Linux uname -v = #2 Wed Dec 19 15:50:23 PST 2007 /usr/bin/uname -p = VIA Esther processor 1500MHz /bin/uname -X = unknown /bin/arch = i686 /usr/bin/arch -k = unknown /usr/convex/getsysinfo = unknown /usr/bin/hostinfo = unknown /bin/machine = unknown /usr/bin/oslevel = unknown /bin/universe = unknown PATH: /usr/local/sbin PATH: /usr/sbin PATH: /sbin PATH: /usr/local/bin PATH: /usr/bin PATH: /bin PATH: /usr/games PATH: /usr/lib/java/bin PATH: /usr/lib/java/jre/bin PATH: /usr/lib/qt/bin PATH: /usr/share/texmf/bin ## ----------- ## ## Core tests. ## ## ----------- ## configure:2108: checking for gcc configure:2124: found /usr/bin/gcc configure:2135: result: gcc configure:2373: checking for C compiler version configure:2380: gcc --version >&5 gcc (GCC) 4.1.2 Copyright (C) 2006 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. configure:2383: $? = 0 configure:2390: gcc -v >&5 Reading specs from /usr/lib/gcc/i486-slackware-linux/4.1.2/specs Target: i486-slackware-linux Configured with: ../gcc-4.1.2/configure --prefix=/usr --enable-shared --enable-languages=ada,c,c++,fortran,java,objc --enable-threads=posix --enable-__cxa_atexit --disable-checki ng --with-gnu-ld --verbose --with-arch=i486 --target=i486-slackware-linux --host=i486-slackware-linux Thread model: posix gcc version 4.1.2 configure:2393: $? = 0 configure:2400: gcc -V >&5 gcc: '-V' option must have argument configure:2403: $? = 1 configure:2426: checking for C compiler default output file name configure:2453: gcc conftest.c >&5 configure:2456: $? = 0 configure:2494: result: a.out configure:2511: checking whether the C compiler works configure:2521: ./a.out configure:2524: $? = 0 configure:2541: result: yes configure:2548: checking whether we are cross compiling configure:2550: result: no configure:2553: checking for suffix of executables configure:2560: gcc -o conftest conftest.c >&5 configure:2563: $? = 0 configure:2587: result: configure:2593: checking for suffix of object files configure:2619: gcc -c conftest.c >&5 configure:2622: $? = 0 configure:2645: result: o configure:2649: checking whether we are using the GNU C compiler configure:2678: gcc -c conftest.c >&5 configure:2684: $? = 0 configure:2701: result: yes configure:2706: checking whether gcc accepts -g configure:2736: gcc -c -g conftest.c >&5 configure:2742: $? = 0 configure:2841: result: yes configure:2858: checking for gcc option to accept ISO C89 configure:2932: gcc -c -g -O2 conftest.c >&5 configure:2938: $? = 0 configure:2961: result: none needed configure:3022: checking for ranlib configure:3038: found /usr/bin/ranlib configure:3049: result: ranlib configure:3119: checking for a BSD-compatible install configure:3175: result: /usr/bin/ginstall -c configure:3198: checking how to run the C preprocessor configure:3238: gcc -E conftest.c configure:3244: $? = 0 configure:3275: gcc -E conftest.c conftest.c:8:28: error: ac_nonexistent.h: No such file or directory configure:3281: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | /* end confdefs.h. */ | #include <ac_nonexistent.h> configure:3314: result: gcc -E configure:3343: gcc -E conftest.c configure:3349: $? = 0 configure:3380: gcc -E conftest.c conftest.c:8:28: error: ac_nonexistent.h: No such file or directory configure:3386: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | /* end confdefs.h. */ | #include <ac_nonexistent.h> configure:3424: checking for grep that handles long lines and -e configure:3498: result: /usr/bin/grep configure:3503: checking for egrep configure:3581: result: /usr/bin/grep -E configure:3586: checking for ANSI C header files configure:3616: gcc -c -g -O2 conftest.c >&5 configure:3622: $? = 0 configure:3721: gcc -o conftest -g -O2 conftest.c >&5 configure:3724: $? = 0 configure:3730: ./conftest configure:3733: $? = 0 configure:3750: result: yes configure:3763: checking for an ANSI C-conforming const configure:3838: gcc -c -g -O2 conftest.c >&5 configure:3844: $? = 0 configure:3859: result: yes configure:3869: checking for inline configure:3895: gcc -c -g -O2 conftest.c >&5 configure:3901: $? = 0 configure:3919: result: inline configure:3938: checking whether time.h and sys/time.h may both be included configure:3968: gcc -c -g -O2 conftest.c >&5 configure:3974: $? = 0 configure:3989: result: yes configure:4001: checking whether gcc needs -traditional configure:4043: result: no configure:4051: checking for current directory configure:4055: result: /usr/src/alsa-driver configure:4353: checking cross compile configure:4365: result: configure:4380: checking for directory with kernel source configure:4401: result: /lib/modules/2.6.23.11/source configure:4404: checking for directory with kernel build configure:4416: result: /lib/modules/2.6.23.11/build configure:4421: checking for kernel linux/version.h configure:4436: result: yes configure:4440: checking for kernel linux/autoconf.h configure:4455: result: yes configure:4459: checking for kernel version configure:4544: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 configure:4547: $? = 0 configure:4553: ./conftest configure:4556: $? = 0 configure:4591: result: 2.6.23.11 configure:4594: checking for GCC version configure:4658: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:41: warning: incompatible implicit declaration of built-in function 'exit' configure:4661: $? = 0 configure:4667: ./conftest configure:4670: $? = 0 configure:4777: result: Kernel compiler: gcc 4.1.2 Used compiler: gcc (GCC) 4.1.2 configure:4829: checking for built-in ALSA configure:4867: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:14: warning: incompatible implicit declaration of built-in function 'exit' configure:4870: $? = 0 configure:4876: ./conftest configure:4879: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #ifndef CONFIG_SND | exit(1); | #else | exit(0); | #endif | } | configure:4889: result: no configure:4907: checking for existing ALSA module configure:4945: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:16: warning: incompatible implicit declaration of built-in function 'exit' configure:4948: $? = 0 configure:4954: ./conftest configure:4957: $? = 0 configure:4959: result: yes configure:4982: checking for Red Hat kernel configure:5003: result: auto configure:5008: checking for Red Hat kernel configure:5046: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:15: warning: incompatible implicit declaration of built-in function 'exit' configure:5049: $? = 0 configure:5055: ./conftest configure:5058: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(RED_HAT_LINUX_KERNEL) && !defined(RED_HAT_LINUX_KERNEL_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:5068: result: no configure:5081: checking for SUSE kernel configure:5102: result: auto configure:5107: checking for SUSE kernel configure:5145: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:15: warning: incompatible implicit declaration of built-in function 'exit' configure:5148: $? = 0 configure:5154: ./conftest configure:5157: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(CONFIG_SUSE_KERNEL) && !defined(CONFIG_SUSE_KERNEL_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:5167: result: no configure:5187: checking for CONFIG_EXPERIMENTAL configure:5198: result: yes configure:5211: checking for kernel linux/config.h configure:5232: result: no configure:5261: checking to modify of kernel linux/kmod.h configure:5269: result: no configure:5286: checking for kernel linux/utsrelease.h configure:5297: result: yes configure:5323: checking for kernel linux/compiler.h configure:5334: result: yes configure:5360: checking for kernel linux/pm.h configure:5371: result: yes configure:5397: checking for kernel linux/spinlock.h configure:5408: result: yes configure:5434: checking for kernel linux/irq.h configure:5445: result: yes configure:5471: checking for kernel linux/threads.h configure:5482: result: yes configure:5508: checking for kernel linux/rwsem.h configure:5519: result: yes configure:5545: checking for kernel linux/gameport.h configure:5556: result: yes configure:5582: checking for kernel media/v4l2-dev.h configure:5593: result: yes configure:5619: checking for kernel linux/devfs_fs_kernel.h configure:5640: result: no configure:5656: checking for kernel linux/highmem.h configure:5667: result: yes configure:5693: checking for kernel linux/workqueue.h configure:5704: result: yes configure:5739: checking for kernel linux/dma-mapping.h configure:5750: result: yes configure:5776: checking for kernel asm/hw_irq.h configure:5787: result: yes configure:5813: checking for kernel linux/device.h configure:5824: result: yes configure:5852: checking for kernel linux/platform_device.h configure:5863: result: yes configure:5891: checking for kernel linux/isa.h configure:5902: result: yes configure:5930: checking for kernel linux/jiffies.h configure:5941: result: yes configure:5971: checking for kernel linux/compat.h configure:5982: result: yes configure:6010: checking for kernel linux/log2.h configure:6021: result: yes configure:6047: checking for kernel linux/adb.h configure:6058: result: yes configure:6086: checking for kernel linux/cuda.h configure:6097: result: yes configure:6125: checking for kernel linux/pmu.h configure:6136: result: yes configure:6164: checking for kernel linux/moduleparam.h configure:6175: result: yes configure:6201: checking for kernel linux/syscalls.h configure:6212: result: yes configure:6240: checking for kernel linux/firmware.h configure:6251: result: yes configure:6279: checking for kernel linux/err.h configure:6290: result: yes configure:6318: checking for kernel linux/bitmap.h configure:6329: result: yes configure:6357: checking for kernel linux/mutex.h configure:6368: result: yes configure:6396: checking for kernel linux/latency.h configure:6407: result: yes configure:6435: checking for kernel linux/kthread.h configure:6446: result: yes configure:6474: checking for kernel asm/irq_regs.h configure:6485: result: yes configure:6540: checking for kernel module symbol versions configure:6578: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:17: warning: incompatible implicit declaration of built-in function 'exit' configure:6581: $? = 0 configure:6587: ./conftest configure:6590: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(CONFIG_MODVERSIONS) && !defined(CONFIG_MODVERSIONS_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:6600: result: no configure:6617: checking for has ioport support configure:6655: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:19: warning: incompatible implicit declaration of built-in function 'exit' configure:6658: $? = 0 configure:6664: ./conftest configure:6667: $? = 0 configure:6669: result: yes configure:6695: checking for PCI support in kernel configure:6733: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:20: warning: incompatible implicit declaration of built-in function 'exit' configure:6736: $? = 0 configure:6742: ./conftest configure:6745: $? = 0 configure:6747: result: yes configure:6770: checking for I2C driver in kernel configure:6814: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:20: warning: incompatible implicit declaration of built-in function 'exit' configure:6817: $? = 0 configure:6823: ./conftest configure:6826: $? = 0 configure:6852: result: unknown configure:6959: checking for firmware loader configure:7003: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:20: warning: incompatible implicit declaration of built-in function 'exit' configure:7006: $? = 0 configure:7012: ./conftest configure:7015: $? = 0 configure:7041: result: yes configure:7062: checking for input subsystem in kernel configure:7106: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:20: warning: incompatible implicit declaration of built-in function 'exit' configure:7109: $? = 0 configure:7115: ./conftest configure:7118: $? = 0 configure:7144: result: yes configure:7154: checking for directory to store kernel modules configure:7181: result: /lib/modules/2.6.23.11/kernel/sound configure:7184: checking for verbose procfs configure:7200: result: on configure:7209: checking for verbose printk configure:7225: result: on configure:7235: checking for debug level configure:7256: result: full configure:7298: checking for ISA support in kernel configure:7336: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:24: warning: incompatible implicit declaration of built-in function 'exit' configure:7339: $? = 0 configure:7345: ./conftest configure:7348: $? = 0 configure:7350: result: yes configure:7372: checking for processor type configure:7509: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:24: warning: incompatible implicit declaration of built-in function 'exit' configure:7512: $? = 0 configure:7518: ./conftest configure:7521: $? = 0 configure:7523: result: x86_32 configure:8006: checking for ISA DMA API configure:8044: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:24: warning: incompatible implicit declaration of built-in function 'exit' configure:8047: $? = 0 configure:8053: ./conftest configure:8056: $? = 0 configure:8058: result: yes configure:8294: checking for SMP configure:8332: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:23: warning: incompatible implicit declaration of built-in function 'exit' configure:8335: $? = 0 configure:8341: ./conftest configure:8344: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(CONFIG_SMP) && !defined(CONFIG_SMP_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:8354: result: no configure:8382: checking for Video device support in kernel configure:8420: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:25: warning: incompatible implicit declaration of built-in function 'exit' configure:8423: $? = 0 configure:8429: ./conftest configure:8432: $? = 0 configure:8434: result: yes configure:8455: checking for ISA PnP driver in kernel configure:8459: result: yes configure:8472: checking for PnP driver in kernel configure:8476: result: yes configure:8493: checking for Kernel ISA-PnP support configure:8531: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:27: warning: incompatible implicit declaration of built-in function 'exit' configure:8534: $? = 0 configure:8540: ./conftest configure:8543: $? = 0 configure:8545: result: yes configure:8690: checking for strlcpy configure:8726: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Werror conftest.c >&5 configure:8732: $? = 0 configure:8737: result: yes configure:8758: checking for snprintf configure:8793: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Werror conftest.c >&5 configure:8799: $? = 0 configure:8804: result: yes configure:8825: checking for vsnprintf configure:8867: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Werror conftest.c >&5 configure:8873: $? = 0 configure:8878: result: yes configure:8899: checking for scnprintf configure:8934: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Werror conftest.c >&5 configure:8940: $? = 0 configure:8945: result: yes configure:8966: checking for sscanf configure:9001: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Werror conftest.c >&5 configure:9007: $? = 0 configure:9012: result: yes configure:9033: checking for vmalloc_to_page configure:9071: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/seqlock.h:29, from /lib/modules/2.6.23.11/build/include/linux/time.h:8, from conftest.c:31: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness In file included from /lib/modules/2.6.23.11/build/include/linux/irq.h:23, from /lib/modules/2.6.23.11/build/include/asm/hardirq.h:5, from /lib/modules/2.6.23.11/build/include/linux/hardirq.h:7, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:11, from conftest.c:33: /lib/modules/2.6.23.11/build/include/asm/irq.h:15:25: error: irq_vectors.h: No such file or directory In file included from /lib/modules/2.6.23.11/build/include/asm/hardirq.h:5, from /lib/modules/2.6.23.11/build/include/linux/hardirq.h:7, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:11, from conftest.c:33: /lib/modules/2.6.23.11/build/include/linux/irq.h: At top level: /lib/modules/2.6.23.11/build/include/linux/irq.h:180: error: 'NR_IRQS' undeclared here (not in a function) configure:9077: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/time.h> | #include <linux/sched.h> | #include <linux/interrupt.h> | #include <linux/mm.h> | | int | main () | { | | struct page * (*func)(void*); | func = vmalloc_to_page; | | ; | return 0; | } configure:9088: result: no configure:9103: checking for old kmod configure:9137: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 conftest.c: In function 'main': conftest.c:37: warning: too many arguments for format configure:9143: $? = 0 configure:9148: result: no configure:9169: checking for PDE configure:9205: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/wait.h:22, from /lib/modules/2.6.23.11/build/include/linux/fs.h:272, from conftest.c:31: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:9211: $? = 0 configure:9216: result: yes configure:9237: checking for pci_set_consistent_dma_mask configure:9272: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/pci.h:48, from conftest.c:32: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:9278: $? = 0 configure:9283: result: yes configure:9304: checking for pci_dev_present configure:9339: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/pci.h:48, from conftest.c:33: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:9345: $? = 0 configure:9350: result: yes configure:9371: checking for msleep configure:9406: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 configure:9412: $? = 0 configure:9417: result: yes configure:9439: checking for msleep_interrupt configure:9474: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 configure:9480: $? = 0 configure:9485: result: yes configure:9507: checking for msecs_to_jiffies configure:9543: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Iinclude conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/seqlock.h:29, from /lib/modules/2.6.23.11/build/include/linux/time.h:8, from /lib/modules/2.6.23.11/build/include/linux/jiffies.h:7, from conftest.c:36: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:44: warning: assignment from incompatible pointer type configure:9549: $? = 0 configure:9554: result: yes configure:9575: checking for tty->count is the atomic type configure:9610: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/wait.h:22, from /lib/modules/2.6.23.11/build/include/linux/fs.h:272, from /lib/modules/2.6.23.11/build/include/linux/tty.h:9, from conftest.c:37: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:44: error: request for member 'counter' in something not a structure or union configure:9616: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/tty.h> | | int | main () | { | | struct tty_struct tty; | atomic_read(&tty.count); | | ; | return 0; | } configure:9627: result: no configure:9643: checking for video_get_drvdata configure:9679: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Iinclude conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/seqlock.h:29, from /lib/modules/2.6.23.11/build/include/linux/time.h:8, from /lib/modules/2.6.23.11/build/include/linux/videodev2.h:59, from /lib/modules/2.6.23.11/build/include/linux/videodev.h:15, from conftest.c:37: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:9685: $? = 0 configure:9690: result: yes configure:9717: checking for V4L1 layer configure:9755: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:41: warning: incompatible implicit declaration of built-in function 'exit' configure:9758: $? = 0 configure:9764: ./conftest configure:9767: $? = 0 configure:9769: result: yes configure:9792: checking for io_remap_pfn_range configure:9827: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME -Werror-implicit-function-declaration conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/mmzone.h:7, from /lib/modules/2.6.23.11/build/include/linux/gfp.h:4, from /lib/modules/2.6.23.11/build/include/linux/mm.h:8, from conftest.c:38: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:9833: $? = 0 configure:9838: result: yes configure:9926: checking for kcalloc configure:9961: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/mmzone.h:7, from /lib/modules/2.6.23.11/build/include/linux/gfp.h:4, from /lib/modules/2.6.23.11/build/include/linux/slab.h:14, from conftest.c:39: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:46: warning: assignment from incompatible pointer type configure:9967: $? = 0 configure:9972: result: yes configure:9993: checking for kstrdup configure:10028: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 conftest.c: In function 'main': conftest.c:47: warning: assignment from incompatible pointer type configure:10034: $? = 0 configure:10039: result: yes configure:10060: checking for kzalloc configure:10095: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/mmzone.h:7, from /lib/modules/2.6.23.11/build/include/linux/gfp.h:4, from /lib/modules/2.6.23.11/build/include/linux/slab.h:14, from conftest.c:41: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:48: warning: assignment from incompatible pointer type configure:10101: $? = 0 configure:10106: result: yes configure:10128: checking for create_workqueue with flags configure:10162: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/timer.h:4, from /lib/modules/2.6.23.11/build/include/linux/workqueue.h:8, from conftest.c:42: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c:48:31: error: macro "create_workqueue" passed 2 arguments, but takes just 1 conftest.c: In function 'main': conftest.c:48: error: 'create_workqueue' undeclared (first use in this function) conftest.c:48: error: (Each undeclared identifier is reported only once conftest.c:48: error: for each function it appears in.) configure:10168: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/workqueue.h> | | int | main () | { | | create_workqueue("abcd", 0); | | ; | return 0; | } configure:10179: result: no configure:10195: checking for saved_config_space in pci_dev configure:10230: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/pci.h:48, from conftest.c:42: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:10236: $? = 0 configure:10241: result: yes configure:10263: checking for new pci_save_state configure:10298: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/pci.h:48, from conftest.c:43: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:50: warning: 'pci' is used uninitialized in this function configure:10304: $? = 0 configure:10309: result: yes configure:10330: checking for register_sound_special_device configure:10365: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/wait.h:22, from /lib/modules/2.6.23.11/build/include/linux/fs.h:272, from /lib/modules/2.6.23.11/build/include/linux/sound.h:6, from conftest.c:44: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:51: warning: assignment from incompatible pointer type configure:10371: $? = 0 configure:10376: result: yes configure:10633: checking for driver version configure:10646: result: 1.0.15 configure:10649: checking for sequencer support configure:10673: result: yes configure:10680: checking for OSS/Free emulation configure:10713: result: yes configure:10722: checking for OSS PCM plugin system inclusion configure:10741: result: yes configure:10750: checking for RTC callback support in kernel configure:10791: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:8, from /lib/modules/2.6.23.11/build/include/linux/rtc.h:102, from conftest.c:55: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness In file included from /lib/modules/2.6.23.11/build/include/linux/irq.h:23, from /lib/modules/2.6.23.11/build/include/asm/hardirq.h:5, from /lib/modules/2.6.23.11/build/include/linux/hardirq.h:7, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:11, from /lib/modules/2.6.23.11/build/include/linux/rtc.h:102, from conftest.c:55: /lib/modules/2.6.23.11/build/include/asm/irq.h:15:25: error: irq_vectors.h: No such file or directory In file included from /lib/modules/2.6.23.11/build/include/asm/hardirq.h:5, from /lib/modules/2.6.23.11/build/include/linux/hardirq.h:7, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:11, from /lib/modules/2.6.23.11/build/include/linux/rtc.h:102, from conftest.c:55: /lib/modules/2.6.23.11/build/include/linux/irq.h: At top level: /lib/modules/2.6.23.11/build/include/linux/irq.h:180: error: 'NR_IRQS' undeclared here (not in a function) conftest.c: In function 'main': conftest.c:63: warning: unused variable 'cb' configure:10797: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/version.h> | #include <linux/rtc.h> | #include <linux/mc146818rtc.h> | | int | main () | { | | #ifdef RTC_IRQ | rtc_task_t *cb = 0; | #else | #error | #endif | | ; | return 0; | } configure:10808: result: no configure:10831: checking for HPET support configure:10869: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:57: warning: incompatible implicit declaration of built-in function 'exit' configure:10872: $? = 0 configure:10878: ./conftest configure:10881: $? = 0 configure:10883: result: yes configure:10905: checking for dynamic minor numbers configure:10923: result: no configure:10927: checking for support of old API configure:10941: result: yes configure:10951: checking for Procfs support configure:10989: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:58: warning: incompatible implicit declaration of built-in function 'exit' configure:10992: $? = 0 configure:10998: ./conftest configure:11001: $? = 0 configure:11003: result: yes configure:11026: checking for USB support configure:11064: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:58: warning: incompatible implicit declaration of built-in function 'exit' configure:11067: $? = 0 configure:11073: ./conftest configure:11076: $? = 0 configure:11078: result: yes configure:11175: checking for class_simple configure:11210: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/kobject.h:21, from /lib/modules/2.6.23.11/build/include/linux/device.h:16, from conftest.c:55: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:61: error: 'class_simple_create' undeclared (first use in this function) conftest.c:61: error: (Each undeclared identifier is reported only once conftest.c:61: error: for each function it appears in.) configure:11216: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | #define CONFIG_SND_SUPPORT_OLD_API 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/device.h> | | int | main () | { | | struct class_simple *(*foo)(struct module *, char *) = class_simple_create; | foo(NULL,NULL); | | ; | return 0; | } configure:11227: result: no configure:11244: checking for old driver suspend/resume callbacks configure:11279: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/kobject.h:21, from /lib/modules/2.6.23.11/build/include/linux/device.h:16, from conftest.c:55: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:62: error: too many arguments to function 'driver.resume' configure:11285: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | #define CONFIG_SND_SUPPORT_OLD_API 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/device.h> | | int | main () | { | | static struct device_driver driver; | driver.resume(NULL, 0); | | ; | return 0; | } configure:11296: result: no configure:11313: checking for removal of page-reservation for nopage/mmap configure:11347: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/mmzone.h:7, from /lib/modules/2.6.23.11/build/include/linux/gfp.h:4, from /lib/modules/2.6.23.11/build/include/linux/mm.h:8, from conftest.c:55: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:61: warning: unused variable 'func' configure:11353: $? = 0 configure:11358: result: yes configure:11381: checking for nested class_device configure:11416: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/kobject.h:21, from /lib/modules/2.6.23.11/build/include/linux/device.h:16, from conftest.c:56: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:11422: $? = 0 configure:11427: result: yes configure:11450: checking for new IRQ handler configure:11485: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:8, from conftest.c:57: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness In file included from /lib/modules/2.6.23.11/build/include/linux/irq.h:23, from /lib/modules/2.6.23.11/build/include/asm/hardirq.h:5, from /lib/modules/2.6.23.11/build/include/linux/hardirq.h:7, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:11, from conftest.c:57: /lib/modules/2.6.23.11/build/include/asm/irq.h:15:25: error: irq_vectors.h: No such file or directory In file included from /lib/modules/2.6.23.11/build/include/asm/hardirq.h:5, from /lib/modules/2.6.23.11/build/include/linux/hardirq.h:7, from /lib/modules/2.6.23.11/build/include/linux/interrupt.h:11, from conftest.c:57: /lib/modules/2.6.23.11/build/include/linux/irq.h: At top level: /lib/modules/2.6.23.11/build/include/linux/irq.h:180: error: 'NR_IRQS' undeclared here (not in a function) configure:11491: $? = 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | #define CONFIG_SND_SUPPORT_OLD_API 1 | #define CONFIG_SND_REMOVE_PAGE_RESERVE 1 | #define CONFIG_SND_NESTED_CLASS_DEVICE 1 | /* end confdefs.h. */ | | #define __KERNEL__ | #include <linux/config.h> | #include <linux/interrupt.h> | | int | main () | { | | irq_handler_t handle = 0; | handle(0, 0); | | ; | return 0; | } configure:11502: result: no configure:11518: checking for gfp_t configure:11553: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/asm/thread_info.h:16, from /lib/modules/2.6.23.11/build/include/linux/thread_info.h:34, from /lib/modules/2.6.23.11/build/include/linux/preempt.h:9, from /lib/modules/2.6.23.11/build/include/linux/spinlock.h:49, from /lib/modules/2.6.23.11/build/include/linux/mmzone.h:7, from /lib/modules/2.6.23.11/build/include/linux/gfp.h:4, from conftest.c:58: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness conftest.c: In function 'main': conftest.c:64: warning: unused variable 'flags' configure:11559: $? = 0 configure:11564: result: yes configure:11586: checking for PnP suspend/resume configure:11621: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/kobject.h:21, from /lib/modules/2.6.23.11/build/include/linux/device.h:16, from /lib/modules/2.6.23.11/build/include/linux/pnp.h:11, from conftest.c:58: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:11627: $? = 0 configure:11632: result: yes configure:11656: checking for new unlocked/compat_ioctl configure:11692: gcc -c -Wall -fno-strict-aliasing -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include -Iinclude -nostdinc -iwithprefix include -DKBUILD_STR(s)=#s -DKBUILD_BASENAME=KBUILD_STR(dummy) -DKBUILD_MODNAME=KBUILD_BASENAME conftest.c >&5 In file included from /lib/modules/2.6.23.11/build/include/linux/prefetch.h:14, from /lib/modules/2.6.23.11/build/include/linux/list.h:8, from /lib/modules/2.6.23.11/build/include/linux/wait.h:22, from /lib/modules/2.6.23.11/build/include/linux/fs.h:272, from conftest.c:59: /lib/modules/2.6.23.11/build/include/asm/processor.h: In function 'cpuid_count': /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 1 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 2 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 3 of 'native_cpuid' differ in signedness /lib/modules/2.6.23.11/build/include/asm/processor.h:611: warning: pointer targets in passing argument 4 of 'native_cpuid' differ in signedness configure:11698: $? = 0 configure:11703: result: yes configure:11735: checking for x86-compatible PC configure:11773: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:63: warning: incompatible implicit declaration of built-in function 'exit' configure:11776: $? = 0 configure:11782: ./conftest configure:11785: $? = 0 configure:11787: result: yes configure:11810: checking for High-Res timers configure:11848: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:63: warning: incompatible implicit declaration of built-in function 'exit' configure:11851: $? = 0 configure:11857: ./conftest configure:11860: $? = 0 configure:11862: result: yes configure:11905: checking for PCMCIA support configure:11943: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:61: warning: incompatible implicit declaration of built-in function 'exit' configure:11946: $? = 0 configure:11952: ./conftest configure:11955: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | #define CONFIG_SND_SUPPORT_OLD_API 1 | #define CONFIG_SND_REMOVE_PAGE_RESERVE 1 | #define CONFIG_SND_NESTED_CLASS_DEVICE 1 | #define CONFIG_HAVE_GFP_T 1 | #define CONFIG_HAVE_PNP_SUSPEND 1 | #define CONFIG_SND_HAVE_NEW_IOCTL 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(CONFIG_PCMCIA) && !defined(CONFIG_PCMCIA_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:11965: result: no configure:11979: checking for PCMCIA module support configure:12017: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:61: warning: incompatible implicit declaration of built-in function 'exit' configure:12020: $? = 0 configure:12026: ./conftest configure:12029: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | #define CONFIG_SND_SUPPORT_OLD_API 1 | #define CONFIG_SND_REMOVE_PAGE_RESERVE 1 | #define CONFIG_SND_NESTED_CLASS_DEVICE 1 | #define CONFIG_HAVE_GFP_T 1 | #define CONFIG_HAVE_PNP_SUSPEND 1 | #define CONFIG_SND_HAVE_NEW_IOCTL 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(CONFIG_PCMCIA_MODULE) && !defined(CONFIG_PCMCIA_MODULE_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:12039: result: no configure:12073: checking for PC9800 support in kernel configure:12111: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:61: warning: incompatible implicit declaration of built-in function 'exit' configure:12114: $? = 0 configure:12120: ./conftest configure:12123: $? = 1 configure: program exited with status 1 configure: failed program was: | /* confdefs.h. */ | #define PACKAGE_NAME "" | #define PACKAGE_TARNAME "" | #define PACKAGE_VERSION "" | #define PACKAGE_STRING "" | #define PACKAGE_BUGREPORT "" | #define STDC_HEADERS 1 | #define TIME_WITH_SYS_TIME 1 | #define CONFIG_SND_MODULE 1 | #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" | #define CONFIG_HAVE_DUMP_STACK 1 | #define CONFIG_HAS_IOPORT 1 | #define CONFIG_SND_VERBOSE_PROCFS 1 | #define CONFIG_SND_VERBOSE_PRINTK 1 | #define CONFIG_SND_DEBUG 1 | #define CONFIG_SND_DEBUG_MEMORY 1 | #define CONFIG_SND_ISA 1 | #define CONFIG_ISAPNP_KERNEL 1 | #define CONFIG_PNP_KERNEL 1 | #define CONFIG_SND_ISAPNP 1 | #define CONFIG_SND_PNP 1 | #define CONFIG_HAVE_STRLCPY 1 | #define CONFIG_HAVE_SNPRINTF 1 | #define CONFIG_HAVE_VSNPRINTF 1 | #define CONFIG_HAVE_SCNPRINTF 1 | #define CONFIG_HAVE_SSCANF 1 | #define CONFIG_HAVE_PDE 1 | #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 | #define CONFIG_HAVE_PCI_DEV_PRESENT 1 | #define CONFIG_HAVE_MSLEEP 1 | #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 | #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 | #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 | #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 | #define CONFIG_HAVE_KCALLOC 1 | #define CONFIG_HAVE_KSTRDUP 1 | #define CONFIG_HAVE_KZALLOC 1 | #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 | #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 | #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 | #define CONFIG_SND_VERSION "1.0.15" | #define CONFIG_SND_DATE "" | #define CONFIG_SND_SEQUENCER_MODULE 1 | #define CONFIG_SND_SEQ_DUMMY_MODULE 1 | #define CONFIG_SND_OSSEMUL 1 | #define CONFIG_SND_MIXER_OSS_MODULE 1 | #define CONFIG_SND_PCM_OSS_MODULE 1 | #define CONFIG_SND_SEQUENCER_OSS 1 | #define CONFIG_SND_PCM_OSS_PLUGINS 1 | #define CONFIG_SND_SUPPORT_OLD_API 1 | #define CONFIG_SND_REMOVE_PAGE_RESERVE 1 | #define CONFIG_SND_NESTED_CLASS_DEVICE 1 | #define CONFIG_HAVE_GFP_T 1 | #define CONFIG_HAVE_PNP_SUSPEND 1 | #define CONFIG_SND_HAVE_NEW_IOCTL 1 | /* end confdefs.h. */ | | #include "/lib/modules/2.6.23.11/build/include/linux/autoconf.h" | int main( void ) { | #if !defined(CONFIG_X86_PC9800) && !defined(CONFIG_X86_PC9800_MODULE) | exit(1); | #else | exit(0); | #endif | } | configure:12133: result: no configure:12151: checking for parallel port support configure:12189: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:63: warning: incompatible implicit declaration of built-in function 'exit' configure:12192: $? = 0 configure:12198: ./conftest configure:12201: $? = 0 configure:12203: result: yes configure:12306: checking for power management configure:12344: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:63: warning: incompatible implicit declaration of built-in function 'exit' configure:12347: $? = 0 configure:12353: ./conftest configure:12356: $? = 0 configure:12358: result: yes configure:12383: checking for CONFIG_HAS_DMA configure:12421: gcc -o conftest -g -O2 -I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include conftest.c >&5 conftest.c: In function 'main': conftest.c:63: warning: incompatible implicit declaration of built-in function 'exit' configure:12424: $? = 0 configure:12430: ./conftest configure:12433: $? = 0 configure:12435: result: yes configure:12460: checking for which soundcards to compile driver for configure:14896: result: all configure:16802: creating ./config.status ## ---------------------- ## ## Running config.status. ## ## ---------------------- ## This file was extended by config.status, which was generated by GNU Autoconf 2.61. Invocation command line was CONFIG_FILES = CONFIG_HEADERS = CONFIG_LINKS = CONFIG_COMMANDS = $ ./config.status on viziframe config.status:946: creating version config.status:946: creating Makefile.conf config.status:1081: WARNING: Makefile.conf.in seems to ignore the --datarootdir setting config.status:946: creating snddevices config.status:946: creating utils/alsa-driver.spec config.status:946: creating utils/buildrpm config.status:946: creating toplevel.config config.status:946: creating utils/alsasound config.status:946: creating utils/alsasound.posix config.status:946: creating include/pci_ids_compat.h config.status:946: creating include/i2c-id_compat.h config.status:946: creating include/config.h config.status:946: creating include/config1.h config.status:946: creating include/version.h config.status:946: creating include/autoconf-extra.h ## ---------------- ## ## Cache variables. ## ## ---------------- ## ac_cv_c_compiler_gnu=yes ac_cv_c_const=yes ac_cv_c_inline=inline ac_cv_env_CC_set= ac_cv_env_CC_value= ac_cv_env_CFLAGS_set= ac_cv_env_CFLAGS_value= ac_cv_env_CPPFLAGS_set= ac_cv_env_CPPFLAGS_value= ac_cv_env_CPP_set= ac_cv_env_CPP_value= ac_cv_env_LDFLAGS_set= ac_cv_env_LDFLAGS_value= ac_cv_env_LIBS_set= ac_cv_env_LIBS_value= ac_cv_env_build_alias_set= ac_cv_env_build_alias_value= ac_cv_env_host_alias_set= ac_cv_env_host_alias_value= ac_cv_env_target_alias_set= ac_cv_env_target_alias_value= ac_cv_header_stdc=yes ac_cv_header_time=yes ac_cv_objext=o ac_cv_path_EGREP='/usr/bin/grep -E' ac_cv_path_GREP=/usr/bin/grep ac_cv_path_install='/usr/bin/ginstall -c' ac_cv_prog_CPP='gcc -E' ac_cv_prog_ac_ct_CC=gcc ac_cv_prog_ac_ct_RANLIB=ranlib ac_cv_prog_cc_c89= ac_cv_prog_cc_g=yes ac_cv_prog_gcc_traditional=no ## ----------------- ## ## Output variables. ## ## ----------------- ## AR='ar' ARCH='i386' AS='as' CC='gcc' CFLAGS='-g -O2' CONFIG_AC97_BUS='m' CONFIG_ALPHA='' CONFIG_ARCH_AT91='' CONFIG_ARCH_PXA='' CONFIG_ARCH_S3C2410='' CONFIG_ARCH_SA1100='' CONFIG_ARM='' CONFIG_ARM_AMBA='' CONFIG_ATMEL_SSC='' CONFIG_BROKEN='' CONFIG_CPU_SUBTYPE_SH7760='' CONFIG_EXPERIMENTAL='y' CONFIG_FW_LOADER='y' CONFIG_GSC='' CONFIG_HAS_DMA='y' CONFIG_HAS_IOMEM='' CONFIG_HAS_IOPORT='y' CONFIG_HIGH_RES_TIMERS='y' CONFIG_HPET='y' CONFIG_I2C='' CONFIG_I2C_POWERMAC='' CONFIG_I2C_SENSOR='' CONFIG_INPUT='y' CONFIG_ISA='y' CONFIG_ISAPNP='y' CONFIG_ISAPNP_KERNEL='y' CONFIG_ISA_DMA_API='y' CONFIG_L3='' CONFIG_MACH_ETI_B1='' CONFIG_MACH_ETI_C1='' CONFIG_MACH_NEO1973_GTA01='' CONFIG_MACH_POODLE='' CONFIG_MACH_SMDK2443='' CONFIG_MACH_TOSA='' CONFIG_MIPS='' CONFIG_PARISC='' CONFIG_PARPORT='y' CONFIG_PCI='y' CONFIG_PCMCIA='' CONFIG_PM='y' CONFIG_PNP='y' CONFIG_PNP_KERNEL='y' CONFIG_PPC32='' CONFIG_PPC64='' CONFIG_PPC='' CONFIG_PPC_PMAC='' CONFIG_PROC_FS='y' CONFIG_PS3_PS3AV='' CONFIG_PXA_SHARP_C7XX='' CONFIG_PXA_SHARP_CXX00='' CONFIG_RTC='' CONFIG_SBUS='' CONFIG_SGI='' CONFIG_SH_DMABRG='' CONFIG_SH_DREAMCAST='' CONFIG_SND='m' CONFIG_SND_AC97_CODEC='m' CONFIG_SND_AC97_POWER_SAVE='y' CONFIG_SND_AC97_POWER_SAVE_DEFAULT='0' CONFIG_SND_AD1816A='m' CONFIG_SND_AD1848='m' CONFIG_SND_AD1848_LIB='m' CONFIG_SND_AD1889='m' CONFIG_SND_ADLIB='m' CONFIG_SND_AICA='' CONFIG_SND_ALI5451='m' CONFIG_SND_ALS100='m' CONFIG_SND_ALS300='m' CONFIG_SND_ALS4000='m' CONFIG_SND_AOA='' CONFIG_SND_AOA_FABRIC_LAYOUT='' CONFIG_SND_AOA_ONYX='' CONFIG_SND_AOA_SOUNDBUS='' CONFIG_SND_AOA_SOUNDBUS_I2S='' CONFIG_SND_AOA_TAS='' CONFIG_SND_AOA_TOONIE='' CONFIG_SND_ARMAACI='' CONFIG_SND_ASIHPI='' CONFIG_SND_AT73C213='' CONFIG_SND_AT73C213_TARGET_BITRATE='' CONFIG_SND_AT91_SOC='' CONFIG_SND_AT91_SOC_ETI_B1_WM8731='' CONFIG_SND_AT91_SOC_ETI_SLAVE='' CONFIG_SND_AT91_SOC_SSC='' CONFIG_SND_ATIIXP='m' CONFIG_SND_ATIIXP_MODEM='m' CONFIG_SND_AU1X00='' CONFIG_SND_AU8810='m' CONFIG_SND_AU8820='m' CONFIG_SND_AU8830='m' CONFIG_SND_AZT2320='m' CONFIG_SND_AZT3328='m' CONFIG_SND_BIT32_EMUL='' CONFIG_SND_BT87X='m' CONFIG_SND_BT87X_OVERCLOCK='y' CONFIG_SND_CA0106='m' CONFIG_SND_CMI8330='m' CONFIG_SND_CMIPCI='m' CONFIG_SND_CS4231='m' CONFIG_SND_CS4231_LIB='m' CONFIG_SND_CS4232='m' CONFIG_SND_CS4236='m' CONFIG_SND_CS4281='m' CONFIG_SND_CS46XX='m' CONFIG_SND_CS46XX_NEW_DSP='y' CONFIG_SND_CS5530='m' CONFIG_SND_CS5535AUDIO='' CONFIG_SND_DARLA20='m' CONFIG_SND_DARLA24='m' CONFIG_SND_DATE='' CONFIG_SND_DEBUG='y' CONFIG_SND_DEBUG_DETECT='' CONFIG_SND_DEBUG_MEMORY='y' CONFIG_SND_DT019X='m' CONFIG_SND_DUMMY='m' CONFIG_SND_DYNAMIC_MINORS='' CONFIG_SND_ECHO3G='m' CONFIG_SND_EMU10K1='m' CONFIG_SND_EMU10K1X='m' CONFIG_SND_ENS1370='m' CONFIG_SND_ENS1371='m' CONFIG_SND_ES1688='m' CONFIG_SND_ES18XX='m' CONFIG_SND_ES1938='m' CONFIG_SND_ES1968='m' CONFIG_SND_ES968='m' CONFIG_SND_FM801='m' CONFIG_SND_FM801_TEA575X='m' CONFIG_SND_FM801_TEA575X_BOOL='y' CONFIG_SND_GINA20='m' CONFIG_SND_GINA24='m' CONFIG_SND_GUSCLASSIC='m' CONFIG_SND_GUSEXTREME='m' CONFIG_SND_GUSMAX='m' CONFIG_SND_GUS_SYNTH='m' CONFIG_SND_HARMONY='' CONFIG_SND_HDA_CODEC_ANALOG='y' CONFIG_SND_HDA_CODEC_ATIHDMI='y' CONFIG_SND_HDA_CODEC_CMEDIA='y' CONFIG_SND_HDA_CODEC_CONEXANT='y' CONFIG_SND_HDA_CODEC_REALTEK='y' CONFIG_SND_HDA_CODEC_SI3054='y' CONFIG_SND_HDA_CODEC_SIGMATEL='y' CONFIG_SND_HDA_CODEC_VIA='y' CONFIG_SND_HDA_GENERIC='y' CONFIG_SND_HDA_HWDEP='y' CONFIG_SND_HDA_INTEL='m' CONFIG_SND_HDA_POWER_SAVE='y' CONFIG_SND_HDA_POWER_SAVE_DEFAULT='0' CONFIG_SND_HDSP='m' CONFIG_SND_HDSPM='m' CONFIG_SND_HPET='' CONFIG_SND_HWDEP='m' CONFIG_SND_ICE1712='m' CONFIG_SND_ICE1724='m' CONFIG_SND_INDIGO='m' CONFIG_SND_INDIGODJ='m' CONFIG_SND_INDIGOIO='m' CONFIG_SND_INTEL8X0='m' CONFIG_SND_INTEL8X0M='m' CONFIG_SND_INTERWAVE='m' CONFIG_SND_INTERWAVE_STB='m' CONFIG_SND_KERNELDIR='/lib/modules/2.6.23.11/source' CONFIG_SND_KORG1212='m' CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL='y' CONFIG_SND_LAYLA20='m' CONFIG_SND_LAYLA24='m' CONFIG_SND_LOOPBACK='m' CONFIG_SND_MAESTRO3='m' CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL='y' CONFIG_SND_MIA='m' CONFIG_SND_MIRO='m' CONFIG_SND_MIXART='m' CONFIG_SND_MIXER_OSS='m' CONFIG_SND_ML403_AC97CR='' CONFIG_SND_MONA='m' CONFIG_SND_MPU401='m' CONFIG_SND_MPU401_UART='m' CONFIG_SND_MSND_PINNACLE='' CONFIG_SND_MTPAV='m' CONFIG_SND_MTS64='m' CONFIG_SND_MVERSION='' CONFIG_SND_NM256='m' CONFIG_SND_OPL3SA2='m' CONFIG_SND_OPL3_LIB='m' CONFIG_SND_OPL4_LIB='m' CONFIG_SND_OPTI92X_AD1848='m' CONFIG_SND_OPTI92X_CS4231='m' CONFIG_SND_OPTI93X='m' CONFIG_SND_OSSEMUL='y' CONFIG_SND_OXYGEN='m' CONFIG_SND_OXYGEN_LIB='m' CONFIG_SND_PC98_CS4232='' CONFIG_SND_PCM='m' CONFIG_SND_PCM_OSS='m' CONFIG_SND_PCM_OSS_PLUGINS='y' CONFIG_SND_PCM_XRUN_DEBUG='y' CONFIG_SND_PCSP='' CONFIG_SND_PCXHR='m' CONFIG_SND_PDAUDIOCF='' CONFIG_SND_PDPLUS='m' CONFIG_SND_PORTMAN2X4='m' CONFIG_SND_POWERMAC='' CONFIG_SND_POWERMAC_AUTO_DRC='' CONFIG_SND_PS3='' CONFIG_SND_PS3_DEFAULT_START_DELAY='' CONFIG_SND_PXA2XX_AC97='' CONFIG_SND_PXA2XX_I2SOUND='m' CONFIG_SND_PXA2XX_PCM='' CONFIG_SND_PXA2XX_SOC='' CONFIG_SND_PXA2XX_SOC_AC97='' CONFIG_SND_PXA2XX_SOC_CORGI='' CONFIG_SND_PXA2XX_SOC_I2S='' CONFIG_SND_PXA2XX_SOC_POODLE='' CONFIG_SND_PXA2XX_SOC_SPITZ='' CONFIG_SND_PXA2XX_SOC_TOSA='' CONFIG_SND_RAWMIDI='m' CONFIG_SND_RIPTIDE='m' CONFIG_SND_RME32='m' CONFIG_SND_RME9652='m' CONFIG_SND_RME96='m' CONFIG_SND_RTCTIMER='' CONFIG_SND_S3C2410='' CONFIG_SND_S3C2443_SOC_AC97='' CONFIG_SND_S3C24XX_SOC='' CONFIG_SND_S3C24XX_SOC_I2S='' CONFIG_SND_S3C24XX_SOC_LN2440SBC_ALC650='' CONFIG_SND_S3C24XX_SOC_NEO1973_WM8753='' CONFIG_SND_S3C24XX_SOC_SMDK2443_WM9710='' CONFIG_SND_SA11XX_UDA1341='' CONFIG_SND_SB16='m' CONFIG_SND_SB16_CSP='y' CONFIG_SND_SB16_CSP_FIRMWARE_IN_KERNEL='y' CONFIG_SND_SB16_DSP='m' CONFIG_SND_SB8='m' CONFIG_SND_SB8_DSP='m' CONFIG_SND_SBAWE='m' CONFIG_SND_SB_COMMON='m' CONFIG_SND_SC6000='m' CONFIG_SND_SEQUENCER='m' CONFIG_SND_SEQUENCER_OSS='y' CONFIG_SND_SEQ_DUMMY='m' CONFIG_SND_SEQ_RTCTIMER_DEFAULT='' CONFIG_SND_SERIALMIDI='' CONFIG_SND_SERIAL_U16550='m' CONFIG_SND_SGALAXY='m' CONFIG_SND_SH7760_AC97='' CONFIG_SND_SOC='m' CONFIG_SND_SOC_AC97_BUS='y' CONFIG_SND_SOC_AC97_CODEC='' CONFIG_SND_SOC_CS4270='' CONFIG_SND_SOC_CS4270_HWMUTE='' CONFIG_SND_SOC_CS4270_VD33_ERRATA='' CONFIG_SND_SOC_PCM_SH7760='' CONFIG_SND_SOC_SH4_HAC='' CONFIG_SND_SOC_SH4_SSI='' CONFIG_SND_SOC_TLV320AIC3X='' CONFIG_SND_SOC_WM8731='' CONFIG_SND_SOC_WM8750='' CONFIG_SND_SOC_WM8753='' CONFIG_SND_SOC_WM9712='' CONFIG_SND_SONICVIBES='m' CONFIG_SND_SSCAPE='m' CONFIG_SND_SUN_AMD7930='' CONFIG_SND_SUN_CS4231='' CONFIG_SND_SUN_DBRI='' CONFIG_SND_SUPPORT_OLD_API='y' CONFIG_SND_TIMER='m' CONFIG_SND_TRIDENT='m' CONFIG_SND_USB_AUDIO='m' CONFIG_SND_USB_CAIAQ='m' CONFIG_SND_USB_CAIAQ_INPUT='y' CONFIG_SND_USB_USX2Y='' CONFIG_SND_VERBOSE_PRINTK='y' CONFIG_SND_VERBOSE_PROCFS='y' CONFIG_SND_VERSION='1.0.15' CONFIG_SND_VIA82XX='m' CONFIG_SND_VIA82XX_MODEM='m' CONFIG_SND_VIRMIDI='m' CONFIG_SND_VIRTUOSO='m' CONFIG_SND_VX222='m' CONFIG_SND_VXPOCKET='' CONFIG_SND_VX_LIB='m' CONFIG_SND_WAVEFRONT='m' CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL='y' CONFIG_SND_YMFPCI='m' CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL='y' CONFIG_SOC_AU1000='' CONFIG_SOC_AU1100='' CONFIG_SOC_AU1500='' CONFIG_SOUND='m' CONFIG_SOUND_PRIME='' CONFIG_SPARC32='' CONFIG_SPARC64='' CONFIG_SPARC='' CONFIG_SUPERH64='' CONFIG_SUPERH='' CONFIG_USB='y' CONFIG_VIDEO_DEV='y' CONFIG_VIDEO_V4L1='y' CONFIG_X86='' CONFIG_X86_32='' CONFIG_X86_64='' CONFIG_X86_PC9800='' CONFIG_X86_PC='y' CONFIG_XILINX_VIRTEX='' CPP='gcc -E' CPPFLAGS='' CROSS_COMPILE='' DEFS='-DHAVE_CONFIG_H' ECHO_C='' ECHO_N='-n' ECHO_T='' EGREP='/usr/bin/grep -E' EXEEXT='' EXTRA_INCLUDES='' GENKSYMS='/sbin/genksyms -k 2.6.23' GREP='/usr/bin/grep' INSTALL_DATA='${INSTALL} -m 644' INSTALL_PROGRAM='${INSTALL}' INSTALL_SCRIPT='${INSTALL}' KERNEL_INC='-I/lib/modules/2.6.23.11/build/include -I/lib/modules/2.6.23.11/source/include' KLD='ld' LDFLAGS='' LIBOBJS='' LIBS='' LTLIBOBJS='' MAKE_ADDS='' NEW_KBUILD='y' OBJEXT='o' PACKAGE_BUGREPORT='' PACKAGE_NAME='' PACKAGE_STRING='' PACKAGE_TARNAME='' PACKAGE_VERSION='' PATH_SEPARATOR=':' RANLIB='ranlib' SHELL='/bin/sh' SRCDIR='/usr/src/alsa-driver' ac_ct_CC='gcc' bindir='${exec_prefix}/bin' build_alias='' c_opts='-O2 -march=i386 -Wdeclaration-after-statement -Wno-pointer-sign -DLINUX' datadir='${datarootdir}' datarootdir='${prefix}/share' docdir='${datarootdir}/doc/${PACKAGE}' dvidir='${docdir}' exec_prefix='${prefix}' host_alias='' htmldir='${docdir}' includedir='${prefix}/include' infodir='${datarootdir}/info' kaversion='2.6.23.11' kextraversion='.11' kpatchlevel='6' ksublevel='23' kversion='2' libdir='${exec_prefix}/lib' libexecdir='${exec_prefix}/libexec' localedir='${datarootdir}/locale' localstatedir='${prefix}/var' m_opts='' mandir='${datarootdir}/man' moddir='/lib/modules/2.6.23.11/kernel/sound' moddir_tree='y' modsubdir='kernel/sound' msmp='' oldincludedir='/usr/include' pdfdir='${docdir}' prefix='/usr' processor='x86_32' program_transform_name='s,x,x,' psdir='${docdir}' sbindir='${exec_prefix}/sbin' sharedstatedir='${prefix}/com' sysconfdir='${prefix}/etc' target_alias='' ## ----------- ## ## confdefs.h. ## ## ----------- ## #define PACKAGE_NAME "" #define PACKAGE_TARNAME "" #define PACKAGE_VERSION "" #define PACKAGE_STRING "" #define PACKAGE_BUGREPORT "" #define STDC_HEADERS 1 #define TIME_WITH_SYS_TIME 1 #define CONFIG_SND_MODULE 1 #define CONFIG_SND_KERNELDIR "/lib/modules/2.6.23.11/source" #define CONFIG_HAVE_DUMP_STACK 1 #define CONFIG_HAS_IOPORT 1 #define CONFIG_SND_VERBOSE_PROCFS 1 #define CONFIG_SND_VERBOSE_PRINTK 1 #define CONFIG_SND_DEBUG 1 #define CONFIG_SND_DEBUG_MEMORY 1 #define CONFIG_SND_ISA 1 #define CONFIG_ISAPNP_KERNEL 1 #define CONFIG_PNP_KERNEL 1 #define CONFIG_SND_ISAPNP 1 #define CONFIG_SND_PNP 1 #define CONFIG_HAVE_STRLCPY 1 #define CONFIG_HAVE_SNPRINTF 1 #define CONFIG_HAVE_VSNPRINTF 1 #define CONFIG_HAVE_SCNPRINTF 1 #define CONFIG_HAVE_SSCANF 1 #define CONFIG_HAVE_PDE 1 #define CONFIG_HAVE_PCI_CONSISTENT_DMA_MASK 1 #define CONFIG_HAVE_PCI_DEV_PRESENT 1 #define CONFIG_HAVE_MSLEEP 1 #define CONFIG_HAVE_MSLEEP_INTERRUPTIBLE 1 #define CONFIG_HAVE_MSECS_TO_JIFFIES 1 #define CONFIG_HAVE_VIDEO_GET_DRVDATA 1 #define CONFIG_HAVE_IO_REMAP_PFN_RANGE 1 #define CONFIG_HAVE_KCALLOC 1 #define CONFIG_HAVE_KSTRDUP 1 #define CONFIG_HAVE_KZALLOC 1 #define CONFIG_HAVE_PCI_SAVED_CONFIG 1 #define CONFIG_HAVE_NEW_PCI_SAVE_STATE 1 #define CONFIG_HAVE_REGISTER_SOUND_SPECIAL_DEVICE 1 #define CONFIG_SND_VERSION "1.0.15" #define CONFIG_SND_DATE "" #define CONFIG_SND_SEQUENCER_MODULE 1 #define CONFIG_SND_SEQ_DUMMY_MODULE 1 #define CONFIG_SND_OSSEMUL 1 #define CONFIG_SND_MIXER_OSS_MODULE 1 #define CONFIG_SND_PCM_OSS_MODULE 1 #define CONFIG_SND_SEQUENCER_OSS 1 #define CONFIG_SND_PCM_OSS_PLUGINS 1 #define CONFIG_SND_SUPPORT_OLD_API 1 #define CONFIG_SND_REMOVE_PAGE_RESERVE 1 #define CONFIG_SND_NESTED_CLASS_DEVICE 1 #define CONFIG_HAVE_GFP_T 1 #define CONFIG_HAVE_PNP_SUSPEND 1 #define CONFIG_SND_HAVE_NEW_IOCTL 1 #define CONFIG_SOUND_MODULE 1 #define CONFIG_SND_TIMER_MODULE 1 #define CONFIG_SND_PCM_MODULE 1 #define CONFIG_SND_HWDEP_MODULE 1 #define CONFIG_SND_RAWMIDI_MODULE 1 #define CONFIG_SND_SEQ_DUMMY_MODULE 1 #define CONFIG_SND_SUPPORT_OLD_API 1 #define CONFIG_PROC_FS_MODULE 1 #define CONFIG_SND_PCM_XRUN_DEBUG 1 #define CONFIG_HPET_MODULE 1 #define CONFIG_SND_MPU401_UART_MODULE 1 #define CONFIG_SND_OPL3_LIB_MODULE 1 #define CONFIG_SND_OPL4_LIB_MODULE 1 #define CONFIG_SND_VX_LIB_MODULE 1 #define CONFIG_SND_AC97_CODEC_MODULE 1 #define CONFIG_SND_DUMMY_MODULE 1 #define CONFIG_SND_VIRMIDI_MODULE 1 #define CONFIG_SND_MTPAV_MODULE 1 #define CONFIG_SND_MTS64_MODULE 1 #define CONFIG_PARPORT_MODULE 1 #define CONFIG_SND_SERIAL_U16550_MODULE 1 #define CONFIG_SND_MPU401_MODULE 1 #define CONFIG_SND_PORTMAN2X4_MODULE 1 #define CONFIG_SND_LOOPBACK_MODULE 1 #define CONFIG_X86_PC_MODULE 1 #define CONFIG_HIGH_RES_TIMERS_MODULE 1 #define CONFIG_SND_AD1848_LIB_MODULE 1 #define CONFIG_SND_CS4231_LIB_MODULE 1 #define CONFIG_SND_SB_COMMON_MODULE 1 #define CONFIG_SND_SB8_DSP_MODULE 1 #define CONFIG_SND_SB16_DSP_MODULE 1 #define CONFIG_SND_ADLIB_MODULE 1 #define CONFIG_SND_AD1816A_MODULE 1 #define CONFIG_PNP_MODULE 1 #define CONFIG_SND_AD1848_MODULE 1 #define CONFIG_SND_ALS100_MODULE 1 #define CONFIG_SND_AZT2320_MODULE 1 #define CONFIG_SND_CMI8330_MODULE 1 #define CONFIG_SND_CS4231_MODULE 1 #define CONFIG_SND_CS4232_MODULE 1 #define CONFIG_SND_CS4236_MODULE 1 #define CONFIG_SND_DT019X_MODULE 1 #define CONFIG_SND_ES968_MODULE 1 #define CONFIG_SND_ES1688_MODULE 1 #define CONFIG_SND_ES18XX_MODULE 1 #define CONFIG_SND_SC6000_MODULE 1 #define CONFIG_SND_GUS_SYNTH_MODULE 1 #define CONFIG_SND_GUSCLASSIC_MODULE 1 #define CONFIG_SND_GUSEXTREME_MODULE 1 #define CONFIG_SND_GUSMAX_MODULE 1 #define CONFIG_SND_INTERWAVE_MODULE 1 #define CONFIG_SND_INTERWAVE_STB_MODULE 1 #define CONFIG_SND_OPL3SA2_MODULE 1 #define CONFIG_SND_OPTI92X_AD1848_MODULE 1 #define CONFIG_SND_OPTI92X_CS4231_MODULE 1 #define CONFIG_SND_OPTI93X_MODULE 1 #define CONFIG_SND_MIRO_MODULE 1 #define CONFIG_SND_SB8_MODULE 1 #define CONFIG_SND_SB16_MODULE 1 #define CONFIG_SND_SBAWE_MODULE 1 #define CONFIG_SND_SB16_CSP 1 #define CONFIG_SND_SB16_CSP_FIRMWARE_IN_KERNEL 1 #define CONFIG_SND_SGALAXY_MODULE 1 #define CONFIG_SND_SSCAPE_MODULE 1 #define CONFIG_SND_WAVEFRONT_MODULE 1 #define CONFIG_SND_WAVEFRONT_FIRMWARE_IN_KERNEL 1 #define CONFIG_SND_AD1889_MODULE 1 #define CONFIG_SND_ALS300_MODULE 1 #define CONFIG_SND_ALS4000_MODULE 1 #define CONFIG_SND_ALI5451_MODULE 1 #define CONFIG_SND_ATIIXP_MODULE 1 #define CONFIG_SND_ATIIXP_MODEM_MODULE 1 #define CONFIG_SND_AU8810_MODULE 1 #define CONFIG_SND_AU8820_MODULE 1 #define CONFIG_SND_AU8830_MODULE 1 #define CONFIG_SND_AZT3328_MODULE 1 #define CONFIG_SND_BT87X_MODULE 1 #define CONFIG_SND_BT87X_OVERCLOCK 1 #define CONFIG_SND_CA0106_MODULE 1 #define CONFIG_SND_CMIPCI_MODULE 1 #define CONFIG_SND_CS4281_MODULE 1 #define CONFIG_SND_CS46XX_MODULE 1 #define CONFIG_SND_CS46XX_NEW_DSP 1 #define CONFIG_SND_CS5530_MODULE 1 #define CONFIG_SND_DARLA20_MODULE 1 #define CONFIG_SND_GINA20_MODULE 1 #define CONFIG_SND_LAYLA20_MODULE 1 #define CONFIG_SND_DARLA24_MODULE 1 #define CONFIG_SND_GINA24_MODULE 1 #define CONFIG_SND_LAYLA24_MODULE 1 #define CONFIG_SND_MONA_MODULE 1 #define CONFIG_SND_MIA_MODULE 1 #define CONFIG_SND_ECHO3G_MODULE 1 #define CONFIG_SND_INDIGO_MODULE 1 #define CONFIG_SND_INDIGOIO_MODULE 1 #define CONFIG_SND_INDIGODJ_MODULE 1 #define CONFIG_SND_EMU10K1_MODULE 1 #define CONFIG_SND_EMU10K1X_MODULE 1 #define CONFIG_SND_ENS1370_MODULE 1 #define CONFIG_SND_ENS1371_MODULE 1 #define CONFIG_SND_ES1938_MODULE 1 #define CONFIG_SND_ES1968_MODULE 1 #define CONFIG_SND_FM801_MODULE 1 #define CONFIG_SND_FM801_TEA575X_BOOL 1 #define CONFIG_SND_FM801_TEA575X_MODULE 1 #define CONFIG_SND_HDA_INTEL_MODULE 1 #define CONFIG_SND_HDA_HWDEP 1 #define CONFIG_SND_HDA_CODEC_REALTEK 1 #define CONFIG_SND_HDA_CODEC_ANALOG 1 #define CONFIG_SND_HDA_CODEC_SIGMATEL 1 #define CONFIG_SND_HDA_CODEC_VIA 1 #define CONFIG_SND_HDA_CODEC_ATIHDMI 1 #define CONFIG_SND_HDA_CODEC_CONEXANT 1 #define CONFIG_SND_HDA_CODEC_CMEDIA 1 #define CONFIG_SND_HDA_CODEC_SI3054 1 #define CONFIG_SND_HDA_GENERIC 1 #define CONFIG_SND_HDA_POWER_SAVE 1 #define CONFIG_SND_HDA_POWER_SAVE_DEFAULT 0 #define CONFIG_SND_HDSP_MODULE 1 #define CONFIG_SND_HDSPM_MODULE 1 #define CONFIG_SND_ICE1712_MODULE 1 #define CONFIG_SND_ICE1724_MODULE 1 #define CONFIG_SND_INTEL8X0_MODULE 1 #define CONFIG_SND_INTEL8X0M_MODULE 1 #define CONFIG_SND_KORG1212_MODULE 1 #define CONFIG_SND_KORG1212_FIRMWARE_IN_KERNEL 1 #define CONFIG_SND_MAESTRO3_MODULE 1 #define CONFIG_SND_MAESTRO3_FIRMWARE_IN_KERNEL 1 #define CONFIG_SND_MIXART_MODULE 1 #define CONFIG_SND_NM256_MODULE 1 #define CONFIG_SND_PCXHR_MODULE 1 #define CONFIG_SND_RIPTIDE_MODULE 1 #define CONFIG_SND_RME32_MODULE 1 #define CONFIG_SND_RME96_MODULE 1 #define CONFIG_SND_RME9652_MODULE 1 #define CONFIG_SND_SONICVIBES_MODULE 1 #define CONFIG_SND_TRIDENT_MODULE 1 #define CONFIG_SND_VIA82XX_MODULE 1 #define CONFIG_SND_VIA82XX_MODEM_MODULE 1 #define CONFIG_SND_VX222_MODULE 1 #define CONFIG_SND_YMFPCI_MODULE 1 #define CONFIG_SND_YMFPCI_FIRMWARE_IN_KERNEL 1 #define CONFIG_SND_AC97_POWER_SAVE 1 #define CONFIG_SND_AC97_POWER_SAVE_DEFAULT 0 #define CONFIG_SND_PDPLUS_MODULE 1 #define CONFIG_SND_OXYGEN_LIB_MODULE 1 #define CONFIG_SND_VIRTUOSO_MODULE 1 #define CONFIG_SND_OXYGEN_MODULE 1 #define CONFIG_SND_PXA2XX_I2SOUND_MODULE 1 #define CONFIG_SND_USB_AUDIO_MODULE 1 #define CONFIG_SND_USB_CAIAQ_MODULE 1 #define CONFIG_SND_USB_CAIAQ_INPUT 1 #define CONFIG_SND_SOC_AC97_BUS 1 #define CONFIG_SND_SOC_MODULE 1 configure: exit 0 -- Thank you again for your help, and please let me know if I can help in any way... -- Darryl House -------------- Original message ---------------------- From: Takashi Iwai <tiwai(a)suse.de> > At Thu, 20 Dec 2007 00:37:46 +0000, > forumjunk(a)comcast.net wrote: > > > > > > For VIA platforms, to avoid the driver compile error in pci.h, reconfigure > your kernel to remove SMP support, and set the architecture type to > PC-compatible. > > > > (you already know this, I'm sure): > > Recompile the kernel (as a paranoid, I always trash > /lib/modules/<kernel_version> before making modules_install). Install the > kernel, of course, and reconfigure your boot loader and/or initrd and reboot. > > > > There are still several fixes that need to be made to get all of this to > compile cleanly, but I haven't got the permanent fixes nailed down yet. I'm > almost positive that these are alsa-driver configure errors or omissions. I > can't (yet) find a kernel config that will avoid them. The fixes below are > harmless, and they will allow you to get the driver compiled. > > My guess is that the error of configure script is the only point we > need to fix. > > > Add this definition to the top section of acore/memory_wrapper.c after the > line that reads #include <linux/version.h> > > > > #define VMALLOC_VMADDR(x) ((unsigned long)(x)) > > > > Change these lines (45 and 46 or close to that) of acore/memory_wrapper.c > from: > > > > pmd = pmd_offset(pgd, lpage); > > pte = pte_offset(pmd, lpage); > > > > to > > > > pmd = pmd_offset((pud_t *)pgd, lpage); > > pte = pte_offset_kernel(pmd, lpage); > > > > The driver will then compile cleanly (at least with no errors or warnings > relating to any of these changes) > > Don't do this. This part shouldn't be compiled at all for recent > kernels. This is, again, the problem of configure script that > detected the codes wrongly. > > > So, could you guys show config.log? Otherwise it's hard to know > what's really wrong. > > > thanks, > > Takashi

1 0

Re: [alsa-devel] asihpi driver -> kernel
by Takashi Iwai 20 Dec '07

20 Dec '07

Hi Eliot, [it seems that you Cc'ed to the old alsa-devel ML so the last one seemed missing.] At Tue, 16 Oct 2007 14:47:15 +1300, Eliot Blennerhassett wrote: > > Hi Takashi, > > still trying to move forward on this... > > Looking for "what is the first step" change to make in your opinion? The first step would be to get closer to linux kernel coding style. Namely, - get rid of C++ style function names, variables and field names (no hungarian!) - remove API wrappers Eventually, we may get rid of unneeded typedefs, too. Then, we can begin with simplifying the driver structure. Currently, asihpi and hpi are almost indepedent implementations. This makes life harder. > Takashi Iwai wrote: > > At Wed, 05 Sep 2007 13:00:35 +1200, > > Eliot Blennerhassett wrote: > >> Hi Takashi, > >> > >> I'd like to talk more about what you think is needed to get asihpi driver > >> ready for kernel tree. > >> > >> I think for now we should put aside cosmetic coding style changes (eg. comment > >> style, whitespace etc.) these are fixable with combination of automated tools > >> and some monkey work. > >> > >> So, lets start with the hard part. If we can't solve that there is no point in > >> cosmetics! > >> > >> A while ago you said "The specific problem I'm concerned is the "glue-code" > >> style in HPI driver. The OS-independent wrapper code is often disliked by > >> kernel maintainers. " > >> > >> Can you point to specific examples. Even better suggest how you think it > >> should be done better. Either way I need to consider how your concerns could > >> be addressed without breaking our codebase for other OSes etc. > >> > >> thanks and regards > > > > The problem is that the whole "wrapping" is superfluous when you > > concentrate on Linux driver. Merging into the linux kernel tree means > > that the driver would be dedicated to Linux. It's no longer your > > child, and it's no longer for every other OS. Thus, "common codebase" > > doesn't exist per se. > > > > For example, the whole definitions in hpios_linux_kernel.h should go > > away. The driver should use the native kernel API instead. > > Even this is not so hard, as it a mechanical process to replace wrapped > calls with linux native calls. OK, then it's fine. > > I know it's a very hard decision. Supporting other OSes is very > > important in most cases. OTOH, the slim-down is the best way to make > > the driver code maintainable for other people. Honestly, I find it > > also hard to maintain/debug the current ASIHPI code. It's too > > different from the normal codes, and understanding it alone takes too > > much energy before analyzing the problem. > > This is where I need more input, because to _me_ (of course) the code is > understandable. > > Is it the whole source or just parts that are hard. > E.g toplevel asihpi.c Ok/No? The code in asihpi.c is readable enough. But, other parts are different. The difficulty is that the driver structure isn't straight (apart from the coding style issues). We have an entry point for PCI driver _and_ indepdendent HPI initialization. This is already puzzling because HPI is also assigned to the corresponding PCI device. Also, the message union/struct is another stuff that is pretty hard to follow. Since we use no C++, union is often bad for readability. Could it be a bit simpler? > > The coding-style issues are similar problems. When C++-style codes > > appear in the tree of C-codes, they are strange. Simply so. It makes > > difficult to find the problematic part. "A buggy part looks buggy" -- > > it's the whole point of having a uniform coding style. > > Again, a mechanical process to change comments and style etc. Then it's fine with me. Let's try and see the progress. > > These problems (different from others) make it impossible to merge to > > the upstream tree, so far. > > > > > > > >> Eliot > >> BTW I'm happy to conduct this discussion publically on alsa-devel, some other > >> people might learn something too... > > > > I think it's a good thing, too. Feel free to add alsa-devel to Cc in > > the next reply with citation. > > > >> For reference here is the source tree diagram. > >> > >> asihpi.c alsa driver toplevel > >> | > >> hpifunc.c message/response pack/unpack > >> | > >> hpimsgx.c hpi stream ownership, backend selection, some init > >> | > >> hpi6205.c hpi6000.c hpi4000.c card family drivers > >> | > >> hpicmn.c, hpios_linux.c hpidspcd.c hpidebug.c library code > >> > >> hpimod.c probing, HPI ioctl > >> > > > > > > thanks, > > > > Takashi thanks, Takashi

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[alsa-devel] [PATCH] The alc268 input_mux should be a selector instead of mixer
by Jiang zhe 20 Dec '07

20 Dec '07

Hi, According to the [0003659], the node 0x23,0x24 is a selector. I checked the alc268 spec on the REALTEK website and it showed that they were selectors indeed. However, current code implement the alc268 input_mux in a mixer way. Signed-off-by: Jiang Zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20071219/alsa-kernel/pci/hda/patch_realtek.c 2007-12-18 09:00:08.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_realtek.c 2007-12-20 10:09:11.000000000 +0800 @@ -9196,19 +9196,13 @@ static struct hda_verb alc268_base_init_ {0x1c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, {0x1d, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - /* FIXME: use matrix-type input source selection */ - /* Mixer elements: 0x18, 19, 1a, 1c, 14, 15, 0b */ - /* Input mixer1: unmute Mic, F-Mic, Line, CD inputs */ - /* Input mixer2 */ - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x23, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, - - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x00 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x01 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x02 << 8))}, - {0x24, AC_VERB_SET_AMP_GAIN_MUTE, (0x7000 | (0x03 << 8))}, + /* Unmute Selector 23h,24h and set the default input to mic-in */ + + {0x23, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x24, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x24, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + { } }; @@ -9253,29 +9247,14 @@ static int alc268_mux_enum_put(struct sn { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct alc_spec *spec = codec->spec; - const struct hda_input_mux *imux = spec->input_mux; + unsigned int adc_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); static hda_nid_t capture_mixers[3] = { 0x23, 0x24 }; hda_nid_t nid = capture_mixers[adc_idx]; - unsigned int *cur_val = &spec->cur_mux[adc_idx]; - unsigned int i, idx; - idx = ucontrol->value.enumerated.item[0]; - if (idx >= imux->num_items) - idx = imux->num_items - 1; - if (*cur_val == idx) - return 0; - for (i = 0; i < imux->num_items; i++) { - unsigned int v = (i == idx) ? 0 : HDA_AMP_MUTE; - snd_hda_codec_amp_stereo(codec, nid, HDA_INPUT, - imux->items[i].index, - HDA_AMP_MUTE, v); - snd_hda_codec_write_cache(codec, nid, 0, - AC_VERB_SET_CONNECT_SEL, - idx ); - } - *cur_val = idx; - return 1; + return snd_hda_input_mux_put(codec, spec->input_mux, ucontrol, + nid, + &spec->cur_mux[adc_idx]); } static struct snd_kcontrol_new alc268_capture_alt_mixer[] = {

2 1

[alsa-devel] [PATCH]Device ID for Toshiba laptop which uses AD1986A
by Jiang zhe 20 Dec '07

20 Dec '07

Hi, Please refer to the [0003662]. The model laptop-eapd get rid of the high-pitched noise. Signed-Off-by: Jiang Zhe <zhe.jiang(a)intel.com> --- alsa-driver-hg20071219/alsa-kernel/pci/hda/patch_analog.c 2007-12-20 09:40:37.000000000 +0800 +++ b/alsa-kernel/pci/hda/patch_analog.c 2007-12-20 09:23:05.000000000 +0800 @@ -859,6 +859,7 @@ static struct snd_pci_quirk ad1986a_cfg_ SND_PCI_QUIRK(0x17aa, 0x1017, "Lenovo A60", AD1986A_3STACK), SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo N100", AD1986A_LAPTOP_AUTOMUTE), SND_PCI_QUIRK(0x17c0, 0x2017, "Samsung M50", AD1986A_LAPTOP), + SND_PCI_QUIRK(0x1179, 0xff40, "Toshiba", AD1986A_LAPTOP_EAPD), {} };

2 1

[alsa-devel] Event notification
by Miloslaw Smyk 20 Dec '07

20 Dec '07

Hi, My app needs to know about the events occurring to a sound card. Specifically, I am interested in start, stop, pause and volume change. While I could monitor volume by similar means as alsamixer does (is there any doc on this apart from the source code?), I am not sure if monitoring other events is at all possible. I would appreciate any suggestions as to how to do this. Kind Regards, Milek

2 1

[alsa-devel] pcm file question
by Leigh Stewart 20 Dec '07

20 Dec '07

hey all, im writing a little app that needs to tie into the audio in/out-put from another alsa application. in other words i need to take all audio produced by the app and do some work with it, and supply data to its audio input. im not sure but i think the pcm_file plugin will allow me to do what i need to do. i.e. i as i understand it i could tie the audio output to a fifo file using the plugin, and then read from the file. anyway im not sure how write a program that uses the plugin code. i wrote something like ... > #include <alsa/asoundlib.h> > #include <iostream> > #include <alsa/pcm.h> > #include <alsa/pcm_plugin.h> > ... > err = snd_pcm_file_open (&pcm_handle, pcm_name, (char*)0, fdo, (char*)0, > fdi, "raw", 777, capture_handle, 0); > if(err < 0) { > fprintf (stderr, "cannot open pcm_file device %s (%s)\n", pcm_name, > snd_strerror (err)); > exit (1); > } but there are no method signatures in pcm_plugin.h. i've tried building alsa from source 1.0.15. i see method signatures for copy and dmix, and even jack, but not for file. how can i compile the plugin in? can i even use pcm_file programmatically? can anyone refer me to any examples? thanks, lstewart

1 0

[alsa-devel] [PATCH 3/12] neo1973: ASoC include pathname fix
by Harald Welte 19 Dec '07

19 Dec '07

[PATCH] Fix s3c24xx include file path changes in asoc driver Signed-off-by: Harald Welte <laforge(a)openmoko.org> --- Index: linux-2.6.22/sound/soc/s3c24xx/neo1973_wm8753.c =================================================================== --- linux-2.6.22.orig/sound/soc/s3c24xx/neo1973_wm8753.c +++ linux-2.6.22/sound/soc/s3c24xx/neo1973_wm8753.c @@ -30,13 +30,15 @@ #include <asm/mach-types.h> #include <asm/hardware/scoop.h> -#include <asm/arch/regs-iis.h> #include <asm/arch/regs-clock.h> #include <asm/arch/regs-gpio.h> #include <asm/hardware.h> #include <asm/arch/audio.h> #include <asm/io.h> #include <asm/arch/spi-gpio.h> + +#include <asm/plat-s3c24xx/regs-iis.h> + #include "../codecs/wm8753.h" #include "lm4857.h" #include "s3c24xx-pcm.h" Index: linux-2.6.22/sound/soc/s3c24xx/s3c24xx-i2s.c =================================================================== --- linux-2.6.22.orig/sound/soc/s3c24xx/s3c24xx-i2s.c +++ linux-2.6.22/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -33,13 +33,14 @@ #include <asm/hardware.h> #include <asm/io.h> -#include <asm/arch/regs-iis.h> #include <asm/arch/regs-gpio.h> #include <asm/arch/regs-clock.h> #include <asm/arch/audio.h> #include <asm/dma.h> #include <asm/arch/dma.h> +#include <asm/plat-s3c24xx/regs-iis.h> + #include "s3c24xx-pcm.h" #include "s3c24xx-i2s.h" -- - Harald Welte <laforge(a)openmoko.org> http://openmoko.org/ ============================================================================ Software for the world's first truly open Free Software mobile phone

2 2

[alsa-devel] snd-hda-intel support for SPDIF-in wiht vt1708 card?
by John Stile 19 Dec '07

19 Dec '07

I am trying to use a Via VT1708 audio card ( uses the snd-hda-intel.ko driver) to record from SPDIF-in. An off-list reply to my posting to alsa-users list suggested that snd-hda-intel.ko might not support SPDIF-in with the VT1708 card, which brought me here. Can someone who works on the Intel HDA driver tell me if it supports SPDIF-in on the VT1708 audio card? Should 'arecord -L' show "Digital Audio Input"? Environment: Audio Card: VT1708 ALSA version: 1.0.15 (as gentoo packages) Driver: snd-hda-intel.ko (In-kernel) Kernel: 2.6.23-gentoo-r3 Distribution: Gentoo Systemboard: PX10000G (VIA pico-itx) Case: Serener GS-L08 case

3 7

[alsa-devel] interesting message, but not sure where its coming from
by Christopher Johnson 19 Dec '07

19 Dec '07

I'm working on a complicated program that uses the ALSA library in an embedded environment. I have things working pretty well, although occasionally I get the following message printed on the console: buffer empty, loop_cnt 4727, loop_cnt_of_last_copy 4682, last period elapsed 4700, elapsed flag 0, chip_data->prev_cnt F000 buffer empty, loop_cnt 4758, loop_cnt_of_last_copy 4727, last period elapsed 4700, elapsed flag 0, chip_data->prev_cnt F000 I know this isn't my code, and I can't find anything that looks like this in the alsa library or the kernel. Does anybody recognize this?

2 1

[alsa-devel] Fix capture mixers of ALC662 models
by Herton Ronaldo Krzesinski 19 Dec '07

19 Dec '07

The below patches fixes capture mixers of ALC662 models. The first one applies to current hg tree, the second one to Linus tree (2.6.24-rcX). The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> --- alsa-kernel/pci/hda/patch_realtek.c.orig 2007-12-18 21:57:02.000000000 -0200 +++ alsa-kernel/pci/hda/patch_realtek.c 2007-12-18 22:01:24.000000000 -0200 @@ -12364,17 +12364,6 @@ static struct snd_kcontrol_new alc662_3S HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12398,17 +12387,6 @@ static struct snd_kcontrol_new alc662_3S HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12422,17 +12400,6 @@ static struct snd_kcontrol_new alc662_le HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -12463,6 +12430,25 @@ static struct snd_kcontrol_new alc662_ch { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc662_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, + }, + { } /* end */ +}; + static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -12585,26 +12571,6 @@ static struct hda_verb alc662_auto_init_ { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -12746,8 +12712,7 @@ static struct alc_config_preset alc662_p .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, The commit that added support for ASUS P701 eeepc also changed the mixers of other ALC662 models, duplicating entries for the Capture items, making them to not work anymore. This fixes it by removing duplicated entries using where possible the common alc662_capture_mixer. Also alc662_capture_mixer should use alc662* functions and not alc882 (I checked /proc/asound/card0/codec* on an eepc model and it's ok). Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com> diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1c50278..85e54d1 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -11728,17 +11728,6 @@ static struct snd_kcontrol_new alc662_3ST_2ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11762,17 +11751,6 @@ static struct snd_kcontrol_new alc662_3ST_6ch_mixer[] = { HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_VOLUME("PC Speaker Playback Volume", 0x0b, 0x05, HDA_INPUT), HDA_CODEC_MUTE("PC Speaker Playback Switch", 0x0b, 0x05, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11786,17 +11764,6 @@ static struct snd_kcontrol_new alc662_lenovo_101e_mixer[] = { HDA_CODEC_MUTE("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc662_mux_enum_info, - .get = alc662_mux_enum_get, - .put = alc662_mux_enum_put, - }, { } /* end */ }; @@ -11827,6 +11794,25 @@ static struct snd_kcontrol_new alc662_chmode_mixer[] = { { } /* end */ }; +/* capture mixer elements */ +static struct snd_kcontrol_new alc662_capture_mixer[] = { + HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* The multiple "Capture Source" controls confuse alsamixer + * So call somewhat different.. + */ + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 1, + .info = alc662_mux_enum_info, + .get = alc662_mux_enum_get, + .put = alc662_mux_enum_put, + }, + { } /* end */ +}; + static struct hda_verb alc662_init_verbs[] = { /* ADC: mute amp left and right */ {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, @@ -11949,26 +11935,6 @@ static struct hda_verb alc662_auto_init_verbs[] = { { } }; -/* capture mixer elements */ -static struct snd_kcontrol_new alc662_capture_mixer[] = { - HDA_CODEC_VOLUME("Capture Volume", 0x09, 0x0, HDA_INPUT), - HDA_CODEC_MUTE("Capture Switch", 0x09, 0x0, HDA_INPUT), - { - .iface = SNDRV_CTL_ELEM_IFACE_MIXER, - /* The multiple "Capture Source" controls confuse alsamixer - * So call somewhat different.. - * FIXME: the controls appear in the "playback" view! - */ - /* .name = "Capture Source", */ - .name = "Input Source", - .count = 1, - .info = alc882_mux_enum_info, - .get = alc882_mux_enum_get, - .put = alc882_mux_enum_put, - }, - { } /* end */ -}; - static void alc662_lenovo_101e_ispeaker_automute(struct hda_codec *codec) { unsigned int present; @@ -12109,8 +12075,7 @@ static struct alc_config_preset alc662_presets[] = { .input_mux = &alc662_capture_source, }, [ALC662_5ST_DIG] = { - .mixers = { alc662_base_mixer, alc662_chmode_mixer, - alc662_capture_mixer }, + .mixers = { alc662_base_mixer, alc662_chmode_mixer }, .init_verbs = { alc662_init_verbs }, .num_dacs = ARRAY_SIZE(alc662_dac_nids), .dac_nids = alc662_dac_nids, -- []'s Herton

2 3

[alsa-devel] [PATCH] Dmux mixer conflict
by Matthew Ranostay 19 Dec '07

19 Dec '07

DMUX mixers need to be created before input source mixers, otherwise some STAC9228 codecs fail to record correctly or at all. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r fbe3540c8df4 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Dec 18 15:27:34 2007 -0700 @@ -834,15 +834,16 @@ static int stac92xx_build_controls(struc if (err < 0) return err; - for (i = 0; i < spec->num_mixers; i++) { - err = snd_hda_add_new_ctls(codec, spec->mixers[i]); - if (err < 0) - return err; - } if (spec->num_dmuxes > 0) { stac_dmux_mixer.count = spec->num_dmuxes; err = snd_ctl_add(codec->bus->card, snd_ctl_new1(&stac_dmux_mixer, codec)); + if (err < 0) + return err; + } + + for (i = 0; i < spec->num_mixers; i++) { + err = snd_hda_add_new_ctls(codec, spec->mixers[i]); if (err < 0) return err; }

2 2

[alsa-devel] rate plugin issue
by Takashi Iwai 19 Dec '07

19 Dec '07

Hi, I committed your patch now on HG tree, as it's likely OK from the possible breakage on other apps, too. Anyway, I swear that I'll never read any mails during my next vacation. That was so exhausting to be dragged to a lengthy discussion for band-aiding a badly designed compoment... :) Takashi

3 11

[alsa-devel] speaker-test stalls
by Dave Dillow 19 Dec '07

19 Dec '07

While working with Fernando to get the sis7019 driver working in his buildroot setup, we kept running into a lockup of the PCM stream when using speaker-test. aplay was fine, as was other I finally tracked it down to a combination of uclibc not doing block reads, which led to XRUNs, and what I thought was issue in snd_pcm_lib_write1() when trying to fill up the tail of the PCM buffer with a large write, when there was an intervening small write. As it turns out, the lockup problem may simply be in speaker-write's sw_params settings -- it calls snd_pcm_sw_params_set_avail_min() to set the minium write to the period size, and snd_pcm_sw_params_set_start_threshold() to start once the buffer has been filled. This doesn't account for odd wave file lengths and XRUNs. The problem arises when the wave data is not a multiple of the period size, and you get an XRUN when there is less than buffer_size samples remaining in the current loop. So for example, period size is 8192, buffer is 32768, and the wave file has 71042 samples (Front_Left.wav for speaker-test). speaker-test will break that into 8 writes of 8192 samples, and one of 5506. If you get a XRUN in the last 32k of the file, the PCM will be stopped, and prepared, and we'll start loading data into again. Because we have a tail of 5506 samples, we get to the point where the available space is less than our minimum, and then snd_pcm_lib_write1() will wait forever for the available space to be >= runtime->avail_min, even though the channel is stopped. [snip from a trace, patch to pcm_lib.c attached] writei size 8192 avail 32768 min 8192 xfer 1 state 2 writei size 8192 avail 24576 min 8192 xfer 1 state 2 writei size 5506 avail 16384 min 8192 xfer 1 state 2 writei size 8192 avail 10878 min 8192 xfer 1 state 2 writei size 8192 avail 2686 min 8192 xfer 1 state 2 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 writei loop avail 2686 min 8192 Assuming that this is desired behavior from the kernel side, then I see three options to fix this in speaker-test: 1) snd_pcm_sw_params_set_avail_min() to 1 sample 2) snd_pcm_sw_params_set_start_threshold() to start when (nperiods - 1) are ready (or be even more aggressive) 3) call snd_pcm_drain() at the end of each loop to clear out the buffer. This won't help if the wave file has fewer samples than the buffer size. Options 1 or 2 seem to be best. Thoughts? Dave

2 5

[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h
by Andrew Paprocki 19 Dec '07

19 Dec '07

Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/hda_codec.c Tue Dec 18 21:48:58 2007 -0500 @@ -1435,7 +1435,8 @@ int snd_hda_create_spdif_out_ctls(struct return err; } codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/hda_codec.h Tue Dec 18 21:48:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#define AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,10 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +/* Note: This LR_SWAP pincap is defined in the Realtek ALC883 specification, + * but is marked reserved in the Intel HDA specification. + */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r fbe3540c8df4 -r 54a57ee44eb6 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Tue Dec 18 18:05:52 2007 +0100 +++ b/pci/hda/patch_realtek.c Tue Dec 18 21:48:58 2007 -0500 @@ -572,7 +572,7 @@ static int alc_spdif_ctrl_get(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -586,7 +586,7 @@ static int alc_spdif_ctrl_put(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

2 1

[alsa-devel] Firmware problem
by Adrian McMenamin 19 Dec '07

19 Dec '07

My user - yes there is one - of the aica sound driver on the Dreamcast is having a problem with formware loading. This is what he has sent me - anyone know what is wrong? When loading snd-aica it gives: [ 55.119676] ALSA sound/sh/aica.c:641: ALSA Driver for Yamaha AICA Super Inter then after like 15 or 20 seconds it will give: FATAL: Error inserting snd_aica (/lib/modules/2.6.24-rc4/kernel/sound/sh/snd-aica.ko): Unknown symbol in module, or unknown parameter (see dmesg) and dmesg gives this: [ 54.632531] fill_kobj_path: path = '/module/snd_page_alloc' [ 54.643434] kobject notes: registering. parent: snd_page_alloc, set: <NULL> [ 54.645602] kobject_uevent_env [ 54.646181] kobject filter function caused the event to drop! [ 54.669703] kobject soundcore: registering. parent: <NULL>, set: module [ 54.671737] kobject holders: registering. parent: soundcore, set: <NULL> [ 54.673616] kobject_uevent_env [ 54.675375] kobject filter function caused the event to drop! [ 54.677096] kobject_uevent_env [ 54.678780] fill_kobj_path: path = '/module/soundcore' [ 54.690116] kobject notes: registering. parent: soundcore, set: <NULL> [ 54.690904] kobject_uevent_env [ 54.693891] kobject filter function caused the event to drop! [ 54.695845] device class 'sound': registering [ 54.697748] kobject sound: registering. parent: <NULL>, set: class [ 54.699657] kobject_uevent_env [ 54.701350] fill_kobj_path: path = '/class/sound' [ 54.787597] kobject snd: registering. parent: <NULL>, set: module [ 54.789664] kobject holders: registering. parent: snd, set: <NULL> [ 54.791557] kobject_uevent_env [ 54.793340] kobject filter function caused the event to drop! [ 54.795228] kobject_uevent_env [ 54.797010] fill_kobj_path: path = '/module/snd' [ 54.808092] kobject notes: registering. parent: snd, set: <NULL> [ 54.808886] kobject_uevent_env [ 54.811852] kobject filter function caused the event to drop! [ 54.849584] kobject snd_timer: registering. parent: <NULL>, set: module [ 54.850378] kobject holders: registering. parent: snd_timer, set: <NULL> [ 54.853372] kobject_uevent_env [ 54.855118] kobject filter function caused the event to drop! [ 54.856902] kobject_uevent_env [ 54.858680] fill_kobj_path: path = '/module/snd_timer' [ 54.869736] kobject notes: registering. parent: snd_timer, set: <NULL> [ 54.871802] kobject_uevent_env [ 54.872452] kobject filter function caused the event to drop! [ 54.875569] DEV: registering device: ID = 'timer' [ 54.877400] kobject sound: registering. parent: virtual, set: <NULL> [ 54.881274] kobject_uevent_env [ 54.883351] unset subsystem caused the event to drop! [ 54.886599] kobject timer: registering. parent: sound, set: devices [ 54.888593] kobject_uevent_env [ 54.890420] fill_kobj_path: path = '/devices/virtual/sound/timer' [ 55.000017] kobject snd_pcm: registering. parent: <NULL>, set: module [ 55.002208] kobject holders: registering. parent: snd_pcm, set: <NULL> [ 55.004074] kobject_uevent_env [ 55.004636] kobject filter function caused the event to drop! [ 55.007502] kobject_uevent_env [ 55.009274] fill_kobj_path: path = '/module/snd_pcm' [ 55.020295] kobject notes: registering. parent: snd_pcm, set: <NULL> [ 55.021108] kobject_uevent_env [ 55.024045] kobject filter function caused the event to drop! [ 55.053022] kobject snd_aica: registering. parent: <NULL>, set: module [ 55.055131] kobject holders: registering. parent: snd_aica, set: <NULL> [ 55.057074] kobject_uevent_env [ 55.057643] kobject filter function caused the event to drop! [ 55.060587] kobject_uevent_env [ 55.061260] fill_kobj_path: path = '/module/snd_aica' [ 55.073491] kobject notes: registering. parent: snd_aica, set: <NULL> [ 55.074288] kobject_uevent_env [ 55.077295] kobject filter function caused the event to drop! [ 55.079239] bus platform: add driver AICA [ 55.079929] kobject AICA: registering. parent: <NULL>, set: drivers [ 55.082948] kobject_uevent_env [ 55.084712] fill_kobj_path: path = '/bus/platform/drivers/AICA' [ 55.089946] Registering platform device 'AICA'. Parent at platform [ 55.091828] DEV: registering device: ID = 'AICA' [ 55.093558] kobject AICA: registering. parent: platform, set: devices [ 55.093744] bus platform: add device AICA [ 55.093849] kobject_uevent_env [ 55.093918] fill_kobj_path: path = '/devices/platform/AICA' [ 55.094045] platform: Matched Device AICA with Driver AICA [ 55.094096] platform: Probing driver AICA with device AICA [ 55.094492] DEV: registering device: ID = 'card0' [ 55.094564] kobject sound: registering. parent: AICA, set: <NULL> [ 55.094646] kobject_uevent_env [ 55.094682] unset subsystem caused the event to drop! [ 55.094739] kobject card0: registering. parent: sound, set: devices [ 55.094888] kobject_uevent_env [ 55.094959] fill_kobj_path: path = '/devices/platform/AICA/sound/card0' [ 55.095133] DEV: registering device: ID = 'pcmC0D0p' [ 55.095195] kobject pcmC0D0p: registering. parent: card0, set: devices [ 55.095454] kobject_uevent_env [ 55.095528] fill_kobj_path: path = '/devices/platform/AICA/sound/card0/pcmC0D0p' [ 55.095745] DEV: registering device: ID = 'controlC0' [ 55.095811] kobject controlC0: registering. parent: card0, set: devices [ 55.095980] kobject_uevent_env [ 55.096051] fill_kobj_path: path = '/devices/platform/AICA/sound/card0/controlC0' [ 55.119676] ALSA sound/sh/aica.c:641: ALSA Driver for Yamaha AICA Super Intelligent Sound Processor [ 55.128339] bound device 'AICA' to driver 'AICA' [ 55.128394] platform: Bound Device AICA to Driver AICA [ 57.018452] DEV: registering device: ID = 'firmware-AICA' [ 57.019121] kobject firmware: registering. parent: AICA, set: <NULL> [ 57.022447] kobject_uevent_env [ 57.023025] unset subsystem caused the event to drop! [ 57.023099] kobject firmware-AICA: registering. parent: firmware, set: devices [ 57.027620] kobject_uevent_env [ 57.029196] kobject filter function caused the event to drop! [ 57.029842] kobject_uevent_env [ 57.032535] fill_kobj_path: path = '/devices/platform/AICA/firmware/firmware-AICA' [ 115.955604] DEV: Unregistering device. ID = 'firmware-AICA' [ 115.955788] kobject_uevent_env [ 115.955898] fill_kobj_path: path = '/devices/platform/AICA/firmware/firmware-AICA' [ 115.956090] kobject firmware-AICA: cleaning up [ 115.956144] kobject firmware: cleaning up [ 115.960813] kobject holders: unregistering [ 115.960863] kobject_uevent_env [ 115.960900] kobject filter function caused the event to drop! [ 115.960972] kobject holders: cleaning up [ 115.961018] kobject snd_aica: unregistering [ 115.961059] kobject_uevent_env [ 115.961128] fill_kobj_path: path = '/module/snd_aica'

2 1

[alsa-devel] How do I use SNDRV_PCM_INFO_JOINT_DUPLEX?
by Timur Tabi 19 Dec '07

19 Dec '07

I'm working on an ASoC driver. I have a limitation in my hardware that if I have an active substream, I cannot re-program the hardware registers that control the sample size. This means, for example, if I'm playing 16-bit audio, and I want to start capture, I cannot program the sample size into the hardware. My only choice is to have already programmed the capture sample size *before* I started playback. The easiest way to implement this is to make sure that the second stream has the same sample size as the first. That way, I won't need to reprogram the registers, because they'll already be programmed correctly. I discovered the SNDRV_PCM_INFO_JOINT_DUPLEX option, but I can't figure out how to use it. I know it needs to be set in my snd_pcm_hardware.info, but then what? How do I tell ALSA that the "joint" part is just that the sample sizes must be the same? -- Timur Tabi Linux kernel developer at Freescale

2 1

[alsa-devel] uniwill (0x1558, 0x9075) dup
by Andy Shevchenko 18 Dec '07

18 Dec '07

Hi! I investigated the duplicate problem and try to compare two cases. I don't deep to search of the specifications, but the current variant (with asus-laptop) should be more correct. In last HG snapshot second case already comented. I just noticed on my reseaches. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

1 0

[alsa-devel] [PATCH] hda: STAC927x DMIC Cleanup
by Matthew Ranostay 18 Dec '07

18 Dec '07

Cleaned up STAC927x and added several subsystem id's for more laptops. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 4a204306b295 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Fri Dec 14 14:42:41 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Fri Dec 14 12:37:16 2007 -0700 @@ -107,6 +107,7 @@ enum { STAC_D965_3ST, STAC_D965_5ST, STAC_DELL_3ST, + STAC_DELL_BIOS, STAC_927X_MODELS }; @@ -1408,22 +1409,24 @@ static unsigned int dell_3st_pin_configs static unsigned int dell_3st_pin_configs[14] = { 0x02211230, 0x02a11220, 0x01a19040, 0x01114210, 0x01111212, 0x01116211, 0x01813050, 0x01112214, - 0x403003fa, 0x40000100, 0x40000100, 0x404003fb, + 0x403003fa, 0x90a60040, 0x90a60040, 0x404003fb, 0x40c003fc, 0x40000100 }; static unsigned int *stac927x_brd_tbl[STAC_927X_MODELS] = { - [STAC_D965_REF] = ref927x_pin_configs, - [STAC_D965_3ST] = d965_3st_pin_configs, - [STAC_D965_5ST] = d965_5st_pin_configs, - [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_D965_REF] = ref927x_pin_configs, + [STAC_D965_3ST] = d965_3st_pin_configs, + [STAC_D965_5ST] = d965_5st_pin_configs, + [STAC_DELL_3ST] = dell_3st_pin_configs, + [STAC_DELL_BIOS] = NULL, }; static const char *stac927x_models[STAC_927X_MODELS] = { - [STAC_D965_REF] = "ref", - [STAC_D965_3ST] = "3stack", - [STAC_D965_5ST] = "5stack", - [STAC_DELL_3ST] = "dell-3stack", + [STAC_D965_REF] = "ref", + [STAC_D965_3ST] = "3stack", + [STAC_D965_5ST] = "5stack", + [STAC_DELL_3ST] = "dell-3stack", + [STAC_DELL_BIOS] = "dell-bios", }; static struct snd_pci_quirk stac927x_cfg_tbl[] = { @@ -1450,13 +1453,21 @@ static struct snd_pci_quirk stac927x_cfg SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2003, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2002, "Intel D965", STAC_D965_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2001, "Intel D965", STAC_D965_3ST), - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_D965_3ST), /* Dell 3 stack systems */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f3, "Dell Inspiron 1420", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f7, "Dell XPS M1730", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01dd, "Dell Dimension E520", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01ed, "Dell ", STAC_DELL_3ST), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f4, "Dell ", STAC_DELL_3ST), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0227, "Dell Vostro 1400 ", STAC_DELL_3ST), + /* Dell 3 stack systems with verb table in BIOS */ + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022f, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x022e, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0242, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0243, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x02ff, "Dell ", STAC_DELL_BIOS), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_DELL_BIOS), /* 965 based 5 stack systems */ - SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0209, "Dell XPS 1330", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2301, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2302, "Intel D965", STAC_D965_5ST), SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2303, "Intel D965", STAC_D965_5ST), @@ -1992,6 +2003,7 @@ static int stac92xx_add_dyn_out_pins(str for (i = 0; i < codec->num_nodes; i++) { wcaps = codec->wcaps[i]; wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; + if (wtype == AC_WID_AUD_OUT && !(wcaps & AC_WCAP_DIGITAL)) num_dacs++; } @@ -2079,7 +2091,6 @@ static int stac92xx_auto_fill_dac_nids(s wcaps = snd_hda_param_read(codec, conn[j], AC_PAR_AUDIO_WIDGET_CAP); wtype = (wcaps & AC_WCAP_TYPE) >> AC_WCAP_TYPE_SHIFT; - if (wtype != AC_WID_AUD_OUT || (wcaps & AC_WCAP_DIGITAL)) continue; @@ -3293,8 +3304,11 @@ static int patch_stac927x(struct hda_cod stac927x_models, stac927x_cfg_tbl); again: - if (spec->board_config < 0) { - snd_printdd(KERN_INFO "hda_codec: Unknown model for STAC927x, using BIOS defaults\n"); + if (spec->board_config < 0 || + stac927x_brd_tbl[spec->board_config] != NULL) { + if (spec->board_config < 0) + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + "STAC927x, using BIOS defaults\n"); err = stac92xx_save_bios_config_regs(codec); if (err < 0) { stac92xx_free(codec); @@ -3306,59 +3320,46 @@ static int patch_stac927x(struct hda_cod stac92xx_set_config_regs(codec); } + spec->adc_nids = stac927x_adc_nids; + spec->mux_nids = stac927x_mux_nids; + spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + spec->multiout.dac_nids = spec->dac_nids; + switch (spec->board_config) { case STAC_D965_3ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + case STAC_D965_5ST: + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; break; - case STAC_D965_5ST: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + case STAC_DELL_BIOS: + case STAC_DELL_3ST: + /* GPIO2 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000004; + spec->dmic_nids = stac927x_dmic_nids; + spec->num_dmics = STAC927X_NUM_DMICS; + spec->init = d965_core_init; spec->mixer = stac927x_mixer; + spec->dmux_nids = stac927x_dmux_nids; break; default: - spec->adc_nids = stac927x_adc_nids; - spec->mux_nids = stac927x_mux_nids; - spec->num_muxes = ARRAY_SIZE(stac927x_mux_nids); - spec->num_adcs = ARRAY_SIZE(stac927x_adc_nids); + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x00000001; + spec->num_dmics = 0; + spec->init = stac927x_core_init; spec->mixer = stac927x_mixer; } - switch (codec->subsystem_id) { - case 0x10280242: /* STAC 9228 */ - case 0x102801f3: - case 0x1028020A: - case 0x10280209: - spec->dmic_nids = stac927x_dmic_nids; - spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nids = stac927x_dmux_nids; - - /* Enable DMIC0 */ - stac92xx_set_config_reg(codec, 0x13, 0x90a60040); - - /* GPIO2 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000004; - break; - default: - spec->num_dmics = 0; - - /* GPIO0 High = Enable EAPD */ - spec->gpio_mask = spec->gpio_data = 0x00000001; - } - - spec->multiout.dac_nids = spec->dac_nids; spec->aloopback_mask = 0x40; spec->aloopback_shift = 0; + stac92xx_enable_gpio_mask(codec); - err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); if (!err) { if (spec->board_config < 0) {

3 2

[alsa-devel] snd-hda-intel: HG regression - NULL pointer dereference
by Ivan N. Zlatev 18 Dec '07

18 Dec '07

When loading snd-hda-intel from Alsa HG I get a NULL pointer dereference and the trace below. The card is with the following codec: Codec: SigmaTel STAC9228 Address: 0 Vendor Id: 0x83847616 Subsystem Id: 0x10280227 Revision Id: 0x100201 The trace: ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2792: autoconfig: line_outs=1 (0xf/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2796: speaker_outs=1 (0xd/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2800: hp_outs=1 (0xa/0x0/0x0/0x0/0x0) ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/hda_codec.c:2808: inputs: mic=0x13, fmic=0x0, line=0xe, fline=0x0, cd=0x0, aux=0x0 ALSA /home/ivanz/svn/alsa/alsa-driver/pci/hda/../../alsa-kernel/pci/hda/patch_sigmatel.c:2088: dac_nids=2 (0x5/0x4/0x0/0x0/0x0) BUG: unable to handle kernel NULL pointer dereference at virtual address 00000000 printing eip: f91cfcb0 *pde = 00000000 Oops: 0000 [#1] SMP last sysfs file: /devices/pci0000:00/0000:00:1c.1/0000:0c:00.0/cmd Modules linked in: snd_hda_intel snd_pcm snd_timer snd_page_alloc snd_hwdep snd soundcore nls_iso8859_1 nls_cp437 vfat fat usb_storage ide_core rndis_host cdc_ether usbnet mii aes_i586 ieee80211_crypt_ccmp af_packet xt_tcpudp xt_pkttype ipt_LOG xt_limit vboxdrv ipt_REJECT xt_state iptable_mangle iptable_nat nf_nat iptable_filter nf_conntrack_ipv4 nf_conntrack nfnetlink ip_tables ip6_tables x_tables cpufreq_conservative cpufreq_userspace cpufreq_powersave acpi_cpufreq speedstep_lib microcode apparmor loop dm_mod rfcomm l2cap nvidia(P) ipw3945 sdhci ohci1394 ieee1394 mmc_core ieee80211 hci_usb ac ieee80211_crypt battery i2c_i801 button tg3 rtc_cmos bluetooth rtc_core sr_mod rtc_lib cdrom intel_agp firmware_class i2c_core agpgart sg usbhid hid ff_memless sd_mod ehci_hcd uhci_hcd usbcore edd ext3 mbcache jbd fan ata_piix ahci libata scsi_mod thermal processor CPU: 1 EIP: 0060:[<f91cfcb0>] Tainted: P N VLI EFLAGS: 00010246 (2.6.22.13-0.3-default #1) EIP is at stac92xx_dmux_enum_put+0x28/0x3c [snd_hda_intel] eax: 00000000 ebx: 00000000 ecx: f54b7014 edx: d7e241ec esi: d7e24014 edi: f71e6014 ebp: 00000000 esp: d9ea1ea0 ds: 007b es: 007b fs: 00d8 gs: 0033 ss: 0068 Process alsactl (pid: 25097, ti=d9ea0000 task=c7457030 task.ti=d9ea0000) Stack: 00000000 d7e240a4 f54b7014 cbe43694 fffffff3 f911c2b3 cbe43694 c2175154 df8a4c14 df8a4d60 df8a4c14 00000000 f54b7014 df8a4dc4 f911cd85 cb940b7c c03c5aec f52693c0 f52693c0 bf8ef020 c2175154 c0172331 c016e87a dfbef2c0 Call Trace: [<f911c2b3>] snd_ctl_elem_write+0xa6/0xe6 [snd] [<f911cd85>] snd_ctl_ioctl+0x494/0x7af [snd] [<c0172331>] chrdev_open+0x0/0x133 [<c016e87a>] __dentry_open+0xe4/0x178 [<c016e988>] nameidata_to_filp+0x24/0x33 [<c016e9ce>] do_filp_open+0x37/0x3e [<f911c8f1>] snd_ctl_ioctl+0x0/0x7af [snd] [<c017a13d>] do_ioctl+0x21/0xa0 [<c017a3f3>] vfs_ioctl+0x237/0x249 [<c017a451>] sys_ioctl+0x4c/0x67 [<c0104e22>] sysenter_past_esp+0x6b/0xa9 [<c02c0000>] unix_bind+0x5e/0x28e ======================= Code: 31 c0 c3 57 89 d1 56 53 83 ec 08 8b 78 60 8b 5a 3c 2b 58 44 8b 77 4c 8d 84 9e 90 00 00 00 8b 96 8c 00 00 00 89 44 24 04 8b 46 6c <0f> b7 04 58 89 04 24 89 f8 e8 99 24 ff ff 5f 5a 5b 5e 5f c3 53 EIP: [<f91cfcb0>] stac92xx_dmux_enum_put+0x28/0x3c [snd_hda_intel] SS:ESP 0068:d9ea1ea0 -- Ivan N. Zlatev Web: http://www.i-nZ.net "It's all some kind of whacked out conspiracy."

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[alsa-devel] Status of Dolby Digital on-the-fly encoding? (via a52 plugin)
by Dr. Alexander K. Seewald 18 Dec '07

18 Dec '07

I am trying to output Dolby Digital data via coaxial cable to the Logitech Z-5450 5.1 speaker set. Presumably, a52 is the right plugin for this, taking six channels and combining them into the right format. However, I cannot seem to make the Z-5450 understand the format... it always displays "No digital data" and does not output anything. Now, the documentation is very sparse on this, but states that Dolby Digital is supported, as well as DTS and DTS with 96kHz sampling rate and 24bit samples. The Z-5450 can play digital PCM stereo data, but this only activates left and right speaker (it is possible to set an effect to distribute sound to the other speakers, but this is insufficient for my purpose). I can play digital DVD 5.1 sound by attaching the coaxial cable to a DVD player. As of now, speakertest -c 6 tries to play sound on all speakers, but only left and right respond. For all other speakers, "No digital data" is displayed on the Z-5450 console. This is consistent with the Z-5450 not able to play six-channel PCM data. However, using -Dplug:a52 and .asoundrc = pcm.a52 { type a52 } (with various settings) always shows "No digital data" and no sound output. I had a sample DTS 96/24 file a while ago which was able to activate all speakers, so the data connection is ok. But sadly I have misplaced it; in any case DTS 96/24 support is not what I'd expect in the near future. Has anyone successfully output Dolby Digital (A52, AC3, ..) data via coax to the Z-5450 or a similar model? I'm using the newest alsa drivers and plugins on an 2.6.11, and the stereo-PCM output has been working for years... but this one keeps bugging me. Best, Alex -- Dr. Alexander K. Seewald +43(664)1106886 Seewald Solutions ------------------------------------------------ Information wants to be free; Information also wants to be expensive (S.Brant) --------------- alex.seewald.at ----------------

2 7

[alsa-devel] [PATCH] cs4270: wrong sample rate when CONFIG_SND_SOC_CS4270_VD33_ERRATA is set
by Timur Tabi 18 Dec '07

18 Dec '07

When CONFIG_SND_SOC_CS4270_VD33_ERRATA is set, there was a mismatch between the mclk_ratios[] and cs4270_mode_ratios[] arrays. The two arrays have been merged and code has been shuffled. One side effect is that the cs4270_set_dai_sysclk() and cs4270_set_dai_fmt() functions are available only if I2C has been enabled. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/codecs/cs4270.c | 261 ++++++++++++++++++++++----------------------- 1 files changed, 128 insertions(+), 133 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index abac628..bebb4ba 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -48,12 +48,130 @@ struct cs4270_private { unsigned int mode; /* The mode (I2S or left-justified) */ }; -/* The number of MCLK/LRCK ratios supported by the CS4270 */ -#define NUM_MCLK_RATIOS 9 +/* + * The codec isn't really big-endian or little-endian, since the I2S + * interface requires data to be sent serially with the MSbit first. + * However, to support BE and LE I2S devices, we specify both here. That + * way, ALSA will always match the bit patterns. + */ +#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ + SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ + SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ + SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) + +#ifdef USE_I2C + +/* CS4270 registers addresses */ +#define CS4270_CHIPID 0x01 /* Chip ID */ +#define CS4270_PWRCTL 0x02 /* Power Control */ +#define CS4270_MODE 0x03 /* Mode Control */ +#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ +#define CS4270_TRANS 0x05 /* Transition Control */ +#define CS4270_MUTE 0x06 /* Mute Control */ +#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ +#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ + +#define CS4270_FIRSTREG 0x01 +#define CS4270_LASTREG 0x08 +#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) -/* The actual MCLK/LRCK ratios, in increasing numerical order */ -static unsigned int mclk_ratios[NUM_MCLK_RATIOS] = - {64, 96, 128, 192, 256, 384, 512, 768, 1024}; +/* Bit masks for the CS4270 registers */ +#define CS4270_CHIPID_ID 0xF0 +#define CS4270_CHIPID_REV 0x0F +#define CS4270_PWRCTL_FREEZE 0x80 +#define CS4270_PWRCTL_PDN_ADC 0x20 +#define CS4270_PWRCTL_PDN_DAC 0x02 +#define CS4270_PWRCTL_PDN 0x01 +#define CS4270_MODE_SPEED_MASK 0x30 +#define CS4270_MODE_1X 0x00 +#define CS4270_MODE_2X 0x10 +#define CS4270_MODE_4X 0x20 +#define CS4270_MODE_SLAVE 0x30 +#define CS4270_MODE_DIV_MASK 0x0E +#define CS4270_MODE_DIV1 0x00 +#define CS4270_MODE_DIV15 0x02 +#define CS4270_MODE_DIV2 0x04 +#define CS4270_MODE_DIV3 0x06 +#define CS4270_MODE_DIV4 0x08 +#define CS4270_MODE_POPGUARD 0x01 +#define CS4270_FORMAT_FREEZE_A 0x80 +#define CS4270_FORMAT_FREEZE_B 0x40 +#define CS4270_FORMAT_LOOPBACK 0x20 +#define CS4270_FORMAT_DAC_MASK 0x18 +#define CS4270_FORMAT_DAC_LJ 0x00 +#define CS4270_FORMAT_DAC_I2S 0x08 +#define CS4270_FORMAT_DAC_RJ16 0x18 +#define CS4270_FORMAT_DAC_RJ24 0x10 +#define CS4270_FORMAT_ADC_MASK 0x01 +#define CS4270_FORMAT_ADC_LJ 0x00 +#define CS4270_FORMAT_ADC_I2S 0x01 +#define CS4270_TRANS_ONE_VOL 0x80 +#define CS4270_TRANS_SOFT 0x40 +#define CS4270_TRANS_ZERO 0x20 +#define CS4270_TRANS_INV_ADC_A 0x08 +#define CS4270_TRANS_INV_ADC_B 0x10 +#define CS4270_TRANS_INV_DAC_A 0x02 +#define CS4270_TRANS_INV_DAC_B 0x04 +#define CS4270_TRANS_DEEMPH 0x01 +#define CS4270_MUTE_AUTO 0x20 +#define CS4270_MUTE_ADC_A 0x08 +#define CS4270_MUTE_ADC_B 0x10 +#define CS4270_MUTE_POLARITY 0x04 +#define CS4270_MUTE_DAC_A 0x01 +#define CS4270_MUTE_DAC_B 0x02 + +/* + * Clock Ratio Selection for Master Mode with I2C enabled + * + * The data for this chart is taken from Table 5 of the CS4270 reference + * manual. + * + * This table is used to determine how to program the Mode Control register. + * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling + * rates the CS4270 currently supports. + * + * Each element in this array corresponds to the ratios in mclk_ratios[]. + * These two arrays need to be in sync. + * + * 'speed_mode' is the corresponding bit pattern to be written to the + * MODE bits of the Mode Control Register + * + * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of + * the Mode Control Register. + * + * In situations where a single ratio is represented by multiple speed + * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick + * double-speed instead of quad-speed. However, the CS4270 errata states + * that Divide-By-1.5 can cause failures, so we avoid that mode where + * possible. + * + * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not + * work if VD = 3.3V. If this effects you, select the + * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will + * never select any sample rates that require divide-by-1.5. + */ +static struct { + unsigned int ratio; + u8 speed_mode; + u8 mclk; +} cs4270_mode_ratios[] = { + {64, CS4270_MODE_4X, CS4270_MODE_DIV1}, +#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA + {96, CS4270_MODE_4X, CS4270_MODE_DIV15}, +#endif + {128, CS4270_MODE_2X, CS4270_MODE_DIV1}, + {192, CS4270_MODE_4X, CS4270_MODE_DIV3}, + {256, CS4270_MODE_1X, CS4270_MODE_DIV1}, + {384, CS4270_MODE_2X, CS4270_MODE_DIV3}, + {512, CS4270_MODE_1X, CS4270_MODE_DIV2}, + {768, CS4270_MODE_1X, CS4270_MODE_DIV3}, + {1024, CS4270_MODE_1X, CS4270_MODE_DIV4} +}; + +/* The number of MCLK/LRCK ratios supported by the CS4270 */ +#define NUM_MCLK_RATIOS ARRAY_SIZE(cs4270_mode_ratios) /* * Determine the CS4270 samples rates. @@ -97,7 +215,7 @@ static int cs4270_set_dai_sysclk(struct snd_soc_codec_dai *codec_dai, cs4270->mclk = freq; for (i = 0; i < NUM_MCLK_RATIOS; i++) { - unsigned int rate = freq / mclk_ratios[i]; + unsigned int rate = freq / cs4270_mode_ratios[i].ratio; rates |= snd_pcm_rate_to_rate_bit(rate); if (rate < rate_min) rate_min = rate; @@ -155,80 +273,6 @@ static int cs4270_set_dai_fmt(struct snd_soc_codec_dai *codec_dai, } /* - * The codec isn't really big-endian or little-endian, since the I2S - * interface requires data to be sent serially with the MSbit first. - * However, to support BE and LE I2S devices, we specify both here. That - * way, ALSA will always match the bit patterns. - */ -#define CS4270_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | \ - SNDRV_PCM_FMTBIT_S18_3LE | SNDRV_PCM_FMTBIT_S18_3BE | \ - SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ - SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_3BE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) - -#ifdef USE_I2C - -/* CS4270 registers addresses */ -#define CS4270_CHIPID 0x01 /* Chip ID */ -#define CS4270_PWRCTL 0x02 /* Power Control */ -#define CS4270_MODE 0x03 /* Mode Control */ -#define CS4270_FORMAT 0x04 /* Serial Format, ADC/DAC Control */ -#define CS4270_TRANS 0x05 /* Transition Control */ -#define CS4270_MUTE 0x06 /* Mute Control */ -#define CS4270_VOLA 0x07 /* DAC Channel A Volume Control */ -#define CS4270_VOLB 0x08 /* DAC Channel B Volume Control */ - -#define CS4270_FIRSTREG 0x01 -#define CS4270_LASTREG 0x08 -#define CS4270_NUMREGS (CS4270_LASTREG - CS4270_FIRSTREG + 1) - -/* Bit masks for the CS4270 registers */ -#define CS4270_CHIPID_ID 0xF0 -#define CS4270_CHIPID_REV 0x0F -#define CS4270_PWRCTL_FREEZE 0x80 -#define CS4270_PWRCTL_PDN_ADC 0x20 -#define CS4270_PWRCTL_PDN_DAC 0x02 -#define CS4270_PWRCTL_PDN 0x01 -#define CS4270_MODE_SPEED_MASK 0x30 -#define CS4270_MODE_1X 0x00 -#define CS4270_MODE_2X 0x10 -#define CS4270_MODE_4X 0x20 -#define CS4270_MODE_SLAVE 0x30 -#define CS4270_MODE_DIV_MASK 0x0E -#define CS4270_MODE_DIV1 0x00 -#define CS4270_MODE_DIV15 0x02 -#define CS4270_MODE_DIV2 0x04 -#define CS4270_MODE_DIV3 0x06 -#define CS4270_MODE_DIV4 0x08 -#define CS4270_MODE_POPGUARD 0x01 -#define CS4270_FORMAT_FREEZE_A 0x80 -#define CS4270_FORMAT_FREEZE_B 0x40 -#define CS4270_FORMAT_LOOPBACK 0x20 -#define CS4270_FORMAT_DAC_MASK 0x18 -#define CS4270_FORMAT_DAC_LJ 0x00 -#define CS4270_FORMAT_DAC_I2S 0x08 -#define CS4270_FORMAT_DAC_RJ16 0x18 -#define CS4270_FORMAT_DAC_RJ24 0x10 -#define CS4270_FORMAT_ADC_MASK 0x01 -#define CS4270_FORMAT_ADC_LJ 0x00 -#define CS4270_FORMAT_ADC_I2S 0x01 -#define CS4270_TRANS_ONE_VOL 0x80 -#define CS4270_TRANS_SOFT 0x40 -#define CS4270_TRANS_ZERO 0x20 -#define CS4270_TRANS_INV_ADC_A 0x08 -#define CS4270_TRANS_INV_ADC_B 0x10 -#define CS4270_TRANS_INV_DAC_A 0x02 -#define CS4270_TRANS_INV_DAC_B 0x04 -#define CS4270_TRANS_DEEMPH 0x01 -#define CS4270_MUTE_AUTO 0x20 -#define CS4270_MUTE_ADC_A 0x08 -#define CS4270_MUTE_ADC_B 0x10 -#define CS4270_MUTE_POLARITY 0x04 -#define CS4270_MUTE_DAC_A 0x01 -#define CS4270_MUTE_DAC_B 0x02 - -/* * A list of addresses on which this CS4270 could use. I2C addresses are * 7 bits. For the CS4270, the upper four bits are always 1001, and the * lower three bits are determined via the AD2, AD1, and AD0 pins @@ -315,53 +359,6 @@ static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, } /* - * Clock Ratio Selection for Master Mode with I2C enabled - * - * The data for this chart is taken from Table 5 of the CS4270 reference - * manual. - * - * This table is used to determine how to program the Mode Control register. - * It is also used by cs4270_set_dai_sysclk() to tell ALSA which sampling - * rates the CS4270 currently supports. - * - * Each element in this array corresponds to the ratios in mclk_ratios[]. - * These two arrays need to be in sync. - * - * 'speed_mode' is the corresponding bit pattern to be written to the - * MODE bits of the Mode Control Register - * - * 'mclk' is the corresponding bit pattern to be wirten to the MCLK bits of - * the Mode Control Register. - * - * In situations where a single ratio is represented by multiple speed - * modes, we favor the slowest speed. E.g, for a ratio of 128, we pick - * double-speed instead of quad-speed. However, the CS4270 errata states - * that Divide-By-1.5 can cause failures, so we avoid that mode where - * possible. - * - * ERRATA: There is an errata for the CS4270 where divide-by-1.5 does not - * work if VD = 3.3V. If this effects you, select the - * CONFIG_SND_SOC_CS4270_VD33_ERRATA Kconfig option, and the driver will - * never select any sample rates that require divide-by-1.5. - */ -static struct { - u8 speed_mode; - u8 mclk; -} cs4270_mode_ratios[NUM_MCLK_RATIOS] = { - {CS4270_MODE_4X, CS4270_MODE_DIV1}, /* 64 */ -#ifndef CONFIG_SND_SOC_CS4270_VD33_ERRATA - {CS4270_MODE_4X, CS4270_MODE_DIV15}, /* 96 */ -#endif - {CS4270_MODE_2X, CS4270_MODE_DIV1}, /* 128 */ - {CS4270_MODE_4X, CS4270_MODE_DIV3}, /* 192 */ - {CS4270_MODE_1X, CS4270_MODE_DIV1}, /* 256 */ - {CS4270_MODE_2X, CS4270_MODE_DIV3}, /* 384 */ - {CS4270_MODE_1X, CS4270_MODE_DIV2}, /* 512 */ - {CS4270_MODE_1X, CS4270_MODE_DIV3}, /* 768 */ - {CS4270_MODE_1X, CS4270_MODE_DIV4} /* 1024 */ -}; - -/* * Program the CS4270 with the given hardware parameters. * * The .dai_ops functions are used to provide board-specific data, like @@ -388,7 +385,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream, ratio = cs4270->mclk / rate; /* MCLK/LRCK ratio */ for (i = 0; i < NUM_MCLK_RATIOS; i++) { - if (mclk_ratios[i] == ratio) + if (cs4270_mode_ratios[i].ratio == ratio) break; } @@ -669,7 +666,7 @@ error: return ret; } -#endif +#endif /* USE_I2C*/ struct snd_soc_codec_dai cs4270_dai = { .name = "CS4270", @@ -687,10 +684,6 @@ struct snd_soc_codec_dai cs4270_dai = { .rates = 0, .formats = CS4270_FORMATS, }, - .dai_ops = { - .set_sysclk = cs4270_set_dai_sysclk, - .set_fmt = cs4270_set_dai_fmt, - } }; EXPORT_SYMBOL_GPL(cs4270_dai); @@ -752,6 +745,8 @@ static int cs4270_probe(struct platform_device *pdev) if (codec->control_data) { /* Initialize codec ops */ cs4270_dai.ops.hw_params = cs4270_hw_params; + cs4270_dai.dai_ops.set_sysclk = cs4270_set_dai_sysclk; + cs4270_dai.dai_ops.set_fmt = cs4270_set_dai_fmt; #ifdef CONFIG_SND_SOC_CS4270_HWMUTE cs4270_dai.dai_ops.digital_mute = cs4270_mute; #endif -- 1.5.2.4

2 1

[alsa-devel] rfe: keep SND_PCI_QUIRK groups are sorted by vendor and product id pair
by Andy Shevchenko 17 Dec '07

17 Dec '07

Hi! Some to be discussed. I found the SND_PCI_QUIRK group (lines with this macro under one structure) is unsorted. Almost in all cases no sense to make so. I wrote python script to keep lists sorted. Please, comment this. P.S. Known bug is a deletion of the empty lines under group. #!/usr/bin/python # -*- coding: UTF-8 -*- # vim: ts=4 sw=4 et: """ The sript sorts SND_PCI_QUIRK groups by vendor and product id pair. """ __author__ = "Andy Shevchenko <andy(a)smile.org.ua>" __revision__ = "$Id$" import sys import os import re def read_file(fname): """ Read given file """ fdes = open(fname, 'rt') result = fdes.readlines() fdes.close() return result def save_file(fname, lines): """ Write lines to the given file """ fdes = open(fname, 'wt') fdes.writelines(lines) fdes.flush() fdes.close() def parse_quirks(quirks): """ Parse buffer as SND_PCI_QUIRK group """ rexp_begin = re.compile("^\s*[/\*]*\s*SND_PCI_QUIRK$(?P<vid>[^,]+),\s*(?P<pid>[^,]+)") rexp_end = re.compile("$,\s*(/\*.+\*/)?\s*$") xbuffer = {} # vendor, product: prefix lines, line, postfix lines flag = False vpid = 0 prefix = [] postfix = [] for line in quirks: if line.strip() == '': continue rmatch = rexp_begin.match(line) if rmatch: if flag == True: xbuffer[vpid][2].extend(postfix) postfix = [] flag = False vpid = (int(rmatch.group('vid'), 16) << 16) + int(rmatch.group('pid'), 16) if xbuffer.has_key(vpid): print "Warning: the line '%s' is duplicate!" % line.strip() xbuffer[vpid][2].append(line) else: xbuffer[vpid] = [prefix, line, []] prefix = [] if not rexp_end.search(line): flag = True else: if flag == False: prefix.append(line) else: postfix.append(line) return xbuffer def sort_quirks(quirks): """ Sort SND_PCI_QUIRK group by vendor and product id pair """ xbuffer = [] keys = quirks.keys() keys.sort() for key in keys: quirk = quirks[key] xbuffer.extend(quirk[0]) xbuffer.append(quirk[1]) xbuffer.extend(quirk[2]) return xbuffer def catch_quirk_lines(lines): """ Catch SND_PCI_QUIRK group from lines and sort it """ rexp_begin = re.compile("\s*struct\s+snd_pci_quirk\s+.+\s*=\s*{\s*$") flag = False xbuffer = [] quirks = [] for line in lines: if flag == False and rexp_begin.search(line): xbuffer.append(line) quirks.extend(xbuffer) flag = True xbuffer = [] continue if flag == True and line.strip() == '{}': quirks.extend(sort_quirks(parse_quirks(xbuffer))) flag = False xbuffer = [] xbuffer.append(line) quirks.extend(xbuffer) return quirks def main(): """ MAIN """ try: fname = sys.argv[1] except IndexError: print "Usage: %s <input file>" % os.path.basename(sys.argv[0]) sys.exit(1) lines = read_file(fname) quirks = catch_quirk_lines(lines) save_file('%s.sorted' % fname, quirks) if __name__ == '__main__': main() # End of file -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

2 2

[alsa-devel] Is the mixer interface in the kernel protect against concurrent access?
by Hans-Christian Egtvedt 17 Dec '07

17 Dec '07

Hi, I am solving a locking bug in the at73c213 SPI sound driver, and wondered if the mixer API is protected against concurrent access in ALSA? Or should I implement a mutex to protect my hardware against concurrent access? -- Mvh Hans-Christian Egtvedt

2 2

[alsa-devel] Notebook Toshiba satellite x200-21g with realtek snd-hda-intel audio device still not supported (codec ALC268)
by Daniele (Mastro) 17 Dec '07

17 Dec '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 hi all i'm now to this mailing list i've bought a laptop and i would like to help you supporting my sound card what works: 2 speaker, low volume (it should be a 5+1 audio card) what doesn't - -) low volume - -) inoperating headphone jack - -) not working mic - -) no subwoofer - -) may be some other stuff not tested ( SPDIF? hardware midi?) with speaker-test -t wav -c 6 only front left/right channel works my kernel is a debian kernel Linux Luffy 2.6.23-1-686-bigmem #1 SMP Wed Dec 5 02:42:05 UTC 2007 i686 GNU/Linux some info about the audio card: lspci -n 00:1b.0 0403: 8086:284b (rev 03) lspci -vvv 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) Subsystem: Toshiba America Info Systems Unknown device ff00 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at f6600000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=55mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [70] Express Unknown type IRQ 0 Device: Supported: MaxPayload 128 bytes, PhantFunc 0, ExtTag- Device: Latency L0s <64ns, L1 <1us Device: Errors: Correctable- Non-Fatal- Fatal- Unsupported- Device: RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ Device: MaxPayload 128 bytes, MaxReadReq 128 bytes Link: Supported Speed unknown, Width x0, ASPM unknown, Port 0 Link: Latency L0s <64ns, L1 <1us Link: ASPM Disabled CommClk- ExtSynch- Link: Speed unknown, Width x0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) cat /proc/asound/card0/codec#0 Codec: Realtek ALC268 Address: 0 Vendor Id: 0x10ec0268 Subsystem Id: 0x1179ff0a Revision Id: 0x100003 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x1d: Stereo Amp-Out Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [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: 0x40: OUT 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: [0x00 0x00] Pincap 0x081003c: IN OUT HP EAPD Detect Pin Default 0x0221101f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 1 0x10 Node 0x16 [Pin Complex] wcaps 0x40010c: Mono Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x0e Node 0x17 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 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 0x99a30931: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown Pin-ctls: 0x24: IN Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x02, stepsize=0x4f, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x083734: IN OUT Detect Pin Default 0x0281103e: [Jack] Line In at Ext Front 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 0x4016852d: [N/A] Speaker at Ext N/A Conn = Digital, Color = Purple Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400380: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, 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: [0x1f 0x1f] 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: [0x8a 0x8a] Connection: 7 0x18 0x19 0x1a* 0x1c 0x14 0x15 0x13 many of this think doesn't mean a thing to me but i will be glad to help you if you ask.. i don't mind compiling some other driver/kernel/tool for this just ask me one last note: searching the internet i've found this page: https://wiki.ubuntu.com/Gutsy_Intel_HD_Audio_Controller if you go down you find that the Toshiba Satellite x205-S7483 has the exact same problem here. I think they have the same card searching the Realtek web site i didn't find any 6 channel audio card with the ALC268 codec may be this mean the ALC268 is not the right codec for my audio card or this is a really new model/heavily Toshiba modified device please ask me if you need any more info or want me to do some test thanks you all for your work - -- Daniele -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHZTelgSF24OYDe4YRAm9cAJ0cWzwxUT4QZWAuTYi0ajyteb8dUgCguWlz 9rvQrAqdJBZRfHQ5yRU5PaU= =4wBW -----END PGP SIGNATURE----- - To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

2 2

[alsa-devel] [RFC PATCH] alsactl: skip inactive controls
by Dave Dillow 17 Dec '07

17 Dec '07

When using alsactl to save or restore the card settings, it currently skips over controls that don't have the appropriate read/write permissions. It should also skip over inactive controls, otherwise it will get an error when it tries to access that control, and will fail to save the card state (or fully restore it.) --- I noticed this while adding surround support to SiS7019. I think the problem affects other cards, as I am following the pattern used by trident, rme*, cmipci, emu10k1 and others. This is only for discussion, since I've not tested this (nor even compiled it) but I think it will fix the issue. diff -r d3ba62cd2776 alsactl/state.c --- a/alsactl/state.c Fri Dec 14 15:29:51 2007 +0100 +++ b/alsactl/state.c Sat Dec 15 14:06:35 2007 -0500 @@ -250,7 +250,8 @@ static int get_control(snd_ctl_t *handle return err; } - if (!snd_ctl_elem_info_is_readable(info)) + if (snd_ctl_elem_info_is_inactive(info) || + !snd_ctl_elem_info_is_readable(info)) return 0; snd_ctl_elem_value_set_id(ctl, id); err = snd_ctl_elem_read(handle, ctl); @@ -1306,7 +1307,8 @@ static int set_control(snd_ctl_t *handle } } - if (!snd_ctl_elem_info_is_writable(info)) + if (snd_ctl_elem_info_is_inactive(info) || + !snd_ctl_elem_info_is_writable(info)) return 0; snd_ctl_elem_value_set_numid(ctl, numid1);

2 1

[alsa-devel] [PATCH] hda-codec - Fix definition of AC_KNBCAP_DELTA to match spec
by Andrew Paprocki 17 Dec '07

17 Dec '07

AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the Intel HDA spec, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r 4a204306b295 -r 3681a8bf780e pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Fri Dec 14 14:42:41 2007 +0100 +++ b/pci/hda/hda_codec.h Fri Dec 14 12:27:35 2007 -0500 @@ -234,7 +234,7 @@ enum { /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<8) +#define AC_KNBCAP_DELTA (1<<7) /* * Control Parameters

2 1

[alsa-devel] [PATCH] alsa-lib: Broken SNDRV_PCM_IOCTL_TTSTAMP entry
by Matthew Ranostay 17 Dec '07

17 Dec '07

Fixed a IOCTL entry that was broken with a recent update to alsa-driver. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 0b2df0817fb6 include/sound/asound.h --- a/include/sound/asound.h Fri Dec 14 15:19:03 2007 +0100 +++ b/include/sound/asound.h Sun Dec 16 19:38:55 2007 -0700 @@ -463,7 +463,7 @@ enum { enum { SNDRV_PCM_IOCTL_PVERSION = _IOR('A', 0x00, int), SNDRV_PCM_IOCTL_INFO = _IOR('A', 0x01, struct sndrv_pcm_info), - SNDRV_PCM_IOCTL_TSTAMP = _IOW('A', 0x02, int), + SNDRV_PCM_IOCTL_TTSTAMP = _IOW('A', 0x03, int), SNDRV_PCM_IOCTL_HW_REFINE = _IOWR('A', 0x10, struct sndrv_pcm_hw_params), SNDRV_PCM_IOCTL_HW_PARAMS = _IOWR('A', 0x11, struct sndrv_pcm_hw_params), SNDRV_PCM_IOCTL_HW_FREE = _IO('A', 0x12), diff -r 0b2df0817fb6 src/pcm/pcm_hw.c --- a/src/pcm/pcm_hw.c Fri Dec 14 15:19:03 2007 +0100 +++ b/src/pcm/pcm_hw.c Sun Dec 16 19:38:55 2007 -0700 @@ -978,9 +978,9 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, if (SNDRV_PROTOCOL_VERSION(2, 0, 5) <= ver) { int on = 1; - if (ioctl(fd, SNDRV_PCM_IOCTL_TSTAMP, &on) < 0) { + if (ioctl(fd, SNDRV_PCM_IOCTL_TTSTAMP, &on) < 0) { ret = -errno; - SNDMSG("TSTAMP failed\n"); + SNDMSG("TTSTAMP failed\n"); return ret; } }

2 1

[alsa-devel] Altec Lansing FX5051 USB Speakers
by Jeremy Nickurak 16 Dec '07

16 Dec '07

Some number of months ago I posted a request here for any suggestions in getting a set of USB speakers from Altec Lansing, model# FX5051, working with ALSA. I wasn't able to follow up in detail at the time, so I'm hoping try again. They're a set of 5.1 speakers with a built in USB sound card. Sound does appear to work, but the mixer is locked at 100%. There is no analog manual mixer, and the digital control dial exhibits the same problem as alsamixer, so I have to find a way to adjust the usb device's mixer volume in order to make the speakers usable. The speakers work fine in Windows XP, without installing any drivers, so it would seem that this device is operating based on some standard. In windows, adjusting the mixer via the digital control dial, or via the windows mixer control both result in the LED display moving in sync to indicate the new volume setting. In alsa, adjusting the digital control dial doesn't appear to have any effect, other then waking the LED display from its low-light mode. Attempting to adjust volume from alsamixer likewise wakes the display from low-light mode, but also fails to significantly change the mixer setting. When I say significantly, I mean that the mixer flips between about 98% and 100%, when using the Up/Down keys (or PgUp or PgDown). End appears to drop the volume to 0%, but Home doesn't restore it. Pressing the device's 'Power' button also behaves strangely. I can turn the device off, but it is turned back on shortly afterward. No usb connect/disconnect events appear here, instead, the power button seems to function more like a mute. amixer -c 2 contents yields: atrus@kedri:~$ amixer -c 2 contents numid=3,iface=MIXER,name='Tone Control - Bass' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=1 numid=4,iface=MIXER,name='Tone Control - Treble' ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 : values=3 numid=1,iface=MIXER,name='PCM Playback Switch' ; type=BOOLEAN,access=rw------,values=1 : values=on numid=2,iface=MIXER,name='PCM Playback Volume' ; type=INTEGER,access=rw---R--,values=2,min=0,max=49,step=0 : values=48,48 | dBscale-min=-49.00dB,step=1.00dB,mute=0 I'm attaching the contents of 'lsusb -v'. Any suggestions would be appreciated, and any requests for further information about this device will be accomodated ASAP. Thanks in advance. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

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[alsa-devel] [PATCH v3 1/2] sis7019: support the SiS 7019 Audio Accelerator
by Dave Dillow 14 Dec '07

14 Dec '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth support at the moment, but audio playback and capture with two periods per buffer has seen extensive use. Arbitrary period and buffer sizes (with multiple periods per buffer) have seen light testing, but are believed to be production ready. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-kernel hg repo. Expect surround sound and S/PDIF support soon. I plan on doing that as a separate patch, as that gives a bisection point for bug-hunting. I'm not tied to that, though. Changes since last posting: * Fix the use of the capture timing channel (multiple periods per buffer) * Fix checkpatch warnings * Remove space from the name pci/sis7019.c | 1461 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/sis7019.h | 342 +++++++++++++ pci/Kconfig | 10 pci/Makefile | 2 4 files changed, 1815 insertions(+) diff -r eb09731e73d7 pci/Kconfig --- a/pci/Kconfig Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Kconfig Thu Dec 13 21:28:56 2007 -0500 @@ -802,6 +802,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND && X86 && !X86_64 + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff -r eb09731e73d7 pci/Makefile --- a/pci/Makefile Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Makefile Thu Dec 13 21:28:56 2007 -0500 @@ -23,6 +23,7 @@ snd-maestro3-objs := maestro3.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_MAESTRO3) += snd-maestr obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff -r eb09731e73d7 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Thu Dec 13 21:28:56 2007 -0500 @@ -0,0 +1,1461 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = 1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a + * documented range of 8-0xfff8 samples. Given that they are 0-based, + * that places our period/buffer range at 9-0xfff9 samples. That makes the + * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and + * max samples / min samples gives us the max periods in a buffer. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the documented hardware minimum offset */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *voice, + struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *timing = voice->timing; + void __iomem *play_base = timing->ctrl_base; + void __iomem *wave_base = timing->wave_base; + u16 buffer_size, period_size; + u32 format, control, sso_eso, delta; + u32 vperiod, sso, reg; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + /* The interrupt handler implements the timing syncronization, so + * setup its state. + */ + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = voice->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; + + /* Using unsigned samples with the all-zero silence buffer + * forces the output to the lower rail, killing playback. + * So ignore unsigned vs signed -- it doesn't change the timing. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + control = timing->buffer_size - 1; + control |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + /* We've done the math, now configure the channel. + */ + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(control, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *rec_base = voice->ctrl_base; + u32 format, dma_addr, control; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_CAPTURE_DMA_LOOP; + + /* If we've got more than two periods per buffer, then we have + * use a timing voice to clock out the periods. Otherwise, we can + * use the capture channel's interrupts. + */ + if (voice->timing) { + sis_prepare_timing_voice(voice, substream); + } else { + control |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + control |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(control, rec_base + SIS_CAPTURE_DMA_CONTROL); + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + /* We have 64 voices, and the driver currently records from + * only one channel, though that could change in the future. + */ + rc = snd_pcm_new(sis->card, "SiS7019", 0, 64, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * + * FIXME: Maximum attenuation is set for the music amp, which will + * need to change if we start using the synth engine. + */ + outl(0xffff0000, io + SIS_WEVCR); + + /* Ensure that the wave engine is in normal operating mode. + */ + outl(0, io + SIS_WECCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +#ifdef CONFIG_PM +static int sis_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(sis->pcm); + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[2]); + + /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. + */ + if (sis->irq >= 0) { + synchronize_irq(sis->irq); + free_irq(sis->irq, sis); + sis->irq = -1; + } + + /* Save the internal state away + */ + for (i = 0; i < 4; i++) { + memcpy_fromio(sis->suspend_state[i], ioaddr, 4096); + ioaddr += 4096; + } + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int sis_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "sis7019: unable to re-enable device\n"); + goto error; + } + + if (sis_chip_init(sis)) { + printk(KERN_ERR "sis7019: unable to re-init controller\n"); + goto error; + } + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + goto error; + } + + /* Restore saved state, then clear out the page we use for the + * silence buffer. + */ + for (i = 0; i < 4; i++) { + memcpy_toio(ioaddr, sis->suspend_state[i], 4096); + ioaddr += 4096; + } + + memset(sis->suspend_state[0], 0, 4096); + + sis->irq = pci->irq; + pci_set_master(pci); + + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[2]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + +error: + snd_card_disconnect(card); + return -EIO; +} +#endif /* CONFIG_PM */ + +static int sis_alloc_suspend(struct sis7019 *sis) +{ + int i; + + /* We need 16K to store the internal wave engine state during a + * suspend, but we don't need it to be contiguous, so play nice + * with the memory system. We'll also use this area for a silence + * buffer. + */ + for (i = 0; i < SIS_SUSPEND_PAGES; i++) { + sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL); + if (!sis->suspend_state[i]) + return -ENOMEM; + } + memset(sis->suspend_state[0], 0, 4096); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "30-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + mutex_init(&sis->ac97_mutex); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_alloc_suspend(sis); + if (rc < 0) { + printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + int rc; + + rc = -ENOENT; + if (!enable) + goto error_out; + + rc = -ENOMEM; + card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + strcpy(card->driver, "SiS7019"); + strcpy(card->shortname, "SiS7019"); + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + +#ifdef CONFIG_PM + .suspend = sis_suspend, + .resume = sis_resume, +#endif +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); diff -r eb09731e73d7 pci/sis7019.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.h Thu Dec 13 21:28:56 2007 -0500 @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000 +#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */

2 2

Re: [alsa-devel] [Pkg-alsa-devel] Bug#438118: alsa-utils: aplay non-blocking mode isn't working
by Elimar Riesebieter 14 Dec '07

14 Dec '07

forwarded 438118 alsa-devel(a)alsa-project.org stop Hi all, can one please have a lokk at this? aplay -N isn't working for the OP. On Thu, 16 Aug 2007 the mental interface of Anders Boström told: > >>>>> "ER" == Elimar Riesebieter <riesebie(a)lxtec.de> writes: > > >> strace aplay -N : > >> > >> ... > >> open("/usr/lib/alsa-lib/libasound_module_rate_speexrate.so", O_RDONLY) = -1 ENOENT (No such file or directory) > > ER> This file is provided by > ER> libasound2-plugins: /usr/lib/alsa-lib/libasound_module_rate_speexrate.so > > ER> Please install libasound2-plugins and try again. > > Thanks for the suggestion. > > Unfortunately, it didn't solve my problem, no sound when non-blocking > mode is used. New strace: > > strace aplay -N : > ... > open("/usr/lib/alsa-lib/libasound_module_rate_speexrate.so", O_RDONLY) = 5 > read(5, "\177ELF\2\1\1\0\0\0\0\0\0\0\0\0\3\0>\0\1\0\0\0\20\22\0"..., 832) = 832 > fstat(5, {st_mode=S_IFREG|0644, st_size=20848, ...}) = 0 > mmap(NULL, 2116136, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 5, 0) = 0x2b9f792fe000 > mprotect(0x2b9f79302000, 2097152, PROT_NONE) = 0 > mmap(0x2b9f79502000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 5, 0x4000) = 0x2b9f79502000 > close(5) = 0 > ioctl(4, 0xc2604110, 0x7fff3269dec0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269dec0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269de00) = 0 > ioctl(4, 0xc2604110, 0x7fff3269de00) = 0 > ioctl(4, 0xc2604110, 0x7fff3269ddd0) = 0 > ioctl(4, 0xc2604110, 0x7fff3269ddc0) = 0 > ioctl(4, 0xc2604111, 0x7fff3269ddc0) = 0 > ioctl(4, 0xc0884113, 0x7fff3269dd10) = 0 > ioctl(4, 0x80184132, 0x7fff3269dc70) = 0 > mmap(NULL, 49152, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0x2b9f79503000 > ioctl(4, 0xc0884113, 0x624e88) = 0 > ioctl(4, 0x4140, 0) = 0 > ioctl(4, 0xc0884113, 0x624e88) = 0 > read(3, "\377eO\333\337\370\344}\335_\377\330^l\356\361`j\336_l"..., 1001) = 1001 > read(3, "n{u^\342zl~e~_[\365_y\362_\363\350\337n[\347nmi`\367y\344"..., 604) = 604 > ioctl(4, 0x4144, 0x2b9f785d8000) = -1 EAGAIN (Resource temporarily unavailable) > close(3) = 0 > ioctl(4, 0x4143, 0x3) = 0 > munmap(0x2b9f79503000, 49152) = 0 > ioctl(4, 0x4112, 0) = 0 > close(4) = 0 > munmap(0x2b9f785d7000, 4096) = 0 > munmap(0x2b9f785d8000, 4096) = 0 > munmap(0x2b9f792fe000, 2116136) = 0 > exit_group(0) = ? I have no idea, so forwarded to alsa-dev then. THX in advance Elimar -- Experience is something you don't get until just after you need it!

5 8

[alsa-devel] [PATCH] hda: Added more 92HD71 codecs
by Matthew Ranostay 14 Dec '07

14 Dec '07

Added more codecs to the 92HD71 family, as well as support for several that don't have an analog mixer. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 51415b2a4cbb pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Dec 13 17:51:00 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Dec 13 14:10:59 2007 -0500 @@ -532,9 +532,24 @@ static struct hda_verb stac92hd71bxx_cor { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + /* unmute mono out node */ + { 0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, +}; + +static struct hda_verb stac92hd71bxx_analog_core_init[] = { + /* set master volume and direct control */ + { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* connect headphone jack to dac1 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, /* connect ports 0d and 0f to audio mixer */ { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ @@ -714,9 +729,31 @@ static struct snd_kcontrol_new stac92hd7 { } /* end */ }; +static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(1), + STAC_INPUT_SOURCE(2), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x19, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x0, 0x1a, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), + + HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), /* hardware gain controls */ HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x18, 0x0, HDA_OUTPUT), @@ -729,9 +766,6 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), - HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), { } /* end */ }; @@ -3088,11 +3122,24 @@ again: stac92xx_set_config_regs(codec); } + switch (codec->vendor_id) { + case 0x111d76b6: /* 4 Port without Analog Mixer */ + case 0x111d76b7: + case 0x111d76b4: /* 6 Port without Analog Mixer */ + case 0x111d76b5: + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71bxx_core_init; + break; + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + spec->aloopback_mask = 0x20; + spec->aloopback_shift = 0; + spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ stac92xx_enable_gpio_mask(codec); - - spec->init = stac92hd71bxx_core_init; - spec->mixer = stac92hd71bxx_mixer; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; @@ -3702,8 +3749,17 @@ struct hda_codec_preset snd_hda_preset_s { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, + { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b3, .name = "92HD71B7X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b4, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b5, .name = "92HD71B6X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b6, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, + { .id = 0x111d76b7, .name = "92HD71B5X", .patch = patch_stac92hd71bxx }, {} /* terminator */ };

2 1

[alsa-devel] [BUG] Kernel Bugzilla: ALSA bugs
by Natalie Protasevich 14 Dec '07

14 Dec '07

Hello, There are couple of open ALSA bus on bugzilla.kernel.org, seems like they weren't noticed yet. Can someone take look at those please. Thanks! hp_only quirk needed for lifebook p7010 http://bugzilla.kernel.org/show_bug.cgi?id=9100 Kernel: 2.6.22.X on lifebook p7010 I need to add options snd-intel8x0 ac97_quirk=hp_only to have the master volume work properly snd-au8830 has strange I/O problems http://bugzilla.kernel.org/show_bug.cgi?id=9367 Kernel:2.6.22 jack only works in 'playback' mode, none 'duplex' or 'capture' works. snd_hda_intel doesn't mute front speakers when headset is plugged in http://bugzilla.kernel.org/show_bug.cgi?id=9338 Kernel: 2.6.23+ Vaio VGN-FZ19VN

1 0

[alsa-devel] [PATCH v3 2/2] sis7019: add support to driver package
by Dave Dillow 14 Dec '07

14 Dec '07

The sis7019 driver uses __ffs(), which first became available in kernel 2.5.2.6. This adds that compatibility function for x86 to adriver.h, and adds a more useful error message for other platforms that try to use it. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-driver hg repo. It builds against 2.6.23, but I've not tested against earlier versions. pci/sis7019.c | 2 ++ include/adriver.h | 18 ++++++++++++++++++ 2 files changed, 20 insertions(+) diff -r 9eda190b55c3 include/adriver.h --- a/include/adriver.h Tue Dec 04 12:46:46 2007 +0100 +++ b/include/adriver.h Thu Dec 13 22:50:16 2007 -0500 @@ -1445,4 +1445,22 @@ static inline unsigned char snd_pci_revi #endif #endif +#if LINUX_VERSION_CODE < KERNEL_VERSION(2, 5, 3) +#if defined(__i386__) +static inline unsigned long __ffs(unsigned long word) +{ + __asm__("bsfl %1,%0" + :"=r" (word) + :"rm" (word)); + return word; +} +#else +static inline unsigned long __ffs(unsigned long word) +{ + __asm__("need_asm_for_your_arch_in_adriver.h"); + return word; +} +#endif +#endif + #endif /* __SOUND_LOCAL_DRIVER_H */ diff -r 9eda190b55c3 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Thu Dec 13 22:50:16 2007 -0500 @@ -0,0 +1,2 @@ +#include "../alsa-kernel/pci/sis7019.c" +EXPORT_NO_SYMBOLS;

1 0

[alsa-devel] Alsa library for the arm-linux architecture
by Felipe Uderman 13 Dec '07

13 Dec '07

Hello, I need to cross compile a project to the arm-linux platform that depends on the Alsa Library. My problem is that my cross development environment does not include the Alsa Library. I have downloaded the Alsa sources, and I checked that there are some instructions on how to port the library to other architectures. I would like to know if there are ports of the Alsa Library for the arm-linux architecture availible for download, or if I really have to cross compile it. Thank you for your help! Felipe

3 3

[alsa-devel] Dell-830 Intel Core2 Duo
by Alex Garin 13 Dec '07

13 Dec '07

Hi, I have this error when I installed Ubuntu 7.10 on this computer Dell-830 Intel Core2 Duo : No volume control GStreamer plugins and/or devices found. Find some things on the web, as I installed and compiled alsa-<driver, lib, utils> but still do not works. Any help please ? Thank you ! -- Alex Garin - CGI / GMA Gestion Intégrée des Technologies / Integrated Technology Management 1350 Boulevard Rene-Levesque Ouest, Montréal, Québec, H3G 1T4 (514) 415-3000 ext. 5194, (514) 415-3962 fax

1 0

[alsa-devel] [PATCH] hda: 92HD73 DMIC Amps
by Matthew Ranostay 13 Dec '07

13 Dec '07

Changed hardware gain mixers for the digital mic's from HDA_OUTPUT to HDA_INPUT. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r 51415b2a4cbb pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Thu Dec 13 17:51:00 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Thu Dec 13 13:25:46 2007 -0500 @@ -626,8 +626,8 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -657,8 +657,8 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), @@ -688,8 +688,8 @@ static struct snd_kcontrol_new stac92hd7 STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), /* hardware gain controls */ - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), - HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_INPUT), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT),

1 0

[alsa-devel] [PATCH 1/2] sis7019: support the SiS 7019 Audio Accelerator
by Dave Dillow 13 Dec '07

13 Dec '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth support at the moment, but audio playback and capture with two periods per buffer has seen extensive use. Arbitrary period and buffer sizes (with multiple periods per buffer) have seen light testing, but are believed to be production ready. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-kernel hg repo. Changes since last posting: * Cleanups as suggested * Fix a few mistaken hardware limits (off-by-one) pci/sis7019.c | 1462 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/sis7019.h | 342 +++++++++++++ pci/Kconfig | 10 pci/Makefile | 2 4 files changed, 1816 insertions(+) diff -r eb09731e73d7 pci/Kconfig --- a/pci/Kconfig Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Kconfig Sat Dec 08 18:46:18 2007 -0500 @@ -802,6 +802,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND && X86 && !X86_64 + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff -r eb09731e73d7 pci/Makefile --- a/pci/Makefile Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/Makefile Sat Dec 08 18:46:18 2007 -0500 @@ -23,6 +23,7 @@ snd-maestro3-objs := maestro3.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_MAESTRO3) += snd-maestr obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff -r eb09731e73d7 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Sat Dec 08 18:46:18 2007 -0500 @@ -0,0 +1,1462 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS 7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index = SNDRV_DEFAULT_IDX1; /* Index 0-MAX */ +static char *id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int enable = 1; + +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param(id, charp, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param(enable, bool, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW offset parameters (Loop End, Stop Sample, End Sample) have a + * documented range of 8-0xfff8 samples. Given that they are 0-based, + * that places our period/buffer range at 9-0xfff9 samples. That makes the + * max buffer size 0xfff9 samples * 2 channels * 2 bytes per sample, and + * max samples / min samples gives us the max periods in a buffer. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff9 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff9 * 4), + .periods_min = 1, + .periods_max = (0xfff9 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the documented hardware minimum offset */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Helper function: must hold sis->voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags = ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff9); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff9); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *timing, struct voice *cap, + struct snd_pcm_runtime *runtime) +{ + u16 buffer_size, period_size; + u32 vperiod, sso; + + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = cap->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + struct voice *timing = voice->timing; + void __iomem *rec_base = voice->ctrl_base; + void __iomem *play_base = NULL; + void __iomem *wave_base = NULL; + u32 format, dma_addr, rec_ctrl; + u16 leo; + + /* The following variables are only used if there is a timing channel. + */ + u32 uninitialized_var(timing_ctrl); + u32 uninitialized_var(sso_eso); + u32 uninitialized_var(delta); + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + rec_ctrl = leo | SIS_CAPTURE_DMA_LOOP; + if (timing) { + /* We're using a timing voice, so the capture voice will + * not need to generate interrupts. + */ + sis_prepare_timing_voice(timing, voice, runtime); + + timing_ctrl = timing->buffer_size - 1; + timing_ctrl |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1;; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + play_base = timing->ctrl_base; + wave_base = timing->wave_base; + } else { + /* We've just got one or two periods, so we can use the + * capture voice's interrupt generation. + */ + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + /* We're ready to program the channel(s) + */ + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(rec_ctrl, rec_base + SIS_CAPTURE_DMA_CONTROL); + + if (play_base) { + u32 reg; + + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(timing_ctrl, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, + wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + } + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + /* We have 64 voices, and the driver currently records from + * only one channel, though that could change in the future. + */ + rc = snd_pcm_new(sis->card, "SiS 7019", 0, 64, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS 7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * + * FIXME: Maximum attenuation is set for the music amp, which will + * need to change if we start using the synth engine. + */ + outl(0xffff0000, io + SIS_WEVCR); + + /* Ensure that the wave engine is in normal operating mode. + */ + outl(0, io + SIS_WECCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +#ifdef CONFIG_PM +static int sis_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(sis->pcm); + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[2]); + + /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. + */ + if (sis->irq >= 0) { + synchronize_irq(sis->irq); + free_irq(sis->irq, sis); + sis->irq = -1; + } + + /* Save the internal state away + */ + for (i = 0; i < 4; i++ ) { + memcpy_fromio(sis->suspend_state[i], ioaddr, 4096); + ioaddr += 4096; + } + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int sis_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "sis7019: unable to re-enable device\n"); + goto error; + } + + if (sis_chip_init(sis)) { + printk(KERN_ERR "sis7019: unable to re-init controller\n"); + goto error; + } + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + goto error; + } + + /* Restore saved state, then clear out the page we use for the + * silence buffer. + */ + for (i = 0; i < 4; i++ ) { + memcpy_toio(ioaddr, sis->suspend_state[i], 4096); + ioaddr += 4096; + } + + memset(sis->suspend_state[0], 0, 4096); + + sis->irq = pci->irq; + pci_set_master(pci); + + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[2]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + +error: + snd_card_disconnect(card); + return -EIO; +} +#endif /* CONFIG_PM */ + +static int sis_alloc_suspend(struct sis7019 *sis) +{ + int i; + + /* We need 16K to store the internal wave engine state during a + * suspend, but we don't need it to be contiguous, so play nice + * with the memory system. We'll also use this area for a silence + * buffer. + */ + for (i = 0; i < SIS_SUSPEND_PAGES; i++) { + sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL); + if (!sis->suspend_state[i]) + return -ENOMEM; + } + memset(sis->suspend_state[0], 0, 4096); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "30-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + mutex_init(&sis->ac97_mutex); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS 7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_alloc_suspend(sis); + if (rc < 0) { + printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + int rc; + + rc = -ENOENT; + if (!enable) + goto error_out; + + rc = -ENOMEM; + card = snd_card_new(index, id, THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + strcpy(card->driver, "SiS 7019"); + strcpy(card->shortname, "SiS 7019"); + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + +#ifdef CONFIG_PM + .suspend = sis_suspend, + .resume = sis_resume, +#endif +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); diff -r eb09731e73d7 pci/sis7019.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.h Sat Dec 08 18:46:18 2007 -0500 @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000 +#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */

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[alsa-devel] [PATCH 2/2] sis7019: add support for pre-2.6 kernels
by Dave Dillow 13 Dec '07

13 Dec '07

The sis7019 driver uses __ffs(), which is available in 2.6 kernels, but not in most earlier ones. This patches the driver to use a local version assembly version, as the hardware is SiS 55x only (i486/Pentium-ish). Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-driver hg repo. It has been compiled and tested under 2.6.23. pci/sis7019.patch | 52 ++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/Makefile | 3 ++- 2 files changed, 54 insertions(+), 1 deletion(-) diff -r 9eda190b55c3 pci/Makefile --- a/pci/Makefile Tue Dec 04 12:46:46 2007 +0100 +++ b/pci/Makefile Sat Dec 08 18:45:46 2007 -0500 @@ -10,7 +10,7 @@ include $(SND_TOPDIR)/Makefile.conf # clean-files := ad1889.c atiixp.c atiixp_modem.c bt87x.c cmipci.c ens1370.c \ - fm801.c intel8x0.c maestro3.c via82xx.c via82xx_modem.c + fm801.c intel8x0.c maestro3.c sis7019.c via82xx.c via82xx_modem.c obj-$(CONFIG_SND) += pdplus/ asihpi/ oxygen/ @@ -31,5 +31,6 @@ fm801.c: fm801.patch $(SND_TOPDIR)/alsa- fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c +sis7019.c: sis7019.patch $(SND_TOPDIR)/alsa-kernel/pci/sis7019.c via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c via82xx_modem.c: via82xx_modem.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx_modem.c diff -r 9eda190b55c3 pci/sis7019.patch --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.patch Sat Dec 08 18:45:46 2007 -0500 @@ -0,0 +1,52 @@ +--- sis7019.c.orig 2007-12-08 18:32:09.000000000 -0500 ++++ sis7019.c 2007-12-08 18:31:35.000000000 -0500 +@@ -31,7 +31,17 @@ + #include <sound/core.h> + #include <sound/ac97_codec.h> + #include <sound/initval.h> +-#include "sis7019.h" ++#include "../alsa-kernel/pci/sis7019.h" ++ ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2,6,0) ++static inline unsigned long sis_ffs(unsigned long word) ++{ ++ __asm__("bsfl %1,%0" ++ :"=r" (word) ++ :"rm" (word)); ++ return word; ++} ++#endif + + MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); + MODULE_DESCRIPTION("SiS 7019"); +@@ -290,7 +300,11 @@ + int bit; + + while (status) { ++#if LINUX_VERSION_CODE < KERNEL_VERSION(2,6,0) ++ bit = sis_ffs(status); ++#else + bit = __ffs(status); ++#endif + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); +@@ -791,9 +805,9 @@ + + /* The following variables are only used if there is a timing channel. + */ +- u32 uninitialized_var(timing_ctrl); +- u32 uninitialized_var(sso_eso); +- u32 uninitialized_var(delta); ++ u32 timing_ctrl = 0; ++ u32 sso_eso = 0; ++ u32 delta = 0; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. +@@ -1460,3 +1474,5 @@ + + module_init(sis7019_init); + module_exit(sis7019_exit); ++ ++EXPORT_NO_SYMBOLS

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[alsa-devel] ALSA external control plugin question
by Ben C 13 Dec '07

13 Dec '07

Hi, I have a newbie question. I'm trying to understand how the external control plugin fits into the picture. I have an external PCM plugin that does some audio processing before sending it to the slave audio device. And I want to control the behavior of the processing with this control plugin. So before any playback is happening, any setting changes via the control plugin need to be stored somewhere, so when playback happens, it could be used. Also during playback, the control plugin needs to find and talk to the PCM plugin so settings could be changed during playback. Is there some sort of internal mechanism that allows this type of setting storage/retrieval and communication between control & PCM plugins to happen, or do I have to roll my own outside of ALSA? I've tried to find examples, I've found the pulse control and maemo in the alsa-plugin tarball. But they both seem to use some sort of external server/device. What I'm looking for is ways to control an external PCM plugin that simply runs some audio processing algorithm, there's no additional server or device associated with it (well, just the slave, which would be specified in the config). Currently I have the PCM plugin working, but it can only read the settings from the alsa config file. Once playback starts it'll stick with the original settings. I also have the basics of the external control plugin working, but the controls currently don't do anything, since I don't know how to propagate the changes to the PCM plugin. I've been searching on the web but haven't found much. I'm willing to buy books too, but I can't seem to find books on ALSA. I would appreciate any explanation, pointers to documents, or even just pointing me to existing source code that has done this. Thank you for your time and your help. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[alsa-devel] [PATCH] hda-intel: Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI.
by Johannes Stezenbach 13 Dec '07

13 Dec '07

Hi, the internal CD connector on my Asus M2N-SLI Deluxe board doesn't work without the patch below, /proc/asound/card0/codec#0 shows that the pin is disabled (Pin-ctls: 0x00). I used kernel linux-2.6.24-rc4-190-g94545ba from Linus' git tree. During my experiments I found that loading the snd-hda-intel driver with model=auto also enables the CD connector. It seems that for my board (BIOS version 1102) the BIOS configuration is correct and the "Asus M2N-SLI" quirk shouldn't be necessary? I get the following debug output: ALSA sound/pci/hda/hda_intel.c:727: codec_mask = 0x1 ALSA sound/pci/hda/hda_codec.c:2140: hda_codec: model 'auto' is selected ALSA sound/pci/hda/hda_codec.c:2760: autoconfig: line_outs=4 (0x12/0x16/0x24/0x25/0x0) ALSA sound/pci/hda/hda_codec.c:2764: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:2768: hp_outs=1 (0x11/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:2776: inputs: mic=0x17, fmic=0x14, line=0x15, fline=0x0, cd=0x18, aux=0x0 (which matches the board's actual connections) Anyway, if you think the patch below is correct the please apply. It works fine for me. There is one thing that I don't get however: If I run aplay it creates an additional "PCM" mixer control, which however doesn't have any effect. I guess this is intended for softvol, but I don't need it. How do I stop aplay from creating this control? (If it matters, I use Debian unstable, alsa-lib 1.0.15.) Thanks, Johannes --- Enable Analog CD Input from internal ATAPI connector on Asus M2N-SLI. Signed-off-by: Johannes Stezenbach <js(a)sig21.net> diff --git a/sound/pci/hda/patch_analog.c b/sound/pci/hda/patch_analog.c index 196ad3c..8c2aff4 100644 --- a/sound/pci/hda/patch_analog.c +++ b/sound/pci/hda/patch_analog.c @@ -2079,6 +2079,8 @@ static struct hda_verb ad1988_6stack_init_verbs[] = { {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x3c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_ZERO}, {0x34, AC_VERB_SET_CONNECT_SEL, 0x0}, + /* Analog CD Input */ + {0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, { } };

2 1

[alsa-devel] [PATCH] hda: Added STAC92HD73 support.
by Matthew Ranostay 13 Dec '07

13 Dec '07

Added support for new STAC92HD73 family of codecs. Additionally added features for multiple analog loopbacks, and multiple dmux mixers. Regression testing for the analog loopback changes for STAC9205 and STAC9274D completed with any issues, as well for the dmux changes. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r eb09731e73d7 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Dec 04 11:28:40 2007 -0500 @@ -62,6 +62,11 @@ enum { }; enum { + STAC_92HD73XX_REF, + STAC_92HD73XX_MODELS +}; + +enum { STAC_92HD71BXX_REF, STAC_92HD71BXX_MODELS }; @@ -118,6 +123,7 @@ struct sigmatel_spec { unsigned int gpio_mute: 1; unsigned int gpio_mask, gpio_data; + unsigned int aloopback_mask; /* playback */ struct hda_multi_out multiout; @@ -130,7 +136,7 @@ struct sigmatel_spec { unsigned int num_muxes; hda_nid_t *dmic_nids; unsigned int num_dmics; - hda_nid_t dmux_nid; + hda_nid_t *dmux_nids; hda_nid_t dig_in_nid; /* pin widgets */ @@ -145,7 +151,7 @@ struct sigmatel_spec { /* capture source */ struct hda_input_mux *dinput_mux; - unsigned int cur_dmux; + unsigned int cur_dmux[2]; struct hda_input_mux *input_mux; unsigned int cur_mux[3]; @@ -176,12 +182,38 @@ static hda_nid_t stac9200_dac_nids[1] = 0x02, }; +static hda_nid_t stac92hd73xx_adc_nids[2] = { + 0x1a, 0x1b +}; + +#define STAC92HD73XX_NUM_DMICS 2 +static hda_nid_t stac92hd73xx_dmic_nids[STAC92HD73XX_NUM_DMICS + 1] = { + 0x13, 0x14, 0 +}; + +#define STAC92HD73_DAC_COUNT 5 +static hda_nid_t stac92hd73xx_dac_nids[STAC92HD73_DAC_COUNT] = { + 0x15, 0x16, 0x17, 0x18, 0x19, +}; + +static hda_nid_t stac92hd73xx_mux_nids[4] = { + 0x28, 0x29, 0x2a, 0x2b, +}; + +static hda_nid_t stac92hd73xx_dmux_nids[2] = { + 0x20, 0x21, +}; + static hda_nid_t stac92hd71bxx_adc_nids[2] = { 0x12, 0x13, }; static hda_nid_t stac92hd71bxx_mux_nids[2] = { 0x1a, 0x1b +}; + +static hda_nid_t stac92hd71bxx_dmux_nids[1] = { + 0x1c, }; static hda_nid_t stac92hd71bxx_dac_nids[2] = { @@ -226,6 +258,10 @@ static hda_nid_t stac927x_mux_nids[3] = 0x15, 0x16, 0x17 }; +static hda_nid_t stac927x_dmux_nids[1] = { + 0x1b, +}; + #define STAC927X_NUM_DMICS 2 static hda_nid_t stac927x_dmic_nids[STAC927X_NUM_DMICS + 1] = { 0x13, 0x14, 0 @@ -239,6 +275,10 @@ static hda_nid_t stac9205_mux_nids[2] = 0x19, 0x1a }; +static hda_nid_t stac9205_dmux_nids[1] = { + 0x1d, +}; + #define STAC9205_NUM_DMICS 2 static hda_nid_t stac9205_dmic_nids[STAC9205_NUM_DMICS + 1] = { 0x17, 0x18, 0 @@ -257,6 +297,12 @@ static hda_nid_t stac922x_pin_nids[10] = static hda_nid_t stac922x_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, 0x0f, 0x10, 0x11, 0x15, 0x1b, +}; + +static hda_nid_t stac92hd73xx_pin_nids[12] = { + 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, + 0x0f, 0x10, 0x11, 0x12, 0x13, + 0x14, 0x22 }; static hda_nid_t stac92hd71bxx_pin_nids[10] = { @@ -289,8 +335,9 @@ static int stac92xx_dmux_enum_get(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.enumerated.item[0] = spec->cur_dmux; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + + ucontrol->value.enumerated.item[0] = spec->cur_dmux[dmux_idx]; return 0; } @@ -299,9 +346,10 @@ static int stac92xx_dmux_enum_put(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int dmux_idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); return snd_hda_input_mux_put(codec, spec->dinput_mux, ucontrol, - spec->dmux_nid, &spec->cur_dmux); + spec->dmux_nids[dmux_idx], &spec->cur_dmux[dmux_idx]); } static int stac92xx_mux_enum_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) @@ -337,9 +385,11 @@ static int stac92xx_aloopback_get(struct struct snd_ctl_elem_value *ucontrol) { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); - struct sigmatel_spec *spec = codec->spec; - - ucontrol->value.integer.value[0] = spec->aloopback; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); + struct sigmatel_spec *spec = codec->spec; + + ucontrol->value.integer.value[0] = spec->aloopback & + (spec->aloopback_mask << idx); return 0; } @@ -348,24 +398,29 @@ static int stac92xx_aloopback_put(struct { struct hda_codec *codec = snd_kcontrol_chip(kcontrol); struct sigmatel_spec *spec = codec->spec; + unsigned int idx = snd_ctl_get_ioffidx(kcontrol, &ucontrol->id); unsigned int dac_mode; - unsigned int val; - + unsigned int val, idx_val; + + idx_val = (spec->aloopback_mask & 0xFF) << idx; val = !!ucontrol->value.integer.value[0]; - if (spec->aloopback == val) + if ((spec->aloopback | idx) == val) return 0; - spec->aloopback = val; - + spec->aloopback ^= idx_val; + + /* Only return the bits defined by the shift value of the + * first two bytes of the mask */ dac_mode = snd_hda_codec_read(codec, codec->afg, 0, - kcontrol->private_value & 0xFFFF, 0x0); - - if (spec->aloopback) { + kcontrol->private_value & 0xFFFF, 0x0) >> + ((spec->aloopback_mask & 0xFF00) >> 8); + + if (spec->aloopback & idx_val) { snd_hda_power_up(codec); - dac_mode |= 0x40; + dac_mode |= idx_val; } else { snd_hda_power_down(codec); - dac_mode &= ~0x40; + dac_mode &= ~idx_val; } snd_hda_codec_write_cache(codec, codec->afg, 0, @@ -384,6 +439,86 @@ static struct hda_verb stac9200_eapd_ini /* set dac0mux for dac converter */ {0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, {0x08, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, + {} +}; + +static struct hda_verb stac92hd73xx_6ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x00}, + {} +}; + +static struct hda_verb stac92hd73xx_8ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00}, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02}, + /* connect hp ports to dac3 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x03}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x03}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, + {} +}; + +static struct hda_verb stac92hd73xx_10ch_core_init[] = { + /* set master volume and direct control */ + { 0x1f, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, + /* setup audio connections */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x00 }, + { 0x10, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { 0x11, AC_VERB_SET_CONNECT_SEL, 0x02 }, + /* dac3 is connected to import3 mux */ + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, 0xb07f}, + /* connect hp ports to dac4 */ + { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x04}, + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x04}, + /* setup adcs to point to mixer */ + { 0x20, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x21, AC_VERB_SET_CONNECT_SEL, 0x0b}, + { 0x0b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Front Mic */ + { 0x0c, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Mic */ + { 0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, /* Line In */ + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x10, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + { 0x11, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + /* setup import muxs */ + { 0x28, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x29, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2a, AC_VERB_SET_CONNECT_SEL, 0x01}, + { 0x2b, AC_VERB_SET_CONNECT_SEL, 0x03}, {} }; @@ -460,11 +595,11 @@ static struct hda_verb stac9205_core_ini .put = stac92xx_mux_enum_put, \ } -#define STAC_ANALOG_LOOPBACK(verb_read,verb_write) \ +#define STAC_ANALOG_LOOPBACK(verb_read, verb_write, cnt) \ { \ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ .name = "Analog Loopback", \ - .count = 1, \ + .count = cnt, \ .info = stac92xx_aloopback_info, \ .get = stac92xx_aloopback_get, \ .put = stac92xx_aloopback_put, \ @@ -481,6 +616,99 @@ static struct snd_kcontrol_new stac9200_ { } /* end */ }; +static struct snd_kcontrol_new stac92hd73xx_6ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 3), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_8ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 4), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + +static struct snd_kcontrol_new stac92hd73xx_10ch_mixer[] = { + STAC_DIGITAL_INPUT_SOURCE(2), + STAC_ANALOG_LOOPBACK(0xFA0, 0x7A1, 5), + + /* hardware gain controls */ + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x0, 0x13, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_IDX("Digital Mic Volume", 0x1, 0x14, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x20, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x20, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x21, 0x0, HDA_OUTPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x21, 0x0, HDA_OUTPUT), + + HDA_CODEC_VOLUME("Front Mic Mixer Capture Volume", 0x1d, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Mixer Capture Switch", 0x1d, 0, HDA_INPUT), + + HDA_CODEC_VOLUME("Mic Mixer Capture Volume", 0x1d, 0x1, HDA_INPUT), + HDA_CODEC_MUTE("Mic Mixer Capture Switch", 0x1d, 0x1, HDA_INPUT), + + HDA_CODEC_VOLUME("Line In Mixer Capture Volume", 0x1d, 0x2, HDA_INPUT), + HDA_CODEC_MUTE("Line In Mixer Capture Switch", 0x1d, 0x2, HDA_INPUT), + + HDA_CODEC_VOLUME("DAC Mixer Capture Volume", 0x1d, 0x3, HDA_INPUT), + HDA_CODEC_MUTE("DAC Mixer Capture Switch", 0x1d, 0x3, HDA_INPUT), + + HDA_CODEC_VOLUME("CD Mixer Capture Volume", 0x1d, 0x4, HDA_INPUT), + HDA_CODEC_MUTE("CD Mixer Capture Switch", 0x1d, 0x4, HDA_INPUT), + { } /* end */ +}; + static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), @@ -513,7 +741,7 @@ static struct snd_kcontrol_new stac9205_ static struct snd_kcontrol_new stac9205_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(2), - STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1d, 0x0, HDA_OUTPUT), @@ -543,7 +771,7 @@ static struct snd_kcontrol_new stac927x_ static struct snd_kcontrol_new stac927x_mixer[] = { STAC_DIGITAL_INPUT_SOURCE(1), STAC_INPUT_SOURCE(3), - STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB, 1), HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1b, 0x0, HDA_OUTPUT), @@ -851,6 +1079,27 @@ static struct snd_pci_quirk stac925x_cfg SND_PCI_QUIRK(0x107b, 0x0461, "Gateway NX560XL", STAC_MA6), SND_PCI_QUIRK(0x107b, 0x0681, "Gateway NX860", STAC_PA6), SND_PCI_QUIRK(0x1002, 0x437b, "Gateway MX6453", STAC_M2_2), + {} /* terminator */ +}; + +static unsigned int ref92hd73xx_pin_configs[12] = { + 0x02214030, 0x02a19040, 0x01a19020, 0x02214030, + 0x0181302e, 0x01014010, 0x01014020, 0x01014030, + 0x02319040, 0x90a000f0, 0x90a000f0, 0x01452050, +}; + +static unsigned int *stac92hd73xx_brd_tbl[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = ref92hd73xx_pin_configs, +}; + +static const char *stac92hd73xx_models[STAC_92HD73XX_MODELS] = { + [STAC_92HD73XX_REF] = "ref", +}; + +static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { + /* SigmaTel reference board */ + SND_PCI_QUIRK(PCI_VENDOR_ID_INTEL, 0x2668, + "DFI LanParty", STAC_92HD73XX_REF), {} /* terminator */ }; @@ -2030,7 +2279,7 @@ static int stac92xx_auto_create_dmic_inp continue; num_cons = snd_hda_get_connections(codec, - spec->dmux_nid, + spec->dmux_nids[0], con_lst, HDA_MAX_NUM_INPUTS); for (j = 0; j < num_cons; j++) @@ -2211,7 +2460,8 @@ static int stac92xx_parse_auto_config(st spec->mixers[spec->num_mixers++] = spec->kctl_alloc; spec->input_mux = &spec->private_imux; - spec->dinput_mux = &spec->private_dimux; + if (!spec->dinput_mux) + spec->dinput_mux = &spec->private_dimux; return 1; } @@ -2686,6 +2936,111 @@ static int patch_stac925x(struct hda_cod } err = -EINVAL; } + if (err < 0) { + stac92xx_free(codec); + return err; + } + + codec->patch_ops = stac92xx_patch_ops; + + return 0; +} + +static struct hda_input_mux stac92hd73xx_dmux = { + .num_items = 4, + .items = { + { "Analog Inputs", 0x0b }, + { "CD", 0x08 }, + { "Digital Mic 1", 0x09 }, + { "Digital Mic 2", 0x0a }, + } +}; + +static int patch_stac92hd73xx(struct hda_codec *codec) +{ + struct sigmatel_spec *spec; + hda_nid_t conn[STAC92HD73_DAC_COUNT + 2]; + int err = 0; + + spec = kzalloc(sizeof(*spec), GFP_KERNEL); + if (spec == NULL) + return -ENOMEM; + + codec->spec = spec; + spec->num_pins = ARRAY_SIZE(stac92hd73xx_pin_nids); + spec->pin_nids = stac92hd73xx_pin_nids; + spec->board_config = snd_hda_check_board_config(codec, + STAC_92HD73XX_MODELS, + stac92hd73xx_models, + stac92hd73xx_cfg_tbl); +again: + if (spec->board_config < 0) { + snd_printdd(KERN_INFO "hda_codec: Unknown model for" + " STAC92HD73XX, using BIOS defaults\n"); + err = stac92xx_save_bios_config_regs(codec); + if (err < 0) { + stac92xx_free(codec); + return err; + } + spec->pin_configs = spec->bios_pin_configs; + } else { + spec->pin_configs = stac92hd73xx_brd_tbl[spec->board_config]; + stac92xx_set_config_regs(codec); + } + + spec->multiout.num_dacs = snd_hda_get_connections(codec, 0x0a, + conn, STAC92HD73_DAC_COUNT + 2) - 1; + + if (spec->multiout.num_dacs < 0) { + printk(KERN_WARNING "hda_codec: Could not determine " + "number of channels defaulting to DAC count\n"); + spec->multiout.num_dacs = STAC92HD73_DAC_COUNT; + } + + switch (spec->multiout.num_dacs) { + case 0x3: /* 6 Channel */ + spec->mixer = stac92hd73xx_6ch_mixer; + spec->init = stac92hd73xx_6ch_core_init; + break; + case 0x4: /* 8 Channel */ + spec->multiout.hp_nid = 0x18; + spec->mixer = stac92hd73xx_8ch_mixer; + spec->init = stac92hd73xx_8ch_core_init; + break; + case 0x5: /* 10 Channel */ + spec->multiout.hp_nid = 0x19; + spec->mixer = stac92hd73xx_10ch_mixer; + spec->init = stac92hd73xx_10ch_core_init; + }; + + spec->multiout.dac_nids = stac92hd73xx_dac_nids; + spec->aloopback_mask = 0x801; + + spec->mux_nids = stac92hd73xx_mux_nids; + spec->adc_nids = stac92hd73xx_adc_nids; + spec->dmic_nids = stac92hd73xx_dmic_nids; + spec->dmux_nids = stac92hd73xx_dmux_nids; + + spec->num_muxes = ARRAY_SIZE(stac92hd73xx_mux_nids); + spec->num_adcs = ARRAY_SIZE(stac92hd73xx_adc_nids); + spec->num_dmics = STAC92HD73XX_NUM_DMICS; + spec->dinput_mux = &stac92hd73xx_dmux; + /* GPIO0 High = Enable EAPD */ + spec->gpio_mask = spec->gpio_data = 0x000001; + stac92xx_enable_gpio_mask(codec); + + err = stac92xx_parse_auto_config(codec, 0x22, 0x24); + + if (!err) { + if (spec->board_config < 0) { + printk(KERN_WARNING "hda_codec: No auto-config is " + "available, default to model=ref\n"); + spec->board_config = STAC_92HD73XX_REF; + goto again; + } + err = -EINVAL; + } + if (err < 0) { stac92xx_free(codec); return err; @@ -2736,7 +3091,7 @@ again: spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids; - spec->dmux_nid = 0x1c; + spec->dmux_nids = stac92hd71bxx_dmux_nids; spec->num_muxes = ARRAY_SIZE(stac92hd71bxx_mux_nids); spec->num_adcs = ARRAY_SIZE(stac92hd71bxx_adc_nids); @@ -2931,7 +3286,7 @@ static int patch_stac927x(struct hda_cod case 0x10280209: spec->dmic_nids = stac927x_dmic_nids; spec->num_dmics = STAC927X_NUM_DMICS; - spec->dmux_nid = 0x1b; + spec->dmux_nids = stac927x_dmux_nids; /* Enable DMIC0 */ stac92xx_set_config_reg(codec, 0x13, 0x90a60040); @@ -2947,6 +3302,7 @@ static int patch_stac927x(struct hda_cod } spec->multiout.dac_nids = spec->dac_nids; + spec->aloopback_mask = 0x40; stac92xx_enable_gpio_mask(codec); err = stac92xx_parse_auto_config(codec, 0x1e, 0x20); @@ -3004,11 +3360,12 @@ static int patch_stac9205(struct hda_cod spec->num_muxes = ARRAY_SIZE(stac9205_mux_nids); spec->dmic_nids = stac9205_dmic_nids; spec->num_dmics = STAC9205_NUM_DMICS; - spec->dmux_nid = 0x1d; + spec->dmux_nids = stac9205_dmux_nids; spec->init = stac9205_core_init; spec->mixer = stac9205_mixer; + spec->aloopback_mask = 0x40; spec->multiout.dac_nids = spec->dac_nids; switch (spec->board_config){ @@ -3337,6 +3694,8 @@ struct hda_codec_preset snd_hda_preset_s { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, + { .id = 0x111d7675, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d76b0, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, {} /* terminator */ };

2 3

[alsa-devel] [patch 13/18] alsa: nopage
by npiggin＠suse.de 13 Dec '07

13 Dec '07

Convert ALSA from nopage to fault. Switch from OOM to SIGBUS if the resource is not available. Signed-off-by: Nick Piggin <npiggin(a)suse.de> Cc: perex(a)perex.cz Cc: tiwai(a)suse.de Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org --- sound/core/pcm_native.c | 59 ++++++++++++++++++++---------------------------- 1 file changed, 25 insertions(+), 34 deletions(-) Index: linux-2.6/sound/core/pcm_native.c =================================================================== --- linux-2.6.orig/sound/core/pcm_native.c +++ linux-2.6/sound/core/pcm_native.c @@ -3018,26 +3018,23 @@ static unsigned int snd_pcm_capture_poll /* * mmap status record */ -static struct page * snd_pcm_mmap_status_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_status_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->status); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->status); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_status = { - .nopage = snd_pcm_mmap_status_nopage, + .fault = snd_pcm_mmap_status_fault, }; static int snd_pcm_mmap_status(struct snd_pcm_substream *substream, struct file *file, @@ -3061,26 +3058,23 @@ static int snd_pcm_mmap_status(struct sn /* * mmap control record */ -static struct page * snd_pcm_mmap_control_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_control_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; - struct page * page; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - page = virt_to_page(runtime->control); - get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = virt_to_page(runtime->control); + get_page(vmf->page); + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_control = { - .nopage = snd_pcm_mmap_control_nopage, + .fault = snd_pcm_mmap_control_fault, }; static int snd_pcm_mmap_control(struct snd_pcm_substream *substream, struct file *file, @@ -3117,10 +3111,10 @@ static int snd_pcm_mmap_control(struct s #endif /* coherent mmap */ /* - * nopage callback for mmapping a RAM page + * fault callback for mmapping a RAM page */ -static struct page *snd_pcm_mmap_data_nopage(struct vm_area_struct *area, - unsigned long address, int *type) +static int snd_pcm_mmap_data_fault(struct vm_area_struct *area, + struct vm_fault *vmf) { struct snd_pcm_substream *substream = area->vm_private_data; struct snd_pcm_runtime *runtime; @@ -3130,33 +3124,30 @@ static struct page *snd_pcm_mmap_data_no size_t dma_bytes; if (substream == NULL) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; runtime = substream->runtime; - offset = area->vm_pgoff << PAGE_SHIFT; - offset += address - area->vm_start; - snd_assert((offset % PAGE_SIZE) == 0, return NOPAGE_SIGBUS); + offset = vmf->pgoff << PAGE_SHIFT; dma_bytes = PAGE_ALIGN(runtime->dma_bytes); if (offset > dma_bytes - PAGE_SIZE) - return NOPAGE_SIGBUS; + return VM_FAULT_SIGBUS; if (substream->ops->page) { page = substream->ops->page(substream, offset); - if (! page) - return NOPAGE_OOM; /* XXX: is this really due to OOM? */ + if (!page) + return VM_FAULT_SIGBUS; } else { vaddr = runtime->dma_area + offset; page = virt_to_page(vaddr); } get_page(page); - if (type) - *type = VM_FAULT_MINOR; - return page; + vmf->page = page; + return 0; } static struct vm_operations_struct snd_pcm_vm_ops_data = { .open = snd_pcm_mmap_data_open, .close = snd_pcm_mmap_data_close, - .nopage = snd_pcm_mmap_data_nopage, + .fault = snd_pcm_mmap_data_fault, }; /* --

2 1

[alsa-devel] [PATCH] Initial support of the Mitac 8252D (based on ALC883)
by Andy Shevchenko 13 Dec '07

13 Dec '07

Hi! The attached patch adds initial support of the Mitac 8252D (http://www.mitac-mtc.com.tw/English/products/8252Dspec.htm). Working: - Front speakers (volume + mute) - Center/LFE speakers (volume + mute) - HP out (with Front Volume) - HP individual mute switch - HP Jack sense - Front Mic and its volume Not tested: - external mic and its volume Not working while now: - Mic Jack sense Questionable: - is Mic have Jack sense? - one or two Mic volume controls? - CD/Line-in: presense in the mixer Signed-off-by: Andy Shevchenko <andy(a)smile.org.ua> --- alsa-kernel/pci/hda/patch_realtek.c.orig 2007-12-05 17:44:28.000000000 +0300 +++ alsa-kernel/pci/hda/patch_realtek.c 2007-12-05 20:16:46.000000000 +0300 @@ -185,6 +185,7 @@ ALC883_HAIER_W66, ALC888_6ST_HP, ALC888_3ST_HP, + ALC883_MITAC, ALC883_AUTO, ALC883_MODEL_LAST, }; @@ -6342,6 +6343,36 @@ { } /* end */ }; +static struct snd_kcontrol_new alc883_mitac_mixer[] = { + HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), + HDA_CODEC_VOLUME_MONO("Center Playback Volume", 0x0e, 1, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME_MONO("LFE Playback Volume", 0x0e, 2, 0x0, HDA_OUTPUT), + HDA_BIND_MUTE_MONO("Center Playback Switch", 0x0e, 1, 2, HDA_INPUT), + HDA_BIND_MUTE_MONO("LFE Playback Switch", 0x0e, 2, 2, HDA_INPUT), + HDA_CODEC_MUTE("Headphone Playback Switch", 0x15, 0x0, HDA_OUTPUT), + HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Mic Boost", 0x18, 0, HDA_INPUT), + HDA_CODEC_MUTE("Mic Playback Switch", 0x0b, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Playback Volume", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Front Mic Boost", 0x19, 0, HDA_INPUT), + HDA_CODEC_MUTE("Front Mic Playback Switch", 0x0b, 0x1, HDA_INPUT), + HDA_CODEC_VOLUME("Capture Volume", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_MUTE("Capture Switch", 0x08, 0x0, HDA_INPUT), + HDA_CODEC_VOLUME_IDX("Capture Volume", 1, 0x09, 0x0, HDA_INPUT), + HDA_CODEC_MUTE_IDX("Capture Switch", 1, 0x09, 0x0, HDA_INPUT), + { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + /* .name = "Capture Source", */ + .name = "Input Source", + .count = 2, + .info = alc883_mux_enum_info, + .get = alc883_mux_enum_get, + .put = alc883_mux_enum_put, + }, + { } /* end */ +}; + static struct snd_kcontrol_new alc883_3ST_2ch_mixer[] = { HDA_CODEC_VOLUME("Front Playback Volume", 0x0c, 0x0, HDA_OUTPUT), HDA_BIND_MUTE("Front Playback Switch", 0x0c, 2, HDA_INPUT), @@ -6774,6 +6805,67 @@ { } }; +/* toggle speaker-output according to the hp-jack state */ +static void alc883_mitac_hp_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x15, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); +} + +/* auto-toggle front mic */ +/* +static void alc883_mitac_mic_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_codec_read(codec, 0x18, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x0b, HDA_INPUT, 1, HDA_AMP_MUTE, bits); +} +*/ + +static void alc883_mitac_automute(struct hda_codec *codec) +{ + alc883_mitac_hp_automute(codec); + /* alc883_mitac_mic_automute(codec); */ +} + +static void alc883_mitac_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc883_mitac_hp_automute(codec); + break; + case ALC880_MIC_EVENT: + /* alc883_mitac_mic_automute(codec); */ + break; + } +} + +static struct hda_verb alc883_mitac_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Subwoofer */ + {0x17, AC_VERB_SET_CONNECT_SEL, 0x02}, + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + + /* enable unsolicited event */ + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + /* {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, */ + + { } /* end */ +}; + static struct hda_verb alc883_tagra_verbs[] = { {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x0c, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, @@ -7156,6 +7246,7 @@ [ALC883_HAIER_W66] = "haier-w66", [ALC888_6ST_HP] = "6stack-hp", [ALC888_3ST_HP] = "3stack-hp", + [ALC883_MITAC] = "mitac", [ALC883_AUTO] = "auto", }; @@ -7190,6 +7281,7 @@ SND_PCI_QUIRK(0x1025, 0, "Acer laptop", ALC883_ACER), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), SND_PCI_QUIRK(0x161f, 0x2054, "Medion laptop", ALC883_MEDION), + SND_PCI_QUIRK(0x1071, 0x8253, "Mitac 8252d", ALC883_MITAC), SND_PCI_QUIRK(0x1071, 0x8258, "Evesham Voyaeger", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x8086, 0xd601, "D102GGC", ALC883_3ST_6ch), SND_PCI_QUIRK(0x17aa, 0x101e, "Lenovo 101e", ALC883_LENOVO_101E_2ch), @@ -7437,6 +7529,19 @@ .need_dac_fix = 1, .input_mux = &alc883_capture_source, }, + [ALC883_MITAC] = { + .mixers = { alc883_mitac_mixer }, + .init_verbs = { alc883_init_verbs, alc883_mitac_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .num_adc_nids = ARRAY_SIZE(alc883_adc_nids), + .adc_nids = alc883_adc_nids, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_2ch_modes), + .channel_mode = alc883_3ST_2ch_modes, + .input_mux = &alc883_capture_source, + .unsol_event = alc883_mitac_unsol_event, + .init_hook = alc883_mitac_automute, + }, }; -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

3 5

[alsa-devel] [patch] Device ID for MSI L745
by zhejiang 13 Dec '07

13 Dec '07

Hi, Please refer to the [0003641] on bugtracking system. The user found that "model=targa-2ch-dig" works and provided the ID. diff -Nur a/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c b/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c --- a/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c 2007-12-04 09:00:07.000000000 +0800 +++ b/alsa-driver-hg20071211/alsa-kernel/pci/hda/patch_realtek.c 2007-12-11 15:30:20.000000000 +0800 @@ -7173,6 +7173,7 @@ SND_PCI_QUIRK(0x1462, 0x7327, "MSI", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1462, 0x0349, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x0579, "MSI", ALC883_TARGA_2ch_DIG), + SND_PCI_QUIRK(0x1462, 0x040d, "MSI", ALC883_TARGA_2ch_DIG), SND_PCI_QUIRK(0x1462, 0x3729, "MSI S420", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3ef9, "MSI", ALC883_TARGA_DIG), SND_PCI_QUIRK(0x1462, 0x3b7f, "MSI", ALC883_TARGA_2ch_DIG),

2 1

[alsa-devel] [PATCH] 1616(M) cardbus improvements
by Ctirad Fertr 13 Dec '07

13 Dec '07

Hello, this patch improves E-Mu 1616(M) cardbus support. It adds definitions of the new Microdock and 1010 cardbus registers (thanks again for descriptions James) and improves mixer for this card. Now you can use S/PDIF and ADAT on Mirodock and also use headpohone output on host cardbus card as another independent output. Signed-off-by: Ctirad Fertr <c.fertr(a)gmail.com> ------ Ctirad

3 7

[alsa-devel] [PATCH] hda-codec - Fix definition of AC_KNBCAP_DELTA to match datasheet.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Fix definition of AC_KNBCAP_DELTA to match datasheet. AC_KNBCAP_DELTA is incorrectly defined as (1<<8). According to the ALC883 datasheet rev 1.3, this is bit 7 after AC_KNBCAP_NUM_STEPS which is a 0x7f mask. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r da002fb740a42b4fd6dcbd45693f9bee3c828812 -r 65b577e690e95e4ace950c3f986270dd3e992192 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 +++ b/pci/hda/hda_codec.h Mon Dec 03 23:02:51 2007 -0500 @@ -238,7 +238,7 @@ enum { /* Volume knobs capabilities */ #define AC_KNBCAP_NUM_STEPS (0x7f<<0) -#define AC_KNBCAP_DELTA (1<<8) +#define AC_KNBCAP_DELTA (1<<7) /* * Control Parameters

2 1

[alsa-devel] [PATCH] hda-codec - Add missing #defines (and 1 rename) in hda_codec.h.
by Andrew Paprocki 13 Dec '07

13 Dec '07

hda-codec - Add missing #defines (and 1 rename) in hda_codec.h. Added AC_VERB_GET_DIGI_CONVERT_2 and renamed AC_VERB_GET_DIGI_CONVERT to AC_VERB_GET_DIGI_CONVERT_1 to stay consistent with the SET variants. Added AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK, AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK, and AC_PINCAP_LR_SWAP. The missing fields were listed in the ALC883 datasheet rev 1.3. Signed-off-by: Andrew Paprocki <andrew(a)ishiboo.com> diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.c Mon Dec 03 22:52:58 2007 -0500 @@ -1434,8 +1434,9 @@ int snd_hda_create_spdif_out_ctls(struct if (err < 0) return err; } - codec->spdif_ctls = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + codec->spdif_ctls = + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0); codec->spdif_status = convert_to_spdif_status(codec->spdif_ctls); return 0; } @@ -1482,7 +1483,7 @@ static int snd_hda_spdif_in_status_get(s unsigned short val; unsigned int sbits; - val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0); + val = snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT_1, 0); sbits = convert_to_spdif_status(val); ucontrol->value.iec958.status[0] = sbits; ucontrol->value.iec958.status[1] = sbits >> 8; @@ -1533,7 +1534,8 @@ int snd_hda_create_spdif_in_ctls(struct return err; } codec->spdif_in_enable = - snd_hda_codec_read(codec, nid, 0, AC_VERB_GET_DIGI_CONVERT, 0) & + snd_hda_codec_read(codec, nid, 0, + AC_VERB_GET_DIGI_CONVERT_1, 0) & AC_DIG1_ENABLE; return 0; } diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/hda_codec.h --- a/pci/hda/hda_codec.h Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/hda_codec.h Mon Dec 03 22:52:58 2007 -0500 @@ -77,12 +77,14 @@ enum { #define AC_VERB_GET_PIN_SENSE 0x0f09 #define AC_VERB_GET_BEEP_CONTROL 0x0f0a #define AC_VERB_GET_EAPD_BTLENABLE 0x0f0c -#define AC_VERB_GET_DIGI_CONVERT 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_1 0x0f0d +#define AC_VERB_GET_DIGI_CONVERT_2 0x0f0e #define AC_VERB_GET_VOLUME_KNOB_CONTROL 0x0f0f /* f10-f1a: GPIO */ #define AC_VERB_GET_GPIO_DATA 0x0f15 #define AC_VERB_GET_GPIO_MASK 0x0f16 #define AC_VERB_GET_GPIO_DIRECTION 0x0f17 +#deifne AC_VERB_GET_GPIO_UNSOLICITED_RSP_MASK 0x0f19 #define AC_VERB_GET_CONFIG_DEFAULT 0x0f1c /* f20: AFG/MFG */ #define AC_VERB_GET_SUBSYSTEM_ID 0x0f20 @@ -110,6 +112,7 @@ enum { #define AC_VERB_SET_GPIO_DATA 0x715 #define AC_VERB_SET_GPIO_MASK 0x716 #define AC_VERB_SET_GPIO_DIRECTION 0x717 +#define AC_VERB_SET_GPIO_UNSOLICITED_RSP_MASK 0x719 #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_0 0x71c #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_1 0x71d #define AC_VERB_SET_CONFIG_DEFAULT_BYTES_2 0x71e @@ -190,6 +193,7 @@ enum { #define AC_PINCAP_OUT (1<<4) /* output capable */ #define AC_PINCAP_IN (1<<5) /* input capable */ #define AC_PINCAP_BALANCE (1<<6) /* balanced I/O capable */ +#define AC_PINCAP_LR_SWAP (1<<7) /* L/R swap */ #define AC_PINCAP_VREF (0x37<<8) #define AC_PINCAP_VREF_SHIFT 8 #define AC_PINCAP_EAPD (1<<16) /* EAPD capable */ diff -r eb09731e73d7928c70ba776c1719149d112ee96a -r da002fb740a42b4fd6dcbd45693f9bee3c828812 pci/hda/patch_realtek.c --- a/pci/hda/patch_realtek.c Mon Dec 03 17:08:40 2007 +0100 +++ b/pci/hda/patch_realtek.c Mon Dec 03 22:52:58 2007 -0500 @@ -564,7 +564,7 @@ static int alc_spdif_ctrl_get(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long *valp = ucontrol->value.integer.value; unsigned int val = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, 0x00); + AC_VERB_GET_DIGI_CONVERT_1, 0x00); *valp = (val & mask) != 0; return 0; @@ -578,7 +578,7 @@ static int alc_spdif_ctrl_put(struct snd unsigned char mask = (kcontrol->private_value >> 16) & 0xff; long val = *ucontrol->value.integer.value; unsigned int ctrl_data = snd_hda_codec_read(codec, nid, 0, - AC_VERB_GET_DIGI_CONVERT, + AC_VERB_GET_DIGI_CONVERT_1, 0x00); /* Set/unset the masked control bit(s) as needed */

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[alsa-devel] M-audio Delta Audiophile 192 drivers
by Manel Clos 13 Dec '07

13 Dec '07

Hi all! We purchased a M-audio Delta Audiophile 192, which uses the ICE1724 chipset. Using Ubuntu gutsy 7.10 and alsa 1.0.14. PCM is working, but as soon as we start Rosegarden the complete system hangs (SysRq doesn't even work). Also, the input of the card is not shown in gnome-mixer. Is there anything we can do to help the development of the driver? We are able to compile kernel + patches, do C programming, test PCM, Midi... Thanks in advance!

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[alsa-devel] Question about pcm_formats[] array in sound/core/pcm_misc.c
by Timur Tabi 13 Dec '07

13 Dec '07

I'm confused about the entries for unsigned 24-bit sound in the pcm_formats[] array: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, and [SNDRV_PCM_FORMAT_U24_3BE] = { .width = 24, .phys = 24, .le = 0, .signd = 0, .silence = { 0x80, 0x00, 0x00 }, }, SNDRV_PCM_FORMAT_U24_BE means a 24-bit sample packed into a 32-bit word. So I presume that if the 24-bit sample is: 0x123456 then when that sample is written to memory, it looks like this: X X + 1 X + 2 X + 3 0x00 0x12 0x34 0x56 (where X is some address in memory). If that's so, shouldn't the entry for SNDRV_PCM_FORMAT_U24_BE look like this: [SNDRV_PCM_FORMAT_U24_BE] = { .width = 24, .phys = 32, .le = 0, .signd = 0, .silence = { 0x00, 0x80, 0x00, 0x00 }, }, ^^^^^^ -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Sound problems in PCCHIPS A13G+
by alsa-devel.20.pris＠spamgourmet.com 11 Dec '07

11 Dec '07

My motherboard: PCCHIPS A13G+ http://www.pcchips.com.tw/PCCWebSite...uID=44&LanID=0 I'm using Ubuntu 7.10 - the Gutsy Gibbon, I was using Feisty before and had the same problems. The interesting thing: the system sounds all work. Some example .ogg files that came with Ubuntu also work. There is one example file however "fables_01_01_aesop.spx" which only gives a very low sound, almost inaudible. The real strange thing happens with ".avi" and mp3 files. They sound distorted and very low, almost inaudible. Some have better/louder sound, but still distorted, but I can understand what is being spoken, others not. Thanks for any help! If you need more information just ask. Technical details follow(including the Alsa Audio Debug): The sound hardware of my Mobo is in the NVIDIA MCP61S Single chipset. This chipset consists of: GeForce(R) 6100 GPU and MCP nForce(R) 405 codec(less /proc/asound/card0/codec#0): Codec: Realtek ALC660-VD Address: 0 Vendor Id: 0x10ec0660 Subsystem Id: 0x1019c601 Revision Id: 0x100001 The aplay command lists another codec: ALC861VD aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Here is the Alsa Audio Debug: ALSA Audio Debug v0.1.0 - Mon Nov 12 00:00:12 BRST 2007 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux 2.6.22-14-generic #1 SMP Sun Oct 14 23:05:12 GMT 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_hda_intel 263712 4 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 3 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 15 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device snd_page_alloc 11400 2 snd_hda_intel,snd_pcm Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xfe028000 irq 16 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control cat: /proc/asound/hwdep: No such file or directory 00-00: ALC861VD Analog : ALC861VD Analog : playback 1 : capture 2 Client info cur clients : 3 peak clients : 3 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Dev Snd --------------------------------------------------- controlC0 pcmC0D0c pcmC0D0p seq timer CPU ------------------------------------------------------- model name : AMD Processor model unknown cpu MHz : 1900.000 RAM ------------------------------------------------------- MemTotal: 969524 kB SwapTotal: 1951888 kB Hardware -------------------------------------------------- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control

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[alsa-devel] Steinberg Midex 8 driver
by Jason Woodward 11 Dec '07

11 Dec '07

I am wondering if there is still work being done on a driver for the Steinberg Midex 8. I found a message from May 2007 by Norman Schleicher that midi out is working. I had given up hope on this long ago, and am excited at the possiblity of using my Midex in linux. I would love to have even a partially working driver to play with. Thanks, Jason Woodward

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[alsa-devel] Capturing from Line and Mic simultaneously
by Christophe Osuna 11 Dec '07

11 Dec '07

Hi, I would like to capture sound from the Line and Mic inputs simultaneously. The sound chip seems capable to do so: inside alsamixer, in the "Capture" tab I can select up to three subdevices, and from the "Playback" tab I can set the input source one to "Mic" and the input source two to "Line" (and three to whatever). When trying to use several output ports at the same time I use four, six or eight channels. I tried the same thing for input (i.e. use four channels) but that failed. I guess I have to use subdevices but I have no idea on how to start working with them. Could anyone: * tell me if ALSA is limited to two capture channels, and in that case what should be done; * point me to some documentation showing what to do, or give me some general hints Thanks.

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[alsa-devel] cmipci: have only surround40 on 6 ch card
by SeNS 11 Dec '07

11 Dec '07

Hello, All! Sorry for repost; probably, it's a driver problem. I have 6 channel CMI8738-based card (it works in 6 channel mode under win32; and works before under Linux with ALSA). =============================================== lspci output: "0000:00:0a.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10)" =============================================== my kernel version 2.4.34 alsa version 1.0.15 =============================================== cat /proc/asound/cards 0 [CMI8738 ]: CMI8738 - C-Media CMI8738 C-Media CMI8738 (model 37) at 0xe000, irq 10 =============================================== cat /proc/asound/card0/cmipci C-Media PCI CMI8738 (model 37) at 0xe000, irq 10 00: 02 00 01 00 04: 10 1c 00 00 08: 03 00 00 01 0c: 00 00 01 00 10: d4 00 00 40 14: 00 00 00 00 18: 00 80 80 08 1c: fa 4c 04 33 20: 10 01 00 00 24: 20 01 ff 00 28: 00 00 00 00 2c: 00 00 00 00 30: ff ff ff ff 34: 00 00 00 00 38: 00 00 00 00 3c: 00 00 00 00 =============================================== WBR, SeNS

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Re: [alsa-devel] [alsa-cvslog] alsa-kernel: cmipci: disable "Modem" control on version 39 or newer chips
by Thierry Vignaud 10 Dec '07

10 Dec '07

Clemens Ladisch <clemens(a)ladisch.de> writes: > changeset: 5582:5082de4abb26 > tag: tip > user: cladisch > date: Mon Dec 10 12:28:52 2007 +0100 > files: pci/cmipci.c > description: > cmipci: disable "Modem" control on version 39 or newer chips > > On version 39 or newer chips, we better remove the "Modem" control > because this register bit now mutes the front channels of the > multichannel stream. shouldn't that deserve a comment in the code? > diff -r eb09731e73d7 -r 5082de4abb26 pci/cmipci.c > --- a/pci/cmipci.c Mon Dec 03 17:08:40 2007 +0100 > +++ b/pci/cmipci.c Mon Dec 10 12:28:52 2007 +0100 > @@ -2651,10 +2651,8 @@ static struct snd_kcontrol_new snd_cmipc > }; > > /* card control switches */ > -static struct snd_kcontrol_new snd_cmipci_control_switches[] __devinitdata = { > - // DEFINE_CARD_SWITCH("Joystick", joystick), /* now module option */ > - DEFINE_CARD_SWITCH("Modem", modem), > -}; > +static struct snd_kcontrol_new snd_cmipci_modem_switch __devinitdata = > +DEFINE_CARD_SWITCH("Modem", modem); > > > static int __devinit snd_cmipci_mixer_new(struct cmipci *cm, int pcm_spdif_device) > @@ -2735,9 +2733,9 @@ static int __devinit snd_cmipci_mixer_ne > } > > /* card switches */ > - sw = snd_cmipci_control_switches; > - for (idx = 0; idx < ARRAY_SIZE(snd_cmipci_control_switches); idx++, sw++) { > - err = snd_ctl_add(cm->card, snd_ctl_new1(sw, cm)); > + if (cm->chip_version < 39) { > + err = snd_ctl_add(cm->card, > + snd_ctl_new1(&snd_cmipci_modem_switch, cm)); > if (err < 0) > return err; > } > > > ------------------------------------------------------------------------- > SF.Net email is sponsored by: > Check out the new SourceForge.net Marketplace. > It's the best place to buy or sell services for > just about anything Open Source. > http://sourceforge.net/services/buy/index.php > _______________________________________________ > Alsa-cvslog mailing list > Alsa-cvslog(a)lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/alsa-cvslog

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[alsa-devel] Segfault in mix_areas3_smp with Edirol UA-25 in advance-mode
by Lasse Kärkkäinen 10 Dec '07

10 Dec '07

No asound.conf, just the default dmix configuration. Trying to play audio via this card randomly causes crashes. Might be a race condition between closing the PCM at the same time as opening a new one or something like that, but I didn't debug it further. I get this with both Xine and GStreamer, using ALSA as audio output. Using other sound cards or using UA-25 without advance mode¹ there are no segfaults. Advanced Linux Sound Architecture Driver Version 1.0.14 (Fri Jul 20 09:12:58 2007 UTC). alsa-lib version: 1.0.14a GStreamer: #0 0x00002b54714d58dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b54714d6477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b54714d6a4e in snd_pcm_dmix_start () from /usr/lib/libasound.so.2 #3 0x00002b54714bff60 in snd_pcm_rate_start () from /usr/lib/libasound.so.2 #4 0x00002b54714ac357 in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b54714b4ca2 in snd_pcm_mmap_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaf89dc1c in gst_alsasink_write (asink=0x1bf1620, data=0x1a51a40, length=3760) at gstalsasink.c:786 #7 0x00002aaaadf46ca0 in audioringbuffer_thread_func (buf=<value optimized out>) at gstaudiosink.c:226 #8 0x00002b5472bf72b4 in g_thread_create_proxy () from /usr/lib/libglib-2.0.so.0 #9 0x00002b547202bfe7 in start_thread () from /lib/libpthread.so.0 Xine: #0 0x00002b9c574d88dc in mix_areas3_smp () from /usr/lib/libasound.so.2 #1 0x00002b9c574d9477 in snd_pcm_dmix_sync_area () from /usr/lib/libasound.so.2 #2 0x00002b9c574d98fa in snd_pcm_dmix_mmap_commit () from /usr/lib/libasound.so.2 #3 0x00002b9c574b96f0 in snd_pcm_plugin_write_areas () from /usr/lib/libasound.so.2 #4 0x00002b9c574af29a in snd_pcm_write_areas () from /usr/lib/libasound.so.2 #5 0x00002b9c574b9052 in snd_pcm_plugin_writei () from /usr/lib/libasound.so.2 #6 0x00002aaaaefa8078 in ao_alsa_write () from /usr/lib64/xine/plugins/1.1.8/xineplug_ao_out_alsa.so #7 0x00002b9c542397a7 in ao_loop () from /usr/lib/libxine.so.1 #8 0x00002b9c5802efe7 in start_thread () from /lib/libpthread.so.0 ¹) There is a switch on this USB sound card for switching between modes. In advance mode the device is not standard USB Audio device, but it supports 24 bit packed samples and sample rates higher than 44.1 kHz. Otherwise it is a standard USB audio device, but only supports 16 bit 44.1 kHz.

2 2

[alsa-devel] Query regarding external alsa plugins
by Pharaoh . 10 Dec '07

10 Dec '07

Hi I am writing an external IO type plugin for my board since my audio driver is going to be a user space driver i.e. no alsa kernel driver because of some technical reasons. My pluging works okay and now I want to provide a mixer control interface like volume control and playback path selection etc. Is there anything like mixer plugin etc which can be intergrated with my existing plugin? Can I get a test code somewhere? Regards, Sandeep.

1 1

[alsa-devel] hdsp rpm support
by Roman Haefeli 10 Dec '07

10 Dec '07

hi all i just wanted to ask quickly what the current state is of support for the hdsp rpm. i found some mails in the alsa-user list archive, mentioning that adding support is not so difficult and is going to be done within a week. this was in january 2005. if i can help with something, i'd be glad to do so. i am not a coder, but i have a rpm box myself. roman ___________________________________________________________ Telefonate ohne weitere Kosten vom PC zum PC: http://messenger.yahoo.de

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[alsa-devel] can't use dmixer
by feng_tian 08 Dec '07

08 Dec '07

Hi, all: I have a problem with dmix of alsa. I'm using alsa v1.0.11rc2 now. And make it running on ATMEL AT91SAM9260EK board. It runs ok without Dmix. But when I tried: bash> aplay -D dmixer test.wav The errors are: ALSA lib pcm_direct.c:790:(snd_pcm_direct_initialize_slave) slave plugin does not support mmap interleaved or mmap noninterleaved access ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to initialize slave aplay: main:547: audio open error: Invalid argument The configuration of my alsa in /etc/asound.con is: pcm.card0 { type hw card 0 } pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } ctl.dmixer { type hw card 0 device 0 } pcm.dsp { type plug slave.pcm "dmixer" #use our new PCM here } ctl.mixer{ type hw card 0 } bash> aplay -L gives: PCM list: hw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type hw card $CARD device $DEV subdevice $SUBDEV } plughw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type plug slave.pcm { type hw card $CARD device $DEV subdevice $SUBDEV } } plug { @args.0 SLAVE @args.SLAVE { type string } type plug slave.pcm $SLAVE } shm { @args.0 SOCKET @args.1 PCM @args.SOCKET { type string } @args.PCM { type string } type shm server $SOCKET pcm $PCM } tee { @args.0 SLAVE @args.1 FILE @args.2 FORMAT @args.SLAVE { type string } @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm $SLAVE file $FILE format $FORMAT } file { @args.0 FILE @args.1 FORMAT @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT } null { type null } cards 'cards.pcm' front 'cards.pcm.front' rear 'cards.pcm.rear' center_lfe 'cards.pcm.center_lfe' side 'cards.pcm.side' surround40 'cards.pcm.surround40' surround41 'cards.pcm.surround41' surround50 'cards.pcm.surround50' surround51 'cards.pcm.surround51' surround71 'cards.pcm.surround71' iec958 'cards.pcm.iec958' spdif 'cards.pcm.iec958' modem 'cards.pcm.modem' phoneline 'cards.pcm.phoneline' dmix 'cards.pcm.dmix' dsnoop 'cards.pcm.dsnoop' card0 { type hw card 0 } default { type plug slave.pcm dmixer } dmixer { type dmix ipc_key 1024 slave { pcm 'hw:0,0' format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } dsp { type plug slave.pcm dmixer } Is there anyone experienced in this kind of error? Any helps or advices is wecome. BR Feng

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[alsa-devel] rme hdsp pcmcia (and probably pci) broken
by Fernando Lopez-Lezcano 08 Dec '07

08 Dec '07

Hi, anyone has a pcmcia rme card running? And working? With a reasonably recent alsa? (this probably applies also for the regular pci rme cards, I've had reports on the Planet CCRMA list about problems that sound similar to this one). My test setup is on fc7 running: alsa-lib 1.0.15 alsa-tools 1.0.15 alsa-firmware 1.0.15 After a sequence of reboots this is what I get: 2.6.22.1-1.rt5.1 + alsa driver modules from svn 20070702.165519 --> OK 2.6.22.5-1.rt9.2 + alsa driver modules from hg20070830 --> OK 2.6.22.6-1.rt9.5 + alsa driver modules from hg20070830 --> OK 2.6.22.10-1.rt9.1 + alsa driver modules from hg20071012 --> BROKEN 2.6.23.1-1.rt11.1 + alsa drive modules from hg20071124 --> BROKEN BROKEN means: after starting hdspmixer I get no sound from the analog outputs (all previous tries play sound correctly with the default preset in hdspmixer). Some channels do appear (using a Digiface II). In the "broken" drivers I can get sound with the default preset if I send a signal to channels 7 and 12, all other stay muted. Something broke drastically in the driver between 2007.08.30 and 2007.10.12. Any help appreciated... -- Fernando

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Re: [alsa-devel] Please help me with USB Audio drivers.
by Clemens Ladisch 07 Dec '07

07 Dec '07

(please don't top-post) kevin liu wrote: > Yeah, I get silence, mplayer not abort. > Also I can not record anything from the device I am using. It's possible that the device needs some special configuration to select what is to be recorded from the audio interface. Please show the output of "lsusb -v" for this device. Regards, Clemens

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Re: [alsa-devel] New Realtec ALC833 Codec in Fujitsu Siemens Xi2428
by Die Mumie 07 Dec '07

07 Dec '07

So, first: I have a -HDMI Connector -Jack for Headphones (HP) and SPDIF -Jack for Mic or Line -Two built in Mics * -Built in Speakers (I Guess that they are Stereo, but I'm not sure) * What I can test is: Headphones Internel Speakers Internal Microphone so I'm going trough the model list. 3stack-dig - Sound at HP, no IS, no Mic 6stack-dig - Front gives Sound at HP, Surround at IS, no Mic 3stack-6ch - Front gives Sound at HP, Surround at IS, no Mic 3stack-6ch-dig - Front gives Sound at HP, no IS, no Mic 6stack-dig-demo - Front gives sound at HP and IS at the same time, no switching by Jack, no Mic acer - Front gives Sound at HP, Surround at IS, no Mic acer-aspire - Switchs between HP and IS by plugging the HP, Headphones switch Enables/Disables sound on HP, no Mic medion - no Sound medion-md2 - sound on HP when HP switch is enabled, no mic targa-dig - sound on HP when HP switch is enabled, Surround gives on IS, no Mic targa-2ch-dig - sound on HP when HP switch is enabled, no mic laptop-eapd - Front on HP, Surround on IS, no mic lenovo-101e - Sound on Front Jack switching, no Mic lenovo-nb0763 - Sound on Speaker Jack switching, no Mic lenovo-ms7195-dig - Sound on Front same time at IS and HP, no Mic haier-w66 - Front on HP, no Mic 6stack-hp - Nothing 3stack-hp - Front on HP, Center on IS, no Mic I hope this helps Andy Shevchenko <andy(a)smile.org.ua> schrieb: Hi Die Mumie! On Wed, Dec 05, 2007 at 01:47:07AM +0100, Die Mumie wrote next: > Also, try one of the supported models for the ALC883: > 3stack-dig 3-jack with SPDIF I/O > 6stack-dig 6-jack digital with SPDIF I/O > 3stack-6ch 3-jack 6-channel > 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O > 6stack-dig-demo 6-jack digital for Intel demo board > acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) > acer-aspire Acer Aspire 9810 > medion Medion Laptops > medion-md2 Medion MD2 > targa-dig Targa/MSI > targa-2ch-dig Targs/MSI with 2-channel > laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) > lenovo-101e Lenovo 101E > lenovo-nb0763 Lenovo NB0763 > lenovo-ms7195-dig Lenovo MS7195 > haier-w66 Haier W66 > 6stack-hp HP machines with 6stack (Nettle boards) > 3stack-hp HP machines with 3stack (Lucknow, Samba boards) > auto auto-config reading BIOS (default) Did you hear some sound in the HP or internel speakers with someone of these models? If yes, please, say what is wrong with that sound. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua --------------------------------- Ihr erstes Fernweh? Wo gibt es den schönsten Strand.

2 1

[alsa-devel] Please help me with USB Audio drivers.
by kevin liu 07 Dec '07

07 Dec '07

Dear: I am using a tv card which has a standard usb audio interface, when I plug the card, snd-usb-audio finds my usb audio interface, And I use mplayer to test the effect, unluckly, I could not get and sound. And I give mplayer these parameters: """ mplayer alsa:adevice=hw.1 ,0:volume=100:immediatemode=0:amode=1:audiorate=48000:forc eaudio """ I am getting so frustrated by this problem. I don't know how to let USB Audio driver to drive my audio interface. Please help me.

2 1

[alsa-devel] Can I control volume of each virtual device?
by feng_tian 07 Dec '07

07 Dec '07

Hi, all: We are on a project using alsa. There are three applications will use the sound card, but we only have one physical card. My plan is to create virtual device for each application. How I'm gonna control the volume of each device, which interface to use? I want this feature because when one application need to be exclusive, it can mute the other virtual devices. Any comments or advices is welcome. Thanks BR Feng

2 1

[alsa-devel] Making ALSA apps run on OSS driver
by Pharaoh . 06 Dec '07

06 Dec '07

Hi For my OMAP 2430 board, I have a working OSS driver , but all my audio/video applications are written for ALSA, since we had all ALSA drivers for previous board. Now I want to use the OSS driver as it is for 1 week and run my applications using it, I know this is not the best thing to do but currently I dont have time to write ALSA driver from scratch. I have heard about OSS emulation in ALSA i.e. any read/write to /dev/dsp will go to my ALSA driver but is there any way to do the reverse i.e. apps using ALSA interface should somehow talk to OSS driver? What all changes I need to make and where? -Pharaoh.

2 2

Re: [alsa-devel] New Realtec ALC833 Codec in Fujitsu Siemens Xi2428
by Die Mumie 05 Dec '07

05 Dec '07

Ok, doing as you wish. There is no useable Documentation for this Notebook, because ist seems to work with "great Vista", all I could find is a Guide for installing the internal Microphone under Vista. I'll attach 3 files, -The Vista guide -The Alsa-Info output -The dmesg output Tobin Davis <tdavis(a)dsl-only.net> schrieb: The helps immensely, thanks. For some reason, pastebin seems to be having issues. Could you email me directly the alsa-info.txt file that the alsa-info.sh script generated? It should be in your /tmp directory. Also, if you have a user's manual in pdf form for your system, please attach it as well (english version). This may gleen some info about the specs of your system that may have been overlooked, especially the mic. The last bit I'd like you to do, if you are up for it, is to rebuild the drivers with debugging turned on. To do this, download alsa-driver-1.0.15.tar.bz2 and extract it. Then, in the directory it extracted to, type "./configure --with-debug=detect --with-cards=hda-intel && make && make install". When you load this driver, set model=auto. This will dump the bios configuration for the card (if any) into dmesg. I need that output as well. I'll try to work on this this week, but no guarantees. We had some serious flooding this last weekend, and I'm busy fixing my house. Based on the info below, I'll need to create a separate model for your system. It may take a few patches, but it will happen. Tobin On Wed, 2007-12-05 at 12:26 +0100, Die Mumie wrote: So, first: I have a -HDMI Connector -Jack for Headphones (HP) and SPDIF -Jack for Mic or Line -Two built in Mics * -Built in Speakers (I Guess that they are Stereo, but I'm not sure) * What I can test is: Headphones Internel Speakers Internal Microphone so I'm going trough the model list. 3stack-dig - Sound at HP, no IS, no Mic 6stack-dig - Front gives Sound at HP, Surround at IS, no Mic 3stack-6ch - Front gives Sound at HP, Surround at IS, no Mic 3stack-6ch-dig - Front gives Sound at HP, no IS, no Mic 6stack-dig-demo - Front gives sound at HP and IS at the same time, no switching by Jack, no Mic acer - Front gives Sound at HP, Surround at IS, no Mic acer-aspire - Switchs between HP and IS by plugging the HP, Headphones switch Enables/Disables sound on HP, no Mic medion - no Sound medion-md2 - sound on HP when HP switch is enabled, no mic targa-dig - sound on HP when HP switch is enabled, Surround gives on IS, no Mic targa-2ch-dig - sound on HP when HP switch is enabled, no mic laptop-eapd - Front on HP, Surround on IS, no mic lenovo-101e - Sound on Front Jack switching, no Mic lenovo-nb0763 - Sound on Speaker Jack switching, no Mic lenovo-ms7195-dig - Sound on Front same time at IS and HP, no Mic haier-w66 - Front on HP, no Mic 6stack-hp - Nothing 3stack-hp - Front on HP, Center on IS, no Mic I hope this helps Andy Shevchenko <andy(a)smile.org.ua> schrieb: Hi Die Mumie! On Wed, Dec 05, 2007 at 01:47:07AM +0100, Die Mumie wrote next: > Also, try one of the supported models for the ALC883: > 3stack-dig 3-jack with SPDIF I/O > 6stack-dig 6-jack digital with SPDIF I/O > 3stack-6ch 3-jack 6-channel > 3stack-6ch-dig 3-jack 6-channel with SPDIF I/O > 6stack-dig-demo 6-jack digital for Intel demo board > acer Acer laptops (Travelmate 3012WTMi, Aspire 5600, etc) > acer-aspire Acer Aspire 9810 > medion Medion Laptops > medion-md2 Medion MD2 > targa-dig Targa/MSI > targa-2ch-dig Targs/MSI with 2-channel > laptop-eapd 3-jack with SPDIF I/O and EAPD (Clevo M540JE, M550JE) > lenovo-101e Lenovo 101E > lenovo-nb0763 Lenovo NB0763 > lenovo-ms7195-dig Lenovo MS7195 > haier-w66 Haier W66 > 6stack-hp HP machines with 6stack (Nettle boards) > 3stack-hp HP machines with 3stack (Lucknow, Samba boards) > auto auto-config reading BIOS (default) Did you hear some sound in the HP or internel speakers with someone of these models? If yes, please, say what is wrong with that sound. -- Tobin Davis There are two types of people in this world, good and bad. The good sleep better, but the bad seem to enjoy the waking hours much more. -- Woody Allen --------------------------------- Jetzt Mails schnell in einem Vorschaufenster überfliegen. Dies und viel mehr bietet das neue Yahoo! Mail. name=root&type=33&description=/tmp/alsa-info.txt&expiry=&s=Submit+Post&content= !!################################ !!ALSA Information Script v 0.4.34 !!################################ !!Script ran on: Wed Dec 5 01:29:45 CET 2007 !!Linux Distribution !!------------------ Gentoo Base System release 1.12.9 !!Kernel Information !!------------------ Kernel release: 2.6.22-gentoo-r9 Operating System: GNU/Linux Architecture: i686 Processor: Intel(R) Core(TM)2 Duo CPU T7250 @ 2.00GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.15 Library version: 1.0.14a Utilities version: 1.0.14 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf0600000 irq 23 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:284b (rev 03) Subsystem: 1734:1107 -- Region 0: Memory at f0402000 (64-bit, non-prefetchable) [size=128] Region 2: Memory at f0400000 (64-bit, non-prefetchable) [size=8K] !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel enable : N id : <NULL> index : -1 model : <NULL> position_fix : 0 power_save : 0 power_save_controller : Y probe_mask : -1 single_cmd : N !!HDA-Intel Codec information !!--------------------------- Codec: Motorola Si3054 Address: 0 Vendor Id: 0x10573055 Subsystem Id: 0x10573055 Revision Id: 0x100700 Modem Function Group: 0x1 Codec: Realtek ALC883 Address: 1 Vendor Id: 0x10ec0883 Subsystem Id: 0x17341236 Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 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: [0x00 0x00] PCM: rates [0x160]: 44100 48000 96000 bits [0x6]: 16 20 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=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: [0x00 0x00] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 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: [0x1f 0x1f] 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: [0x12 0x12] 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: [0x00 0x00] 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 0x0121441f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green EAPD: 0x0 Pin-ctls: 0xc0: OUT HP 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 0x99130110: [Fixed] Speaker at Int ATAPI Conn = ATAPI, Color = Unknown EAPD: 0x0 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: [0x80 0x80] Pincap 0x083e: IN OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black EAPD: 0x0 Pin-ctls: 0x20: IN 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: [0x80 0x80] Pincap 0x083e: IN OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black EAPD: 0x0 Pin-ctls: 0x20: IN 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 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink EAPD: 0x0 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 0x99a3093f: [Fixed] Mic at Int ATAPI Conn = ATAPI, Color = Unknown EAPD: 0x0 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 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black EAPD: 0x0 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 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black EAPD: 0x0 Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black EAPD: 0x0 Pin-ctls: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x598301f0: [N/A] Line In at Int ATAPI Conn = ATAPI, Color = Unknown EAPD: 0x0 Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x99430120: [Fixed] SPDIF Out at Int ATAPI Conn = ATAPI, Color = Unknown EAPD: 0x0 Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black EAPD: 0x0 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] [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 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 !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Dec 4 21:29 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Dec 4 21:29 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 5 Dec 4 21:29 /dev/snd/hwC0D1 crw-rw---- 1 root audio 116, 24 Dec 4 21:29 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Dec 4 21:29 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 17 Dec 4 21:29 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 26 Dec 4 21:29 /dev/snd/pcmC0D2c crw-rw---- 1 root audio 116, 30 Dec 4 21:29 /dev/snd/pcmC0D6c crw-rw---- 1 root audio 116, 22 Dec 4 21:29 /dev/snd/pcmC0D6p crw-rw---- 1 root audio 116, 1 Dec 4 20:29 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Dec 4 21:29 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: ALC883 Digital [ALC883 Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: Intel [HDA Intel], device 2: ALC883 Analog [ALC883 Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 card 0: Intel [HDA Intel], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 18 [58%] [-19.50dB] [on] Front Right: Playback 18 [58%] [-19.50dB] [on] Simple mixer control 'Front Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Front Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [on] Front Right: Capture 0 [0%] [-12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 0 [0%] [-12.00dB] [on] Front Right: Capture 0 [0%] [-12.00dB] [on] Simple mixer control 'Caller ID',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Input Source',0 Capabilities: enum Items: 'Mic' 'Front Mic' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' Item0: 'Mic' Simple mixer control 'Off-hook',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!All Loaded Modules !!------------------ Module ipv6 bridge llc snd_pcm_oss snd_mixer_oss snd_seq_oss snd_seq_midi_event snd_seq snd_seq_device iwl4965 snd_hda_intel snd_pcm snd_timer snd_page_alloc snd_hwdep snd nvidia ACPI: PCI interrupt for device 0000:00:1b.0 disabled ACPI: PCI Interrupt 0000:00:1b.0[A] -> GSI 22 (level, low) -> IRQ 23 PCI: Setting latency timer of device 0000:00:1b.0 to 64 hda_codec: Unknown model for ALC883, trying auto-probe from BIOS... ALSA /var/tmp/portage/media-sound/alsa-driver-9999/work/alsa-driver/pci/hda/hda_codec.c:2792: autoconfig: line_outs=1 (0x15/0x0/0x0/0x0/0x0) ALSA /var/tmp/portage/media-sound/alsa-driver-9999/work/alsa-driver/pci/hda/hda_codec.c:2796: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA /var/tmp/portage/media-sound/alsa-driver-9999/work/alsa-driver/pci/hda/hda_codec.c:2800: hp_outs=1 (0x14/0x0/0x0/0x0/0x0) ALSA /var/tmp/portage/media-sound/alsa-driver-9999/work/alsa-driver/pci/hda/hda_codec.c:2808: inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0

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[alsa-devel] Mindprint DI-MOD USB
by Christian Schumann 05 Dec '07

05 Dec '07

Hi, in janoary, I wrote to this list about an extension of the usb-audio driver for the Mindprint DI-MOD USB, but neither hat the necessary information at hand, no became a reply. In the meantime I did some research and here's what I found out: The DI-MOD USB is pretty usb-audio class compliant with the exception of the spdif output, which has to be activated manaually. I did this with the windows driver, but it would be nice to be able to do that with ALSA. I have a Envoice MkII with the DI-MOD USB, and my suggestion would be to automatically enable the spdif output when initializing the driver. The spdif output does not have any other effect on the card functionality. This could for example be included into the setup quirks for usb interfaces. The info I got from Markus Medau at usb-audio.com is the following: Digital: Vendorrequest, Typ: Out/Vendorinterface, bReq setCur, wVal 0x00, wIndex 0x04, data length 1: data 0x01 Analog: Vendorrequest, Typ: Out/Vendorinterface, bReq setCur, wVal 0x00, wIndex 0x02, data length 1: data 0x00 I don't have experience in writing drivers, but if someone can give me a clue where to start, I could give it a try. Regards Christian -- ------------------------------------------------------------------------ Dipl.-Phys. Christian Schumann |Technische Universitaet Kaiserslautern Mail: schumann(a)physik.uni-kl.de | Fachbereich Physik Tel.: 0631/205-4842 (Office) | http://www.physik.uni-kl.de Tel.: 0631/205-4843 (Lab) | Fax.: 0631/205-3902 | Post: Erwin-Schroedinger-Strasse, D-67663 Kaiserslautern -------------------------------------------------------------------------

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[alsa-devel] imedabid@yahoo.fr
by abid imed 05 Dec '07

05 Dec '07

imedabid(a)yahoo.fr _____________________________________________________________________________ Ne gardez plus qu'une seule adresse mail ! Copiez vos mails vers Yahoo! Mail

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[alsa-devel] Alsa problems: can't use dmix
by feng_tian 05 Dec '07

05 Dec '07

Hi, all: I have a problem with dmix of alsa. I'm using alsa v1.0.11rc2 now. And make it running on ATMEL AT91SAM9260EK board. It runs ok without Dmix. But when I tried: bash> aplay -D dmixer test.wav The errors are: ALSA lib pcm_direct.c:790:(snd_pcm_direct_initialize_slave) slave plugin does not support mmap interleaved or mmap noninterleaved access ALSA lib pcm_dmix.c:864:(snd_pcm_dmix_open) unable to initialize slave aplay: main:547: audio open error: Invalid argument The configuration of my alsa in /etc/asound.con is: pcm.card0 { type hw card 0 } pcm.!default { type plug slave.pcm "dmixer" } pcm.dmixer { type dmix ipc_key 1024 slave { pcm "hw:0,0" format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } ctl.dmixer { type hw card 0 device 0 } pcm.dsp { type plug slave.pcm "dmixer" #use our new PCM here } ctl.mixer{ type hw card 0 } bash> aplay -L gives: PCM list: hw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type hw card $CARD device $DEV subdevice $SUBDEV } plughw { @args.0 CARD @args.1 DEV @args.2 SUBDEV @args.CARD { type string default { @func getenv vars { 0 ALSA_PCM_CARD 1 ALSA_CARD } default { @func refer name 'defaults.pcm.card' } } } @args.DEV { type integer default { @func igetenv vars { 0 ALSA_PCM_DEVICE } default { @func refer name 'defaults.pcm.device' } } } @args.SUBDEV { type integer default { @func refer name 'defaults.pcm.subdevice' } } type plug slave.pcm { type hw card $CARD device $DEV subdevice $SUBDEV } } plug { @args.0 SLAVE @args.SLAVE { type string } type plug slave.pcm $SLAVE } shm { @args.0 SOCKET @args.1 PCM @args.SOCKET { type string } @args.PCM { type string } type shm server $SOCKET pcm $PCM } tee { @args.0 SLAVE @args.1 FILE @args.2 FORMAT @args.SLAVE { type string } @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm $SLAVE file $FILE format $FORMAT } file { @args.0 FILE @args.1 FORMAT @args.FILE { type string } @args.FORMAT { type string default raw } type file slave.pcm null file $FILE format $FORMAT } null { type null } cards 'cards.pcm' front 'cards.pcm.front' rear 'cards.pcm.rear' center_lfe 'cards.pcm.center_lfe' side 'cards.pcm.side' surround40 'cards.pcm.surround40' surround41 'cards.pcm.surround41' surround50 'cards.pcm.surround50' surround51 'cards.pcm.surround51' surround71 'cards.pcm.surround71' iec958 'cards.pcm.iec958' spdif 'cards.pcm.iec958' modem 'cards.pcm.modem' phoneline 'cards.pcm.phoneline' dmix 'cards.pcm.dmix' dsnoop 'cards.pcm.dsnoop' card0 { type hw card 0 } default { type plug slave.pcm dmixer } dmixer { type dmix ipc_key 1024 slave { pcm 'hw:0,0' format S16_LE period_time 0 period_size 1024 buffer_size 4096 rate 22050 } bindings { 0 0 1 1 } } dsp { type plug slave.pcm dmixer } Any helps or advices is wecome. BR Feng

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[alsa-devel] New Realtec ALC833 Codec in Fujitsu Siemens Xi2428
by Die Mumie 05 Dec '07

05 Dec '07

Hi, I've been told to post here, so I do. I have a Notebook with a Intel hda sound device, the /proc/asound/card0/codec#1 returns Realtec ALC833 but there is no model provided by the alsa driver that runs this Soundcard properly. Am I wrong here? If yes, then I'm sorry. --------------------------------- Heute schon einen Blick in die Zukunft von E-Mails wagen? Versuchen Sie´s mit dem neuen Yahoo! Mail.

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[alsa-devel] POSIX clocks and ALSA
by Heikki Lindholm 04 Dec '07

04 Dec '07

Hello, Some years ago there was some talk about UST support in Linux, but the support never happened. With the hrtimers patch (and I'm not quite sure if even earlier?) CLOCK_MONOTONIC would seem like a fairly good UST time source. What I'd like to see, is a selectable clock for ALSA timestamping, e.g. something like snd_sw_params_clock(..., clockid_t clk). Would this seem plausible? I don't know that much about ALSA internals, so, no idea whether different clocks on different pcms/whatever would quickly turn into an unmanageable mess. -- Heikki Lindholm

3 15

[alsa-devel] handling x-runs
by srinivasa bp 04 Dec '07

04 Dec '07

Hi all, I am new to ALSA. But was able to bring up two applications to play and record pcm. But now i am facing problems with X-Runs . I am using the API snd_pcm_prepare but somehow not able to recover from it. Can any one please guide me regarding or please let me know if there is any doc available to do this .... thanks -- - Regards, Srini

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Re: [alsa-devel] rme multiface I=>pccard error -12
by rmgls＠free.fr 04 Dec '07

04 Dec '07

> At Fri, 30 Nov 2007 15:11:08 +0100, > rmgls(a)free.fr wrote: >> >> >> Hello, >> >> on a laptop the old multiface I with a pccard (1) doesn't attach. >> acore returns error -12. >> >> details: >> dmesg: >> pccard: CardBus card inserted into slot 0 >> ALSA /root/alsa-driver-hg20071129/acore/init.c:174: cannot find the slot for index 0 (range 0-0), error: -16 > Check /proc/asound/cards. The index 0 is likely already used by > another device. Since the recent distro loads the sound drivers > automatically via udev, you'll have to set index for all drivers > properly. > If it's snd-hda-intel, add the option like below: > options snd-hda-intel index=1 > Takashi Hello Takashi, thanks for this info, and sorry for the noise. all worlks fine now. Best regards rmgls(a)free.fr

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[alsa-devel] May be typos in the current ALSA?
by Andy Shevchenko 03 Dec '07

03 Dec '07

Hi! I think the following notices should be reviewed as a typo fix. Could you tell me do I wrong? /* patch_realtek.c */ ... static struct snd_pci_quirk alc861_cfg_tbl[] = { SND_PCI_QUIRK(0x1043, 0x1205, "ASUS W7J", ALC861_3ST), SND_PCI_QUIRK(0x1043, 0x1335, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1338, "ASUS F2/3", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x13d7, "ASUS A9rp", ALC861_ASUS_LAPTOP), >>> SND_PCI_QUIRK(0x1584, 0x9075, "Airis Praxis N1212", ALC861_ASUS_LAPTOP), SND_PCI_QUIRK(0x1043, 0x1393, "ASUS", ALC861_ASUS), SND_PCI_QUIRK(0x1043, 0x81cb, "ASUS P1-AH2", ALC861_3ST_DIG), SND_PCI_QUIRK(0x1179, 0xff00, "Toshiba", ALC861_TOSHIBA), /* FIXME: the entry below breaks Toshiba A100 (model=auto works!) * Any other models that need this preset? */ /* SND_PCI_QUIRK(0x1179, 0xff10, "Toshiba", ALC861_TOSHIBA), */ SND_PCI_QUIRK(0x1584, 0x9072, "Uniwill m31", ALC861_UNIWILL_M31), >>> SND_PCI_QUIRK(0x1584, 0x9075, "Uniwill", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1584, 0x2b01, "Uniwill X40AIx", ALC861_UNIWILL_M31), SND_PCI_QUIRK(0x1849, 0x0660, "Asrock 939SLI32", ALC660_3ST), SND_PCI_QUIRK(0x8086, 0xd600, "Intel", ALC861_3ST), SND_PCI_QUIRK(0x1462, 0x7254, "HP dx2200 (MSI MS-7254)", ALC861_3ST), SND_PCI_QUIRK(0x1462, 0x7297, "HP dx2250 (MSI MS-7297)", ALC861_3ST), {} }; ... The second record was added by myself (I filed bug report) when I discover RoverBook Voyager V200. First one was added later without checks. And the lines at the same file ... /* * generic initialization of ADC, input mixers and output mixers */ static struct hda_verb alc883_auto_init_verbs[] = { ... /* FIXME: use matrix-type input source selection */ /* Mixer elements: 0x18, 19, 1a, 1b, 1c, 1d, 14, 15, 16, 17, 0b */ /* Input mixer1 */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, /* {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, /* Input mixer2 */ {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(1)}, {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(2)}, /* {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(3)}, */ >>> {0x23, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(4)}, { } }; The same initialization is present above in few lines. P.S. I've donloaded ALSA just few hours ago in daily snapshot tarball form. -- With best regards, Andy Shevchenko. mailto: andy(a)smile.org.ua

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[alsa-devel] rme multiface I=>pccard error -12
by rmgls＠free.fr 03 Dec '07

03 Dec '07

Hello, on a laptop the old multiface I with a pccard (1) doesn't attach. acore returns error -12. details: dmesg: pccard: CardBus card inserted into slot 0 ALSA /root/alsa-driver-hg20071129/acore/init.c:174: cannot find the slot for index 0 (range 0-0), error: -16 RME Hammerfall DSP: probe of 0000:07:00.0 failed with error -12 config: machine: sony vaio s5 kernel 2-6-24-rc3 all alsa-*-hg as you seen. sound box: multiface I and pccard (I), the old one. Socket 0 Bridge: [yenta_cardbus] (bus ID: 0000:06:05.0) lspcmcia: Configuration: state: on ready: yes Voltage: 3.3V Vcc: 3.3V Vpp: 3.3V lspci -v 06:05.0 CardBus bridge: Texas Instruments PCI7420 CardBus Controller Subsystem: Sony Corporation Unknown device 818f Flags: bus master, medium devsel, latency 168, IRQ 17 Memory at b0008000 (32-bit, non-prefetchable) [size=4K] Bus: primary=06, secondary=07, subordinate=0a, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 54000000-57fff000 I/O window 0: 00002400-000024ff I/O window 1: 00002800-000028ff 16-bit legacy interface ports at 0001 07:00.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall DSP (rev 10) Flags: medium devsel, IRQ 17 Memory at 54000000 (32-bit, non-prefetchable) [disabled] [size=64K] cat /dev/sndstat: ... 0: alc-260 any help would be appreciated. thanks. best regards raoul rmgls(a)free.fr

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[alsa-devel] Patch - switching rate in STAC9460 codec of Prodigy192
by Pavel Hofman 03 Dec '07

03 Dec '07

Hi, Here is a patch for switching rate in STAC9460 codec. Thanks, Pavel. Signed-off-by: Pavel Hofman <dustin(a)seznam.cz> * support for switching rate in STAC9460 - using set_rate_val of the akm infrastructure * listing all STAC9460 registers in proc * disabling mpu401 device for Prodigy192 - otherwise the currently flawed mpu401 code hangs kernel when opening the midi device * removing old unused commented-out code

2 1

[alsa-devel] I2C fix for ice1724
by Pavel Hofman 03 Dec '07

03 Dec '07

Hello Takashi, Below is a patch for ice1724 I2C read code. Thanks, Pavel. Signed-off-by: Pavel Hofman <dustin(a)seznam.cz> * adding i2c busy wait before sending device address to prevent reading bogus data diff -r 7f4b25ff79d6 pci/ice1712/ice1724.c --- a/pci/ice1712/ice1724.c Fri Nov 30 17:59:25 2007 +0100 +++ b/pci/ice1712/ice1724.c Fri Nov 30 21:17:36 2007 +0100 @@ -1958,6 +1958,7 @@ unsigned char snd_vt1724_read_i2c(struct unsigned char val; mutex_lock(&ice->i2c_mutex); + wait_i2c_busy(ice); outb(addr, ICEREG1724(ice, I2C_BYTE_ADDR)); outb(dev & ~VT1724_I2C_WRITE, ICEREG1724(ice, I2C_DEV_ADDR)); wait_i2c_busy(ice);

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[alsa-devel] help
by YanBob 02 Dec '07

02 Dec '07

hello, my linux is ubuntu with Linux bob-laptop 2.6.22-14-generic #1 SMP Sun Oct 14 23:05:12 GMT 2007 i686 GNU/Linux. now the problem is that,no sound comes out when i use ubuntu. i have google some pages, and followed the steps as they said. But i turns out that my sound card is still not working. At the end of this email, i send you some inf. about my sound card. And my laptop model is Lenovo F41. Could you please help me? thanks a million!! bob@bob-laptop:~$ aplay -l **** PLAYBACK硬件设备列表 **** 卡 0: Intel [HDA Intel], 设备 0: ALC262 Analog [ALC262 Analog] 子设备: 1/1 子设备：#0: subdevice #0 卡 0: Intel [HDA Intel], 设备 1: ALC262 Digital [ALC262 Digital] 子设备: 1/1 子设备：#0: subdevice #0 卡 0: Intel [HDA Intel], 设备 6: Si3054 Modem [Si3054 Modem] 子设备: 1/1 子设备：#0: subdevice #0 bob@bob-laptop:~$ lspci -vv|grep Audio 00:1b.0 Audio device: Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03) bob@bob-laptop:~$ cat /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xf8300000 irq 22 bob@bob-laptop:~$ lsmod|grep snd snd_hda_intel 263712 1 snd_pcm_oss 44672 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80388 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4740 0 snd_seq_oss 33152 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8448 2 snd_seq_oss,snd_seq_midi snd_seq 53232 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 2 snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 54660 11 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8800 1 snd snd_page_alloc 11400 2 snd_hda_intel,snd_pcm bob@bob-laptop:~$ cat /proc/asound/version Advanced Linux Sound Architecture Driver Version 1.0.14 (Thu May 31 09:03:25 2007 UTC). ___________________________________________________________ 进入雅虎游戏嘉年华，赢取液晶显示器！ http://cn.mail.yahoo.com/promo/carnival07/

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Re: [alsa-devel] multi pcm and mmap problem
by Takashi Iwai 02 Dec '07

02 Dec '07

At Wed, 21 Nov 2007 16:27:52 -0500, Tim wrote: > > On Wednesday 21 November 2007 06:01:22 am you wrote: > > At Wed, 21 Nov 2007 01:58:41 -0500, > > > > terminator356(a)users.sourceforge.net wrote: > > > After 2 years I've finally traced and fixed a problem > > > which was preventing me from using anything after > > > ALSA 1.0.9b ! > > > > > > In ALSA lib 1.0.9b: > > > pcm/pcm_generic.c: > > > int snd_pcm_generic_channel_info() > > > { > > > snd_pcm_generic_t *generic = pcm->private_data; > > > return snd_pcm_channel_info(generic->slave, info); > > > } > > > > > > But in ALSA lib 1.0.15: > > > pcm/pcm_generic.c: > > > int snd_pcm_generic_channel_info() > > > { > > > snd_pcm_generic_t *generic = pcm->private_data; > > > if (pcm->mmap_shadow) { > > > /* No own buffer is required - the plugin won't change > > > * the data on the buffer, or do safely on-the-place > > > * conversion > > > */ > > > return snd_pcm_channel_info(generic->slave, info); > > > } else { > > > // Tim: TESTED: This is now called instead of the other > > > one! /* Allocate own buffer */ > > > return snd_pcm_channel_info_shm(pcm, info, -1); > > > } > > > } > > > > > > My multi PCM consists of a plughw:0,2 and a hw:1,0 > > > In ALSA 1.0.9b, all five PCMs ALSA creates for this arrangement > > > create their areas with mmap(). And it works. > > > But in ALSA 1.0.15 my hw:1,0 and its multi PCM areas are > > > created with mmap(), and my plughw:0,2 and its PCMs areas > > > are created with malloc(). > > > Can't hear/record any sound from the plughw:0,2 part of the multi. > > > > > > I have verified that ALSA 1.0.15 is copying the plughw:0,2 HW PCM > > > areas to its LINEAR PCM areas, but NOT finally to its multi PCM areas. > > > It just copies all zeros. > > > That's the reason I hear/record no sound from the plughw:0,2 part > > > of the multi PCM. > > > So I reverted the function's code back to 1.0.9b and presto! > > > It works now, the same as in ALSA 1.0.9b > > > > > > So please tell me, am I missing something? > > > > Well, first of all, could you elaborate what is the real problem you > > are facing? Judging from your description above, it sounds like a > > clear bug, but I'm not sure where to start. > > > > In addition to the detailed bug description, a testcase program would > > be greatly helpful. > > > > OK. First, here's my simple testing .asoundrc : > pcm.multi_rec { > type multi; > slaves.a.pcm "hw:1,0"; > slaves.a.channels 12; > slaves.b.pcm "plughw:0,2"; > slaves.b.channels 16; > bindings.0.slave a; > bindings.0.channel 4; > bindings.1.slave b; > bindings.1.channel 10; > } > ctl.multi_rec { > type hw; > card 0; > } > > The two cards are SBLive! (emu10k1) and Delta1010LT (ice1712). > > And here is the command line I run: > arecord -I -v -t wav -c2 -r 48000 -f S32_LE -Dmulti_rec -d 3 \ > --buffer-size 128 arecord-multi-ice-emu.wav > > My problem is under ALSA 1.0.15 no sound is recorded from slave 'b' - > (the plughw:0,2 channel). > When I look at the two generated wav files, the first has data, and the second > has all zeros. > > So as I said before, I found that ALSA is copying data from the > plughw:0,2 HW PCM area to its LINEAR PCM area (i.e. the actual conversion > is taking place), but then somehow the data is not finally copied to its > multi slave b PCM area - just all zeros are copied. > Hence all zeros are stored in the second wav file. > > But with ALSA 1.0.9b, since all the areas are allocated with mmap(), > it works fine. It is only after ALSA 1.0.9b, that I get the problem. > Because ALSA 1.0.15 is allocating both the plughw:0,2 LINEAR PCM area > and multi slave slave b's PCM area with malloc(). > It seems ALSA doesn't want to copy to areas created with malloc(). OK, point taken. The problem is that the multi plugin creates unnecessary local buffer. It works somehow when the slave is a hw, but it doesn't for other types. Try the patch below. This should fix the problem. Now the multi plugin just shadows the slave buffers via mmap. > > > Should I add something more to my .asoundrc ? > > > I see that besides {type multi ... }, there is a {type share ...} > > > defined. *** Does the 'share' have any use here? Can't find any docs on > > > it. > > > > The share type is used only for share plugin, so you can ignore it. > > > > > Also: > > > Yes, I can see now, why you made this code change. > > > mmap() has limits, so you switched over to malloc(). > > > Those limits force the user to use small buffers > > > and/or reduce the number of channels created. > > > > No, mmap was never used between plugins. It was only for the hw layer > > even in the earlier verson. In the earlier version, SHM was used > > instead. But, the plugin buffer is referred only locally, so it makes > > no sense to use SHM for that. That's why now it was switched to > > malloc. (snd_pcm_channel_info_shm() is a rest of the old behavior, > > and thus its name is confusing...) > > > My point was this: > Under ALSA 1.0.9b, (and with my fix to ALSA 1.0.15 to force it to use > mmap() always - see above), if I run: > arecord -I -v -t wav -c2 -r 48000 -f S32_LE -Dmulti_rec -d 3 \ > --buffer-size 128 arecord-multi-ice-emu.wav > it will run. But if I run > arecord -I -v -t wav -c2 -r 48000 -f S32_LE -Dmulti_rec -d 3 \ > --buffer-size 256 arecord-multi-ice-emu.wav > it fails with "mmap() failed: Invalid argument". > I quadrupled the available locked memory with a line in > /etc/security/limits.conf > * - memlock 128 > but the second command line still fails. > What am I doing wrong? How can I increase mmap() limits? It must be irrelevant now... Takashi diff -r 3539f279ec38 src/pcm/pcm_multi.c --- a/src/pcm/pcm_multi.c Wed Nov 21 12:19:43 2007 +0100 +++ b/src/pcm/pcm_multi.c Fri Nov 23 14:40:39 2007 +0100 @@ -690,13 +690,43 @@ static snd_pcm_sframes_t snd_pcm_multi_m return size; } -static int snd_pcm_multi_mmap(snd_pcm_t *pcm ATTRIBUTE_UNUSED) +static int snd_pcm_multi_munmap(snd_pcm_t *pcm) { + free(pcm->mmap_channels); + free(pcm->running_areas); + pcm->mmap_channels = NULL; + pcm->running_areas = NULL; return 0; } -static int snd_pcm_multi_munmap(snd_pcm_t *pcm ATTRIBUTE_UNUSED) +static int snd_pcm_multi_mmap(snd_pcm_t *pcm) { + snd_pcm_multi_t *multi = pcm->private_data; + unsigned int c; + + pcm->mmap_channels = calloc(pcm->channels, + sizeof(pcm->mmap_channels[0])); + pcm->running_areas = calloc(pcm->channels, + sizeof(pcm->running_areas[0])); + if (!pcm->mmap_channels || !pcm->running_areas) { + snd_pcm_multi_munmap(pcm); + return -ENOMEM; + } + + /* Copy the slave mmapped buffer data */ + for (c = 0; c < pcm->channels; c++) { + snd_pcm_multi_channel_t *chan = &multi->channels[c]; + snd_pcm_t *slave; + if (chan->slave_idx < 0) { + snd_pcm_multi_munmap(pcm); + return -ENXIO; + } + slave = multi->slaves[chan->slave_idx].pcm; + pcm->mmap_channels[c] = + slave->mmap_channels[chan->slave_channel]; + pcm->running_areas[c] = + slave->running_areas[chan->slave_channel]; + } return 0; } @@ -850,6 +880,7 @@ int snd_pcm_multi_open(snd_pcm_t **pcmp, return err; } pcm->mmap_rw = 1; + pcm->mmap_shadow = 1; /* has own mmap method */ pcm->ops = &snd_pcm_multi_ops; pcm->fast_ops = &snd_pcm_multi_fast_ops; pcm->private_data = multi;

3 9

[alsa-devel] Why is a USB capture device showing up in "aplay -L"?
by Chris Rankin 01 Dec '07

01 Dec '07

Hi, I have a webcam which has microphone-only USB audio capability. However, not only has this device stopped recording anything but it has also vanished from the "arecord -L" list and appeared in the "aplay -L" one instead: $ arecord -L default:CARD=Audigy2 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Default Audio Device front:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Front speakers rear:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Rear speakers center_lfe:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Center and Subwoofer speakers side:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Side speakers surround40:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 4.0 Surround output to Front and Rear speakers surround41:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) and: $ aplay -L default:CARD=Audigy2 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Default Audio Device front:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Front speakers rear:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Rear speakers center_lfe:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Center and Subwoofer speakers side:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback Side speakers surround40:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 4.0 Surround output to Front and Rear speakers surround41:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=Audigy2,DEV=0 Audigy 2 ZS [SB0353], ADC Capture/Standard PCM Playback IEC958 (S/PDIF) Digital Audio Output null Discard all samples (playback) or generate zero samples (capture) default:CARD=WebCam USB Device 0x46d:0x8b4, USB Audio Default Audio Device front:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio Front speakers surround40:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio 4.0 Surround output to Front and Rear speakers surround41:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio 4.1 Surround output to Front, Rear and Subwoofer speakers surround50:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio 5.0 Surround output to Front, Center and Rear speakers surround51:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio 5.1 Surround output to Front, Center, Rear and Subwoofer speakers surround71:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio 7.1 Surround output to Front, Center, Side, Rear and Woofer speakers iec958:CARD=WebCam,DEV=0 USB Device 0x46d:0x8b4, USB Audio IEC958 (S/PDIF) Digital Audio Output Here is my .asoundrc file: ctl.Audigy2 { type hw card 0 } pcm.Audigy2 { type plug slave { pcm "hw:0,0" } } ctl.webcam { type hw card 1 } pcm.webcam { type plug slave { pcm "hw:1,0" } } and here is what /proc/asound looks like: ]$ ls -als /proc/asound/ total 0 0 dr-xr-xr-x 6 root root 0 2007-12-01 15:55 . 0 dr-xr-xr-x 172 root root 0 2007-12-01 14:37 .. 0 lrwxrwxrwx 1 root root 5 2007-12-01 15:55 Audigy2 -> card0 0 dr-xr-xr-x 11 root root 0 2007-12-01 15:55 card0 0 dr-xr-xr-x 3 root root 0 2007-12-01 15:55 card1 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 cards 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 devices 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 hwdep 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 modules 0 dr-xr-xr-x 2 root root 0 2007-12-01 15:55 oss 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 pcm 0 dr-xr-xr-x 2 root root 0 2007-12-01 15:55 seq 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 timers 0 -r--r--r-- 1 root root 0 2007-12-01 15:55 version 0 lrwxrwxrwx 1 root root 5 2007-12-01 15:55 WebCam -> card1 This happens with both Fedora 7 and Fedora 8. Could there be a problem with the snd-usb-audio kernel module, please? Cheers, Chris ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

1 0

Re: [alsa-devel] alsa-devel on vger
by Jaroslav Kysela 30 Nov '07

30 Nov '07

On Thu, 22 Nov 2007, Takashi Iwai wrote: > At Wed, 21 Nov 2007 20:14:34 +0100 (CET), > Jaroslav Kysela wrote: > > > > On Wed, 21 Nov 2007, Rene Herman wrote: > > > > > On 21-11-07 17:54, Takashi Iwai wrote: > > > > > > > What about the plan to move alsa-devel to vger? > > > > Shouldn't we do that soon? > > > > > > Jaroslav hasn't yet pointed alsa-devel(a)alsa-project.org at vger. > > > It's a little unfortunate that David wouldn't just transfer > > > subscriptions as it seems that's where things stalled again. Given > > > our still recent move from sourceforge it really should be okay to > > > just move everyone over. > > > > > > I noticed that the entirety of people thanking David for setting up > > > the list consisted of alsa-devel moderators, you and me, and I > > > suspect most other subscribers don't feel there's actually much > > > wrong with things as they are today and mostly consider moving again > > > a nuisance. The one other reaction to the move was a slightly > > > annoyed reaction about the re-subscribing... > > > > I'm playing with idea to redirect alsa-devel(a)alsa-project.org to vger and > > leave current mailing list (which is subscribed to vger now too). So every > > e-mail to this list will go through vger, and resubscription will be only > > choice of subscriber. Opinions? > > A couple of things: > > - the official address (on kernel tree and wiki) should be vger, not > alsa-project.org, and promote to re-subscribe. Changing e-mail address of list is something I don't want. The official e-mail address for alsa-devel mailing list was always alsa-devel(a)alsa-project.org (even in time when this list was on sourceforge). Unfortunately, vger mailing list does not accept simple redirected e-mails from alsa-devel(a)alsa-project.org so I postponed any actions. > - open alsa-project.org list again. I'll stop list admin works. I tried to count some statistics and at the time spamassassin and mailman filtering rules catch 92% of spam messages coming to alsa-devel. Unfortunately remaining 8% means 20 spam messages daily (without moderation). Could moderators confirm this value? Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

3 7

[alsa-devel] [PATCH] sound/isa: kill pnp_resource_change.
by Rene Herman 30 Nov '07

30 Nov '07

Hi Takashi. This removes the pnp_resource_change use from the ALSA ISAPnP drivers. In 2.4 these were useful in providing an easy path to setting the resources, but in 2.6 they retain function as a layering violation only. This makes for a nice cleanup (-550 lines) of ALSA but moreover, ALSA is the only remaining user of pnp_init_resource_table(), pnp_resource_change() and pnp_manual_config_dev() (and, in fact, of "struct pnp_resource_table") in the tree outide of drivers/pnp itself meaning it makes for more cleanup potential inside the PnP layer. Thomas Renninger acked their removal from that side, you did from the ALSA side (CC list just copied from that thread). Against current alsa-kernel HG. Many more potential cleanups in there, but this _only_ removes the pnp_resource_change code. Compile tested against current alsa-kernel HG and compile- and use-tested against 2.6.23.x (few offsets). Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Documentation/ALSA-Configuration.txt | 35 ------------ isa/ad1816a/ad1816a.c | 55 ++----------------- isa/als100.c | 49 ---------------- isa/azt2320.c | 50 ----------------- isa/cmi8330.c | 42 +------------- isa/cs423x/cs4236.c | 101 +++++++---------------------------- isa/dt019x.c | 48 +--------------- isa/es18xx.c | 58 +++----------------- isa/gus/interwave.c | 39 +------------ isa/opl3sa2.c | 36 ------------ isa/opti9xx/opti92x-ad1848.c | 50 +---------------- isa/sb/es968.c | 27 --------- isa/sb/sb16.c | 39 ------------- isa/wavefront/wavefront.c | 55 +------------------ 14 files changed, 65 insertions(+), 619 deletions(-) diff --git a/Documentation/ALSA-Configuration.txt b/Documentation/ALSA-Configuration.txt --- a/Documentation/ALSA-Configuration.txt +++ b/Documentation/ALSA-Configuration.txt @@ -150,13 +150,6 @@ Prior to version 0.9.0rc4 options had a Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips. - port - port # for AD1816A chip (PnP setup) - mpu_port - port # for MPU-401 UART (PnP setup) - fm_port - port # for OPL3 (PnP setup) - irq - IRQ # for AD1816A chip (PnP setup) - mpu_irq - IRQ # for MPU-401 UART (PnP setup) - dma1 - first DMA # for AD1816A chip (PnP setup) - dma2 - second DMA # for AD1816A chip (PnP setup) clockfreq - Clock frequency for AD1816A chip (default = 0, 33000Hz) This module supports multiple cards, autoprobe and PnP. @@ -203,14 +196,6 @@ Prior to version 0.9.0rc4 options had a Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips. - port - port # for ALS100 (SB16) chip (PnP setup) - irq - IRQ # for ALS100 (SB16) chip (PnP setup) - dma8 - 8-bit DMA # for ALS100 (SB16) chip (PnP setup) - dma16 - 16-bit DMA # for ALS100 (SB16) chip (PnP setup) - mpu_port - port # for MPU-401 UART (PnP setup) - mpu_irq - IRQ # for MPU-401 (PnP setup) - fm_port - port # for OPL3 FM (PnP setup) - This module supports multiple cards, autoprobe and PnP. The power-management is supported. @@ -304,15 +289,6 @@ Prior to version 0.9.0rc4 options had a Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only). - port - port # for AZT2320 chip (PnP setup) - wss_port - port # for WSS (PnP setup) - mpu_port - port # for MPU-401 UART (PnP setup) - fm_port - FM port # for AZT2320 chip (PnP setup) - irq - IRQ # for AZT2320 (WSS) chip (PnP setup) - mpu_irq - IRQ # for MPU-401 UART (PnP setup) - dma1 - 1st DMA # for AZT2320 (WSS) chip (PnP setup) - dma2 - 2nd DMA # for AZT2320 (WSS) chip (PnP setup) - This module supports multiple cards, PnP and autoprobe. The power-management is supported. @@ -504,13 +480,6 @@ Prior to version 0.9.0rc4 options had a Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP only) - - port - Port # (PnP setup) - mpu_port - Port # for MPU-401 (PnP setup) - fm_port - Port # for FM OPL-3 (PnP setup) - irq - IRQ # (PnP setup) - mpu_irq - IRQ # for MPU-401 (PnP setup) - dma8 - DMA # (PnP setup) This module supports multiple cards. This module is enabled only with ISA PnP support. @@ -609,10 +578,6 @@ Prior to version 0.9.0rc4 options had a Module for sound cards based on ESS ES968 chip (PnP only). - port - port # for ES968 (SB8) chip (PnP setup) - irq - IRQ # for ES968 (SB8) chip (PnP setup) - dma1 - DMA # for ES968 (SB8) chip (PnP setup) - This module supports multiple cards, PnP and autoprobe. The power-management is supported. diff --git a/isa/ad1816a/ad1816a.c b/isa/ad1816a/ad1816a.c --- a/isa/ad1816a/ad1816a.c +++ b/isa/ad1816a/ad1816a.c @@ -61,20 +61,6 @@ MODULE_PARM_DESC(id, "ID string for ad18 MODULE_PARM_DESC(id, "ID string for ad1816a based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable ad1816a based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for ad1816a driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for ad1816a driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for ad1816a driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for ad1816a driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for ad1816a driver."); -module_param_array(dma1, int, NULL, 0444); -MODULE_PARM_DESC(dma1, "1st DMA # for ad1816a driver."); -module_param_array(dma2, int, NULL, 0444); -MODULE_PARM_DESC(dma2, "2nd DMA # for ad1816a driver."); module_param_array(clockfreq, int, NULL, 0444); MODULE_PARM_DESC(clockfreq, "Clock frequency for ad1816a driver (default = 0)."); @@ -117,16 +103,12 @@ static int __devinit snd_card_ad1816a_pn const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -EBUSY; - } + acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); if (acard->devmpu == NULL) { mpu_port[dev] = -1; @@ -134,25 +116,10 @@ static int __devinit snd_card_ad1816a_pn } pdev = acard->dev; - pnp_init_resource_table(cfg); - - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], port[dev], 16); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "AUDIO PnP configure failure\n"); - kfree(cfg); return -EBUSY; } @@ -162,20 +129,11 @@ static int __devinit snd_card_ad1816a_pn dma2[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - if (acard->devmpu == NULL) { - kfree(cfg); + if (acard->devmpu == NULL) return 0; - } + pdev = acard->devmpu; - pnp_init_resource_table(cfg); - - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { printk(KERN_ERR PFX "MPU401 PnP configure failure\n"); @@ -186,7 +144,6 @@ static int __devinit snd_card_ad1816a_pn mpu_irq[dev] = pnp_irq(pdev, 0); } - kfree(cfg); return 0; } diff --git a/isa/als100.c b/isa/als100.c --- a/isa/als100.c +++ b/isa/als100.c @@ -63,20 +63,6 @@ MODULE_PARM_DESC(id, "ID string for als1 MODULE_PARM_DESC(id, "ID string for als100 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable als100 based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for als100 driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for als100 driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for als100 driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for als100 driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for als100 driver."); -module_param_array(dma8, int, NULL, 0444); -MODULE_PARM_DESC(dma8, "8-bit DMA # for als100 driver."); -module_param_array(dma16, int, NULL, 0444); -MODULE_PARM_DESC(dma16, "16-bit DMA # for als100 driver."); struct snd_card_als100 { int dev_no; @@ -111,38 +97,20 @@ static int __devinit snd_card_als100_pnp const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } + acard->devmpu = pnp_request_card_device(card, id->devs[1].id, acard->dev); acard->devopl = pnp_request_card_device(card, id->devs[2].id, acard->dev); pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (dma16[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma16[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -152,13 +120,6 @@ static int __devinit snd_card_als100_pnp pdev = acard->devmpu; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) goto __mpu_error; @@ -176,11 +137,6 @@ static int __devinit snd_card_als100_pnp pdev = acard->devopl; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], fm_port[dev], 4); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "OPL3 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) goto __fm_error; @@ -195,7 +151,6 @@ static int __devinit snd_card_als100_pnp fm_port[dev] = -1; } - kfree(cfg); return 0; } diff --git a/isa/azt2320.c b/isa/azt2320.c --- a/isa/azt2320.c +++ b/isa/azt2320.c @@ -72,22 +72,6 @@ MODULE_PARM_DESC(id, "ID string for azt2 MODULE_PARM_DESC(id, "ID string for azt2320 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable azt2320 based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for azt2320 driver."); -module_param_array(wss_port, long, NULL, 0444); -MODULE_PARM_DESC(wss_port, "WSS Port # for azt2320 driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for azt2320 driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for azt2320 driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for azt2320 driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for azt2320 driver."); -module_param_array(dma1, int, NULL, 0444); -MODULE_PARM_DESC(dma1, "1st DMA # for azt2320 driver."); -module_param_array(dma2, int, NULL, 0444); -MODULE_PARM_DESC(dma2, "2nd DMA # for azt2320 driver."); struct snd_card_azt2320 { int dev_no; @@ -121,43 +105,19 @@ static int __devinit snd_card_azt2320_pn const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; - acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } acard->devmpu = pnp_request_card_device(card, id->devs[1].id, NULL); pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (wss_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], wss_port[dev], 4); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -169,13 +129,6 @@ static int __devinit snd_card_azt2320_pn pdev = acard->devmpu; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) goto __mpu_error; @@ -191,7 +144,6 @@ static int __devinit snd_card_azt2320_pn mpu_port[dev] = -1; } - kfree (cfg); return 0; } diff --git a/isa/cmi8330.c b/isa/cmi8330.c --- a/isa/cmi8330.c +++ b/isa/cmi8330.c @@ -286,39 +286,21 @@ static int __devinit snd_cmi8330_pnp(int const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->cap = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->cap == NULL) { - kfree(cfg); + if (acard->cap == NULL) return -EBUSY; - } + acard->play = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->play == NULL) { - kfree(cfg); + if (acard->play == NULL) return -EBUSY; - } pdev = acard->cap; - pnp_init_resource_table(cfg); - /* allocate AD1848 resources */ - if (wssport[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], wssport[dev], 8); - if (wssdma[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], wssdma[dev], 1); - if (wssirq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], wssirq[dev], 1); - - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP manual resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "CMI8330/C3D (AD1848) PnP configure failure\n"); - kfree(cfg); return -EBUSY; } wssport[dev] = pnp_port_start(pdev, 0); @@ -327,23 +309,10 @@ static int __devinit snd_cmi8330_pnp(int /* allocate SB16 resources */ pdev = acard->play; - pnp_init_resource_table(cfg); - if (sbport[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], sbport[dev], 16); - if (sbdma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], sbdma8[dev], 1); - if (sbdma16[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], sbdma16[dev], 1); - if (sbirq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], sbirq[dev], 1); - - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP manual resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "CMI8330/C3D (SB16) PnP configure failure\n"); - kfree(cfg); return -EBUSY; } sbport[dev] = pnp_port_start(pdev, 0); @@ -351,7 +320,6 @@ static int __devinit snd_cmi8330_pnp(int sbdma16[dev] = pnp_dma(pdev, 1); sbirq[dev] = pnp_irq(pdev, 0); - kfree(cfg); return 0; } #endif diff --git a/isa/cs423x/cs4236.c b/isa/cs423x/cs4236.c --- a/isa/cs423x/cs4236.c +++ b/isa/cs423x/cs4236.c @@ -270,29 +270,9 @@ MODULE_DEVICE_TABLE(pnp_card, snd_cs423x MODULE_DEVICE_TABLE(pnp_card, snd_cs423x_pnpids); /* WSS initialization */ -static int __devinit snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) -{ - int err; - - pnp_init_resource_table(cfg); - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 4); - if (fm_port[dev] != SNDRV_AUTO_PORT && fm_port[dev] > 0) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (sb_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], sb_port[dev], 16); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev] < 0 ? 4 : dma2[dev], 1); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR IDENT " WSS PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { +static int __devinit snd_cs423x_pnp_init_wss(int dev, struct pnp_dev *pdev) +{ + if (pnp_activate_dev(pdev) < 0) { printk(KERN_ERR IDENT " WSS PnP configure failed for WSS (out of resources?)\n"); return -EBUSY; } @@ -311,19 +291,9 @@ static int __devinit snd_cs423x_pnp_init } /* CTRL initialization */ -static int __devinit snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) -{ - int err; - - pnp_init_resource_table(cfg); - if (cport[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], cport[dev], 8); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR IDENT " CTRL PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { +static int __devinit snd_cs423x_pnp_init_ctrl(int dev, struct pnp_dev *pdev) +{ + if (pnp_activate_dev(pdev) < 0) { printk(KERN_ERR IDENT " CTRL PnP configure failed for WSS (out of resources?)\n"); return -EBUSY; } @@ -333,21 +303,9 @@ static int __devinit snd_cs423x_pnp_init } /* MPU initialization */ -static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) -{ - int err; - - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] >= 0) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR IDENT " MPU401 PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { +static int __devinit snd_cs423x_pnp_init_mpu(int dev, struct pnp_dev *pdev) +{ + if (pnp_activate_dev(pdev) < 0) { printk(KERN_ERR IDENT " MPU401 PnP configure failed for WSS (out of resources?)\n"); mpu_port[dev] = SNDRV_AUTO_PORT; mpu_irq[dev] = SNDRV_AUTO_IRQ; @@ -368,15 +326,8 @@ static int __devinit snd_card_cs4232_pnp static int __devinit snd_card_cs4232_pnp(int dev, struct snd_card_cs4236 *acard, struct pnp_dev *pdev) { - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; - if (snd_cs423x_pnp_init_wss(dev, acard->wss, cfg) < 0) { - kfree(cfg); + if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) return -EBUSY; - } - kfree(cfg); cport[dev] = -1; return 0; } @@ -386,43 +337,33 @@ static int __devinit snd_card_cs423x_pnp struct pnp_card_link *card, const struct pnp_card_device_id *id) { - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; - acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL); if (acard->wss == NULL) - goto error; + return -EBUSY; acard->ctrl = pnp_request_card_device(card, id->devs[1].id, NULL); if (acard->ctrl == NULL) - goto error; + return -EBUSY; if (id->devs[2].id[0]) { acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); if (acard->mpu == NULL) - goto error; + return -EBUSY; } /* WSS initialization */ - if (snd_cs423x_pnp_init_wss(dev, acard->wss, cfg) < 0) - goto error; + if (snd_cs423x_pnp_init_wss(dev, acard->wss) < 0) + return -EBUSY; /* CTRL initialization */ if (acard->ctrl && cport[dev] > 0) { - if (snd_cs423x_pnp_init_ctrl(dev, acard->ctrl, cfg) < 0) - goto error; + if (snd_cs423x_pnp_init_ctrl(dev, acard->ctrl) < 0) + return -EBUSY; } /* MPU initialization */ if (acard->mpu && mpu_port[dev] > 0) { - if (snd_cs423x_pnp_init_mpu(dev, acard->mpu, cfg) < 0) - goto error; - } - kfree(cfg); - return 0; - - error: - kfree(cfg); - return -EBUSY; + if (snd_cs423x_pnp_init_mpu(dev, acard->mpu) < 0) + return -EBUSY; + } + return 0; } #endif /* CONFIG_PNP */ diff --git a/isa/dt019x.c b/isa/dt019x.c --- a/isa/dt019x.c +++ b/isa/dt019x.c @@ -56,18 +56,6 @@ MODULE_PARM_DESC(id, "ID string for DT-0 MODULE_PARM_DESC(id, "ID string for DT-019X based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable DT-019X based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for dt019x driver."); -module_param_array(mpu_port, long, NULL, 0444); -MODULE_PARM_DESC(mpu_port, "MPU-401 port # for dt019x driver."); -module_param_array(fm_port, long, NULL, 0444); -MODULE_PARM_DESC(fm_port, "FM port # for dt019x driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for dt019x driver."); -module_param_array(mpu_irq, int, NULL, 0444); -MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for dt019x driver."); -module_param_array(dma8, int, NULL, 0444); -MODULE_PARM_DESC(dma8, "8-bit DMA # for dt019x driver."); struct snd_card_dt019x { struct pnp_dev *dev; @@ -95,36 +83,20 @@ static int __devinit snd_card_dt019x_pnp const struct pnp_card_device_id *pid) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; - acard->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree (cfg); + if (acard->dev == NULL) return -ENODEV; - } + acard->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); acard->devopl = pnp_request_card_device(card, pid->devs[2].id, NULL); pdev = acard->dev; - pnp_init_resource_table(cfg); - - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "DT-019X AUDIO the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "DT-019X AUDIO pnp configure failure\n"); - kfree(cfg); return err; } @@ -135,15 +107,7 @@ static int __devinit snd_card_dt019x_pnp port[dev],irq[dev],dma8[dev]); pdev = acard->devmpu; - if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port[dev], 2); - if (mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "DT-019X MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { pnp_release_card_device(pdev); @@ -162,11 +126,6 @@ static int __devinit snd_card_dt019x_pnp pdev = acard->devopl; if (pdev != NULL) { - pnp_init_resource_table(cfg); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], fm_port[dev], 4); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "DT-019X OPL3 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { pnp_release_card_device(pdev); @@ -181,7 +140,6 @@ static int __devinit snd_card_dt019x_pnp fm_port[dev] = -1; } - kfree(cfg); return 0; } diff --git a/isa/es18xx.c b/isa/es18xx.c --- a/isa/es18xx.c +++ b/isa/es18xx.c @@ -2035,31 +2035,9 @@ MODULE_DEVICE_TABLE(pnp, snd_audiodrive_ MODULE_DEVICE_TABLE(pnp, snd_audiodrive_pnpbiosids); /* PnP main device initialization */ -static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev, - struct pnp_resource_table *cfg) -{ - int err; - - pnp_init_resource_table(cfg); - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], mpu_port[dev], 2); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_device_is_isapnp(pdev)) { - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_ERR PFX "PnP manual resources are invalid, using auto config\n"); - } - err = pnp_activate_dev(pdev); - if (err < 0) { +static int __devinit snd_audiodrive_pnp_init_main(int dev, struct pnp_dev *pdev) +{ + if (pnp_activate_dev(pdev) < 0) { snd_printk(KERN_ERR PFX "PnP configure failure (out of resources?)\n"); return -EBUSY; } @@ -2087,16 +2065,9 @@ static int __devinit snd_audiodrive_pnp( static int __devinit snd_audiodrive_pnp(int dev, struct snd_audiodrive *acard, struct pnp_dev *pdev) { - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; acard->dev = pdev; - if (snd_audiodrive_pnp_init_main(dev, acard->dev, cfg) < 0) { - kfree(cfg); + if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) return -EBUSY; - } - kfree(cfg); return 0; } @@ -2125,33 +2096,24 @@ static int __devinit snd_audiodrive_pnpc struct pnp_card_link *card, const struct pnp_card_device_id *id) { - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); - - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -EBUSY; - } + acard->devc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (acard->devc == NULL) { - kfree(cfg); + if (acard->devc == NULL) return -EBUSY; - } + /* Control port initialization */ if (pnp_activate_dev(acard->devc) < 0) { - kfree(cfg); snd_printk(KERN_ERR PFX "PnP control configure failure (out of resources?)\n"); return -EAGAIN; } snd_printdd("pnp: port=0x%llx\n", (unsigned long long)pnp_port_start(acard->devc, 0)); - if (snd_audiodrive_pnp_init_main(dev, acard->dev, cfg) < 0) { - kfree(cfg); + if (snd_audiodrive_pnp_init_main(dev, acard->dev) < 0) return -EBUSY; - } - kfree(cfg); + return 0; } #endif /* CONFIG_PNP */ diff --git a/isa/gus/interwave.c b/isa/gus/interwave.c --- a/isa/gus/interwave.c +++ b/isa/gus/interwave.c @@ -560,50 +560,27 @@ static int __devinit snd_interwave_pnp(i const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; iwcard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (iwcard->dev == NULL) { - kfree(cfg); + if (iwcard->dev == NULL) return -EBUSY; - } + #ifdef SNDRV_STB iwcard->devtc = pnp_request_card_device(card, id->devs[1].id, NULL); - if (iwcard->devtc == NULL) { - kfree(cfg); + if (iwcard->devtc == NULL) return -EBUSY; - } #endif /* Synth & Codec initialization */ pdev = iwcard->dev; - pnp_init_resource_table(cfg); - if (port[dev] != SNDRV_AUTO_PORT) { - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - pnp_resource_change(&cfg->port_resource[1], port[dev] + 0x100, 12); - pnp_resource_change(&cfg->port_resource[2], port[dev] + 0x10c, 4); - } - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (dma2[dev] < 0) - pnp_resource_change(&cfg->dma_resource[1], 4, 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "InterWave - Synth - the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { - kfree(cfg); snd_printk(KERN_ERR "InterWave PnP configure failure (out of resources?)\n"); return err; } if (pnp_port_start(pdev, 0) + 0x100 != pnp_port_start(pdev, 1) || pnp_port_start(pdev, 0) + 0x10c != pnp_port_start(pdev, 2)) { - kfree(cfg); snd_printk(KERN_ERR "PnP configure failure (wrong ports)\n"); return -ENOENT; } @@ -620,21 +597,15 @@ static int __devinit snd_interwave_pnp(i #ifdef SNDRV_STB /* Tone Control initialization */ pdev = iwcard->devtc; - pnp_init_resource_table(cfg); - if (port_tc[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port_tc[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "InterWave - ToneControl - the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { - kfree(cfg); snd_printk(KERN_ERR "InterWave ToneControl PnP configure failure (out of resources?)\n"); return err; } port_tc[dev] = pnp_port_start(pdev, 0); snd_printdd("isapnp IW: tone control port=0x%lx\n", port_tc[dev]); #endif - kfree(cfg); return 0; } #endif /* CONFIG_PNP */ diff --git a/isa/opl3sa2.c b/isa/opl3sa2.c --- a/isa/opl3sa2.c +++ b/isa/opl3sa2.c @@ -610,39 +610,8 @@ static int __devinit snd_opl3sa2_pnp(int static int __devinit snd_opl3sa2_pnp(int dev, struct snd_opl3sa2 *chip, struct pnp_dev *pdev) { - struct pnp_resource_table * cfg; - int err; - - cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); - if (!cfg) { - snd_printk(KERN_ERR PFX "cannot allocate pnp cfg\n"); - return -ENOMEM; - } - /* PnP initialization */ - pnp_init_resource_table(cfg); - if (sb_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], sb_port[dev], 16); - if (wss_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], wss_port[dev], 8); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], fm_port[dev], 4); - if (midi_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[3], midi_port[dev], 2); - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[4], port[dev], 2); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - err = pnp_manual_config_dev(pdev, cfg, 0); - if (err < 0) - snd_printk(KERN_WARNING "PnP manual resources are invalid, using auto config\n"); - err = pnp_activate_dev(pdev); - if (err < 0) { - kfree(cfg); - snd_printk(KERN_ERR "PnP configure failure (out of resources?) err = %d\n", err); + if (pnp_activate_dev(pdev) < 0) { + snd_printk(KERN_ERR "PnP configure failure (out of resources?)\n"); return -EBUSY; } sb_port[dev] = pnp_port_start(pdev, 0); @@ -657,7 +626,6 @@ static int __devinit snd_opl3sa2_pnp(int pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", sb_port[dev], wss_port[dev], fm_port[dev], midi_port[dev]); snd_printdd("%sPnP OPL3-SA: control port=0x%lx, dma1=%i, dma2=%i, irq=%i\n", pnp_device_is_pnpbios(pdev) ? "BIOS" : "ISA", port[dev], dma1[dev], dma2[dev], irq[dev]); - kfree(cfg); return 0; } #endif /* CONFIG_PNP */ diff --git a/isa/opti9xx/opti92x-ad1848.c b/isa/opti9xx/opti92x-ad1848.c --- a/isa/opti9xx/opti92x-ad1848.c +++ b/isa/opti9xx/opti92x-ad1848.c @@ -1690,53 +1690,19 @@ static int __devinit snd_card_opti9xx_pn const struct pnp_card_device_id *pid) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; chip->dev = pnp_request_card_device(card, pid->devs[0].id, NULL); - if (chip->dev == NULL) { - kfree(cfg); + if (chip->dev == NULL) return -EBUSY; - } + chip->devmpu = pnp_request_card_device(card, pid->devs[1].id, NULL); pdev = chip->dev; - pnp_init_resource_table(cfg); - -#ifdef OPTi93X - if (port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port + 4, 4); -#else - if (pid->driver_data != 0x0924 && port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], port, 4); -#endif /* OPTi93X */ - if (irq != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq, 1); - if (dma1 != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1, 1); -#if defined(CS4231) || defined(OPTi93X) - if (dma2 != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2, 1); -#else -#ifdef snd_opti9xx_fixup_dma2 - snd_opti9xx_fixup_dma2(pdev); -#endif -#endif /* CS4231 || OPTi93X */ -#ifdef OPTi93X - if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port, 4); -#else - if (fm_port > 0 && fm_port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], fm_port, 4); -#endif - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "AUDIO the requested resources are invalid, using auto config\n"); + err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure: %d\n", err); - kfree(cfg); return err; } @@ -1756,15 +1722,6 @@ static int __devinit snd_card_opti9xx_pn pdev = chip->devmpu; if (pdev && mpu_port > 0) { - pnp_init_resource_table(cfg); - - if (mpu_port != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], mpu_port, 2); - if (mpu_irq != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], mpu_irq, 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "AUDIO pnp configure failure\n"); @@ -1775,7 +1732,6 @@ static int __devinit snd_card_opti9xx_pn mpu_irq = pnp_irq(pdev, 0); } } - kfree(cfg); return pid->driver_data; } #endif /* CONFIG_PNP */ diff --git a/isa/sb/es968.c b/isa/sb/es968.c --- a/isa/sb/es968.c +++ b/isa/sb/es968.c @@ -49,12 +49,6 @@ MODULE_PARM_DESC(id, "ID string for es96 MODULE_PARM_DESC(id, "ID string for es968 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable es968 based soundcard."); -module_param_array(port, long, NULL, 0444); -MODULE_PARM_DESC(port, "Port # for es968 driver."); -module_param_array(irq, int, NULL, 0444); -MODULE_PARM_DESC(irq, "IRQ # for es968 driver."); -module_param_array(dma8, int, NULL, 0444); -MODULE_PARM_DESC(dma8, "8-bit DMA # for es968 driver."); struct snd_card_es968 { struct pnp_dev *dev; @@ -86,40 +80,23 @@ static int __devinit snd_card_es968_pnp( const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; + acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); dma8[dev] = pnp_dma(pdev, 1); irq[dev] = pnp_irq(pdev, 0); - kfree(cfg); return 0; } diff --git a/isa/sb/sb16.c b/isa/sb/sb16.c --- a/isa/sb/sb16.c +++ b/isa/sb/sb16.c @@ -257,44 +257,21 @@ static int __devinit snd_card_sb16_pnp(i const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table * cfg = kmalloc(sizeof(struct pnp_resource_table), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; acard->dev = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->dev == NULL) { - kfree(cfg); + if (acard->dev == NULL) return -ENODEV; - } + #ifdef SNDRV_SBAWE_EMU8000 acard->devwt = pnp_request_card_device(card, id->devs[1].id, acard->dev); #endif /* Audio initialization */ pdev = acard->dev; - pnp_init_resource_table(cfg); - - /* override resources */ - - if (port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], port[dev], 16); - if (mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], mpu_port[dev], 2); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[2], fm_port[dev], 4); - if (dma8[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma8[dev], 1); - if (dma16[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma16[dev], 1); - if (irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], irq[dev], 1); - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR PFX "AUDIO the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR PFX "AUDIO pnp configure failure\n"); - kfree(cfg); return err; } port[dev] = pnp_port_start(pdev, 0); @@ -311,17 +288,6 @@ static int __devinit snd_card_sb16_pnp(i /* WaveTable initialization */ pdev = acard->devwt; if (pdev != NULL) { - pnp_init_resource_table(cfg); - - /* override resources */ - - if (awe_port[dev] != SNDRV_AUTO_PORT) { - pnp_resource_change(&cfg->port_resource[0], awe_port[dev], 4); - pnp_resource_change(&cfg->port_resource[1], awe_port[dev] + 0x400, 4); - pnp_resource_change(&cfg->port_resource[2], awe_port[dev] + 0x800, 4); - } - if ((pnp_manual_config_dev(pdev, cfg, 0)) < 0) - snd_printk(KERN_ERR PFX "WaveTable the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { goto __wt_error; @@ -339,7 +305,6 @@ __wt_error: awe_port[dev] = -1; } #endif - kfree(cfg); return 0; } diff --git a/isa/wavefront/wavefront.c b/isa/wavefront/wavefront.c --- a/isa/wavefront/wavefront.c +++ b/isa/wavefront/wavefront.c @@ -104,21 +104,15 @@ snd_wavefront_pnp (int dev, snd_wavefron const struct pnp_card_device_id *id) { struct pnp_dev *pdev; - struct pnp_resource_table *cfg = kmalloc(sizeof(*cfg), GFP_KERNEL); int err; - if (!cfg) - return -ENOMEM; - /* Check for each logical device. */ /* CS4232 chip (aka "windows sound system") is logical device 0 */ acard->wss = pnp_request_card_device(card, id->devs[0].id, NULL); - if (acard->wss == NULL) { - kfree(cfg); + if (acard->wss == NULL) return -EBUSY; - } /* there is a game port at logical device 1, but we ignore it completely */ @@ -133,25 +127,19 @@ snd_wavefront_pnp (int dev, snd_wavefron if (use_cs4232_midi[dev]) { acard->mpu = pnp_request_card_device(card, id->devs[2].id, NULL); - if (acard->mpu == NULL) { - kfree(cfg); + if (acard->mpu == NULL) return -EBUSY; - } } /* The ICS2115 synth is logical device 4 */ acard->synth = pnp_request_card_device(card, id->devs[3].id, NULL); - if (acard->synth == NULL) { - kfree(cfg); + if (acard->synth == NULL) return -EBUSY; - } /* PCM/FM initialization */ pdev = acard->wss; - - pnp_init_resource_table(cfg); /* An interesting note from the Tropez+ FAQ: @@ -165,23 +153,9 @@ snd_wavefront_pnp (int dev, snd_wavefron */ - if (cs4232_pcm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], cs4232_pcm_port[dev], 4); - if (fm_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[1], fm_port[dev], 4); - if (dma1[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[0], dma1[dev], 1); - if (dma2[dev] != SNDRV_AUTO_DMA) - pnp_resource_change(&cfg->dma_resource[1], dma2[dev], 1); - if (cs4232_pcm_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->irq_resource[0], cs4232_pcm_irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "PnP WSS the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "PnP WSS pnp configure failure\n"); - kfree(cfg); return err; } @@ -195,22 +169,9 @@ snd_wavefront_pnp (int dev, snd_wavefron pdev = acard->synth; - pnp_init_resource_table(cfg); - - if (ics2115_port[dev] != SNDRV_AUTO_PORT) { - pnp_resource_change(&cfg->port_resource[0], ics2115_port[dev], 16); - } - - if (ics2115_port[dev] != SNDRV_AUTO_IRQ) { - pnp_resource_change(&cfg->irq_resource[0], ics2115_irq[dev], 1); - } - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "PnP ICS2115 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "PnP ICS2115 pnp configure failure\n"); - kfree(cfg); return err; } @@ -226,15 +187,6 @@ snd_wavefront_pnp (int dev, snd_wavefron pdev = acard->mpu; - pnp_init_resource_table(cfg); - - if (cs4232_mpu_port[dev] != SNDRV_AUTO_PORT) - pnp_resource_change(&cfg->port_resource[0], cs4232_mpu_port[dev], 2); - if (cs4232_mpu_irq[dev] != SNDRV_AUTO_IRQ) - pnp_resource_change(&cfg->port_resource[0], cs4232_mpu_irq[dev], 1); - - if (pnp_manual_config_dev(pdev, cfg, 0) < 0) - snd_printk(KERN_ERR "PnP MPU401 the requested resources are invalid, using auto config\n"); err = pnp_activate_dev(pdev); if (err < 0) { snd_printk(KERN_ERR "PnP MPU401 pnp configure failure\n"); @@ -258,7 +210,6 @@ snd_wavefront_pnp (int dev, snd_wavefron ics2115_port[dev], ics2115_irq[dev]); - kfree(cfg); return 0; }

6 19

Re: [alsa-devel] Edirol UA-4FX
by David Hubbard 30 Nov '07

30 Nov '07

Hello Luis, and Alsa list, I recently got an Edirol UA-4FX. It works just fine as a generic USB Audio device. I've written up all the details at http://alsa.opensrc.org/index.php/Edirol_UA-4FX (shamelessly stolen from the Edirol_UA-25 page). The the UA-4FX needs a patch to usbaudio.c create_ua700_ua25_quirk() and usbquirks.h, and then it has full support. (MIDI I/O, sample rates up to 96 kHz) I've made an attempt, mostly to get it working on my machine. Can you review it? Thanks, David

2 3

[alsa-devel] Export symbols in alsa-lib
by Takashi Iwai 30 Nov '07

30 Nov '07

Hi, during the discussions with LSB guys, it turned out that the current ALSA-lib has many exported symbols that are not actually considered to be public. I fixed these unneeded exports in the HG tree. If you find some problems with this change, please let me know. thanks, Takashi

1 0

[alsa-devel] ALSA API as standards
by Takashi Iwai 30 Nov '07

30 Nov '07

Hi, I've talked with some people workiong on LSB about the possibility to put libasound as a part of the next LSB (surprisingly, they would like it more than I). I proposed that we should define the minimal set at first as the standard instead of taking all existing API functions. An interesting input was that they did some statistics and counted the actual used functions/variables in ALSA API over distros. The data is attached below. I just put the number of apps and functions there to simplify. (It includes __* functions that are actaully internal ones, and I'm wondering why their numbers are so high. Anyway, they shouldn't be a part of standard.) So, first, just take a look at it. You'll see how many functions are not actually used at all. What I'd like to do now are: - create a list of "standard" functions to be passed to LSB - clean up and actually drop some non-used functions - remove some functions from export list (in Versions file) I'll try to create a list. Basically it'll be almost as same as the list of functions with count > 0, plus some related ones. Then, let's discuss and clean up... BTW, regarding Versions. The versioned symbols for newly added functions seem not set properly as expected. We have entries ALSA_0.9 { global: __snd_*; _snd_*; snd_*; local: *; }; and thus the rest functions after this all belong to ALSA_0.9 in the end (except for functions explicitly defined with obsolete macros in pcm.c). It means, we'll need to stick with ALSA_0.9 tag no matter which version is added. Only if the function API is changed and double definitions are needed, we'll tag to another version... Oh well. Takashi AppUsage Iname 36 snd_pcm_hw_params_set_format 36 snd_pcm_hw_params_set_access 36 snd_pcm_hw_params_any 36 snd_pcm_open 36 snd_pcm_hw_params 36 snd_pcm_prepare 35 snd_strerror 35 snd_pcm_close 35 snd_pcm_writei 34 snd_pcm_hw_params_set_channels 29 snd_pcm_sw_params_current 28 snd_pcm_sw_params 28 snd_pcm_sw_params_set_start_threshold 27 snd_pcm_readi 25 snd_pcm_drop 24 snd_pcm_avail_update 23 snd_pcm_state 23 snd_pcm_resume 22 snd_ctl_open 22 snd_ctl_close 22 snd_card_next 22 snd_pcm_sw_params_set_avail_min 22 snd_pcm_start 21 snd_ctl_pcm_next_device 21 snd_pcm_hw_params_malloc 20 snd_pcm_hw_params_free 20 snd_pcm_nonblock 19 snd_pcm_pause 19 snd_pcm_info_set_stream 19 snd_ctl_pcm_info 19 snd_pcm_info_get_name 19 snd_pcm_status 19 snd_pcm_info_set_device 18 snd_pcm_hw_params_set_rate_near 18 snd_pcm_sw_params_set_xfer_align 18 snd_pcm_info_set_subdevice 17 snd_ctl_card_info_get_name 17 snd_ctl_card_info 17 snd_pcm_hw_params_sizeof 16 snd_pcm_hw_params_get_format_mask 16 snd_pcm_status_get_avail 16 snd_pcm_hw_params_set_period_size_near 16 snd_pcm_format_physical_width 16 snd_lib_error_set_handler 16 snd_pcm_format_mask_test 15 snd_pcm_sw_params_free 15 snd_pcm_info_free 15 snd_ctl_card_info_get_id 15 snd_pcm_sw_params_malloc 15 snd_pcm_info_malloc 14 snd_pcm_format_mask_malloc 14 snd_pcm_sw_params_sizeof 14 snd_seq_set_client_name 14 snd_seq_open 14 snd_pcm_format_mask_free 13 snd_pcm_format_width 13 snd_pcm_format_big_endian 13 snd_pcm_format_linear 13 snd_pcm_build_linear_format 13 snd_ctl_card_info_free 13 snd_pcm_info_get_id 13 snd_seq_create_simple_port 13 snd_seq_close 13 snd_ctl_card_info_malloc 13 snd_pcm_status_free 13 snd_pcm_format_signed 13 snd_pcm_info_get_subdevices_count 13 snd_pcm_status_malloc 13 __snd_pcm_hw_params_set_tick_time_last 13 __snd_pcm_hw_params_get_period_time 13 __snd_pcm_hw_params_get_periods 13 __snd_pcm_hw_params_set_access_last 13 __snd_pcm_hw_params_set_buffer_size_near 13 __snd_pcm_hw_params_get_periods_max 13 __snd_pcm_hw_params_get_period_size_min 13 __snd_pcm_hw_params_set_format_last 13 __snd_pcm_hw_params_set_period_time_first 13 __snd_pcm_hw_params_set_period_time_last 13 __snd_pcm_hw_params_set_channels_last 13 __snd_pcm_hw_params_get_tick_time_min 13 __snd_pcm_hw_params_set_buffer_size_first 13 __snd_pcm_hw_params_get_period_time_max 13 __snd_pcm_hw_params_get_tick_time_max 13 __snd_pcm_hw_params_get_rate 13 __snd_pcm_hw_params_get_subformat 13 __snd_pcm_hw_params_get_tick_time 13 __snd_pcm_hw_params_get_rate_max 13 __snd_pcm_hw_params_get_period_size 13 __snd_pcm_hw_params_set_periods_first 13 __snd_pcm_hw_params_get_access 13 __snd_pcm_hw_params_set_rate_near 13 __snd_pcm_hw_params_set_buffer_size_last 13 __snd_pcm_hw_params_set_tick_time_first 13 __snd_pcm_hw_params_get_periods_min 13 __snd_pcm_hw_params_get_rate_min 13 __snd_pcm_hw_params_set_channels_first 13 __snd_pcm_hw_params_get_channels_max 13 __snd_pcm_hw_params_get_buffer_time_min 13 __snd_pcm_hw_params_get_period_size_max 13 __snd_pcm_hw_params_get_channels_min 13 __snd_pcm_hw_params_get_buffer_time_max 13 __snd_pcm_hw_params_get_buffer_size_max 13 __snd_pcm_hw_params_get_buffer_size 13 __snd_pcm_hw_params_get_buffer_time 13 __snd_pcm_hw_params_get_period_time_min 13 __snd_pcm_hw_params_set_format_first 13 __snd_pcm_hw_params_set_period_size_first 13 __snd_pcm_hw_params_get_channels 13 __snd_pcm_hw_params_set_rate_last 13 __snd_pcm_hw_params_get_format 13 __snd_pcm_hw_params_set_buffer_time_near 13 __snd_pcm_hw_params_get_buffer_size_min 13 __snd_pcm_hw_params_set_channels_near 13 __snd_pcm_hw_params_set_subformat_last 13 __snd_pcm_hw_params_set_subformat_first 13 __snd_pcm_hw_params_set_rate_first 13 __snd_pcm_hw_params_set_period_size_last 13 __snd_pcm_hw_params_set_period_size_near 13 __snd_pcm_hw_params_set_period_time_near 13 __snd_pcm_hw_params_set_tick_time_near 13 __snd_pcm_hw_params_set_periods_near 13 __snd_pcm_hw_params_set_buffer_time_last 13 __snd_pcm_hw_params_set_periods_last 13 __snd_pcm_hw_params_set_buffer_time_first 11 snd_seq_event_input 11 snd_pcm_drain 11 snd_pcm_hw_params_get_buffer_size 10 snd_mixer_selem_get_playback_volume 10 snd_mixer_selem_register 10 snd_mixer_attach 10 snd_mixer_selem_get_playback_volume_range 10 snd_pcm_hw_params_set_periods_near 10 snd_pcm_hw_params_get_period_size 10 snd_mixer_open 10 snd_mixer_load 10 snd_mixer_close 9 snd_mixer_elem_next 9 snd_mixer_selem_has_playback_volume 9 snd_pcm_delay 9 snd_rawmidi_read 9 snd_seq_client_id 9 snd_rawmidi_close 9 snd_mixer_selem_has_playback_switch 9 snd_midi_event_new 9 snd_mixer_first_elem 9 snd_rawmidi_open 8 snd_pcm_hw_params_get_rate 8 snd_seq_poll_descriptors 8 snd_mixer_selem_set_playback_switch_all 8 snd_mixer_selem_get_name 8 snd_mixer_selem_get_capture_volume 8 snd_rawmidi_write 8 snd_mixer_selem_set_playback_volume 8 snd_seq_port_info_sizeof 8 snd_mixer_selem_has_capture_volume 8 snd_mixer_selem_has_capture_switch 8 snd_midi_event_free 8 snd_mixer_selem_set_capture_switch_all 8 snd_rawmidi_nonblock 7 snd_rawmidi_info_free 7 snd_seq_client_info_sizeof 7 snd_pcm_sw_params_set_silence_size 7 snd_rawmidi_info_malloc 7 snd_ctl_rawmidi_info 7 snd_rawmidi_info_get_name 7 snd_midi_event_encode_byte 7 snd_ctl_rawmidi_next_device 7 snd_rawmidi_info_set_device 7 snd_seq_port_info_set_port 7 snd_rawmidi_info_set_stream 7 snd_pcm_hw_params_set_buffer_size_near 7 snd_rawmidi_info_set_subdevice 7 snd_rawmidi_info_get_id 7 snd_rawmidi_info_get_subdevices_count 7 snd_mixer_selem_get_capture_volume_range 7 snd_mixer_selem_is_playback_mono 7 snd_pcm_sw_params_set_stop_threshold 7 snd_pcm_hw_params_can_pause 6 snd_ctl_card_info_get_mixername 6 snd_seq_free_event 6 snd_pcm_hw_params_set_buffer_time_near 6 snd_pcm_hw_params_get_channels_max 6 snd_seq_poll_descriptors_count 6 snd_mixer_selem_get_capture_switch 6 snd_pcm_sw_params_set_silence_threshold 6 snd_mixer_selem_has_playback_channel 6 snd_pcm_status_sizeof 6 snd_pcm_hw_params_set_period_time_near 6 snd_mixer_selem_set_capture_volume 6 snd_config_search 6 snd_mixer_selem_channel_name 6 snd_seq_query_next_port 6 snd_pcm_info_sizeof 6 snd_pcm_hw_params_get_channels_min 6 snd_pcm_hw_params_set_rate 6 snd_mixer_selem_get_playback_switch 6 snd_mixer_selem_is_capture_mono 6 snd_seq_port_info_set_client 6 snd_mixer_selem_is_active 6 snd_mixer_selem_has_capture_channel 6 snd_pcm_hw_params_test_format 6 snd_config_update 5 snd_seq_port_subscribe_set_sender 5 snd_seq_port_subscribe_set_dest 5 snd_config_iterator_end 5 snd_config_iterator_next 5 snd_seq_port_info_get_name 5 snd_seq_port_info_get_capability 5 snd_config_iterator_first 5 snd_seq_connect_to 5 snd_pcm_hw_params_get_channels 5 snd_pcm_hw_params_get_buffer_size_min 5 snd_seq_subscribe_port 5 snd_pcm_hw_params_get_period_size_min 5 snd_seq_port_info_get_addr 5 snd_seq_event_input_pending 5 snd_seq_query_next_client 5 snd_seq_event_output 5 snd_config_iterator_entry 5 snd_seq_client_info_get_client 5 snd_seq_drain_output 5 snd_seq_client_info_set_client 5 snd_config_get_string 5 snd_seq_port_info_get_port 4 snd_pcm_hw_params_get_buffer_size_max 4 snd_pcm_hw_params_get_rate_min 4 snd_pcm_bytes_to_frames 4 snd_ctl_card_info_sizeof 4 snd_pcm_hw_params_get_rate_max 4 snd_pcm_hw_params_can_resume 4 snd_card_get_name 4 snd_pcm_hw_params_get_period_size_max 4 snd_pcm_frames_to_bytes 4 snd_pcm_status_get_trigger_tstamp 4 snd_pcm_wait 3 snd_pcm_poll_descriptors 3 snd_pcm_hw_params_is_double 3 snd_pcm_hw_params_is_batch 3 snd_pcm_hw_params_get_tick_time 3 snd_pcm_hw_params_can_sync_start 3 snd_pcm_status_get_state 3 snd_seq_port_info_set_name 3 snd_pcm_hw_params_get_periods_max 3 snd_pcm_hw_params_get_period_time 3 snd_async_add_pcm_handler 3 snd_seq_port_subscribe_sizeof 3 snd_pcm_mmap_commit 3 snd_pcm_hw_params_test_rate 3 snd_pcm_hw_params_get_access 3 snd_pcm_hw_params_can_overrange 3 snd_pcm_writen 3 snd_pcm_hw_params_get_format 3 snd_pcm_hw_params_is_half_duplex 3 snd_seq_event_output_direct 3 snd_pcm_hw_params_is_joint_duplex 3 snd_seq_connect_from 3 snd_pcm_hw_params_get_buffer_time 3 snd_config_get_id 3 snd_seq_client_info_get_name 3 snd_pcm_mmap_begin 3 snd_seq_port_info_set_capability 3 snd_pcm_hw_params_can_mmap_sample_resolution 3 snd_pcm_poll_descriptors_count 3 snd_pcm_link 3 snd_config 3 snd_pcm_hw_params_get_periods 3 snd_pcm_hw_params_set_channels_near 3 snd_pcm_hw_params_test_access 3 snd_pcm_hw_params_is_block_transfer 3 snd_pcm_hw_params_set_rate_resample 2 snd_seq_disconnect_from 2 snd_config_get_type 2 snd_mixer_selem_id_sizeof 2 snd_pcm_hw_params_get_period_time_max 2 snd_pcm_poll_descriptors_revents 2 snd_seq_delete_simple_port 2 snd_pcm_access_mask_test 2 snd_ctl_elem_value_clear 2 snd_config_get_integer 2 snd_pcm_access_name 2 snd_seq_port_info_get_client 2 snd_ctl_elem_value_set_integer 2 snd_mixer_selem_id_get_name 2 snd_pcm_mmap_readi 2 snd_ctl_elem_info_is_readable 2 snd_hctl_close 2 snd_ctl_elem_info_malloc 2 snd_pcm_mmap_writei 2 snd_ctl_name 2 snd_hwdep_close 2 snd_hwdep_open 2 snd_hwdep_info 2 snd_pcm_hw_params_get_access_mask 2 snd_pcm_format_set_silence 2 snd_hctl_load 2 snd_hctl_find_elem 2 snd_hctl_free 2 snd_pcm_sw_params_set_tstamp_mode 2 snd_pcm_hw_params_get_periods_min 2 snd_pcm_hw_params_test_channels 2 snd_pcm_sw_params_set_xrun_mode 2 snd_ctl_elem_value_get_integer 2 snd_pcm_hw_params_get_tick_time_max 2 snd_async_handler_get_callback_private 2 snd_hctl_open 2 snd_pcm_hw_params_get_tick_time_min 2 snd_pcm_readn 2 snd_seq_unsubscribe_port 2 snd_hctl_elem_info 2 snd_pcm_access_mask_sizeof 2 snd_seq_disconnect_to 2 snd_pcm_hw_params_dump 2 snd_pcm_hw_params_get_period_time_min 2 snd_pcm_rewind 2 snd_seq_client_info_get_type 2 snd_pcm_hw_params_copy 2 snd_seq_create_port 2 snd_pcm_type 2 snd_hwdep_info_sizeof 2 snd_hctl_elem_write 2 snd_ctl_elem_value_free 2 snd_ctl_elem_info_get_count 2 snd_pcm_hw_params_set_period_size 2 snd_seq_port_info_get_type 2 snd_ctl_elem_id_set_name 2 snd_ctl_elem_id_clear 2 snd_ctl_elem_info_free 2 snd_pcm_hwsync 2 snd_pcm_info 2 snd_seq_port_info_set_type 2 snd_mixer_selem_set_playback_volume_all 2 snd_ctl_elem_info_clear 2 snd_pcm_state_name 2 snd_pcm_sw_params_get_boundary 2 snd_hctl_elem_read 2 snd_ctl_elem_id_malloc 2 snd_ctl_elem_id_set_interface 2 snd_pcm_format_mask_sizeof 2 snd_ctl_elem_id_free 2 snd_rawmidi_poll_descriptors_count 2 snd_rawmidi_poll_descriptors 2 snd_pcm_hw_params_get_buffer_time_max 2 snd_midi_event_decode 2 snd_pcm_format_cpu_endian 2 snd_ctl_elem_value_malloc 2 snd_mixer_selem_set_capture_volume_all 2 snd_ctl_elem_info_get_step 2 snd_seq_port_subscribe_free 2 snd_output_stdio_attach 2 snd_ctl_elem_info_get_min 2 snd_seq_port_subscribe_malloc 2 snd_mixer_selem_id_get_index 2 snd_pcm_status_get_tstamp 2 snd_pcm_hw_params_get_buffer_time_min 2 snd_pcm_format_name 2 snd_ctl_elem_info_get_max 1 snd_seq_port_subscribe_copy 1 snd_pcm_sw_params_set_sleep_min 1 snd_timer_info_get_resolution 1 snd_ctl_hwdep_next_device 1 snd_midi_event_init 1 snd_mixer_selem_get_id 1 snd_pcm_forward 1 snd_pcm_hw_params_test_period_size 1 snd_mixer_find_selem 1 snd_config_add 1 snd_pcm_areas_silence 1 snd_timer_poll_descriptors_count 1 snd_pcm_hw_free 1 snd_timer_params_set_ticks 1 snd_seq_poll_descriptors_revents 1 snd_seq_port_info_free 1 snd_pcm_hw_params_get_sbits 1 snd_seq_query_subscribe_set_root 1 snd_pcm_format_description 1 snd_pcm_status_dump 1 snd_pcm_sw_params_dump 1 snd_seq_get_client_info 1 snd_seq_query_subscribe_get_root 1 snd_timer_params_set_auto_start 1 snd_timer_params 1 snd_timer_open 1 snd_mixer_selem_id_set_name 1 snd_pcm_area_copy 1 snd_rawmidi_poll_descriptors_revents 1 snd_timer_read 1 snd_timer_id_malloc 1 snd_seq_port_subscribe_get_exclusive 1 snd_mixer_selem_has_common_volume 1 snd_timer_params_get_ticks 1 snd_seq_alloc_queue 1 snd_asoundlib_version 1 snd_seq_port_info_set_port_specified 1 snd_timer_close 1 snd_mixer_detach 1 snd_timer_info_malloc 1 snd_seq_client_info_set_name 1 snd_seq_port_subscribe_get_sender 1 snd_seq_query_subscribe_get_addr 1 snd_seq_query_subscribe_get_queue 1 snd_seq_port_subscribe_get_time_real 1 snd_seq_port_subscribe_set_exclusive 1 snd_config_substitute 1 snd_seq_query_subscribe_get_time_real 1 snd_timer_stop 1 snd_timer_info 1 snd_seq_query_subscribe_get_exclusive 1 snd_output_close 1 snd_seq_port_subscribe_get_time_update 1 snd_seq_port_subscribe_set_time_update 1 snd_seq_port_subscribe_get_dest 1 snd_timer_params_malloc 1 snd_seq_port_subscribe_get_queue 1 snd_config_copy 1 snd_seq_query_subscribe_get_index 1 snd_seq_set_client_info 1 snd_seq_port_subscribe_set_queue 1 snd_seq_port_info_malloc 1 snd_config_get_ascii 1 snd_seq_queue_tempo_set_ppq 1 snd_seq_query_subscribe_set_type 1 snd_pcm_hw_params_get_rate_numden 1 snd_midi_event_reset_encode 1 snd_seq_queue_tempo_sizeof 1 snd_pcm_open_lconf 1 snd_ctl_hwdep_info 1 snd_pcm_hw_params_get_subformat 1 snd_pcm_hw_params_set_periods 1 snd_seq_query_subscribe_sizeof 1 snd_mixer_elem_get_type 1 snd_seq_query_subscribe_get_time_update 1 snd_seq_query_subscribe_set_index 1 snd_config_search_alias_hooks 1 snd_card_get_longname 1 snd_config_imake_integer 1 snd_seq_query_port_subscribers 1 snd_seq_queue_tempo_set_tempo 1 snd_config_update_free_global 1 snd_hwdep_info_set_device 1 snd_seq_get_any_client_info 1 snd_pcm_dump 1 snd_seq_set_port_info 1 snd_seq_get_port_info 1 snd_mixer_selem_has_playback_switch_joined 1 snd_hwdep_poll_descriptors 1 snd_pcm_format_size 1 snd_seq_delete_port 1 snd_seq_control_queue 1 snd_seq_event_length 1 snd_config_delete 1 snd_mixer_selem_id_set_index 1 snd_pcm_hw_params_set_periods_min 1 snd_seq_port_info_set_midi_channels 1 snd_mixer_selem_has_playback_volume_joined 1 snd_seq_free_queue 1 snd_hwdep_ioctl 1 snd_pcm_hw_params_set_periods_integer 1 snd_seq_parse_address 1 snd_timer_poll_descriptors 1 snd_async_del_handler 1 snd_pcm_status_get_avail_max 1 snd_seq_get_any_port_info 1 snd_hwdep_info_get_iface 1 snd_mixer_selem_set_playback_switch 1 snd_seq_port_subscribe_set_time_real 1 snd_timer_start 1 snd_hwdep_info_get_name 1 snd_midi_event_encode 1 snd_pcm_status_get_delay 1 snd_hwdep_read 1 snd_seq_set_queue_tempo 0 snd_instr_header_get_id 0 snd_pcm_unlink 0 snd_pcm_hw_params_set_access_first 0 snd_ctl_rawmidi_prefer_subdevice 0 snd_pcm_hw_params_test_buffer_size 0 snd_pcm_info_get_subdevices_avail 0 snd_pcm_sw_params_get_silence_size 0 snd_ctl_elem_list_malloc 0 snd_output_buffer_open 0 snd_pcm_samples_to_bytes 0 snd_pcm_null_open 0 snd_pcm_hook_get_private 0 snd_pcm_status_get_htstamp 0 snd_ctl_elem_value_get_bytes 0 snd_ctl_elem_list_set_offset 0 snd_ctl_elem_value_get_name 0 snd_timer_params_set_early_event 0 snd_pcm_meter_get_rate 0 snd_config_save 0 snd_pcm_meter_get_boundary 0 snd_pcm_meter_get_bufsize 0 snd_seq_remove_events_get_queue 0 snd_pcm_mulaw_open 0 snd_hwdep_info_get_id 0 snd_hctl_elem_prev 0 snd_pcm_info_get_sync 0 snd_config_search_definition 0 snd_pcm_multi_open 0 snd_pcm_meter_search_scope 0 snd_hwdep_dsp_status_get_dsp_loaded 0 snd_ctl_event_elem_get_numid 0 snd_hctl_poll_descriptors 0 snd_pcm_mmap_writen 0 snd_mixer_elem_empty 0 snd_pcm_info_get_device 0 snd_instr_header_get_alias 0 snd_pcm_hw_params_set_period_size_max 0 snd_pcm_hw_params_set_period_size_integer 0 snd_pcm_hw_params_set_period_size_first 0 snd_pcm_hw_params_set_buffer_time_min 0 snd_pcm_hw_params_get_fifo_size 0 snd_mixer_elem_get_callback_private 0 snd_ctl_elem_info_set_id 0 snd_pcm_hw_params_set_format_last 0 snd_seq_port_info_get_timestamp_real 0 snd_pcm_hw_params_set_period_time_min 0 snd_pcm_scope_set_name 0 snd_seq_set_client_pool 0 snd_pcm_hw_params_set_periods_last 0 snd_pcm_hw_params_set_periods_first 0 snd_pcm_hw_params_set_period_time_minmax 0 snd_pcm_hw_params_set_period_time_max 0 snd_pcm_hw_params_set_period_size_minmax 0 snd_pcm_hw_params_set_period_size_min 0 snd_pcm_hw_params_set_period_time_last 0 snd_pcm_hw_params_set_period_time_first 0 snd_pcm_hw_params_set_period_time 0 snd_config_get_integer64 0 snd_pcm_hw_params_set_access_last 0 snd_pcm_meter_get_now 0 snd_pcm_hw_params_set_buffer_size_last 0 snd_pcm_hw_open_fd 0 snd_pcm_hw_params_set_buffer_size 0 snd_seq_reset_pool_input 0 snd_ctl_card_info_get_card 0 snd_pcm_hw_params_get_subformat_mask 0 snd_ctl_elem_remove 0 snd_pcm_scope_s16_get_channel_buffer 0 snd_timer_query_next_device 0 snd_pcm_hw_params_set_channels_first 0 snd_pcm_hw_params_set_buffer_time_minmax 0 snd_pcm_hw_params_set_buffer_time_max 0 snd_pcm_scope_set_ops 0 snd_pcm_hw_params_set_channels_min 0 snd_pcm_hw_params_set_channels_last 0 snd_ctl_elem_info_set_interface 0 snd_pcm_hw_params_set_buffer_size_min 0 snd_pcm_hw_params_set_buffer_size_max 0 snd_hctl_elem_get_index 0 snd_pcm_hw_params_set_buffer_time_first 0 snd_pcm_hw_params_set_buffer_time 0 snd_pcm_hw_params_test_tick_time 0 snd_pcm_hw_params_test_subformat 0 snd_seq_get_input_buffer_size 0 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snd_pcm_direct_poll_revents 0 snd_pcm_direct_server_create 0 _snd_hwdep_hw_open 0 snd_interval_mulkdiv 0 snd_rawmidi_virtual_open 0 snd_pcm_hw_param_set_max 0 snd_pcm_hw_param_set_mask 0 _snd_func_private_card_driver_dlsym_config_evaluate_001 0 __snd_hwdep_hw_open_dlsym_hwdep_001 0 snd_pcm_direct_semaphore_create_or_connect 0 snd_pcm_direct_client_discard 0 snd_rawmidi_virtual_ops 0 _snd_timer_query_hw_open 0 __snd_pcm_multi_open_dlsym_pcm_001 0 __snd_pcm_hooks_open_dlsym_pcm_001 0 snd_pcm_direct_shm_create_or_connect 0 __snd_pcm_hook_ctl_elems_install_dlsym_pcm_001 0 snd_pcm_hw_param_set_last 0 snd_atomic_read_wait 0 snd_pcm_hw_param_set_near 0 snd_pcm_slave_conf 0 _snd_pcm_boundary 0 __snd_pcm_adpcm_open_dlsym_pcm_001 0 snd_pcm_direct_initialize_poll_fd 0 __snd_ctl_shm_open_dlsym_control_001 0 __snd_timer_params_set_filter 0 snd_pcm_open_slave 0 __snd_pcm_asym_open_dlsym_pcm_001 0 _snd_pcm_hw_param_set_max 0 __snd_pcm_alaw_open_dlsym_pcm_001 0 __snd_ctl_hw_open_dlsym_control_001 0 snd_interval_add 0 snd_pcm_hw_param_get_min 0 snd_interval_div 0 snd_hwdep_hw_open 0 snd_timer_query_hw_open 0 snd_pcm_adpcm_decode 0 __snd_pcm_sw_params_get_silence_threshold 0 snd_pcm_new 0 _snd_pcm_mmap_hw_ptr 0 __snd_pcm_copy_open_dlsym_pcm_001 0 snd_pcm_direct_async 0 __snd_pcm_hw_open_dlsym_pcm_001 0 __snd_timer_query_info 0 snd_pcm_plugin_fast_ops 0 snd_pcm_direct_client_connect 0 __snd_pcm_shm_open_dlsym_pcm_001 0 snd_pcm_plug_open 0 snd_pcm_areas_from_buf 0 _snd_pcm_hw_param_set_interval 0 snd_pcm_linear_get_index 0 snd_pcm_areas_from_bufs 0 snd_pcm_async 0 __snd_pcm_dmix_open_dlsym_pcm_001 0 snd_pcm_read_mmap 0 snd_pcm_channel_info_shm 0 _snd_rawmidi_virtual_open 0 _snd_ctl_poll_descriptor 0 __snd_timer_query_status 0 __snd_timer_query_params 0 _snd_ctl_hw_open 0 _snd_config_hook_load_dlsym_config_hook_001 0 snd_pcm_direct_mmap 0 snd_pcm_set_appl_ptr 0 __snd_timer_query_hw_open_dlsym_timer_query_001 0 snd_timer_hw_open 0 _snd_func_private_string_dlsym_config_evaluate_001 0 _snd_func_datadir_dlsym_config_evaluate_001 0 __snd_pcm_mulaw_open_dlsym_pcm_001 0 _snd_func_private_pcm_subdevice_dlsym_config_evaluate_001 0 _snd_func_pcm_id_dlsym_config_evaluate_001 0 __snd_pcm_null_open_dlsym_pcm_001 0 __snd_pcm_plug_open_dlsym_pcm_001 0 snd_shm_area_destructor 0 snd_pcm_linear_get32_index 0 snd_pcm_direct_info 0 __snd_pcm_sw_params_get_silence_size 0 __snd_pcm_sw_params_get_start_threshold 0 __snd_pcm_sw_params_get_sleep_min 0 __snd_pcm_share_open_dlsym_pcm_001 0 __snd_timer_params_get_exclusive 0 __snd_pcm_sw_params_get_avail_min 0 __snd_pcm_status_get_trigger_htstamp 0 __snd_timer_params_set_exclusive 0 __snd_timer_hw_open_dlsym_timer_001 0 snd_pcm_mulaw_encode 0 snd_pcm_alaw_decode 0 snd_pcm_hw_param_always_eq 0 snd_pcm_hw_refine 0 __snd_pcm_sw_params_get_stop_threshold 0 __snd_rawmidi_hw_open_dlsym_rawmidi_001 0 snd_pcm_conf_generic_id 0 snd_pcm_hw_param_never_eq 0 _snd_seq_hw_open 0 snd_pcm_hw_param_get_max 0 snd_pcm_hw_param_name 0 snd_pcm_mmap_appl_forward 0 _snd_pcm_hw_params_refine 0 _snd_pcm_hw_param_refine 0 snd_pcm_hw_param_get_mask 0 _snd_pcm_hw_param_set_minmax 0 snd_pcm_hw_param_set_integer 0 snd_pcm_hw_param_set 0 snd_pcm_set_hw_ptr 0 snd_pcm_hw_param_set_first 0 _snd_rawmidi_hw_open 0 _snd_pcm_plug_open 0 __snd_seq_hw_open_dlsym_seq_001 0 snd_pcm_direct_sw_params 0 _snd_pcm_poll_descriptor 0 __snd_pcm_ladspa_open_dlsym_pcm_001 0 _snd_func_getenv_dlsym_config_evaluate_001 0 __snd_pcm_hw_params_set_access_first 0 snd_pcm_hw_param_dump 0 __snd_pcm_meter_open_dlsym_pcm_001 0 _snd_func_refer_dlsym_config_evaluate_001 0 snd_pcm_read_areas 0 __snd_pcm_route_open_dlsym_pcm_001 0 __snd_pcm_iec958_open_dlsym_pcm_001 0 __snd_pcm_dshare_open_dlsym_pcm_001 0 snd_pcm_adpcm_encode 0 _snd_pcm_hw_params_any 0 _snd_pcm_hw_param_set_min 0 _snd_pcm_hw_param_set_empty 0 _snd_pcm_hw_param_set 0 _snd_pcm_hw_param_set_mask 0 snd_pcm_hw_param_empty 0 snd_pcm_plugin_init 0 _snd_func_igetenv_dlsym_config_evaluate_001 0 snd_is_local 0 _snd_config_hook_load_for_all_cards_dlsym_config_hook_001 0 snd_interval_refine_min 0 snd_pcm_munmap 0 snd_interval_refine_set 0 snd_pcm_hw_param_set_minmax 0 snd_rawmidi_hw_open 0 snd_receive_fd 0 snd_pcm_direct_parse_bindings 0 snd_ctl_hw_ops 0 snd_ctl_hw_open 0 __snd_pcm_rate_open_dlsym_pcm_001 0 __snd_ctl_elem_info_get_dimensions 0 _snd_ctl_shm_open 0 snd_interval_refine_max 0 snd_pcm_meter_open 0 __snd_rawmidi_virtual_open_dlsym_rawmidi_001 0 snd_pcm_hw_param_get_interval 0 snd_pcm_write_areas 0 snd_pcm_direct_hw_params 0 snd_ctl_new 0 _snd_func_card_id_dlsym_config_evaluate_001 0 _snd_func_card_driver_dlsym_config_evaluate_001 0 snd_pcm_link_hw_ptr 0 snd_pcm_unlink_hw_ptr 0 __snd_ctl_elem_info_get_dimension 0 __snd_pcm_sw_params_get_xfer_align 0 snd_pcm_linear_convert_index 0 snd_pcm_direct_hw_free 0 __snd_timer_params_get_filter 0 snd_interval_sub 0 snd_ctl_async 0 _snd_timer_hw_open 0 snd_pcm_linear_getput 0 snd_pcm_unlink_appl_ptr 0 snd_pcm_linear_put_index 0 snd_pcm_direct_channel_info 0 snd_seq_create_event 0 __snd_pcm_status_get_htstamp 0 snd_send_fd 0 snd_pcm_linear_put32_index 0 snd_interval_muldiv 0 snd_pcm_write_mmap 0 snd_pcm_linear_convert 0 snd_pcm_hw_fast_tstamp 0 snd_pcm_hw_refine_soft 0 snd_pcm_hw_param_refine_near 0 snd_pcm_mmap_hw_forward 0 snd_pcm_direct_munmap 0 snd_determine_driver 0 snd_interval_refine 0 snd_pcm_free 0 snd_pcm_link_appl_ptr 0 snd_pcm_hw_params_slave 0 snd_pcm_hw_param_set_min 0 snd_pcm_hw_refine_slave 0 snd_pcm_mmap_hw_backward 0 snd_interval_refine_first 0 snd_interval_mul 0 snd_pcm_hw_param_any 0 snd_pcm_direct_nonblock 0 snd_func_private_pcm_subdevice 0 snd_func_pcm_id 0 __snd_pcm_type_name 0 snd_pcm_mulaw_decode 0 snd_interval_print 0 __snd_pcm_linear_open_dlsym_pcm_001 0 snd_interval_muldivk 0 snd_pcm_alaw_encode 0 __snd_pcm_dsnoop_open_dlsym_pcm_001 0 snd_seq_hw_open 0 _snd_func_concat_dlsym_config_evaluate_001 0 _snd_pcm_meter_open 0 snd_seq_hw_ops 0 snd_ctl_shm_ops 0 snd_pcm_mmap 0 __snd_pcm_file_open_dlsym_pcm_001 0 snd_pcm_mmap_appl_backward 0 snd_ctl_shm_open 0 snd_timer_async 0 snd_pcm_lfloat_open 0 _snd_pcm_lfloat_open 0 snd_pcm_lfloat_convert_integer_float 0 snd_pcm_lfloat_convert_float_integer 0 snd_pcm_lfloat_put_s32_index 0 __snd_pcm_lfloat_open_dlsym_pcm_001 0 snd_pcm_lfloat_get_s32_index 0 snd_ctl_elem_add_integer64 0 snd_pcm_softvol_open 0 _snd_pcm_softvol_open 0 snd_ctl_elem_add_integer 0 snd_ctl_elem_add_iec958 0 snd_ctl_elem_info_is_user 0 snd_ctl_elem_add_boolean 0 __snd_pcm_softvol_open_dlsym_pcm_001 0 snd_timer_ginfo_get_name 0 snd_timer_ginfo_get_resolution 0 snd_pcm_ioplug_set_param_minmax 0 snd_timer_ginfo_get_resolution_max 0 snd_pcm_ioplug_reinit_status 0 snd_async_handler_get_timer 0 snd_async_add_timer_handler 0 snd_pcm_ioplug_set_param_list 0 snd_timer_ginfo_get_card 0 snd_pcm_ioplug_params_reset 0 snd_err_msg 0 snd_timer_ginfo_malloc 0 snd_timer_ginfo_get_tid 0 snd_timer_ginfo_copy 0 snd_timer_ginfo_get_clients 0 snd_pcm_hw_params_set_export_buffer 0 snd_pcm_ioplug_delete 0 snd_pcm_ioplug_mmap_areas 0 snd_timer_ginfo_get_id 0 snd_timer_ginfo_set_tid 0 snd_timer_ginfo_get_flags 0 snd_pcm_ioplug_create 0 snd_pcm_extplug_create 0 snd_pcm_extplug_params_reset 0 snd_timer_ginfo_sizeof 0 snd_pcm_extplug_delete 0 snd_pcm_extplug_set_param_minmax 0 snd_pcm_hw_params_get_rate_resample 0 snd_timer_ginfo_get_resolution_min 0 snd_pcm_extplug_set_slave_param_list 0 snd_names_list 0 snd_pcm_extplug_set_param_list 0 snd_names_list_free 0 snd_timer_ginfo_free 0 snd_pcm_hw_params_get_export_buffer 0 snd_pcm_parse_control_id 0 snd_pcm_extplug_set_slave_param_minmax 0 snd_pcm_generic_rewind 0 snd_pcm_generic_sw_params 0 snd_pcm_generic_start 0 snd_pcm_direct_set_timer_params 0 snd_dlobj_cache_add 0 snd_pcm_generic_link 0 snd_ext_parm_set_minmax 0 snd_pcm_generic_state 0 snd_pcm_generic_writei 0 snd_pcm_generic_mmap 0 snd_pcm_generic_async 0 snd_pcm_generic_info 0 snd_pcm_direct_clear_timer_queue 0 snd_pcm_generic_drain 0 snd_pcm_generic_drop 0 snd_pcm_generic_close 0 snd_pcm_generic_hw_free 0 snd_pcm_generic_nonblock 0 snd_pcm_generic_munmap 0 snd_pcm_generic_hw_refine 0 snd_pcm_generic_hw_params 0 snd_pcm_generic_forward 0 snd_pcm_generic_writen 0 snd_ext_parm_clear 0 snd_pcm_generic_readi 0 snd_ext_parm_mask_refine 0 snd_dlobj_cache_cleanup 0 snd_dlobj_cache_lookup 0 snd_pcm_generic_channel_info 0 snd_pcm_generic_poll_descriptors 0 snd_pcm_generic_hwsync 0 snd_pcm_generic_prepare 0 snd_pcm_generic_avail_update 0 snd_ext_parm_set_list 0 snd_pcm_generic_poll_descriptors_count 0 snd_pcm_generic_status 0 snd_pcm_generic_readn 0 snd_pcm_generic_resume 0 snd_pcm_generic_pause 0 snd_interval_list 0 snd_pcm_generic_delay 0 snd_ext_parm_interval_refine 0 snd_pcm_generic_reset 0 snd_pcm_generic_unlink 0 snd_pcm_generic_poll_revents 0 snd_pcm_generic_mmap_commit 0 snd_pcm_direct_timer_stop 0 snd_pcm_wait_nocheck 0 snd_mixer_sbasic_get_private 0 snd_mixer_class_get_mixer 0 snd_mixer_get_hctl 0 snd_mixer_class_set_private_free 0 snd_mixer_class_copy 0 snd_mixer_elem_free 0 snd_mixer_simple_basic_register 0 snd_mixer_class_get_private 0 snd_mixer_selem_set_playback_dB_all 0 snd_mixer_class_get_event 0 snd_mixer_selem_is_enum_capture 0 snd_mixer_sbasic_set_private 0 snd_mixer_class_free 0 snd_mixer_sbasic_set_private_free 0 snd_mixer_selem_set_playback_dB 0 snd_mixer_elem_get_private 0 snd_mixer_class_sizeof 0 snd_mixer_elem_new 0 snd_mixer_selem_get_capture_dB_range 0 snd_mixer_class_get_compare 0 snd_mixer_selem_get_playback_dB_range 0 snd_mixer_sbasic_info 0 snd_mixer_class_set_event 0 snd_mixer_selem_get_capture_dB 0 snd_mixer_class_set_compare 0 snd_mixer_simple_none_register 0 snd_mixer_selem_set_capture_dB_all 0 snd_mixer_selem_is_enum_playback 0 snd_ctl_ext_delete 0 snd_mixer_selem_set_capture_dB 0 snd_mixer_class_set_private 0 snd_mixer_detach_hctl 0 snd_mixer_attach_hctl 0 snd_mixer_class_malloc 0 snd_mixer_selem_get_playback_dB 0 snd_ctl_ext_create 0 snd_pcm_direct_resume 0 _snd_func_pcm_args_by_class_dlsym_config_evaluate_001 0 snd_func_pcm_args_by_class 0 snd_pcm_direct_open_secondary_client 0 snd_mixer_selem_compare 0 snd_pcm_recover 0 snd_pcm_set_params 0 snd_pcm_get_params 0 snd_pcm_direct_parse_open_conf 0 _snd_seq_open_lconf 0 snd_config_check_hop 0 snd_config_set_hop 0 snd_pcm_generic_link_fd 0 _snd_pcm_link_descriptors 0 snd_pcm_generic_link2 0 snd_pcm_rate_get_default_converter 0 snd_hctl_elem_tlv_read 0 snd_ctl_elem_tlv_write 0 snd_ctl_elem_tlv_command 0 snd_hctl_elem_tlv_write 0 snd_ctl_elem_info_is_tlv_commandable 0 snd_ctl_elem_info_is_tlv_writable 0 snd_ctl_elem_info_is_tlv_readable 0 snd_hctl_elem_tlv_command 0 snd_ctl_elem_tlv_read 0 _snd_pcm_rate_linear_open 0 snd_pcm_direct_initialize_secondary_slave

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[alsa-devel] Some drivers don't create all necessary sysfs symlinks
by Lennart Poettering 30 Nov '07

30 Nov '07

Hi! Apparently not all ALSA drivers set the "device" symlink properly. This creates problems with HAL which then is unable to manage permissions to those files properly. One example of such a driver is EMU10K1X. For more information, see: https://bugzilla.redhat.com/show_bug.cgi?id=370821#c26 Also reported on the ALSA BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3575 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

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[alsa-devel] Problems while using mplayer based on ALSA
by 30 Nov '07

30 Nov '07

Hi All: Im having problems while using mplayer 1.0 to play music based on alsa 1.0.15. Im using an ARM9 system and montavista linux 2.6.18. Here's what the problem is and what's modified in the mplayer.conf: *************************************************************************************************************** / $ mplayer /tmp/a.wav MPlayer 1.0rc2-4.2.0 (C) 2000-2007 MPlayer Team CPU: ARM 115 audio & 237 video codecs Playing /tmp/a.wav. Audio file file format detected. ========================================================================== Opening audio decoder: [pcm] Uncompressed PCM audio decoder AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== [AO_ALSA] alsa-lib: conf.c:2699:(snd_config_hooks_call) Cannot open shared library (null) [AO_ALSA] alsa-lib: conf.c:3076:(snd_config_update_r) hooks failed, removing configuration [AO_ALSA] Playback open error: No such file or directory Could not open/initialize audio device -> no sound. Audio: no sound Video: no video Exiting... (End of file) ***************************************************************************************************************** /etc/mplayer/mplayer.conf ################## # audio settings # ################## # Specify default audio driver (see -ao help for a list). ao=alsa # Use SDL audio driver with the esd subdriver by default. #ao = sdl:esd # Specify the mixer device. #mixer = /dev/mixer # Resample the sound to 44100Hz with the lavcresample audio filter. #af=lavcresample=44100 **************************************************************************************************************** Can anybody tell me what going wrong with this? Thanks in advance! semiyd _________________________________________________________________ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-us&source=…

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[alsa-devel] HDA or AC97 for Nvidia MCP61S?
by alsa-devel.20.pris＠spamgourmet.com 30 Nov '07

30 Nov '07

Hello, as I already reported on another email I'm having problems with my sound: http://thread.gmane.org/gmane.linux.alsa.devel/49412 My motherboard is PCCHIPS A13G+. The sound hardware of my Mobo is in the NVIDIA MCP61S Single chipset. This chipset consists of: GeForce(R) 6100 GPU and MCP nForce(R) 405 Codec: Realtek ALC660-VD When looking at the soundcard matrix for Nvidia there are 2 options listed: http://www.alsa-project.org/main/index.php/Matrix:Vendor-Nvidia AC97 nForce-1 - nForce-4 HDA nForce-430 and later Since my chipset is nForce 405 should I select the first option AC97? But everyone seems to advise me to use the HDA option. When I look at the details for HDA I see only "Intel HDA driver" and nothing about which particular chipsets are supported. On the AC97 option there is a list of chips but all are from Intel and none from Nvidia. So I'm confused. Which soundcard is the correct one for me? Is my particular chipset supported at all? Thanks!

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[alsa-devel] pcm test app failure on PowerPC
by John Faith 29 Nov '07

29 Nov '07

Hello, I am using alsa-lib 1.0.15 on an embedded PowerPC board. I can play audio with aplay, but gstreamer fails, so I built and ran the 'pcm' app in alsa-lib-1.0.15/test/. pcm gives: # export LIBASOUND_DEBUG=1 # ./pcm Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_BE, 1 channels Sine wave rate is 440.0000Hz Using transfer method: write ALSA lib pcm_plug.c:773:(snd_pcm_plug_hw_refine_schange) Unable to find an usable access for 'plughw:0,0' ALSA ERROR hw_params: set (FORMAT) value = S16_BE : Invalid argument ACCESS: RW_INTERLEAVED FORMAT: S8 U8 S16_LE S16_BE U16_LE U16_BE S24_LE S24_BE U24_LE U24_BE S32_LE S32_BE U32_LE U32_BE FLOAT_LE FLOAT_BE SUBFORMAT: STD SAMPLE_BITS: [4 64] FRAME_BITS: [4 640000] CHANNELS: [1 10000] RATE: [4000 4294967295) PERIOD_TIME: [25000 300000] PERIOD_SIZE: [100 1288490189) PERIOD_BYTES: [50 4294967295) PERIODS: (0 12884902) BUFFER_TIME: [1 4294967295] BUFFER_SIZE: [100 1288490188] BUFFER_BYTES: [50 4294967295] TICK_TIME: 4000 Sample format not available for playback: Invalid argument Setting of hwparams failed: Invalid argument Is the driver saying it can do format S16_BE somewhere, but fails when the format is actually set? How can I dig deeper on this? Thanks in advance for any ideas! , John

1 1

[alsa-devel] PCM Volume on FSC V5505
by Jan-Simon Möller 29 Nov '07

29 Nov '07

Hi ! With the patches, the sound works - well done! But I need to make sure the PCM volume is max @ 50% . If its over 50% sound is clipped and ugly (overload?). Here's cat /proc/asound/card0/codec#0 when in use (patched kernel with your patches for v5505) Codec: Conexant CX20549 (Venice) Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x1734110e Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x140]: 48000 96000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x10 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x28 0x28] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x92170110: [Fixed] Speaker at Int Front Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Power: 0x0 Connection: 2 0x19 0x17* Node 0x11 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0x28 0x28] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0121401f: [Jack] HP Out at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Power: 0x0 Connection: 2 0x19 0x17* Node 0x12 [Pin Complex] wcaps 0x40058d: Stereo Amp-Out Amp-Out caps: ofs=0x2b, nsteps=0x2b, stepsize=0x05, mute=1 Amp-Out vals: [0xab 0xab] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x01a19020: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x20: IN Power: 0x0 Connection: 2 0x19* 0x17 Node 0x13 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x214571f0: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Yellow Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a7012e: [Fixed] Mic at Int Riser Conn = Analog, Color = Unknown Pin-ctls: 0x24: IN Node 0x15 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400001f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x00: Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x07, nsteps=0x07, stepsize=0x0b, mute=1 Amp-Out vals: [0x06] Node 0x17 [Audio Mixer] wcaps 0x20050b: Stereo Amp-In Amp-In caps: ofs=0x14, nsteps=0x2b, stepsize=0x05, mute=1 Amp-In vals: [0x14 0x14] [0xab 0xab] [0xa0 0xa0] [0x80 0x80] [0x80 0x80] Power: 0x0 Connection: 5 0x19 0x14 0x12 0x11 0x15 Node 0x18 [Audio Output] wcaps 0x211: Stereo Digital PCM: rates [0x40]: 48000 bits [0x6]: 16 20 formats [0x5]: PCM AC3 Node 0x19 [Audio Output] wcaps 0xc11: Stereo PCM: rates [0x540]: 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power: 0x0 Node 0x1a [Audio Input] wcaps 0x100d0b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x17, stepsize=0x05, mute=1 Amp-In vals: [0x17 0x17] [0x00 0x00] [0x00 0x00] [0x00 0x00] [0x00 0x00] Power: 0x0 Connection: 5 0x17 0x14 0x12* 0x11 0x15 Node 0x1b [Vendor Defined Widget] wcaps 0xf00000: Mono Best regards Jan-Simon

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Re: [alsa-devel] mbox II
by rmgls＠free.fr 29 Nov '07

29 Nov '07

On Wed, 28 Nov 2007 10:30:43 +0100 "Clemens Ladisch" <cladisch(a)fastmail.net> wrote: ... >> aplay plughw:0 (plughw:0,0) (plughw:0,1) (plughw:0,2) coq.wav >> >> this time no coredump, aplay try to play, but no sound on the monitoring >> of the mbox II, a complete silence. >> >> - perhaps the mixer is muted! >> >> i cannot adjust the mixer: >> amixer >> no elem> found. > Some USB devices don't have a software-controllable mixer. > Please show the output of "lsusb -v" for this device. lsusb -v ... Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.00 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0dba idProduct 0x3000 bcdDevice 1.38 iManufacturer 1 Digidesign iProduct 2 Mbox 2 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 646 bNumInterfaces 7 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 254 Application Specific Interface bInterfaceSubClass 1 Device Firmware Update bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 2 Mbox 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 96 bInCollection 4 baInterfaceNr( 0) 2 baInterfaceNr( 1) 3 baInterfaceNr( 2) 4 baInterfaceNr( 3) 5 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 8 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bSourceID 7 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 1 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 2 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 3 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 2 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 132 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 2 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 132 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 2 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 6 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 8193 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x00c0 1x 192 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 13 Transfer Type Isochronous Synch Type Synchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 6 bAlternateSetting 0 bNumEndpoints 3 bInterfaceClass 255 Vendor Specific Class bInterfaceSubClass 0 bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 1 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x02 EP 2 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 Device Status: 0x0000 (Bus Powered) Best Regards, raoul rmgls(a)free.Fr

1 0

[alsa-devel] [usb-audio] M-Audio FastTrack Pro giving invalid sample rates
by Lasse Kärkkäinen 29 Nov '07

29 Nov '07

Resending because nobody reacted the last time. This is still broken with 2.6.23.1, except for the fact that it doesn't seem to be possible to adjust bConfigurationValue anymore (this is not exposed under sysfs at least on my system), so I am stuck with configuration 1. Original message follows: This bug has been around for years... ALSA seems to think that the card supports a wide range of sampling rates, but it does not manage to set them in practice. I do not know which is faulty, ALSA rate setting or the USB info sent by the device, but the problem needs to be fixed. I did some measurements with the alsarate program, attached to my other post (subject: ALSA C++ API). Using bConfigurationValue=1 and measuring the actual rate with alsarate, I get >>> Testing playback Opening ALSA device hw:0,1 at 44100 Hz. Got 44100 Hz, 2 channels, period 1.02 ms. 96.27 seconds, measured rate 45807 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 44100 Hz. Got 44100 Hz, 2 channels, period 1.02 ms. 97.29 seconds, measured rate 45330 Hz. As expected, 44.1 kHz works. The measured rate seems a bit higher than it should be, but I assume that this is an alsarate inaccuracy rather than an actual phenomenon. >>> Testing playback Opening ALSA device hw:0,1 at 48000 Hz. Got 44100 Hz, 2 channels, period 1 ms. 96.27 seconds, measured rate 45808 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 48000 Hz. Got 48000 Hz, 2 channels, period 1 ms. 105.89 seconds, measured rate 45330 Hz. Using 48 kHz for playback is not possible, the snd_pcm_hw_params_rate_near returns a new rate of 44.1 kHz. This is all fine and the measured rate still is around 44.1 kHz as it should. In the capture part, we see the first problem: rate_near returned 48 kHz, but the measured value hasn't changed. So, the application thinks that it is recording at 48 kHz, but ALSA only gives it data at 44.1 kHz. This is a bug in ALSA. >>> Testing playback Opening ALSA device hw:0,1 at 8000 Hz. Got 44100 Hz, 2 channels, period 1.02 ms. 96.27 seconds, measured rate 45808 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 8000 Hz. Got 8000 Hz, 2 channels, period 2 ms. 17.66 seconds, measured rate 45309 Hz. Trying to use 8 kHz yields similar results to the 48 kHz test. No problems with playback, but the capture part reports that it is using 8 kHz, even though it really uses 44.1 kHz. >>> Testing playback Opening ALSA device hw:0,1 at 96000 Hz. Got 44100 Hz, 2 channels, period 1.02 ms. 96.27 seconds, measured rate 45808 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 96000 Hz. Got 48000 Hz, 2 channels, period 1 ms. 105.89 seconds, measured rate 45330 Hz. 96 kHz makes little difference. Now the capture rate is reported to be 48 kHz, but the actual rates are still always 44.1 kHz. That was all done using the default configuration. When using the other configuration, bConfigurationValue=2, the following results are gathered: >>> Testing playback Opening ALSA device hw:0,1 at 44100 Hz. Got 44100 Hz, 2 channels, period 1.02 ms. 96.27 seconds, measured rate 45808 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 44100 Hz. Got 44100 Hz, 2 channels, period 1.02 ms. 97.29 seconds, measured rate 45330 Hz. 44.1 kHz is identical to that in the other configuration. >>> Testing playback Opening ALSA device hw:0,1 at 48000 Hz. Got 48000 Hz, 2 channels, period 1 ms. 96.29 seconds, measured rate 49849 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 48000 Hz. Got 48000 Hz, 2 channels, period 1 ms. 97.29 seconds, measured rate 49339 Hz. 48 kHz now works properly for both playback and capture (note: measured rate over 49 kHz, but the measurements displaying about 1.5 kHz too high is consistent). >>> Testing playback Opening ALSA device hw:0,1 at 8000 Hz. Got 8000 Hz, 2 channels, period 2 ms. 15.89 seconds, measured rate 50336 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 8000 Hz. Got 8000 Hz, 2 channels, period 2 ms. 16.22 seconds, measured rate 49313 Hz. 8 kHz still doesn't work. ALSA reports that it does, but in reality gives 48 kHz (within the bounds of the measurement accuracy). >>> Testing playback Opening ALSA device hw:0,1 at 96000 Hz. Got 48000 Hz, 2 channels, period 1 ms. 96.29 seconds, measured rate 49849 Hz. >>> Testing capture Opening ALSA device hw:0,1 at 96000 Hz. Got 48000 Hz, 2 channels, period 1 ms. 97.29 seconds, measured rate 49339 Hz. 96 kHz works kinda like it should, except for the fact that the hardware should support that rate as well. Summa summarum: Config 1 only uses 44.1 kHz, but claims to support 8-48 kHz on capture. Config 2 only uses 44.1 and 48 kHz, but claims to support 8-48 kHz on both capture and playback. The card in reality, according to specs, supports up to 96 kHz. According to lspci (attached), it supports everything within the 8-96 kHz range. Bus 001 Device 010: ID 0763:2012 Midiman Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0763 Midiman idProduct 0x2012 bcdDevice 1.00 iManufacturer 1 M-Audio iProduct 2 FastTrack Pro iSerial 0 bNumConfigurations 2 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 326 bNumInterfaces 5 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 MaxPower 200mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 2 FastTrack Pro AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 75 bInCollection 4 baInterfaceNr( 0) 2 baInterfaceNr( 1) 3 baInterfaceNr( 2) 4 baInterfaceNr( 3) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 8 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bSourceID 7 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 1 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 2 FastTrack Pro MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 65 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 44100 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 1165 bNumInterfaces 6 bConfigurationValue 2 iConfiguration 0 bmAttributes 0x80 MaxPower 200mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 2 FastTrack Pro AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 97 bInCollection 5 baInterfaceNr( 0) 2 baInterfaceNr( 1) 3 baInterfaceNr( 2) 4 baInterfaceNr( 3) 5 baInterfaceNr( 4) 1 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bSourceID 5 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 7 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 8 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bSourceID 7 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 2 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 1 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0601 Analog Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 3 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 3 MIDI Streaming bInterfaceProtocol 0 iInterface 2 FastTrack Pro MIDIStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 65 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 1 Embedded bJackID 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 6 bDescriptorType 36 bDescriptorSubtype 2 (MIDI_IN_JACK) bJackType 2 External bJackID 2 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 1 Embedded bJackID 3 bNrInputPins 1 baSourceID( 0) 2 BaSourcePin( 0) 1 iJack 0 MIDIStreaming Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (MIDI_OUT_JACK) bJackType 2 External bJackID 4 bNrInputPins 1 baSourceID( 0) 1 BaSourcePin( 0) 1 iJack 0 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x01 EP 1 OUT bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 1 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 2 Transfer Type Bulk Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 0 bRefresh 0 bSynchAddress 0 MIDIStreaming Endpoint Descriptor: bLength 5 bDescriptorType 37 bDescriptorSubtype 1 (GENERAL) bNumEmbMIDIJack 1 baAssocJackID( 0) 3 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 tSamFreq[ 2] 88200 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 4 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 6 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 5 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 6 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 6 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 5 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x03 EP 3 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 131 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 6 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 20 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 4 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 tSamFreq[ 2] 88200 tSamFreq[ 3] 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 4 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 132 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 6 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 5 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 132 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 6 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 6 bNumEndpoints 2 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 132 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x84 EP 4 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 bRefresh 6 bSynchAddress 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 7 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 7 bDelay 1 frames wFormatTag 8193 IEC1937_AC-3 AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 3 (FORMAT_TYPE_III) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x04 EP 4 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 4 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 2 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0128 1x 296 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 5 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 4 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 3 bBitResolution 24 bSamFreqType 0 Continuous tLowerSamFreq 8000 tUpperSamFreq 96000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x86 EP 6 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x00 bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Bus 001 Device 006: ID 041e:4058 Creative Technology, Ltd Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 Common Class bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 idVendor 0x041e Creative Technology, Ltd idProduct 0x4058 bcdDevice 1.00 iManufacturer 1 Creative Labs iProduct 0 iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 1073 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 MaxPower 500mA UNRECOGNIZED: 08 0b 00 02 0e 03 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 1 Video Control bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x000a 1x 10 bytes bInterval 5 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0080 1x 128 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0200 1x 512 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0400 1x 0 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0b00 2x 768 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 5 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x0c00 2x 0 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 6 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x1380 3x 896 bytes bInterval 1 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 7 bNumEndpoints 1 bInterfaceClass 14 Video bInterfaceSubClass 2 Video Streaming bInterfaceProtocol 0 iInterface 0 Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x1400 3x 0 bytes bInterval 1 UNRECOGNIZED: 08 0b 02 02 01 02 00 00 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 40 bInCollection 1 baInterfaceNr( 0) 3 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0602 Digital Audio Interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 1 bControlSize 1 bmaControls( 0) 0x00 bmaControls( 1) 0x43 Mute Volume Automatic Gain bmaControls( 2) 0x43 Mute Volume Automatic Gain iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 2 iTerminal 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 3 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 35 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 9 Discrete tSamFreq[ 0] 8000 tSamFreq[ 1] 11025 tSamFreq[ 2] 12000 tSamFreq[ 3] 16000 tSamFreq[ 4] 22050 tSamFreq[ 5] 24000 tSamFreq[ 6] 32000 tSamFreq[ 7] 44100 tSamFreq[ 8] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x83 EP 3 IN bmAttributes 1 Transfer Type Isochronous Synch Type None Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 4 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Device Qualifier (for other device speed): bLength 10 bDescriptorType 6 bcdUSB 2.00 bDeviceClass 239 Miscellaneous Device bDeviceSubClass 2 Common Class bDeviceProtocol 1 Interface Association bMaxPacketSize0 64 bNumConfigurations 1 Bus 001 Device 004: ID 1532:0101 Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x1532 idProduct 0x0101 bcdDevice 21.00 iManufacturer 1 Razer iProduct 2 Razer Copperhead Laser Mouse iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 59 bNumInterfaces 2 bConfigurationValue 1 iConfiguration 0 bmAttributes 0xa0 Remote Wakeup MaxPower 200mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Devices bInterfaceSubClass 0 No Subclass bInterfaceProtocol 2 Mouse iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 10.01 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 73 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 8 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Devices bInterfaceSubClass 1 Boot Interface Subclass bInterfaceProtocol 1 Keyboard iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 10.01 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 47 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x82 EP 2 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0010 1x 16 bytes bInterval 8

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Re: [alsa-devel] Pls don't require exclusive Alsa audio device access, use snd_pcm_dmix_open ()
by thomas schorpp 28 Nov '07

28 Nov '07

thomas schorpp wrote: > thomas schorpp wrote: >> since dmix is enabled by libasound2 by default for cards not >> supporting h/w mixing. >> >> skype can access the audio device using snd_pcm_dmix_open () if other >> app has already opened it, ekiga throws error msgboxes. skype 5089 skype mem CHR 116,24 7792 /dev/snd/pcmC0D0c skype 5089 skype mem CHR 116,16 7784 /dev/snd/pcmC0D0p kio_file 5422 schorpp 0r CHR 1,3 956 /dev/null kio_uiser 6025 schorpp 0r CHR 1,3 956 /dev/null iceape-bi 30161 schorpp mem CHR 116,16 7784 /dev/snd/pcmC0D0p iceape-bi 30161 schorpp 0r CHR 1,3 956 /dev/null iceape-bi 30161 schorpp 66r CHR 116,33 7473 /dev/snd/timer iceape-bi 30161 schorpp 67u CHR 116,16 7784 /dev/snd/pcmC0D0p >> >> y >> tom >> > > hmm. takeback. future removal warning in alsa lib doc. well for > experiments (in ptlib plugin): > > Index: sound_alsa.cxx > =================================================================== > --- sound_alsa.cxx (Revision 18763) > +++ sound_alsa.cxx (Arbeitskopie) > @@ -344,13 +344,15 @@ > } > } > > - if (snd_pcm_open (&os_handle, real_device_name, stream, > SND_PCM_NONBLOCK) < 0) { > + if (snd_pcm_open (&os_handle, real_device_name, stream, > SND_PCM_NONBLOCK) < 0) > > - PTRACE (1, "ALSA\tOpen Failed"); > - return FALSE; > - } > - else > - snd_pcm_nonblock (os_handle, 0); > + if (_snd_pcm_dmix_open (&os_handle, real_device_name, snd_config_t* > root, snd_config_t* sconf, stream, SND_PCM_NONBLOCK) < 0) { > + > + PTRACE (1, "ALSA\tOpen Failed"); > + return FALSE; > + } > + else > + snd_pcm_nonblock (os_handle, 0); > > /* save internal parameters */ > device = real_device_name; > > i cant find any sample code for the missing params, google is full of > alsa user issues ;) > > y > tom > @alsa devs any hints?

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[alsa-devel] [PATCH 2/2] sis7019: add support for pre-2.6 kernels
by Dave Dillow 28 Nov '07

28 Nov '07

The sis7019 driver uses __ffs(), which is available in 2.6 kernels, but not in most earlier ones. This patches the driver to use a local version assembly version, as the hardware is SiS 55x only (i486/Pentium-ish). Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-driver hg repo. I've not build tested this version, as the build system currently hates me. However, this is 95% identical to what I was using against 1.0.13. Only the location of the __ffs -> sis_ffs change is different. pci/sis7019.patch | 32 ++++++++++++++++++++++++++++++++ pci/Makefile | 3 ++- 2 files changed, 34 insertions(+), 1 deletion(-) diff -r ab42d23439ef pci/Makefile --- a/pci/Makefile Tue Nov 27 16:05:25 2007 +0100 +++ b/pci/Makefile Wed Nov 28 01:39:04 2007 -0500 @@ -10,7 +10,7 @@ include $(SND_TOPDIR)/Makefile.conf # clean-files := ad1889.c atiixp.c atiixp_modem.c bt87x.c cmipci.c ens1370.c \ - fm801.c intel8x0.c maestro3.c via82xx.c via82xx_modem.c + fm801.c intel8x0.c maestro3.c sis7019.c via82xx.c via82xx_modem.c obj-$(CONFIG_SND) += pdplus/ asihpi/ oxygen/ @@ -31,5 +31,6 @@ fm801.c: fm801.patch $(SND_TOPDIR)/alsa- fm801.c: fm801.patch $(SND_TOPDIR)/alsa-kernel/pci/fm801.c intel8x0.c: intel8x0.patch $(SND_TOPDIR)/alsa-kernel/pci/intel8x0.c maestro3.c: maestro3.patch $(SND_TOPDIR)/alsa-kernel/pci/maestro3.c +sis7019.c: sis7019.patch $(SND_TOPDIR)/alsa-kernel/pci/sis7019.c via82xx.c: via82xx.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx.c via82xx_modem.c: via82xx_modem.patch $(SND_TOPDIR)/alsa-kernel/pci/via82xx_modem.c diff -r ab42d23439ef pci/sis7019.patch --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.patch Wed Nov 28 01:39:04 2007 -0500 @@ -0,0 +1,32 @@ +--- ../alsa-kernel/pci/sis7019.c 2007-11-28 01:02:26.000000000 -0500 ++++ sis7019.c 2007-11-28 01:36:27.000000000 -0500 +@@ -33,6 +33,14 @@ + #include <sound/initval.h> + #include "sis7019.h" + ++static inline unsigned long sis_ffs(unsigned long word) ++{ ++ __asm__("bsfl %1,%0" ++ :"=r" (word) ++ :"rm" (word)); ++ return word; ++} ++ + MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); + MODULE_DESCRIPTION("SiS 7019"); + MODULE_LICENSE("GPL"); +@@ -289,7 +297,7 @@ + int bit; + + while (status) { +- bit = __ffs(status); ++ bit = sis_ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); +@@ -1464,3 +1472,5 @@ + + module_init(sis7019_init); + module_exit(sis7019_exit); ++ ++EXPORT_NO_SYMBOLS

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[alsa-devel] [PATCH 1/2] sis7019: support the SiS 7019 Audio Accelerator
by Dave Dillow 28 Nov '07

28 Nov '07

Basic audio support for the SiS 7019 Audio Accelerator as found in the SiS 55x SoC. There is currently no synth support at the moment, but audio playback and capture with two periods per buffer has seen extensive use. Arbitrary period and buffer sizes (with multiple periods per buffer) have seen light testing, but are believed to be production ready. Signed-off-by: David Dillow <dave(a)thedillows.org> --- This is against the alsa-kernel hg repo. Changes since last posting: * Removed unnecessary locking * Added power management support * Reworked capture period timing to use dedicated silence buffer * Changed header location * Fixed muting of left stereo channel * Fixed XRUNs with non-integral periods per buffer * Better enforcement of hardware limits pci/sis7019.c | 1466 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ pci/sis7019.h | 342 +++++++++++++ pci/Kconfig | 10 pci/Makefile | 2 4 files changed, 1820 insertions(+) diff -r 17223a4918b4 pci/Kconfig --- a/pci/Kconfig Tue Nov 27 15:27:17 2007 +0100 +++ b/pci/Kconfig Wed Nov 28 01:39:33 2007 -0500 @@ -802,6 +802,16 @@ config SND_RME9652 To compile this driver as a module, choose M here: the module will be called snd-rme9652. +config SND_SIS7019 + tristate "SiS 7019 Audio Accelerator" + depends on SND + select SND_AC97_CODEC + help + Say Y here to include support for the SiS 7019 Audio Accelerator. + + To compile this driver as a module, choose M here: the module + will be called snd-sis7019. + config SND_SONICVIBES tristate "S3 SonicVibes" depends on SND diff -r 17223a4918b4 pci/Makefile --- a/pci/Makefile Tue Nov 27 15:27:17 2007 +0100 +++ b/pci/Makefile Wed Nov 28 01:39:33 2007 -0500 @@ -23,6 +23,7 @@ snd-maestro3-objs := maestro3.o snd-maestro3-objs := maestro3.o snd-rme32-objs := rme32.o snd-rme96-objs := rme96.o +snd-sis7019-objs := sis7019.o snd-sonicvibes-objs := sonicvibes.o snd-via82xx-objs := via82xx.o snd-via82xx-modem-objs := via82xx_modem.o @@ -48,6 +49,7 @@ obj-$(CONFIG_SND_MAESTRO3) += snd-maestr obj-$(CONFIG_SND_MAESTRO3) += snd-maestro3.o obj-$(CONFIG_SND_RME32) += snd-rme32.o obj-$(CONFIG_SND_RME96) += snd-rme96.o +obj-$(CONFIG_SND_SIS7019) += snd-sis7019.o obj-$(CONFIG_SND_SONICVIBES) += snd-sonicvibes.o obj-$(CONFIG_SND_VIA82XX) += snd-via82xx.o obj-$(CONFIG_SND_VIA82XX_MODEM) += snd-via82xx-modem.o diff -r 17223a4918b4 pci/sis7019.c --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.c Wed Nov 28 01:39:33 2007 -0500 @@ -0,0 +1,1466 @@ +/* + * Driver for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/pci.h> +#include <linux/time.h> +#include <linux/moduleparam.h> +#include <linux/interrupt.h> +#include <linux/delay.h> +#include <sound/core.h> +#include <sound/ac97_codec.h> +#include <sound/initval.h> +#include "sis7019.h" + +MODULE_AUTHOR("David Dillow <dave(a)thedillows.org>"); +MODULE_DESCRIPTION("SiS 7019"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{SiS,SiS7019 Audio Accelerator}}"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */ +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /* Enable this card */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for SiS7019 Audio Accelerator."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for SiS7019 Audio Accelerator."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable SiS7019 Audio Accelerator."); + +static struct pci_device_id snd_sis7019_ids[] = { + { PCI_DEVICE(PCI_VENDOR_ID_SI, 0x7019) }, + { 0, } +}; + +MODULE_DEVICE_TABLE(pci, snd_sis7019_ids); + +/* There are three timing modes for the voices. + * + * For both playback and capture, when the buffer is one or two periods long, + * we use the hardware's built-in Mid-Loop Interrupt and End-Loop Interrupt + * to let us know when the periods have ended. + * + * When performing playback with more than two periods per buffer, we set + * the "Stop Sample Offset" and tell the hardware to interrupt us when we + * reach it. We then update the offset and continue on until we are + * interrupted for the next period. + * + * Capture channels do not have a SSO, so we allocate a playback channel to + * use as a timer for the capture periods. We use the SSO on the playback + * channel to clock out virtual periods, and adjust the virtual period length + * to maintain synchronization. This algorithm came from the Trident driver. + * + * FIXME: It'd be nice to make use of some of the synth features in the + * hardware, but a woeful lack of documentation is a significant roadblock. + */ +struct voice { + u16 flags; +#define VOICE_IN_USE 1 +#define VOICE_CAPTURE 2 +#define VOICE_SSO_TIMING 4 +#define VOICE_SYNC_TIMING 8 + u16 sync_cso; + u16 period_size; + u16 buffer_size; + u16 sync_period_size; + u16 sync_buffer_size; + u32 sso; + u32 vperiod; + struct snd_pcm_substream *substream; + struct voice *timing; + void __iomem *ctrl_base; + void __iomem *wave_base; + void __iomem *sync_base; + int num; +}; + +/* We need four pages to store our wave parameters during a suspend. If + * we're not doing power management, we still need to allocate a page + * for the silence buffer. + */ +#ifdef CONFIG_PM +#define SIS_SUSPEND_PAGES 4 +#else +#define SIS_SUSPEND_PAGES 1 +#endif + +struct sis7019 { + unsigned long ioport; + void __iomem *ioaddr; + int irq; + int codecs_present; + + struct pci_dev *pci; + struct snd_pcm *pcm; + struct snd_card *card; + struct snd_ac97 *ac97[3]; + + /* Protect against more than one thread hitting the AC97 + * registers (in a more polite manner than pounding the hardware + * semaphore) + */ + struct mutex ac97_mutex; + + /* voice_lock protects allocation/freeing of the voice descriptions + */ + spinlock_t voice_lock; + + struct voice voices[64]; + struct voice capture_voice; + + /* Allocate pages to store the internal wave state during + * suspends. When we're operating, this can be used as a silence + * buffer for a timing channel. + */ + void *suspend_state[SIS_SUSPEND_PAGES]; + + int silence_users; + dma_addr_t silence_dma_addr; +}; + +#define SIS_PRIMARY_CODEC_PRESENT 0x0001 +#define SIS_SECONDARY_CODEC_PRESENT 0x0002 +#define SIS_TERTIARY_CODEC_PRESENT 0x0004 + +/* The HW has a limit in the loop end offset parameter of 0xfff8 samples, + * so our max buffer size is 0xfff8 samples * 2 channels * 2 bytes per sample. + * The minimum offset for LEO, SSO, and ESO is 9 samples, so that bounds our + * period. Given the two bounds, we get our maximum number of periods. + * + * We'll add a constraint upon open that limits the period and buffer sample + * size to values that are legal for the hardware. + */ +static struct snd_pcm_hardware sis_playback_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_CONTINUOUS, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff8 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff8 * 4), + .periods_min = 1, + .periods_max = (0xfff8 / 9), +}; + +static struct snd_pcm_hardware sis_capture_hw_info = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_SYNC_START | + SNDRV_PCM_INFO_RESUME), + .formats = (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_U16_LE), + .rates = SNDRV_PCM_RATE_48000, + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 1, + .channels_max = 2, + .buffer_bytes_max = (0xfff7 * 4), + .period_bytes_min = 9, + .period_bytes_max = (0xfff7 * 4), + .periods_min = 1, + .periods_max = (0xfff8 / 9), +}; + +static void sis_update_sso(struct voice *voice, u16 period) +{ + void __iomem *base = voice->ctrl_base; + + voice->sso += period; + if (voice->sso >= voice->buffer_size) + voice->sso -= voice->buffer_size; + + /* Enforce the hardware limit */ + if (voice->sso < 8) + voice->sso = 8; + + /* The SSO is in the upper 16 bits of the register. */ + writew(voice->sso & 0xffff, base + SIS_PLAY_DMA_SSO_ESO + 2); +} + +static void sis_update_voice(struct voice *voice) +{ + if (voice->flags & VOICE_SSO_TIMING) { + sis_update_sso(voice, voice->period_size); + } else if (voice->flags & VOICE_SYNC_TIMING) { + int sync; + + /* If we've not hit the end of the virtual period, update + * our records and keep going. + */ + if (voice->vperiod > voice->period_size) { + voice->vperiod -= voice->period_size; + if (voice->vperiod < voice->period_size) + sis_update_sso(voice, voice->vperiod); + else + sis_update_sso(voice, voice->period_size); + return; + } + + /* Calculate our relative offset between the target and + * the actual CSO value. Since we're operating in a loop, + * if the value is more than half way around, we can + * consider ourselves wrapped. + */ + sync = voice->sync_cso; + sync -= readw(voice->sync_base + SIS_CAPTURE_DMA_FORMAT_CSO); + if (sync > (voice->sync_buffer_size / 2)) + sync -= voice->sync_buffer_size; + + /* If sync is positive, then we interrupted too early, and + * we'll need to come back in a few samples and try again. + * There's a minimum wait, as it takes some time for the DMA + * engine to startup, etc... + */ + if (sync > 0) { + if (sync < 16) + sync = 16; + sis_update_sso(voice, sync); + return; + } + + /* Ok, we interrupted right on time, or (hopefully) just + * a bit late. We'll adjst our next waiting period based + * on how close we got. + * + * We need to stay just behind the actual channel to ensure + * it really is past a period when we get our interrupt -- + * otherwise we'll fall into the early code above and have + * a minimum wait time, which makes us quite late here, + * eating into the user's time to refresh the buffer, esp. + * if using small periods. + * + * If we're less than 9 samples behind, we're on target. + */ + if (sync > -9) + voice->vperiod = voice->sync_period_size + 1; + else + voice->vperiod = voice->sync_period_size - 4; + + if (voice->vperiod < voice->buffer_size) { + sis_update_sso(voice, voice->vperiod); + voice->vperiod = 0; + } else + sis_update_sso(voice, voice->period_size); + + sync = voice->sync_cso + voice->sync_period_size; + if (sync >= voice->sync_buffer_size) + sync -= voice->sync_buffer_size; + voice->sync_cso = sync; + } + + snd_pcm_period_elapsed(voice->substream); +} + +static void sis_voice_irq(u32 status, struct voice *voice) +{ + int bit; + + while (status) { + bit = __ffs(status); + status >>= bit + 1; + voice += bit; + sis_update_voice(voice); + voice++; + } +} + +static irqreturn_t sis_interrupt(int irq, void *dev) +{ + struct sis7019 *sis = dev; + unsigned long io = sis->ioport; + struct voice *voice; + u32 intr, status; + + /* We only use the DMA interrupts, and we don't enable any other + * source of interrupts. But, it is possible to see an interupt + * status that didn't actually interrupt us, so eliminate anything + * we're not expecting to avoid falsely claiming an IRQ, and an + * ensuing endless loop. + */ + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + if (!intr) + return IRQ_NONE; + + do { + status = inl(io + SIS_PISR_A); + if (status) { + sis_voice_irq(status, sis->voices); + outl(status, io + SIS_PISR_A); + } + + status = inl(io + SIS_PISR_B); + if (status) { + sis_voice_irq(status, &sis->voices[32]); + outl(status, io + SIS_PISR_B); + } + + status = inl(io + SIS_RISR); + if (status) { + voice = &sis->capture_voice; + if (!voice->timing) + snd_pcm_period_elapsed(voice->substream); + + outl(status, io + SIS_RISR); + } + + outl(intr, io + SIS_GISR); + intr = inl(io + SIS_GISR); + intr &= SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS | + SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS; + } while (intr); + + return IRQ_HANDLED; +} + +static u32 sis_rate_to_delta(unsigned int rate) +{ + u32 delta; + + /* This was copied from the trident driver, but it seems its gotten + * around a bit... nevertheless, it works well. + * + * We special case 44100 and 8000 since rounding with the equation + * does not give us an accurate enough value. For 11025 and 22050 + * the equation gives us the best answer. All other frequencies will + * also use the equation. JDW + */ + if (rate == 44100) + delta = 0xeb3; + else if (rate == 8000) + delta = 0x2ab; + else if (rate == 48000) + delta = 0x1000; + else + delta = (((rate << 12) + 24000) / 48000) & 0x0000ffff; + return delta; +} + +static void __sis_map_silence(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + if (!sis->silence_users) + sis->silence_dma_addr = pci_map_single(sis->pci, + sis->suspend_state[0], + 4096, PCI_DMA_TODEVICE); + sis->silence_users++; +} + +static void __sis_unmap_silence(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + sis->silence_users--; + if (!sis->silence_users) + pci_unmap_single(sis->pci, sis->silence_dma_addr, 4096, + PCI_DMA_TODEVICE); +} + +static void sis_free_voice(struct sis7019 *sis, struct voice *voice) +{ + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->timing) { + __sis_unmap_silence(sis); + voice->timing->flags = ~(VOICE_IN_USE | VOICE_SSO_TIMING | + VOICE_SYNC_TIMING); + voice->timing = NULL; + } + voice->flags &= ~(VOICE_IN_USE | VOICE_SSO_TIMING | VOICE_SYNC_TIMING); + spin_unlock_irqrestore(&sis->voice_lock, flags); +} + +static struct voice *__sis_alloc_playback_voice(struct sis7019 *sis) +{ + /* Must hold the voice_lock on entry */ + struct voice *voice; + int i; + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + if (voice->flags & VOICE_IN_USE) + continue; + voice->flags |= VOICE_IN_USE; + goto found_one; + } + voice = NULL; + +found_one: + return voice; +} + +static struct voice *sis_alloc_playback_voice(struct sis7019 *sis) +{ + struct voice *voice; + unsigned long flags; + + spin_lock_irqsave(&sis->voice_lock, flags); + voice = __sis_alloc_playback_voice(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + + return voice; +} + +static int sis_alloc_timing_voice(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + unsigned int period_size, buffer_size; + unsigned long flags; + int needed; + + /* If there are one or two periods per buffer, we don't need a + * timing voice, as we can use the capture channel's interrupts + * to clock out the periods. + */ + period_size = params_period_size(hw_params); + buffer_size = params_buffer_size(hw_params); + needed = (period_size != buffer_size && + period_size != (buffer_size / 2)); + + if (needed && !voice->timing) { + spin_lock_irqsave(&sis->voice_lock, flags); + voice->timing = __sis_alloc_playback_voice(sis); + if (voice->timing) + __sis_map_silence(sis); + spin_unlock_irqrestore(&sis->voice_lock, flags); + if (!voice->timing) + return -ENOMEM; + voice->timing->substream = substream; + } else if (!needed && voice->timing) { + sis_free_voice(sis, voice); + voice->timing = NULL; + } + + return 0; +} + +static int sis_playback_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice; + + voice = sis_alloc_playback_voice(sis); + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_playback_hw_info; + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff8); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff8); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_substream_close(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + + sis_free_voice(sis, voice); + return 0; +} + +static int sis_playback_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int sis_hw_free(struct snd_pcm_substream *substream) +{ + return snd_pcm_lib_free_pages(substream); +} + +static int sis_pcm_playback_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + void __iomem *ctrl_base = voice->ctrl_base; + void __iomem *wave_base = voice->wave_base; + u32 format, dma_addr, control, sso_eso, delta, reg; + u16 leo; + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format |= SIS_PLAY_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_PLAY_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_PLAY_DMA_FORMAT_MONO; + + /* The baseline setup is for a single period per buffer, and + * we add bells and whistles as needed from there. + */ + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + control = leo | SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_LEO; + sso_eso = leo; + + if (runtime->period_size == (runtime->buffer_size / 2)) { + control |= SIS_PLAY_DMA_INTR_AT_MLP; + } else if (runtime->period_size != runtime->buffer_size) { + voice->flags |= VOICE_SSO_TIMING; + voice->sso = runtime->period_size - 1; + voice->period_size = runtime->period_size; + voice->buffer_size = runtime->buffer_size; + + control &= ~SIS_PLAY_DMA_INTR_AT_LEO; + control |= SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso |= (runtime->period_size - 1) << 16; + } + + delta = sis_rate_to_delta(runtime->rate); + + /* Ok, we're ready to go, set up the channel. + */ + writel(format, ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(dma_addr, ctrl_base + SIS_PLAY_DMA_BASE); + writel(control, ctrl_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, ctrl_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + + /* Force PCI writes to post. */ + readl(ctrl_base); + + return 0; +} + +static int sis_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + unsigned long io = sis->ioport; + struct snd_pcm_substream *s; + struct voice *voice; + void *chip; + int starting; + u32 record = 0; + u32 play[2] = { 0, 0 }; + + /* No locks needed, as the PCM core will hold the locks on the + * substreams, and the HW will only start/stop the indicated voices + * without changing the state of the others. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + case SNDRV_PCM_TRIGGER_RESUME: + starting = 1; + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + case SNDRV_PCM_TRIGGER_SUSPEND: + starting = 0; + break; + default: + return -EINVAL; + } + + snd_pcm_group_for_each_entry(s, substream) { + /* Make sure it is for us... */ + chip = snd_pcm_substream_chip(s); + if (chip != sis) + continue; + + voice = s->runtime->private_data; + if (voice->flags & VOICE_CAPTURE) { + record |= 1 << voice->num; + voice = voice->timing; + } + + /* voice could be NULL if this a recording stream, and it + * doesn't have an external timing channel. + */ + if (voice) + play[voice->num / 32] |= 1 << (voice->num & 0x1f); + + snd_pcm_trigger_done(s, substream); + } + + if (starting) { + if (record) + outl(record, io + SIS_RECORD_START_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_START_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_START_B_REG); + } else { + if (record) + outl(record, io + SIS_RECORD_STOP_REG); + if (play[0]) + outl(play[0], io + SIS_PLAY_STOP_A_REG); + if (play[1]) + outl(play[1], io + SIS_PLAY_STOP_B_REG); + } + return 0; +} + +static snd_pcm_uframes_t sis_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + u32 cso; + + cso = readl(voice->ctrl_base + SIS_PLAY_DMA_FORMAT_CSO); + cso &= 0xffff; + return cso; +} + +static int sis_capture_open(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = &sis->capture_voice; + unsigned long flags; + + /* FIXME: The driver only supports recording from one channel + * at the moment, but it could support more. + */ + spin_lock_irqsave(&sis->voice_lock, flags); + if (voice->flags & VOICE_IN_USE) + voice = NULL; + else + voice->flags |= VOICE_IN_USE; + spin_unlock_irqrestore(&sis->voice_lock, flags); + + if (!voice) + return -EAGAIN; + + voice->substream = substream; + runtime->private_data = voice; + runtime->hw = sis_capture_hw_info; + runtime->hw.rates = sis->ac97[0]->rates[AC97_RATES_ADC]; + snd_pcm_limit_hw_rates(runtime); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + 9, 0xfff8); + snd_pcm_hw_constraint_minmax(runtime, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + 9, 0xfff8); + snd_pcm_set_sync(substream); + return 0; +} + +static int sis_capture_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + int rc; + + rc = snd_ac97_set_rate(sis->ac97[0], AC97_PCM_LR_ADC_RATE, + params_rate(hw_params)); + if (rc) + goto out; + + rc = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (rc < 0) + goto out; + + rc = sis_alloc_timing_voice(substream, hw_params); + +out: + return rc; +} + +static void sis_prepare_timing_voice(struct voice *timing, struct voice *cap, + struct snd_pcm_runtime *runtime) +{ + u16 buffer_size, period_size; + u32 vperiod, sso; + + timing->flags |= VOICE_SYNC_TIMING; + timing->sync_base = cap->ctrl_base; + timing->sync_cso = runtime->period_size - 1; + timing->sync_period_size = runtime->period_size; + timing->sync_buffer_size = runtime->buffer_size; + + /* Set our initial buffer and period as large as we can given a + * single page of silence. + */ + buffer_size = 4096 / runtime->channels; + buffer_size /= snd_pcm_format_size(runtime->format, 1); + period_size = buffer_size; + + /* Initially, we want to interrupt just a bit behind the end of + * the period we're clocking out. 10 samples seems to give a good + * delay. + * + * We want to spread our interrupts throughout the virtual period, + * so that we don't end up with two interrupts back to back at the + * end -- this helps minimize the effects of any jitter. Adjust our + * clocking period size so that the last period is at least a fourth + * of a full period. + * + * This is all moot if we don't need to use virtual periods. + */ + vperiod = runtime->period_size + 10; + if (vperiod > period_size) { + u16 tail = vperiod % period_size; + u16 quarter_period = period_size / 4; + + if (tail && tail < quarter_period) { + u16 loops = vperiod / period_size; + + tail = quarter_period - tail; + tail += loops - 1; + tail /= loops; + period_size -= tail; + } + + sso = period_size - 1; + } else { + /* The initial period will fit inside the buffer, so we + * don't need to use virtual periods -- disable them. + */ + period_size = runtime->period_size; + sso = vperiod - 1; + vperiod = 0; + } + + timing->period_size = period_size; + timing->buffer_size = buffer_size; + timing->sso = sso; + timing->vperiod = vperiod; +} + +static int sis_pcm_capture_prepare(struct snd_pcm_substream *substream) +{ + struct sis7019 *sis = snd_pcm_substream_chip(substream); + struct snd_pcm_runtime *runtime = substream->runtime; + struct voice *voice = runtime->private_data; + struct voice *timing = voice->timing; + void __iomem *rec_base = voice->ctrl_base; + void __iomem *play_base = NULL; + void __iomem *wave_base = NULL; + u32 format, dma_addr, rec_ctrl; + u16 leo; + + /* The following variables are only used if there is a timing channel. + */ + u32 uninitialized_var(timing_ctrl); + u32 uninitialized_var(sso_eso); + u32 uninitialized_var(delta); + + /* We rely on the PCM core to ensure that the parameters for this + * substream do not change on us while we're programming the HW. + */ + format = 0; + if (snd_pcm_format_width(runtime->format) == 8) + format = SIS_CAPTURE_DMA_FORMAT_8BIT; + if (!snd_pcm_format_signed(runtime->format)) + format |= SIS_CAPTURE_DMA_FORMAT_UNSIGNED; + if (runtime->channels == 1) + format |= SIS_CAPTURE_DMA_FORMAT_MONO; + + dma_addr = runtime->dma_addr; + leo = runtime->buffer_size - 1; + rec_ctrl = leo | SIS_CAPTURE_DMA_LOOP; + if (timing) { + /* We're using a timing voice, so the capture voice will + * not need to generate interrupts. + */ + sis_prepare_timing_voice(timing, voice, runtime); + + timing_ctrl = timing->buffer_size - 1; + timing_ctrl |= SIS_PLAY_DMA_LOOP | SIS_PLAY_DMA_INTR_AT_SSO; + sso_eso = timing->buffer_size - 1;; + sso_eso |= timing->sso << 16; + + delta = sis_rate_to_delta(runtime->rate); + + play_base = timing->ctrl_base; + wave_base = timing->wave_base; + } else { + /* We've just got one or two periods, so we can use the + * capture voice's interrupt generation. + */ + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_LEO; + if (runtime->period_size != runtime->buffer_size) + rec_ctrl |= SIS_CAPTURE_DMA_INTR_AT_MLP; + } + + /* We're ready to program the channel(s) + */ + writel(format, rec_base + SIS_CAPTURE_DMA_FORMAT_CSO); + writel(dma_addr, rec_base + SIS_CAPTURE_DMA_BASE); + writel(rec_ctrl, rec_base + SIS_CAPTURE_DMA_CONTROL); + + if (play_base) { + u32 reg; + + writel(format, play_base + SIS_PLAY_DMA_FORMAT_CSO); + writel(sis->silence_dma_addr, play_base + SIS_PLAY_DMA_BASE); + writel(timing_ctrl, play_base + SIS_PLAY_DMA_CONTROL); + writel(sso_eso, play_base + SIS_PLAY_DMA_SSO_ESO); + + for (reg = 0; reg < SIS_WAVE_SIZE; reg += 4) + writel(0, wave_base + reg); + + writel(SIS_WAVE_GENERAL_WAVE_VOLUME, + wave_base + SIS_WAVE_GENERAL); + writel(delta << 16, wave_base + SIS_WAVE_GENERAL_ARTICULATION); + writel(SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE | + SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE | + SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE, + wave_base + SIS_WAVE_CHANNEL_CONTROL); + } + + /* Force the writes to post. */ + readl(rec_base); + + return 0; +} + +static struct snd_pcm_ops sis_playback_ops = { + .open = sis_playback_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_playback_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_playback_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static struct snd_pcm_ops sis_capture_ops = { + .open = sis_capture_open, + .close = sis_substream_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = sis_capture_hw_params, + .hw_free = sis_hw_free, + .prepare = sis_pcm_capture_prepare, + .trigger = sis_pcm_trigger, + .pointer = sis_pcm_pointer, +}; + +static int __devinit sis_pcm_create(struct sis7019 *sis) +{ + struct snd_pcm *pcm; + int rc; + + rc = snd_pcm_new(sis->card, "SiS 7019", 0, 40, 1, &pcm); + if (rc) + return rc; + + pcm->private_data = sis; + strcpy(pcm->name, "SiS 7019"); + sis->pcm = pcm; + + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &sis_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &sis_capture_ops); + + /* Try to preallocate some memory, but it's not the end of the + * world if this fails. + */ + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(sis->pci), 64*1024, 128*1024); + + return 0; +} + +static unsigned short sis_ac97_rw(struct sis7019 *sis, int codec, u32 cmd) +{ + unsigned long io = sis->ioport; + unsigned short val = 0xffff; + u16 status; + u16 rdy; + int count; + const static u16 codec_ready[3] = { + SIS_AC97_STATUS_CODEC_READY, + SIS_AC97_STATUS_CODEC2_READY, + SIS_AC97_STATUS_CODEC3_READY, + }; + + rdy = codec_ready[codec]; + + + /* Get the AC97 semaphore -- software first, so we don't spin + * pounding out IO reads on the hardware semaphore... + */ + mutex_lock(&sis->ac97_mutex); + + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + goto timeout; + + /* ... and wait for any outstanding commands to complete ... + */ + count = 0xffff; + do { + status = inw(io + SIS_AC97_STATUS); + if ((status & rdy) && !(status & SIS_AC97_STATUS_BUSY)) + break; + + udelay(1); + } while (--count); + + if (!count) + goto timeout_sema; + + /* ... before sending our command and waiting for it to finish ... + */ + outl(cmd, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* ... and reading the results (if any). + */ + val = inl(io + SIS_AC97_CMD) >> 16; + +timeout_sema: + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); +timeout: + mutex_unlock(&sis->ac97_mutex); + + if (!count) { + printk(KERN_ERR "sis7019: ac97 codec %d timeout cmd 0x%08x\n", + codec, cmd); + } + + return val; +} + +static void sis_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_WRITE, + SIS_AC97_CMD_CODEC2_WRITE, + SIS_AC97_CMD_CODEC3_WRITE, + }; + sis_ac97_rw(ac97->private_data, ac97->num, + (val << 16) | (reg << 8) | cmd[ac97->num]); +} + +static unsigned short sis_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +{ + const static u32 cmd[3] = { + SIS_AC97_CMD_CODEC_READ, + SIS_AC97_CMD_CODEC2_READ, + SIS_AC97_CMD_CODEC3_READ, + }; + return sis_ac97_rw(ac97->private_data, ac97->num, + (reg << 8) | cmd[ac97->num]); +} + +static int __devinit sis_mixer_create(struct sis7019 *sis) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + static struct snd_ac97_bus_ops ops = { + .write = sis_ac97_write, + .read = sis_ac97_read, + }; + int rc; + + memset(&ac97, 0, sizeof(ac97)); + ac97.private_data = sis; + + rc = snd_ac97_bus(sis->card, 0, &ops, NULL, &bus); + if (!rc && sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[0]); + ac97.num = 1; + if (!rc && (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[1]); + ac97.num = 2; + if (!rc && (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT)) + rc = snd_ac97_mixer(bus, &ac97, &sis->ac97[2]); + + /* If we return an error here, then snd_card_free() should + * free up any ac97 codecs that got created, as well as the bus. + */ + return rc; +} + +static void sis_free_suspend(struct sis7019 *sis) +{ + int i; + + for (i = 0; i < SIS_SUSPEND_PAGES; i++) + kfree(sis->suspend_state[i]); +} + +static int sis_chip_free(struct sis7019 *sis) +{ + /* Reset the chip, and disable all interrputs. + */ + outl(SIS_GCR_SOFTWARE_RESET, sis->ioport + SIS_GCR); + udelay(10); + outl(0, sis->ioport + SIS_GCR); + outl(0, sis->ioport + SIS_GIER); + + /* Now, free everything we allocated. + */ + if (sis->irq >= 0) + free_irq(sis->irq, sis); + + if (sis->ioaddr) + iounmap(sis->ioaddr); + + pci_release_regions(sis->pci); + pci_disable_device(sis->pci); + + sis_free_suspend(sis); + return 0; +} + +static int sis_dev_free(struct snd_device *dev) +{ + struct sis7019 *sis = dev->device_data; + return sis_chip_free(sis); +} + +static int sis_chip_init(struct sis7019 *sis) +{ + unsigned long io = sis->ioport; + void __iomem *ioaddr = sis->ioaddr; + u16 status; + int count; + int i; + + /* Reset the audio controller + */ + outl(SIS_GCR_SOFTWARE_RESET, io + SIS_GCR); + udelay(10); + outl(0, io + SIS_GCR); + + /* Get the AC-link semaphore, and reset the codecs + */ + count = 0xffff; + while ((inw(io + SIS_AC97_SEMA) & SIS_AC97_SEMA_BUSY) && --count) + udelay(1); + + if (!count) + return -EIO; + + outl(SIS_AC97_CMD_CODEC_COLD_RESET, io + SIS_AC97_CMD); + udelay(10); + + count = 0xffff; + while ((inw(io + SIS_AC97_STATUS) & SIS_AC97_STATUS_BUSY) && --count) + udelay(1); + + /* Now that we've finished the reset, find out what's attached. + */ + status = inl(io + SIS_AC97_STATUS); + if (status & SIS_AC97_STATUS_CODEC_READY) + sis->codecs_present |= SIS_PRIMARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC2_READY) + sis->codecs_present |= SIS_SECONDARY_CODEC_PRESENT; + if (status & SIS_AC97_STATUS_CODEC3_READY) + sis->codecs_present |= SIS_TERTIARY_CODEC_PRESENT; + + /* All done, let go of the semaphore, and check for errors + */ + outl(SIS_AC97_SEMA_RELEASE, io + SIS_AC97_SEMA); + if (!sis->codecs_present || !count) + return -EIO; + + /* Let the hardware know that the audio driver is alive, + * and enable PCM slots on the AC-link for L/R playback (3 & 4) and + * record channels. We're going to want to use Variable Rate Audio + * for recording, to avoid needlessly resampling from 48kHZ. + */ + outl(SIS_AC97_CONF_AUDIO_ALIVE, io + SIS_AC97_CONF); + outl(SIS_AC97_CONF_AUDIO_ALIVE | SIS_AC97_CONF_PCM_LR_ENABLE | + SIS_AC97_CONF_PCM_CAP_MIC_ENABLE | + SIS_AC97_CONF_PCM_CAP_LR_ENABLE | + SIS_AC97_CONF_CODEC_VRA_ENABLE, io + SIS_AC97_CONF); + + /* All AC97 PCM slots should be sourced from sub-mixer 0. + */ + outl(0, io + SIS_AC97_PSR); + + /* There is only one valid DMA setup for a PCI environment. + */ + outl(SIS_DMA_CSR_PCI_SETTINGS, io + SIS_DMA_CSR); + + /* Reset the syncronization groups for all of the channels + * to be asyncronous. If we start doing SPDIF or 5.1 sound, etc. + * we'll need to change how we handle these. Until then, we just + * assign sub-mixer 0 to all playback channels, and avoid any + * attenuation on the audio. + */ + outl(0, io + SIS_PLAY_SYNC_GROUP_A); + outl(0, io + SIS_PLAY_SYNC_GROUP_B); + outl(0, io + SIS_PLAY_SYNC_GROUP_C); + outl(0, io + SIS_PLAY_SYNC_GROUP_D); + outl(0, io + SIS_MIXER_SYNC_GROUP); + + for (i = 0; i < 64; i++) { + writel(i, SIS_MIXER_START_ADDR(ioaddr, i)); + writel(SIS_MIXER_RIGHT_NO_ATTEN | SIS_MIXER_LEFT_NO_ATTEN | + SIS_MIXER_DEST_0, SIS_MIXER_ADDR(ioaddr, i)); + } + + /* Don't attenuate any audio set for the wave amplifier. + * + * FIXME: Maximum attenuation is set for the music amp, which will + * need to change if we start using the synth engine. + */ + outl(0xffff0000, io + SIS_WEVCR); + + /* Ensure that the wave engine is in normal operating mode. + */ + outl(0, io + SIS_WECCR); + + /* Go ahead and enable the DMA interrupts. They won't go live + * until we start a channel. + */ + outl(SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE | + SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE, io + SIS_GIER); + + return 0; +} + +#ifdef CONFIG_PM +static int sis_suspend(struct pci_dev *pci, pm_message_t state) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); + snd_pcm_suspend_all(sis->pcm); + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_suspend(sis->ac97[2]); + + /* snd_pcm_suspend_all() stopped all channels, so we're quiescent. + */ + if (sis->irq >= 0) { + synchronize_irq(sis->irq); + free_irq(sis->irq, sis); + sis->irq = -1; + } + + /* Save the internal state away + */ + for (i = 0; i < 4; i++ ) { + memcpy_fromio(sis->suspend_state[i], ioaddr, 4096); + ioaddr += 4096; + } + + pci_disable_device(pci); + pci_save_state(pci); + pci_set_power_state(pci, pci_choose_state(pci, state)); + return 0; +} + +static int sis_resume(struct pci_dev *pci) +{ + struct snd_card *card = pci_get_drvdata(pci); + struct sis7019 *sis = card->private_data; + void __iomem *ioaddr = sis->ioaddr; + int i; + + pci_set_power_state(pci, PCI_D0); + pci_restore_state(pci); + + if (pci_enable_device(pci) < 0) { + printk(KERN_ERR "sis7019: unable to re-enable device\n"); + goto error; + } + + if (sis_chip_init(sis)) { + printk(KERN_ERR "sis7019: unable to re-init controller\n"); + goto error; + } + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "sis7019: unable to regain IRQ %d\n", pci->irq); + goto error; + } + + /* Restore saved state, then clear out the page we use for the + * silence buffer. + */ + for (i = 0; i < 4; i++ ) { + memcpy_toio(ioaddr, sis->suspend_state[i], 4096); + ioaddr += 4096; + } + + memset(sis->suspend_state[0], 0, 4096); + + sis->irq = pci->irq; + pci_set_master(pci); + + if (sis->codecs_present & SIS_PRIMARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[0]); + if (sis->codecs_present & SIS_SECONDARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[1]); + if (sis->codecs_present & SIS_TERTIARY_CODEC_PRESENT) + snd_ac97_resume(sis->ac97[2]); + + snd_power_change_state(card, SNDRV_CTL_POWER_D0); + return 0; + +error: + snd_card_disconnect(card); + return -EIO; +} +#endif /* CONFIG_PM */ + +static int sis_alloc_suspend(struct sis7019 *sis) +{ + int i; + + /* We need 16K to store the internal wave engine state during a + * suspend, but we don't need it to be contiguous, so play nice + * with the memory system. We'll also use this area for a silence + * buffer. + */ + for (i = 0; i < SIS_SUSPEND_PAGES; i++) { + sis->suspend_state[i] = kmalloc(4096, GFP_KERNEL); + if (!sis->suspend_state[i]) + return -ENOMEM; + } + memset(sis->suspend_state[0], 0, 4096); + + return 0; +} + +static int __devinit sis_chip_create(struct snd_card *card, + struct pci_dev *pci) +{ + struct sis7019 *sis = card->private_data; + struct voice *voice; + static struct snd_device_ops ops = { + .dev_free = sis_dev_free, + }; + int rc; + int i; + + rc = pci_enable_device(pci); + if (rc) + goto error_out; + + if (pci_set_dma_mask(pci, DMA_30BIT_MASK) < 0) { + printk(KERN_ERR "sis7019: architecture does not support " + "30-bit PCI busmaster DMA"); + goto error_out_enabled; + } + + memset(sis, 0, sizeof(*sis)); + mutex_init(&sis->ac97_mutex); + spin_lock_init(&sis->voice_lock); + sis->card = card; + sis->pci = pci; + sis->irq = -1; + sis->ioport = pci_resource_start(pci, 0); + + rc = pci_request_regions(pci, "SiS 7019"); + if (rc) { + printk(KERN_ERR "sis7019: unable request regions\n"); + goto error_out_enabled; + } + + rc = -EIO; + sis->ioaddr = ioremap_nocache(pci_resource_start(pci, 1), 0x4000); + if (!sis->ioaddr) { + printk(KERN_ERR "sis7019: unable to remap MMIO, aborting\n"); + goto error_out_cleanup; + } + + rc = sis_alloc_suspend(sis); + if (rc < 0) { + printk(KERN_ERR "sis7019: unable to allocate state storage\n"); + goto error_out_cleanup; + } + + rc = sis_chip_init(sis); + if (rc) + goto error_out_cleanup; + + if (request_irq(pci->irq, sis_interrupt, IRQF_DISABLED|IRQF_SHARED, + card->shortname, sis)) { + printk(KERN_ERR "unable to allocate irq %d\n", sis->irq); + goto error_out_cleanup; + } + + sis->irq = pci->irq; + pci_set_master(pci); + + for (i = 0; i < 64; i++) { + voice = &sis->voices[i]; + voice->num = i; + voice->ctrl_base = SIS_PLAY_DMA_ADDR(sis->ioaddr, i); + voice->wave_base = SIS_WAVE_ADDR(sis->ioaddr, i); + } + + voice = &sis->capture_voice; + voice->flags = VOICE_CAPTURE; + voice->num = SIS_CAPTURE_CHAN_AC97_PCM_IN; + voice->ctrl_base = SIS_CAPTURE_DMA_ADDR(sis->ioaddr, voice->num); + + rc = snd_device_new(card, SNDRV_DEV_LOWLEVEL, sis, &ops); + if (rc) + goto error_out_cleanup; + + snd_card_set_dev(card, &pci->dev); + + return 0; + +error_out_cleanup: + sis_chip_free(sis); + +error_out_enabled: + pci_disable_device(pci); + +error_out: + return rc; +} + +static int __devinit snd_sis7019_probe(struct pci_dev *pci, + const struct pci_device_id *pci_id) +{ + struct snd_card *card; + struct sis7019 *sis; + static int dev; + int rc; + + rc = -ENODEV; + if (dev >= SNDRV_CARDS) + goto error_out; + + rc = -ENOENT; + if (!enable[dev]) { + dev++; + goto error_out; + } + + rc = -ENOMEM; + card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(*sis)); + if (!card) + goto error_out; + + strcpy(card->driver, "SiS 7019"); + strcpy(card->shortname, "SiS 7019"); + rc = sis_chip_create(card, pci); + if (rc) + goto card_error_out; + + sis = card->private_data; + + rc = sis_mixer_create(sis); + if (rc) + goto card_error_out; + + rc = sis_pcm_create(sis); + if (rc) + goto card_error_out; + + snprintf(card->longname, sizeof(card->longname), + "%s Audio Accelerator with %s at 0x%lx, irq %d", + card->shortname, snd_ac97_get_short_name(sis->ac97[0]), + sis->ioport, sis->irq); + + rc = snd_card_register(card); + if (rc) + goto card_error_out; + + pci_set_drvdata(pci, card); + dev++; + return 0; + +card_error_out: + snd_card_free(card); + +error_out: + return rc; +} + +static void __devexit snd_sis7019_remove(struct pci_dev *pci) +{ + snd_card_free(pci_get_drvdata(pci)); + pci_set_drvdata(pci, NULL); +} + +static struct pci_driver sis7019_driver = { + .name = "SiS7019", + .id_table = snd_sis7019_ids, + .probe = snd_sis7019_probe, + .remove = __devexit_p(snd_sis7019_remove), + +#ifdef CONFIG_PM + .suspend = sis_suspend, + .resume = sis_resume, +#endif +}; + +static int __init sis7019_init(void) +{ + return pci_register_driver(&sis7019_driver); +} + +static void __exit sis7019_exit(void) +{ + pci_unregister_driver(&sis7019_driver); +} + +module_init(sis7019_init); +module_exit(sis7019_exit); diff -r 17223a4918b4 pci/sis7019.h --- /dev/null Thu Jan 01 00:00:00 1970 +0000 +++ b/pci/sis7019.h Wed Nov 28 01:39:33 2007 -0500 @@ -0,0 +1,342 @@ +#ifndef __sis7019_h__ +#define __sis7019_h__ + +/* + * Definitions for SiS7019 Audio Accelerator + * + * Copyright (C) 2004-2007, David Dillow + * Written by David Dillow <dave(a)thedillows.org> + * Inspired by the Trident 4D-WaveDX/NX driver. + * + * All rights reserved. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation, version 2. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + */ + + +/* General Control Register */ +#define SIS_GCR 0x00 +#define SIS_GCR_MACRO_POWER_DOWN 0x80000000 +#define SIS_GCR_MODEM_ENABLE 0x00010000 +#define SIS_GCR_SOFTWARE_RESET 0x00000001 + +/* General Interrupt Enable Register */ +#define SIS_GIER 0x04 +#define SIS_GIER_MODEM_TIMER_IRQ_ENABLE 0x00100000 +#define SIS_GIER_MODEM_RX_DMA_IRQ_ENABLE 0x00080000 +#define SIS_GIER_MODEM_TX_DMA_IRQ_ENABLE 0x00040000 +#define SIS_GIER_AC97_GPIO1_IRQ_ENABLE 0x00020000 +#define SIS_GIER_AC97_GPIO0_IRQ_ENABLE 0x00010000 +#define SIS_GIER_AC97_SAMPLE_TIMER_IRQ_ENABLE 0x00000010 +#define SIS_GIER_AUDIO_GLOBAL_TIMER_IRQ_ENABLE 0x00000008 +#define SIS_GIER_AUDIO_RECORD_DMA_IRQ_ENABLE 0x00000004 +#define SIS_GIER_AUDIO_PLAY_DMA_IRQ_ENABLE 0x00000002 +#define SIS_GIER_AUDIO_WAVE_ENGINE_IRQ_ENABLE 0x00000001 + +/* General Interrupt Status Register */ +#define SIS_GISR 0x08 +#define SIS_GISR_MODEM_TIMER_IRQ_STATUS 0x00100000 +#define SIS_GISR_MODEM_RX_DMA_IRQ_STATUS 0x00080000 +#define SIS_GISR_MODEM_TX_DMA_IRQ_STATUS 0x00040000 +#define SIS_GISR_AC97_GPIO1_IRQ_STATUS 0x00020000 +#define SIS_GISR_AC97_GPIO0_IRQ_STATUS 0x00010000 +#define SIS_GISR_AC97_SAMPLE_TIMER_IRQ_STATUS 0x00000010 +#define SIS_GISR_AUDIO_GLOBAL_TIMER_IRQ_STATUS 0x00000008 +#define SIS_GISR_AUDIO_RECORD_DMA_IRQ_STATUS 0x00000004 +#define SIS_GISR_AUDIO_PLAY_DMA_IRQ_STATUS 0x00000002 +#define SIS_GISR_AUDIO_WAVE_ENGINE_IRQ_STATUS 0x00000001 + +/* DMA Control Register */ +#define SIS_DMA_CSR 0x10 +#define SIS_DMA_CSR_PCI_SETTINGS 0x0000001d +#define SIS_DMA_CSR_CONCURRENT_ENABLE 0x00000200 +#define SIS_DMA_CSR_PIPELINE_ENABLE 0x00000100 +#define SIS_DMA_CSR_RX_DRAIN_ENABLE 0x00000010 +#define SIS_DMA_CSR_RX_FILL_ENABLE 0x00000008 +#define SIS_DMA_CSR_TX_DRAIN_ENABLE 0x00000004 +#define SIS_DMA_CSR_TX_LOWPRI_FILL_ENABLE 0x00000002 +#define SIS_DMA_CSR_TX_HIPRI_FILL_ENABLE 0x00000001 + +/* Playback Channel Start Registers */ +#define SIS_PLAY_START_A_REG 0x14 +#define SIS_PLAY_START_B_REG 0x18 + +/* Playback Channel Stop Registers */ +#define SIS_PLAY_STOP_A_REG 0x1c +#define SIS_PLAY_STOP_B_REG 0x20 + +/* Recording Channel Start Register */ +#define SIS_RECORD_START_REG 0x24 + +/* Recording Channel Stop Register */ +#define SIS_RECORD_STOP_REG 0x28 + +/* Playback Interrupt Status Registers */ +#define SIS_PISR_A 0x2c +#define SIS_PISR_B 0x30 + +/* Recording Interrupt Status Register */ +#define SIS_RISR 0x34 + +/* AC97 AC-link Playback Source Register */ +#define SIS_AC97_PSR 0x40 +#define SIS_AC97_PSR_MODEM_HEADSET_SRC_MIXER 0x0f000000 +#define SIS_AC97_PSR_MODEM_LINE2_SRC_MIXER 0x00f00000 +#define SIS_AC97_PSR_MODEM_LINE1_SRC_MIXER 0x000f0000 +#define SIS_AC97_PSR_PCM_LFR_SRC_MIXER 0x0000f000 +#define SIS_AC97_PSR_PCM_SURROUND_SRC_MIXER 0x00000f00 +#define SIS_AC97_PSR_PCM_CENTER_SRC_MIXER 0x000000f0 +#define SIS_AC97_PSR_PCM_LR_SRC_MIXER 0x0000000f + +/* AC97 AC-link Command Register */ +#define SIS_AC97_CMD 0x50 +#define SIS_AC97_CMD_DATA_MASK 0xffff0000 +#define SIS_AC97_CMD_REG_MASK 0x0000ff00 +#define SIS_AC97_CMD_CODEC3_READ 0x0000000d +#define SIS_AC97_CMD_CODEC3_WRITE 0x0000000c +#define SIS_AC97_CMD_CODEC2_READ 0x0000000b +#define SIS_AC97_CMD_CODEC2_WRITE 0x0000000a +#define SIS_AC97_CMD_CODEC_READ 0x00000009 +#define SIS_AC97_CMD_CODEC_WRITE 0x00000008 +#define SIS_AC97_CMD_CODEC_WARM_RESET 0x00000005 +#define SIS_AC97_CMD_CODEC_COLD_RESET 0x00000004 +#define SIS_AC97_CMD_DONE 0x00000000 + +/* AC97 AC-link Semaphore Register */ +#define SIS_AC97_SEMA 0x54 +#define SIS_AC97_SEMA_BUSY 0x00000001 +#define SIS_AC97_SEMA_RELEASE 0x00000000 + +/* AC97 AC-link Status Register */ +#define SIS_AC97_STATUS 0x58 +#define SIS_AC97_STATUS_AUDIO_D2_INACT_SECS 0x03f00000 +#define SIS_AC97_STATUS_MODEM_ALIVE 0x00002000 +#define SIS_AC97_STATUS_AUDIO_ALIVE 0x00001000 +#define SIS_AC97_STATUS_CODEC3_READY 0x00000400 +#define SIS_AC97_STATUS_CODEC2_READY 0x00000200 +#define SIS_AC97_STATUS_CODEC_READY 0x00000100 +#define SIS_AC97_STATUS_WARM_RESET 0x00000080 +#define SIS_AC97_STATUS_COLD_RESET 0x00000040 +#define SIS_AC97_STATUS_POWERED_DOWN 0x00000020 +#define SIS_AC97_STATUS_NORMAL 0x00000010 +#define SIS_AC97_STATUS_READ_EXPIRED 0x00000004 +#define SIS_AC97_STATUS_SEMAPHORE 0x00000002 +#define SIS_AC97_STATUS_BUSY 0x00000001 + +/* AC97 AC-link Audio Configuration Register */ +#define SIS_AC97_CONF 0x5c +#define SIS_AC97_CONF_AUDIO_ALIVE 0x80000000 +#define SIS_AC97_CONF_WARM_RESET_ENABLE 0x40000000 +#define SIS_AC97_CONF_PR6_ENABLE 0x20000000 +#define SIS_AC97_CONF_PR5_ENABLE 0x10000000 +#define SIS_AC97_CONF_PR4_ENABLE 0x08000000 +#define SIS_AC97_CONF_PR3_ENABLE 0x04000000 +#define SIS_AC97_CONF_PR2_PR7_ENABLE 0x02000000 +#define SIS_AC97_CONF_PR0_PR1_ENABLE 0x01000000 +#define SIS_AC97_CONF_AUTO_PM_ENABLE 0x00800000 +#define SIS_AC97_CONF_PCM_LFE_ENABLE 0x00080000 +#define SIS_AC97_CONF_PCM_SURROUND_ENABLE 0x00040000 +#define SIS_AC97_CONF_PCM_CENTER_ENABLE 0x00020000 +#define SIS_AC97_CONF_PCM_LR_ENABLE 0x00010000 +#define SIS_AC97_CONF_PCM_CAP_MIC_ENABLE 0x00002000 +#define SIS_AC97_CONF_PCM_CAP_LR_ENABLE 0x00001000 +#define SIS_AC97_CONF_PCM_CAP_MIC_FROM_CODEC3 0x00000200 +#define SIS_AC97_CONF_PCM_CAP_LR_FROM_CODEC3 0x00000100 +#define SIS_AC97_CONF_CODEC3_PM_VRM 0x00000080 +#define SIS_AC97_CONF_CODEC_PM_VRM 0x00000040 +#define SIS_AC97_CONF_CODEC3_VRA_ENABLE 0x00000020 +#define SIS_AC97_CONF_CODEC_VRA_ENABLE 0x00000010 +#define SIS_AC97_CONF_CODEC3_PM_EAC 0x00000008 +#define SIS_AC97_CONF_CODEC_PM_EAC 0x00000004 +#define SIS_AC97_CONF_CODEC3_EXISTS 0x00000002 +#define SIS_AC97_CONF_CODEC_EXISTS 0x00000001 + +/* Playback Channel Sync Group registers */ +#define SIS_PLAY_SYNC_GROUP_A 0x80 +#define SIS_PLAY_SYNC_GROUP_B 0x84 +#define SIS_PLAY_SYNC_GROUP_C 0x88 +#define SIS_PLAY_SYNC_GROUP_D 0x8c +#define SIS_MIXER_SYNC_GROUP 0x90 + +/* Wave Engine Config and Control Register */ +#define SIS_WECCR 0xa0 +#define SIS_WECCR_TESTMODE_MASK 0x00300000 +#define SIS_WECCR_TESTMODE_NORMAL 0x00000000 +#define SIS_WECCR_TESTMODE_BYPASS_NSO_ALPHA 0x00100000 +#define SIS_WECCR_TESTMODE_BYPASS_FC 0x00200000 +#define SIS_WECCR_TESTMODE_BYPASS_WOL 0x00300000 +#define SIS_WECCR_RESONANCE_DELAY_MASK 0x00060000 +#define SIS_WECCR_RESONANCE_DELAY_NONE 0x00000000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1F00 0x00020000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1E00 0x00040000 +#define SIS_WECCR_RESONANCE_DELAY_FC_1C00 0x00060000 +#define SIS_WECCR_IGNORE_CHANNEL_PARMS 0x00010000 +#define SIS_WECCR_COMMAND_CHANNEL_ID_MASK 0x0003ff00 +#define SIS_WECCR_COMMAND_MASK 0x00000007 +#define SIS_WECCR_COMMAND_NONE 0x00000000 +#define SIS_WECCR_COMMAND_DONE 0x00000000 +#define SIS_WECCR_COMMAND_PAUSE 0x00000001 +#define SIS_WECCR_COMMAND_TOGGLE_VEG 0x00000002 +#define SIS_WECCR_COMMAND_TOGGLE_MEG 0x00000003 +#define SIS_WECCR_COMMAND_TOGGLE_VEG_MEG 0x00000004 + +/* Wave Engine Volume Control Register */ +#define SIS_WEVCR 0xa4 +#define SIS_WEVCR_LEFT_MUSIC_ATTENUATION_MASK 0xff000000 +#define SIS_WEVCR_RIGHT_MUSIC_ATTENUATION_MASK 0x00ff0000 +#define SIS_WEVCR_LEFT_WAVE_ATTENUATION_MASK 0x0000ff00 +#define SIS_WEVCR_RIGHT_WAVE_ATTENUATION_MASK 0x000000ff + +/* Wave Engine Interrupt Status Registers */ +#define SIS_WEISR_A 0xa8 +#define SIS_WEISR_B 0xac + + +/* Playback DMA parameters (paramter RAM) */ +#define SIS_PLAY_DMA_OFFSET 0x0000 +#define SIS_PLAY_DMA_SIZE 0x10 +#define SIS_PLAY_DMA_ADDR(addr, num) \ + ((num * SIS_PLAY_DMA_SIZE) + (addr) + SIS_PLAY_DMA_OFFSET) + +#define SIS_PLAY_DMA_FORMAT_CSO 0x00 +#define SIS_PLAY_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_PLAY_DMA_FORMAT_8BIT 0x00040000 +#define SIS_PLAY_DMA_FORMAT_MONO 0x00020000 +#define SIS_PLAY_DMA_CSO_MASK 0x0000ffff +#define SIS_PLAY_DMA_BASE 0x04 +#define SIS_PLAY_DMA_CONTROL 0x08 +#define SIS_PLAY_DMA_STOP_AT_SSO 0x04000000 +#define SIS_PLAY_DMA_RELEASE 0x02000000 +#define SIS_PLAY_DMA_LOOP 0x01000000 +#define SIS_PLAY_DMA_INTR_AT_SSO 0x00080000 +#define SIS_PLAY_DMA_INTR_AT_ESO 0x00040000 +#define SIS_PLAY_DMA_INTR_AT_LEO 0x00020000 +#define SIS_PLAY_DMA_INTR_AT_MLP 0x00010000 +#define SIS_PLAY_DMA_LEO_MASK 0x0000ffff +#define SIS_PLAY_DMA_SSO_ESO 0x0c +#define SIS_PLAY_DMA_SSO_MASK 0xffff0000 +#define SIS_PLAY_DMA_ESO_MASK 0x0000ffff + +/* Capture DMA parameters (paramter RAM) */ +#define SIS_CAPTURE_DMA_OFFSET 0x0800 +#define SIS_CAPTURE_DMA_SIZE 0x10 +#define SIS_CAPTURE_DMA_ADDR(addr, num) \ + ((num * SIS_CAPTURE_DMA_SIZE) + (addr) + SIS_CAPTURE_DMA_OFFSET) + +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_0 0 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_1 1 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_2 2 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_3 3 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_4 4 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_5 5 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_6 6 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_7 7 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_8 8 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_9 9 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_10 10 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_11 11 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_12 12 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_13 13 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_14 14 +#define SIS_CAPTURE_CHAN_MIXER_ROUTE_BACK_15 15 +#define SIS_CAPTURE_CHAN_AC97_PCM_IN 16 +#define SIS_CAPTURE_CHAN_AC97_MIC_IN 17 +#define SIS_CAPTURE_CHAN_AC97_LINE1_IN 18 +#define SIS_CAPTURE_CHAN_AC97_LINE2_IN 19 +#define SIS_CAPTURE_CHAN_AC97_HANDSE_IN 20 + +#define SIS_CAPTURE_DMA_FORMAT_CSO 0x00 +#define SIS_CAPTURE_DMA_MONO_MODE_MASK 0xc0000000 +#define SIS_CAPTURE_DMA_MONO_MODE_AVG 0x00000000 +#define SIS_CAPTURE_DMA_MONO_MODE_LEFT 0x40000000 +#define SIS_CAPTURE_DMA_MONO_MODE_RIGHT 0x80000000 +#define SIS_CAPTURE_DMA_FORMAT_UNSIGNED 0x00080000 +#define SIS_CAPTURE_DMA_FORMAT_8BIT 0x00040000 +#define SIS_CAPTURE_DMA_FORMAT_MONO 0x00020000 +#define SIS_CAPTURE_DMA_CSO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_BASE 0x04 +#define SIS_CAPTURE_DMA_CONTROL 0x08 +#define SIS_CAPTURE_DMA_STOP_AT_SSO 0x04000000 +#define SIS_CAPTURE_DMA_RELEASE 0x02000000 +#define SIS_CAPTURE_DMA_LOOP 0x01000000 +#define SIS_CAPTURE_DMA_INTR_AT_LEO 0x00020000 +#define SIS_CAPTURE_DMA_INTR_AT_MLP 0x00010000 +#define SIS_CAPTURE_DMA_LEO_MASK 0x0000ffff +#define SIS_CAPTURE_DMA_RESERVED 0x0c + + +/* Mixer routing list start pointer (parameter RAM) */ +#define SIS_MIXER_START_OFFSET 0x1000 +#define SIS_MIXER_START_SIZE 0x04 +#define SIS_MIXER_START_ADDR(addr, num) \ + ((num * SIS_MIXER_START_SIZE) + (addr) + SIS_MIXER_START_OFFSET) + +#define SIS_MIXER_START_MASK 0x0000007f + +/* Mixer routing table (parameter RAM) */ +#define SIS_MIXER_OFFSET 0x1400 +#define SIS_MIXER_SIZE 0x04 +#define SIS_MIXER_ADDR(addr, num) \ + ((num * SIS_MIXER_SIZE) + (addr) + SIS_MIXER_OFFSET) + +#define SIS_MIXER_RIGHT_ATTENUTATION_MASK 0xff000000 +#define SIS_MIXER_RIGHT_NO_ATTEN 0xff000000 +#define SIS_MIXER_LEFT_ATTENUTATION_MASK 0x00ff0000 +#define SIS_MIXER_LEFT_NO_ATTEN 0x00ff0000 +#define SIS_MIXER_NEXT_ENTRY_MASK 0x00007f00 +#define SIS_MIXER_NEXT_ENTRY_NONE 0x00000000 +#define SIS_MIXER_DEST_MASK 0x0000007f +#define SIS_MIXER_DEST_0 0x00000020 +#define SIS_MIXER_DEST_1 0x00000021 +#define SIS_MIXER_DEST_2 0x00000022 +#define SIS_MIXER_DEST_3 0x00000023 +#define SIS_MIXER_DEST_4 0x00000024 +#define SIS_MIXER_DEST_5 0x00000025 +#define SIS_MIXER_DEST_6 0x00000026 +#define SIS_MIXER_DEST_7 0x00000027 +#define SIS_MIXER_DEST_8 0x00000028 +#define SIS_MIXER_DEST_9 0x00000029 +#define SIS_MIXER_DEST_10 0x0000002a +#define SIS_MIXER_DEST_11 0x0000002b +#define SIS_MIXER_DEST_12 0x0000002c +#define SIS_MIXER_DEST_13 0x0000002d +#define SIS_MIXER_DEST_14 0x0000002e +#define SIS_MIXER_DEST_15 0x0000002f + +/* Wave Engine Control Parameters (parameter RAM) */ +#define SIS_WAVE_OFFSET 0x2000 +#define SIS_WAVE_SIZE 0x40 +#define SIS_WAVE_ADDR(addr, num) \ + ((num * SIS_WAVE_SIZE) + (addr) + SIS_WAVE_OFFSET) + +#define SIS_WAVE_GENERAL 0x00 +#define SIS_WAVE_GENERAL_WAVE_VOLUME 0x80000000 +#define SIS_WAVE_GENERAL_MUSIC_VOLUME 0x00000000 +#define SIS_WAVE_GENERAL_VOLUME_MASK 0x7f000000 +#define SIS_WAVE_GENERAL_ARTICULATION 0x04 +#define SIS_WAVE_GENERAL_ARTICULATION_DELTA_MASK 0x3fff0000 +#define SIS_WAVE_ARTICULATION 0x08 +#define SIS_WAVE_TIMER 0x0c +#define SIS_WAVE_GENERATOR 0x10 +#define SIS_WAVE_CHANNEL_CONTROL 0x14 +#define SIS_WAVE_CHANNEL_CONTROL_FIRST_SAMPLE 0x80000000 +#define SIS_WAVE_CHANNEL_CONTROL_AMP_ENABLE 0x40000000 +#define SIS_WAVE_CHANNEL_CONTROL_FILTER_ENABLE 0x20000000 +#define SIS_WAVE_CHANNEL_CONTROL_INTERPOLATE_ENABLE 0x10000000 +#define SIS_WAVE_LFO_EG_CONTROL 0x18 +#define SIS_WAVE_LFO_EG_CONTROL_2 0x1c +#define SIS_WAVE_LFO_EG_CONTROL_3 0x20 +#define SIS_WAVE_LFO_EG_CONTROL_4 0x24 + +#endif /* __sis7019_h__ */

2 5

[alsa-devel] [PATCH][RFC] pcm: add exclusion between hw_params and prepare callbacks
by Dave Dillow 28 Nov '07

28 Nov '07

While working on the locking in the sis7019 driver, and as also noted by Trent Piepho, there is a race between threads calling into the driver's hw_params callback and its prepare callback. This can lead to incorrect/inconsistent data being programmed into the audio device, and lead to unexpected results, if not a possible system crash. While it is arguably a bug in the user space program, we should protect the system and provide some assurance to the driver code about concurrency between these two callbacks. The attached patch adds a mutex to the two paths to provide the locking without giving up the ability for those calls to schedule, as documented in the API. It just to get a discussion going, as the code to be restructured a bit to clean up the error exit paths. It is also likely there are other callbacks that should be excluded when either of these are executing as well. This applies to the base kernel tree. Dave --- include/sound/pcm.h | 2 ++ sound/core/pcm.c | 1 + sound/core/pcm_native.c | 20 +++++++++++++++++--- 3 files changed, 20 insertions(+), 3 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 73334e0..989e84f 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -388,6 +388,8 @@ struct snd_pcm_substream { struct snd_info_entry *proc_prealloc_entry; struct snd_info_entry *proc_prealloc_max_entry; #endif + /* -- hw_param/prepare exclusion -- */ + struct mutex hw_param_mutex; /* misc flags */ unsigned int hw_opened: 1; }; diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 2743414..d728594 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -651,6 +651,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) list_add_tail(&substream->link_list, &substream->self_group.substreams); spin_lock_init(&substream->timer_lock); atomic_set(&substream->mmap_count, 0); + mutex_init(&substream->hw_param_mutex); prev = substream; } return 0; diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59b29cd..638d78a 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -373,6 +373,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, snd_assert(substream != NULL, return -ENXIO); runtime = substream->runtime; snd_assert(runtime != NULL, return -ENXIO); + mutex_lock(&substream->hw_param_mutex); snd_pcm_stream_lock_irq(substream); switch (runtime->status->state) { case SNDRV_PCM_STATE_OPEN: @@ -380,6 +381,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_PREPARED: break; default: + mutex_unlock(&substream->hw_param_mutex); snd_pcm_stream_unlock_irq(substream); return -EBADFD; } @@ -387,8 +389,10 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, #if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE) if (!substream->oss.oss) #endif - if (atomic_read(&substream->mmap_count)) + if (atomic_read(&substream->mmap_count)) { + mutex_unlock(&substream->hw_param_mutex); return -EBADFD; + } params->rmask = ~0U; err = snd_pcm_hw_refine(substream, params); @@ -450,6 +454,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, remove_acceptable_latency(substream->latency_id); if ((usecs = period_to_usecs(runtime)) >= 0) set_acceptable_latency(substream->latency_id, usecs); + mutex_unlock(&substream->hw_param_mutex); return 0; _error: /* hardware might be unuseable from this time, @@ -458,6 +463,7 @@ static int snd_pcm_hw_params(struct snd_pcm_substream *substream, runtime->status->state = SNDRV_PCM_STATE_OPEN; if (substream->ops->hw_free != NULL) substream->ops->hw_free(substream); + mutex_unlock(&substream->hw_param_mutex); return err; } @@ -1345,9 +1351,10 @@ static struct action_ops snd_pcm_action_prepare = { static int snd_pcm_prepare(struct snd_pcm_substream *substream, struct file *file) { - int res; struct snd_card *card = substream->pcm->card; + struct snd_pcm_substream *s; int f_flags; + int res; if (file) f_flags = file->f_flags; @@ -1355,9 +1362,16 @@ static int snd_pcm_prepare(struct snd_pcm_substream *substream, f_flags = substream->f_flags; snd_power_lock(card); - if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0) + if ((res = snd_power_wait(card, SNDRV_CTL_POWER_D0)) >= 0) { + down_read(&snd_pcm_link_rwsem); + snd_pcm_group_for_each_entry(s, substream) + mutex_lock(&s->hw_param_mutex); res = snd_pcm_action_nonatomic(&snd_pcm_action_prepare, substream, f_flags); + snd_pcm_group_for_each_entry(s, substream) + mutex_unlock(&s->hw_param_mutex); + up_read(&snd_pcm_link_rwsem); + } snd_power_unlock(card); return res; }

2 1

Re: [alsa-devel] mbox II
by rmgls＠free.fr 28 Nov '07

28 Nov '07

Hi Clemens and list, ... > try plughw ... i tried: aplay plughw:0 (plughw:0,0) (plughw:0,1) (plughw:0,2) coq.wav this time no coredump, aplay try to play, but no sound on the monitoring of the mbox II, a complete silence. - perhaps the mixer is muted! i cannot adjust the mixer: amiser no elements found. the technical support contacted tell me: they have no tech doc for the box. i will try on the duc forum on digidisign.com. i am unsure to have something there! otherwise, is a wrapper to the win driver suitable, or even possible? thanks. best regards Raoul rmgls(a)free.fr

2 1

[alsa-devel] No sound on Intel DQ965GF motherboard -- SOLVED
by Eric S. Raymond 28 Nov '07

28 Nov '07

Thanks to all who tried to help. The ALSA 1.0.15 backport modules solved my problem, you can consider the DQ965GF fully supported now. -- >>esr>>

1 0

Re: [alsa-devel] [PATCH 2/3] PNP cleanups - Unify the pnp macros to access resources in the pnp resource table
by Rene Herman 27 Nov '07

27 Nov '07

On 20-11-07 15:19, Thomas Renninger wrote: > At the end is some example code how things could get even more cleaned > up. It shows how I think pnp layer and one example driver would get > adjusted. There are not that much drivers making use of > pnp_resource_change... The ALSA ISA-PnP drivers do in what I personally consider a bad layering violation. Brought up not doing that a while ago on alsa-devel, but no discussion ensued. If the PnP maintainer(s) agree with me though, I'd be more than happy to rip all of that out of ALSA. (added alsa-devel) Rene.

3 4

[alsa-devel] 1.0.15: volume levels drift on HDA with STAC92XX codec
by Chuck Ebbert 26 Nov '07

26 Nov '07

We have two reports now of unstable volume levels: https://bugzilla.redhat.com/show_bug.cgi?id=361051 https://bugzilla.redhat.com/show_bug.cgi?id=354981 This is with kernel 2.6.23 plus the two ALSA merge patches from 2.6.23-rc1.

6 25

[alsa-devel] [PATCH] add number of periods constraint to snd-aoa [try 3]
by Heikki O Lindholm 26 Nov '07

26 Nov '07

From: Heikki Lindholm <holindho(a)cs.helsinki.fi> The aoa driver is not specifying constraints on number of periods, and, it seems, it might end with a non-integer number, which it cannot deal with. Fix by adding a proper constraint. Signed-off-by: Heikki Lindholm <holindho(a)cs.helsinki.fi> --- try 3: fixed coding style diff -r 1b54a8725ded aoa/soundbus/i2sbus/i2sbus-pcm.c --- a/aoa/soundbus/i2sbus/i2sbus-pcm.c Wed Nov 14 17:07:17 2007 +0100 +++ b/aoa/soundbus/i2sbus/i2sbus-pcm.c Fri Nov 23 14:16:07 2007 +0200 @@ -194,6 +194,12 @@ static int i2sbus_pcm_open(struct i2sbus hw->period_bytes_max = 16384; hw->periods_min = 3; hw->periods_max = MAX_DBDMA_COMMANDS; + err = snd_pcm_hw_constraint_integer(pi->substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + result = err; + goto out_unlock; + } list_for_each_entry(cii, &sdev->codec_list, list) { if (cii->codec->open) { err = cii->codec->open(cii, pi->substream);

5 10

[alsa-devel] ALSA Leaking memory
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! Apparently alsa-libs .15 is leaking memory when using snd_pcm_open() on some devices that cannot be opened. On my USB wecam which has only a single mono audio stream, opening it as "front:3" (which of course fails) leaks a bit of memory. Consider this example: <snip> #include <asoundlib.h> #include <assert.h> int main(int argc, char *argv[]) { snd_pcm_t *pcm; if (snd_pcm_open(&pcm, "front:3", SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK) == 0) { int r = snd_pcm_close(pcm); assert(r == 0); } snd_config_update_free_global(); return 0; } </snip> And there they are gone, the 10 bytes of memory. Here's a snippet from valgrinding this program: ==22198== 10 bytes in 1 blocks are definitely lost in loss record 1 of 3 ==22198== at 0x4023765: malloc (vg_replace_malloc.c:149) ==22198== by 0x417C44F: strdup (strdup.c:43) ==22198== by 0x4071954: snd_determine_driver (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406E80C: snd_config_hook_load_for_all_cards (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406EC7C: (within /usr/lib/libasound.so.2.0.0) ==22198== by 0x406F896: snd_config_searcha_hooks (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406FA74: snd_config_search_alias_hooks (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x406FCD9: snd_config_search_definition (in /usr/lib/libasound.so.2.0.0) ==22198== by 0x40952FE: (within /usr/lib/libasound.so.2.0.0) ==22198== by 0x80484D7: main (in /home/lennart/alsa-test) And there seem to be more. When PulseAudio starts up it probes for available audio devices. On my machine there are 5 sound cards, which are all probed with device strings like "surround51:0" and so on, and in the end some memory with a caller history like the one shown above is shown, and another one: ==22229== 235 (180 direct, 55 indirect) bytes in 5 blocks are definitely lost in loss record 6 of 6 ==22229== at 0x4023AA4: calloc (vg_replace_malloc.c:279) ==22229== by 0x46967B8: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x4696C48: snd_config_make (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x4696CD0: snd_config_imake_pointer (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46EA573: _snd_pcm_hook_ctl_elems_install (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46EAC10: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46EB2BC: _snd_pcm_hooks_open (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C0CD4: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C1467: snd_pcm_open_slave (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46D232B: _snd_pcm_route_open (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C0CD4: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C1467: snd_pcm_open_slave (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46E47CC: _snd_pcm_empty_open (in /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C0CD4: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x46C13D2: (within /usr/lib/libasound.so.2.0.0) ==22229== by 0x412E007: pa_alsa_open_by_device_id (alsa-util.c:529) ==22229== by 0x4135904: module_alsa_sink_LTX_pa__init (module-alsa-sink.c:781) ==22229== by 0x419CD2B: pa_module_load (module.c:115) ==22229== by 0x4164D33: hal_device_load_alsa (module-hal-detect.c:204) ==22229== by 0x4165313: hal_device_add (module-hal-detect.c:323) Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 6

[alsa-devel] snd_pcm_name() returns NULL is used on a surround51: device
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! snd_pcm_name() returns NULL if used shortly after opening a "surround51" device. ("surround51" is just an example, it returns NULL for a lot of other devices, too.) Also reported as bug in the BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3563 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

Re: [alsa-devel] How to cont the sound cards?
by Takashi Iwai 26 Nov '07

26 Nov '07

At Mon, 26 Nov 2007 12:53:52 +0200, Cosmin Humeniuc wrote: > > Thank you for the reply, Takashi. > > The problem is, I'm not sure what should I loop through. I saw this > list on the wiki ( > http://www.alsa-project.org/main/index.php/DeviceNames ) but I don't > know how it was obtained. The code would be like: int card = -1; while (!snd_card_next(&card) && card >= 0) { ... } Takashi > > Cosmin > > On 11/23/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 19 Nov 2007 09:12:51 +0200, > > Cosmin Humeniuc wrote: > > > > > > Hi, > > > > > > I am trying to develop an application using the ALSA library. I read > > > in a book about the snd_cards function, but it doesn't seem to be > > > defined anywhere. (The book is rather old, though, it talks about the > > > 0.5.x series). > > > > > > The compiler doesn't complain, but the linker gives me an "undefined > > > reference to `snd_cards'" error. I assume this function is deprecated, > > > but what can I use instead? > > > > Oh that myth. I myself already forgot about that function :) > > > > On ALSA 1.x, there is no corresponding function. You'll need to > > loop and check the presence of each card. > > > > > > Takashi > > >

1 0

[alsa-devel] [PATCH] Use different buffer metrics in the PulseAudio plugin
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! Please commit this patch to alsa-plugins we ship in Fedora: http://cvs.fedoraproject.org/viewcvs/*checkout*/devel/alsa-plugins/1.0.14-b… It increases the "pre-buffering level" (i.e. start threshold) to the full buffer size minus one period. This makes PA work a little bit more like normal audio devices, and makes a few drop outs go away for software which uses very small period sizes. It also increases the initial maximum buffer size, which allows a small overcommit. That's not really an issue, but cleaner nonetheless so I smuggled it into this patch. Also reported in the ALSA BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3578 Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

2 1

[alsa-devel] If /dev/snd/controlCx EACCES vs. ENODEV mixup
by Lennart Poettering 26 Nov '07

26 Nov '07

Hi! If /dev/snd/controlC0 is not accessible due to a permission problem (EACCES), then alsa-libs will lie and return ENODEV. This is quite a bit confusing: <snip> $ sudo chmod 000 /dev/snd/controlC0 $ aplay -D hw:0 ALSA lib pcm_hw.c:1207:(_snd_pcm_hw_open) Invalid value for card aplay: main:546: audio open error: No such device </snip> (and strace shows that the actual problem is EACESS, as one would assume. I also filed this as bug on the BTS: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3600 BTW, what's the status of the BTS? Takashi asked me to post all my issues on the ML instead of on the BTS. So why have the BTS at all? It's even linked on the alsa-project.org front page under "I found a bug!", which is a bit misleading. If this Mantis thing is not liked at all, would it be possible to make some replacement available? Maybe just use the kernel bugzilla on bugzilla.kernel.org? It's very fast, and certainly more fun to work with than with Mantis. Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net ICQ# 11060553 http://0pointer.net/lennart/ GnuPG 0x1A015CC4

3 9

[alsa-devel] [PATCH] add support for Kore controller 2
by Daniel Mack 26 Nov '07

26 Nov '07

Added support for Native Instrument's Kore controller 2. This device has no audio but MIDI, input devices and ALSA controllers only. Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

2 1

[alsa-devel] [PATCH] soc/wm8731: Fix stereo mixer controls
by Ville Syrjälä 26 Nov '07

26 Nov '07

soc/wm8731: Fix stereo mixer controls Disable the simultaneous load feature for the line in and headphone out volume registers. This allows left and right volume levels to be controlled separately. Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 7ca0b52..57fb95a 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -562,13 +562,13 @@ static int wm8731_init(struct snd_soc_device *socdev) /* set the update bits */ reg = wm8731_read_reg_cache(codec, WM8731_LOUT1V); - wm8731_write(codec, WM8731_LOUT1V, reg | 0x0100); + wm8731_write(codec, WM8731_LOUT1V, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_ROUT1V); - wm8731_write(codec, WM8731_ROUT1V, reg | 0x0100); + wm8731_write(codec, WM8731_ROUT1V, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_LINVOL); - wm8731_write(codec, WM8731_LINVOL, reg | 0x0100); + wm8731_write(codec, WM8731_LINVOL, reg & ~0x0100); reg = wm8731_read_reg_cache(codec, WM8731_RINVOL); - wm8731_write(codec, WM8731_RINVOL, reg | 0x0100); + wm8731_write(codec, WM8731_RINVOL, reg & ~0x0100); wm8731_add_controls(codec); wm8731_add_widgets(codec); -- Ville Syrjälä syrjala(a)sci.fi http://www.sci.fi/~syrjala/

2 1

[alsa-devel] Audigy LS driver
by Daniel Sabo 26 Nov '07

26 Nov '07

Hi, I'm interested in helping with the ca0106 driver. My understanding is that the dmix and multi plugins can't be used with each other, which means that right now it not possible to have a dmixed surround sound device with this card. The driver has 4 stereo devices on it (one for each DSP on the board?), would it be possible for me to merge devices 0,1, and 2 into a single 6 channel device? I have some experience with driver development doing ARM SoC stuff, but this is my first time messing with a linux driver. Thanks, - Daniel Sabo

2 1

[alsa-devel] Semantics of a single period per buffer?
by Dave Dillow 25 Nov '07

25 Nov '07

While refining the sis7019 for submission, I found a possible issue with having a single period per buffer. The easy way out is to set the minimum number of periods to 2, but I see other drivers allow a single period. This leads to a question about the semantics of a single period that I've not been able to resolve using Google. Any one have any thoughts or pointers to a discussion on having a single period per buffer? Should it be a one-shot play, or just let it hit XRUN every time? Other ideas? Thanks, Dave

2 2

[alsa-devel] Inappropriate ioctl for device on alsalib with uclibc and arm platform
by Federico Fuga 23 Nov '07

23 Nov '07

Hi, Maybe I am a little bit off-topic here, but I think this question may be related with a bug (maybe not in the alsa library, but I am a little confused now), so I apologize in advance. I am building a uclibc-based system on arm board. I have compiled the library 1.0.14a with uClibc 0.9.29 and kernel 2.6.23.1 / kernel headers 2.6.22. Settings the config params of mixer with "alsactl restore" fails with an "inappropriate ioctl for device" error: # alsactl restore alsactl: set_control:1159: Cannot write control '2:0:0:Speaker Playback Volume:0' : Inappropriate ioctl for device while traceing with strace I see that: open("/usr/lib/libasound.so.2", O_RDONLY) = 4 fstat(4, {st_mode=S_IFREG|0755, st_size=716844, ...}) = 0 close(4) = 0 open("/dev/snd/controlC0", O_RDONLY) = 4 close(4) = 0 open("/dev/snd/controlC0", O_RDWR) = 4 ioctl(4, USBDEVFS_CONTROL, 0xbeab09ec) = 0 ioctl(4, UI_DEV_CREATE, 0xbeab0ad0) = 0 ioctl(4, USBDEVFS_CONNECTINFO, 0xbeab0654) = 0 ioctl(4, USBDEVFS_HUB_PORTINFO, 0xbeab0768) = -1 ENOTTY (Inappropriate ioctl for device) write(2, "alsactl", 7alsactl) = 7 but aplay see all the board and seems to work properly: # aplay -l **** List of PLAYBACK Hardware Devices **** card 0: EMX270 [EM-X270], device 0: AC97 HiFi AC97 HiFi-AC97-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: EMX270 [EM-X270], device 1: AC97 Aux AC97 Aux-AC97-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 # aplay -L default:CARD=EMX270 EM-X270, Default Audio Device null Discard all samples (playback) or generate zero samples (capture) # speaker-test speaker-test 1.0.14 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 8 to 32768 Period size range from 8 to 2040 Using max buffer size 32768 Periods = 4 was set period_size = 1024 was set buffer_size = 32768 0 - Front Left Time per period = 2.528627 0 - Front Left Time per period = 2.999174 But I hear no sound (the levels are all off). In a similar installation based on glibc (Armstrong linux) with the same version of kernel, alsa and all other stuff works correctly. root@em-x270:~$ cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 33: : timer The device nodes in /dev/sound are created by udev, and major/minor codes and permissions seems ok. Can you give me some hint on how to solve this issue? Thank you in advance ing. Federico Fuga

2 4

[alsa-devel] How to cont the sound cards?
by Cosmin Humeniuc 23 Nov '07

23 Nov '07

Hi, I am trying to develop an application using the ALSA library. I read in a book about the snd_cards function, but it doesn't seem to be defined anywhere. (The book is rather old, though, it talks about the 0.5.x series). The compiler doesn't complain, but the linker gives me an "undefined reference to `snd_cards'" error. I assume this function is deprecated, but what can I use instead? Cosmin

2 1

[alsa-devel] Lock free ring buffers and SMP safety
by Matt Bucknall 23 Nov '07

23 Nov '07

Hello, Yes, this old chestnut. I've spent the best part of today trying to ascertain whether or not, in practical terms, a single-producer, single-consumer lock free ring-buffer can be written purely in C/C++ and be SMP safe on multiple architectures without memory barriers. Information I've found has been largely conflicting so I've given up trying to get a conclusive answer. Instead I'll just go with the flow. What is the general consensus amongst ALSA application developers when it comes to implementing this kind of buffer? Is it possible to use memory barriers through some user space API call that I don't know about? Thanks in advance for any comments/suggestions. Matt.

2 1

Re: [alsa-devel] Mbox II DigiDisign
by Clemens Ladisch 23 Nov '07

23 Nov '07

rmgls(a)free.fr wrote: > on jeudi 22 novembre 2007 Clemens Ladisch" <cladisch(a)fastmail.net wrote: > >> aplay plays nothing and reports: (alsa-lib pcm error). > > > What exactly does it report? > > it reports > > segfault! Does this also happen when you run "aplay -D plughw coq.wav"? > Program received signal SIGSEGV, Segmentation fault. > [Switching to Thread 0xb7cfb6b0 (LWP 21167)] > 0xb7f0389c in ?? () from /usr/lib/libasound.so.2 > (gdb) bt > #0 0xb7f0389c in ?? () from /usr/lib/libasound.so.2 > #1 0x00038007 in ?? () > ... > > if needed i'll build a non stripped aplay. That would be helpful. Regards, Clemens

1 0

[alsa-devel] [PATCH] ASoC: S3C2412 IIS driver
by Ben Dooks 23 Nov '07

23 Nov '07

S3C2412 SoC IIS support for ALSA/ASoC Signed-off-by: Ben Dooks <ben-linux(a)fluff.org> diff -urpN -X linux-2.6.24-rc3/Documentation/dontdiff linux-2.6.24-rc3/sound/soc/s3c24xx/Kconfig linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/Kconfig --- linux-2.6.24-rc3/sound/soc/s3c24xx/Kconfig 2007-11-19 16:12:18.000000000 +0000 +++ linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/Kconfig 2007-11-19 21:21:57.000000000 +0000 @@ -10,6 +10,9 @@ config SND_S3C24XX_SOC config SND_S3C24XX_SOC_I2S tristate +config SND_S3C2412_SOC_I2S + tristate + config SND_S3C2443_SOC_AC97 tristate select AC97_BUS diff -urpN -X linux-2.6.24-rc3/Documentation/dontdiff linux-2.6.24-rc3/sound/soc/s3c24xx/Makefile linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/Makefile --- linux-2.6.24-rc3/sound/soc/s3c24xx/Makefile 2007-10-09 21:31:38.000000000 +0100 +++ linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/Makefile 2007-11-19 21:21:57.000000000 +0000 @@ -1,11 +1,13 @@ # S3c24XX Platform Support snd-soc-s3c24xx-objs := s3c24xx-pcm.o snd-soc-s3c24xx-i2s-objs := s3c24xx-i2s.o +snd-soc-s3c2412-i2s-objs := s3c2412-i2s.o snd-soc-s3c2443-ac97-objs := s3c2443-ac97.o obj-$(CONFIG_SND_S3C24XX_SOC) += snd-soc-s3c24xx.o obj-$(CONFIG_SND_S3C24XX_SOC_I2S) += snd-soc-s3c24xx-i2s.o obj-$(CONFIG_SND_S3C2443_SOC_AC97) += snd-soc-s3c2443-ac97.o +obj-$(CONFIG_SND_S3C2412_SOC_I2S) += snd-soc-s3c2412-i2s.o # S3C24XX Machine Support snd-soc-neo1973-wm8753-objs := neo1973_wm8753.o diff -urpN -X linux-2.6.24-rc3/Documentation/dontdiff linux-2.6.24-rc3/sound/soc/s3c24xx/s3c2412-i2s.c linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/s3c2412-i2s.c --- linux-2.6.24-rc3/sound/soc/s3c24xx/s3c2412-i2s.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/s3c2412-i2s.c 2007-11-19 21:28:16.000000000 +0000 @@ -0,0 +1,686 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2006 Wolfson Microelectronics PLC. + * Graeme Gregory graeme.gregory(a)wolfsonmicro.com + * linux(a)wolfsonmicro.com + * + * Copyright (c) 2007, 2004-2005 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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. +*/ + +#include <linux/init.h> +#include <linux/module.h> +#include <linux/device.h> +#include <linux/delay.h> +#include <linux/clk.h> + +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/soc.h> +#include <asm/hardware.h> + +#include <asm/io.h> +#include <asm/dma.h> + +#include <asm/plat-s3c24xx/regs-s3c2412-iis.h> + +#include <asm/arch/regs-gpio.h> +#include <asm/arch/audio.h> +#include <asm/arch/dma.h> + +#include "s3c24xx-pcm.h" +#include "s3c2412-i2s.h" + +#define S3C2412_I2S_DEBUG 0 +#define S3C2412_I2S_DEBUG_CON 0 + +#if S3C2412_I2S_DEBUG +#define DBG(x...) printk(KERN_INFO x) +#else +#define DBG(x...) do { } while (0) +#endif + +static struct s3c2410_dma_client s3c2412_dma_client_out = { + .name = "I2S PCM Stereo out" +}; + +static struct s3c2410_dma_client s3c2412_dma_client_in = { + .name = "I2S PCM Stereo in" +}; + +static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_out = { + .client = &s3c2412_dma_client_out, + .channel = DMACH_I2S_OUT, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .dma_size = 4, +}; + +static struct s3c24xx_pcm_dma_params s3c2412_i2s_pcm_stereo_in = { + .client = &s3c2412_dma_client_in, + .channel = DMACH_I2S_IN, + .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .dma_size = 4, +}; + +struct s3c2412_i2s_info { + struct device *dev; + void __iomem *regs; + struct clk *iis_clk; + struct clk *iis_pclk; + struct clk *iis_cclk; +}; + +static struct s3c2412_i2s_info s3c2412_i2s; + +#define bit_set(v,b) (((v) & (b)) ? 1 : 0) + +#if S3C2412_I2S_DEBUG_CON +static void dbg_showcon(const char *fn, u32 con) +{ + printk(KERN_DEBUG "%s: LRI=%d, TXFEMPT=%d, RXFEMPT=%d, TXFFULL=%d, RXFFULL=%d\n", fn, + bit_set(con, S3C2412_IISCON_LRINDEX), + bit_set(con, S3C2412_IISCON_TXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_RXFIFO_EMPTY), + bit_set(con, S3C2412_IISCON_TXFIFO_FULL), + bit_set(con, S3C2412_IISCON_RXFIFO_FULL)); + + printk(KERN_DEBUG "%s: PAUSE: TXDMA=%d, RXDMA=%d, TXCH=%d, RXCH=%d\n", + fn, + bit_set(con, S3C2412_IISCON_TXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_RXDMA_PAUSE), + bit_set(con, S3C2412_IISCON_TXCH_PAUSE), + bit_set(con, S3C2412_IISCON_RXCH_PAUSE)); + printk(KERN_DEBUG "%s: ACTIVE: TXDMA=%d, RXDMA=%d, IIS=%d\n", fn, + bit_set(con, S3C2412_IISCON_TXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_RXDMA_ACTIVE), + bit_set(con, S3C2412_IISCON_IIS_ACTIVE)); +} +#else +static inline void dbg_showcon(const char *fn, u32 con) +{ +} +#endif + +/* Turn on or off the transmission path. */ +static void s3c2412_snd_txctrl(int on) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + DBG("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_TXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_TXDMA_PAUSE; + con &= ~S3C2412_IISCON_TXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXONLY: + case S3C2412_IISMOD_MODE_TXRX: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_RXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "TXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } else { + /* Note, we do not have any indication that the FIFO problems + * tha the S3C2410/2440 had apply here, so we should be able + * to disable the DMA and TX without resetting the FIFOS. + */ + + con |= S3C2412_IISCON_TXDMA_PAUSE; + con |= S3C2412_IISCON_TXCH_PAUSE; + con &= ~S3C2412_IISCON_TXDMA_ACTIVE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_RXONLY; + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + con &= ~S3C2412_IISCON_IIS_ACTIVE; + break; + + default: + dev_err(i2s->dev, "TXDIS: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } + + fic = readl(regs + S3C2412_IISFIC); + dbg_showcon(__func__, con); + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + +static void s3c2412_snd_rxctrl(int on) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + void __iomem *regs = i2s->regs; + u32 fic, con, mod; + + DBG("%s(%d)\n", __func__, on); + + fic = readl(regs + S3C2412_IISFIC); + con = readl(regs + S3C2412_IISCON); + mod = readl(regs + S3C2412_IISMOD); + + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); + + if (on) { + con |= S3C2412_IISCON_RXDMA_ACTIVE | S3C2412_IISCON_IIS_ACTIVE; + con &= ~S3C2412_IISCON_RXDMA_PAUSE; + con &= ~S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_TXRX: + case S3C2412_IISMOD_MODE_RXONLY: + /* do nothing, we are in the right mode */ + break; + + case S3C2412_IISMOD_MODE_TXONLY: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXRX; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(mod, regs + S3C2412_IISMOD); + writel(con, regs + S3C2412_IISCON); + } else { + /* See txctrl notes on FIFOs. */ + + con &= ~S3C2412_IISCON_RXDMA_ACTIVE; + con |= S3C2412_IISCON_RXDMA_PAUSE; + con |= S3C2412_IISCON_RXCH_PAUSE; + + switch (mod & S3C2412_IISMOD_MODE_MASK) { + case S3C2412_IISMOD_MODE_RXONLY: + con &= ~S3C2412_IISCON_IIS_ACTIVE; + mod &= ~S3C2412_IISMOD_MODE_MASK; + break; + + case S3C2412_IISMOD_MODE_TXRX: + mod &= ~S3C2412_IISMOD_MODE_MASK; + mod |= S3C2412_IISMOD_MODE_TXONLY; + break; + + default: + dev_err(i2s->dev, "RXEN: Invalid MODE in IISMOD\n"); + } + + writel(con, regs + S3C2412_IISCON); + writel(mod, regs + S3C2412_IISMOD); + } + + fic = readl(regs + S3C2412_IISFIC); + DBG("%s: IIS: CON=%x MOD=%x FIC=%x\n", __func__, con, mod, fic); +} + + +/* + * Wait for the LR signal to allow synchronisation to the L/R clock + * from the codec. May only be needed for slave mode. + */ +static int s3c2412_snd_lrsync(void) +{ + u32 iiscon; + unsigned long timeout = jiffies + msecs_to_jiffies(5); + + DBG("Entered %s\n", __func__); + + while (1) { + iiscon = readl(s3c2412_i2s.regs + S3C2412_IISCON); + if (iiscon & S3C2412_IISCON_LRINDEX) + break; + + if (timeout < jiffies) { + printk(KERN_ERR "%s: timeout\n", __func__); + return -ETIMEDOUT; + } + } + + return 0; +} + +/* + * Check whether CPU is the master or slave + */ +static inline int s3c2412_snd_is_clkmaster(void) +{ + u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + + DBG("Entered %s\n", __func__); + + iismod &= S3C2412_IISMOD_MASTER_MASK; + return !(iismod == S3C2412_IISMOD_SLAVE); +} + +/* + * Set S3C2412 I2S DAI format + */ +static int s3c2412_i2s_set_fmt(struct snd_soc_cpu_dai *cpu_dai, + unsigned int fmt) +{ + u32 iismod; + + + DBG("Entered %s\n", __func__); + + iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("hw_params r: IISMOD: %x \n", iismod); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_SLAVE; + break; + case SND_SOC_DAIFMT_CBS_CFS: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_MASTER_INTERNAL; + break; + default: + DBG("unknwon master/slave format\n"); + return -EINVAL; + } + + iismod &= ~S3C2412_IISMOD_SDF_MASK; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + iismod |= S3C2412_IISMOD_SDF_MSB; + break; + case SND_SOC_DAIFMT_LEFT_J: + iismod |= S3C2412_IISMOD_SDF_LSB; + break; + case SND_SOC_DAIFMT_I2S: + iismod |= S3C2412_IISMOD_SDF_IIS; + break; + default: + DBG("Unknown data format\n"); + return -EINVAL; + } + + writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("hw_params w: IISMOD: %x \n", iismod); + return 0; +} + +static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + u32 iismod; + + DBG("Entered %s\n", __func__); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_out; + else + rtd->dai->cpu_dai->dma_data = &s3c2412_i2s_pcm_stereo_in; + + /* Working copies of register */ + iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("%s: r: IISMOD: %x\n", __func__, iismod); + + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S8: + iismod |= S3C2412_IISMOD_8BIT; + break; + case SNDRV_PCM_FORMAT_S16_LE: + iismod &= ~S3C2412_IISMOD_8BIT; + break; + } + + writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); + DBG("%s: w: IISMOD: %x\n", __func__, iismod); + return 0; +} + +static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd) +{ + int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); + unsigned long irqs; + int ret = 0; + + DBG("Entered %s\n", __func__); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + /* On start, ensure that the FIFOs are cleared and reset. */ + + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + s3c2412_i2s.regs + S3C2412_IISFIC); + + /* clear again, just in case */ + writel(0x0, s3c2412_i2s.regs + S3C2412_IISFIC); + + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (!s3c2412_snd_is_clkmaster()) { + ret = s3c2412_snd_lrsync(); + if (ret) + goto exit_err; + } + + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(1); + else + s3c2412_snd_txctrl(1); + + local_irq_restore(irqs); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + local_irq_save(irqs); + + if (capture) + s3c2412_snd_rxctrl(0); + else + s3c2412_snd_txctrl(0); + + local_irq_restore(irqs); + break; + default: + ret = -EINVAL; + break; + } + +exit_err: + return ret; +} + +/* default table of all avaialable root fs divisors */ +static unsigned int s3c2412_iis_fs[] = { 256, 512, 384, 768, 0 }; + +#define ABS(x) (((x) < 0) ? (-(x)) : (x)) + +int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk) +{ + unsigned long clkrate = clk_get_rate(clk); + unsigned int div; + unsigned int fsclk; + unsigned int actual; + unsigned int fs; + unsigned int fsdiv; + signed int deviation = 0; + unsigned int best_fs = 0; + unsigned int best_div = 0; + unsigned int best_rate = 0; + unsigned int best_deviation = INT_MAX; + + + if (fstab == NULL) + fstab = s3c2412_iis_fs; + + for (fs = 0;; fs++) { + fsdiv = s3c2412_iis_fs[fs]; + + if (fsdiv == 0) + break; + + fsclk = clkrate / fsdiv; + div = fsclk / rate; + + if ((fsclk % rate) > (rate / 2)) + div++; + + if (div <= 1) + continue; + + actual = clkrate / (fsdiv * div); + deviation = actual - rate; + + printk(KERN_DEBUG "%dfs: div %d => result %d, deviation %d\n", + fsdiv, div, actual, deviation); + + deviation = ABS(deviation); + + if (deviation < best_deviation) { + best_fs = fsdiv; + best_div = div; + best_rate = actual; + best_deviation = deviation; + } + + if (deviation == 0) + break; + } + + printk(KERN_DEBUG "best: fs=%d, div=%d, rate=%d\n", + best_fs, best_div, best_rate); + + info->fs_div = best_fs; + info->clk_div = best_div; + + return 0; +} + +EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate); + +/* + * Set S3C2412 Clock source + */ +static int s3c2412_i2s_set_sysclk(struct snd_soc_cpu_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + u32 iismod = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + + DBG("%s(%p, %d, %u, %d)\n", __func__, cpu_dai, clk_id, + freq, dir); + + switch (clk_id) { + case S3C2412_CLKSRC_PCLK: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_MASTER_INTERNAL; + break; + case S3C2412_CLKSRC_I2SCLK: + iismod &= ~S3C2412_IISMOD_MASTER_MASK; + iismod |= S3C2412_IISMOD_MASTER_EXTERNAL; + break; + default: + return -EINVAL; + } + + writel(iismod, s3c2412_i2s.regs + S3C2412_IISMOD); + return 0; +} + +/* + * Set S3C2412 Clock dividers + */ +static int s3c2412_i2s_set_clkdiv(struct snd_soc_cpu_dai *cpu_dai, + int div_id, int div) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + u32 reg; + + DBG("%s(%p, %d, %d)\n", __func__, cpu_dai, div_id, div); + + switch (div_id) { + case S3C2412_DIV_BCLK: + reg = readl(i2s->regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_BCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + + DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C2412_DIV_RCLK: + if (div > 3) { + /* convert value to bit field */ + + switch (div) { + case 256: + div = S3C2412_IISMOD_RCLK_256FS; + break; + + case 384: + div = S3C2412_IISMOD_RCLK_384FS; + break; + + case 512: + div = S3C2412_IISMOD_RCLK_512FS; + break; + + case 768: + div = S3C2412_IISMOD_RCLK_768FS; + break; + + default: + return -EINVAL; + } + } + + reg = readl(s3c2412_i2s.regs + S3C2412_IISMOD); + reg &= ~S3C2412_IISMOD_RCLK_MASK; + writel(reg | div, i2s->regs + S3C2412_IISMOD); + DBG("%s: MOD=%08x\n", __func__, readl(i2s->regs + S3C2412_IISMOD)); + break; + + case S3C2412_DIV_PRESCALER: + if (div >= 0) { + writel((div << 8) | S3C2412_IISPSR_PSREN, + i2s->regs + S3C2412_IISPSR); + } else { + writel(0x0, i2s->regs + S3C2412_IISPSR); + } + DBG("%s: PSR=%08x\n", __func__, readl(i2s->regs + S3C2412_IISPSR)); + break; + + default: + return -EINVAL; + } + + return 0; +} + +struct clk *s3c2412_get_iisclk(void) +{ + return s3c2412_i2s.iis_clk; +} + +EXPORT_SYMBOL_GPL(s3c2412_get_iisclk); + + +static int s3c2412_i2s_probe(struct platform_device *pdev) +{ + DBG("Entered %s\n", __func__); + + s3c2412_i2s.dev = &pdev->dev; + + s3c2412_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s.iis_pclk=clk_get(&pdev->dev, "iis"); + if (s3c2412_i2s.iis_pclk == NULL) { + DBG("failed to get iis_clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + s3c2412_i2s.iis_cclk=clk_get(&pdev->dev, "i2sclk"); + if (s3c2412_i2s.iis_cclk == NULL) { + DBG("failed to get i2sclk clock\n"); + iounmap(s3c2412_i2s.regs); + return -ENODEV; + } + + clk_set_parent(s3c2412_i2s.iis_cclk, clk_get(NULL, "mpll")); + + clk_enable(s3c2412_i2s.iis_pclk); + clk_enable(s3c2412_i2s.iis_cclk); + + s3c2412_i2s.iis_clk = s3c2412_i2s.iis_pclk; + + /* Configure the I2S pins in correct mode */ + s3c2410_gpio_cfgpin(S3C2410_GPE0, S3C2410_GPE0_I2SLRCK); + s3c2410_gpio_cfgpin(S3C2410_GPE1, S3C2410_GPE1_I2SSCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE2, S3C2410_GPE2_CDCLK); + s3c2410_gpio_cfgpin(S3C2410_GPE3, S3C2410_GPE3_I2SSDI); + s3c2410_gpio_cfgpin(S3C2410_GPE4, S3C2410_GPE4_I2SSDO); + + s3c2412_snd_txctrl(0); + s3c2412_snd_rxctrl(0); + + return 0; +} + +#define S3C2412_I2S_RATES \ + (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ + SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000) + +struct snd_soc_cpu_dai s3c2412_i2s_dai = { + .name = "s3c2412-i2s", + .id = 0, + .type = SND_SOC_DAI_I2S, + .probe = s3c2412_i2s_probe, + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = S3C2412_I2S_RATES, + .formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = { + .trigger = s3c2412_i2s_trigger, + .hw_params = s3c2412_i2s_hw_params, + }, + .dai_ops = { + .set_fmt = s3c2412_i2s_set_fmt, + .set_clkdiv = s3c2412_i2s_set_clkdiv, + .set_sysclk = s3c2412_i2s_set_sysclk, + }, +}; + +EXPORT_SYMBOL_GPL(s3c2412_i2s_dai); + +/* Module information */ +MODULE_AUTHOR("Ben Dooks, <ben(a)simtec.co.uk>"); +MODULE_DESCRIPTION("S3C2412 I2S SoC Interface"); +MODULE_LICENSE("GPL"); diff -urpN -X linux-2.6.24-rc3/Documentation/dontdiff linux-2.6.24-rc3/sound/soc/s3c24xx/s3c2412-i2s.h linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/s3c2412-i2s.h --- linux-2.6.24-rc3/sound/soc/s3c24xx/s3c2412-i2s.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/s3c2412-i2s.h 2007-11-19 21:21:57.000000000 +0000 @@ -0,0 +1,38 @@ +/* sound/soc/s3c24xx/s3c2412-i2s.c + * + * ALSA Soc Audio Layer - S3C2412 I2S driver + * + * Copyright (c) 2007 Simtec Electronics + * http://armlinux.simtec.co.uk/ + * Ben Dooks <ben(a)simtec.co.uk> + * + * 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. +*/ + +#ifndef __SND_SOC_S3C24XX_S3C2412_I2S_H +#define __SND_SOC_S3C24XX_S3C2412_I2S_H __FILE__ + +#define S3C2412_DIV_BCLK (1) +#define S3C2412_DIV_RCLK (2) +#define S3C2412_DIV_PRESCALER (3) + +#define S3C2412_CLKSRC_PCLK (0) +#define S3C2412_CLKSRC_I2SCLK (1) + +extern struct clk *s3c2412_get_iisclk(void); + +extern struct snd_soc_cpu_dai s3c2412_i2s_dai; + +struct s3c2412_rate_calc { + unsigned int clk_div; /* for prescaler */ + unsigned int fs_div; /* for root frame clock */ +}; + +extern int s3c2412_iis_calc_rate(struct s3c2412_rate_calc *info, + unsigned int *fstab, + unsigned int rate, struct clk *clk); + +#endif /* __SND_SOC_S3C24XX_S3C2412_I2S_H */

3 2

[alsa-devel] S3C2412: suspend and resume support
by Ben Dooks 23 Nov '07

23 Nov '07

Support for suspend/resume for the S3C2412 ASoC IIS core driver. Signed-off-by: Ben Dooks <ben-linux(a)fluff.org> diff -urpN -X linux-2.6.24-rc3-asoc1/Documentation/dontdiff linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/s3c2412-i2s.c linux-2.6.24-rc3-asoc2/sound/soc/s3c24xx/s3c2412-i2s.c --- linux-2.6.24-rc3-asoc1/sound/soc/s3c24xx/s3c2412-i2s.c 2007-11-19 21:28:16.000000000 +0000 +++ linux-2.6.24-rc3-asoc2/sound/soc/s3c24xx/s3c2412-i2s.c 2007-11-20 12:29:46.000000000 +0000 @@ -79,6 +79,10 @@ struct s3c2412_i2s_info { struct clk *iis_clk; struct clk *iis_pclk; struct clk *iis_cclk; + + u32 suspend_iismod; + u32 suspend_iiscon; + u32 suspend_iispsr; }; static struct s3c2412_i2s_info s3c2412_i2s; @@ -381,7 +385,7 @@ static int s3c2412_i2s_trigger(struct sn case SNDRV_PCM_TRIGGER_START: /* On start, ensure that the FIFOs are cleared and reset. */ - writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH, + writel(capture ? S3C2412_IISFIC_RXFLUSH : S3C2412_IISFIC_TXFLUSH , s3c2412_i2s.regs + S3C2412_IISFIC); /* clear again, just in case */ @@ -645,6 +649,63 @@ static int s3c2412_i2s_probe(struct plat return 0; } +#ifdef CONFIG_PM +static int s3c2412_i2s_suspend(struct platform_device *dev, + struct snd_soc_cpu_dai *dai) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + u32 iismod; + + if (dai->active) { + i2s->suspend_iismod = readl(i2s->regs + S3C2412_IISMOD); + i2s->suspend_iiscon = readl(i2s->regs + S3C2412_IISCON); + i2s->suspend_iispsr = readl(i2s->regs + S3C2412_IISPSR); + + /* some basic suspend checks */ + + iismod = readl(i2s->regs + S3C2412_IISMOD); + + if (iismod & S3C2412_IISCON_RXDMA_ACTIVE) + dev_warn(&dev->dev, "%s: RXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_TXDMA_ACTIVE) + dev_warn(&dev->dev, "%s: TXDMA active?\n", __func__); + + if (iismod & S3C2412_IISCON_IIS_ACTIVE) + dev_warn(&dev->dev, "%s: IIS active\n", __func__); + } + + return 0; +} + +static int s3c2412_i2s_resume(struct platform_device *pdev, + struct snd_soc_cpu_dai *dai) +{ + struct s3c2412_i2s_info *i2s = &s3c2412_i2s; + + dev_info(&pdev->dev, "dai_active %d, IISMOD %08x, IISCON %08x\n", + dai->active, i2s->suspend_iismod, i2s->suspend_iiscon); + + if (dai->active) { + writel(i2s->suspend_iiscon, i2s->regs + S3C2412_IISCON); + writel(i2s->suspend_iismod, i2s->regs + S3C2412_IISMOD); + writel(i2s->suspend_iispsr, i2s->regs + S3C2412_IISPSR); + + writel(S3C2412_IISFIC_RXFLUSH | S3C2412_IISFIC_TXFLUSH, + i2s->regs + S3C2412_IISFIC); + + ndelay(250); + writel(0x0, i2s->regs + S3C2412_IISFIC); + + } + + return 0; +} +#else +#define s3c2412_i2s_suspend NULL +#define s3c2412_i2s_resume NULL +#endif /* CONFIG_PM */ + #define S3C2412_I2S_RATES \ (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 | SNDRV_PCM_RATE_16000 | \ SNDRV_PCM_RATE_22050 | SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ @@ -655,6 +716,8 @@ struct snd_soc_cpu_dai s3c2412_i2s_dai = .id = 0, .type = SND_SOC_DAI_I2S, .probe = s3c2412_i2s_probe, + .suspend= s3c2412_i2s_suspend, + .resume = s3c2412_i2s_resume, .playback = { .channels_min = 2, .channels_max = 2,

4 3

[alsa-devel] [PATCH] add number of periods constraint to snd-aoa [try 2]
by Heikki O Lindholm 23 Nov '07

23 Nov '07

From: Heikki Lindholm <holindho(a)cs.helsinki.fi> The aoa driver is not specifying constraints on number of periods, and, it seems, it might end with a non-integer number, which it cannot deal with. Fix by adding a proper constraint. Signed-off-by: Heikki Lindholm <holindho(a)cs.helsinki.fi> --- try 2: previous try didn't work on anything but Onyx diff -r 1b54a8725ded aoa/soundbus/i2sbus/i2sbus-pcm.c --- a/aoa/soundbus/i2sbus/i2sbus-pcm.c Wed Nov 14 17:07:17 2007 +0100 +++ b/aoa/soundbus/i2sbus/i2sbus-pcm.c Fri Nov 23 11:53:38 2007 +0200 @@ -194,6 +194,12 @@ static int i2sbus_pcm_open(struct i2sbus hw->period_bytes_max = 16384; hw->periods_min = 3; hw->periods_max = MAX_DBDMA_COMMANDS; + if ((err = snd_pcm_hw_constraint_integer( + pi->substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS)) < 0) { + result = err; + goto out_unlock; + } list_for_each_entry(cii, &sdev->codec_list, list) { if (cii->codec->open) { err = cii->codec->open(cii, pi->substream);

2 1

[alsa-devel] [PATCH] add number of periods constraint to snd-aoa
by Heikki O Lindholm 23 Nov '07

23 Nov '07

From: Heikki Lindholm <holindho(a)cs.helsinki.fi> The aoa driver is not specifying constraints on number of periods, and, it seems, it might end with a non-integer number, which it cannot deal with. Fix by adding a proper constraint. Signed-off-by: Heikki Lindholm <holindho(a)cs.helsinki.fi> --- diff -r 1b54a8725ded aoa/soundbus/i2sbus/i2sbus-pcm.c --- a/aoa/soundbus/i2sbus/i2sbus-pcm.c Wed Nov 14 17:07:17 2007 +0100 +++ b/aoa/soundbus/i2sbus/i2sbus-pcm.c Fri Nov 23 11:35:17 2007 +0200 @@ -196,7 +196,10 @@ static int i2sbus_pcm_open(struct i2sbus hw->periods_max = MAX_DBDMA_COMMANDS; list_for_each_entry(cii, &sdev->codec_list, list) { if (cii->codec->open) { - err = cii->codec->open(cii, pi->substream); + if ((err = cii->codec->open(cii, pi->substream)) == 0) + err = snd_pcm_hw_constraint_integer( + pi->substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); if (err) { result = err; /* unwind */

2 1

[alsa-devel] test through vger #1 - please ignore
by Jaroslav Kysela 23 Nov '07

23 Nov '07

please ignore ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project - To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

1 0

[alsa-devel] test 3 - please ignore
by Jaroslav Kysela 23 Nov '07

23 Nov '07

test 3 - please ignore ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

1 0

[alsa-devel] Mbox II DigiDisign
by rmgls＠free.fr 22 Nov '07

22 Nov '07

hello all, Can you tell me please if the external usb audio interface mbox 2 from digidisign is supported by alsa? Here is my config, sony laptop, with an integrated alc260 chip (snd-hda). and the mbox on debian. now the different debian messages: dmesg ... usb 2-1: new full speed USB device using uhci_hcd and address 3 usb 2-1: config 1 interface 2 has no altsetting 1 usb 2-1: config 1 interface 3 has no altsetting 1 usb 2-1: config 1 interface 4 has no altsetting 1 usb 2-1: config 1 interface 5 has no altsetting 1 usb 2-1: configuration #1 chosen from 1 choice usbcore: registered new interface driver snd-usb-audio usb 2-1: USB disconnect, address 3 usb 2-1: new full speed USB device using uhci_hcd and address 4 usb 2-1: config 1 interface 2 has no altsetting 1 usb 2-1: config 1 interface 3 has no altsetting 1 usb 2-1: config 1 interface 4 has no altsetting 1 usb 2-1: config 1 interface 5 has no altsetting 1 usb 2-1: configuration #1 chosen from 1 choice cat /dev/sndstat Sound Driver:3.8.1a-980706 (ALSA v1.0.15 emulation code) Kernel: Linux mydomain.music 2.6.22-2-686 #1 SMP Fri Aug 31 00:24:01 UTC 2007 i686 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: HDA Intel at 0x80000000 irq 19 Digidesign Mbox 2 at usb-0000:00:1d.1-1, full speed Audio devices: 0: ALC260 Analog (DUPLEX) 1: USB Audio (DUPLEX) Synth devices: NOT ENABLED IN CONFIG Midi devices: NOT ENABLED IN CONFIG Timers: 7: system timer Mixers: 0: Realtek ALC260 1: USB Mixer aplay plays nothing and reports: (alsa-lib pcm error). alsamixer -c 1 reports: no mixer elem found. a google search does not provide no interesting technical sheet, at least i did not found any. have you something please: datasheets, config? thanks in advance. Best regards rmgls(a)free.fr

2 1

Re: [alsa-devel] Broken playback
by Clemens Ladisch 22 Nov '07

22 Nov '07

Gabriele Melia wrote: > > snd_pcm_hw_params_set_buffer_size_near(), but the allowed values are > > hardware dependent. You might want to add your own circular buffer that > > is bigger than ALSA's. > > So your suggestion is to limit my buffer_size after decode to be same of > ALSA's? No; my suggestion is to use your own buffer which is big enough for your decoder, and use this buffer to fill ALSA's buffer, whenever possible. If one call to your decoder function needs so much time that ALSA's buffer can become emptied, you need another thread to re-fill the ALSA buffer in parallel with the decoding. Regards, Clemens

1 0

[alsa-devel] alsa-devel on vger
by Takashi Iwai 22 Nov '07

22 Nov '07

Hi, What about the plan to move alsa-devel to vger? Shouldn't we do that soon? Takashi

6 5

[alsa-devel] [PATCH 2/2] caiaq/NativeInstruments: control support
by Daniel Mack 21 Nov '07

21 Nov '07

Hi, This patch adds support for input controllers on "Kore controller" and uses ALSA's control API to enable LED switching on several hardware products as well as some internal audio related functions of "Audio 8 DJ". Subject: SOUND: caiaq - add control API and more input features From: Daniel Mack <daniel(a)caiaq.de> SOUND: caiaq - add control API and more input features - added support for all input controllers on Native Instrument's "Kore controller". - added ALSA controls to switch LEDs on "RigKontrol 2", "RigKontrol3", "Audio Kontrol 1" and "Kore controller". - added ALSA controls to switch input mode, software lock and ground lift features on "Audio 8 DJ". Signed-off-by: Daniel Mack <daniel(a)caiaq.de>

2 3

[alsa-devel] [PATCH 1/2] caiaq/NativeInstruments: input handling fixes
by Daniel Mack 21 Nov '07

21 Nov '07

Hi, I'm re-posting a patch from Dmitry Torokhov which cleans up some input layer related things. It hasn't made its way to any repository yet but my subsequent patch needs it as dependency. I Cc'ed Dmitry to avoid confusion. Subject: SOUND: caiaq - misc input handling fixes From: Dmitry Torokhov <dmitry.torokhov(a)gmail.com> SOUND: caiaq - misc input handling fixes - link input device with its parent so that it placed in proper spot in sysfs hierarchy - drivers that allow changing their keymaps should use private copy of the keymap so that one instance of a device does not affect another instance - it is preferred for drivers to properly set up input_dev->phys to help userspace locate devices - drivers should use usb_to_input_id(), or perform endianess conversion, themselves, otherwise ID is not correct on big-endian boxes - whitespace and formatting cleanup Signed-off-by: Dmitry Torokhov <dtor(a)mail.ru> Acked-by: Daniel Mack <daniel(a)caiaq.de>

3 5

[alsa-devel] ASoC merge
by Mark Brown 21 Nov '07

21 Nov '07

The following patch series merges some of the outstanding changes in the linux-2.6-asoc tree. It updates the core and existing drivers and adds some of the new drivers currently in the tree.

9 62

Re: [alsa-devel] Broken playback
by Clemens Ladisch 21 Nov '07

21 Nov '07

Gabriele Melia wrote: > > Try non-blocking mode (see the snd_pcm_nonblock function). In non- > > blocking mode, the writei call will only write as much data as can be > > written without waiting (or returns -EAGAIN if the buffer is already > > full at the moment). > > I'm already in this case Then the problem is not that the writei call takes too long. An underrun can happen only when your code takes too much time between two consecutive calls to writei. > How to set the buffer size of the circular buffer in which i store the > pcm that is then called by alsa? snd_pcm_hw_params_set_buffer_size_near(), but the allowed values are hardware dependent. You might want to add your own circular buffer that is bigger than ALSA's. HTH Clemens

1 0

Re: [alsa-devel] Broken playback
by Clemens Ladisch 21 Nov '07

21 Nov '07

Gabriele Melia wrote: > i have already tried to read and decode some data in order to accumulate > 1 second of data to prevent broken audio, > but nothing change, analysing the issue i have found that broken audio > comes when the snd_pcm_writei is called with a size that is 5000. So it > takes time to dispach all this size. Try non-blocking mode (see the snd_pcm_nonblock function). In non- blocking mode, the writei call will only write as much data as can be written without waiting (or returns -EAGAIN if the buffer is already full at the moment). HTH Clemens

1 0

[alsa-devel] Latency and timestamps
by Claudio Matsuoka 21 Nov '07

21 Nov '07

Hi, I'm adding latency control to an application and didn't find much documentation about the pcm status functions aside from a very brief description and the latency.c example. What exactly are the "trigger timestamp" and "now timestamp" returned by snd_pcm_status_get_trigger_tstamp() and snd_pcm_status_get_tstamp()? I collected both from my output pcm handle to find values (in tv_sec.tv_usec format) such as 1195518180.938427 938427.557702 1195518180.938441 938441.557702 1195518181.300395 300395.557702 1195518181.300441 300441.557702 where the "trigger" tv_sec matches the "now" tv_usec, and the "trigger" tv_usec varies in large steps. Also the example latency.c says: 00703 showinmax(in_max); 00704 if (p_tstamp.tv_sec == p_tstamp.tv_sec && 00705 p_tstamp.tv_usec == c_tstamp.tv_usec) 00706 printf("Hardware sync\n"); 00707 snd_pcm_drop(chandle); In line 704, shouldn't it be p_tstamp.tv_sec == c_tstamp.tv_sec? My final goal is to sync on-screen events with the sound output. I'd like to do something like storing output metadata associated with a certain timestamp and collecting it back based on the timestamp of the currently playing pcm data (like the OSS sequencer echoback events), is that possible, or is there a better way to do this?

3 6

[alsa-devel] latency issues
by paul blakeley 21 Nov '07

21 Nov '07

I am currently trying to measure the latency through my alsa application. I will briefly explain my application and set-up. The application has a test harness which sends audio data to my application. To measure the latency the test harness reads in a .wav file(one which I have generated. This contains a short pulse of a sine wave. The rest being zero data.) and also drives the serial port at the point were it detects the sine pulse. As previously explained the audio data is passed through my application and the alsa driver. The difference then between the start of the sine pulse and the output from the sound card is the latency through the system. I measure this with an oscilloscope. Also just to note the test harness is driven from the same sound card as the application, so there should be no difference clocks. Now that I have explained the test scenario, I will explain the behaviour that I am experiencing. To start off, starting the test harness at different instances revealed that the latency measurement is different by maybe 20ms or so, although the alsa buffer levels indicate all is correct. Once started the behaviour is such that the difference drifts ever so slightly. After around 10 minutes the output flips to the beginning and the process starts again. This happens continuously. Without establishing the cause of this problem will ultimately mean the measurement is useless. Could this issue be related to the serial port, alsa buffering, kernel scheduling etc. Can somebody with the relevant experience help. Please.

1 0

[alsa-devel] Broken playback
by Gabriele Melia 21 Nov '07

21 Nov '07

Hi All, i'm new so sorry in advance if i do any errors. I have integrated my hw decoder inside ffmpeg that is working fine, but while playing through alsa the sound comes broken in the sense that it is playing for 3 seconds, stops for 1 plays again for 3 sec and stop again and so on. How I can delay the palyback in order to accumulate some decoded buffers to prevent this and to not afflict the decoder when playback is stopping? Thanks, Gabriele _________________________________________________________________ Scarica GRATIS le emoticon della tua squadra del cuore e il calendario di serie A! http://www.emoticons-livemessenger.com/pages/msnitcalcio/index.htm

2 1

[alsa-devel] multi pcm and mmap problem
by terminator356＠users.sourceforge.net 21 Nov '07

21 Nov '07

After 2 years I've finally traced and fixed a problem which was preventing me from using anything after ALSA 1.0.9b ! In ALSA lib 1.0.9b: pcm/pcm_generic.c: int snd_pcm_generic_channel_info() { snd_pcm_generic_t *generic = pcm->private_data; return snd_pcm_channel_info(generic->slave, info); } But in ALSA lib 1.0.15: pcm/pcm_generic.c: int snd_pcm_generic_channel_info() { snd_pcm_generic_t *generic = pcm->private_data; if (pcm->mmap_shadow) { /* No own buffer is required - the plugin won't change * the data on the buffer, or do safely on-the-place * conversion */ return snd_pcm_channel_info(generic->slave, info); } else { // Tim: TESTED: This is now called instead of the other one! /* Allocate own buffer */ return snd_pcm_channel_info_shm(pcm, info, -1); } } My multi PCM consists of a plughw:0,2 and a hw:1,0 In ALSA 1.0.9b, all five PCMs ALSA creates for this arrangement create their areas with mmap(). And it works. But in ALSA 1.0.15 my hw:1,0 and its multi PCM areas are created with mmap(), and my plughw:0,2 and its PCMs areas are created with malloc(). Can't hear/record any sound from the plughw:0,2 part of the multi. I have verified that ALSA 1.0.15 is copying the plughw:0,2 HW PCM areas to its LINEAR PCM areas, but NOT finally to its multi PCM areas. It just copies all zeros. That's the reason I hear/record no sound from the plughw:0,2 part of the multi PCM. So I reverted the function's code back to 1.0.9b and presto! It works now, the same as in ALSA 1.0.9b So please tell me, am I missing something? Should I add something more to my .asoundrc ? I see that besides {type multi ... }, there is a {type share ...} defined. *** Does the 'share' have any use here? Can't find any docs on it. Also: Yes, I can see now, why you made this code change. mmap() has limits, so you switched over to malloc(). Those limits force the user to use small buffers and/or reduce the number of channels created. Otherwise aplay, arecord, and jack etc complain mmap() failed. In ALSA 1.0.9b, and after my change to ALSA 1.0.15, I tried ulimit() -l, and editing /etc/security/limits.conf I quadrupled the available locked memory, but aplay, arecord etc still won't let me increase the buffer size beyond what it was BEFORE I edited limits.conf *** What am I doing wrong? Thanks Tim.

2 1

[alsa-devel] alsa timestamps
by Heikki Lindholm 21 Nov '07

21 Nov '07

Hello list, I took up some old dusty code of mine that uses snd_pcm_state followed by snd_pcm_status_get_tstamp when in capture mode. The code used to work, but now the returned timestamps are all zeroes. Is there some API change done recently or is the whole timestamping deprecated or something? I've tried with different drivers on ubuntu's alsa .14 and gentoo's .14. I've also tried mmap'ed and r/w modes, and I'm setting the TSTAMP_MMAP sw param. -- Heikki Lindholm

3 8

Re: [alsa-devel] [alsa-cvslog] alsa-lib: pcm - Limit the avail_min minimum size
by Jaroslav Kysela 21 Nov '07

21 Nov '07

On Tue, 20 Nov 2007, Takashi Iwai wrote: > changeset: 2352:39d34d6a4587 > tag: tip > user: tiwai > date: Tue Nov 20 15:29:10 2007 +0100 > files: src/pcm/pcm.c > description: > pcm - Limit the avail_min minimum size > > Fix avail_min if it's less than period_size. The too small avail_min > is simply useless and the cause of CPU hog with rate plugin. > > > diff -r b1d1733e52f8 -r 39d34d6a4587 src/pcm/pcm.c > --- a/src/pcm/pcm.c Mon Nov 19 08:07:19 2007 +0100 > +++ b/src/pcm/pcm.c Tue Nov 20 15:29:10 2007 +0100 > @@ -5577,6 +5577,12 @@ int snd_pcm_sw_params_set_avail_min(snd_ > #endif > { > assert(pcm && params); > + /* Fix avail_min if it's below period size. The period_size > + * defines the minimal wake-up timing accuracy, so it doesn't > + * make sense to set below that. > + */ > + if (val < pcm->period_size) > + val = pcm->period_size; > params->avail_min = val; > return 0; > } I think that this patch is wrong. We may use system timers to increase (fine-tune) "interrupt" latencies for pcm streams - see tick time in driver and library. Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project

2 1

[alsa-devel] hda-intel codec realtec ALC883
by Jochen Maes 20 Nov '07

20 Nov '07

hey all, just writing because in order to setup my alsa card i had to do some extra steps that might be documented better. steps taken: 1) add to /etc/modules.d/alsa: options snd-hda-intel model=acer-aspire (this is documented) 2) launch alsamixer, go to the capture screen, select the "capture" and press spacebar + set to 75, d the same for "capture 1" the capture simply didn't work otherwise. kind regards, SeJo -- "Defer no time, delays have dangerous ends" "Ne humanus crede" Jochen Maes http://sejo.be http://monitron.org http://www.freenode.net Codec: Realtek ALC883 Address: 0 Vendor Id: 0x10ec0883 Subsystem Id: 0x10250090 Revision Id: 0x100002 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A Node 0x02 [Audio Output] wcaps 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: [0x16 0x16] 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: [0x17 0x17] 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] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 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: [0x19 0x19] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] 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: [0x80 0x80] [0x80 0x80] 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 0xc2211110: [Both] HP Out at Ext Front Conn = 1/8, Color = Black 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 0x411111f0: [N/A] Speaker 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 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black 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 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black 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 0x02a19930: [Jack] Mic at Ext Front 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: [0x02 0x02] [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 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black 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 0x0281313f: [Jack] Line In at Ext Front 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: [0x00 0x00] Pincap 0x08173e: IN OUT HP Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0xc0: OUT HP Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 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: 0x00: Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x4015130d: [N/A] Speaker at Ext N/A Conn = Optical, Color = Black Pin-ctls: 0x00: Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x02451120: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black Pin-ctls: 0x00: Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 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] [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 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] [0x80 0x80] Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x25 0x0b 00:00.0 Host bridge: Intel Corporation Mobile 945GM/PM/GMS, 943/940GML and 945GT Express Memory Controller Hub (rev 03) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information <?> 00:02.0 VGA compatible controller: Intel Corporation Mobile 945GM/GMS, 943/940GML Express Integrated Graphics Controller (rev 03) (prog-if 00 [VGA controller]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 17 Region 0: Memory at d0200000 (32-bit, non-prefetchable) [size=512K] Region 1: I/O ports at 5088 [size=8] Region 2: Memory at c0000000 (32-bit, prefetchable) [size=256M] Region 3: Memory at d0300000 (32-bit, non-prefetchable) [size=256K] Capabilities: [90] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Address: 00000000 Data: 0000 Capabilities: [d0] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00:02.1 Display controller: Intel Corporation Mobile 945GM/GMS, 943/940GML Express Integrated Graphics Controller (rev 03) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Region 0: Memory at d0280000 (32-bit, non-prefetchable) [size=512K] Capabilities: [d0] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 22 Region 0: Memory at d0340000 (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 (v1) Root Complex Integrated Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <64ns, L1 <1us ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #0, Speed unknown, Width x0, ASPM unknown, Latency L0 <64ns, L1 <1us ClockPM- Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed unknown, Width x0, TrErr- Train- SlotClk- DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Virtual Channel <?> Capabilities: [130] Root Complex Link <?> 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 02) (prog-if 00 [Normal decode]) Control: I/O- Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 unlimited ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #1, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <1us, L1 <4us ClockPM- Suprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x0, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surpise+ Slot # 2, PowerLimit 6.500000; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet- Interlock- Changed: MRL- PresDet- LinkState- RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Address: 00000000 Data: 0000 Capabilities: [90] Subsystem: Acer Incorporated [ALI] Unknown device 0090 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> 00:1c.1 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 2 (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 unlimited ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #2, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <1us, L1 <4us ClockPM- Suprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x0, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surpise+ Slot # 3, PowerLimit 6.500000; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet- Interlock- Changed: MRL- PresDet- LinkState- RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Address: 00000000 Data: 0000 Capabilities: [90] Subsystem: Acer Incorporated [ALI] Unknown device 0090 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=04, subordinate=04, sec-latency=0 Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 unlimited ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #3, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <1us, L1 <4us ClockPM- Suprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk- ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x0, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surpise+ Slot # 4, PowerLimit 6.500000; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet- CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet- Interlock- Changed: MRL- PresDet- LinkState- RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Address: 00000000 Data: 0000 Capabilities: [90] Subsystem: Acer Incorporated [ALI] Unknown device 0090 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> 00:1c.3 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 4 (rev 02) (prog-if 00 [Normal decode]) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Bus: primary=00, secondary=05, subordinate=05, sec-latency=0 Memory behind bridge: d0100000-d01fffff Secondary status: 66MHz- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA+ VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [40] Express (v1) Root Port (Slot+), MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s unlimited, L1 unlimited ExtTag- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd- ExtTag- PhantFunc- AuxPwr- NoSnoop- MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr- FatalErr- UnsuppReq- AuxPwr+ TransPend- LnkCap: Port #4, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <256ns, L1 <4us ClockPM- Suprise- LLActRep+ BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM- AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive+ BWMgmt- ABWMgmt- SltCap: AttnBtn- PwrCtrl- MRL- AttnInd- PwrInd- HotPlug+ Surpise+ Slot # 5, PowerLimit 6.500000; Interlock- NoCompl- SltCtl: Enable: AttnBtn- PwrFlt- MRL- PresDet+ CmdCplt- HPIrq- LinkChg- Control: AttnInd Unknown, PwrInd Unknown, Power- Interlock- SltSta: Status: AttnBtn- PowerFlt- MRL- CmdCplt- PresDet+ Interlock- Changed: MRL- PresDet- LinkState- RootCtl: ErrCorrectable- ErrNon-Fatal- ErrFatal- PMEIntEna- CRSVisible- RootCap: CRSVisible- RootSta: PME ReqID 0000, PMEStatus- PMEPending- Capabilities: [80] Message Signalled Interrupts: Mask- 64bit- Queue=0/0 Enable- Address: 00000000 Data: 0000 Capabilities: [90] Subsystem: Acer Incorporated [ALI] Unknown device 0090 Capabilities: [a0] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [100] Virtual Channel <?> Capabilities: [180] Root Complex Link <?> 00:1d.0 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #1 (rev 02) (prog-if 00 [UHCI]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 21 Region 4: I/O ports at 50a0 [size=32] 00:1d.1 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #2 (rev 02) (prog-if 00 [UHCI]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin B routed to IRQ 19 Region 4: I/O ports at 50c0 [size=32] 00:1d.2 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #3 (rev 02) (prog-if 00 [UHCI]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin C routed to IRQ 18 Region 4: I/O ports at 50e0 [size=32] 00:1d.3 USB Controller: Intel Corporation 82801G (ICH7 Family) USB UHCI Controller #4 (rev 02) (prog-if 00 [UHCI]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin D routed to IRQ 17 Region 4: I/O ports at 5400 [size=32] 00:1d.7 USB Controller: Intel Corporation 82801G (ICH7 Family) USB2 EHCI Controller (rev 02) (prog-if 20 [EHCI]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Interrupt: pin A routed to IRQ 21 Region 0: Memory at d0544000 (32-bit, non-prefetchable) [size=1K] Capabilities: [50] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=375mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [58] Debug port: BAR=1 offset=00a0 00:1e.0 PCI bridge: Intel Corporation 82801 Mobile PCI Bridge (rev e2) (prog-if 01 [Subtractive decode]) Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Bus: primary=00, secondary=06, subordinate=0a, sec-latency=32 Memory behind bridge: d0000000-d00fffff Secondary status: 66MHz- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort+ <SERR- <PERR- BridgeCtl: Parity- SERR- NoISA- VGA- MAbort- >Reset- FastB2B- PriDiscTmr- SecDiscTmr- DiscTmrStat- DiscTmrSERREn- Capabilities: [50] Subsystem: Acer Incorporated [ALI] Unknown device 0090 00:1f.0 ISA bridge: Intel Corporation 82801GBM (ICH7-M) LPC Interface Bridge (rev 02) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0 Capabilities: [e0] Vendor Specific Information <?> 00:1f.2 IDE interface: Intel Corporation 82801GBM/GHM (ICH7 Family) SATA IDE Controller (rev 02) (prog-if 80 [Master]) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem- BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz+ UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx+ Latency: 0 Interrupt: pin B routed to IRQ 19 Region 0: I/O ports at 01f0 [size=8] Region 1: I/O ports at 03f4 [size=1] Region 2: I/O ports at 0170 [size=8] Region 3: I/O ports at 0374 [size=1] Region 4: I/O ports at 5440 [size=16] Capabilities: [70] Power Management version 2 Flags: PMEClk- DSI- D1- D2- AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 00:1f.3 SMBus: Intel Corporation 82801G (ICH7 Family) SMBus Controller (rev 02) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin B routed to IRQ 10 Region 4: I/O ports at 5420 [size=32] 05:00.0 Network controller: Intel Corporation PRO/Wireless 3945ABG Network Connection (rev 02) Subsystem: Intel Corporation Unknown device 1001 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR+ FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 0, Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 10 Region 0: Memory at d0100000 (32-bit, non-prefetchable) [size=4K] Capabilities: [c8] Power Management version 2 Flags: PMEClk- DSI+ D1- D2- AuxCurrent=0mA PME(D0+,D1-,D2-,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Capabilities: [d0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Address: 0000000000000000 Data: 0000 Capabilities: [e0] Express (v1) Legacy Endpoint, MSI 00 DevCap: MaxPayload 128 bytes, PhantFunc 0, Latency L0s <512ns, L1 unlimited ExtTag- AttnBtn- AttnInd- PwrInd- RBE- FLReset- DevCtl: Report errors: Correctable- Non-Fatal- Fatal- Unsupported- RlxdOrd+ ExtTag- PhantFunc- AuxPwr- NoSnoop+ MaxPayload 128 bytes, MaxReadReq 128 bytes DevSta: CorrErr- UncorrErr+ FatalErr- UnsuppReq+ AuxPwr+ TransPend- LnkCap: Port #0, Speed 2.5GT/s, Width x1, ASPM L0s L1, Latency L0 <128ns, L1 <64us ClockPM+ Suprise- LLActRep- BwNot- LnkCtl: ASPM Disabled; RCB 64 bytes Disabled- Retrain- CommClk+ ExtSynch- ClockPM+ AutWidDis- BWInt- AutBWInt- LnkSta: Speed 2.5GT/s, Width x1, TrErr- Train- SlotClk+ DLActive- BWMgmt- ABWMgmt- Capabilities: [100] Advanced Error Reporting <?> Capabilities: [140] Device Serial Number 99-dd-c6-ff-ff-de-18-00 06:01.0 Ethernet controller: Broadcom Corporation BCM4401-B0 100Base-TX (rev 02) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=fast >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 64 Interrupt: pin A routed to IRQ 20 Region 0: Memory at d0000000 (32-bit, non-prefetchable) [size=8K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=2 PME- 06:04.0 CardBus bridge: ENE Technology Inc CB-712/4 Cardbus Controller (rev 10) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Latency: 168, Cache Line Size: 128 bytes Interrupt: pin A routed to IRQ 17 Region 0: Memory at d0002000 (32-bit, non-prefetchable) [size=4K] Bus: primary=06, secondary=07, subordinate=0a, sec-latency=176 Memory window 0: 50000000-53fff000 (prefetchable) Memory window 1: 54000000-57fff000 I/O window 0: 00001400-000014ff I/O window 1: 00001800-000018ff BridgeCtl: Parity- SERR- ISA- VGA- MAbort- >Reset+ 16bInt+ PostWrite+ 16-bit legacy interface ports at 0001 06:04.1 FLASH memory: ENE Technology Inc ENE PCI Memory Stick Card Reader Controller (rev 01) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin B routed to IRQ 11 Region 0: Memory at d0003000 (32-bit, non-prefetchable) [disabled] [size=128] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 06:04.2 SD Host controller: ENE Technology Inc ENE PCI Secure Digital Card Reader Controller (rev 01) (prog-if 01) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin B routed to IRQ 16 Region 0: Memory at d0003400 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 06:04.3 FLASH memory: ENE Technology Inc FLASH memory: ENE Technology Inc: (rev 01) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin B routed to IRQ 11 Region 0: Memory at d0003800 (32-bit, non-prefetchable) [disabled] [size=128] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- 06:04.4 FLASH memory: ENE Technology Inc SD/MMC Card Reader Controller (rev 01) Subsystem: Acer Incorporated [ALI] Unknown device 0090 Control: I/O- Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- Interrupt: pin B routed to IRQ 16 Region 0: Memory at d0003100 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME-

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[alsa-devel] [PATCH 0/3] PNP cleanups - Version 2
by Thomas Renninger 20 Nov '07

20 Nov '07

Changes: - Introduce pnp_irq_no instead of pnp_irq_start (same for dma). - Let irq and dma resource structs never access .end. .end was always only a copy of .start for dma and irq Some background what I want to do next to allocate resources on the fly: I want to use pnp_{port,dma,irq,mem} macros mainly in loops to check whether a resource got allocated: e.g.: - if (idx >= PNP_MAX_DMA) { + if (!pnp_dma(dev, idx)) { or: - for (idx = 0; idx < PNP_MAX_IRQ; idx++) { + for (idx = 0; pnp_irq(dev, idx); idx++) { that would look like this in pnp.h: +#define pnp_port(dev,bar) ((dev)->res.allocated_ports > (bar) \ + ? (&(dev)->res.port_resource[(bar)]) : NULL) I'd like to use krealloc to realloc portions of e.g. 8 port resources on the fly. As krealloc modifies the addresses of the array, it must be insured that these are only used at pnp parse time. This seem to work fine for pnpacpi, were most of the: echo {get,fill,clean,...} >/sys/devices/pnp0/*/resources interface does not work for me :) It should be possible to get this working, if I could get some testing help with these interfaces, that would be great. Is any tool making use of this? The "get" looks evil. It reparses the devices' BIOS information. This should work as long as not more resources are attached to a device compared to the initial parse and krealloc must pop in, this should be the case... I wonder whether this get and possibly other sysfs interfaces in drivers/pnp/interface.c can be reverted (it's debug only?). Then the whole parsing code could theoretically be made __init data. Bjorn? Also the pnp_resource_change exported symbol could be a problem, if resources are simply defined by the driver, even there were none (allocated) found by the pnp{isa,bios,acpi} parser. But this should be "workaroundable" by doing one malloc/krealloc portion there. It must be called before insert_resource(..) got called anyway (which passes the address of a struct resource to be registered as global used IO/System Memory). Invalidating the passed struct resource pointers afterwards would be fatal. Just found that problem, I haven't looked at it in detail. Any help, comments, etc. are very much appreciated, Thomas

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[alsa-devel] [PATCH] mm snapshot broken-out-2007-11-20-01-45 build failer tumbler/snapper
by Kamalesh Babulal 20 Nov '07

20 Nov '07

Hi Andrew, The kernel build fails, with following error CC sound/ppc/tumbler.o sound/ppc/tumbler.c: In function ‘snapper_get_capture_source’: sound/ppc/tumbler.c:812: error: ‘union <anonymous>’ has no member named ‘value’ sound/ppc/tumbler.c: In function ‘snapper_put_capture_source’: sound/ppc/tumbler.c:824: error: ‘union <anonymous>’ has no member named ‘enuemerated’ make[2]: *** [sound/ppc/tumbler.o] Error 1 make[1]: *** [sound/ppc] Error 2 make: *** [sound] Error 2 Signed-off-by: Kamalesh Babulal <kamalesh(a)linx.vnet.ibm.com> -- --- linux-2.6.24-rc3/sound/ppc/tumbler.c 2007-11-20 16:13:42.000000000 +0530 +++ linux-2.6.24-rc3/sound/ppc/~tumbler.c 2007-11-20 19:14:27.000000000 +0530 @@ -809,7 +809,7 @@ static int snapper_get_capture_source(st struct pmac_tumbler *mix = chip->mixer_data; snd_assert(mix, return -ENODEV); - ucontrol->value.enumerated.value[0] = mix->capture_source; + ucontrol->value.enumerated.item[0] = mix->capture_source; return 0; } @@ -821,7 +821,7 @@ static int snapper_put_capture_source(st int change; snd_assert(mix, return -ENODEV); - change = ucontrol->value.enuemerated.item[0] != mix->capture_source; + change = ucontrol->value.enumerated.item[0] != mix->capture_source; if (change) { mix->capture_source = !!ucontrol->value.enumerated.item[0]; snapper_set_capture_source(mix); -- Thanks & Regards, Kamalesh Babulal, Linux Technology Center, IBM, ISTL.

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[alsa-devel] Duplex syncronization in ALSA
by Fredrik Lingvall 20 Nov '07

20 Nov '07

Hi all, I am an researcher in the field of acoustic (ultrasonic) signal processing. Normally we use dedicated hardware where the transmit and receive processes are synchronized. Now I want to use a multi-channel sound card (in my case a RME Multiface card) to perform, for example, simple beamforming. Typically, one transmit pulses on an array of transducers (speakers) where each pulse is delayed a certain amount to create focusing or beemsteering. By knowing the soundspeed in the propagation medium one can calculate the double path time-of-flight to various points and then, by beamforming the received signals, one can get an image of the objects that are in front of the acoustic array. Now to my problem: in order to do the processing one need to know the delay in the system from transmit to receive. I have looked at the ALSA lib API docs and some code examples (pcm.c, aplay etc) and it looks like the playback and capture processes are independent in ALSA (i.e., there is no write-and-read-at-the-same-time function). The latency inherent in the soundcard isn't that much of a problem if it is know. It is, however a problem if the latency from transmit to receive is not constant from measurement to measurement. I have written some code (for Octave) that uses MMAP and poll:ing but I don't see how to synchronize the write and read processes (they use two different poll fd:s). What is the best way to do this? Is it possible at all? Regards /Fredrik

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[alsa-devel] snd_pcm_status_get_avail() values
by Markus Korber 20 Nov '07

20 Nov '07

Hello, I've got a driver with the following settings: ,---- | static const struct snd_pcm_hardware svenm_pcm_hardware = { | .info = SNDRV_PCM_INFO_MMAP | | SNDRV_PCM_INFO_MMAP_VALID | | SNDRV_PCM_INFO_BLOCK_TRANSFER | | SNDRV_PCM_INFO_PAUSE | | SNDRV_PCM_INFO_RESUME | | SNDRV_PCM_INFO_INTERLEAVED, | .formats = SNDRV_PCM_FMTBIT_S16_LE | | SNDRV_PCM_FMTBIT_U16_LE, | .channels_min = 2, /* 2 = only stereo */ | .channels_max = 2, | .period_bytes_min = 8*1024, | .period_bytes_max = 8*1024, | .periods_min = 2, | .periods_max = 2, | .buffer_bytes_max = 4*8*1024, | }; `---- ,----[ _pcm_prepare() ] | chip->playback.hw_buffer_size = prtd->period_bytes * 2; /* byte size */ | chip->playback.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream); `---- Since my HW only allows for a period number of 2, I would like to always receive period_bytes_min(max) input bytes from an application (mplayer, xine, etc.) so that I can always fill one complete HW output buffer. The application (mplayer) does [3] to determine the number of bytes it is allowed to write with snd_pcm_writei() (for logging see [4]): ,----[ 3 ] | bytes_to_write = snd_pcm_status_get_avail(status) * bytes_per_sample; `---- (Here 1 sample equals 4 bytes, S16_LE, stereo.) ,----[ 4 ] | fill_audio_out_buffers: bytes_to_write=16384 | alsa-play: frames=4096, len=16384 | fill_audio_out_buffers: bytes_to_write=8192 | alsa-play: frames=2048, len=8192 | fill_audio_out_buffers: bytes_to_write=8192 | alsa-play: frames=2048, len=8192 | fill_audio_out_buffers: bytes_to_write=8192 | alsa-play: frames=2048, len=8192 | fill_audio_out_buffers: bytes_to_write=16384 | alsa-play: frames=4096, len=16384 | fill_audio_out_buffers: bytes_to_write=16384 | alsa-play: frames=4096, len=16384 `---- Furthermore, I'm using the pcm-indirect framework[1] and my hw_ptr always toggles between 0 and period_bytes[2] (bufnum refers to the HW playback buffer number). ,----[ 1 ] | static snd_pcm_uframes_t | chip_pcm_pointer(struct snd_pcm_substream *substream) | { | ... | return snd_pcm_indirect_playback_pointer(substream, &chip->playback, | prtd->playback_hw_ptr); | } `---- ,----[ 2 ] | if (bufnum == 0) | prtd->playback_hw_ptr = 0; | else | prtd->playback_hw_ptr = prtd->period_bytes; `---- Thus, how is the value returned by snd_pcm_status_get_avail determined? I mean, if my pointer always toggles between 0 and period_bytes[2], how can ALSA return the values logged in [4]? Would this mean that I'm not able to process (in the application) the samples fast enough and the HW buffer runs empty in the meantime? (I would expect snd_pcm_status_get_avail to always returns 8192.) Or is there an additional parameter or constraint to somehow limit the input sample number from an application? Thanks. Regards, Markus Korber

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[alsa-devel] Strange bug in .asoundrc ??
by Dominique Larchey-Wendling 19 Nov '07

19 Nov '07

Dear all, It seems that neither the current ctl.hw function nor the following simplified function ctl.!hw work correctly when the argument CARD is given to the function. It works however when no argument is given as the following test examples show it. It have alsa-lib version 1.0.14-0.4.rc3.fc7 installed on FC7. Is this a know bug of alsa-lib ? It is strange because this bug does not show up with pcm.hw for example. --- .asoundrc ------------------------ ctl.!hw { @args [ CARD ] @args.CARD { type string default "0" } type hw card $CARD } ctl.rha "hw" ctl.tata "hw:0" ctl.toto "hw:1" ------------------------------------------------ rha % amixer -D rha info Card rha 'Intel'/'HDA Intel at 0xfebfc000 irq 16' Mixer name : 'SigmaTel STAC9200' Components : 'HDA:83847690' Controls : 6 Simple ctrls : 4 rha % amixer -D tata info ALSA lib control.c:781:(snd_ctl_open_conf) Invalid type for CTL tata definition amixer: Control device tata open error: Invalid argument rha % amixer -D toto info ALSA lib control.c:781:(snd_ctl_open_conf) Invalid type for CTL toto definition amixer: Control device toto open error: Invalid argument

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[alsa-devel] No sound on Intel DQ965GF motherboard
by Eric S. Raymond 19 Nov '07

19 Nov '07

Thank you, alsa developers, for the prompt resolution of my previous sound issue with the Thinkpad X61. Unfortunately, I've now turned up an apparently different problem on very similar hardware. Sound doesn't work on my DQ965GF motherboard under Ubuntu 7.10. Here's what I've found out about it. I'm writing in hope that someone on the ALSA devel list can suggest diagnostics to pin down the problem exactly. The device has PCI ID [8086:284b] (rev 02). It is an Intel 82801H. This appears to be an Ubuntu-specific error. At http://www.linuxquestions.org/questions/linux-hardware-18/intel-dq965gf-mob… I found a post implying that at least the rear jacks on the mobo are active under Fedora Core 6. I have therefore filed an Ubuntu bug at https://bugs.launchpad.net/ubuntu/+bug/162410 but seen no response. This also appears to be a very hardware-configuration-specific error. I have an Intel HDA audio device on my X61 laptop with PCI ID [8086:284b] (rev 03), another 82801H, that works just fine under Ubuntu 7.10 (the alsa version is nominally 1.0.14 but some 1.10.15 fixes, including the right one, were backported). Note the 03 as opposed to 02 subrev. aplay -l, which is supposed to list sound playback devices, returns no listing on the DQ965GF. As a check, on the X61 where it works it returns this: card 0: Intel [HDA Intel], device 0: AD198x Analog [AD198x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod also returns suggestively different results. On the laptop, its output includes this line. snd_hda_intel 337192 1 On the DQ965GF, here are the sound device lines: snd_hda_intel 337192 0 snd_usb_audio 96640 1 snd_pcm_oss 50048 0 snd_mixer_oss 20096 1 snd_pcm_oss snd_pcm 94344 3 snd_hda_intel,snd_usb_audio,snd_pcm_oss snd_usb_lib 20352 1 snd_usb_audio snd_seq_dummy 5380 0 snd_seq_oss 36864 0 snd_seq_midi 11008 0 snd_seq_midi_event 9984 2 snd_seq_oss,snd_seq_midi snd_seq 62496 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 27272 2 snd_pcm,snd_seq snd_rawmidi 29824 2 snd_usb_lib,snd_seq_midi snd_seq_device 10260 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq,snd_rawmidi snd_hwdep 12168 1 snd_usb_audio snd 69288 13 snd_hda_intel,snd_usb_audio,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_oss,snd_seq,snd_timer,snd_rawmidi,snd_seq_device,snd_hwdep soundcore 10272 1 snd snd_page_alloc 12560 2 snd_hda_intel,snd_pcm usbcore 161584 8 usbhid,snd_usb_audio,snd_usb_lib,xpad,gspca,ehci_hcd,uhci_hcd Note that snd_hda_intel has no users and that snd_usb_audio has 1. On the DQ965GF, the Sound Preferences tab on my Gnome toolbar thinks the sound device is "USB Device 0x46d:0x8ad (ALsa mixer)". I don't know what tool GNOME is running to determine this. This arch-linux thread gave me some insight: http://bugs.archlinux.org/task/6617 It appears the snd_usb_audio driver is a red herring, or at least a separate bug. I checked, and my dmesg log does indeed say "hda-intel: no codecs found!" on boot, which suggests that snd_hda_intel is loading and then failing to initialize properly, accounting for the usage count of 0. I tried reverting the kernel patch mentioned in the arch thread (e.g. recompiling the snd_hda_intel driver with #define AZX_MAX_CODECS 4) but this did *not* solve the problem. Any suggestions would be appreciated. I need my music back! -- <a href="http://www.catb.org/~esr/">Eric S. Raymond</a> The common argument that crime is caused by poverty is a kind of slander on the poor. -- H. L. Mencken

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Re: [alsa-devel] E-mu 1212M ADAT channels missing ?
by Ctirad Fertr 19 Nov '07

19 Nov '07

Dne Monday 19 of November 2007 03:04:15 Kenneth Long napsal(a): > yeah, that is it.. it use to be hw 0,0. Its now 0,2. > > There seems to be a lot more xruns now... > is that from something new? > my kernel is now 2.6.23-gentoo-r1 There shouldn't be any changes in this aspect. Try to increase size of jack buffers and check if jack samplerate matches samplerate in alsamixer. ------ Ctirad > thank you. > > --- Ctirad Fertr <c.fertr(a)gmail.com> wrote: > > Dne Sunday 18 of November 2007 05:54:37 Kenneth Long > > > > napsal(a): > > > I'm using qjackctl to launch jackd and before, I > > > > would > > > > > see the PCM channels appear under "Connect" > > > > button. > > > > > Alsa_pcm only shows two channels "capture_1 and > > > capture_2. Before there were 16 channels... > > > > Do you use correct capture device in qjackctl? The > > correct device for > > multichannel capture is 0,2 (in case your card is > > 0). > > > > ------ > > Ctirad > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > > > hello

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[alsa-devel] [BUG] New Kernel Bugs
by Natalie Protasevich 19 Nov '07

19 Nov '07

This is the listing of the open bugs that are relatively new, around 2.6.22 and up. They are vaguely classified by specific area. (not a full list, there are more :) The good part is that reporters of the bugs below are still around and haven't dissipated, or disposed of their hardware, so it is a good time to get the bugs. Those bugzillas that have been started as regressions on Rafael's list are not mentioned here so far, since they are being tracked as new regressions already. It would be appreciated if the corresponding maintenance team could take a look, close off any which are fixed and see if they can fix any which aren't. NOTE: when replying to this email, please add the bug number to the Subject in the form [Bug 1234] so that bugzilla will capture the discussion. Thanks. ACPI==================================================================== System does not load without acpi=off ide=nodma noapic http://bugzilla.kernel.org/show_bug.cgi?id=9358 Kernel: 2.6.23.1 ACPI Error attaching device data http://bugzilla.kernel.org/show_bug.cgi?id=9354 Kernel: 2.6.24-rc2 /proc/acpi/battery displays Incorrect voltages http://bugzilla.kernel.org/show_bug.cgi?id=9341 Kernel: 2.6.23.1 PATA scan: ACPI Exception AE_AML_PACKAGE_LIMIT... is beyond end of object http://bugzilla.kernel.org/show_bug.cgi?id=9320 Kernel: 2.6.24-rc2 (Tejun: calling _GTF without calling _STM first. _GTM doesn't have any prerequisite (it can't). Can someone familiar with ACPI tell me why the method is failing? At any rate, libata should work fine regardless of ACPI failures. Maybe it's time to start blacklist to skip ATA-ACPI for some boards to avoid those annoying messages during boot) ACPI Battery Info in /sys but not /proc/acpi http://bugzilla.kernel.org/show_bug.cgi?id=9183 Kernel: 2.6.23-rc8-mm2 When using ACPI on a Compaq Presario V6221EU the laptop goes into deadlock after a random amount of time http://bugzilla.kernel.org/show_bug.cgi?id=9118 Kernel: 2.6.23-rc6 ACPI video driver should validate brightness level before setting it via _BCM http://bugzilla.kernel.org/show_bug.cgi?id=9277 Kernel: 2.6.23 VIDEO/DVB dvb driver reboot system http://bugzilla.kernel.org/show_bug.cgi?id=9357 Kernel: 2.6.21.5 PLATFORM=============================================================== xipImage is built so that uBoot cant run it (ARM) http://bugzilla.kernel.org/show_bug.cgi?id=9356 Kernel: 2.6.21 Samsung R20 - ACPI: PCI Root Bridge [PCI0] (0000:00) http://bugzilla.kernel.org/show_bug.cgi?id=9339 Kernel: 2.6.24 (boot is very long ..MP-BIOS bug: 8254 timer not connected to IO-APIC then the boot stop at : ACPI: PCI Root Bridge [PCI0] (0000:00) (during 3 minutes, and boot continue) system_64.h: switch_to inline asm should be more robbust wrt optimizations http://bugzilla.kernel.org/show_bug.cgi?id=9302 Kernel: 2.6.24-rc1 with CONFIG_NO_HZ and/or CONFIG_HPET_TIMER set kernel 2.6.23 doesn't boot (ARM, Timer) http://bugzilla.kernel.org/show_bug.cgi?id=9229 Kernel: 2.6.23 NETWORKING=========================================================== RTNLGRP_ND_USEROPT does not report ifindex (IPv6) http://bugzilla.kernel.org/show_bug.cgi?id=9349 Kernel: 2.6.24+ a kernel error happend in the func: __skb_dequeue when using in pfifo_fast_dequeue http://bugzilla.kernel.org/show_bug.cgi?id=9342 Kernel: 2.6.11.1 - reporter asked to try recent kernel e100 does not work after boot http://bugzilla.kernel.org/show_bug.cgi?id=9336 Kernel: 2.6.23.1 2.6.23.1-smp kernel panic (network-related) http://bugzilla.kernel.org/show_bug.cgi?id=9318 Kernel: 2.6.23.1 Infiniband panic sundance -> 4port D-Link System Inc DFE-580TX -> Log errors http://bugzilla.kernel.org/show_bug.cgi?id=9311 Kernel: 2.6.22.9 via-rhine driver stalls with: PHY status 786d, resetting... http://bugzilla.kernel.org/show_bug.cgi?id=9300 Kernel: 2.6.23+ Weird network problems with 2.6.23-rc2 http://bugzilla.kernel.org/show_bug.cgi?id=9080 http://lkml.org/lkml/2007/8/11/40 - description rt2500pci: low TCP throughput (wireless) http://bugzilla.kernel.org/show_bug.cgi?id=9273 Kernel: 2.6.24-rc1 This is a regression Unable to build wifi network between zd1201 and b43 http://bugzilla.kernel.org/show_bug.cgi?id=9237 Kernel: 2.6.24-rc1 Crash after module unload in b43 (wireless) http://bugzilla.kernel.org/show_bug.cgi?id=9233 Kernel: 2.6.24-rc1 (net typhoon) "no descs for cmd, had (needed) 0 (1) cmd, 31 (7) resp" http://bugzilla.kernel.org/show_bug.cgi?id=9225 Kernel: 2.6.23.1 IDE/SATA========================================================= pata_pdc202xx_old excessive ATA bus errors http://bugzilla.kernel.org/show_bug.cgi?id=9337 2.6.24-rc2 Drive seagate ST380011AS needs to be blacklisted http://bugzilla.kernel.org/show_bug.cgi?id=9309 Kernel: 2.6.22.X DVD-RAM umount and disk free bug http://bugzilla.kernel.org/show_bug.cgi?id=9265 Kernel: 2.6.15 (asked to try current kernel) FILE SYSTEMS======================================================= ext4: delalloc space accounting problem drops data http://bugzilla.kernel.org/show_bug.cgi?id=9329 Kernel: 2.6.24-rc1 POSIX Access Control Lists cause bogus file system check errors http://bugzilla.kernel.org/show_bug.cgi?id=9241 Kernel: 2.6.23.1 MEMORY MANAGEMENT================================================ My system hangs when it has no more free memory to allocate via malloc() http://bugzilla.kernel.org/show_bug.cgi?id=9316 Kernel: 2.6.23 User program, "My system hangs when it has no more free memory to allocate via malloc()" BUG: unable to handle kernel paging request at virtual address 26121228/kswapd0[231] exited with preempt_count 1 http://bugzilla.kernel.org/show_bug.cgi?id=9305 EIP is at free_block+0x6d/0xe4 Kernel: 2.6.22.6 POWER MANAGEMENT================================================== IBM X41 looses time after Suspend2Disk http://bugzilla.kernel.org/show_bug.cgi?id=9314 Kernel: 2.6.23 Suspend to RAM resume hangs on a tickless (NO_HZ) kernel http://bugzilla.kernel.org/show_bug.cgi?id=9275 Kernel: 2.6.23 This is HP notebook nc6320 T2400 945GM VIDEO DRIVERS======================================================== No text consoles with FRAMEBUFFER_CONSOLE_DETECT_PRIMARY http://bugzilla.kernel.org/show_bug.cgi?id=9310 Kernel: 2.6.24-rc1 This is a regression PARALLEL PORT======================================================== LPC IT8705 POST port making noise on parallel port http://bugzilla.kernel.org/show_bug.cgi?id=9306 Kernel: 2.6.16+ I/O STORAGE=========================================================== kernel bug from pktcdvd http://bugzilla.kernel.org/show_bug.cgi?id=9294 Kernel: 2.6.23 After pci-e video card was installed, pci add-on usb card & firewire card fail http://bugzilla.kernel.org/show_bug.cgi?id=9223 Kernel: 2.6.20 (testing of latest kernel requested) SCSI================================================================== qla2xxx: driver initialization does not complete when booting with Port connected http://bugzilla.kernel.org/show_bug.cgi?id=9267 Kernel: 2.6.23.1 SOUND ALSA============================================================ Unable to load snd-hda-intel module: Unknown symbol in module, or unknown parameter http://bugzilla.kernel.org/show_bug.cgi?id=9242 Kernel: 2.6.24-rc1 usbaudio microphone: regular sound distortion on several Logitech Webcams http://bugzilla.kernel.org/show_bug.cgi?id=9230 Kernel: 2.6.22.9 HID==================================================================== Kernel NULL pointer dereference at :usbhid:hiddev_ioctl+0x2f/0xabc http://bugzilla.kernel.org/show_bug.cgi?id=9216 Kernel: 2.6.23.1 Looks like this is a regression

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[alsa-devel] Can this be a driver problem? Sound works fine for ogg files but not for avi
by alsa-devel.20.pris＠spamgourmet.com 18 Nov '07

18 Nov '07

Hello, as I have already reported I'm having problems with my sound: http://news.gmane.org/find-root.php?group=gmane.linux.alsa.devel&article=49… Takashi Iwai suggested installing the latest version of the driver. But, before I do this I want to make sure that it is a driver problem and not some other problem. The thing is, on my system(Ubuntu Gutsy Gibbon) I get perfect sound playing the example ogg files that come with the system. The system sounds seem to work fine also. But I have problems when trying to listen to avi or mp3 files. Can this still be an Alsa problem or is this a problem of some incorrect configuration? Thanks!

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[alsa-devel] E-mu 1212M/1820M/1616M development
by Ctirad Fertr 18 Nov '07

18 Nov '07

Hello, I just bought an 1616M PCMCIA in addition to my old 1212m v1 and want to help improve the current driver. I wrote one little patch for those cards in the past (and some bigger patches for my experiments which were never released) and now I have some time to continue. I'm going to add missing switches into mixer, make some changes in mixer to match the real I/O setup of each particular card model and perhaps also write some graphics patchmix like app. Unfortunatelly, I'm unable to fix the most important bugs (at least from my point of view) dua to lack of documentation and knowlenge of DSP programming. These are: 1. non working 24bit playback. It should be on device 0,2 This device exists, but cannot be openned for playback.It is the last thing needed for serious DAW. 2. Non working S/PDIF (and probably ADAT, too) on 1616(M) microdock ports. I think tese ports uses different set of registers than PCI 1010 host card. Am I right? 3. Midi I/O works well on 1212m v1, but only when 48kHz is set. At 44.1kHz signalling is corrupted (different UART timing? How to fix it?). On 1616m microdock MIDI I/O does not work at all. I welcome any comments, advices and patches ;) ---- Ctirad

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[alsa-devel] [CONF] All C-Media Codecs based 5.1 Channel USB- Headsets
by thomas schorpp 18 Nov '07

18 Nov '07

Report to devlist as requested from USB-Audio.conf header: 1. all this headsets with windriver inf files + manual showing c-media driver must have the cm106x series 7.1 channel usb-codec in. there's no other from c-media for the design of 5.1 channel usb headsets available at this time, see: http://www.cmedia.com.tw/?q=en/usb_audio 2. the headset interface shows up with 8 channels in lsusb -v , 1 8-ch pcm in /proc, and must be driven with surround7.1 pcm plug instead of surround5.1 or it will not work. speaker-test program works with -c6 -Dsurround7.1, test with -c8 works too, but shows that no SideL/R -speakers connected/routed/downmixed ( Speedlink Medusa 5.1 Mobile USB Headset, cm106L) 3. it works instantly without extra config using programs that allow specifying (-D)surround7.1 like VDR and MPlayer, VLC did not so far (maybe can do with special commandline) and therere maybe other unconfigurable (or buggy?) players that need a missing extra cm106 CARD configfile showing those apps a 5.1 channel device instead a 7.1. i've done none so far, the config file language (especially routing and mappings) is not well documented, sorry. y tom

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[alsa-devel] Is there a function I can call from my ISR to tell ALSA to abort playback?
by Timur Tabi 16 Nov '07

16 Nov '07

In my ISR, I call snd_pcm_period_elapsed() to tell it that a period has elapsed. I can also detect if playback has aborted for some reason (e.g. DMA controller has died). Is there a function I can call in my ISR to tell ALSA to abort playback or capture? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] [PATCH] HDA-Intel - Add support for RV610/RV630 HDMI audio
by Herton Ronaldo Krzesinski 16 Nov '07

16 Nov '07

The Audio interface on HD2400/HD2600 cards isn't currently detected by snd-hda-intel. I added missing pci device ids for RV610 and RV630, but I only had a HD2400 pro card to test, where now the audio interface is detected (and no need to change patch_atihdmi.c, as the codec vendor id remains 0x1002aa01 for which we already have an entry there). I also couldn't test if sound pass-trough is ok (and I don't know how to), but at least now the device is detected. Signed-off-by: Herton Ronaldo Krzesinski <herton(a)mandriva.com.br> --- a/sound/pci/hda/hda_intel.c.orig 2007-11-02 01:59:29.000000000 -0200 +++ b/sound/pci/hda/hda_intel.c 2007-11-02 01:59:58.000000000 -0200 @@ -104,6 +104,8 @@ MODULE_SUPPORTED_DEVICE("{{Intel, ICH6}, "{ATI, RS690}," "{ATI, RS780}," "{ATI, R600}," + "{ATI, RV630}," + "{ATI, RV610}," "{VIA, VT8251}," "{VIA, VT8237A}," "{SiS, SIS966}," @@ -1945,6 +1947,8 @@ static struct pci_device_id azx_ids[] = { 0x1002, 0x7919, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS690 HDMI */ { 0x1002, 0x960c, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RS780 HDMI */ { 0x1002, 0xaa00, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI R600 HDMI */ + { 0x1002, 0xaa08, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV630 HDMI */ + { 0x1002, 0xaa10, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ATIHDMI }, /* ATI RV610 HDMI */ { 0x1106, 0x3288, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_VIA }, /* VIA VT8251/VT8237A */ { 0x1039, 0x7502, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_SIS }, /* SIS966 */ { 0x10b9, 0x5461, PCI_ANY_ID, PCI_ANY_ID, 0, 0, AZX_DRIVER_ULI }, /* ULI M5461 */ -- []'s Herton

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