June 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] Listing raw midi devices "The Right Way"?
by Jacob Ole Juul Kolding 18 Jun '08

18 Jun '08

Hello all I'm working on an app that used to run on Linux but then moved to OS X and gained MIDI capabilities. Now it's moving back to Linux so I need to be able to read midi events from devices. I already found a code example that reads raw midi packages form a specified device but I need to have a complete list of ALL raw midi capable devices and ports on the system. What is the "Right Way" for an app to build such a list? Sincerely Jacob Kolding Juvul Tech

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[alsa-devel] [PATCH] ASoC: fix PM=n build
by Mark Brown 17 Jun '08

17 Jun '08

From: Randy Dunlap <randy.dunlap(a)oracle.com> Fix sound/soc build failure when CONFIG_PM=n: linux-next-20080617/sound/soc/soc-core.c:829: error: 'soc_resume_deferred' undeclared (first use in this function) soc3.out:make[3]: *** [sound/soc/soc-core.o] Error 1 Signed-off-by: Randy Dunlap <randy.dunlap(a)oracle.com> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index b931039..bdbbc6a 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -825,8 +825,10 @@ static int soc_probe(struct platform_device *pdev) /* DAPM stream work */ INIT_DELAYED_WORK(&socdev->delayed_work, close_delayed_work); +#ifdef CONFIG_PM /* deferred resume work */ INIT_WORK(&socdev->deferred_resume_work, soc_resume_deferred); +#endif return 0; -- 1.5.2.5

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[alsa-devel] hda-intel on Asus M2N-X Plus NVIDIA GeForce 6100/nForce 430 mobo
by Nigel Henry 17 Jun '08

17 Jun '08

I've just built a new machine, and expected problems with the onboard soundcard, and also have problems with the onboard NIC, but that's another problem, which is resolved by installing my last remaining realtek 8139 PCI card. Anyway the mobo is an Asus M2N-X Plus NVIDIA GeForce 6100/nForce 430, and the Southbridge (which appears to handle allsorts of stuff) is MCP61. the specs for the soundcard from the Asus site are below. ALC662 6-Channel High Definition Audio Azalia CODEC Support Jack-Sensing, Enumeration, Multi-streaming S/PDIF out on back I/O port This is using the Fedora 8 distro, which has alsa driver 1.0.15, and is probably a bit old. The card is not detected at bootup, but I did a modprobe snd-hda-intel, using various model options. the modules were loaded, but a cat /proc/asound/cards showed "no soundcards detected" I don't want to waste too much time on this onboard card, and will probably buy a cheap well supported card, just to get some sounds on this new machine. All the same I've run the alsa-info.sh script, and the link is below. http://pastebin.ca/1048647 Any comments, as always welcome. Btw: My other machines use Audigy2 soundblaster (emu10k1) on the first machine, and Ensoniq (ens1371) on the other machine, and no problems with these cards at all. I cannot repeat my opinions/thoughts about hda-intel cards. Thanks for all your work you folks. Nigel.

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[alsa-devel] Thoughts on ASOC v2 driver architecture
by Jon Smirl 17 Jun '08

17 Jun '08

I'm trying to sort out an architecture that will let all of the code used in my ASOC drivers to be automatically loaded by the kernel. So if I'm confused and have some of this wrong please explain where I've gone astray. First, what are the physical hardware devices present? From the device tree it says I have two i2s controllers each with an i2c controlled codec. It also says I have a system wide DMA controller. And finally there is the "fabric" which represent the specific pcb (populated jacks, etc) used by the system (the root of the device tree). Upon booting the kernel it will find the five open firmware devices (two i2s, two codecs, one DMA) and load the appropriate of drivers for them. Note that this is the system wide DMA driver, not the ASOC one. Three drivers will be loaded and five devices will be created. The codec devices will each register one codec and multiple codec DAIs. Each i2c channel will register a platform_dai. The next part, the platform device, is where I'm having trouble. Why is the DMA subsystem in it's own ASOC device? I already have a separate system wide device for my DMA hardware. Looking at the implementations of a couple of the ASOC DMA drivers, there is nothing global about the driver, everything is done with per channel instances. Should the DMA code instead simply be a library that is linked to by the i2s drivers? And then move the entry points registered by the DMA driver up into the platform_dai? It seems to me like implementation of the DMA code should be a private matter for the platform_dai/i2s driver. Note that I'm not saying that there shouldn't be modularized ASOC DMA code, instead, why should ASOC be aware of this shared code? To see if this model would work I tried to visualize a system where one i2s controller is using DMA and the second is doing programmed IO. So if the DMA API is pushed into the platform_dai that would free up the platform device. It could then be used to support the fabric code (fabric is the motherboard specific bits of code). It should probably be renamed to the machine device instead of platform device. To handle the case of HDA not needing a fabric driver, ASOC would have a default internal machine driver that a fabric driver could override. This also implies that the fabric driver can't be the last driver registered with ASOC. There would be one global fabric driver, but it would get called on a per instance basis. You could implement (made up scenarios) two amps on the motherboard and only enough power to turn one of them on. Or maybe a front panel switch that switches which channel the front panel controls act on. I'd end up with this object model generic machine (optional machine specific fabric override) platform i2s (platform_dai, soc_snd_card) generic codec (choice of codec dai) platform i2s (platform_dai, soc_snd_card) generic codec (choice of codec dai) The platform specific i2s drivers are linked to a shared library of platform DMA code. -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] git-http interface to ALSA git repositories?
by Timur Tabi 17 Jun '08

17 Jun '08

My company's firewall blocks the git protocol, so I need to use http:// to clone git repositories. What is the http path for the ALSA git repository? -- Timur Tabi Linux kernel developer at Freescale

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[alsa-devel] Mute sound
by Jorge Peixoto de Morais Neto 17 Jun '08

17 Jun '08

Hi. Under kernel 2.6.23, I could get sound from the alsa device created by the module snd_bt87x. In kernel 2.6.24, there is no sound. Also, in kernel 2.6.23 the device would only appear if I modprobed the module with the option load_all=1. In kernel 2.6.24, the option load_all=1 is unnecessary for the device to appear. This is probably because I have in 2007 sent an email to this list with information about my board (it was responded by Takashi). If you need any addition other than what I provide below, ask at any time. jorge@jorge ~ $ uname -a Linux jorge 2.6.24-gentoo-r8d #4 Mon Jun 16 17:58:17 BRT 2008 i686 AMD Athlon(tm) XP 2600+ AuthenticAMD GNU/Linux jorge@jorge ~ $ sudo lspci -v 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host Bridge Subsystem: VIA Technologies, Inc. Unknown device 0000 Flags: bus master, 66MHz, medium devsel, latency 8 Memory at e0000000 (32-bit, prefetchable) [size=128M] Capabilities: [a0] AGP version 2.0 Capabilities: [c0] Power Management version 2 00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge (prog-if 00 [Normal decode]) Flags: bus master, 66MHz, medium devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 Memory behind bridge: dde00000-dfefffff Prefetchable memory behind bridge: cdc00000-ddcfffff Capabilities: [80] Power Management version 2 00:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) Subsystem: Realtek Semiconductor Co., Ltd. RT8139 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at ec00 [size=256] Memory at dfffff00 (32-bit, non-prefetchable) [size=256] Capabilities: [50] Power Management version 2 Kernel driver in use: 8139too 00:0b.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) Subsystem: PROLINK Microsystems Corp Unknown device 4011 Flags: bus master, medium devsel, latency 32, IRQ 5 Memory at dddfe000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data <?> Capabilities: [4c] Power Management version 2 Kernel driver in use: bttv Kernel modules: bttv 00:0b.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) Subsystem: PROLINK Microsystems Corp Unknown device 4011 Flags: bus master, medium devsel, latency 32, IRQ 5 Memory at dddff000 (32-bit, prefetchable) [size=4K] Capabilities: [44] Vital Product Data <?> Capabilities: [4c] Power Management version 2 Kernel driver in use: Bt87x Kernel modules: snd-bt87x 00:0c.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) Subsystem: Micro-Star International Co., Ltd. Unknown device 5900 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at e800 [size=256] Capabilities: [c0] Power Management version 2 Kernel driver in use: C-Media PCI Kernel modules: snd-cmipci 00:10.0 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at dc00 [size=32] Capabilities: [80] Power Management version 2 Kernel driver in use: uhci_hcd 00:10.1 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at e000 [size=32] Capabilities: [80] Power Management version 2 Kernel driver in use: uhci_hcd 00:10.2 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) (prog-if 00 [UHCI]) Subsystem: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at e400 [size=32] Capabilities: [80] Power Management version 2 Kernel driver in use: uhci_hcd 00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82) (prog-if 20 [EHCI]) Subsystem: VIA Technologies, Inc. USB 2.0 Flags: bus master, medium devsel, latency 32, IRQ 10 Memory at dffffe00 (32-bit, non-prefetchable) [size=256] Capabilities: [80] Power Management version 2 Kernel driver in use: ehci_hcd 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge Subsystem: VIA Technologies, Inc. Unknown device 0000 Flags: bus master, stepping, medium devsel, latency 0 Capabilities: [c0] Power Management version 2 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. VT82C586/B/VT82C686/A/B/VT8233/A/C/VT8235 PIPC Bus Master IDE Flags: bus master, medium devsel, latency 32, IRQ 255 [virtual] Memory at 000001f0 (32-bit, non-prefetchable) [size=8] [virtual] Memory at 000003f0 (type 3, non-prefetchable) [size=1] [virtual] Memory at 00000170 (32-bit, non-prefetchable) [size=8] [virtual] Memory at 00000370 (type 3, non-prefetchable) [size=1] I/O ports at fc00 [size=16] Capabilities: [c0] Power Management version 2 01:00.0 VGA compatible controller: nVidia Corporation NV17 [GeForce4 MX 440] (rev a3) (prog-if 00 [VGA controller]) Flags: bus master, 66MHz, medium devsel, latency 32, IRQ 11 Memory at de000000 (32-bit, non-prefetchable) [size=16M] Memory at d0000000 (32-bit, prefetchable) [size=128M] Memory at ddc80000 (32-bit, prefetchable) [size=512K] Expansion ROM at dfee0000 [disabled] [size=128K] Capabilities: [60] Power Management version 2 Capabilities: [44] AGP version 2.0 jorge@jorge ~ $ cat /proc/cpuinfo processor : 0 vendor_id : AuthenticAMD cpu family : 6 model : 8 model name : AMD Athlon(tm) XP 2600+ stepping : 1 cpu MHz : 2133.528 cache size : 256 KB fdiv_bug : no hlt_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 1 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 mmx fxsr sse syscall mmxext 3dnowext 3dnow ts bogomips : 4271.41 clflush size : 32 jorge@jorge /proc/asound $ ls Bt878 card0 card1 cards CMI8738 devices hwdep oss pcm timers version jorge@jorge /proc/asound $ cat cards 0 [CMI8738 ]: CMI8738-MC6 - C-Media CMI8738 C-Media CMI8738 (model 55) at 0xe800, irq 11 1 [Bt878 ]: Bt87x - Brooktree Bt878 Brooktree Bt878 at 0xdddff000, irq 5 jorge@jorge /proc/asound $ cat devices 0: [ 0] : control 4: [ 0- 0]: hardware dependent 8: [ 0- 0]: raw midi 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 18: [ 0- 2]: digital audio playback 24: [ 0- 0]: digital audio capture 26: [ 0- 2]: digital audio capture 32: [ 1] : control 33: : timer 56: [ 1- 0]: digital audio capture 57: [ 1- 1]: digital audio capture jorge@jorge /proc/asound $ cat hwdep 00-00: OPL3 FM jorge@jorge /proc/asound $ cat pcm 00-02: CMI8738-MC6 : C-Media PCI IEC958 : playback 1 : capture 1 00-01: CMI8738-MC6 : C-Media PCI 2nd DAC : playback 1 00-00: CMI8738-MC6 : C-Media PCI DAC/ADC : playback 1 : capture 1 01-01: Bt87x Analog : Bt87x Analog : capture 1 01-00: Bt87x Digital : Bt87x Digital : capture 1 jorge@jorge /proc/asound $ cat version Advanced Linux Sound Architecture Driver Version 1.0.15 (Tue Nov 20 19:16:42 2007 UTC). -- Software is like sex: it is better when it is free - Linus Torvalds

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Re: [alsa-devel] Problem regarding CA0106
by Travis Place 16 Jun '08

16 Jun '08

On Mon, 2008-06-16 at 18:00 +0200, Takashi Iwai wrote: > At Tue, 17 Jun 2008 01:52:08 +1000, > Travis Place wrote: > > > > On Mon, 2008-06-16 at 12:38 +0200, Takashi Iwai wrote: > > > At Sat, 14 Jun 2008 03:18:33 +1000, > > > Travis Place wrote: > > > > > > > > Just been speaking with a guy in #alsa about the problems with his "MSI > > > > K8N Diamond MB" which has an onboard "SB Live 24bit" or "Audigy". > > > > > > > > While looking through ca0106_main.c in GIT, i saw the following section: > > > > > > > > /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 > > > > */ > > > > /* SB0438 > > > > * CTRL:CA0106-DAT > > > > * ADC: WM8775SEDS > > > > * DAC: CS4382-KQZ > > > > */ > > > > { .serial = 0x10091462, > > > > .name = "MSI K8N Diamond MB [SB0438]", > > > > .gpio_type = 2, > > > > .i2c_adc = 1 } , > > > > > > > > For this guys sound to work, we had to change the serial line to: > > > > > > > > { .serial = 0x10091102, > > > > > > Does it mean that this number 1009:1102 is missing, or a wrong > > > 1009:1462 entry matches with his hardware? > > > > > Well, it appears to me (due to another report filed previously) that > > the motherboard is a 'newer revision' and has a different ID for the > > soundcard. > > Hm, so you didn't receive alsa-info.sh or "lspci -nv" output? > Of course I got alsa-info.sh output :) (asks himself why he didnt include it though) http://pastebin.ca/1048644 > > > In the former case, it happens often. We just need to add a new entry > > > for the newer model. In the latter case, it's odd. > > > > > > > I think that adding the new ID to the driver wont cause any immediate > > problems. The old value (1009:1462) should remain, until we can confirm > > it against another K8N Diamond motherboard (although im not sure how > > common these boards are). > > Yeah, I guess so, too. But just to be sure... > Well, we'll just have to see what happens, I guess. For the record, in both older ALSA versions, and 1.0.16, changing only the serial seems to be sufficient. So adding another serial should be an easy fix. Thanks, Travis > > Takashi

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[alsa-devel] [Fwd: Re: Problem regarding CA0106]
by Travis Place 16 Jun '08

16 Jun '08

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[alsa-devel] alsa and 2.6.25.4 kernel
by linuxconsole＠free.fr 16 Jun '08

16 Jun '08

Hi, I can't play sound or use mixer with alsa on 2.6.25.4 kernel. I tried all alsa-libs (1.16, 1.17rc1, hg) without success. the file /proc/asound/cards is : cat /proc/asound/cards 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xf0100000 irq 18 strace -e open amixer : ... open("/dev/snd/controlC0", O_RDONLY) = -1 ENODEV (No such device) open("/dev/aloadC0", O_RDONLY) = -1 ENODEV (No such device) ls -l /dev/snd/controlC0 crw-rwxrw- 1 root users 116, 0 Aug 23 2004 /dev/snd/controlC0 what happens ??? Regards. Yann.

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[alsa-devel] Function to update the PCM status in alsa lib?
by anuj aggarwal 16 Jun '08

16 Jun '08

Hi, I am calling snd_pcm_period_elapsed() in my DMA callback routine. After doing that, I check for more data in the 'runtime' struct and pass a new request to xfer data to the codec driver. snd_pcm_period_elapsed() is used to update the pcm status for the next period, by the codec driver. I want to know which function is used by the ALSA library to do the same when it receives a new request from the user to transfer data (i.e. when user calls snd_pcm_writei()). Thanks, Anuj

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[alsa-devel] [PATCH 1/2] ad1848: separate header for registers definition
by Krzysztof Helt 16 Jun '08

16 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Separate AD1848 registers definition into a new header. The new header can be used by compatible CS4231 chip driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch is a start combining ad1848_lib and cs4231_lib functionality. My approach is to make the cs4231_lib module use the ad1848_lib module. This should allow removing most of identical functions from the cs4231_lib and use the functions from the ad1848_lib directly. If this approach is not a good idea, please drop the patch. My current plan is to reuse the ad1848 register constants in the cs423x code and remove duplicated cs4231 constants. This should make the ad1848_lib and cs4231_lib easier to compare for differences. The next step is to use the snd_cs4231 structure instead of the snd_ad1848 structure inside the ad1848_lib. This will make functions from the ad1848_lib taking the same arguments as the functions in the cs4231_lib. The last step would be to export needed functions from the ad1848_lib and use them directly in the cs4231_lib. This should reduce the size of the cs4231_lib while preserving smaller the ad1848_lib for driver which uses only ad1848 functionality. Kind regards, Krzysztof diff -urpN linux-ref/include/sound/ad1848-regs.h linux-mm/include/sound/ad1848-regs.h --- linux-ref/include/sound/ad1848-regs.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-mm/include/sound/ad1848-regs.h 2008-06-15 07:00:08.041351281 +0200 @@ -0,0 +1,104 @@ +#ifndef __SOUND_AD1848_REGS_H +#define __SOUND_AD1848_REGS_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Definitions for AD1847/AD1848/CS4248 chips registers + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* IO ports */ + +#define AD1848P(chip, x) ((chip)->port + c_d_c_AD1848##x) + +#define c_d_c_AD1848REGSEL 0 +#define c_d_c_AD1848REG 1 +#define c_d_c_AD1848STATUS 2 +#define c_d_c_AD1848PIO 3 + +/* codec registers */ + +#define AD1848_LEFT_INPUT 0x00 /* left input control */ +#define AD1848_RIGHT_INPUT 0x01 /* right input control */ +#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ +#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ +#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ +#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ +#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ +#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ +#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ +#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ +#define AD1848_PIN_CTRL 0x0a /* pin control */ +#define AD1848_TEST_INIT 0x0b /* test and initialization */ +#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ +#define AD1848_LOOPBACK 0x0d /* loopback control */ +#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ +#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ + +/* definitions for codec register select port - CODECP( REGSEL ) */ + +#define AD1848_INIT 0x80 /* CODEC is initializing */ +#define AD1848_MCE 0x40 /* mode change enable */ +#define AD1848_TRD 0x20 /* transfer request disable */ + +/* definitions for codec status register - CODECP( STATUS ) */ + +#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ + +/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ + +#define AD1848_ENABLE_MIC_GAIN 0x20 + +#define AD1848_MIXS_LINE1 0x00 +#define AD1848_MIXS_AUX1 0x40 +#define AD1848_MIXS_LINE2 0x80 +#define AD1848_MIXS_ALL 0xc0 + +/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ + +#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ +#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ +#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ +#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ +#define AD1848_STEREO 0x10 /* stereo mode */ +/* bits 3-1 define frequency divisor */ +#define AD1848_XTAL1 0x00 /* 24.576 crystal */ +#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ + +/* definitions for interface control register - AD1848_IFACE_CTRL */ + +#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ +#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ +#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ +#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ +#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ +#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ +#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ + +/* definitions for pin control register - AD1848_PIN_CTRL */ + +#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ +#define AD1848_XCTL1 0x40 /* external control #1 */ +#define AD1848_XCTL0 0x80 /* external control #0 */ + +/* definitions for test and init register - AD1848_TEST_INIT */ + +#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ +#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ + +#endif /* __SOUND_AD1848_REGS_H */ diff -urpN linux-ref/include/sound/ad1848.h linux-mm/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-06-15 06:58:15.293013657 +0200 +++ linux-mm/include/sound/ad1848.h 2008-06-15 00:14:36.728628406 +0200 @@ -25,84 +25,7 @@ #include "pcm.h" #include <linux/interrupt.h> -/* IO ports */ - -#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) - -#define c_d_c_AD1848REGSEL 0 -#define c_d_c_AD1848REG 1 -#define c_d_c_AD1848STATUS 2 -#define c_d_c_AD1848PIO 3 - -/* codec registers */ - -#define AD1848_LEFT_INPUT 0x00 /* left input control */ -#define AD1848_RIGHT_INPUT 0x01 /* right input control */ -#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ -#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ -#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ -#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define AD1848_PIN_CTRL 0x0a /* pin control */ -#define AD1848_TEST_INIT 0x0b /* test and initialization */ -#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ -#define AD1848_LOOPBACK 0x0d /* loopback control */ -#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ -#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define AD1848_INIT 0x80 /* CODEC is initializing */ -#define AD1848_MCE 0x40 /* mode change enable */ -#define AD1848_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ - -#define AD1848_ENABLE_MIC_GAIN 0x20 - -#define AD1848_MIXS_LINE1 0x00 -#define AD1848_MIXS_AUX1 0x40 -#define AD1848_MIXS_LINE2 0x80 -#define AD1848_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ - -#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ -#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ -#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define AD1848_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define AD1848_XTAL1 0x00 /* 24.576 crystal */ -#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - AD1848_IFACE_CTRL */ - -#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ -#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ -#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ -#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ -#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ -#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - AD1848_PIN_CTRL */ - -#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ -#define AD1848_XCTL1 0x40 /* external control #1 */ -#define AD1848_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - AD1848_TEST_INIT */ - -#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ +#include "ad1848-regs.h" /* defines for codec.mode */ ---------------------------------------------------------------------- Tania telefonia internetowa! Sprawdz >>> http://link.interia.pl/f1e2e

3 3

[alsa-devel] Problem regarding CA0106
by Travis Place 16 Jun '08

16 Jun '08

Just been speaking with a guy in #alsa about the problems with his "MSI K8N Diamond MB" which has an onboard "SB Live 24bit" or "Audigy". While looking through ca0106_main.c in GIT, i saw the following section: /* MSI K8N Diamond Motherboard with onboard SB Live 24bit without AC97 */ /* SB0438 * CTRL:CA0106-DAT * ADC: WM8775SEDS * DAC: CS4382-KQZ */ { .serial = 0x10091462, .name = "MSI K8N Diamond MB [SB0438]", .gpio_type = 2, .i2c_adc = 1 } , For this guys sound to work, we had to change the serial line to: { .serial = 0x10091102, Seems to be a bit hackish, and maybe some more investigation is required, but, it got sound to work, and the card is recognised. Perhaps there is (for some odd reason) 2 variants of the onboard sound on that motherboard (Perhaps a rev 1.0 and rev 1.1 motherboard difference?) Let me know if any more information is required, or if I can just make and submit a patch (ill add a new section for the new card, not the hackish way). Thanks, Travis Place

2 1

[alsa-devel] [PATCH 1/2] opti93x: add support for Opti93x codec in cs4231-lib
by Krzysztof Helt 15 Jun '08

15 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch adds support for WSS compatible Opti93x codec to the cs4231-lib. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch was sent some time ago but was not accepted due to regression of the driver on the opti 930 chip. The regression is fixed with already accepted patch, so it should work now. The patch was prepared against the 2.6.26-rc5 kernel. Tested on Opti 930, 931 and 933. The previous repost: http://mailman.alsa-project.org/pipermail/alsa-devel/2008-January/005679.ht… diff -urp linux-rc2/include/sound/cs4231-regs.h linux-rc5/include/sound/cs4231-regs.h --- linux-rc2/include/sound/cs4231-regs.h 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/include/sound/cs4231-regs.h 2008-06-08 00:33:55.041903713 +0200 @@ -177,4 +177,12 @@ #define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ #define CS4236_VERSION 0x9c /* chip version and ID */ +/* definitions for extended registers - OPTI93X */ +#define OPTi931_AUX_LEFT_INPUT 0x10 +#define OPTi931_AUX_RIGHT_INPUT 0x11 +#define OPTi93X_MIC_LEFT_INPUT 0x14 +#define OPTi93X_MIC_RIGHT_INPUT 0x15 +#define OPTi93X_OUT_LEFT 0x16 +#define OPTi93X_OUT_RIGHT 0x17 + #endif /* __SOUND_CS4231_REGS_H */ diff -urp linux-rc2/include/sound/cs4231.h linux-rc5/include/sound/cs4231.h --- linux-rc2/include/sound/cs4231.h 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/include/sound/cs4231.h 2008-06-08 00:35:54.808064369 +0200 @@ -58,6 +58,7 @@ /* compatible, but clones */ #define CS4231_HW_INTERWAVE 0x1000 /* InterWave chip */ #define CS4231_HW_OPL3SA2 0x1101 /* OPL3-SA2 chip, similar to cs4231 */ +#define CS4231_HW_OPTI93X 0x1102 /* Opti 930/931/933 */ /* defines for codec.hwshare */ #define CS4231_HWSHARE_IRQ (1<<0) @@ -120,6 +121,8 @@ unsigned char snd_cs4236_ext_in(struct s void snd_cs4231_mce_up(struct snd_cs4231 *chip); void snd_cs4231_mce_down(struct snd_cs4231 *chip); +void snd_cs4231_overrange(struct snd_cs4231 *chip); + irqreturn_t snd_cs4231_interrupt(int irq, void *dev_id); const char *snd_cs4231_chip_id(struct snd_cs4231 *chip); diff -urp linux-rc2/sound/isa/cs423x/cs4231_lib.c linux-rc5/sound/isa/cs423x/cs4231_lib.c --- linux-rc2/sound/isa/cs423x/cs4231_lib.c 2008-04-17 04:49:44.000000000 +0200 +++ linux-rc5/sound/isa/cs423x/cs4231_lib.c 2008-06-08 00:38:32.130878968 +0200 @@ -119,6 +119,42 @@ static unsigned char snd_cs4231_original 0x00, /* 1f/31 - cbrl */ }; +static unsigned char snd_opti93x_original_image[32] = +{ + 0x00, /* 00/00 - l_mixout_outctrl */ + 0x00, /* 01/01 - r_mixout_outctrl */ + 0x88, /* 02/02 - l_cd_inctrl */ + 0x88, /* 03/03 - r_cd_inctrl */ + 0x88, /* 04/04 - l_a1/fm_inctrl */ + 0x88, /* 05/05 - r_a1/fm_inctrl */ + 0x80, /* 06/06 - l_dac_inctrl */ + 0x80, /* 07/07 - r_dac_inctrl */ + 0x00, /* 08/08 - ply_dataform_reg */ + 0x00, /* 09/09 - if_conf */ + 0x00, /* 0a/10 - pin_ctrl */ + 0x00, /* 0b/11 - err_init_reg */ + 0x0a, /* 0c/12 - id_reg */ + 0x00, /* 0d/13 - reserved */ + 0x00, /* 0e/14 - ply_upcount_reg */ + 0x00, /* 0f/15 - ply_lowcount_reg */ + 0x88, /* 10/16 - reserved/l_a1_inctrl */ + 0x88, /* 11/17 - reserved/r_a1_inctrl */ + 0x88, /* 12/18 - l_line_inctrl */ + 0x88, /* 13/19 - r_line_inctrl */ + 0x88, /* 14/20 - l_mic_inctrl */ + 0x88, /* 15/21 - r_mic_inctrl */ + 0x80, /* 16/22 - l_out_outctrl */ + 0x80, /* 17/23 - r_out_outctrl */ + 0x00, /* 18/24 - reserved */ + 0x00, /* 19/25 - reserved */ + 0x00, /* 1a/26 - reserved */ + 0x00, /* 1b/27 - reserved */ + 0x00, /* 1c/28 - cap_dataform_reg */ + 0x00, /* 1d/29 - reserved */ + 0x00, /* 1e/30 - cap_upcount_reg */ + 0x00 /* 1f/31 - cap_lowcount_reg */ +}; + /* * Basic I/O functions */ @@ -895,7 +931,7 @@ static int snd_cs4231_capture_prepare(st return 0; } -static void snd_cs4231_overrange(struct snd_cs4231 *chip) +void snd_cs4231_overrange(struct snd_cs4231 *chip) { unsigned long flags; unsigned char res; @@ -1054,8 +1090,11 @@ static int snd_cs4231_probe(struct snd_c chip->image[CS4231_IFACE_CTRL] = (chip->image[CS4231_IFACE_CTRL] & ~CS4231_SINGLE_DMA) | (chip->single_dma ? CS4231_SINGLE_DMA : 0); - chip->image[CS4231_ALT_FEATURE_1] = 0x80; - chip->image[CS4231_ALT_FEATURE_2] = chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + if (chip->hardware != CS4231_HW_OPTI93X) { + chip->image[CS4231_ALT_FEATURE_1] = 0x80; + chip->image[CS4231_ALT_FEATURE_2] = + chip->hardware == CS4231_HW_INTERWAVE ? 0xc2 : 0x01; + } ptr = (unsigned char *) &chip->image; snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->reg_lock, flags); @@ -1376,6 +1415,7 @@ const char *snd_cs4231_chip_id(struct sn case CS4231_HW_INTERWAVE: return "AMD InterWave"; case CS4231_HW_OPL3SA2: return chip->card->shortname; case CS4231_HW_AD1845: return "AD1845"; + case CS4231_HW_OPTI93X: return "OPTi 93x"; default: return "???"; } } @@ -1401,8 +1441,13 @@ static int snd_cs4231_new(struct snd_car chip->rate_constraint = snd_cs4231_xrate; chip->set_playback_format = snd_cs4231_playback_format; chip->set_capture_format = snd_cs4231_capture_format; - memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); - + if (chip->hardware == CS4231_HW_OPTI93X) + memcpy(&chip->image, &snd_opti93x_original_image, + sizeof(snd_opti93x_original_image)); + else + memcpy(&chip->image, &snd_cs4231_original_image, + sizeof(snd_cs4231_original_image)); + *rchip = chip; return 0; } @@ -1790,6 +1835,48 @@ CS4231_SINGLE("Loopback Capture Switch", CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1) }; +static struct snd_kcontrol_new snd_opti93x_controls[] = { +CS4231_DOUBLE("Master Playback Switch", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 7, 7, 1, 1), +CS4231_DOUBLE("Master Playback Volume", 0, + OPTi93X_OUT_LEFT, OPTi93X_OUT_RIGHT, 1, 1, 31, 1), +CS4231_DOUBLE("PCM Playback Switch", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 7, 7, 1, 1), +CS4231_DOUBLE("PCM Playback Volume", 0, + CS4231_LEFT_OUTPUT, CS4231_RIGHT_OUTPUT, 0, 0, 31, 1), +CS4231_DOUBLE("FM Playback Switch", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("FM Playback Volume", 0, + CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Line Playback Switch", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 7, 7, 1, 1), +CS4231_DOUBLE("Line Playback Volume", 0, + CS4231_LEFT_LINE_IN, CS4231_RIGHT_LINE_IN, 0, 0, 15, 1), +CS4231_DOUBLE("Mic Playback Switch", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Mic Playback Volume", 0, + OPTi93X_MIC_LEFT_INPUT, OPTi93X_MIC_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Mic Boost", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("CD Playback Switch", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("CD Playback Volume", 0, + CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, + OPTi931_AUX_LEFT_INPUT, OPTi931_AUX_RIGHT_INPUT, 1, 1, 15, 1), +CS4231_DOUBLE("Capture Volume", 0, + CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Capture Source", + .info = snd_cs4231_info_mux, + .get = snd_cs4231_get_mux, + .put = snd_cs4231_put_mux, +} +}; + int snd_cs4231_mixer(struct snd_cs4231 *chip) { struct snd_card *card; @@ -1802,10 +1889,22 @@ int snd_cs4231_mixer(struct snd_cs4231 * strcpy(card->mixername, chip->pcm->name); - for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { - if ((err = snd_ctl_add(card, snd_ctl_new1(&snd_cs4231_controls[idx], chip))) < 0) - return err; - } + if (chip->hardware == CS4231_HW_OPTI93X) + for (idx = 0; idx < ARRAY_SIZE(snd_opti93x_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_opti93x_controls[idx], + chip)); + if (err < 0) + return err; + } + else + for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], + chip)); + if (err < 0) + return err; + } return 0; } @@ -1815,6 +1914,7 @@ EXPORT_SYMBOL(snd_cs4236_ext_out); EXPORT_SYMBOL(snd_cs4236_ext_in); EXPORT_SYMBOL(snd_cs4231_mce_up); EXPORT_SYMBOL(snd_cs4231_mce_down); +EXPORT_SYMBOL(snd_cs4231_overrange); EXPORT_SYMBOL(snd_cs4231_interrupt); EXPORT_SYMBOL(snd_cs4231_chip_id); EXPORT_SYMBOL(snd_cs4231_create); --------------------------------------------------------------- Sprawdz jak zdobyc zdrowy usmiech! 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2 4

[alsa-devel] Vendor support in Alsa Drivers
by Emmanuel Kasper 15 Jun '08

15 Jun '08

Hello With a couple of other french folks, we are building a kind of webshop for Linux/FreeBSD compatible hardware. We will reference only devices which work with a free driver, so no nvidia binary thingie or atheros HAL. Based on the information found on the alsa wiki we plan to reference the following sound devices: M-Audio Audiophile 24/96 ( snd-ice1712 driver ) Terratec AUREON 5.1 M-Audio Fast Track usb Hercules Muse Pocket LT Ion TTUSB usb turnable ( these 4 working with snd-usb-driver ) We would be also interested to indicate on each product description, if the vendor of the product, or the chipset maker for that matters had been helpful with developers trying to write a free driver. So for the devices mentionned above, was any of the vendor helpful by providing programming documentation or sample hardware ? If yes we will mention it on the product description with an extra tag like "programmer friendly" Regards -- Emmanuel Kasper System Administrator www.libera.cc Find, compare, buy Linux & FreeBSD compatible hardware

6 7

[alsa-devel] input source selection not working (snd-hda-intel)
by Bruno Schneider 14 Jun '08

14 Jun '08

Hi all, I'm using a computer with a ASUS P5LD2-X motherboard. The audio controller, is a Realtek ALC662. I'm using an updated Debian Lenny (testing) system. No matter what I set as the input source, sound is always recorded from the back microphone jack. Before posting here, I tried to get help from my Debian list and from the alsa-user list [1]. I'm starting to think this is a bug on the snd-hda-intel module. Please help. [1] http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg22667.html -- Bruno Schneider http://www.dcc.ufla.br/~bruno/

1 0

[alsa-devel] [PATCH][ASoC v2] Update Freescale MPC8610HPCD fabric driver to support multiple codecs
by Timur Tabi 14 Jun '08

14 Jun '08

Update the Freescale MPC8610 HPCD fabric driver to create a new platform device for each SSI in the system, instead of registering multiple PCMs under one card. This is necessary for alsamixer to control the individual codecs. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/mpc8610_hpcd.c | 236 +++++++++++++++++++++--------------------- 1 files changed, 117 insertions(+), 119 deletions(-) diff --git a/sound/soc/fsl/mpc8610_hpcd.c b/sound/soc/fsl/mpc8610_hpcd.c index 8813291..2cf7d95 100644 --- a/sound/soc/fsl/mpc8610_hpcd.c +++ b/sound/soc/fsl/mpc8610_hpcd.c @@ -40,13 +40,10 @@ struct mpc8610_hpcd_data { struct snd_soc_card soc_card; struct ccsr_guts __iomem *guts; - unsigned int num_configs; - struct snd_soc_pcm_config configs[MAX_SSI]; - struct mpc8610_hpcd_names { - char pcm_name[32]; - char ssi_name[32]; - char codec_name[32]; - } names[MAX_SSI]; + struct snd_soc_pcm_config pcm_config; + char pcm_name[32]; + char ssi_name[32]; + char codec_name[32]; }; /** @@ -91,29 +88,25 @@ static int mpc8610_hpcd_audio_init(struct snd_soc_card *soc_card) { struct snd_soc_codec *codec; struct mpc8610_hpcd_data *soc_card_data = soc_card->private_data; - unsigned int i; int ret = 0; - for (i = 0; i < soc_card_data->num_configs; i++) { - codec = snd_soc_card_get_codec(soc_card, - soc_card_data->names[i].codec_name, - soc_card_data->configs[i].codec_num); + codec = snd_soc_card_get_codec(soc_card, soc_card_data->codec_name, + soc_card_data->pcm_config.codec_num); - if (!codec) { - dev_err(soc_card->dev, "could not find codec\n"); - return -ENODEV; - } + if (!codec) { + dev_err(soc_card->dev, "could not find codec\n"); + return -ENODEV; + } - /* The codec driver should have called - * snd_soc_card_config_codec() by now. - */ - ret = snd_soc_card_init_codec(codec, soc_card); - if (ret < 0) { - dev_err(soc_card->dev, - "could not initialize codec %s-%u\n", - codec->name, codec->num); - continue; - } + /* The codec driver should have called + * snd_soc_card_config_codec() by now. + */ + ret = snd_soc_card_init_codec(codec, soc_card); + if (ret < 0) { + dev_err(soc_card->dev, + "could not initialize codec %s-%u\n", + codec->name, codec->num); + return -ENODEV; } return 0; @@ -316,10 +309,16 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) struct snd_soc_card *soc_card = NULL; struct mpc8610_hpcd_data *soc_card_data; struct device_node *guts_np = NULL; - struct device_node *ssi_np; - unsigned int ssi = 0; + struct device_node *ssi_np = + (struct device_node *)platform_get_resource(pdev, 0, 0)->start; int ret = -ENODEV; + struct device_node *codec_np; + const char *compat; + const char *p; + unsigned int bus; + unsigned int address; + soc_card_data = kzalloc(sizeof(struct mpc8610_hpcd_data), GFP_KERNEL); if (!soc_card_data) { dev_err(&pdev->dev, "could not allocate card structure\n"); @@ -353,100 +352,68 @@ static int mpc8610_hpcd_probe(struct platform_device *pdev) soc_card->private_data = soc_card_data; soc_card->dev = &pdev->dev; - /* Scan the device tree for SSI nodes. Each one we find that has a - * codec is registered. Remember, we only support MAX_SSI of these. - */ - for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { - struct snd_soc_pcm_config *pcm_config; - struct mpc8610_hpcd_names *name; - struct device_node *codec_np; - const char *compat; - const char *p; - unsigned int bus; - unsigned int address; - int i; - - name = &soc_card_data->names[ssi]; - pcm_config = &soc_card_data->configs[ssi]; - - pcm_config->name = name->pcm_name; - pcm_config->codec = name->codec_name; - pcm_config->codec_dai = name->codec_name; - pcm_config->platform = "fsl-elo"; - pcm_config->cpu_dai = name->ssi_name; - pcm_config->ops = &mpc8610_hpcd_ops; - pcm_config->playback = 1; - pcm_config->capture = 1; - - sprintf(name->pcm_name, "MPC8610HPCD.%u", ssi); - - /* Get the SSI name */ - i = of_get_integer(ssi_np, "cell-index"); - if (i < 0) { - dev_err(soc_card->dev, "no cell-index for %s\n", - ssi_np->full_name); - continue; - } - sprintf(name->ssi_name, "ssi%u", i); - - /* Get the codec name and ID */ - codec_np = find_codec(ssi_np); - if (!codec_np) { - dev_err(soc_card->dev, "missing codec node for %s\n", - ssi_np->full_name); - continue; - } - - /* We assume that the first (or only) string in the compatible - * field is the one that counts. - */ - compat = of_get_property(codec_np, "compatible", NULL); - if (!compat) { - dev_err(soc_card->dev, - "missing compatible property for %s\n", - ssi_np->full_name); - continue; - } - - /* We only care about the part after the comma */ - p = strchr(compat, ','); - strcpy(name->codec_name, p ? p + 1 : compat); - - /* Now determine the I2C bus and address of the codec */ - bus = of_get_integer(of_get_parent(codec_np), "cell-index"); - if (bus < 0) { - dev_err(soc_card->dev, - "cannot determine I2C bus number for %s\n", - codec_np->full_name); - continue; - } + soc_card_data->pcm_config.name = soc_card_data->pcm_name; + soc_card_data->pcm_config.codec = soc_card_data->codec_name; + soc_card_data->pcm_config.codec_dai = soc_card_data->codec_name; + soc_card_data->pcm_config.platform = "fsl-elo"; + soc_card_data->pcm_config.cpu_dai = soc_card_data->ssi_name; + soc_card_data->pcm_config.ops = &mpc8610_hpcd_ops; + soc_card_data->pcm_config.playback = 1; + soc_card_data->pcm_config.capture = 1; + + sprintf(soc_card_data->pcm_name, "MPC8610HPCD.%u", pdev->id); + + /* Get the SSI name */ + sprintf(soc_card_data->ssi_name, "ssi%u", pdev->id); + + /* Get the codec name and ID */ + codec_np = find_codec(ssi_np); + if (!codec_np) { + dev_err(soc_card->dev, "missing codec node for %s\n", + ssi_np->full_name); + goto error; + } - address = of_get_integer(codec_np, "reg"); - if (address < 0) { - dev_err(soc_card->dev, - "cannot determine I2C address for %s\n", - codec_np->full_name); - continue; - } + /* We assume that the first (or only) string in the compatible + * field is the one that counts. + */ + compat = of_get_property(codec_np, "compatible", NULL); + if (!compat) { + dev_err(soc_card->dev, + "missing compatible property for %s\n", + ssi_np->full_name); + goto error; + } - pcm_config->codec_num = bus << 16 | address; + /* We only care about the part after the comma */ + p = strchr(compat, ','); + strcpy(soc_card_data->codec_name, p ? p + 1 : compat); - dev_dbg(soc_card->dev, "registering cpu %s with codec %s-%x\n", - pcm_config->cpu_dai, pcm_config->codec, - pcm_config->codec_num); + /* Now determine the I2C bus and address of the codec */ + bus = of_get_integer(of_get_parent(codec_np), "cell-index"); + if (bus < 0) { + dev_err(soc_card->dev, + "cannot determine I2C bus number for %s\n", + codec_np->full_name); + goto error; + } - if (++ssi == MAX_SSI) - break; + address = of_get_integer(codec_np, "reg"); + if (address < 0) { + dev_err(soc_card->dev, + "cannot determine I2C address for %s\n", + codec_np->full_name); + goto error; } - ret = snd_soc_card_create_pcms(soc_card, soc_card_data->configs, ssi); + soc_card_data->pcm_config.codec_num = bus << 16 | address; + + ret = snd_soc_card_create_pcms(soc_card, &soc_card_data->pcm_config, 1); if (ret) { dev_err(soc_card->dev, "could not create PCMs\n"); return ret; } - soc_card_data->num_configs = ssi; - platform_set_drvdata(pdev, soc_card); ret = snd_soc_card_register(soc_card); @@ -493,7 +460,8 @@ static struct platform_driver mpc8610_hpcd_driver = { .remove = __devexit_p(mpc8610_hpcd_remove), }; -static struct platform_device *pdev; +/* Platform device pointers for each SSI */ +static struct platform_device *pdev[MAX_SSI]; /** * mpc8610_hpcd_init: fabric driver initialization. @@ -502,6 +470,9 @@ static struct platform_device *pdev; */ static int __init mpc8610_hpcd_init(void) { + struct resource res; + struct device_node *ssi_np; + unsigned int i = 0; int ret; pr_info("Freescale MPC8610 HPCD ASoC fabric driver\n"); @@ -512,12 +483,35 @@ static int __init mpc8610_hpcd_init(void) return ret; } - pdev = platform_device_register_simple(mpc8610_hpcd_driver.driver.name, - 0, NULL, 0); - if (!pdev) { - pr_err("mpc8610-hpcd: could not register device\n"); - platform_driver_unregister(&mpc8610_hpcd_driver); - return ret; + memset(pdev, 0, sizeof(pdev)); + + memset(&res, 0, sizeof(res)); + res.name = "ssi"; + + for_each_compatible_node(ssi_np, NULL, "fsl,mpc8610-ssi") { + int id; + + id = of_get_integer(ssi_np, "cell-index"); + if (id < 0) { + pr_err("mpc8610-hpcd: no cell-index for %s\n", + ssi_np->full_name); + continue; + } + if (!find_codec(ssi_np)) + /* No codec node? Skip it */ + continue; + + res.start = (resource_size_t) ssi_np; + + pdev[i] = platform_device_register_simple( + mpc8610_hpcd_driver.driver.name, id, &res, 1); + if (!pdev[i]) { + pr_err("mpc8610-hpcd: could not register %s\n", + ssi_np->full_name); + continue; + } + + i++; } return 0; @@ -530,7 +524,11 @@ static int __init mpc8610_hpcd_init(void) */ static void __exit mpc8610_hpcd_exit(void) { - platform_device_unregister(pdev); + unsigned int i; + + for (i = 0; i < MAX_SSI; i++) + platform_device_unregister(pdev[i]); + platform_driver_unregister(&mpc8610_hpcd_driver); } -- 1.5.5

4 26

[alsa-devel] Specs available for Sweex External 5.1 with digital in/out USB 2.0 / C-Media CM6207
by Erik Slagter 14 Jun '08

14 Jun '08

Hi there, Today I bought this sound device, I was hoping it had the C-Media CM650? USB chip which is already supported by alsa (according to the list). Instead it has a CM6207 inside. The good news is that it supports USB audio "class" and it's more or less usable, some mixer outputs are available (although they're all called "speaker") and with some trickery the various outputs can be used. The other good news is that Sweex supplies the "official" datasheet from C-media which I think is quite interesting and may very well be useful to resolve the last few issues. These are the actual issues (so far, I didn't yet test it exhaustively): - using 2 channel output the sound is garbled; it looks like the "wrong" 2 channel alt setting is used (output of /proc/asound/card?/stream0 on request), 4/6 channel output is OK though. - some high squeeking noise although C-Media claims the IC to be high-end (or maybe their definition of high-end differs from mine ;-)) - all outputs are labelled "speaker 1" - only two outputs have a mute knob, the others cannot be silenced in any way, only turned down - one of the two outputs with the mute knob doesn't appear to do anything (or maybe it's the sp/dif output?) - the other (with mute) controls the front output - the others (without mute) don't seem to do anything at all Is this a typical case of (a) bug(s) in the usb audio class driver or should I think of "vendor extensions" here? Would any of the developers have some interest in the datasheet of the chip and or any data of this thing? Of course I'd like it very much if the device would be fully supported but I can only ask nicely ;-)

4 9

[alsa-devel] [PATCH] hda: Add support for 92HD73xxx codecs
by Matthew Ranostay 13 Jun '08

13 Jun '08

Added support for new family of IDT codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a0..115cee6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -349,7 +349,7 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { static hda_nid_t stac92hd71bxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x14, 0x18, 0x19, 0x1e, 0x0f }; static hda_nid_t stac927x_pin_nids[14] = { @@ -641,15 +641,21 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 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}, /* Speaker */ + /* connect port 0d to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x02}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0x0d */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + +static struct hda_verb stac92hd71bxx_enable_portf[] = { + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for nodes 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -1313,20 +1319,20 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x0181302e, 0x01019020, 0x90a000f0, 0x90a000f0, + 0x01452050, 0x01114010, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x40f000f0, 0x90a000f0, 0x40f000f0, + 0x4f0000f0, 0x23014250, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x23a1902e, 0x40f000f0, 0x40f000f0, 0x40f000f0, + 0x044413b0, 0x23014250, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -2633,7 +2639,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + if (def_conf && (def_conf == AC_JACK_PORT_NONE)) continue; nid = spec->dmic_nids[i]; @@ -3103,13 +3109,14 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3614,12 +3621,39 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: + 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; + case 0x111d7603: /* 6 Port with Analog Mixer */ + case 0x111d7608: + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + if (codec->vendor_id != 0x111d7608) + snd_hda_sequence_write(codec, stac92hd71bxx_enable_portf); + else { + /* skip the VSW on port f */ + spec->num_pins--; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + } + if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); @@ -3634,19 +3668,6 @@ 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; @@ -3666,7 +3687,6 @@ again: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->multiout.num_dacs = 1; @@ -4306,10 +4326,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .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 },

2 3

[alsa-devel] [PATCH 1/2] ASoC: Check for exact register match in wm97xx_reset()
by Mark Brown 13 Jun '08

13 Jun '08

To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f..4739011 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9712_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9712_reg[0]) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba3301..0db9637 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == wm9713_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != wm9713_reg[0]) return -EIO; return 0; } -- 1.5.5.4

2 2

[alsa-devel] [PATCH 1/2] ASoC: Check for exact register match in wm97xx_reset()
by Mark Brown 13 Jun '08

13 Jun '08

To provide added robustness in case an AC97 controller reads back all zeros in error cases check for an exact match when testing to see if resets have brought the codec back. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm9712.c | 4 ++-- sound/soc/codecs/wm9713.c | 4 ++-- 2 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm9712.c b/sound/soc/codecs/wm9712.c index 28ac66f..09e2985 100644 --- a/sound/soc/codecs/wm9712.c +++ b/sound/soc/codecs/wm9712.c @@ -589,12 +589,12 @@ static int wm9712_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == 0x6174) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != 0x6174) goto err; return 0; diff --git a/sound/soc/codecs/wm9713.c b/sound/soc/codecs/wm9713.c index aba3301..131bb5b 100644 --- a/sound/soc/codecs/wm9713.c +++ b/sound/soc/codecs/wm9713.c @@ -1076,12 +1076,12 @@ int wm9713_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); - if (!(ac97_read(codec, 0) & 0x8000)) + if (ac97_read(codec, 0) == 0x6174) return 1; } soc_ac97_ops.reset(codec->ac97); - if (ac97_read(codec, 0) & 0x8000) + if (ac97_read(codec, 0) != 0x6174) return -EIO; return 0; } -- 1.5.5.4

2 3

[alsa-devel] ALSA 1.0.17rc1 release
by Jaroslav Kysela 13 Jun '08

13 Jun '08

Hi all, ALSA 1.0.17rc1 packages (except alsa-plugins & alsa-oss - no changes) are available for download on http://www.alsa-project.org . Only difference between testing release is a change in alsa-driver for better support of RedHat kernels. http://www.alsa-project.org/main/index.php/Changes_v1.0.16_v1.0.17rc1 Jaroslav ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 5

[alsa-devel] Names for ALSA controls
by Juergen Beisert 13 Jun '08

13 Jun '08

Hi, when I register these mixer controls: Line Playback Switch Line Playback Route Mono Playback Switch Mono Playback Route Phone Playback Switch Phone Playback Route Headphone Playback Switch Headphone Playback Route PCM Playback Switch PCM Playback Volume Line Bypass Capture Switch Line Bypass Capture Volume ADC Capture Route Running "alsamixer" fails with: ALSA lib simple_none.c:1218:(simple_add1) helem (MIXER,'Headphone Playback Switch',0,0,0) appears twice or more It seems there are restrictions what kind of control name I can use for each control, but what restrictions? Are there better names for my output amplifiers? Each of it I can enable/disable (-> Switch) and select a source (-> Route) independently. Regards, Juergen -- Dipl.-Ing. Juergen Beisert | http://www.pengutronix.de Pengutronix - Linux Solutions for Science and Industry Handelsregister: Amtsgericht Hildesheim, HRA 2686 Vertretung Sued/Muenchen, Germany Phone: +49-8766-939 228 | Fax: +49-5121-206917-9

2 2

[alsa-devel] [PATCH][ASoC v2] Add checks for codec DAI number during match-up
by Timur Tabi 13 Jun '08

13 Jun '08

Update the Codec DAI matching scheme to take the DAI number into account, and not just the name. This is necessary for systems with multiple codecs that use the Codec ID number to differentiate them. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/soc-core.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a88e3a0..cc543e6 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -801,6 +801,7 @@ codec_dai: /* yes, then check the codec_dai list */ list_for_each_entry(codec_dai, &codec_dai_list, list) { if (!strcmp(config->codec_dai, codec_dai->name) && + (codec_dai->id == config->codec_num) && try_module_get(codec_dai->dev->driver->owner)) { dbg("ASoC %s %s: Match for %s %s\n", soc_card->name, config->name, @@ -1767,6 +1768,7 @@ struct snd_soc_dai *snd_soc_register_codec_dai( dai->playback = template->playback; dai->capture = template->capture; dai->name = template->name; + dai->id = template->id; dai->ac97_control = template->ac97_control; mutex_lock(&client_mutex); list_add(&dai->list, &codec_dai_list); -- 1.5.5

2 1

[alsa-devel] [PATCH] maestro3: Fix hw volume on HP OmniBook
by Ville Syrjälä 13 Jun '08

13 Jun '08

Make the hw volume buttons work correctly on some HP OmniBook laptops. The original quirk was apparently applied a bit too early and it was also lacking some critial register writes. This improved sequence was discovered by trial and error (like the original sequence). Tested and found working on OB500 and OB6000 laptops. Signed-off-by: Ville Syrjala <syrjala(a)sci.fi> --- sound/pci/maestro3.c | 42 +++++++++++++++++++++++++++--------------- 1 file changed, 27 insertions(+), 15 deletions(-) Index: linux-2.6.25.4/sound/pci/maestro3.c =================================================================== --- linux-2.6.25.4.orig/sound/pci/maestro3.c +++ linux-2.6.25.4/sound/pci/maestro3.c @@ -2427,6 +2427,29 @@ snd_m3_amp_enable(struct snd_m3 *chip, i outw(0xffff, io + GPIO_MASK); } +static void +snd_m3_hv_init(struct snd_m3 *chip) +{ + unsigned long io = chip->iobase; + u16 val = GPI_VOL_DOWN | GPI_VOL_UP; + + if (!chip->is_omnibook) + return; + + /* + * Volume buttons on some HP OmniBook laptops + * require some GPIO magic to work correctly. + */ + outw(0xffff, io + GPIO_MASK); + outw(0x0000, io + GPIO_DATA); + + outw(~val, io + GPIO_MASK); + outw(inw(io + GPIO_DIRECTION) & ~val, io + GPIO_DIRECTION); + outw(val, io + GPIO_MASK); + + outw(0xffff, io + GPIO_MASK); +} + static int snd_m3_chip_init(struct snd_m3 *chip) { @@ -2442,21 +2465,6 @@ snd_m3_chip_init(struct snd_m3 *chip) DISABLE_LEGACY); pci_write_config_word(pcidev, PCI_LEGACY_AUDIO_CTRL, w); - if (chip->is_omnibook) { - /* - * Volume buttons on some HP OmniBook laptops don't work - * correctly. This makes them work for the most part. - * - * Volume up and down buttons on the laptop side work. - * Fn+cursor_up (volme up) works. - * Fn+cursor_down (volume down) doesn't work. - * Fn+F7 (mute) works acts as volume up. - */ - outw(~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_MASK); - outw(inw(io + GPIO_DIRECTION) & ~(GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DIRECTION); - outw((GPI_VOL_DOWN|GPI_VOL_UP), io + GPIO_DATA); - outw(0xffff, io + GPIO_MASK); - } pci_read_config_dword(pcidev, PCI_ALLEGRO_CONFIG, &n); n &= ~(HV_CTRL_ENABLE | REDUCED_DEBOUNCE | HV_BUTTON_FROM_GD); n |= chip->hv_config; @@ -2644,6 +2652,8 @@ static int m3_resume(struct pci_dev *pci snd_m3_enable_ints(chip); snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0; } @@ -2783,6 +2793,8 @@ snd_m3_create(struct snd_card *card, str snd_m3_amp_enable(chip, 1); + snd_m3_hv_init(chip); + tasklet_init(&chip->hwvol_tq, snd_m3_update_hw_volume, (unsigned long)chip); if (request_irq(pci->irq, snd_m3_interrupt, IRQF_SHARED, -- Ville Syrjälä syrjala(a)sci.fi http://www.sci.fi/~syrjala/

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[alsa-devel] alsa midi tutorial
by andy baxter 12 Jun '08

12 Jun '08

hello, Can anyone recommend a good, reasonably simple, tutorial on programming for the alsa midi system? What I want to do is feed drum trigger signals from an arduino board (a usb development board that lets you interface electronics to the computer) to a midi sequencer. I had a look at the following page from the alsa api reference: http://www.alsa-project.org/alsa-doc/alsa-lib/seq.html but I'm finding it a bit hard to follow - I'm looking for something a bit simpler with code examples to work through that will give me a way in to alsa midi programming. Does anyone know of anything like this? thanks, andy baxter.

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