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[alsa-devel] MAYA44 PCI ALSA driver for 300$
by Piotr Makowski 16 Sep '07

16 Sep '07

Hi all, I have just created the project "ESI MAYA44 PCI ALSA Driver" on getacoder.com. http://www.getacoder.com/projects/esi_maya_pci_alsa_driver_59542.html If anybody is interested in MAYA44 case, please bid or watch the post. -- Regards, Piotr Makowski

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[alsa-devel] Per option CFLAGS?
by Rene Herman 15 Sep '07

15 Sep '07

Hi Kai, Sam. I have a single file foo.c that I want to generate two (ALSA) modules from, snd-foo2000.ko and snd-foo2001.ko, by compiling with either FOO2000 or FOO2001 defined. I can do this, and ALSA does this a few times, by providing dummy foo2000.c and foo2001.c files, like: === foo2000.c #define FOO2000 #include "foo.c" === and a regular Makefile === foo2000-objs := foo2000.o foo2001-objs := foo2001.o obj-$(CONFIG_SND_FOO2000) += snd-foo2000.o obj-$(CONFIG_SND_F002001) += snd-foo2001.o === That #include is a little lame though. Is there a nicer way? I noticed the per-file CFLAGS, but given that it's one source file for both, that doesn't fit. Rene.

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[alsa-devel] Sony VAIO FZ19VN (snd_hda_intel)
by Pablo Yaggi 15 Sep '07

15 Sep '07

Hi, i bought a sony vaio fz19vn, i try both modes for the snd_hda_intel, vaio and vaio-ar, the same problem happen with them, when i plugin the headset internal speakers are not muted, so here is the card info. I read the changelog for version 1.0.15 and it says automute was added for the AD1986 which has jack detection, but this is a different codec. I have programming skills but i'm not familiarized with alsa internals, so ask me anything you need and i try to help. It would be nice a control to enable/disable automute and another two to enable/disable internal and external speakers. Best Regards Pablo *************************** /proc/asound/card0/codec#1: *************************** Codec: Conexant ID 2c06 Address: 1 Vendor Id: 0x14f12c06 Subsystem Id: 0x104d1700 Revision Id: 0x100000 Modem Function Group: 0x2 *************************** /proc/asound/card0/codec#0 : *************************** Codec: SigmaTel STAC9872AK Address: 0 Vendor Id: 0x83847662 Subsystem Id: 0x104d2300 Revision Id: 0x100201 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 Node 0x02 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Power: 0x0 Node 0x03 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x04 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0xff 0xff] Power: 0x0 Node 0x05 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out Amp-Out caps: N/A Amp-Out vals: [0x7f 0x7f] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x07 Node 0x07 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x00 0x00] Connection: 1 0x0e Node 0x08 [Audio Input] wcaps 0x1d0541: Stereo Power: 0x0 Connection: 1 0x09 Node 0x09 [Audio Selector] wcaps 0x300903: Stereo Amp-In Amp-In caps: N/A Amp-In vals: [0x0c 0x0c] Connection: 1 0x15 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173c: IN OUT HP Detect Pin Default 0x04214020: [Jack] HP Out at Ext Right Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x02 Node 0x0b [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0814: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x04 Node 0x0c [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0814: OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x03 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x08173c: IN OUT HP Detect Pin Default 0x04a15030: [Jack] Mic at Ext Right Conn = 1/8, Color = Red Pin-ctls: 0x24: IN Connection: 1 0x02 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0824: IN Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x20: IN Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0814: OUT Detect Pin Default 0x90170110: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x05 Node 0x10 [Audio Output] wcaps 0x40211: Stereo Digital PCM: rates [0x3e0]: 44100 48000 88200 96000 176400 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Node 0x11 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 2 0x10* 0x09 Node 0x12 [Audio Input] wcaps 0x140311: Stereo Digital PCM: rates [0x160]: 44100 48000 96000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Connection: 1 0x13 Node 0x13 [Pin Complex] wcaps 0x440381: Stereo Digital Pincap 0x0834: IN OUT Detect Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x90a7013e: [Fixed] Mic at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x04, stepsize=0x27, mute=1 Amp-Out vals: [0x00 0x00] Connection: 4 0x0a 0x0d* 0x14 0x02 Node 0x16 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=0 Amp-Out vals: [0x00] Node 0x17 [Volume Knob Widget] wcaps 0x600000: Mono Node 0x18 [Audio Output] wcaps 0x40201: Stereo Digital

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[alsa-devel] Ensoniq SoundScape VIVO90
by Krzysztof Helt 14 Sep '07

14 Sep '07

Hi, I acquired Ensoniq VIVO90 card. I tried sscape driver but it does not work. I quickly looked at the oss driver in the kernel and it works completely different (uses different pnp resources for codec and mpu). I want to ask if somebody tested this driver and if so what was the Ensoniq card model. If it worked only for non-pnp models I would integrate at least for VIVO cards. BTW. I found the old post which describes difference between VIVO and VIVO90. It is below the email. Regards, Krzysztof Hannu Savolainen (ha...(a)voxware.pp.fi, ha...(a)4front-tech.com) wrote: Temat: Re: Ensoniq VIVO vs. VIVO 90 Bill Plenderleith <nospam!plend...(a)nortel.ca> writes: >Are VIVO & VIVO 90 different products? >Can anyone summarize the diffences? The only difference is that in VIVO90 the wave table output signal goes through the codec/mixer chip (AD1845). In the original VIVO it's summed to the output of AD1845. This difference is'n really anything important. Best regards, Hannu

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[alsa-devel] SALSA: snd_pcm_avail_update
by J. Scott Merritt 14 Sep '07

14 Sep '07

Using salsa-lib-0.0.11, cross-compiling to ARM ... Despite my best efforts, snd_pcm_avail_update is always returning "broken pipe". I have checked the pcm_state of the Capture Handle just before snd_pcm_avail_update and it is "RUNNING'. But immediately after snd_pcm_avail_update it always returns "XRUN". Furthermore, this same sequence seems to work properly with the standard ALSA-Lib. Test program is attached. Any suggestions appreciated. Thanks, Scott.

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[alsa-devel] How do I update the ALSA SoundCard Matrix?
by Timur Tabi 14 Sep '07

14 Sep '07

I'm looking at http://bugtrack.alsa-project.org/main/index.php/Matrix:Main. I'm trying to add information on the CS4270 driver that I wrote, but I just don't get it. I know how to edit normal Wiki pages, but this looks different. It looks like all the information is obtained from some database somewhere. This page http://bugtrack.alsa-project.org/main/index.php/Help:Howto-Edit supposedly tells me how to edit, but it's useless. It doesn't really tell me *how* to edit, it just tells me that it can. So what am I missing? How do I add information on a new driver? I'll need to add information on a new vendor soon, too. -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] [PATCH] sc6000: documentation fix
by Krzysztof Helt 14 Sep '07

14 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes MPU-401 irq values list in documentation. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp alsa-driver-1.0.15rc2/sound/Documentation/ALSA-Configuration.txt alsa-driver-new/sound/Documentation/ALSA-Configuration.txt --- alsa-driver-1.0.15rc2/sound/Documentation/ALSA-Configuration.txt 2007-09-12 13:33:51.000000000 +0200 +++ alsa-driver-new/sound/Documentation/ALSA-Configuration.txt 2007-09-13 23:12:23.920824623 +0200 @@ -1679,7 +1679,7 @@ Prior to version 0.9.0rc4 options had a port - Port # (0x220 or 0x240) mss_port - MSS Port # (0x530 or 0xe80) irq - IRQ # (5,7,9,10,11) - mpu_irq - MPU-401 IRQ # (ditto) + mpu_irq - MPU-401 IRQ # (5,7,9,10) ,0 - no MPU-401 irq dma - DMA # (1,3,0) This module supports multiple cards.

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[alsa-devel] ALSA changes in 2.6.23-rc6 for Intel HDA
by Jon Smirl 13 Sep '07

13 Sep '07

The ALSA changes in 2.6.23-rc6 for Intel HDA made me change my volume from about 30% to 75% to get the same level of sound. Was this intentional? My ALSA info is here. http://pastebin.ca/692226 -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] [PATCH] New Aztech Sound Galaxy driver.
by Rene Herman 13 Sep '07

13 Sep '07

Good day. This is not being submitted yet... This is a new driver for Aztech Sound Galaxy soundcards. At the moment, it supports all AZT2316A/R based cards which make up the bulk of SG cards. It's been tested and found to be working with: 1. FCC-ID I38-MMSN824: Aztech Sound Galaxy Pro 16 AB 2. FCC-ID I38-MMSN826: Aztech Sound Galaxy Waverider 32+ 3. FCC-ID I38-MMSN830: Trust Sound Expert DeLuxe 16+ 4. FCC-ID I38-MMSN847: Trust Sound Expert DeLuxe Wave 32-3D 5. FCC-ID I38-MMSN852: Aztech Sound Galaxy Waverider Pro 32-3D where: 1. I38-MMSN824 = AZT2316A + CS4248-KL, without EEPROM 2. I38-MMSN826 = AZT2316A + AD1845XP, with EEPROM 3. I38-MMSN830 = AZT2316A + CS4248-KL, without EEPROM, and some cards: AZT2316A + AD1845-XP, without EEPROM 4. I38-MMSN847 = AZT2316R + CS4231A-KL, with EEPROM 5. I38-MMSN852 = AZT2316R + CS4231A-KL, with EEPROM All cards with a settings EEPROM have been tested with their JMPCFG jumper set to both EEPROM and SOFTWARE, and work correctly in both situations. All cards support full-duplex. The ones with AD1845XP codecs seem to be a little flakey, where starting a capture _sometimes_ hangs a concurrent playback, and sometimes not. Except the ones with a CS4248-KL codec, all cards experience a "cs4231_mce_down - auto calibration time out (1)" upon loading which does not seem to have any further consequences. Both this and the previous point may be cause for debugging cs4231_lib. Later... Ram -- included you in the CC because you had a: FCC-ID: I38-MMSN846: Aztech Sound Galaxy Nova 16 Extra II-3D, AZT2316R + CS4231A-KL, without EEPROM The attached patch is against 2.6.22.x and if you're setup to test, feel free and I'll add your card to the "was tested on" list. If you're not, don't bother too much though -- it'll work (and sorry for the long delay by the way...). As to the AZT2316A/R chipset series, this driver seems complete and correct. I also have a couple of Aztech Sound Galaxy cards with an AZT1605 chipset (and one even older Sound Galaxy) though and those don't quite/fully work yet as their EEPROMs are layed out slightly different -- they also store default volumes in the EEPROM for example. Got about half of it mapped out already but can't work on it tomorrow so thought I'd post now. Finally I have a bunch of AZT2320 based cards, which are PnP based, and which I'll merge in as well. Krzysztof -- you won't be surprised to learn that this driver doesn't do auto-probing. It needs all parameters passed in; normally: options snd-galaxy port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 If you don't specify dma2 (or set it equal to dma1) the card's setup in half-duplex mode. I'll document it all upon driver submission. Your (expected) Waverider should also work fine with it. If/when you test it, could you provide the information such as given in the tables above? You have an EEPROM if you have "X24C00" chip sitting on a spot marked "UX2" on the board, close to a crystal (silver shiny thing). If that spot is empty, you have an EEPROM-less variant. Mapping out the rest of AZT1605 (there may also be a duplex diference there) will be sunday or next week. Rene. diff --git a/sound/isa/Kconfig b/sound/isa/Kconfig index cf3803c..02a0f55 100644 --- a/sound/isa/Kconfig +++ b/sound/isa/Kconfig @@ -177,6 +177,19 @@ config SND_ES18XX To compile this driver as a module, choose M here: the module will be called snd-es18xx. +config SND_GALAXY + tristate "Aztech Sound Galaxy Driver" + depends on SND + select SND_CS4231_LIB + select SND_MPU401_UART + select SND_OPL3_LIB + select SND_PCM + help + Say Y here to include support for Aztech Sound Galaxy cards. + + To compile this driver as a module, choose M here: the module + will be called snd-galaxy. + config SND_GUS_SYNTH tristate diff --git a/sound/isa/Makefile b/sound/isa/Makefile index bb317cc..28bf545 100644 --- a/sound/isa/Makefile +++ b/sound/isa/Makefile @@ -12,6 +12,7 @@ snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o +snd-galaxy-objs := galaxy.o # Toplevel Module Dependency obj-$(CONFIG_SND_ADLIB) += snd-adlib.o @@ -23,6 +24,7 @@ obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o +obj-$(CONFIG_SND_GALAXY) += snd-galaxy.o obj-$(CONFIG_SND) += ad1816a/ ad1848/ cs423x/ es1688/ gus/ opti9xx/ \ sb/ wavefront/ diff --git a/sound/isa/galaxy.c b/sound/isa/galaxy.c new file mode 100644 index 0000000..8f8e9af --- /dev/null +++ b/sound/isa/galaxy.c @@ -0,0 +1,674 @@ +/* + * Aztech Sound Galaxy Driver + */ + +#include <sound/driver.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/isa.h> +#include <linux/delay.h> +#include <asm/io.h> +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/cs4231.h> +#include <sound/mpu401.h> +#include <sound/opl3.h> + +#define CRD_NAME "Aztech Sound Galaxy" +#define DRV_NAME "GALAXY" +#define DEV_NAME "galaxy" + +MODULE_DESCRIPTION(CRD_NAME); +MODULE_AUTHOR("Rene Herman"); +MODULE_LICENSE("GPL"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for " CRD_NAME " soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for " CRD_NAME " soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable " CRD_NAME " soundcard."); + +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long wss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static long fm_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; +static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; +static int dma2[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; + +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for " CRD_NAME " driver."); +module_param_array(wss_port, long, NULL, 0444); +MODULE_PARM_DESC(wss_port, "WSS port # for " CRD_NAME " driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for " CRD_NAME " driver."); +module_param_array(fm_port, long, NULL, 0444); +MODULE_PARM_DESC(fm_port, "OPL3 port # for " CRD_NAME " driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for " CRD_NAME " driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for " CRD_NAME " driver."); +module_param_array(dma1, int, NULL, 0444); +MODULE_PARM_DESC(dma1, "Playback DMA # for " CRD_NAME " driver."); +module_param_array(dma2, int, NULL, 0444); +MODULE_PARM_DESC(dma2, "Capture DMA # for " CRD_NAME " driver."); + +/* + * Sound Galaxy 32-bit serial EEPROM + */ + +#define GALAXY_PORT_CONFIG 1024 + +#define GALAXY_CONFIG_SBDMA_DISABLE (0 << 30) +#define GALAXY_CONFIG_SBDMA_0 (1 << 30) +#define GALAXY_CONFIG_SBDMA_1 (2 << 30) +#define GALAXY_CONFIG_SBDMA_3 (3 << 30) + +#define GALAXY_CONFIG_SBIRQ_10 (1 << 29) +#define GALAXY_CONFIG_SBIRQ_7 (1 << 28) +#define GALAXY_CONFIG_SBIRQ_5 (1 << 27) +#define GALAXY_CONFIG_SBIRQ_2 (1 << 26) + +#define GALAXY_CONFIG_SBA_220 (0 << 24) +#define GALAXY_CONFIG_SBA_240 (1 << 24) +#define GALAXY_CONFIG_SBA_260 (2 << 24) +#define GALAXY_CONFIG_SBA_280 (3 << 24) +#define GALAXY_CONFIG_SBA_MASK GALAXY_CONFIG_SBA_280 + +#define GALAXY_CONFIG_CDA_310 (0 << 22) +#define GALAXY_CONFIG_CDA_320 (1 << 22) +#define GALAXY_CONFIG_CDA_340 (2 << 22) +#define GALAXY_CONFIG_CDA_350 (3 << 22) +#define GALAXY_CONFIG_CDA_MASK GALAXY_CONFIG_CDA_350 + +#define GALAXY_CONFIG_MPU_ENABLE (1 << 21) +#define GALAXY_CONFIG_MPUA_300 (0 << 20) +#define GALAXY_CONFIG_MPUA_330 (1 << 20) + +#define GALAXY_CONFIG_GAME_ENABLE (1 << 19) + +#define GALAXY_CONFIG_WSS_ENABLE (1 << 18) +#define GALAXY_CONFIG_WSSA_530 (0 << 16) +#define GALAXY_CONFIG_WSSA_604 (1 << 16) +#define GALAXY_CONFIG_WSSA_E80 (2 << 16) +#define GALAXY_CONFIG_WSSA_F40 (3 << 16) + +#define GALAXY_CONFIG_CDDMA16_DISABLE (0 << 14) +#define GALAXY_CONFIG_CDDMA16_5 (1 << 14) +#define GALAXY_CONFIG_CDDMA16_6 (2 << 14) +#define GALAXY_CONFIG_CDDMA16_7 (3 << 14) +#define GALAXY_CONFIG_CDDMA16_MASK GALAXY_CONFIG_CDDMA16_7 + +#define GALAXY_CONFIG_CDDMA8_DISABLE (0 << 12) +#define GALAXY_CONFIG_CDDMA8_0 (1 << 12) +#define GALAXY_CONFIG_CDDMA8_1 (2 << 12) +#define GALAXY_CONFIG_CDDMA8_3 (3 << 12) +#define GALAXY_CONFIG_CDDMA8_MASK GALAXY_CONFIG_CDDMA8_3 + +#define GALAXY_CONFIG_CD_DISABLE (0 << 8) +#define GALAXY_CONFIG_CD_PANASONIC (1 << 8) +#define GALAXY_CONFIG_CD_SONY (2 << 8) +#define GALAXY_CONFIG_CD_MITSUMI (3 << 8) +#define GALAXY_CONFIG_CD_AZTECH (4 << 8) +#define GALAXY_CONFIG_CD_UNUSED_0 (5 << 8) +#define GALAXY_CONFIG_CD_UNUSED_1 (6 << 8) +#define GALAXY_CONFIG_CD_UNUSED_2 (7 << 8) +#define GALAXY_CONFIG_CD_MASK GALAXY_CONFIG_CD_UNUSED_2 + +#define GALAXY_CONFIG_CDIRQ_15 (1 << 7) +#define GALAXY_CONFIG_CDIRQ_12 (1 << 6) +#define GALAXY_CONFIG_CDIRQ_11 (1 << 5) +#define GALAXY_CONFIG_CDIRQ_5 (1 << 4) +#define GALAXY_CONFIG_CDIRQ_MASK (\ + GALAXY_CONFIG_CDIRQ_15 | GALAXY_CONFIG_CDIRQ_12 |\ + GALAXY_CONFIG_CDIRQ_11 | GALAXY_CONFIG_CDIRQ_5) + +#define GALAXY_CONFIG_MPUIRQ_10 (1 << 3) +#define GALAXY_CONFIG_MPUIRQ_7 (1 << 2) +#define GALAXY_CONFIG_MPUIRQ_5 (1 << 1) +#define GALAXY_CONFIG_MPUIRQ_2 (1 << 0) + +#define GALAXY_CONFIG_MASK (\ + GALAXY_CONFIG_SBA_MASK | GALAXY_CONFIG_CDA_MASK |\ + GALAXY_CONFIG_CDDMA16_MASK | GALAXY_CONFIG_CDDMA8_MASK |\ + GALAXY_CONFIG_CD_MASK | GALAXY_CONFIG_CDIRQ_MASK) + +/* + * WSS 8-bit configuration register + */ + +#define WSS_PORT_CONFIG 0 + +#define WSS_CONFIG_IRQ_7 (1 << 3) +#define WSS_CONFIG_IRQ_9 (2 << 3) +#define WSS_CONFIG_IRQ_10 (3 << 3) +#define WSS_CONFIG_IRQ_11 (4 << 3) + +#define WSS_CONFIG_DUPLEX (1 << 2) + +#define WSS_CONFIG_DMA_0 (1 << 0) +#define WSS_CONFIG_DMA_1 (2 << 0) +#define WSS_CONFIG_DMA_3 (3 << 0) + +#define WSS_PORT_SIGNATURE 3 +#define WSS_SIGNATURE 4 + +/* + * Generic SB DSP registers + */ + +#define DSP_PORT_RESET 0x6 +#define DSP_PORT_READ 0xa +#define DSP_PORT_COMMAND 0xc +#define DSP_PORT_STATUS 0xc +#define DSP_PORT_DATA_AVAIL 0xe + +#define DSP_SIGNATURE 0xaa + +/* + * Sound Galaxy specific DSP commands. + */ + +#define DSP_COMMAND_GALAXY 9 + +#define GALAXY_COMMAND_WSSMODE 0 +#define GALAXY_COMMAND_SBPMODE 1 +#define GALAXY_COMMAND_MPU 2 +#define GALAXY_COMMAND_MPUA 4 + +#define GALAXY_MODE_WSS GALAXY_COMMAND_WSSMODE +#define GALAXY_MODE_SBP GALAXY_COMMAND_SBPMODE + +struct snd_galaxy { + void __iomem *port; + void __iomem *config_port; + void __iomem *wss_port; + struct resource *res_port; + struct resource *res_config_port; + struct resource *res_wss_port; + u32 config; +}; + +static u32 config[SNDRV_CARDS]; +static u8 wss_config[SNDRV_CARDS]; + +static int __devinit snd_galaxy_match(struct device *dev, unsigned int n) +{ + if (!enable[n]) + return 0; + + switch (port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_ERR "%s: please specify port\n", dev->bus_id); + return 0; + case 0x220: + config[n] |= GALAXY_CONFIG_SBA_220; + break; + case 0x240: + config[n] |= GALAXY_CONFIG_SBA_240; + break; + default: + snd_printk(KERN_ERR "%s: invalid port %#lx\n", dev->bus_id, + port[n]); + return 0; + } + + switch (wss_port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_ERR "%s: please specify wss_port\n", + dev->bus_id); + return 0; + case 0x530: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_530; + break; + case 0x604: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_604; + break; + case 0xe80: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_E80; + break; + case 0xf40: + config[n] |= GALAXY_CONFIG_WSS_ENABLE | GALAXY_CONFIG_WSSA_F40; + break; + default: + snd_printk(KERN_ERR "%s: invalid WSS port %#lx\n", dev->bus_id, + wss_port[n]); + return 0; + } + + switch (irq[n]) { + case SNDRV_AUTO_IRQ: + snd_printk(KERN_ERR "%s: please specify irq\n", dev->bus_id); + return 0; + case 7: + wss_config[n] |= WSS_CONFIG_IRQ_7; + break; + case 9: + case 2: + wss_config[n] |= WSS_CONFIG_IRQ_9; + break; + case 10: + wss_config[n] |= WSS_CONFIG_IRQ_10; + break; + case 11: + wss_config[n] |= WSS_CONFIG_IRQ_11; + break; + default: + snd_printk(KERN_ERR "%s: invalid IRQ %d\n", dev->bus_id, + irq[n]); + return 0; + } + + switch (dma1[n]) { + case SNDRV_AUTO_DMA: + snd_printk(KERN_ERR "%s: please specify dma1\n", + dev->bus_id); + return 0; + case 0: + wss_config[n] |= WSS_CONFIG_DMA_0; + break; + case 1: + wss_config[n] |= WSS_CONFIG_DMA_1; + break; + case 3: + wss_config[n] |= WSS_CONFIG_DMA_3; + break; + default: + snd_printk(KERN_ERR "%s: invalid playback DMA %d\n", + dev->bus_id, dma1[n]); + return 0; + } + + if (dma2[n] == SNDRV_AUTO_DMA || dma2[n] == dma1[n]) { + dma2[n] = -1; + goto mpu; + } + + wss_config[n] |= WSS_CONFIG_DUPLEX; + switch (dma2[n]) { + case 1: + if (dma1[n] == 0) + case 0: + break; + default: + snd_printk(KERN_ERR "%s: invalid capture DMA %d\n", + dev->bus_id, dma2[n]); + return 0; + } + +mpu: + switch (mpu_port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_WARNING "%s: mpu_port not specified; not using " + "MPU-401\n", dev->bus_id); + mpu_port[n] = -1; + goto fm; + case 0x300: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_300; + break; + case 0x330: + config[n] |= GALAXY_CONFIG_MPU_ENABLE | GALAXY_CONFIG_MPUA_330; + break; + default: + snd_printk(KERN_ERR "%s: invalid mpu port %#lx\n", dev->bus_id, + mpu_port[n]); + return 0; + } + + if (mpu_irq[n] == irq[n]) { + snd_printk(KERN_ERR "%s: cannot share IRQ between WSS and " + "MPU-401\n", dev->bus_id); + return 0; + } + + switch (mpu_irq[n]) { + case SNDRV_AUTO_IRQ: + snd_printk(KERN_WARNING "%s: mpu_irq not specified; using " + "polling mode\n", dev->bus_id); + mpu_irq[n] = -1; + break; + case 2: + case 9: + config[n] |= GALAXY_CONFIG_MPUIRQ_2; + break; + case 5: + config[n] |= GALAXY_CONFIG_MPUIRQ_5; + break; + case 7: + config[n] |= GALAXY_CONFIG_MPUIRQ_7; + break; + case 10: + config[n] |= GALAXY_CONFIG_MPUIRQ_10; + break; + default: + snd_printk(KERN_ERR "%s: invalid mpu irq %d\n", dev->bus_id, + irq[n]); + return 0; + } + +fm: + switch (fm_port[n]) { + case SNDRV_AUTO_PORT: + snd_printk(KERN_WARNING "%s: fm_port not specified: not using " + "OPL3\n", dev->bus_id); + fm_port[n] = -1; + break; + case 0x388: + break; + default: + snd_printk(KERN_ERR "%s: illegal fm port %#lx\n", dev->bus_id, + fm_port[n]); + return 0; + } + + config[n] |= GALAXY_CONFIG_GAME_ENABLE; + return 1; +} + +static int __devinit __reset(struct snd_galaxy *galaxy) +{ + unsigned long timeout = jiffies + msecs_to_jiffies(100); + + iowrite8(1, galaxy->port + DSP_PORT_RESET); + udelay(10); + iowrite8(0, galaxy->port + DSP_PORT_RESET); + udelay(10); + + while (!(ioread8(galaxy->port + DSP_PORT_DATA_AVAIL) & 0x80)) { + if (time_after(jiffies, timeout)) + return -EIO; + cpu_relax(); + } + if (ioread8(galaxy->port + DSP_PORT_READ) != DSP_SIGNATURE) + return -ENODEV; + + return 0; +} + +static int __devinit __command(struct snd_galaxy *galaxy, unsigned char cmd) +{ + unsigned long timeout = jiffies + msecs_to_jiffies(100); + + while (ioread8(galaxy->port + DSP_PORT_STATUS) & 0x80) { + if (time_after(jiffies, timeout)) + return -EIO; + cpu_relax(); + } + iowrite8(cmd, galaxy->port + DSP_PORT_COMMAND); + + return 0; +} + +static int __devinit snd_galaxy_set(struct snd_galaxy *galaxy, + unsigned char cmd, int set) +{ + int err; + + err = __command(galaxy, DSP_COMMAND_GALAXY); + if (err < 0) + return err; + + err = __command(galaxy, cmd); + if (err < 0) + return err; + + err = __command(galaxy, set ? 255 : 0); + + return err; +} + +static u32 __devinit __get_config(struct snd_galaxy *galaxy) +{ + return ioread32be(galaxy->config_port); +} + +static void __devinit __set_config(struct snd_galaxy *galaxy, u32 config) +{ + unsigned char tmp = ioread8(galaxy->config_port + 4); + + iowrite8(tmp | 0x80, galaxy->config_port + 4); + udelay(10); + iowrite32be(config, galaxy->config_port); + udelay(10); + iowrite8(tmp & 0x7f, galaxy->config_port + 4); +} + +static int __devinit snd_galaxy_config(struct snd_galaxy *galaxy, u32 config) +{ + int err; + + err = snd_galaxy_set(galaxy, GALAXY_COMMAND_MPU, + config & GALAXY_CONFIG_MPU_ENABLE); + if (err < 0) + return err; + + err = snd_galaxy_set(galaxy, GALAXY_COMMAND_MPUA, + config & GALAXY_CONFIG_MPUA_330); + if (err < 0) + return err; + + galaxy->config = __get_config(galaxy); + __set_config(galaxy, (galaxy->config & GALAXY_CONFIG_MASK) | config); + + msleep(100); + return 0; +} + +static int __devinit snd_galaxy_setmode(struct snd_galaxy *galaxy, int mode) +{ + int err; + + err = __reset(galaxy); + if (err < 0) + return err; + + err = __command(galaxy, DSP_COMMAND_GALAXY); + if (err < 0) + return err; + + err = __command(galaxy, mode); + if (err < 0) + return err; + + msleep(100); + return 0; +} + +static unsigned char __devinit __wss_signature(struct snd_galaxy *galaxy) +{ + return ioread8(galaxy->wss_port + WSS_PORT_SIGNATURE) & 0x3f; +} + +static void __devinit __wss_config(struct snd_galaxy *galaxy, u8 wss_config) +{ + iowrite8(wss_config, galaxy->wss_port + WSS_PORT_CONFIG); +} + +static int __devinit snd_galaxy_wss_config(struct snd_galaxy *galaxy, + u8 wss_config) +{ + if (__wss_signature(galaxy) != WSS_SIGNATURE) + return -ENODEV; + + __wss_config(galaxy, wss_config); + + msleep(100); + return 0; +} + +static void snd_galaxy_free(struct snd_card *card) +{ + struct snd_galaxy *galaxy = card->private_data; + + if (galaxy->wss_port) { + __wss_config(galaxy, 0); + ioport_unmap(galaxy->wss_port); + release_and_free_resource(galaxy->res_wss_port); + } + if (galaxy->config_port) { + __set_config(galaxy, galaxy->config); + ioport_unmap(galaxy->config_port); + release_and_free_resource(galaxy->res_config_port); + } + if (galaxy->port) { + ioport_unmap(galaxy->port); + release_and_free_resource(galaxy->res_port); + } +} + +static int __devinit snd_galaxy_probe(struct device *dev, unsigned int n) +{ + struct snd_galaxy *galaxy; + struct snd_cs4231 *chip; + struct snd_card *card; + int err; + + card = snd_card_new(index[n], id[n], THIS_MODULE, sizeof *galaxy); + if (!card) + return -EINVAL; + + snd_card_set_dev(card, dev); + + card->private_free = snd_galaxy_free; + galaxy = card->private_data; + + galaxy->res_port = request_region(port[n], 16, DRV_NAME); + if (!galaxy->res_port) { + err = -EBUSY; + goto error; + } + galaxy->port = ioport_map(port[n], 16); + + err = __reset(galaxy); + if (err < 0) { + snd_printd("%s: DSP not found at %#lx\n", dev->bus_id, port[n]); + goto error; + } + + galaxy->res_config_port = request_region(port[n] + GALAXY_PORT_CONFIG, + 16, DRV_NAME); + if (!galaxy->res_config_port) { + err = -EBUSY; + goto error; + } + galaxy->config_port = ioport_map(port[n] + GALAXY_PORT_CONFIG, 16); + + err = snd_galaxy_config(galaxy, config[n]); + if (err < 0) + goto error; + + galaxy->res_wss_port = request_region(wss_port[n], 4, DRV_NAME); + if (!galaxy->res_wss_port) { + err = -EBUSY; + goto error; + } + galaxy->wss_port = ioport_map(wss_port[n], 4); + + err = snd_galaxy_wss_config(galaxy, wss_config[n]); + if (err < 0) { + snd_printd("%s: WSS not found at %#lx\n", dev->bus_id, + wss_port[n]); + goto error; + } + + err = snd_galaxy_setmode(galaxy, GALAXY_MODE_WSS); + if (err < 0) + goto error; + + strcpy(card->driver, DRV_NAME); + strcpy(card->shortname, DRV_NAME); + + err = snd_cs4231_create(card, wss_port[n] + 4, -1, irq[n], dma1[n], + dma2[n], CS4231_HW_DETECT, 0, &chip); + if (err < 0) + return err; + + sprintf(card->longname, "%s at %#lx, irq %d, dma %d/%d", + card->shortname, chip->port, chip->irq, chip->dma1, chip->dma2); + + err = snd_cs4231_pcm(chip, 0, NULL); + if (err < 0) + return err; + + err = snd_cs4231_mixer(chip); + if (err < 0) + return err; + + err = snd_cs4231_timer(chip, 0, NULL); + if (err < 0) + return err; + + if (mpu_port[n] >= 0) { + err = snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, + mpu_port[n], 0, mpu_irq[n], + IRQF_DISABLED, NULL); + if (err < 0) + goto error; + } + + if (fm_port[n] >= 0) { + struct snd_opl3 *opl3; + + err = snd_opl3_create(card, fm_port[n], fm_port[n] + 2, + OPL3_HW_AUTO, 0, &opl3); + if (err < 0) { + snd_printk(KERN_ERR "%s: no OPL device at %#lx\n", + dev->bus_id, fm_port[n]); + goto error; + } + err = snd_opl3_timer_new(opl3, 1, 2); + if (err < 0) + goto error; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto error; + } + + err = snd_card_register(card); + if (err < 0) + goto error; + + dev_set_drvdata(dev, card); + return 0; + +error: + snd_card_free(card); + return err; +} + +static int __devexit snd_galaxy_remove(struct device *dev, unsigned int n) +{ + snd_card_free(dev_get_drvdata(dev)); + dev_set_drvdata(dev, NULL); + return 0; +} + +static struct isa_driver snd_galaxy_driver = { + .match = snd_galaxy_match, + .probe = snd_galaxy_probe, + .remove = __devexit_p(snd_galaxy_remove), + + .driver = { + .name = DEV_NAME + } +}; + +static int __init alsa_card_galaxy_init(void) +{ + return isa_register_driver(&snd_galaxy_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_galaxy_exit(void) +{ + isa_unregister_driver(&snd_galaxy_driver); +} + +module_init(alsa_card_galaxy_init); +module_exit(alsa_card_galaxy_exit);

2 5

[alsa-devel] tiwai long downtime in this week
by Takashi Iwai 13 Sep '07

13 Sep '07

Hi, I'm attending a conference during this week, so I won't be able to read/post this ML so often. If you need some patches for urgent review or merge, please add my address to Cc. thanks, Takashi

2 4

[alsa-devel] ALSA 1.0.15rc2
by Jaroslav Kysela 13 Sep '07

13 Sep '07

Hi all, I decided to release 1.0.15rc2. The driver, library and python packages are available for download. The 1.0.15rc3 will follow soon (next week). Please, report any problems to this mailing list to fix them for rc3. Changelog: http://www.alsa-project.org/main/index.php/Changes_v1.0.15rc1_v1.0.15rc2 Thanks, Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

1 0

[alsa-devel] wiki sound card matrix
by James Courtier-Dutton 13 Sep '07

13 Sep '07

Hi, In order to update the sound card matrix, do I edit the wiki or the old file in the cvs? James

2 1

[alsa-devel] Bug#407690: marked as done (ld10k1 segfaults on 2.6.19 kernel)
by owner＠bugs.debian.org 13 Sep '07

13 Sep '07

Your message dated Tue, 11 Sep 2007 22:04:37 +0200 with message-id <20070911200437.GS17585(a)frodo.home.lxtec.de> and subject line Bug#407690: [Pkg-alsa-devel] Bug#407690: kernel 2.6.19.2 incompatibel with debian libasound2-dev has caused the attached Bug report to be marked as done. This means that you claim that the problem has been dealt with. If this is not the case it is now your responsibility to reopen the Bug report if necessary, and/or fix the problem forthwith. (NB: If you are a system administrator and have no idea what I am talking about this indicates a serious mail system misconfiguration somewhere. Please contact me immediately.) Debian bug tracking system administrator (administrator, Debian Bugs database)

2 3

[alsa-devel] [PATCH] support for Native Instrument's RigKontrol3
by Daniel Mack 12 Sep '07

12 Sep '07

Hi, this patch addes support for Native Instrument's upcoming RigKontrol3 sound interface. Greets, Daniel

1 0

[alsa-devel] Problems with 32 bit apps on 64 bit OS using dmix and snd-ioctl32
by James Pearson 12 Sep '07

12 Sep '07

I've already sent this to the alsa-user list, but may be this a better place to ask: I have a problem using 32 bit (i386) playback applications on a 64 bit (x86_64) CentOS4/RHEL4 based kernel (based on kernel 2.6.9). I've installed alsa-driver, alsa-lib and alsa-utils 1.0.13 (newer versions of alsa-driver don't build with this kernel) If I try to run a 32 bit copy of aplay on a 64 bit machine that doesn't support hardware mixing (e.g. HDA NVidia) - i.e. uses dmix by default, I get: aplay: pcm_write:1268: write error: Invalid argument and /var/log/messages reports: Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(3) cmd(40045402){00} arg(ffffae88) on /dev/snd/timer Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(5) cmd(c008551a){00} arg(080678a0) on /dev/snd/controlC0 Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(5) cmd(c008551b){00} arg(080678a0) on /dev/snd/controlC0 Sep 4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(3) cmd(000054a0){00} arg(00000320) on /dev/snd/timer I've done some searching to find a solution - and found out that in more recent kernels, snd-ioctl32 has been obsoleted - and if I do use a more modern kernel, then using a 32 bit version of aplay works OK on a newer 64 bit kernel. However, for other (not sound related) reasons, I need to run the RHEL4/CentOS4 2.6.9 kernel ... I had a look at the alsa-driver code, and acore/ioctl32/timer32_new.c contains: #if 0 /* ** FIXME ** * The following four entries are disabled because they conflict * with the TCOC* definitions. * Unfortunately, the current ioctl32 wrapper uses a single * hash table for all devices. Once when the wrapper is fixed * with the table based on devices, they'll be back again. */ MAP_COMPAT(SNDRV_TIMER_IOCTL_START), MAP_COMPAT(SNDRV_TIMER_IOCTL_STOP), MAP_COMPAT(SNDRV_TIMER_IOCTL_CONTINUE), MAP_COMPAT(SNDRV_TIMER_IOCTL_PAUSE), #endif If I change the '#if 0' to '#if 1', recompile and reload with this new snd-ioctl32 module, then the 32 bit aplay works without an error ... although /var/log/messages still reports: Sep 4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(3) cmd(40045402){00} arg(ffffadb8) on /dev/snd/timer Sep 4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(5) cmd(c008551a){00} arg(080678a0) on /dev/snd/controlC0 Sep 4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(5) cmd(c008551b){00} arg(080678a0) on /dev/snd/controlC0 ... but at least I get sound out! So, what is the danger of using these 'disabled' ioctls? I can't find any reference to 'TCOC* definitions' - what are these ??? Thanks James Pearson

2 14

[alsa-devel] Bug#226538: marked as done (alsaconf expects lspci to be in /usr/bin not /sbin)
by owner＠bugs.debian.org 11 Sep '07

11 Sep '07

Your message dated Tue, 11 Sep 2007 22:20:44 +0200 with message-id <20070911202044.GA21118(a)frodo.home.lxtec.de> and subject line alsaconf expects lspci to be in /usr/bin not /sbin has caused the attached Bug report to be marked as done. This means that you claim that the problem has been dealt with. If this is not the case it is now your responsibility to reopen the Bug report if necessary, and/or fix the problem forthwith. (NB: If you are a system administrator and have no idea what I am talking about this indicates a serious mail system misconfiguration somewhere. Please contact me immediately.) Debian bug tracking system administrator (administrator, Debian Bugs database)

1 0

[alsa-devel] Bug#428234: marked as done (linux-sound-base: not empty so not removed)
by owner＠bugs.debian.org 11 Sep '07

11 Sep '07

Your message dated Tue, 11 Sep 2007 22:17:28 +0200 with message-id <20070911201728.GT17585(a)frodo.home.lxtec.de> and subject line [Pkg-alsa-devel] Bug#428234: Bug#428234: linux-sound-base: not empty so not removed has caused the attached Bug report to be marked as done. This means that you claim that the problem has been dealt with. If this is not the case it is now your responsibility to reopen the Bug report if necessary, and/or fix the problem forthwith. (NB: If you are a system administrator and have no idea what I am talking about this indicates a serious mail system misconfiguration somewhere. Please contact me immediately.) Debian bug tracking system administrator (administrator, Debian Bugs database)

1 0

[alsa-devel] Bug#438886: marked as done (alsa-base: alsa modules loaded but /proc/asound/cards remains empty)
by owner＠bugs.debian.org 11 Sep '07

11 Sep '07

Your message dated Tue, 11 Sep 2007 21:24:52 +0200 with message-id <20070911192452.GR17585(a)frodo.home.lxtec.de> and subject line Bug#438886: [Pkg-alsa-devel] Bug#438886: alsa-base: alsa modules loaded but /proc/asound/cards remains empty has caused the attached Bug report to be marked as done. This means that you claim that the problem has been dealt with. If this is not the case it is now your responsibility to reopen the Bug report if necessary, and/or fix the problem forthwith. (NB: If you are a system administrator and have no idea what I am talking about this indicates a serious mail system misconfiguration somewhere. Please contact me immediately.) Debian bug tracking system administrator (administrator, Debian Bugs database)

1 0

[alsa-devel] [PATCH] cs4231-lib: improved waiting after mce_down
by Krzysztof Helt 11 Sep '07

11 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces long msleeps in waiting loops with schedule_timeout() calls. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The cs4231 is quite fast and calibration takes about 2ms at 48kHz (the quickest). Loops wait in 10ms steps which is usually too long. This is done after the ad1848-lib driver, which waits in this improved way. This makes the cs4231-lib and ad1848-lib more similar and should help merging of the two. Regards, Krzysztof diff -urp linux-2.6.22.old/sound/isa/cs423x/cs4231_lib.c linux-2.6.22/sound/isa/cs423x/cs4231_lib.c --- linux-2.6.22.old/sound/isa/cs423x/cs4231_lib.c 2007-09-09 20:33:52.000000000 +0200 +++ linux-2.6.22/sound/isa/cs423x/cs4231_lib.c 2007-09-09 20:31:27.000000000 +0200 @@ -314,6 +314,7 @@ void snd_cs4231_mce_down(struct snd_cs42 { unsigned long flags; int timeout; + long time; snd_cs4231_busy_wait(chip); #if 0 @@ -340,31 +341,34 @@ void snd_cs4231_mce_down(struct snd_cs42 snd_printd("(2) jiffies = %li\n", jiffies); - /* in 10 ms increments, check condition, up to 250 ms */ - timeout = 25; + time = HZ / 4; while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { - if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (2)\n"); + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (time <= 0) { + snd_printk(KERN_ERR "mce_down - " + "auto calibration time out (2)\n"); return; } - msleep(10); + time = schedule_timeout(time); + spin_lock_irqsave(&chip->reg_lock, flags); } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif - /* in 10 ms increments, check condition, up to 100 ms */ - timeout = 10; + + snd_printd("(3) jiffies = %li\n", jiffies); + + time = HZ / 10; while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { - if (--timeout < 0) { - snd_printk(KERN_ERR "mce_down - auto calibration time out (3)\n"); + spin_unlock_irqrestore(&chip->reg_lock, flags); + if (time <= 0) { + snd_printk(KERN_ERR "mce_down - " + "auto calibration time out (3)\n"); return; } - msleep(10); + time = schedule_timeout(time); + spin_lock_irqsave(&chip->reg_lock, flags); } -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); -#endif + snd_printd("(4) jiffies = %li\n", jiffies); + snd_printk("mce_down - exit = 0x%x\n", + cs4231_inb(chip, CS4231P(REGSEL))); } static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size)

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[alsa-devel] Lenovo 3000 N100 Laptop quirk
by Chris Pemberton 11 Sep '07

11 Sep '07

To enable sound on a Lenovo 3000 N100 laptop... In patch_realtek.c (near line # 10882): Change: SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo", ALC861VD_LENOVO), to: SND_PCI_QUIRK(0x17aa, 0x2066, "Lenovo 3000 N100", ALC861VD_3ST), Thanks, Chris

2 3

[alsa-devel] [PATCH] sun-cs4231: improved waiting after MCE down
by Krzysztof Helt 11 Sep '07

11 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch sync sparc driver with x86 isa cs4231 driver patches. It fixes wrong waiting for the auto calibration bit and makes further waiting use much finer granularity. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> ---

2 1

[alsa-devel] [PATCH] [ALSA] Fix CS4270 volume control and optimize I2C operations
by Timur Tabi 11 Sep '07

11 Sep '07

The volume control for the CS4270 ASoC driver was inverted - raising the volume level with alsamixer would decrease the actual volume. This patch also improves the performance of the I2C code (used to change register settings) by only performing an I2C write if the new value is different from the value that's in the register cache. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- This patch applies to the 'mm' branch of perex/alsa.git. I would like this patch to be part of ALSA 1.0.15. sound/soc/codecs/cs4270.c | 21 +++++++++++++-------- 1 files changed, 13 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 43d50a4..5d601ad 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -294,19 +294,24 @@ static unsigned int cs4270_read_reg_cache(struct snd_soc_codec *codec, static int cs4270_i2c_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { + u8 *cache = codec->reg_cache; + if ((reg < CS4270_FIRSTREG) || (reg > CS4270_LASTREG)) return -EIO; - if (i2c_smbus_write_byte_data(codec->control_data, reg, value) == 0) { + /* Only perform an I2C operation if the new value is different */ + if (cache[reg - CS4270_FIRSTREG] != value) { + struct i2c_client *client = codec->control_data; + if (i2c_smbus_write_byte_data(client, reg, value)) { + printk(KERN_ERR "cs4270: I2C write failed\n"); + return -EIO; + } + /* We've written to the hardware, so update the cache */ - u8 *cache = codec->reg_cache; cache[reg - CS4270_FIRSTREG] = value; - return 0; - } else { - printk(KERN_ERR "cs4270: I2C write of register %u failed\n", - reg); - return -EIO; } + + return 0; } /* @@ -521,7 +526,7 @@ static int cs4270_i2c_detach(struct i2c_client *client) /* A list of non-DAPM controls that the CS4270 supports */ static const struct snd_kcontrol_new cs4270_snd_controls[] = { SOC_DOUBLE_R("Master Playback Volume", - CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 0) + CS4270_VOLA, CS4270_VOLB, 0, 0xFF, 1) }; static struct i2c_driver cs4270_i2c_driver = { -- 1.5.2.4

2 1

[alsa-devel] [PATCH] Gallant SC-6000 driver (5th version)
by Krzysztof Helt 11 Sep '07

11 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is port of the Gallant SC-6000 driver from the OSS aedsp16 driver. This card was also sold as AudioExcel DSP 16 and Zoltrix AV302 (Audio Plus True 16). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This version is heavily influenced by Rene's new Sound Galaxy driver. It uses ioread8/iowrite8 functions and managed port mappings. It has fixed memory management on error in the probe function. /* * Driver for Audio Excel DSP 16 soundcard. * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1(a)wp.pl> * * I don't have documentation for this card, I based this driver on the * driver for OSS/Free included in the kernel source. * * 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 */ #include <linux/module.h> #include <linux/isa.h> #include <asm/dma.h> #include <sound/driver.h> #include <sound/core.h> #include <sound/ad1848.h> #include <sound/control.h> #define SNDRV_LEGACY_FIND_FREE_IRQ #define SNDRV_LEGACY_FIND_FREE_DMA #include <sound/initval.h> MODULE_AUTHOR("Krzysztof Helt"); MODULE_DESCRIPTION("Gallant SC-6000"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Gallant SC-6000}}"); 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; /* Enable this card */ static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */ static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x300, 0x310, 0x320, 0x330 */ static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); module_param_array(id, charp, NULL, 0444); MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); module_param_array(port, long, NULL, 0444); MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); module_param_array(mss_port, long, NULL, 0444); MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); module_param_array(mpu_port, long, NULL, 0444); MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); module_param_array(irq, int, NULL, 0444); MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); module_param_array(mpu_irq, int, NULL, 0444); MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); module_param_array(dma, int, NULL, 0444); MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); /* * Bit mapped flags for calling sc6000_init_board(), and saving the current * emulation mode. */ #define INIT_NONE (0) #define INIT_SBPRO (1<<0) #define INIT_MPU401 (1<<1) #define SC6000_AUXC_LEFT 18 #define SC6000_AUXC_RIGHT 19 /* * Commands of SC6000's DSP (SBPRO+special). * Some of them are COMMAND_xx, in the future they may change. */ #define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ #define COMMAND_52 0x52 /* */ #define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ #define COMMAND_5C 0x5c /* */ #define COMMAND_60 0x60 /* */ #define COMMAND_66 0x66 /* */ #define COMMAND_6C 0x6c /* */ #define COMMAND_6E 0x6e /* */ #define COMMAND_88 0x88 /* */ #define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ #define COMMAND_C5 0xc5 /* */ #define GET_DSP_VERSION 0xe1 /* Get DSP Version */ #define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ /* * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port * to have the actual I/O port. * Register permissions are: * (wo) == Write Only * (ro) == Read Only * (w-) == Write * (r-) == Read */ #define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ #define DSP_READ 0x0a /* offset of DSP READ (ro) */ #define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ #define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ #define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ #define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ #define PFX "sc-6000: " /* hardware dependent functions */ /* * sc6000_irq_to_softcfg - Decode irq number into cfg code. */ static inline unsigned char sc6000_irq_to_softcfg(int irq) { unsigned char val = 0; switch (irq) { case 5: val = 0x28; break; case 7: val = 0x8; break; case 9: val = 0x10; break; case 10: val = 0x18; break; case 11: val = 0x20; break; default: break; } return val; } /* * sc6000_dma_to_softcfg - Decode dma number into cfg code. */ static inline unsigned char sc6000_dma_to_softcfg(int dma) { unsigned char val = 0; switch (dma) { case 0: val = 1; break; case 1: val = 2; break; case 3: val = 3; break; default: break; } return val; } /* * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code. */ static inline unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) { unsigned char val = 0; switch (mpu_irq) { case 5: val = 4; break; case 7: val = 0x44; break; case 9: val = 0x84; break; case 10: val = 0xc4; break; default: break; } return val; } static int sc6000_wait_data(int port) { int loop = 1000; unsigned char ret = 0; do { ret = inb(port + DSP_DATAVAIL); if (ret & 0x80) return 0; cpu_relax(); } while (loop--); return -EAGAIN; } static int sc6000_read(int port) { int inbyte; snd_printdd(" Read DSP Byte (0x%x): ", port); if (sc6000_wait_data(port)) { snd_printdd("failure.\n"); return -EBUSY; } inbyte = inb(port + DSP_READ); snd_printdd("read [0x%x]/{%c}.\n", inbyte, inbyte); printk(KERN_DEBUG "read [0x%x]/{%c}.\n", inbyte, inbyte); return inbyte; } static int sc6000_write(int port, int cmd) { unsigned char ret; int loop = 500000; snd_printdd(" Write DSP Byte (0x%x) [0x%x]: ", port, cmd); do { ret = inb(port + DSP_STATUS); /* * DSP ready to receive data if bit 7 of ret == 0 */ if (!(ret & 0x80)) { outb(cmd, port + DSP_COMMAND); snd_printdd("success.\n"); return 1; } } while (loop--); snd_printdd("timeout.\n"); snd_printk(KERN_ERR PFX "DSP Command (0x%x) timeout.\n", cmd); return -EAGAIN; } static int __devinit sc6000_dsp_get_answer(int port, int command, char *data, int data_len) { int len = 0; snd_printdd("Get DSP answer:\n"); if (!sc6000_write(port, command)) { snd_printk(KERN_ERR PFX "CMD 0x%x: failed!\n", command); snd_printdd("failed.\n"); return -EFAULT; } do { int ret = sc6000_read(port); if (ret < 0) { /* * If no more data available, return to the caller, no error if len>0. * We have no other way to know when the string is finished. */ if (len) break; else { snd_printdd("failed.\n"); return -ENXIO; } } data[len++] = ret; } while (len < data_len); snd_printdd("success.\n"); return len; } static int __devinit sc6000_dsp_reset(int port) { snd_printdd("Reset DSP:\n"); outb(1, port + DSP_RESET); udelay(10); outb(0, port + DSP_RESET); udelay(20); if (sc6000_read(port) == 0xaa) { snd_printdd("success.\n"); return 0; } else snd_printdd("failure.\n"); return -EBUSY; } /* detection and initialization */ static int __devinit sc6000_cfg_write(int port, unsigned char softcfg) { if (sc6000_write(port, WRITE_MDIRQ_CFG) < 0) { snd_printk(KERN_ERR PFX "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); return -ENXIO; } if (sc6000_write(port, softcfg) < 0) { snd_printk(KERN_ERR PFX "sc6000_cfg_write: failed!\n"); return -ENXIO; } return 0; } static int __devinit sc6000_setup_board(int port, int config) { int loop = 10; do { if (sc6000_write(port, COMMAND_88) < 0) { snd_printk(KERN_ERR PFX "CMD 0x%x: failed!\n", COMMAND_88); return -ENXIO; } } while ((sc6000_wait_data(port) < 0) && loop--); if (sc6000_read(port) < 0) { snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", COMMAND_88); return -ENXIO; } if (sc6000_cfg_write(port, config) < 0) return -EFAULT; return 0; } static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev) { if (!enable[dev]) return 0; if (port[dev] == SNDRV_AUTO_PORT) { printk(KERN_ERR PFX "specify IO port\n"); return 0; } if (mss_port[dev] == SNDRV_AUTO_IRQ) { printk(KERN_ERR PFX "specify MSS port\n"); return 0; } return 1; } static int __devinit sc6000_init_mss(int port, int config, int mss_port, int mss_config) { snd_printdd("sc6000_init_mss:\n"); if (sc6000_write(port, DSP_INIT_MSS) < 0) { snd_printk(KERN_ERR PFX "sc6000_init_mss [0x%x]: failed!\n", DSP_INIT_MSS); snd_printdd("failure.\n"); return -ENXIO; } mdelay(10); if (sc6000_cfg_write(port, config) < 0) return -ENXIO; outb(mss_config, mss_port); snd_printdd("success.\n"); return 0; } static int __devinit sc6000_init_board(int port, int irq, int dma, int mss_port, int mpu_irq) { char answer[15]; int mss_config = sc6000_irq_to_softcfg(irq) | sc6000_dma_to_softcfg(dma); int config = mss_config | sc6000_mpu_irq_to_softcfg(mpu_irq); printk(KERN_DEBUG "mss_config: %x, config = %x\n",mss_config,config); if (sc6000_dsp_reset(port)) { snd_printk(KERN_ERR PFX "sc6000_dsp_reset: failed!\n"); return -EBUSY; } if (sc6000_dsp_get_answer(port, GET_DSP_COPYRIGHT, answer, 15) <= 0) { snd_printk(KERN_ERR PFX "sc6000_dsp_copyright: failed!\n"); return -ENXIO; } /* * My SC-6000 card return "SC-6000" in DSPCopyright, so * if we have something different, we have to be warned. */ if (strcmp("SC-6000", answer)) snd_printk(KERN_ERR PFX "Warning: non SC-6000 audio card!\n"); if (sc6000_dsp_get_answer(port, GET_DSP_VERSION, answer, 2) < 2) { snd_printk(KERN_ERR PFX "sc6000_dsp_version: failed!\n"); return -ENXIO; } /* * 0x0A == (IRQ 7, DMA 1, MIRQ 0) */ if (sc6000_cfg_write(port, 0x0a) < 0) { snd_printk(KERN_ERR PFX "sc6000_stdcfg: failed!\n"); return -EFAULT; } if (sc6000_setup_board(port, config) < 0) { snd_printk(KERN_ERR PFX "sc6000_setup_board: failed!\n"); return -ENODEV; } if (sc6000_init_mss(port, config, mss_port, mss_config) < 0) { snd_printk(KERN_ERR PFX "Can not initialize" "Microsoft Sound System mode.\n"); return -ENODEV; } return 0; } static int __devinit snd_sc6000_detect(int dev, int irq, int dma) { int err = -ENODEV; snd_printdd("Initializing BASE[0x%x] IRQ[%d] DMA[%d] MIRQ[%d]\n", port, irq, dma, mpu_irq); /* * We must request the port region because these are the I/O * ports to access card's control registers. */ if (!request_region(port[dev], 0x10, "sc-6000 (base)")) { snd_printk(KERN_ERR PFX "SC-6000 port I/O port region is already in use.\n"); return -EBUSY; } err = sc6000_init_board(port[dev],irq,dma,mss_port[dev],mpu_irq[dev]); release_region(port[dev], 0x10); return err; } static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) { static int possible_irqs[] = { 7, 9, 10, 11, -1 }; static int possible_dmas[] = { 1, 3, 0, -1 }; int err; int xirq = irq[dev]; int xdma = dma[dev]; struct snd_card *card; struct snd_ad1848 *chip; card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); if (card == NULL) return -ENOMEM; if (xirq == SNDRV_AUTO_IRQ) { xirq = snd_legacy_find_free_irq(possible_irqs); if (xirq < 0) { snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); err = -EBUSY; goto _err; } } if (xdma == SNDRV_AUTO_DMA) { xdma = snd_legacy_find_free_dma(possible_dmas); if (xdma < 0) { snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); err = -EBUSY; goto _err; } } printk(KERN_DEBUG "irq %d dma %d\n",xirq,xdma); err = snd_sc6000_detect(dev, xirq, xdma); if (err < 0) goto _err; err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, AD1848_HW_DETECT, &chip); if (err < 0) goto _err; card->private_data = chip; err = snd_ad1848_pcm(chip, 0, NULL); if (err < 0) { snd_printdd(PFX "error creating new ad1848 PCM device\n"); goto _err; } err = snd_ad1848_mixer(chip); if (err < 0) { snd_printdd(PFX "error creating new ad1848 mixer\n"); goto _err; } #ifdef NOT_WORKING err = snd_sc6000_mixer(chip); if (err < 0) { snd_printdd(PFX "the mixer rewrite failed\n"); goto _err; } #endif strcpy(card->driver, "Gallant SC-6000"); strcpy(card->shortname, "Gallant SC-6000"); sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d", mss_port[dev], xirq, xdma); snd_card_set_dev(card, devptr); err = snd_card_register(card); if (err < 0) goto _err; dev_set_drvdata(devptr, card); return 0; _err: snd_card_free(card); return err; } static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) { snd_card_free(dev_get_drvdata(devptr)); dev_set_drvdata(devptr, NULL); return 0; } static struct isa_driver snd_sc6000_driver = { .match = snd_sc6000_match, .probe = snd_sc6000_probe, .remove = __devexit_p(snd_sc6000_remove), /* FIXME: suspend/resume */ .driver = { .name = "sc6000", }, }; static int __init alsa_card_sc6000_init(void) { return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS); } static void __exit alsa_card_sc6000_exit(void) { isa_unregister_driver(&snd_sc6000_driver); } module_init(alsa_card_sc6000_init) module_exit(alsa_card_sc6000_exit)

2 2

[alsa-devel] [PATCH] Patch Dreamcast AICA driver to handle Maple bus support
by Adrian McMenamin 11 Sep '07

11 Sep '07

This patches the AICA sound driver for the Dreamcast to handle the well known flakiness of the Dreamcast's G2 bus. This is dependent on getting Maple bus support (see http://lkml.org/lkml/2007/9/9/70) into the kernel. Signed-off by: Adrian McMenamin <adrian(a)mcmen.demon.co.uk> diff --git a/sound/sh/aica.c b/sound/sh/aica.c index 7397865..a42c58f 100644 --- a/sound/sh/aica.c +++ b/sound/sh/aica.c @@ -35,6 +35,7 @@ #include <linux/timer.h> #include <linux/delay.h> #include <linux/workqueue.h> +#include <linux/maple.h> #include <sound/driver.h> #include <sound/core.h> #include <sound/control.h> @@ -43,7 +44,7 @@ #include <sound/info.h> #include <asm/io.h> #include <asm/dma.h> -#include <asm/dreamcast/sysasic.h> +#include <asm/mach/sysasic.h> #include "aica.h" MODULE_AUTHOR("Adrian McMenamin <adrian(a)mcmen.demon.co.uk>"); @@ -218,6 +219,12 @@ static int aica_dma_transfer(int channels, int buffer_size, period_offset = dreamcastcard->clicks; period_offset %= (AICA_PERIOD_NUMBER / channels); runtime = substream->runtime; + /* If maple dma is running, wait for it to finish */ + do{ cpu_relax();} + while (!maple_dma_done()); + /* Turn off the maple dma now */ + ctrl_outl(0, MAPLE_ENABLE); + local_irq_disable(); for (q = 0; q < channels; q++) { err = dma_xfer(AICA_DMA_CHANNEL, (unsigned long) (runtime->dma_area + @@ -232,6 +239,9 @@ static int aica_dma_transfer(int channels, int buffer_size, break; dma_wait_for_completion(AICA_DMA_CHANNEL); } + /* restore maple dma */ + local_irq_enable(); + maplebus_init_hardware(); return err; }

2 1

[alsa-devel] [PATCH] ad1848/cs4231: replace commented out debug code with snd-printd{, d}
by Rene Herman 11 Sep '07

11 Sep '07

Hi Takashi. While I'm at it another "while I'm there" -- replace commented out debug code with snd-printd{,d}. (I saw Krysztof was busy tweaking the sleeping granularity in these functions, so I'm leaving the code alone after this). Rene. diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index 161da11..ace2a9b 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -211,13 +211,14 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) /* end of cleanup sequence */ for (timeout = 12000; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) udelay(100); -#if 0 - printk("(1) timeout = %i\n", timeout); -#endif + + snd_printdd("(1) timeout = %d\n", timeout); + #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_down [0x%lx] - auto calibration time out (0)\n", AD1848P(chip, REGSEL)); #endif + chip->mce_bit &= ~AD1848_MCE; timeout = inb(AD1848P(chip, REGSEL)); outb(chip->mce_bit | (timeout & 0x1f), AD1848P(chip, REGSEL)); @@ -234,9 +235,9 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) * which at the slowest possible rate of 5.5125 kHz means 907 us. */ msleep(1); -#if 0 - printk("(2) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(2) jiffies = %lu\n", jiffies); + time = msecs_to_jiffies(250); while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -247,9 +248,9 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) time = schedule_timeout_interruptible(time); spin_lock_irqsave(&chip->reg_lock, flags); } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(3) jiffies = %lu\n", jiffies); + time = msecs_to_jiffies(100); while (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) { spin_unlock_irqrestore(&chip->reg_lock, flags); @@ -261,10 +262,9 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) spin_lock_irqsave(&chip->reg_lock, flags); } spin_unlock_irqrestore(&chip->reg_lock, flags); -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); -#endif + + snd_printdd("(4) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", inb(AD1848P(chip, REGSEL))); } static unsigned int snd_ad1848_get_count(unsigned char format, diff --git a/sound/isa/cs423x/cs4231_lib.c b/sound/isa/cs423x/cs4231_lib.c index 9e8e0f1..d3a7b59 100644 --- a/sound/isa/cs423x/cs4231_lib.c +++ b/sound/isa/cs423x/cs4231_lib.c @@ -326,9 +326,7 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) int timeout; snd_cs4231_busy_wait(chip); -#if 0 - printk("(1) timeout = %i\n", timeout); -#endif + #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_down [0x%lx] - auto calibration time out (0)\n", (long)CS4231P(REGSEL)); @@ -352,9 +350,9 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) * which at the slowest possible rate of 5.5125 kHz means 907 us. */ msleep(1); -#if 0 - printk("(2) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(1) jiffies = %lu\n", jiffies); + /* in 10 ms increments, check condition, up to 250 ms */ timeout = 25; while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { @@ -364,9 +362,9 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } msleep(10); } -#if 0 - printk("(3) jiffies = %li\n", jiffies); -#endif + + snd_printdd("(2) jiffies = %lu\n", jiffies); + /* in 10 ms increments, check condition, up to 100 ms */ timeout = 10; while (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) { @@ -376,10 +374,9 @@ void snd_cs4231_mce_down(struct snd_cs4231 *chip) } msleep(10); } -#if 0 - printk("(4) jiffies = %li\n", jiffies); - snd_printk("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); -#endif + + snd_printdd("(3) jiffies = %lu\n", jiffies); + snd_printd("mce_down - exit = 0x%x\n", cs4231_inb(chip, CS4231P(REGSEL))); } static unsigned int snd_cs4231_get_count(unsigned char format, unsigned int size)

2 2

[alsa-devel] [PATCH] ad1848: fix AD1848P macro
by Rene Herman 11 Sep '07

11 Sep '07

Hi. Consistent variable naming is a good thing, but let's be a little less sneaky about enforcing it... ;-/ Rene. diff --git a/include/sound/ad1848.h b/include/sound/ad1848.h index b2c3f00..ca0b6c3 100644 --- a/include/sound/ad1848.h +++ b/include/sound/ad1848.h @@ -27,7 +27,7 @@ /* IO ports */ -#define AD1848P( codec, x ) ( (chip) -> port + c_d_c_AD1848##x ) +#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) #define c_d_c_AD1848REGSEL 0 #define c_d_c_AD1848REG 1

2 2

[alsa-devel] [PATCH] hda: BIOS changing subsystem id
by Matthew Ranostay 11 Sep '07

11 Sep '07

Some laptop BIOS change the subsystem id for STAC9205 cards if the microphone isn't toggled on/off in the settings. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff -r f2c43802b1aa pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Aug 21 15:20:26 2007 +0200 +++ b/pci/hda/patch_sigmatel.c Thu Aug 30 15:24:59 2007 -0400 @@ -824,6 +824,8 @@ static struct snd_pci_quirk stac9205_cfg "Dell Precision", STAC_9205_M43xx), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01f9, "Dell Precision", STAC_9205_DELL_M43), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x021b, + "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fa, "Dell Precision", STAC_9205_DELL_M43), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x01fe,

2 1

[alsa-devel] Tascam driver on arm
by ST 10 Sep '07

10 Sep '07

Hi I just tried to build the tascam sound driver SND_USB_USX2Y on arm. But this is not allowed in the Kconfig of the module. Is there a reason for this? Thanks ST

1 0

[alsa-devel] Unable to bypass alsa plugins and write directly to hw:0, 0
by Pharaoh . 10 Sep '07

10 Sep '07

My sound card supports hardware upsampling i.e. I do not need to use the software up sampling done by plugins like dmix etc. For testing hw up sampling, I tried to play an 8K sample rate wav file as follows, but some how it is being upsampled, I think I am missing something. # aplay -Dhw:0,0 8k.wav -v Playing WAVE '8k.wav' : Signed 16 bit Little Endian, Rate 8000 Hz, Stereo Warning: rate is not accurate (requested = 8000Hz, got = 44100Hz) please, try the plug plugin Hardware PCM card 0 'OMAP850 EAC' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22052 period_size : 5513 period_time : 125011 tick_time : 10000 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5513 xfer_align : 5513 start_threshold : 22052 stop_threshold : 22052 silence_threshold: 0 silence_size : 0 boundary : 1445199872 I have attached my alsa.conf. -pharaoh.

2 1

[alsa-devel] Fwd: Re: RME HDSPe Info/Question
by ricman 10 Sep '07

10 Sep '07

Posting reply from kind responder providing additional info... --- Colin Stewart Campbell <colin.campbell(a)comhem.se> wrote: > From: Colin Stewart Campbell <colin.campbell(a)comhem.se> > Subject: Re: [alsa-devel] RME HDSPe Info/Question > Date: Mon, 10 Sep 2007 14:23:28 +0200 > To: ricman <the_ricman(a)yahoo.com> > > Well, except for that the new PCIe cards do all the floating-point > conversion in hardware and present 32-bit float instead of 32-bit > integer data on the bus. > If the alsa driver handles this, then ok. If not then you'll have to > > wait. Those running OSX on Intel machines with PCI slots already got > > hit by this format issue. > > Colin > ____________________________________________________________________________________ Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC

2 1

[alsa-devel] [PATCH] ad1848: replace HZ calculus with msecs_to_jiffies()
by Rene Herman 10 Sep '07

10 Sep '07

Hi Takashi. If I'm not mistaken, any (new) use of HZ these days is considered a bug so while I'm there... Signed-off-by: Rene Herman <rene.herman(a)gmail.com> Rene. diff --git a/sound/isa/ad1848/ad1848_lib.c b/sound/isa/ad1848/ad1848_lib.c index dcd4435..161da11 100644 --- a/sound/isa/ad1848/ad1848_lib.c +++ b/sound/isa/ad1848/ad1848_lib.c @@ -237,7 +237,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) #if 0 printk("(2) jiffies = %li\n", jiffies); #endif - time = HZ / 4; + time = msecs_to_jiffies(250); while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); if (time <= 0) { @@ -250,7 +250,7 @@ static void snd_ad1848_mce_down(struct snd_ad1848 *chip) #if 0 printk("(3) jiffies = %li\n", jiffies); #endif - time = HZ / 10; + time = msecs_to_jiffies(100); while (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) { spin_unlock_irqrestore(&chip->reg_lock, flags); if (time <= 0) {

1 0

[alsa-devel] [PATCH] ad1848 and cs4231 busy loop replacement
by Krzysztof Helt 10 Sep '07

10 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces busy loop with msleep and removes incorrect debug message. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This should fix Rene's problem with the timeout message he got in his new Aztech driver. The problem is that the driver wait for busy bit to go high but it may not happen if the auto-calibration is not set. The result is incorrect debug message (that timeout happened). It is also possible (in theory) to get this message in the ad1848-lib on a slow CPU. The patch just wait a period around the period needed for the auto-calibration. If there is no auto-calibration set, the driver just loses few miliseconds. This may happen only during driver start as both drivers set auto-calibration. If the auto-calibration is set we give CPU more time to do something else then move to next waiting loop (for the auto-calibration bit to get cleared). The msleep delay is calculated as rounded down time of waiting number of cycles (from ad1848k and cs4231a specs) with the highest frequency (48kHz). Rene, please test this patch on the cs4231 codec. diff -urp linux-2.6.22/sound/isa/ad1848/ad1848_lib.c linux-2.6.23/sound/isa/ad1848/ad1848_lib.c --- linux-2.6.22/sound/isa/ad1848/ad1848_lib.c 2007-09-08 23:53:42.958067597 +0200 +++ linux-2.6.23/sound/isa/ad1848/ad1848_lib.c 2007-09-08 23:57:31.255077502 +0200 @@ -229,17 +229,12 @@ static void snd_ad1848_mce_down(struct s spin_unlock_irqrestore(&chip->reg_lock, flags); return; } + /* calibration process */ + msleep(7); + + snd_printd("(2) jiffies = %li\n", jiffies); - for (timeout = 500; timeout > 0 && (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0; timeout--); - if ((snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) == 0) { - snd_printd("mce_down - auto calibration time out (1)\n"); - spin_unlock_irqrestore(&chip->reg_lock, flags); - return; - } -#if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); -#endif time = HZ / 4; while (snd_ad1848_in(chip, AD1848_TEST_INIT) & AD1848_CALIB_IN_PROGRESS) { spin_unlock_irqrestore(&chip->reg_lock, flags); diff -urp linux-2.6.22/sound/isa/cs423x/cs4231_lib.c linux-2.6.23/sound/isa/cs423x/cs4231_lib.c --- linux-2.6.22/sound/isa/cs423x/cs4231_lib.c 2007-09-08 23:53:29.005272473 +0200 +++ linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-09-08 23:56:52.824887490 +0200 @@ -334,19 +334,12 @@ void snd_cs4231_mce_down(struct snd_cs42 !(chip->hardware & (CS4231_HW_CS4231_MASK | CS4231_HW_CS4232_MASK))) { return; } - snd_cs4231_busy_wait(chip); /* calibration process */ + msleep(2); + + snd_printd("(2) jiffies = %li\n", jiffies); - for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--) - udelay(10); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { - snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); - return; - } -#if 0 - printk("(2) timeout = %i, jiffies = %li\n", timeout, jiffies); -#endif /* in 10 ms increments, check condition, up to 250 ms */ timeout = 25; while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) {

2 9

[alsa-devel] alsaseq 0.1 Python module
by Patricio Paez 10 Sep '07

10 Sep '07

Hello all, This was posted in the Python announce list, but it could be of interest here: "The first version of alsaseq, bindings to the ALSA sequencer has been released. alsaseq is a Python module that allows to interact with ALSA sequencer clients. It can create an ALSA client, connect to other clients, send and receive ALSA events immediately or at a scheduled time using a sequencer queue. It provides a subset of the ALSA sequencer capabilities in a simplified model. It is implemented in C language and licensed under the Gnu GPL license version 2 or later. Home http://pp.com.mx/python/alsaseq Download http://pp.com.mx/python/alsaseq/alsaseq-0.1.tar.gz " It is an early and ongoing project, suggestions and comments are welcome. Regards, Patricio Páez <pp at pp.com.mx> Original post: http://mail.python.org/pipermail/python-announce-list/2007-September/006102…

1 0

[alsa-devel] Pausing an audio stream during playback using mixer control interface
by Pharaoh . 10 Sep '07

10 Sep '07

Hi I want to provide a mixer control interface which will pause the playback and resume it as needed. I tried to disable DMA while pausing and enable it while releasing, but this doesn't stop alsa upper layers from inquiring the buffer position and since DMA is not on, it gives a pcm write error, which is quite obvious. I know that, SNDRV_PCM_TRIGGER_PAUSE_PUSH can be used in trigger callback to pause the stream, but I want to give a mixer interface to do the same. As per my understanding this is not possible keeping in mind the current framework, can anyone think of any work around? -pharaoh.

2 1

[alsa-devel] RME HDSPe Info/Question
by ricman 10 Sep '07

10 Sep '07

Howdy! I would like to share a piece of information which I am hoping will be helpful to the ALSA development community. This information is regarding the RME HDSP Multiface/Digiface family of products. We know from http://www.alsa-project.org/main/index.php/Matrix:Vendor-RME and http://www.alsa-project.org/main/index.php/Matrix:Module-hdsp that we have a PCI driver for Linux thanks to Paul Davis, Marcus Andersson, Thomas Charbonnel (and probably others). However, RME recently came out with a new PCI Express (PCI-E) interface for these products. Here is where is gets interesting: I've been trying to ascertain whether I should purchase the PCI-E HDSPe card for my Linux box. I prayed to the RME gods on their forum and none other than Matthias Carstens of RME responded that "The PCIe cards are fully downward compatible, so can be used even with the oldest Windows drivers. So I expect Linux to work with them too." (http://www.rme-audio.de/forum/viewtopic.php?id=1031) Is it possible that he is right? Will the old PCI Linux driver work just the same (or better) with the newer PCI-E card? For some reason I just naturally thought the PCI-E card would need its own special driver. Or is the only way to know to just buy and see if it works? That seem like an expensive way to find out... Thanks! ____________________________________________________________________________________ Choose the right car based on your needs. Check out Yahoo! Autos new Car Finder tool. http://autos.yahoo.com/carfinder/

2 1

[alsa-devel] period_size and relation to number of samples
by Markus Korber 10 Sep '07

10 Sep '07

Hello, one question: If my driver announces the setting below[1], how many samples must an application provide in a single snd_pcm_writei() or snd_pcm_mmap_writei() call? Shouldn't it always send 8192 bytes or 2048 left and right samples (4096 in total) or is it free to send more or less? Logging data in my chip_playback_copy() function shows the following (S16_LE, RW_INTERLEAVED): ,----[ mplayer, aplay: ] | chip_playback_copy (0 -> src: 0xffce1000, #bytes: 0x4000) | chip_playback_copy (1 -> src: 0xffce1000, #bytes: 0x2000) | chip_playback_copy (1 -> src: 0xffce1000, #bytes: 0x2000) `---- ,----[ alsaplayer: ] | chip_playback_copy (0 -> src: 0xffce1000, #bytes: 0x2000) | chip_playback_copy (0 -> src: 0xffce3000, #bytes: 0x2000) | chip_playback_copy (0 -> src: 0xffce3000, #bytes: 0x2000) | [...] `---- Thus mplayer and aplay provide 16384 bytes (4096 l/r samples) in the first call and 8192 bytes (2048 l/r samples) for all subsequent calls. However, alsaplayer provides 8192 bytes (2048 l/r samples) for all calls. Now, what is an application allowed to send and what not? For example, could an application only send 1024 l/r samples and is the driver responsible for buffering the data? Or must it obey the announced period_size and *always* provide 2048 l/r samples? Thanks. ,----[ 1 ] | stream : PLAYBACK | access : RW_INTERLEAVED | format : S16_LE | subformat : STD | channels : 2 | rate : 44100 | exact rate : 44100 (44100/1) | msbits : 16 | buffer_size : 4096 | period_size : 2048 | period_time : 46439 | tick_time : 10000 | tstamp_mode : NONE | period_step : 1 | sleep_min : 0 | avail_min : 2048 | xfer_align : 1 | start_threshold : 2048 | stop_threshold : 4096 | silence_threshold: 0 | silence_size : 0 | boundary : 1073741824 | start_mode: DATA | xrun_mode: STOP | tstamp_mode: NONE | period_step: 1 | sleep_min: 0 | avail_min: 2048 | xfer_align: 1 | silence_threshold: 0 | silence_size: 0 | boundary: 1073741824 `---- Regards, Markus Korber

2 4

Re: [alsa-devel] [Alsa-user] Disable MIDI port creation with ICE1724
by Rene Herman 10 Sep '07

10 Sep '07

On 09/08/2007 03:32 AM, James Roberts wrote: > I need to disable the creation of the ICEnsemble ICE1724 midi device on > my system. I am using a off-brand card (see below) and accessing the > midi device crashes the system hard. > > I have explored udev rules, kernel opts, etc ad nauseum to do this. > > To be clear I do not want to use the midi capabilities of the sound card > at all, but want to keep the sound capabilities. > > lspci: > 00:09.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 > [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) > > amidi -l: > Dir Device Name > IO hw:0,0 ICEnsemble ICE1724 MIDI <- this must be gone forever! > IO hw:1,0,0 BCF2000 MIDI 1 > IO hw:1,0,1 BCF2000 MIDI 2 > O hw:1,0,2 BCF2000 MIDI 3 > > > It appears to get assigned to midiC0D0 > > Am I better off going the udev route and trying to somehow assign a null > value somewhere or is there a way to prevent alsa from creating it? > > Maybe an alias? Sounds like this may warrent a module parameter. Trivial, untested version below, but this needs a module maintainer to decide if there may be better ways to discriminate. Crossposted to alsa-devel... (against 2.6.22.x). Rene. diff --git a/sound/pci/ice1712/ice1724.c b/sound/pci/ice1712/ice1724.c index ee620de..1d64dcc 100644 --- a/sound/pci/ice1712/ice1724.c +++ b/sound/pci/ice1712/ice1724.c @@ -75,6 +75,7 @@ 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 */ static char *model[SNDRV_CARDS]; +static int no_mpu401[SNDRV_CARDS]; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for ICE1724 soundcard."); @@ -84,6 +85,8 @@ module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable ICE1724 soundcard."); module_param_array(model, charp, NULL, 0444); MODULE_PARM_DESC(model, "Use the given board model."); +module_param_array(no_mpu401, int, NULL, 0444); +MODULE_PARM_DESC(no_mpu401, "Mmf mmmff mmfmf mff mmfmmff."); /* Both VT1720 and VT1724 have the same PCI IDs */ @@ -2398,7 +2401,7 @@ static int __devinit snd_vt1724_probe(struct pci_dev *pci, } } - if (! c->no_mpu401) { + if (! c->no_mpu401 && ! no_mpu401[dev]) { if (ice->eeprom.data[ICE_EEP2_SYSCONF] & VT1724_CFG_MPU401) { if ((err = snd_mpu401_uart_new(card, 0, MPU401_HW_ICE1712, ICEREG1724(ice, MPU_CTRL),

2 1

[alsa-devel] alsaplayer configure says Build alsa 0.9.x output plugin: no
by Timur Tabi 10 Sep '07

10 Sep '07

I have a working ALSA driver on my system. I downloaded the latest alsaplayer and when I ran ./configure, it says this: Build alsa 0.9.x output plugin: no I looked at the source code for configure, and I found this: if (eval "$ac_cpp conftest.$ac_ext") 2>&5 | $EGREP "AP_maGiC_VALUE" >/dev/null 2>&1; then have_alsa_final=yes But I don't understand it. What do I need to have to get alsaplayer to build ALSA output support? -- Timur Tabi Linux Kernel Developer @ Freescale

3 3

[alsa-devel] Multiple Stream Playback on HDA-Intel
by Azhar Aga 09 Sep '07

09 Sep '07

Hi, I am trying ao develop an application that would play multiple streams, using the same soundcard on my PC and since this is my first stint in developing application on a pc rather than a native board I am facing problems, for doing this I am using libao to control ALSA. For me to demostrate the multiple stream playback, I need ALSA to direct output of one of the streams on the device hw:0,0 and the other on hw:0,1 [My card has probably two devices as I checked aplay -l option that gave me a list of devices] so that i could connect a set of speakers in one of the PC jacks and the another set in another Jack. Can this be done in using libao. If not then how one direct the sound output to other devices on the same card. Does the configuation file need to be changed? I made a .asoundrc config file in the the root directory but that did not help when I tried to access a hw:0,0 with the command aplay -D hw:0,0 < some_test_file.wav> it gives me the device busy error. ( with or without the .asoundrc aplay -d hw:0,0 gives the above mentioned error) The Card I am using is HDA_intel ICH family that comes with the DELL. -- Thanks and With Regards Azhar Aga

1 0

[alsa-devel] [usb-audio] M-Audio FastTrack Pro giving invalid sample rates
by Lasse Kärkkäinen 09 Sep '07

09 Sep '07

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 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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 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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 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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 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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

1 0

[alsa-devel] random static, distortion on M-audio Revolution 5.1
by Chris Falk 09 Sep '07

09 Sep '07

Hi guys, I've got a strange problem... Although my M-audio Revolution 5.1 sounds great, it will randomly "lock up". This seems to occur randomly regardless of what programs I am running. It has happened while browsing, while listening to mp3s with audacious, or while watching tv with xine. If I am running a program that has audio, it will start playing slower, and the frequencies seem warped to some low freq. There is also static, and some popping noises. If I am not running an audio program, I will hear the static and popping noises, and when I try to play an mp3, it will seem to be stuck at time 00:00. Sometimes the problem will go away if I stop my audio programs upon hearing the static, or if I unload the alsa modules and wait a few minutes before loading them again. The first thing that came to my mind was EM interference. I tried swapping the positions of my pci cards (I have a pchdtv 5500 tuner in addition to the sound card), but the problem did not go away. I thought perhaps there was some defect in the card, but I have not experienced any problems with it when I boot to Windows. Here's some info on my system... 2.6.22-gentoo-r5 Advanced Linux Sound Architecture Driver Version 1.0.14 02:0a.0 Multimedia audio controller: VIA Technologies Inc. VT1720/24 [Envy24PT/HT] PCI Multi-Channel Audio Controller (rev 01) Subsystem: VIA Technologies Inc. M-Audio Revolution 5.1 Flags: bus master, medium devsel, latency 32, IRQ 18 I/O ports at a000 [size=32] I/O ports at a400 [size=128] Capabilities: [80] Power Management version 1 If I am experiencing the problem, and I immediately try to restart alsa using /etc/init.d/alsasound restart, then I get these error messages in dmesg... ice1724: i2c busy timeout ice1724: i2c busy timeout ice1724: i2c busy timeout ice1724: i2c busy timeout ice1724: No matching model found for ID 0x12121212 ice1724: i2c busy timeout ice1724: i2c busy timeout ice1724: Invalid EEPROM version 18 ice1724: i2c busy timeout ice1724: i2c busy timeout ice1724: i2c busy timeout nvidia: module license 'NVIDIA' taints kernel. ice1724: i2c busy timeout ice1724: i2c busy timeout ALSA sound/pci/ice1712/ice1724.c:133: snd_vt1724_ac97_ready: timeout ALSA sound/pci/ice1712/ice1724.c:143: snd_vt1724_ac97_wait_bit: timeout ALSA sound/pci/ice1712/ice1724.c:133: snd_vt1724_ac97_ready: timeout ALSA sound/pci/ice1712/ice1724.c:143: snd_vt1724_ac97_wait_bit: timeout ALSA sound/pci/ice1712/ice1724.c:133: snd_vt1724_ac97_ready: timeout ALSA sound/pci/ice1712/ice1724.c:143: snd_vt1724_ac97_wait_bit: timeout ALSA sound/pci/ice1712/ice1724.c:133: snd_vt1724_ac97_ready: timeout ALSA sound/pci/ac97/ac97_codec.c:2046: AC'97 0 does not respond - RESET ALSA sound/pci/ice1712/ice1724.c:133: snd_vt1724_ac97_ready: timeout ALSA sound/pci/ice1712/ice1724.c:143: snd_vt1724_ac97_wait_bit: timeout ALSA sound/pci/ice1712/ice1724.c:133: snd_vt1724_ac97_ready: timeout ALSA sound/pci/ice1712/ice1724.c:143: snd_vt1724_ac97_wait_bit: timeout ALSA sound/pci/ac97/ac97_codec.c:2055: AC'97 0 access is not valid [0xffffffff], removing mixer. ice1712: cannot initialize pro ac97, skipped This problem is driving me crazy! I visited #alsa and googled all over looking for any clues, but could not find anything. I'm extremely grateful for all of the work you guys do, and for any help you could provide me. Thanks!! Chris

1 0

[alsa-devel] ALSA C++ API
by Lasse Kärkkäinen 08 Sep '07

08 Sep '07

Hi, I have written an experimental wrapper for a few parts of the PCM API, using C++. I intend to keep it header-only, if possible. This minimizes the binary compatibility issues and also improves performance slightly, but unlikely noticeably (should be identical to using the C API directly). Primary goals of this wrapper are safety (errors are handled by exceptions and resource leaks should not be possible) and the simplicity of use, while still keeping it very low-level, close to the C API. Does anyone have interest to develop this further? It would also be nice if it could eventually be included in the alsa-lib distribution. The header and a sample program are attached below. Comments are welcome. #ifndef ALSA_HPP_INCLUDED #define ALSA_HPP_INCLUDED /** * @file Experimental low level C++ API to ALSA. * @licence GNU LGPL 2.1 or later **/ #include <alsa/asoundlib.h> #include <stdexcept> namespace alsa { namespace util { /** * @short FOR INTERNAL USE ONLY. A utility class similar to * boost::noncopyable, duplicated here in order to avoid * a dependency on the Boost library. **/ class noncopyable { protected: noncopyable() {} ~noncopyable() {} private: noncopyable(noncopyable const&); noncopyable const& operator=(noncopyable const&); }; } /** * @short A minimal RAII wrapper for ALSA PCM. * Automatically converts into snd_pcm_t* as needed, so the ALSA C API * can be used directly with this. **/ class PCM: util::noncopyable { snd_pcm_t* handle; public: PCM(char const* device = "default", snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK, int mode = 0) { if (snd_pcm_open(&handle, device, stream, mode) < 0) throw std::runtime_error(std::string("Cannot open ALSA device ") + device); } ~PCM() { snd_pcm_close(handle); } operator snd_pcm_t*() { return handle; } }; /** * @short A RAII wrapper for ALSA HWParams structure. * Automatically converts into snd_pcm_hw_params_t* as needed, so the * ALSA C API can still be used directly with this. * Note: contents need to be loaded before the struct can be used. **/ class HWParams { snd_pcm_hw_params_t* handle; void init() { if (snd_pcm_hw_params_malloc(&handle) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_malloc failed"); } public: HWParams() { init(); } ~HWParams() { snd_pcm_hw_params_free(handle); } HWParams(HWParams const& orig) { init(); *this = orig; } HWParams& operator=(HWParams const& orig) { if (this != &orig) { snd_pcm_hw_params_copy(handle, orig.handle); } return *this; } operator snd_pcm_hw_params_t*() { return handle; } }; /** * @short A helper object for modifying HWParams of a PCM. * * Normally only temporary objects should be created. A typical use case: * alsa::PCM alsa_pcm; // Create a PCM object * alsa::HWConfig(alsa_pcm) // Create a new config space * .set(SND_PCM_ACCESS_RW_INTERLEAVED) * .set(SND_PCM_FORMAT_S16_LE) * .rate_near(rate) * .channels_near(channels) * .period_time_near(period) * .commit(); // Apply the config to the PCM * Note: in case of failure, an exception is thrown. When this happens, * the commit part never gets executed and thus this entire statement * will become void as if it was never executed in the first place. * * If you need to use the C snd_pcm_hw_params_* functions, you may use a * block like this: * { * alsa::HWConfig hw(alsa_pcm); * hw.foo(); * snd_pcm_hw_params_set_period_size(alsa_pcm, hw, 1024, 0); * hw.bar().commit(); * } **/ class HWConfig: util::noncopyable { HWParams params; PCM& pcm; public: HWConfig(PCM& pcm): pcm(pcm) { any(); } operator HWParams&() { return params; } void commit() { if (snd_pcm_hw_params(pcm, params) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params failed"); } HWConfig& any() { if (snd_pcm_hw_params_any(pcm, params) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_any failed"); return *this; } HWConfig& current() { if (snd_pcm_hw_params_current(pcm, params) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_current failed"); return *this; } HWConfig& set(snd_pcm_access_t access) { if (snd_pcm_hw_params_set_access(pcm, params, access) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_access failed"); return *this; } HWConfig& set(snd_pcm_format_t format) { if (snd_pcm_hw_params_set_format(pcm, params, format) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_format failed"); return *this; } HWConfig& rate(unsigned int rate) { if (snd_pcm_hw_params_set_rate(pcm, params, rate, NULL) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_rate failed"); return *this; } HWConfig& rate_near(unsigned int& rate) { if (snd_pcm_hw_params_set_rate_near(pcm, params, &rate, NULL) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_rate_near failed"); return *this; } HWConfig& channels(unsigned int num) { if (snd_pcm_hw_params_set_channels(pcm, params, num) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_channels failed"); return *this; } HWConfig& channels_near(unsigned int& num) { if (snd_pcm_hw_params_set_channels_near(pcm, params, &num) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_channels_near failed"); return *this; } HWConfig& period_time(unsigned int samples) { if (snd_pcm_hw_params_set_period_time(pcm, params, samples, NULL) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_period_time failed"); return *this; } HWConfig& period_time_near(unsigned int& us) { if (snd_pcm_hw_params_set_period_time_near(pcm, params, &us, NULL) < 0) throw std::runtime_error("ALSA snd_pcm_hw_params_set_period_time_near failed"); return *this; } }; } #endif // This program tries to record from your sound card, using a chosen sampling // rate. ALSA may give another rate, if the sound card does not support the // requested rate. Two seconds of audio is recorded and the time spent is // measured in order to find the actual rate of the sound card. // Compile with g++ alsarate.cpp -o alsarate -I. -lasound // ... assuming that you have alsa.hpp in the same folder. // Usage: alsarate [ALSA device] [sampling rate] [time in seconds] // GNU GPLv3 or later. #include <alsa.hpp> #include <sys/time.h> #include <algorithm> #include <iomanip> #include <iostream> #include <sstream> #include <vector> double getTime() { timeval t; if (gettimeofday(&t, NULL) == -1) throw std::runtime_error("gettimeofday failed"); return t.tv_sec + t.tv_usec * 1e-6; } void test(char const* dev, unsigned long reqRate, double seconds, bool play) { std::cerr << ">>> Testing " << (play ? "playback" : "capture") << std::endl; try { std::cerr << "Opening ALSA device " << dev << " at " << reqRate << " Hz." << std::endl; double time = getTime(); unsigned int rate = reqRate; unsigned int channels = 1; unsigned int period = 1; alsa::PCM alsaHandle(dev, play ? SND_PCM_STREAM_PLAYBACK : SND_PCM_STREAM_CAPTURE); alsa::HWConfig(alsaHandle) .set(SND_PCM_ACCESS_RW_INTERLEAVED) .set(SND_PCM_FORMAT_S16_LE) .rate_near(rate) .channels_near(channels) .period_time_near(period) .commit(); std::cerr << "Got " << rate << " Hz, " << channels << " channels, period " << period * 1e-3 << " ms." << std::endl; int nFrames = 0; size_t len = rate * seconds; std::vector<short> buf(2048 * channels); for (size_t f = 0; f < len; f += nFrames) { unsigned int frames = std::min<unsigned int>(buf.size() / channels, len - f); if (play) { nFrames = snd_pcm_writei(alsaHandle, &buf[0], frames); } else { nFrames = snd_pcm_readi(alsaHandle, &buf[0], frames); } if (nFrames < 0) throw std::runtime_error("snd_pcm_readi or snd_pcm_writei failed"); } time = getTime() - time; std::cerr << std::fixed << std::setprecision(2) << time << " seconds, measured rate " << std::setprecision(0) << len / time << " Hz." << std::endl; } catch (std::exception& e) { std::cerr << "FATAL ERROR: " << e.what() << std::endl; } } // Should use Boost, but who has the headers installed? template <typename Out, typename In> Out lexical_cast(In val) { std::stringstream ss; Out ret; ss << val; if (ss >> ret) return ret; throw std::runtime_error("Invalid input, lexical cast failed"); } int main(int argc, char** argv) { char const* dev = argc > 1 ? argv[1] : "default"; unsigned int reqRate = argc > 2 ? lexical_cast<int>(argv[2]) : 48000; double time = argc > 3 ? lexical_cast<double>(argv[3]) : 10.0; test(dev, reqRate, time, true); test(dev, reqRate, time, false); }

1 0

[alsa-devel] [PATCH] Improved support for different bt87x board configurations
by Trent Piepho 08 Sep '07

08 Sep '07

After applying this patch, the bt87x.patch file in alsa-driver doesn't apply. I'm not sure what I'm supposed to do about that? # HG changeset patch # User Trent Piepho <xyzzy(a)speakeasy.org> # Date 1188469025 25200 # Node ID 33d453db23d246dade155a6fc3b91d8437a4b7f5 # Parent 52dfc5244360d2b0b119786596962ff5d0c9f338 snd-bt87x: Improve support for different board types Different cards have different audio configurations, but the driver didn't support this. The only setting it had was the digital rate. This patch adds a board configuration list. Currently, configurable items are the digital rate and the digital data format (for cards with an external ADC), a flag for the absence of an external ADC, and a flag for no connection to the Bt87x internal ADC. This allows cards that don't use the internal ADC to omit the ALSA "Bt87x analog" device and related controls. Cards without an external ADC can omit the "Bt87x digital" device. In order to support the CS5331A ADC used on the Osprey 440 and 2x0 cards, the digital format needs to be different than the default. Support could be added for defining: The connections or lack of them to the Bt87x's internal ADC mux Multiple sample rates for an external ADC (e.g. Osprey) Control of an external mux for an external ADC (e.g. Osprey) The card definitions for cards other than the Ospreys are kept equivalent to their old values. This is likely inaccurate for most cards, as it is doubtful that both an external and the internal ADC would be used. Lacking information on those cards, the behavior is left unchanged. Signed-off-by: Trent Piepho <xyzzy(a)speakeasy.org> diff -r 52dfc5244360 -r 33d453db23d2 pci/bt87x.c --- a/pci/bt87x.c Thu Aug 30 10:22:35 2007 +0200 +++ b/pci/bt87x.c Thu Aug 30 03:17:05 2007 -0700 @@ -147,14 +147,67 @@ MODULE_PARM_DESC(load_all, "Allow to loa /* SYNC, one WRITE per line, one extra WRITE per page boundary, SYNC, JUMP */ #define MAX_RISC_SIZE ((1 + 255 + (PAGE_ALIGN(255 * 4092) / PAGE_SIZE - 1) + 1 + 1) * 8) +/* Cards with configuration information */ +enum snd_bt87x_boardid { + SND_BT87X_BOARD_GENERIC, + SND_BT87X_BOARD_ANALOG, /* board with no external A/D */ + SND_BT87X_BOARD_HAUPPAUGE878, + SND_BT87X_BOARD_OSPREY2x0, + SND_BT87X_BOARD_OSPREY440, + SND_BT87X_BOARD_ATI_TVWONDER, + SND_BT87X_BOARD_WINFAST2000, + SND_BT87X_BOARD_VOODOOTV_200, + SND_BT87X_BOARD_AVPHONE98, +}; + +/* Card configuration */ +struct snd_bt87x_board { + int dig_rate; /* Digital input sampling rate */ + u32 digital_fmt; /* Register settings for digital input */ + unsigned no_analog:1; /* No analog input */ + unsigned no_digital:1; /* No digital input */ +}; + +static const __devinitdata struct snd_bt87x_board snd_bt87x_boards[] = { + [SND_BT87X_BOARD_GENERIC] = { + .dig_rate = 32000, + }, + [SND_BT87X_BOARD_ANALOG] = { + .no_digital = 1, + }, + [SND_BT87X_BOARD_HAUPPAUGE878] = { + .dig_rate = 32000, + }, + [SND_BT87X_BOARD_OSPREY2x0] = { + .dig_rate = 44100, + .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT), + }, + [SND_BT87X_BOARD_OSPREY440] = { + .dig_rate = 32000, + .digital_fmt = CTL_DA_LRI | (1 << CTL_DA_LRD_SHIFT), + .no_analog = 1, + }, + [SND_BT87X_BOARD_ATI_TVWONDER] = { + .dig_rate = 32000, + }, + [SND_BT87X_BOARD_WINFAST2000] = { + .dig_rate = 32000, + }, + [SND_BT87X_BOARD_VOODOOTV_200] = { + .dig_rate = 32000, + }, + [SND_BT87X_BOARD_AVPHONE98] = { + .dig_rate = 48000, + }, +}; + struct snd_bt87x { struct snd_card *card; struct pci_dev *pci; + struct snd_bt87x_board board; void __iomem *mmio; int irq; - - int dig_rate; spinlock_t reg_lock; long opened; @@ -342,9 +395,9 @@ static int snd_bt87x_set_digital_hw(stru { chip->reg_control |= CTL_DA_IOM_DA; runtime->hw = snd_bt87x_digital_hw; - runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->dig_rate); - runtime->hw.rate_min = chip->dig_rate; - runtime->hw.rate_max = chip->dig_rate; + runtime->hw.rates = snd_pcm_rate_to_rate_bit(chip->board.dig_rate); + runtime->hw.rate_min = chip->board.dig_rate; + runtime->hw.rate_max = chip->board.dig_rate; return 0; } @@ -709,9 +762,9 @@ static int __devinit snd_bt87x_create(st chip->mmio = ioremap_nocache(pci_resource_start(pci, 0), pci_resource_len(pci, 0)); if (!chip->mmio) { - snd_bt87x_free(chip); snd_printk(KERN_ERR "cannot remap io memory\n"); - return -ENOMEM; + err = -ENOMEM; + goto fail; } chip->reg_control = CTL_DA_ES2 | CTL_PKTP_16 | (15 << CTL_DA_SDR_SHIFT); @@ -720,54 +773,57 @@ static int __devinit snd_bt87x_create(st snd_bt87x_writel(chip, REG_INT_MASK, 0); snd_bt87x_writel(chip, REG_INT_STAT, MY_INTERRUPTS); - if (request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED, - "Bt87x audio", chip)) { - snd_bt87x_free(chip); - snd_printk(KERN_ERR "cannot grab irq\n"); - return -EBUSY; + err = request_irq(pci->irq, snd_bt87x_interrupt, IRQF_SHARED, + "Bt87x audio", chip); + if (err < 0) { + snd_printk(KERN_ERR "cannot grab irq %d\n", pci->irq); + goto fail; } chip->irq = pci->irq; pci_set_master(pci); synchronize_irq(chip->irq); err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops); - if (err < 0) { - snd_bt87x_free(chip); - return err; - } + if (err < 0) + goto fail; + snd_card_set_dev(card, &pci->dev); *rchip = chip; return 0; -} - -#define BT_DEVICE(chip, subvend, subdev, rate) \ + +fail: + snd_bt87x_free(chip); + return err; +} + +#define BT_DEVICE(chip, subvend, subdev, id) \ { .vendor = PCI_VENDOR_ID_BROOKTREE, \ - .device = chip, \ + .device = PCI_DEVICE_ID_BROOKTREE ## chip, \ .subvendor = subvend, .subdevice = subdev, \ - .driver_data = rate } - -/* driver_data is the default digital_rate value for that device */ + .driver_data = id } +/* driver_data is the card id for that device */ + static struct pci_device_id snd_bt87x_ids[] = { /* Hauppauge WinTV series */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0x13eb, 32000), + BT_DEVICE(_878, 0x0070, 0x13eb, SND_BT87X_BOARD_HAUPPAUGE878), /* Hauppauge WinTV series */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, 0x0070, 0x13eb, 32000), + BT_DEVICE(_879, 0x0070, 0x13eb, SND_BT87X_BOARD_HAUPPAUGE878), /* Viewcast Osprey 200 */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff01, 44100), + BT_DEVICE(_878, 0x0070, 0xff01, SND_BT87X_BOARD_OSPREY2x0), /* Viewcast Osprey 440 (rate is configurable via gpio) */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x0070, 0xff07, 32000), + BT_DEVICE(_878, 0x0070, 0xff07, SND_BT87X_BOARD_OSPREY440), /* ATI TV-Wonder */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1002, 0x0001, 32000), + BT_DEVICE(_878, 0x1002, 0x0001, SND_BT87X_BOARD_ATI_TVWONDER), /* Leadtek Winfast tv 2000xp delux */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x107d, 0x6606, 32000), + BT_DEVICE(_878, 0x107d, 0x6606, SND_BT87X_BOARD_WINFAST2000), /* Voodoo TV 200 */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x121a, 0x3000, 32000), + BT_DEVICE(_878, 0x121a, 0x3000, SND_BT87X_BOARD_VOODOOTV_200), /* AVerMedia Studio No. 103, 203, ...? */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1461, 0x0003, 48000), + BT_DEVICE(_878, 0x1461, 0x0003, SND_BT87X_BOARD_AVPHONE98), /* Prolink PixelView PV-M4900 */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0x1554, 0x4011, 32000), + BT_DEVICE(_878, 0x1554, 0x4011, SND_BT87X_BOARD_GENERIC), /* Pinnacle Studio PCTV rave */ - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, 0xbd11, 0x1200, 32000), + BT_DEVICE(_878, 0xbd11, 0x1200, SND_BT87X_BOARD_GENERIC), { } }; MODULE_DEVICE_TABLE(pci, snd_bt87x_ids); @@ -792,7 +848,7 @@ static struct { static struct pci_driver driver; -/* return the rate of the card, or a negative value if it's blacklisted */ +/* return the id of the card, or a negative value if it's blacklisted */ static int __devinit snd_bt87x_detect_card(struct pci_dev *pci) { int i; @@ -810,12 +866,12 @@ static int __devinit snd_bt87x_detect_ca return -EBUSY; } - snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x, using default rate 32000\n", - pci->device, pci->subsystem_vendor, pci->subsystem_device); + snd_printk(KERN_INFO "unknown card %#04x-%#04x:%#04x\n", + pci->device, pci->subsystem_vendor, pci->subsystem_device); snd_printk(KERN_DEBUG "please mail id, board name, and, " "if it works, the correct digital_rate option to " "<alsa-devel(a)alsa-project.org>\n"); - return 32000; /* default rate */ + return SND_BT87X_BOARD_GENERIC; } static int __devinit snd_bt87x_probe(struct pci_dev *pci, @@ -824,12 +880,15 @@ static int __devinit snd_bt87x_probe(str static int dev; struct snd_card *card; struct snd_bt87x *chip; - int err, rate; - - rate = pci_id->driver_data; - if (! rate) - if ((rate = snd_bt87x_detect_card(pci)) <= 0) + int err; + enum snd_bt87x_boardid boardid; + + if (! pci_id->driver_data) { + if ((err = snd_bt87x_detect_card(pci)) < 0) return -ENODEV; + boardid = err; + } else + boardid = pci_id->driver_data; if (dev >= SNDRV_CARDS) return -ENODEV; @@ -846,27 +905,36 @@ static int __devinit snd_bt87x_probe(str if (err < 0) goto _error; - if (digital_rate[dev] > 0) - chip->dig_rate = digital_rate[dev]; - else - chip->dig_rate = rate; - - err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital"); - if (err < 0) - goto _error; - err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog"); - if (err < 0) - goto _error; - - err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_volume, chip)); - if (err < 0) - goto _error; - err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_boost, chip)); - if (err < 0) - goto _error; - err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_source, chip)); - if (err < 0) - goto _error; + memcpy(&chip->board, &snd_bt87x_boards[boardid], sizeof(chip->board)); + + if (! chip->board.no_digital) { + if (digital_rate[dev] > 0) + chip->board.dig_rate = digital_rate[dev]; + + chip->reg_control |= chip->board.digital_fmt; + + err = snd_bt87x_pcm(chip, DEVICE_DIGITAL, "Bt87x Digital"); + if (err < 0) + goto _error; + } + if (! chip->board.no_analog) { + err = snd_bt87x_pcm(chip, DEVICE_ANALOG, "Bt87x Analog"); + if (err < 0) + goto _error; + err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_volume, chip)); + if (err < 0) + goto _error; + err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_boost, chip)); + if (err < 0) + goto _error; + err = snd_ctl_add(card, snd_ctl_new1(&snd_bt87x_capture_source, chip)); + if (err < 0) + goto _error; + } + snd_printk(KERN_INFO "bt87x%d: Using board %d, %sanalog, %sdigital (rate %d Hz)\n", + dev, boardid, chip->board.no_analog ? "no " : "", + chip->board.no_digital ? "no " : "", + chip->board.dig_rate); strcpy(card->driver, "Bt87x"); sprintf(card->shortname, "Brooktree Bt%x", pci->device); @@ -897,8 +965,8 @@ static void __devexit snd_bt87x_remove(s /* default entries for all Bt87x cards - it's not exported */ /* driver_data is set to 0 to call detection */ static struct pci_device_id snd_bt87x_default_ids[] __devinitdata = { - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_878, PCI_ANY_ID, PCI_ANY_ID, 0), - BT_DEVICE(PCI_DEVICE_ID_BROOKTREE_879, PCI_ANY_ID, PCI_ANY_ID, 0), + BT_DEVICE(_878, PCI_ANY_ID, PCI_ANY_ID, SND_BT87X_BOARD_GENERIC), + BT_DEVICE(_879, PCI_ANY_ID, PCI_ANY_ID, SND_BT87X_BOARD_GENERIC), { } };

4 18

[alsa-devel] Why is ALSA sending STOP/START commands to my driver during MP3 playback?
by Timur Tabi 08 Sep '07

08 Sep '07

I'm developing an ASoC driver, and when I play an MP3 with mpg123, my driver's trigger() function is called with the SNDRV_PCM_TRIGGER_STOP command and then is immediately called again with SNDRV_PCM_TRIGGER_START. Can someone tell me why ALSA would do this? This only happens when playing an MP3. When I use aplay to play a regular Wave file, I don't see this behavior. My driver does not have the SNDRV_PCM_INFO_PAUSE bit set, because I haven't gotten around to testing that capability. I suspect that if I did have that bit set, I would be getting SNDRV_PCM_TRIGGER_PAUSE_PUSH and SNDRV_PCM_TRIGGER_PAUSE_RELEASE calls instead. -- Timur Tabi Linux Kernel Developer @ Freescale

2 1

[alsa-devel] Bizarre ATI SB450 behavior
by Ryan Hughes 07 Sep '07

07 Sep '07

Hi. I've got a little situation over here. Every time I poll any software, it totally believes that the sound-card is installed and configured correctly and working. Nobody complains. But when I try to play sound, none comes out. People usually suggest the alsamixer, to make sure none of my channels are muted. Truly, none of them are muted. Here's the vital stats: lspci | grep -i audio 00:14.2 Audio device: ATI Technologies Inc SB450 HDA Audio (rev 01) aplay -l card 0: SB [HDA ATI SB], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod | grep snd snd_hda_intel 262424 1 snd_pcm_oss 44672 0 snd_mixer_oss 17792 1 snd_pcm_oss snd_pcm 80900 2 snd_hda_intel,snd_pcm_oss snd_seq_dummy 4996 0 snd_seq_oss 35072 0 snd_seq_midi 9728 0 snd_rawmidi 25984 1 snd_seq_midi snd_seq_midi_event 8576 2 snd_seq_oss,snd_seq_midi snd_seq 54000 6 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24196 2 snd_pcm,snd_seq snd_seq_device 9612 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 56324 12 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_seq_dummy,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device soundcore 8672 1 snd snd_page_alloc 11272 2 snd_hda_intel,snd_pcm Interesting pieces of dmesg [ 22.061858] ALSA /usr/src/alsa-driver-1.0.14/pci/hda/../../alsa-kernel/pci/hd a/patch_si3054.c:244: si3054: cannot initialize. EXT MID = 0000 [ 48.948578] hda-intel: Invalid position buffer, using LPIB read method instea d. Am I missing any useful info? Kernel is 2.6.20-16 Alsa is a hand-compiled 1.0.14. Thanks. --Ryan

2 1

Re: [alsa-devel] [PATCH] Gallant SC-6000 driver (4th version)
by Krzysztof Helt 07 Sep '07

07 Sep '07

Rene Herman wrote: > On 09/05/2007 10:25 PM, Krzysztof Helt wrote: > > I have left the ISA autodetection. It makes driver usage simpler for user. > > The base port, mpu port and mss port can be read from jumper settings, > >but irq or dma is not settable. A system may figure out which one is free > >and use it. > The problem is that this "simplification" might actually leave the non-aware > user scratching his head for far longer than if he has to spell things out. I know it may not work all the time, but you can always fall back to giving number. If the simple autodetection works, it helps. > Even without that problem, we have the problem of IRQs that's only _appear_ > to be free. IRQ7 for example will usually be assigned to the first parallel > port but will only show up as taken if the parport driver is loaded. Often, > it'll still actually work, but (again) here for example, it does not. Actually, the parport_pc driver is loaded by the udev by default, so the IRQ7 cannot be requested (I tested - IRQ7 request failed even if I had not used the printer). > It's not like people using legacy ISA in this day and age are not normally > nerds anyway... That's the best argument. > Yes. I am looking at the sound galaxy driver at the moment and am in fact > now trying to think of why I want to use the WSS part of the card and not > just the SBP part. In DOS, the card is setup in SB Pro mode by default as > well, and the CS42xx seems to only be present really for Windows Sound > System compatibility for old 16-bit Windows. IIRC from SB Pro times, the SB Pro specification said it supported only 16 bit of output (ment either 16-bit mono or 8-bit stereo). The SB16 supported stereo 16-bit audio. I don't know what is expected from WSS, but if the SB Pro mode gives full duplex 16-bit stereo, there is no reason to use WSS mode. >> (I do not have the CS4231 based card but I should have it next week - >> Sound Galaxy Waverider). > Am testing on that one currently -- will post the driver before next week. I'll be happy to test it too. > Quite sure that 5 works by the way? On a Sound Galaxy, 5 only works for the > SB part (and 11 only for the WSS). It is the IRQ found most times as the first free during card tests. It worked on all free IRQs from the range (the 10 or 11 is occupied by USB/PS2 and I haven't bothered to make it free). It seems that ASC-9308 is more capable (in this respect) than SG chips. Regards, Krzysztof ---------------------------------------------------- Pierwsza od piętnastu lat książka Václava Havla "Tylko krótko proszę" - już w księgarniach! http://klik.wp.pl/?adr=http%3A%2F%2Fadv.reklama.wp.pl%2Fas%2Fhavel.html&sid…

4 10

[alsa-devel] aplay -M (mmap) omap
by Lopez Cruz, Misael 07 Sep '07

07 Sep '07

Hi, I'm experiencing kernel crashes in an omap platform when playing audio files with 'aplay -M' (mmap option) the second time a file is played. Any suggestion where to start debugging this? Is there a common cause for this failure? Misael

2 1

[alsa-devel] [PATCH] snd-bt87x: Power down audio ADC when not in use
by Trent Piepho 07 Sep '07

07 Sep '07

Sets a bit to power down the Bt87x's internal audio ADC when the ALSA device isn't open, or when it is in "digital mode" using an external ADC.

2 1

[alsa-devel] [PATCH] sun-cs4231: checkpatch fixes
by Krzysztof Helt 07 Sep '07

07 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes white spaces and issues pointed by the checkpatch.pl script. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-2.6.23.ref/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-2.6.23.ref/sound/sparc/cs4231.c 2007-09-04 20:01:03.000000000 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-04 19:58:45.000000000 +0200 @@ -3,7 +3,7 @@ * Copyright (C) 2002 David S. Miller <davem(a)redhat.com> * * Based entirely upon drivers/sbus/audio/cs4231.c which is: - * Copyright (C) 1996, 1997, 1998, 1998 Derrick J Brashear (shadow(a)andrew.cmu.edu) + * Copyright (C) 1996, 1997, 1998 Derrick J Brashear (shadow(a)andrew.cmu.edu) * and also sound/isa/cs423x/cs4231_lib.c which is: * Copyright (c) by Jaroslav Kysela <perex(a)suse.cz> */ @@ -15,6 +15,9 @@ #include <linux/init.h> #include <linux/interrupt.h> #include <linux/moduleparam.h> +#include <linux/irq.h> +#include <linux/io.h> + #include <sound/driver.h> #include <sound/core.h> @@ -25,9 +28,6 @@ #include <sound/initval.h> #include <sound/pcm_params.h> -#include <asm/io.h> -#include <asm/irq.h> - #ifdef CONFIG_SBUS #define SBUS_SUPPORT #include <asm/sbus.h> @@ -41,7 +41,8 @@ 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 */ +/* Enable this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Sun CS4231 soundcard."); @@ -56,19 +57,22 @@ MODULE_SUPPORTED_DEVICE("{{Sun,CS4231}}" #ifdef SBUS_SUPPORT struct sbus_dma_info { - spinlock_t lock; - int dir; - void __iomem *regs; + spinlock_t lock; /* DMA access lock */ + int dir; + void __iomem *regs; }; #endif struct snd_cs4231; struct cs4231_dma_control { - void (*prepare)(struct cs4231_dma_control *dma_cont, int dir); - void (*enable)(struct cs4231_dma_control *dma_cont, int on); - int (*request)(struct cs4231_dma_control *dma_cont, dma_addr_t bus_addr, size_t len); - unsigned int (*address)(struct cs4231_dma_control *dma_cont); - void (*preallocate)(struct snd_cs4231 *chip, struct snd_pcm *pcm); + void (*prepare)(struct cs4231_dma_control *dma_cont, + int dir); + void (*enable)(struct cs4231_dma_control *dma_cont, int on); + int (*request)(struct cs4231_dma_control *dma_cont, + dma_addr_t bus_addr, size_t len); + unsigned int (*address)(struct cs4231_dma_control *dma_cont); + void (*preallocate)(struct snd_cs4231 *chip, + struct snd_pcm *pcm); #ifdef EBUS_SUPPORT struct ebus_dma_info ebus_info; #endif @@ -78,7 +82,7 @@ struct cs4231_dma_control { }; struct snd_cs4231 { - spinlock_t lock; + spinlock_t lock; /* registers access lock */ void __iomem *port; struct cs4231_dma_control p_dma; @@ -102,13 +106,14 @@ struct snd_cs4231 { #define CS4231_MODE_PLAY 0x0001 #define CS4231_MODE_RECORD 0x0002 #define CS4231_MODE_TIMER 0x0004 -#define CS4231_MODE_OPEN (CS4231_MODE_PLAY|CS4231_MODE_RECORD|CS4231_MODE_TIMER) +#define CS4231_MODE_OPEN (CS4231_MODE_PLAY | CS4231_MODE_RECORD | \ + CS4231_MODE_TIMER) unsigned char image[32]; /* registers image */ int mce_bit; int calibrate_mute; - struct mutex mce_mutex; - struct mutex open_mutex; + struct mutex mce_mutex; /* mutex for mce register */ + struct mutex open_mutex; /* mutex for ALSA open/close */ union { #ifdef SBUS_SUPPORT @@ -273,7 +278,8 @@ static u8 __cs4231_readb(struct snd_cs42 #endif } -static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, void __iomem *reg_addr) +static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, + void __iomem *reg_addr) { #ifdef EBUS_SUPPORT if (cp->flags & CS4231_FLAG_EBUS) @@ -301,7 +307,8 @@ static void snd_cs4231_ready(struct snd_ } } -static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, + unsigned char value) { snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG @@ -355,7 +362,7 @@ static void snd_cs4231_busy_wait(struct { int timeout; - /* huh.. looks like this sequence is proper for CS4231A chip (GUS MAX) */ + /* looks like this sequence is proper for CS4231A chip (GUS MAX) */ for (timeout = 5; timeout > 0; timeout--) __cs4231_readb(chip, CS4231U(chip, REGSEL)); @@ -382,7 +389,9 @@ static void snd_cs4231_mce_up(struct snd chip->mce_bit |= CS4231_MCE; timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); if (timeout == 0x80) - snd_printdd("mce_up [%p]: serious init problem - codec still busy\n", chip->port); + snd_printdd("mce_up [%p]: serious init problem - " + "codec still busy\n", + chip->port); if (!(timeout & CS4231_MCE)) __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), CS4231U(chip, REGSEL)); @@ -406,7 +415,9 @@ static void snd_cs4231_mce_down(struct s __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), CS4231U(chip, REGSEL)); if (timeout == 0x80) - snd_printdd("mce_down [%p]: serious init problem - codec still busy\n", chip->port); + snd_printdd("mce_down [%p]: serious init problem - " + "codec still busy\n", + chip->port); if ((timeout & CS4231_MCE) == 0) { spin_unlock_irqrestore(&chip->lock, flags); return; @@ -417,7 +428,8 @@ static void snd_cs4231_mce_down(struct s snd_cs4231_ready(chip); snd_cs4231_ready(chip); - if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { + timeout = snd_cs4231_in(chip, CS4231_TEST_INIT); + if ((timeout & CS4231_CALIB_IN_PROGRESS) == 0) { snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); spin_unlock_irqrestore(&chip->lock, flags); return; @@ -425,10 +437,13 @@ static void snd_cs4231_mce_down(struct s /* in 10ms increments, check condition, up to 250ms */ timeout = 25; - while (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) { + while (snd_cs4231_in(chip, CS4231_TEST_INIT) & + CS4231_CALIB_IN_PROGRESS) { + spin_unlock_irqrestore(&chip->lock, flags); if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (2)\n"); + snd_printk("mce_down - " + "auto calibration time out (2)\n"); return; } msleep(10); @@ -440,7 +455,8 @@ static void snd_cs4231_mce_down(struct s while (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) { spin_unlock_irqrestore(&chip->lock, flags); if (--timeout < 0) { - snd_printk("mce_down - auto calibration time out (3)\n"); + snd_printk("mce_down - " + "auto calibration time out (3)\n"); return; } msleep(10); @@ -461,7 +477,8 @@ static void snd_cs4231_advance_dma(struc BUG_ON(period_size >= (1 << 24)); - if (dma_cont->request(dma_cont, runtime->dma_addr + offset, period_size)) + if (dma_cont->request(dma_cont, + runtime->dma_addr + offset, period_size)) return; (*periods_sent) = ((*periods_sent) + 1) % runtime->periods; } @@ -554,21 +571,32 @@ static unsigned char snd_cs4231_get_rate for (i = 0; i < 14; i++) if (rate == rates[i]) return freq_bits[i]; - // snd_BUG(); + return freq_bits[13]; } -static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, int format, int channels) +static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip, int format, + int channels) { unsigned char rformat; rformat = CS4231_LINEAR_8; switch (format) { - case SNDRV_PCM_FORMAT_MU_LAW: rformat = CS4231_ULAW_8; break; - case SNDRV_PCM_FORMAT_A_LAW: rformat = CS4231_ALAW_8; break; - case SNDRV_PCM_FORMAT_S16_LE: rformat = CS4231_LINEAR_16; break; - case SNDRV_PCM_FORMAT_S16_BE: rformat = CS4231_LINEAR_16_BIG; break; - case SNDRV_PCM_FORMAT_IMA_ADPCM: rformat = CS4231_ADPCM_16; break; + case SNDRV_PCM_FORMAT_MU_LAW: + rformat = CS4231_ULAW_8; + break; + case SNDRV_PCM_FORMAT_A_LAW: + rformat = CS4231_ALAW_8; + break; + case SNDRV_PCM_FORMAT_S16_LE: + rformat = CS4231_LINEAR_16; + break; + case SNDRV_PCM_FORMAT_S16_BE: + rformat = CS4231_LINEAR_16_BIG; + break; + case SNDRV_PCM_FORMAT_IMA_ADPCM: + rformat = CS4231_ADPCM_16; + break; } if (channels > 1) rformat |= CS4231_STEREO; @@ -615,7 +643,8 @@ static void snd_cs4231_calibrate_mute(st spin_unlock_irqrestore(&chip->lock, flags); } -static void snd_cs4231_playback_format(struct snd_cs4231 *chip, struct snd_pcm_hw_params *params, +static void snd_cs4231_playback_format(struct snd_cs4231 *chip, + struct snd_pcm_hw_params *params, unsigned char pdfr) { unsigned long flags; @@ -638,8 +667,9 @@ static void snd_cs4231_playback_format(s mutex_unlock(&chip->mce_mutex); } -static void snd_cs4231_capture_format(struct snd_cs4231 *chip, struct snd_pcm_hw_params *params, - unsigned char cdfr) +static void snd_cs4231_capture_format(struct snd_cs4231 *chip, + struct snd_pcm_hw_params *params, + unsigned char cdfr) { unsigned long flags; @@ -696,7 +726,8 @@ static int snd_cs4231_timer_start(struct chip->image[CS4231_TIMER_LOW] = (unsigned char) ticks); snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, - chip->image[CS4231_ALT_FEATURE_1] | CS4231_TIMER_ENABLE); + chip->image[CS4231_ALT_FEATURE_1] | + CS4231_TIMER_ENABLE); } spin_unlock_irqrestore(&chip->lock, flags); @@ -709,8 +740,9 @@ static int snd_cs4231_timer_stop(struct struct snd_cs4231 *chip = snd_timer_chip(timer); spin_lock_irqsave(&chip->lock, flags); + chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE; snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, - chip->image[CS4231_ALT_FEATURE_1] &= ~CS4231_TIMER_ENABLE); + chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->lock, flags); return 0; @@ -727,8 +759,10 @@ static void __init snd_cs4231_init(struc #endif snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | - CS4231_RECORD_ENABLE | CS4231_RECORD_PIO | + chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | + CS4231_PLAYBACK_PIO | + CS4231_RECORD_ENABLE | + CS4231_RECORD_PIO | CS4231_CALIB_MODE); chip->image[CS4231_IFACE_CTRL] |= CS4231_AUTOCALIB; snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); @@ -741,21 +775,25 @@ static void __init snd_cs4231_init(struc snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, chip->image[CS4231_ALT_FEATURE_1]); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_1, + chip->image[CS4231_ALT_FEATURE_1]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); #ifdef SNDRV_DEBUG_MCE - snd_printdd("init: (3) - afei = 0x%x\n", chip->image[CS4231_ALT_FEATURE_1]); + snd_printdd("init: (3) - afei = 0x%x\n", + chip->image[CS4231_ALT_FEATURE_1]); #endif spin_lock_irqsave(&chip->lock, flags); - snd_cs4231_out(chip, CS4231_ALT_FEATURE_2, chip->image[CS4231_ALT_FEATURE_2]); + snd_cs4231_out(chip, CS4231_ALT_FEATURE_2, + chip->image[CS4231_ALT_FEATURE_2]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_up(chip); spin_lock_irqsave(&chip->lock, flags); - snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, chip->image[CS4231_PLAYBK_FORMAT]); + snd_cs4231_out(chip, CS4231_PLAYBK_FORMAT, + chip->image[CS4231_PLAYBK_FORMAT]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); @@ -838,7 +876,8 @@ static void snd_cs4231_close(struct snd_ chip->image[CS4231_IFACE_CTRL] &= ~(CS4231_PLAYBACK_ENABLE | CS4231_PLAYBACK_PIO | CS4231_RECORD_ENABLE | CS4231_RECORD_PIO); - snd_cs4231_out(chip, CS4231_IFACE_CTRL, chip->image[CS4231_IFACE_CTRL]); + snd_cs4231_out(chip, CS4231_IFACE_CTRL, + chip->image[CS4231_IFACE_CTRL]); spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_mce_down(chip); spin_lock_irqsave(&chip->lock, flags); @@ -867,15 +906,14 @@ static int snd_cs4231_timer_open(struct return 0; } -static int snd_cs4231_timer_close(struct snd_timer * timer) +static int snd_cs4231_timer_close(struct snd_timer *timer) { struct snd_cs4231 *chip = snd_timer_chip(timer); snd_cs4231_close(chip, CS4231_MODE_TIMER); return 0; } -static struct snd_timer_hardware snd_cs4231_timer_table = -{ +static struct snd_timer_hardware snd_cs4231_timer_table = { .flags = SNDRV_TIMER_HW_AUTO, .resolution = 9945, .ticks = 65535, @@ -897,8 +935,9 @@ static int snd_cs4231_playback_hw_params unsigned char new_pdfr; int err; - if ((err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params))) < 0) + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) return err; new_pdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | @@ -934,8 +973,9 @@ static int snd_cs4231_capture_hw_params( unsigned char new_cdfr; int err; - if ((err = snd_pcm_lib_malloc_pages(substream, - params_buffer_bytes(hw_params))) < 0) + err = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + if (err < 0) return err; new_cdfr = snd_cs4231_get_format(chip, params_format(hw_params), params_channels(hw_params)) | @@ -970,7 +1010,8 @@ static void snd_cs4231_overrange(struct res = snd_cs4231_in(chip, CS4231_TEST_INIT); spin_unlock_irqrestore(&chip->lock, flags); - if (res & (0x08 | 0x02)) /* detect overrange only above 0dB; may be user selectable? */ + /* detect overrange only above 0dB; may be user selectable? */ + if (res & (0x08 | 0x02)) chip->capture_substream->runtime->overrange++; } @@ -992,47 +1033,50 @@ static void snd_cs4231_capture_callback( } } -static snd_pcm_uframes_t snd_cs4231_playback_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t snd_cs4231_playback_pointer + (struct snd_pcm_substream *substream) { struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); struct cs4231_dma_control *dma_cont = &chip->p_dma; size_t ptr; - + if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_PLAYBACK_ENABLE)) return 0; ptr = dma_cont->address(dma_cont); if (ptr != 0) ptr -= substream->runtime->dma_addr; - + return bytes_to_frames(substream->runtime, ptr); } -static snd_pcm_uframes_t snd_cs4231_capture_pointer(struct snd_pcm_substream *substream) +static snd_pcm_uframes_t snd_cs4231_capture_pointer + (struct snd_pcm_substream *substream) { struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); struct cs4231_dma_control *dma_cont = &chip->c_dma; size_t ptr; - + if (!(chip->image[CS4231_IFACE_CTRL] & CS4231_RECORD_ENABLE)) return 0; ptr = dma_cont->address(dma_cont); if (ptr != 0) ptr -= substream->runtime->dma_addr; - + return bytes_to_frames(substream->runtime, ptr); } static int __init snd_cs4231_probe(struct snd_cs4231 *chip) { unsigned long flags; - int i, id, vers; + int i; + int id = 0; + int vers = 0; unsigned char *ptr; - id = vers = 0; for (i = 0; i < 50; i++) { mb(); if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) - udelay(2000); + msleep(2); else { spin_lock_irqsave(&chip->lock, flags); snd_cs4231_out(chip, CS4231_MISC_INFO, CS4231_MODE2); @@ -1084,42 +1128,50 @@ static int __init snd_cs4231_probe(struc return 0; /* all things are ok.. */ } -static struct snd_pcm_hardware snd_cs4231_playback = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, +static struct snd_pcm_hardware snd_cs4231_playback = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_MU_LAW | + SNDRV_PCM_FMTBIT_A_LAW | + SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_KNOT | + SNDRV_PCM_RATE_8000_48000, .rate_min = 5510, .rate_max = 48000, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = (32*1024), + .buffer_bytes_max = 32 * 1024, .period_bytes_min = 64, - .period_bytes_max = (32*1024), + .period_bytes_max = 32 * 1024, .periods_min = 1, .periods_max = 1024, }; -static struct snd_pcm_hardware snd_cs4231_capture = -{ - .info = (SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_SYNC_START), - .formats = (SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_IMA_ADPCM | - SNDRV_PCM_FMTBIT_U8 | SNDRV_PCM_FMTBIT_S16_LE | - SNDRV_PCM_FMTBIT_S16_BE), - .rates = SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_8000_48000, +static struct snd_pcm_hardware snd_cs4231_capture = { + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START, + .formats = SNDRV_PCM_FMTBIT_MU_LAW | + SNDRV_PCM_FMTBIT_A_LAW | + SNDRV_PCM_FMTBIT_IMA_ADPCM | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_KNOT | + SNDRV_PCM_RATE_8000_48000, .rate_min = 5510, .rate_max = 48000, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = (32*1024), + .buffer_bytes_max = 32 * 1024, .period_bytes_min = 64, - .period_bytes_max = (32*1024), + .period_bytes_max = 32 * 1024, .periods_min = 1, .periods_max = 1024, }; @@ -1132,7 +1184,8 @@ static int snd_cs4231_playback_open(stru runtime->hw = snd_cs4231_playback; - if ((err = snd_cs4231_open(chip, CS4231_MODE_PLAY)) < 0) { + err = snd_cs4231_open(chip, CS4231_MODE_PLAY); + if (err < 0) { snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } @@ -1152,7 +1205,8 @@ static int snd_cs4231_capture_open(struc runtime->hw = snd_cs4231_capture; - if ((err = snd_cs4231_open(chip, CS4231_MODE_RECORD)) < 0) { + err = snd_cs4231_open(chip, CS4231_MODE_RECORD); + if (err < 0) { snd_free_pages(runtime->dma_area, runtime->dma_bytes); return err; } @@ -1220,9 +1274,11 @@ static int __init snd_cs4231_pcm(struct if (err < 0) return err; - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); - snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_cs4231_capture_ops); - + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_cs4231_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_cs4231_capture_ops); + /* global setup */ pcm->private_data = chip; pcm->info_flags = SNDRV_PCM_INFO_JOINT_DUPLEX; @@ -1258,7 +1314,7 @@ static int __init snd_cs4231_timer(struc return 0; } - + /* * MIXER part */ @@ -1275,7 +1331,8 @@ static int snd_cs4231_info_mux(struct sn uinfo->value.enumerated.items = 4; if (uinfo->value.enumerated.item > 3) uinfo->value.enumerated.item = 3; - strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]); + strcpy(uinfo->value.enumerated.name, + texts[uinfo->value.enumerated.item]); return 0; } @@ -1285,7 +1342,7 @@ static int snd_cs4231_get_mux(struct snd { struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); unsigned long flags; - + spin_lock_irqsave(&chip->lock, flags); ucontrol->value.enumerated.item[0] = (chip->image[CS4231_LEFT_INPUT] & CS4231_MIXS_ALL) >> 6; @@ -1303,7 +1360,7 @@ static int snd_cs4231_put_mux(struct snd unsigned long flags; unsigned short left, right; int change; - + if (ucontrol->value.enumerated.item[0] > 3 || ucontrol->value.enumerated.item[1] > 3) return -EINVAL; @@ -1315,7 +1372,7 @@ static int snd_cs4231_put_mux(struct snd left = (chip->image[CS4231_LEFT_INPUT] & ~CS4231_MIXS_ALL) | left; right = (chip->image[CS4231_RIGHT_INPUT] & ~CS4231_MIXS_ALL) | right; change = left != chip->image[CS4231_LEFT_INPUT] || - right != chip->image[CS4231_RIGHT_INPUT]; + right != chip->image[CS4231_RIGHT_INPUT]; snd_cs4231_out(chip, CS4231_LEFT_INPUT, left); snd_cs4231_out(chip, CS4231_RIGHT_INPUT, right); @@ -1347,7 +1404,7 @@ static int snd_cs4231_get_single(struct int shift = (kcontrol->private_value >> 8) & 0xff; int mask = (kcontrol->private_value >> 16) & 0xff; int invert = (kcontrol->private_value >> 24) & 0xff; - + spin_lock_irqsave(&chip->lock, flags); ucontrol->value.integer.value[0] = (chip->image[reg] >> shift) & mask; @@ -1372,7 +1429,7 @@ static int snd_cs4231_put_single(struct int invert = (kcontrol->private_value >> 24) & 0xff; int change; unsigned short val; - + val = (ucontrol->value.integer.value[0] & mask); if (invert) val = mask - val; @@ -1414,11 +1471,13 @@ static int snd_cs4231_get_double(struct int shift_right = (kcontrol->private_value >> 19) & 0x07; int mask = (kcontrol->private_value >> 24) & 0xff; int invert = (kcontrol->private_value >> 22) & 1; - + spin_lock_irqsave(&chip->lock, flags); - ucontrol->value.integer.value[0] = (chip->image[left_reg] >> shift_left) & mask; - ucontrol->value.integer.value[1] = (chip->image[right_reg] >> shift_right) & mask; + ucontrol->value.integer.value[0] = + (chip->image[left_reg] >> shift_left) & mask; + ucontrol->value.integer.value[1] = + (chip->image[right_reg] >> shift_right) & mask; spin_unlock_irqrestore(&chip->lock, flags); @@ -1445,7 +1504,7 @@ static int snd_cs4231_put_double(struct int invert = (kcontrol->private_value >> 22) & 1; int change; unsigned short val1, val2; - + val1 = ucontrol->value.integer.value[0] & mask; val2 = ucontrol->value.integer.value[1] & mask; if (invert) { @@ -1459,7 +1518,8 @@ static int snd_cs4231_put_double(struct val1 = (chip->image[left_reg] & ~(mask << shift_left)) | val1; val2 = (chip->image[right_reg] & ~(mask << shift_right)) | val2; - change = val1 != chip->image[left_reg] || val2 != chip->image[right_reg]; + change = val1 != chip->image[left_reg]; + change |= val2 != chip->image[right_reg]; snd_cs4231_out(chip, left_reg, val1); snd_cs4231_out(chip, right_reg, val2); @@ -1469,31 +1529,42 @@ static int snd_cs4231_put_double(struct } #define CS4231_SINGLE(xname, xindex, reg, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_single, \ - .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ - .private_value = reg | (shift << 8) | (mask << 16) | (invert << 24) } - -#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, shift_right, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, .index = xindex, \ - .info = snd_cs4231_info_double, \ - .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ - .private_value = left_reg | (right_reg << 8) | (shift_left << 16) | (shift_right << 19) | (mask << 24) | (invert << 22) } +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .index = (xindex), \ + .info = snd_cs4231_info_single, \ + .get = snd_cs4231_get_single, .put = snd_cs4231_put_single, \ + .private_value = (reg) | ((shift) << 8) | ((mask) << 16) | ((invert) << 24) } + +#define CS4231_DOUBLE(xname, xindex, left_reg, right_reg, shift_left, \ + shift_right, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), .index = (xindex), \ + .info = snd_cs4231_info_double, \ + .get = snd_cs4231_get_double, .put = snd_cs4231_put_double, \ + .private_value = (left_reg) | ((right_reg) << 8) | ((shift_left) << 16) | \ + ((shift_right) << 19) | ((mask) << 24) | ((invert) << 22) } static struct snd_kcontrol_new snd_cs4231_controls[] __initdata = { -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, 63, 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, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), -CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), -CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, CS4231_AUX2_RIGHT_INPUT, 0, 0, 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, 63, 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, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 0, CS4231_AUX1_LEFT_INPUT, + CS4231_AUX1_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 0, CS4231_AUX1_LEFT_INPUT, + CS4231_AUX1_RIGHT_INPUT, 0, 0, 31, 1), +CS4231_DOUBLE("Aux Playback Switch", 1, CS4231_AUX2_LEFT_INPUT, + CS4231_AUX2_RIGHT_INPUT, 7, 7, 1, 1), +CS4231_DOUBLE("Aux Playback Volume", 1, CS4231_AUX2_LEFT_INPUT, + CS4231_AUX2_RIGHT_INPUT, 0, 0, 31, 1), CS4231_SINGLE("Mono Playback Switch", 0, CS4231_MONO_CTRL, 7, 1, 1), CS4231_SINGLE("Mono Playback Volume", 0, CS4231_MONO_CTRL, 0, 15, 1), CS4231_SINGLE("Mono Output Playback Switch", 0, CS4231_MONO_CTRL, 6, 1, 1), CS4231_SINGLE("Mono Output Playback Bypass", 0, CS4231_MONO_CTRL, 5, 1, 0), -CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, 15, 0), +CS4231_DOUBLE("Capture Volume", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 0, 0, + 15, 0), { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Capture Source", @@ -1501,14 +1572,15 @@ CS4231_DOUBLE("Capture Volume", 0, CS423 .get = snd_cs4231_get_mux, .put = snd_cs4231_put_mux, }, -CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, 1, 0), +CS4231_DOUBLE("Mic Boost", 0, CS4231_LEFT_INPUT, CS4231_RIGHT_INPUT, 5, 5, + 1, 0), CS4231_SINGLE("Loopback Capture Switch", 0, CS4231_LOOPBACK, 0, 1, 0), CS4231_SINGLE("Loopback Capture Volume", 0, CS4231_LOOPBACK, 2, 63, 1), /* SPARC specific uses of XCTL{0,1} general purpose outputs. */ CS4231_SINGLE("Line Out Switch", 0, CS4231_PIN_CTRL, 6, 1, 1), CS4231_SINGLE("Headphone Out Switch", 0, CS4231_PIN_CTRL, 7, 1, 1) }; - + static int __init snd_cs4231_mixer(struct snd_card *card) { struct snd_cs4231 *chip = card->private_data; @@ -1609,24 +1681,24 @@ static irqreturn_t snd_cs4231_sbus_inter sbus_writel(csr, chip->port + APCCSR); - if ((csr & APC_PDMA_READY) && - (csr & APC_PLAY_INT) && + if ((csr & APC_PDMA_READY) && + (csr & APC_PLAY_INT) && (csr & APC_XINT_PNVA) && !(csr & APC_XINT_EMPT)) snd_cs4231_play_callback(chip); - if ((csr & APC_CDMA_READY) && - (csr & APC_CAPT_INT) && + if ((csr & APC_CDMA_READY) && + (csr & APC_CAPT_INT) && (csr & APC_XINT_CNVA) && !(csr & APC_XINT_EMPT)) snd_cs4231_capture_callback(chip); - + status = snd_cs4231_in(chip, CS4231_IRQ_STATUS); if (status & CS4231_TIMER_IRQ) { if (chip->timer) snd_timer_interrupt(chip->timer, chip->timer->sticks); - } + } if ((status & CS4231_RECORD_IRQ) && (csr & APC_CDMA_READY)) snd_cs4231_overrange(chip); @@ -1643,26 +1715,27 @@ static irqreturn_t snd_cs4231_sbus_inter * SBUS DMA routines */ -static int sbus_dma_request(struct cs4231_dma_control *dma_cont, dma_addr_t bus_addr, size_t len) +static int sbus_dma_request(struct cs4231_dma_control *dma_cont, + dma_addr_t bus_addr, size_t len) { unsigned long flags; u32 test, csr; int err; struct sbus_dma_info *base = &dma_cont->sbus_info; - + if (len >= (1 << 24)) return -EINVAL; spin_lock_irqsave(&base->lock, flags); csr = sbus_readl(base->regs + APCCSR); err = -EINVAL; test = APC_CDMA_READY; - if ( base->dir == APC_PLAY ) + if (base->dir == APC_PLAY) test = APC_PDMA_READY; if (!(csr & test)) goto out; err = -EBUSY; test = APC_XINT_CNVA; - if ( base->dir == APC_PLAY ) + if (base->dir == APC_PLAY) test = APC_XINT_PNVA; if (!(csr & test)) goto out; @@ -1685,7 +1758,7 @@ static void sbus_dma_prepare(struct cs42 test = APC_GENL_INT | APC_PLAY_INT | APC_XINT_ENA | APC_XINT_PLAY | APC_XINT_PEMP | APC_XINT_GENL | APC_XINT_PENA; - if ( base->dir == APC_RECORD ) + if (base->dir == APC_RECORD) test = APC_GENL_INT | APC_CAPT_INT | APC_XINT_ENA | APC_XINT_CAPT | APC_XINT_CEMP | APC_XINT_GENL; csr |= test; @@ -1703,28 +1776,28 @@ static void sbus_dma_enable(struct cs423 if (!on) { sbus_writel(0, base->regs + base->dir + APCNC); sbus_writel(0, base->regs + base->dir + APCNVA); - if ( base->dir == APC_PLAY ) { + if (base->dir == APC_PLAY) { sbus_writel(0, base->regs + base->dir + APCC); sbus_writel(0, base->regs + base->dir + APCVA); } udelay(1200); - } + } csr = sbus_readl(base->regs + APCCSR); shift = 0; - if ( base->dir == APC_PLAY ) + if (base->dir == APC_PLAY) shift = 1; if (on) csr &= ~(APC_CPAUSE << shift); else - csr |= (APC_CPAUSE << shift); + csr |= (APC_CPAUSE << shift); sbus_writel(csr, base->regs + APCCSR); if (on) csr |= (APC_CDMA_READY << shift); else csr &= ~(APC_CDMA_READY << shift); sbus_writel(csr, base->regs + APCCSR); - + spin_unlock_irqrestore(&base->lock, flags); } @@ -1732,14 +1805,14 @@ static unsigned int sbus_dma_addr(struct { struct sbus_dma_info *base = &dma_cont->sbus_info; - return sbus_readl(base->regs + base->dir + APCVA); + return sbus_readl(base->regs + base->dir + APCVA); } static void sbus_dma_preallocate(struct snd_cs4231 *chip, struct snd_pcm *pcm) { snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_SBUS, - snd_dma_sbus_data(chip->dev_u.sdev), - 64*1024, 128*1024); + snd_dma_sbus_data(chip->dev_u.sdev), + 64 * 1024, 128 * 1024); } /* @@ -1861,14 +1934,16 @@ static int __init cs4231_sbus_attach(str #ifdef EBUS_SUPPORT -static void snd_cs4231_ebus_play_callback(struct ebus_dma_info *p, int event, void *cookie) +static void snd_cs4231_ebus_play_callback(struct ebus_dma_info *p, int event, + void *cookie) { struct snd_cs4231 *chip = cookie; - + snd_cs4231_play_callback(chip); } -static void snd_cs4231_ebus_capture_callback(struct ebus_dma_info *p, int event, void *cookie) +static void snd_cs4231_ebus_capture_callback(struct ebus_dma_info *p, + int event, void *cookie) { struct snd_cs4231 *chip = cookie; @@ -1879,7 +1954,8 @@ static void snd_cs4231_ebus_capture_call * EBUS DMA wrappers */ -static int _ebus_dma_request(struct cs4231_dma_control *dma_cont, dma_addr_t bus_addr, size_t len) +static int _ebus_dma_request(struct cs4231_dma_control *dma_cont, + dma_addr_t bus_addr, size_t len) { return ebus_dma_request(&dma_cont->ebus_info, bus_addr, len); } @@ -1980,7 +2056,8 @@ static int __init snd_cs4231_ebus_create chip->port = ioremap(edev->resource[0].start, 0x10); chip->p_dma.ebus_info.regs = ioremap(edev->resource[1].start, 0x10); chip->c_dma.ebus_info.regs = ioremap(edev->resource[2].start, 0x10); - if (!chip->port || !chip->p_dma.ebus_info.regs || !chip->c_dma.ebus_info.regs) { + if (!chip->port || !chip->p_dma.ebus_info.regs || + !chip->c_dma.ebus_info.regs) { snd_cs4231_ebus_free(chip); snd_printdd("cs4231-%d: Unable to map chip registers.\n", dev); return -EIO; @@ -1988,18 +2065,21 @@ static int __init snd_cs4231_ebus_create if (ebus_dma_register(&chip->c_dma.ebus_info)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to register EBUS capture DMA\n", dev); + snd_printdd("cs4231-%d: Unable to register EBUS capture DMA\n", + dev); return -EBUSY; } if (ebus_dma_irq_enable(&chip->c_dma.ebus_info, 1)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to enable EBUS capture IRQ\n", dev); + snd_printdd("cs4231-%d: Unable to enable EBUS capture IRQ\n", + dev); return -EBUSY; } if (ebus_dma_register(&chip->p_dma.ebus_info)) { snd_cs4231_ebus_free(chip); - snd_printdd("cs4231-%d: Unable to register EBUS play DMA\n", dev); + snd_printdd("cs4231-%d: Unable to register EBUS play DMA\n", + dev); return -EBUSY; } if (ebus_dma_irq_enable(&chip->p_dma.ebus_info, 1)) {

2 1

[alsa-devel] How can I make sure the period size is a multiple of 16?
by Timur Tabi 06 Sep '07

06 Sep '07

To work around a hardware issue, my ASoC driver requires the period size to be a multiple of 16. What is the best way to ensure that? -- Timur Tabi Linux Kernel Developer @ Freescale

5 6

[alsa-devel] How to name controls
by Trent Piepho 06 Sep '07

06 Sep '07

I'm trying to improve the audio controls for the cx88-alsa driver, but I'm not sure what names I'm supposed to use. Currently, the driver has a single control called "Capture Volume", but it doesn't adjust the volume of the captured audio, so I don't think this name is correct. The audio routing looks something like this. Keep in mind that this is a TV capture chip and the drviers only support audio capture, not playback. Analog In Source Mute VV V || | || o-----o | || | | ---- \`=====| ADC |----| |---+------------------------> Audio DMA | | ---- | o-----o | ----- Digital Volume________\ / \ / V o-----o | ---- | | \----| |---| DAC |======> Analog Out ---- | | DAC Mute-----------------------^ o-----o Basically, there is an analog input (from the TV tuner), and a line out. The analog input goes through ADC -> DAC steps and then to the line out. There are three controls, one volume and two mute switches. One switch causes the source to be muted, the other only mutes the line out. The volume only controls the line out. So what should I call these controls? Do I call the volume "Playback Volume"? The chip doesn't support playback, so that seems confusing. It's more like "Pass-through Volume", but that's not an option. And what do I call the source mute switch? "Capture Switch"? That doesn't seem quite right either, since it doesn't just mute capture, but also mutes the pass-through audio.

2 2

[alsa-devel] snd_pcm_drain and lockups
by Arve Knudsen 06 Sep '07

06 Sep '07

Hello Users of the PortAudio library on top of ALSA-lib have reported that it may hang in a call to snd_pcm_drain. I have also been able to confirm that this can happen in conjunction with the dmix device at least. Can the ALSA developers please tell me how, if at all, snd_pcm_drain may be safely used? Is this function simply best avoided (snd_pcm_drop being the alternative)? Thanks, Arve Knudsen

2 4

[alsa-devel] Default probe_mask option
by Takashi Iwai 06 Sep '07

06 Sep '07

Hi, it's a known fact that snd-hda-intel driver doesn't work as is on many devices. It's often solved by adding a proper probe_mask option (usually because of non-working modem codec). Also, in some machines, position_fix is necessary. Now, as of ALSA 1.0.15-rc1, the driver has a built-in table for setting the default probe_mask and/or position_fix for known "buggy" devices. I'd like to ask you give feedback for this. If you have a machine that works ONLY with a probe_mask or a position_fix option, let me know. Give the output of "lspci -nv" (only for the audio device, which has class 0403) together with the product name. Then I'll add the entry for your device so that you won't need extra option on 1.0.15-final release. Please make sure that you test the latest driver, at least, 1.0.15-rc1, at best the latest HG version. Don't give a report with 1.0.14 or older versions. It doesn't help. For convenience for people that don't use HG, the daily snapshot tarball can be downloaded from: ftp://ftp.suse.com/pub/projects/alsa/snapshot/driver/ Thanks in advance, Takashi

1 0

[alsa-devel] DMA buffer gets played only once
by Markus Franke 06 Sep '07

06 Sep '07

Dear Alsa Developers/Users, I am working on a SoC Alsa driver for the WM8973. Currently I am facing the problem that after playing once the whole preallocated DMA buffer the playback stops and Alsa returns with ---snip--- ALSA sound/core/pcm_native.c:1526: playback drain error (DMA or IRQ trouble?) ---snap--- As far as I understood the callflow is like this: pcm_prepare() --> pcm_trigger(TRIGGER_START) --> snd_pcm_period_elapsed() from interrupt handler or DMA callback --> pcm_pointer() --> pcm_trigger(TRIGGER_STOP) --> pcm_prepare() --> ..... With each call of pcm_trigger(TRIGGER_START) I am transfering one period of sounddata. For some reason the playback stops once the DMA buffer (in my 8 example it holds 8 periods of data) is finished and the pointer should wrap around to the beginning through pcm_pointer() callback. I checked return values of pcm_pointer() and tried to trace it back through Alsa/SoC stack without success. It seems like that after the 8th period is transferred pcm_trigger(TRIGGER_STOP) isn't getting called. Instead of this I get a lot of calls to pcm_pointer() which result finally in the above mentioned Alsa error message. I checked pcm_pointer() return values but everything seems to be fine I think. Below you can find some verbose printk's I added to the source code: ---snip--- vi1888-i2s: vi1888_i2s_startup()--> vi1888-pcm: vi1888_pcm_open()--> vi1888-i2s: vi1888_i2s_set_dai_fmt()--> vi1888-i2s: vi1888_i2s_set_dai_sysclk()--> vi1888-i2s: vi1888_i2s_hw_params()--> vi1888-pcm: vi1888_pcm_hw_params()--> hw_params: DMA for I2S PCM Stereo out initialized (dma_bytes=32768, period_size=4096) vi1888-i2s: vi1888_i2s_set_dai_fmt()--> vi1888-i2s: vi1888_i2s_set_dai_sysclk()--> vi1888-pcm: vi1888_pcm_hw_params()--> hw_params: DMA for I2S PCM Stereo out initialized (dma_bytes=32768, period_size=4096) vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 1024 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 2048 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 3072 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 4096 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 5120 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 6144 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 7168 vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_prepare()--> vi1888-i2s: vi1888_i2s_prepare()--> vi1888-pcm: vi1888_pcm_trigger()--> vi1888-pcm: vi1888_pcm_dma_userCallback()--> vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 0 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 1024 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 2048 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 3072 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 4096 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 5120 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 6144 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 7168 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 0 vi1888-pcm: vi1888_pcm_pointer()--> pcm_pointer returns 1024 . . . . . . . ALSA sound/core/pcm_native.c:1526: playback drain error (DMA or IRQ trouble?) ---snap--- I get this output when trying to play a 16-bit stereo WAV file. Thanks for any help on this, Regards, Markus Franke

4 9

[alsa-devel] [PATCH] ad1848_lib: replace common delay loop by function
by Krzysztof Helt 06 Sep '07

06 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces a common delay loop by a function. It also uses ARRAY_SIZE macro for the rates table. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- It has the waiting function renamed to snd_ad1848_wait(). diff -urp linux-2.6.23.orig/sound/isa/ad1848/ad1848_lib.c linux-2.6.23/sound/isa/ad1848/ad1848_lib.c --- linux-2.6.23.orig/sound/isa/ad1848/ad1848_lib.c 2007-09-02 12:49:15.281291412 +0200 +++ linux-2.6.23/sound/isa/ad1848/ad1848_lib.c 2007-09-05 18:49:58.510592223 +0200 @@ -70,7 +70,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; @@ -99,24 +99,32 @@ static unsigned char snd_ad1848_original * Basic I/O functions */ -void snd_ad1848_out(struct snd_ad1848 *chip, - unsigned char reg, - unsigned char value) +static void snd_ad1848_wait(struct snd_ad1848 *chip) { int timeout; - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) + for (timeout = 250; timeout > 0; timeout--) { + if ((inb(AD1848P(chip, REGSEL)) & AD1848_INIT) == 0) + break; udelay(100); + } +} + +void snd_ad1848_out(struct snd_ad1848 *chip, + unsigned char reg, + unsigned char value) +{ + snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printk(KERN_WARNING "auto calibration time out - " + "reg = 0x%x, value = 0x%x\n", reg, value); #endif outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); outb(chip->image[reg] = value, AD1848P(chip, REG)); mb(); -#if 0 - printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); -#endif + snd_printdd("codec out - reg 0x%x = 0x%x\n", + chip->mce_bit | reg, value); } EXPORT_SYMBOL(snd_ad1848_out); @@ -124,10 +132,7 @@ EXPORT_SYMBOL(snd_ad1848_out); static void snd_ad1848_dout(struct snd_ad1848 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_wait(chip); outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); outb(value, AD1848P(chip, REG)); mb(); @@ -135,13 +140,11 @@ static void snd_ad1848_dout(struct snd_a static unsigned char snd_ad1848_in(struct snd_ad1848 *chip, unsigned char reg) { - int timeout; - - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x\n", reg); + snd_printk(KERN_WARNING "auto calibration time out - " + "reg = 0x%x\n", reg); #endif outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); mb(); @@ -183,8 +186,7 @@ static void snd_ad1848_mce_up(struct snd unsigned long flags; int timeout; - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_wait(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); @@ -319,11 +321,11 @@ static unsigned char snd_ad1848_get_rate { int i; - for (i = 0; i < 14; i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (rate == rates[i]) return freq_bits[i]; snd_BUG(); - return freq_bits[13]; + return freq_bits[ARRAY_SIZE(rates) - 1]; } static int snd_ad1848_ioctl(struct snd_pcm_substream *substream,

3 2

[alsa-devel] (no subject)
by Krzysztof Helt 06 Sep '07

06 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The linux/of.h header should be used instead of asm/prom.h. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Stephen Rothwell's post to the LKLM pointed that asm/prom.h is replaced by linux/of*.h headers from the new OpenFirmware framework . In case this post got malformed, the patch is attached too. --- old/sound/sparc/dbri.c 2007-09-04 18:38:20.000000000 +0200 +++ new/sound/sparc/dbri.c 2007-09-06 07:24:46.188427350 +0200 @@ -66,7 +66,7 @@ #include <sound/control.h> #include <sound/initval.h> -#include <asm/prom.h> +#include <linux/of.h> #include <asm/sbus.h> #include <asm/atomic.h> ---------------------------------------------------- Podróż w przeszłość - tysiące atrakcji w całej Polsce, tylko we wrześniu - Zajrzyj na stronę Edd.com.pl - Kliknij: http://klik.wp.pl/?adr=www.edd.com.pl&sid=9

2 1

[alsa-devel] [PATCH] Gallant SC-6000 driver (4th version)
by Krzysztof Helt 06 Sep '07

06 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is port of the Gallant SC-6000 driver from the OSS aedsp16 driver. This card was also sold as AudioExcel DSP 16 and Zoltrix AV302 (Audio Plus True 16). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This version fixes most of issues pointed by Rene Herman. Thank Clemens, Takashi and Rene who contributed to this work. I have left the ISA autodetection. It makes driver usage simpler for user. The base port, mpu port and mss port can be read from jumper settings, but irq or dma is not settable. A system may figure out which one is free and use it. I post this driver despite my question if it can be integrated into the sgalaxy driver. I think that I can post it and see if it work for people and maybe contact someone with SC-6600 model so the driver can be extended to handle this chip also. It would replace the OSS driver completely then. I am getting feeling that ad1848 and cs4231 libraries could be merged (or at least produce some common part like mss_lib). This may allow to merge drivers using cs4231-lib and ad1848-lib. I'll look into this (I do not have the CS4231 based card but I should have it next week - Sound Galaxy Waverider). Regards, Krzysztof diff -urpN linux-2.6.22.old/sound/isa/Kconfig linux-2.6.22/sound/isa/Kconfig --- linux-2.6.22.old/sound/isa/Kconfig 2007-08-28 23:14:08.000000000 +0200 +++ linux-2.6.22/sound/isa/Kconfig 2007-09-01 18:38:41.000000000 +0200 @@ -191,6 +191,19 @@ config SND_ES18XX To compile this driver as a module, choose M here: the module will be called snd-es18xx. +config SND_SC6000 + tristate "Gallant SC-6000, Audio Excel DSP 16" + depends on SND + select SND_AD1848_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for Gallant SC-6000 card and clones: + Audio Excel DSP 16 and Zoltrix AV302. + + To compile this driver as a module, choose M here: the module + will be called snd-sc6000. + config SND_GUS_SYNTH tristate diff -urpN linux-2.6.22.old/sound/isa/Makefile linux-2.6.22/sound/isa/Makefile --- linux-2.6.22.old/sound/isa/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22/sound/isa/Makefile 2007-09-03 22:09:58.000000000 +0200 @@ -10,6 +10,7 @@ snd-cmi8330-objs := cmi8330.o snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o +snd-sc6000-objs := sc6000.o snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330 obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o +obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o diff -urpN linux-2.6.22.old/sound/isa/sc6000.c linux-2.6.22/sound/isa/sc6000.c --- linux-2.6.22.old/sound/isa/sc6000.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22/sound/isa/sc6000.c 2007-09-05 21:14:45.000000000 +0200 @@ -0,0 +1,630 @@ +/* + * Driver for Gallant SC-6000 soundcard. This card is also known as + * Audio Excel DSP 16 or Zoltrix AV302. + * These cards use CompuMedia ASC-9308 chip + AD1848 codec. + * + * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1(a)wp.pl> + * + * I don't have documentation for this card. I used the driver + * for OSS/Free included in the kernel source as reference. + * + * 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 + */ + +#include <sound/driver.h> +#include <linux/module.h> +#include <linux/isa.h> +#include <asm/dma.h> +#include <sound/core.h> +#include <sound/ad1848.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#include <sound/control.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Krzysztof Helt"); +MODULE_DESCRIPTION("Gallant SC-6000"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000}," + "{AudioExcel, Audio Excel DSP 16}," + "{Zoltrix, AV302}}"); + +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; /* Enable this card */ +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ +static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + /* 0x300, 0x310, 0x320, 0x330 */ +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); +module_param_array(mss_port, long, NULL, 0444); +MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); +module_param_array(dma, int, NULL, 0444); +MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); + +/* + * Commands of SC6000's DSP (SBPRO+special). + * Some of them are COMMAND_xx, in the future they may change. + */ +#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ +#define COMMAND_52 0x52 /* */ +#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ +#define COMMAND_5C 0x5c /* */ +#define COMMAND_60 0x60 /* */ +#define COMMAND_66 0x66 /* */ +#define COMMAND_6C 0x6c /* */ +#define COMMAND_6E 0x6e /* */ +#define COMMAND_88 0x88 /* Unknown command */ +#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ +#define COMMAND_C5 0xc5 /* */ +#define GET_DSP_VERSION 0xe1 /* Get DSP Version */ +#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ + +/* + * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port + * to have the actual I/O port. + * Register permissions are: + * (wo) == Write Only + * (ro) == Read Only + * (w-) == Write + * (r-) == Read + */ +#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ +#define DSP_READ 0x0a /* offset of DSP READ (ro) */ +#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ +#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ +#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ +#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ + +#define PFX "sc6000: " + +/* hardware dependent functions */ + +/* + * sc6000_irq_to_softcfg - Decode irq number into cfg code. + */ +static __devinit unsigned char sc6000_irq_to_softcfg(int irq) +{ + unsigned char val = 0; + + switch (irq) { + case 5: + val = 0x28; + break; + case 7: + val = 0x8; + break; + case 9: + val = 0x10; + break; + case 10: + val = 0x18; + break; + case 11: + val = 0x20; + break; + default: + break; + } + return val; +} + +/* + * sc6000_dma_to_softcfg - Decode dma number into cfg code. + */ +static __devinit unsigned char sc6000_dma_to_softcfg(int dma) +{ + unsigned char val = 0; + + switch (dma) { + case 0: + val = 1; + break; + case 1: + val = 2; + break; + case 3: + val = 3; + break; + default: + break; + } + return val; +} + +/* + * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code. + */ +static __devinit unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) +{ + unsigned char val = 0; + + switch (mpu_irq) { + case 5: + val = 4; + break; + case 7: + val = 0x44; + break; + case 9: + val = 0x84; + break; + case 10: + val = 0xc4; + break; + default: + break; + } + return val; +} + +static __devinit int sc6000_wait_data(int port) +{ + int loop = 1000; + unsigned char val = 0; + + do { + val = inb(port + DSP_DATAVAIL); + if (val & 0x80) + return 0; + cpu_relax(); + } while (loop--); + + return -EAGAIN; +} + +static __devinit int sc6000_read(int port) +{ + if (sc6000_wait_data(port)) + return -EBUSY; + + return inb(port + DSP_READ); + +} + +static __devinit int sc6000_write(int port, int cmd) +{ + unsigned char val; + int loop = 500000; + + do { + val = inb(port + DSP_STATUS); + /* + * DSP ready to receive data if bit 7 of val == 0 + */ + if (!(val & 0x80)) { + outb(cmd, port + DSP_COMMAND); + return 0; + } + cpu_relax(); + } while (loop--); + + snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd); + + return -EAGAIN; +} + +static int __devinit sc6000_dsp_get_answer(int port, int command, char *data, + int data_len) +{ + int len = 0; + + if (sc6000_write(port, command)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command); + return -EFAULT; + } + + do { + int val = sc6000_read(port); + + if (val < 0) + break; + + data[len++] = val; + + } while (len < data_len); + + /* + * If no more data available, return to the caller, no error if len>0. + * We have no other way to know when the string is finished. + */ + return len ? len : -EIO; +} + +static int __devinit sc6000_dsp_reset(int port) +{ + outb(1, port + DSP_RESET); + udelay(10); + outb(0, port + DSP_RESET); + udelay(20); + if (sc6000_read(port) == 0xaa) + return 0; + return -EBUSY; +} + +/* detection and initialization */ +static int __devinit sc6000_cfg_write(int port, unsigned char softcfg) +{ + + if (sc6000_write(port, WRITE_MDIRQ_CFG)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return -EIO; + } + if (sc6000_write(port, softcfg)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EIO; + } + return 0; +} + +static int __devinit sc6000_setup_board(int port, int config) +{ + int loop = 10; + + do { + if (sc6000_write(port, COMMAND_88)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", + COMMAND_88); + return -EIO; + } + } while ((sc6000_wait_data(port) < 0) && loop--); + + if (sc6000_read(port) < 0) { + snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", + COMMAND_88); + return -EIO; + } + + if (sc6000_cfg_write(port, config)) + return -EFAULT; + + return 0; +} + +static int __devinit sc6000_init_mss(int port, int config, int mss_port, + int mss_config) +{ + if (sc6000_write(port, DSP_INIT_MSS)) { + snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n", + DSP_INIT_MSS); + return -EIO; + } + + msleep(10); + + if (sc6000_cfg_write(port, config)) + return -EIO; + + outb(mss_config, mss_port); + + return 0; +} + +static int __devinit sc6000_init_board(int port, int irq, int dma, + int mss_port, int mpu_irq) +{ + char answer[15]; + char version[2]; + int mss_config = sc6000_irq_to_softcfg(irq) | + sc6000_dma_to_softcfg(dma); + int config = mss_config | + sc6000_mpu_irq_to_softcfg(mpu_irq); + + if (sc6000_dsp_reset(port)) { + snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n"); + return -EBUSY; + } + + if (sc6000_dsp_get_answer(port, GET_DSP_COPYRIGHT, answer, 15) <= 0) { + snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n"); + return -ENODEV; + } + /* + * My SC-6000 card return "SC-6000" in DSPCopyright, so + * if we have something different, we have to be warned. + * Mine returns SC-6000A - KH + */ + if (strncmp("SC-6000", answer, 7)) + snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); + + if (sc6000_dsp_get_answer(port, GET_DSP_VERSION, version, 2) < 2) { + snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n"); + return -ENODEV; + } + printk(KERN_INFO PFX "Detected model: %s, DSP version %d.%d\n", + answer, version[0], version[1]); + + /* + * 0x0A == (IRQ 7, DMA 1, MIRQ 0) + */ + if (sc6000_cfg_write(port, 0x0a)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EFAULT; + } + + if (sc6000_setup_board(port, config) < 0) { + snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + if (sc6000_init_mss(port, config, mss_port, + mss_config) < 0) { + snd_printk(KERN_ERR "Can not initialize" + "Microsoft Sound System mode.\n"); + return -ENODEV; + } + + return 0; +} + +static int __devinit snd_sc6000_detect(int dev, int irq, int dma) +{ + int err = -ENODEV; + + snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", + port[dev], irq, dma, + mpu_irq[dev] == SND_AUTO_IRQ ? 0 : mpu_irq[dev]); + +/* + * We must request the port region because these are the I/O + * ports to access card's control registers. + */ + if (!request_region(port[dev], 0x10, "sc-6000 (base)")) { + snd_printk(KERN_ERR + "SC-6000 port I/O port region is already in use.\n"); + return -EBUSY; + } + + err = sc6000_init_board(port[dev], irq, dma, + mss_port[dev], mpu_irq[dev]); + + return err; +} + +static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +{ + struct snd_card *card = chip->card; + struct snd_ctl_elem_id id1, id2; + int err; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 to FM */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "FM Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + /* reassign AUX1 to CD */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + return 0; +} + +static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify IO port\n"); + return 0; + } + if (mss_port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify MSS port\n"); + return 0; + } + if (port[dev] != 0x220 && port[dev] != 0x240) { + printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n"); + return 0; + } + if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) { + printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n"); + return 0; + } + if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) { + printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]); + return 0; + } + if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) { + printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x30L) != 0x300) { + printk(KERN_ERR PFX "invalid MPU-401 port %lx\n", + mpu_port[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 && + !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) { + printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]); + return 0; + } + return 1; +} + +static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) +{ + static int possible_irqs[] = { 5, 7, 9, 10, 11, -1 }; + static int possible_dmas[] = { 1, 3, 0, -1 }; + int err; + int xirq = irq[dev]; + int xdma = dma[dev]; + struct snd_card *card; + struct snd_ad1848 *chip; + struct snd_opl3 *opl3; + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (!card) + return -ENOMEM; + + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto _err; + } + } + + if (xdma == SNDRV_AUTO_DMA) { + xdma = snd_legacy_find_free_dma(possible_dmas); + if (xdma < 0) { + snd_printk(KERN_ERR "unable to find a free DMA\n"); + err = -EBUSY; + goto _err; + } + } + + err = snd_sc6000_detect(dev, xirq, xdma); + if (err < 0) + goto _err; + + err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, + AD1848_HW_DETECT, &chip); + if (err < 0) + goto _err; + card->private_data = chip; + + err = snd_ad1848_pcm(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR + "error creating new ad1848 PCM device\n"); + goto _err; + } + err = snd_ad1848_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "error creating new ad1848 mixer\n"); + goto _err; + } + err = snd_sc6000_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "the mixer rewrite failed\n"); + goto _err; + } + if (snd_opl3_create(card, + 0x388, 0x388 + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR "no OPL device at 0x%x-0x%x ?\n", + 0x388, 0x388 + 2); + } else { + err = snd_opl3_timer_new(opl3, 0, 1); + if (err < 0) + goto _err; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto _err; + } + + if (mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", + mpu_port[dev]); + } + + strcpy(card->driver, "Gallant SC-6000"); + strcpy(card->shortname, "Gallant SC-6000"); + sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d", + mss_port[dev], xirq, xdma); + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto _err; + + dev_set_drvdata(devptr, card); + return 0; + + _err: + snd_card_free(card); + return err; +} + +static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) +{ + release_region(port[dev], 0x10); + + snd_card_free(dev_get_drvdata(devptr)); + dev_set_drvdata(devptr, NULL); + return 0; +} + +static struct isa_driver snd_sc6000_driver = { + .match = snd_sc6000_match, + .probe = snd_sc6000_probe, + .remove = __devexit_p(snd_sc6000_remove), + /* FIXME: suspend/resume */ + .driver = { + .name = "sc6000", + }, +}; + + +static int __init alsa_card_sc6000_init(void) +{ + return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_sc6000_exit(void) +{ + isa_unregister_driver(&snd_sc6000_driver); +} + +module_init(alsa_card_sc6000_init) +module_exit(alsa_card_sc6000_exit)

2 1

[alsa-devel] Device assignment and hotplug
by Takashi Iwai 05 Sep '07

05 Sep '07

Hi, the recent Linux systems use the auto-probing at boot via udev, and this often screws up the order of device loading, which results in the inconsistent device appearance. A typical case is that you have a PCI onboard and a USB audio device. Sometimes the USB appears as the first device while sometimes (most times) PCI appears as the first. So far, for fixing this order, we use index module option. That is, add the following to module config: options snd-onboard index=0 options snd-usb-audio index=1 But this scheme has a problem. It doesn't specify the order of other possible devices. For example, what if I plug a PCMCIA sound card and reboot? Since no index option is given for that, PCMCIA device may be loaded at any unused slot. If it's loaded before other devices, it goes to slot 0. Now, the onboard driver is loaded, and found that the statically assigned slot 0 is already occupied. Ouch, the driver fails and the device gets unusable. So, obviously, the problem of index option is that it's specific to the driver, and it doesn't give the whole system-wide setting. For overcoming this, I'd like to propose the following new option to snd module. The "slots" option is used to pass the static assignment of known drivers. For example, in the case above, it'd be like the following: options snd slots=snd-onboard,snd-usb-audio And you can omit all index options if you want. When a new device is loaded, then the first two slots are avoided, and it's loaded to the third slot in the end. The experimental patch is below. As you see, the necessary change is minimum, and it doesn't even conflict with the existing index option, too. Any comments? The module name "slots" is OK? BTW, one missing feature is the support for built-in drivers. But I don't think it's needed because the built-in drivers are always loaded in prior to others, and thus you can give simply index options for each. thanks, Takashi diff -r 5debd8716290 core/init.c --- a/core/init.c Wed Sep 05 15:08:23 2007 +0200 +++ b/core/init.c Wed Sep 05 19:00:08 2007 +0200 @@ -42,6 +42,40 @@ EXPORT_SYMBOL(snd_cards); EXPORT_SYMBOL(snd_cards); static DEFINE_MUTEX(snd_card_mutex); + +static char *slots[SNDRV_CARDS]; +module_param_array(slots, charp, NULL, 0444); +MODULE_PARM_DESC(slots, "Module names assigned to the slots."); + +/* return non-zero if the given index is already reserved for another + * module via slots option + */ +static int module_slot_mismatch(struct module *module, int idx) +{ +#ifdef MODULE + char *s1, *s2; + if (!module || !module->name || !slots[idx]) + return 0; + s1 = slots[idx]; + s2 = module->name; + /* compare module name strings + * hyphens are handled as equivalent with underscore + */ + for (;;) { + char c1 = *s1++; + char c2 = *s2++; + if (c1 == '-') + c1 = '_'; + if (c2 == '-') + c2 = '_'; + if (c1 != c2) + return 1; + if (!c1) + break; + } +#endif + return 0; +} #if defined(CONFIG_SND_MIXER_OSS) || defined(CONFIG_SND_MIXER_OSS_MODULE) int (*snd_mixer_oss_notify_callback)(struct snd_card *card, int free_flag); @@ -115,6 +149,8 @@ struct snd_card *snd_card_new(int idx, c for (idx2 = 0; idx2 < SNDRV_CARDS; idx2++) /* idx == -1 == 0xffff means: take any free slot */ if (~snd_cards_lock & idx & 1<<idx2) { + if (module_slot_mismatch(module, idx2)) + continue; idx = idx2; if (idx >= snd_ecards_limit) snd_ecards_limit = idx + 1;

2 2

[alsa-devel] [PATCH] cs4231-lib: replace common delay loop by function
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces a common delay loop by a function. It also uses ARRAY_SIZE macro for the rates table. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-2.6.23.orig/sound/isa/cs423x/cs4231_lib.c linux-2.6.23/sound/isa/cs423x/cs4231_lib.c --- linux-2.6.23.orig/sound/isa/cs423x/cs4231_lib.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-09-05 19:07:35.242811960 +0200 @@ -74,7 +74,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; @@ -134,29 +134,31 @@ static inline u8 cs4231_inb(struct snd_c return inb(chip->port + offset); } -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, - unsigned char mask, unsigned char value) +static void snd_cs4231_wait(struct snd_cs4231 *chip) { int timeout; - unsigned char tmp; for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(100); +} + +static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, + unsigned char mask, unsigned char value) +{ + unsigned char tmp = (chip->image[reg] & mask) | value; + + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif - if (chip->calibrate_mute) { - chip->image[reg] &= mask; - chip->image[reg] |= value; - } else { + chip->image[reg] = tmp; + if (!chip->calibrate_mute) { cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - mb(); - tmp = (chip->image[reg] & mask) | value; + wmb(); cs4231_outb(chip, CS4231P(REG), tmp); - chip->image[reg] = tmp; mb(); } } @@ -176,12 +178,7 @@ static void snd_cs4231_dout(struct snd_c void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); @@ -190,19 +187,13 @@ void snd_cs4231_out(struct snd_cs4231 *c cs4231_outb(chip, CS4231P(REG), value); chip->image[reg] = value; mb(); -#if 0 - printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); -#endif + snd_printdd("codec out - reg 0x%x = 0x%x\n", + chip->mce_bit | reg, value); } unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); @@ -304,8 +295,7 @@ void snd_cs4231_mce_up(struct snd_cs4231 unsigned long flags; int timeout; - for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) - udelay(100); + snd_cs4231_wait(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_up - auto calibration time out (0)\n"); @@ -459,11 +449,11 @@ static unsigned char snd_cs4231_get_rate { int i; - for (i = 0; i < 14; i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (rate == rates[i]) return freq_bits[i]; // snd_BUG(); - return freq_bits[13]; + return freq_bits[ARRAY_SIZE(rates) - 1]; } static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip,

2 1

[alsa-devel] [programmer needed] ESI MAYA44 driver - easy to write
by Piotr Makowski 05 Sep '07

05 Sep '07

ESI are prepared to supply the hardware and make engineers available for the engineer who would like to write ESI MAYA44 driver for linux. It shouldn't be difficult because there is already ice-1724 driver written which will probably need only some modifications to hande the ESI MAYA44 card. Regards, Piotr Makowski

1 0

[alsa-devel] [PATCH] ad1848_lib: replace common delay loop by function
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces a common delay loop by a function. It also uses ARRAY_SIZE macro for the rates table. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- It is the same as for the cs4231_lib.c file. Please do not bounce this patch if you spot something wrong with formatting in the context lines. I'll make a separate patch with checkpatch fixes for this library. diff -urp linux-2.6.23.orig/sound/isa/ad1848/ad1848_lib.c linux-2.6.23/sound/isa/ad1848/ad1848_lib.c --- linux-2.6.23.orig/sound/isa/ad1848/ad1848_lib.c 2007-09-02 12:49:15.281291412 +0200 +++ linux-2.6.23/sound/isa/ad1848/ad1848_lib.c 2007-09-04 23:44:55.684785358 +0200 @@ -70,7 +70,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; @@ -99,24 +99,30 @@ static unsigned char snd_ad1848_original * Basic I/O functions */ -void snd_ad1848_out(struct snd_ad1848 *chip, - unsigned char reg, - unsigned char value) +static void snd_ad1848_ready(struct snd_ad1848 *chip) { int timeout; - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) + for (timeout = 250; timeout > 0; timeout--) { + if ((inb(AD1848P(chip, REGSEL)) & AD1848_INIT) == 0) + break; udelay(100); + } +} + +void snd_ad1848_out(struct snd_ad1848 *chip, + unsigned char reg, + unsigned char value) +{ + snd_ad1848_ready(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printk(KERN_WARNING "auto calibration time out - " + "reg = 0x%x, value = 0x%x\n", reg, value); #endif outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); outb(chip->image[reg] = value, AD1848P(chip, REG)); mb(); -#if 0 - printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); -#endif } EXPORT_SYMBOL(snd_ad1848_out); @@ -124,10 +130,7 @@ EXPORT_SYMBOL(snd_ad1848_out); static void snd_ad1848_dout(struct snd_ad1848 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_ready(chip); outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); outb(value, AD1848P(chip, REG)); mb(); @@ -135,13 +138,11 @@ static void snd_ad1848_dout(struct snd_a static unsigned char snd_ad1848_in(struct snd_ad1848 *chip, unsigned char reg) { - int timeout; - - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_ready(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) - snd_printk(KERN_WARNING "auto calibration time out - reg = 0x%x\n", reg); + snd_printk(KERN_WARNING "auto calibration time out - " + "reg = 0x%x\n", reg); #endif outb(chip->mce_bit | reg, AD1848P(chip, REGSEL)); mb(); @@ -183,8 +184,7 @@ static void snd_ad1848_mce_up(struct snd unsigned long flags; int timeout; - for (timeout = 250; timeout > 0 && (inb(AD1848P(chip, REGSEL)) & AD1848_INIT); timeout--) - udelay(100); + snd_ad1848_ready(chip); #ifdef CONFIG_SND_DEBUG if (inb(AD1848P(chip, REGSEL)) & AD1848_INIT) snd_printk(KERN_WARNING "mce_up - auto calibration time out (0)\n"); @@ -319,11 +319,11 @@ static unsigned char snd_ad1848_get_rate { int i; - for (i = 0; i < 14; i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (rate == rates[i]) return freq_bits[i]; snd_BUG(); - return freq_bits[13]; + return freq_bits[ARRAY_SIZE(rates) - 1]; } static int snd_ad1848_ioctl(struct snd_pcm_substream *substream,

3 4

Re: [alsa-devel] HDA Conexant: is mic boost possible?
by Pablo N. Hess 05 Sep '07

05 Sep '07

> I believe mic boost is on by default. What model are you running this > as? Laptop or Fujitsu? > > Tobin I'm running it as model 'laptop'. Any other hint at what might be causing this different behavior depending on the mic being stereo or mono, then? In case someone needs it, I'll write again the quick description of the problem: > alsa-driver-1.0.14: > playback: ok > recording: > stereo mic: ok > mono mic: no input at all (very high volume already tried) > > alsa-driver-1.0.15rc1: > playback: ok (even better than with alsa-driver-1.0.14) > recording: > stereo mic: ok (seems better than with alsa-driver-1.0.14) > mono mic: *very* very low volume, bad quality, lots of low-volume > noise, too. Thanks a lot, Pablo

1 0

[alsa-devel] [PATCH] sgalaxy: checkpatch fixes
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes white spaces and errors pointed by the checkpatch.pl script. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- A nice side effect: sgalaxy.c gets shorter. diff -urp linux-2.6.23.orig/sound/isa/sgalaxy.c linux-2.6.23/sound/isa/sgalaxy.c --- linux-2.6.23.orig/sound/isa/sgalaxy.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.23/sound/isa/sgalaxy.c 2007-09-04 23:41:11.860030310 +0200 @@ -47,7 +47,8 @@ static int index[SNDRV_CARDS] = SNDRV_DE static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; /* ID for this card */ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; /* Enable this card */ static long sbport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220,0x240 */ -static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530,0xe80,0xf40,0x604 */ +static long wssport[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + /* 0x530,0xe80,0xf40,0x604 */ static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 7,9,10,11 */ static int dma1[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0,1,3 */ @@ -69,14 +70,10 @@ MODULE_PARM_DESC(dma1, "DMA1 # for Sound #define PFX "sgalaxy: " -/* +#define AD1848P1(port, x) (port + c_d_c_AD1848##x) - */ - -#define AD1848P1( port, x ) ( port + c_d_c_AD1848##x ) - -/* from lowlevel/sb/sb.c - to avoid having to allocate a struct snd_sb for the */ -/* short time we actually need it.. */ +/* from lowlevel/sb/sb.c - to avoid having to allocate a struct snd_sb */ +/* for the short time we actually need it.. */ static int snd_sgalaxy_sbdsp_reset(unsigned long port) { @@ -98,7 +95,7 @@ static int __devinit snd_sgalaxy_sbdsp_c unsigned char val) { int i; - + for (i = 10000; i; i--) if ((inb(SBP1(port, STATUS)) & 0x80) == 0) { outb(val, SBP1(port, COMMAND)); @@ -115,45 +112,39 @@ static irqreturn_t snd_sgalaxy_dummy_int static int __devinit snd_sgalaxy_setup_wss(unsigned long port, int irq, int dma) { - static int interrupt_bits[] = {-1, -1, -1, -1, -1, -1, -1, 0x08, -1, + static int interrupt_bits[] = {-1, -1, -1, -1, -1, -1, -1, 0x08, -1, 0x10, 0x18, 0x20, -1, -1, -1, -1}; static int dma_bits[] = {1, 2, 0, 3}; int tmp, tmp1; - if ((tmp = inb(port + 3)) == 0xff) - { + tmp = inb(port + 3); + if (tmp == 0xff) { snd_printdd("I/O address dead (0x%lx)\n", port); return 0; } -#if 0 - snd_printdd("WSS signature = 0x%x\n", tmp); -#endif - if ((tmp & 0x3f) != 0x04 && - (tmp & 0x3f) != 0x0f && - (tmp & 0x3f) != 0x00) { + tmp &= 0x3f; + if (tmp != 0x04 && tmp != 0x0f && tmp != 0x00) { snd_printdd("No WSS signature detected on port 0x%lx\n", port + 3); return 0; } -#if 0 - snd_printdd(PFX "setting up IRQ/DMA for WSS\n"); -#endif - - /* initialize IRQ for WSS codec */ - tmp = interrupt_bits[irq % 16]; - if (tmp < 0) - return -EINVAL; - - if (request_irq(irq, snd_sgalaxy_dummy_interrupt, IRQF_DISABLED, "sgalaxy", NULL)) { + /* initialize IRQ for WSS codec */ + tmp = interrupt_bits[irq % 16]; + if (tmp < 0) + return -EINVAL; + + tmp = request_irq(irq, snd_sgalaxy_dummy_interrupt, IRQF_DISABLED, + "sgalaxy", NULL); + if (tmp) { snd_printk(KERN_ERR "sgalaxy: can't grab irq %d\n", irq); return -EIO; } - outb(tmp | 0x40, port); - tmp1 = dma_bits[dma % 4]; - outb(tmp | tmp1, port); + outb(tmp | 0x40, port); + tmp1 = dma_bits[dma % 4]; + outb(tmp | tmp1, port); free_irq(irq, NULL); @@ -162,10 +153,6 @@ static int __devinit snd_sgalaxy_setup_w static int __devinit snd_sgalaxy_detect(int dev, int irq, int dma) { -#if 0 - snd_printdd(PFX "switching to WSS mode\n"); -#endif - /* switch to WSS mode */ snd_sgalaxy_sbdsp_reset(sbport[dev]); @@ -177,8 +164,10 @@ static int __devinit snd_sgalaxy_detect( } static struct ad1848_mix_elem snd_sgalaxy_controls[] = { -AD1848_DOUBLE("Aux Playback Switch", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 7, 7, 1, 1), -AD1848_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, 0, 0, 31, 0) +AD1848_DOUBLE("Aux Playback Switch", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, + 7, 7, 1, 1), +AD1848_DOUBLE("Aux Playback Volume", 0, SGALAXY_AUXC_LEFT, SGALAXY_AUXC_RIGHT, + 0, 0, 31, 0) }; static int __devinit snd_sgalaxy_mixer(struct snd_ad1848 *chip) @@ -190,28 +179,34 @@ static int __devinit snd_sgalaxy_mixer(s memset(&id1, 0, sizeof(id1)); memset(&id2, 0, sizeof(id2)); - id1.iface = id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; /* reassign AUX0 to LINE */ strcpy(id1.name, "Aux Playback Switch"); strcpy(id2.name, "Line Playback Switch"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) return err; strcpy(id1.name, "Aux Playback Volume"); strcpy(id2.name, "Line Playback Volume"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) return err; /* reassign AUX1 to FM */ strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; strcpy(id2.name, "FM Playback Switch"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) return err; strcpy(id1.name, "Aux Playback Volume"); strcpy(id2.name, "FM Playback Volume"); - if ((err = snd_ctl_rename_id(card, &id1, &id2)) < 0) + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) return err; /* build AUX2 input */ for (idx = 0; idx < ARRAY_SIZE(snd_sgalaxy_controls); idx++) { - if ((err = snd_ad1848_add_ctl_elem(chip, &snd_sgalaxy_controls[idx])) < 0) + err = snd_ad1848_add_ctl_elem(chip, &snd_sgalaxy_controls[idx]); + if (err < 0) return err; } return 0; @@ -241,44 +236,50 @@ static int __devinit snd_sgalaxy_probe(s struct snd_ad1848 *chip; card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); - if (card == NULL) + if (!card) return -ENOMEM; xirq = irq[dev]; if (xirq == SNDRV_AUTO_IRQ) { - if ((xirq = snd_legacy_find_free_irq(possible_irqs)) < 0) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); err = -EBUSY; goto _err; } } xdma1 = dma1[dev]; - if (xdma1 == SNDRV_AUTO_DMA) { - if ((xdma1 = snd_legacy_find_free_dma(possible_dmas)) < 0) { + if (xdma1 == SNDRV_AUTO_DMA) { + xdma1 = snd_legacy_find_free_dma(possible_dmas); + if (xdma1 < 0) { snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); err = -EBUSY; goto _err; } } - if ((err = snd_sgalaxy_detect(dev, xirq, xdma1)) < 0) + err = snd_sgalaxy_detect(dev, xirq, xdma1); + if (err < 0) goto _err; - if ((err = snd_ad1848_create(card, wssport[dev] + 4, - xirq, xdma1, - AD1848_HW_DETECT, &chip)) < 0) + err = snd_ad1848_create(card, wssport[dev] + 4, xirq, xdma1, + AD1848_HW_DETECT, &chip); + if (err < 0) goto _err; card->private_data = chip; - if ((err = snd_ad1848_pcm(chip, 0, NULL)) < 0) { + err = snd_ad1848_pcm(chip, 0, NULL); + if (err < 0) { snd_printdd(PFX "error creating new ad1848 PCM device\n"); goto _err; } - if ((err = snd_ad1848_mixer(chip)) < 0) { + err = snd_ad1848_mixer(chip); + if (err < 0) { snd_printdd(PFX "error creating new ad1848 mixer\n"); goto _err; } - if ((err = snd_sgalaxy_mixer(chip)) < 0) { + err = snd_sgalaxy_mixer(chip); + if (err < 0) { snd_printdd(PFX "the mixer rewrite failed\n"); goto _err; } @@ -290,7 +291,8 @@ static int __devinit snd_sgalaxy_probe(s snd_card_set_dev(card, devptr); - if ((err = snd_card_register(card)) < 0) + err = snd_card_register(card); + if (err < 0) goto _err; dev_set_drvdata(devptr, card); @@ -327,7 +329,8 @@ static int snd_sgalaxy_resume(struct dev chip->resume(chip); snd_ad1848_out(chip, SGALAXY_AUXC_LEFT, chip->image[SGALAXY_AUXC_LEFT]); - snd_ad1848_out(chip, SGALAXY_AUXC_RIGHT, chip->image[SGALAXY_AUXC_RIGHT]); + snd_ad1848_out(chip, SGALAXY_AUXC_RIGHT, + chip->image[SGALAXY_AUXC_RIGHT]); snd_power_change_state(card, SNDRV_CTL_POWER_D0); return 0;

3 6

[alsa-devel] [PATCH] sun-cs4231: use cs4231-regs.h
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces cs4231 registers definitions with common include. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-2.6.23.ref/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-2.6.23.ref/sound/sparc/cs4231.c 2007-09-04 19:04:30.000000000 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-04 19:10:13.000000000 +0200 @@ -130,129 +130,10 @@ static struct snd_cs4231 *cs4231_list; */ /* IO ports */ - -#define CS4231P(chip, x) ((chip)->port + c_d_c_CS4231##x) +#include <sound/cs4231-regs.h> /* XXX offsets are different than PC ISA chips... */ -#define c_d_c_CS4231REGSEL 0x0 -#define c_d_c_CS4231REG 0x4 -#define c_d_c_CS4231STATUS 0x8 -#define c_d_c_CS4231PIO 0xc - -/* codec registers */ - -#define CS4231_LEFT_INPUT 0x00 /* left input control */ -#define CS4231_RIGHT_INPUT 0x01 /* right input control */ -#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ -#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ -#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ -#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define CS4231_PIN_CTRL 0x0a /* pin control */ -#define CS4231_TEST_INIT 0x0b /* test and initialization */ -#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ -#define CS4231_LOOPBACK 0x0d /* loopback control */ -#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ -#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ -#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ -#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ -#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */ -#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */ -#define CS4231_TIMER_LOW 0x14 /* timer low byte */ -#define CS4231_TIMER_HIGH 0x15 /* timer high byte */ -#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */ -#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */ -#define CS4236_EXT_REG 0x17 /* extended register access */ -#define CS4231_IRQ_STATUS 0x18 /* irq status register */ -#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */ -#define CS4231_VERSION 0x19 /* CS4231(A) - version values */ -#define CS4231_MONO_CTRL 0x1a /* mono input/output control */ -#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */ -#define CS4235_LEFT_MASTER 0x1b /* left master output control */ -#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */ -#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */ -#define CS4235_RIGHT_MASTER 0x1d /* right master output control */ -#define CS4231_REC_UPR_CNT 0x1e /* record upper count */ -#define CS4231_REC_LWR_CNT 0x1f /* record lower count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define CS4231_INIT 0x80 /* CODEC is initializing */ -#define CS4231_MCE 0x40 /* mode change enable */ -#define CS4231_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for codec irq status - CS4231_IRQ_STATUS */ - -#define CS4231_PLAYBACK_IRQ 0x10 -#define CS4231_RECORD_IRQ 0x20 -#define CS4231_TIMER_IRQ 0x40 -#define CS4231_ALL_IRQS 0x70 -#define CS4231_REC_UNDERRUN 0x08 -#define CS4231_REC_OVERRUN 0x04 -#define CS4231_PLY_OVERRUN 0x02 -#define CS4231_PLY_UNDERRUN 0x01 - -/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ - -#define CS4231_ENABLE_MIC_GAIN 0x20 - -#define CS4231_MIXS_LINE 0x00 -#define CS4231_MIXS_AUX1 0x40 -#define CS4231_MIXS_MIC 0x80 -#define CS4231_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ - -#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ -#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ -#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ -#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ -#define CS4231_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define CS4231_XTAL1 0x00 /* 24.576 crystal */ -#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - CS4231_IFACE_CTRL */ - -#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ -#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ -#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ -#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ -#define CS4231_RECORD_ENABLE 0x02 /* record enable */ -#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - CS4231_PIN_CTRL */ - -#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ -#define CS4231_XCTL1 0x40 /* external control #1 */ -#define CS4231_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - CS4231_TEST_INIT */ - -#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ - -/* definitions for misc control register - CS4231_MISC_INFO */ - -#define CS4231_MODE2 0x40 /* MODE 2 */ -#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */ -#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */ - -/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */ - -#define CS4231_DACZ 0x01 /* zero DAC when underrun */ -#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */ -#define CS4231_OLB 0x80 /* output level bit */ +#define CS4231U(chip, x) ((chip)->port + ((c_d_c_CS4231##x) << 2)) /* SBUS DMA register defines. */ @@ -412,24 +293,26 @@ static void snd_cs4231_ready(struct snd_ { int timeout; - for (timeout = 250; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + for (timeout = 250; timeout > 0; timeout--) { + int val = __cs4231_readb(chip, CS4231U(chip, REGSEL)); + if ((val & CS4231_INIT) == 0) + break; + udelay(100); + } } static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) snd_printdd("out: auto calibration time out - reg = 0x%x, " "value = 0x%x\n", reg, value); #endif - __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); + __cs4231_writeb(chip, chip->mce_bit | reg, CS4231U(chip, REGSEL)); wmb(); - __cs4231_writeb(chip, value, CS4231P(chip, REG)); + __cs4231_writeb(chip, value, CS4231U(chip, REG)); mb(); } @@ -455,13 +338,13 @@ static unsigned char snd_cs4231_in(struc { snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) snd_printdd("in: auto calibration time out - reg = 0x%x\n", reg); #endif - __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); + __cs4231_writeb(chip, chip->mce_bit | reg, CS4231U(chip, REGSEL)); mb(); - return __cs4231_readb(chip, CS4231P(chip, REG)); + return __cs4231_readb(chip, CS4231U(chip, REG)); } /* @@ -474,13 +357,15 @@ static void snd_cs4231_busy_wait(struct /* huh.. looks like this sequence is proper for CS4231A chip (GUS MAX) */ for (timeout = 5; timeout > 0; timeout--) - __cs4231_readb(chip, CS4231P(chip, REGSEL)); + __cs4231_readb(chip, CS4231U(chip, REGSEL)); /* end of cleanup sequence */ - for (timeout = 500; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) + for (timeout = 500; timeout > 0; timeout--) { + int val = __cs4231_readb(chip, CS4231U(chip, REGSEL)); + if ((val & CS4231_INIT) == 0) + break; msleep(1); + } } static void snd_cs4231_mce_up(struct snd_cs4231 *chip) @@ -491,15 +376,16 @@ static void snd_cs4231_mce_up(struct snd spin_lock_irqsave(&chip->lock, flags); snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) snd_printdd("mce_up - auto calibration time out (0)\n"); #endif chip->mce_bit |= CS4231_MCE; - timeout = __cs4231_readb(chip, CS4231P(chip, REGSEL)); + timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); if (timeout == 0x80) snd_printdd("mce_up [%p]: serious init problem - codec still busy\n", chip->port); if (!(timeout & CS4231_MCE)) - __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), CS4231P(chip, REGSEL)); + __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), + CS4231U(chip, REGSEL)); spin_unlock_irqrestore(&chip->lock, flags); } @@ -511,12 +397,14 @@ static void snd_cs4231_mce_down(struct s spin_lock_irqsave(&chip->lock, flags); snd_cs4231_busy_wait(chip); #ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("mce_down [%p] - auto calibration time out (0)\n", CS4231P(chip, REGSEL)); + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) + snd_printdd("mce_down [%p] - auto calibration time out (0)\n", + CS4231U(chip, REGSEL)); #endif chip->mce_bit &= ~CS4231_MCE; - timeout = __cs4231_readb(chip, CS4231P(chip, REGSEL)); - __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), CS4231P(chip, REGSEL)); + timeout = __cs4231_readb(chip, CS4231U(chip, REGSEL)); + __cs4231_writeb(chip, chip->mce_bit | (timeout & 0x1f), + CS4231U(chip, REGSEL)); if (timeout == 0x80) snd_printdd("mce_down [%p]: serious init problem - codec still busy\n", chip->port); if ((timeout & CS4231_MCE) == 0) { @@ -549,7 +437,7 @@ static void snd_cs4231_mce_down(struct s /* in 10ms increments, check condition, up to 100ms */ timeout = 10; - while (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) { + while (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) { spin_unlock_irqrestore(&chip->lock, flags); if (--timeout < 0) { snd_printk("mce_down - auto calibration time out (3)\n"); @@ -906,8 +794,8 @@ static int snd_cs4231_open(struct snd_cs CS4231_RECORD_IRQ | CS4231_TIMER_IRQ); snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); /* clear IRQ */ - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); /* clear IRQ */ + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */ + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */ snd_cs4231_out(chip, CS4231_IRQ_STATUS, CS4231_PLAYBACK_IRQ | CS4231_RECORD_IRQ | @@ -936,8 +824,8 @@ static void snd_cs4231_close(struct snd_ /* disable IRQ */ spin_lock_irqsave(&chip->lock, flags); snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); /* clear IRQ */ - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); /* clear IRQ */ + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */ + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */ /* now disable record & playback */ @@ -958,8 +846,8 @@ static void snd_cs4231_close(struct snd_ /* clear IRQ again */ snd_cs4231_out(chip, CS4231_IRQ_STATUS, 0); - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); /* clear IRQ */ - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); /* clear IRQ */ + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */ + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); /* clear IRQ */ spin_unlock_irqrestore(&chip->lock, flags); snd_cs4231_calibrate_mute(chip, 0); @@ -1143,7 +1031,7 @@ static int __init snd_cs4231_probe(struc id = vers = 0; for (i = 0; i < 50; i++) { mb(); - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) + if (__cs4231_readb(chip, CS4231U(chip, REGSEL)) & CS4231_INIT) udelay(2000); else { spin_lock_irqsave(&chip->lock, flags); @@ -1161,8 +1049,9 @@ static int __init snd_cs4231_probe(struc spin_lock_irqsave(&chip->lock, flags); - __cs4231_readb(chip, CS4231P(chip, STATUS)); /* clear any pendings IRQ */ - __cs4231_writeb(chip, 0, CS4231P(chip, STATUS)); + /* clear any pendings IRQ */ + __cs4231_readb(chip, CS4231U(chip, STATUS)); + __cs4231_writeb(chip, 0, CS4231U(chip, STATUS)); mb(); spin_unlock_irqrestore(&chip->lock, flags); @@ -1712,7 +1601,7 @@ static irqreturn_t snd_cs4231_sbus_inter struct snd_cs4231 *chip = dev_id; /*This is IRQ is not raised by the cs4231*/ - if (!(__cs4231_readb(chip, CS4231P(chip, STATUS)) & CS4231_GLOBALIRQ)) + if (!(__cs4231_readb(chip, CS4231U(chip, STATUS)) & CS4231_GLOBALIRQ)) return IRQ_NONE; /* ACK the APC interrupt. */

2 1

[alsa-devel] [PATCH] sun-cs4231: memory management fix
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The chip structure is now allocated by snd_card_new() and it must not be released by separate kfree(). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This bug was introduced by the previous patch. --- linux-2.6.23.ref/sound/sparc/cs4231.c 2007-09-03 16:03:46.000000000 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-04 18:02:34.000000000 +0200 @@ -1865,8 +1865,6 @@ static int snd_cs4231_sbus_free(struct s if (chip->port) sbus_iounmap(chip->port, chip->regs_size); - kfree(chip); - return 0; } @@ -2037,8 +2035,6 @@ static int snd_cs4231_ebus_free(struct s if (chip->port) iounmap(chip->port); - kfree(chip); - return 0; }

2 1

[alsa-devel] [PATCH] dbri: conversion to OpenFirmware framework (2nd version)
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch converts the dbri driver to use OF framework. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch requires the previous dbri patch with more code improvements posted yesterday. diff -urp linux-2.6.23.orig/sound/sparc/dbri.c linux-2.6.23/sound/sparc/dbri.c --- linux-2.6.23.orig/sound/sparc/dbri.c 2007-09-04 18:34:14.000000000 +0200 +++ linux-2.6.23/sound/sparc/dbri.c 2007-09-04 18:38:20.638614486 +0200 @@ -66,6 +66,7 @@ #include <sound/control.h> #include <sound/initval.h> +#include <asm/prom.h> #include <asm/sbus.h> #include <asm/atomic.h> @@ -296,8 +297,6 @@ struct dbri_streaminfo { /* This structure holds the information for both chips (DBRI & CS4215) */ struct snd_dbri { - struct snd_card *card; /* ALSA card */ - int regs_size, irq; /* Needed for unload */ struct sbus_dev *sdev; /* SBUS device info */ spinlock_t lock; @@ -318,8 +317,6 @@ struct snd_dbri { struct cs4215 mm; /* mmcodec special info */ /* per stream (playback/record) info */ struct dbri_streaminfo stream_info[DBRI_NO_STREAMS]; - - struct snd_dbri *next; }; #define DBRI_MAX_VOLUME 63 /* Output volume */ @@ -571,8 +568,6 @@ struct snd_dbri { #define DBRI_STREAM(dbri, substream) \ &dbri->stream_info[DBRI_STREAMNO(substream)] -static struct snd_dbri *dbri_list; /* All DBRI devices */ - /* * Short data pipes transmit LSB first. The CS4215 receives MSB first. Grrr. * So we have to reverse the bits. Note: not all bit lengths are supported @@ -748,7 +743,7 @@ static void dbri_reset(struct snd_dbri * } /* Lock must not be held before calling this */ -static void __init dbri_initialize(struct snd_dbri *dbri) +static void __devinit dbri_initialize(struct snd_dbri *dbri) { s32 *cmd; u32 dma_addr; @@ -1308,7 +1303,7 @@ to the DBRI via the CHI interface and fe * Lock must not be held before calling it. */ -static __init void cs4215_setup_pipes(struct snd_dbri *dbri) +static __devinit void cs4215_setup_pipes(struct snd_dbri *dbri) { unsigned long flags; @@ -1341,7 +1336,7 @@ static __init void cs4215_setup_pipes(st dbri_cmdwait(dbri); } -static __init int cs4215_init_data(struct cs4215 *mm) +static __devinit int cs4215_init_data(struct cs4215 *mm) { /* * No action, memory resetting only. @@ -1633,7 +1628,7 @@ static int cs4215_prepare(struct snd_dbr /* * */ -static __init int cs4215_init(struct snd_dbri *dbri) +static __devinit int cs4215_init(struct snd_dbri *dbri) { u32 reg2 = sbus_readl(dbri->regs + REG2); dprintk(D_MM, "cs4215_init: reg2=0x%x\n", reg2); @@ -2218,12 +2213,12 @@ static struct snd_pcm_ops snd_dbri_ops = .pointer = snd_dbri_pointer, }; -static int __devinit snd_dbri_pcm(struct snd_dbri *dbri) +static int __devinit snd_dbri_pcm(struct snd_card *card) { struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(dbri->card, + if ((err = snd_pcm_new(card, /* ID */ "sun_dbri", /* device */ 0, /* playback count */ 1, @@ -2234,9 +2229,9 @@ static int __devinit snd_dbri_pcm(struct snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_dbri_ops); snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &snd_dbri_ops); - pcm->private_data = dbri; + pcm->private_data = card->private_data; pcm->info_flags = 0; - strcpy(pcm->name, dbri->card->shortname); + strcpy(pcm->name, card->shortname); if ((err = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, @@ -2422,14 +2417,14 @@ static struct snd_kcontrol_new dbri_cont CS4215_SINGLE("Mic boost", 4, 4, 1, 1) }; -static int __init snd_dbri_mixer(struct snd_dbri *dbri) +static int __devinit snd_dbri_mixer(struct snd_card *card) { - struct snd_card *card; int idx, err; + struct snd_dbri *dbri; - snd_assert(dbri != NULL && dbri->card != NULL, return -EINVAL); + snd_assert(card != NULL && card->private_data != NULL, return -EINVAL); + dbri = card->private_data; - card = dbri->card; strcpy(card->mixername, card->shortname); for (idx = 0; idx < ARRAY_SIZE(dbri_controls); idx++) { @@ -2485,15 +2480,16 @@ static void dbri_debug_read(struct snd_i } #endif -void snd_dbri_proc(struct snd_dbri *dbri) +void __devinit snd_dbri_proc(struct snd_card *card) { + struct snd_dbri *dbri = card->private_data; struct snd_info_entry *entry; - if (!snd_card_proc_new(dbri->card, "regs", &entry)) + if (!snd_card_proc_new(card, "regs", &entry)) snd_info_set_text_ops(entry, dbri, dbri_regs_read); #ifdef DBRI_DEBUG - if (!snd_card_proc_new(dbri->card, "debug", &entry)) { + if (!snd_card_proc_new(card, "debug", &entry)) { snd_info_set_text_ops(entry, dbri, dbri_debug_read); entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ } @@ -2507,17 +2503,16 @@ void snd_dbri_proc(struct snd_dbri *dbri */ static void snd_dbri_free(struct snd_dbri *dbri); -static int __init snd_dbri_create(struct snd_card *card, +static int __devinit snd_dbri_create(struct snd_card *card, struct sbus_dev *sdev, - struct linux_prom_irqs *irq, int dev) + int irq, int dev) { struct snd_dbri *dbri = card->private_data; int err; spin_lock_init(&dbri->lock); - dbri->card = card; dbri->sdev = sdev; - dbri->irq = irq->pri; + dbri->irq = irq; dbri->dma = sbus_alloc_consistent(sdev, sizeof(struct dbri_dma), &dbri->dma_dvma); @@ -2555,9 +2550,6 @@ static int __init snd_dbri_create(struct return err; } - dbri->next = dbri_list; - dbri_list = dbri; - return 0; } @@ -2577,20 +2569,19 @@ static void snd_dbri_free(struct snd_dbr (void *)dbri->dma, dbri->dma_dvma); } -static int __init dbri_attach(int prom_node, struct sbus_dev *sdev) +static int __devinit dbri_probe(struct of_device *of_dev, + const struct of_device_id *match) { + struct sbus_dev *sdev = to_sbus_device(&of_dev->dev); struct snd_dbri *dbri; - struct linux_prom_irqs irq; + int irq; struct resource *rp; struct snd_card *card; static int dev = 0; int err; - if (sdev->prom_name[9] < 'e') { - printk(KERN_ERR "DBRI: unsupported chip version %c found.\n", - sdev->prom_name[9]); - return -EIO; - } + dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n", + sdev->prom_name, sdev->slot); if (dev >= SNDRV_CARDS) return -ENODEV; @@ -2599,10 +2590,9 @@ static int __init dbri_attach(int prom_n return -ENOENT; } - err = prom_getproperty(prom_node, "intr", (char *)&irq, sizeof(irq)); - if (err < 0) { - printk(KERN_ERR "DBRI-%d: Firmware node lacks IRQ property.\n", - dev); + irq = sdev->irqs[0]; + if (irq <= 0) { + printk(KERN_ERR "DBRI-%d: No IRQ.\n", dev); return -ENODEV; } @@ -2616,24 +2606,26 @@ static int __init dbri_attach(int prom_n rp = &sdev->resource[0]; sprintf(card->longname, "%s at 0x%02lx:0x%016Lx, irq %d", card->shortname, - rp->flags & 0xffL, (unsigned long long)rp->start, irq.pri); + rp->flags & 0xffL, (unsigned long long)rp->start, irq); - if ((err = snd_dbri_create(card, sdev, &irq, dev)) < 0) { + err = snd_dbri_create(card, sdev, irq, dev); + if (err < 0) { snd_card_free(card); return err; } dbri = card->private_data; - err = snd_dbri_pcm(dbri); + err = snd_dbri_pcm(card); if (err < 0) goto _err; - err = snd_dbri_mixer(dbri); + err = snd_dbri_mixer(card); if (err < 0) goto _err; /* /proc file handling */ - snd_dbri_proc(dbri); + snd_dbri_proc(card); + dev_set_drvdata(&of_dev->dev, card); err = snd_card_register(card); if (err < 0) @@ -2652,43 +2644,46 @@ _err: return err; } -/* Probe for the dbri chip and then attach the driver. */ -static int __init dbri_init(void) +static int __devexit dbri_remove(struct of_device *dev) { - struct sbus_bus *sbus; - struct sbus_dev *sdev; - int found = 0; - - /* Probe each SBUS for the DBRI chip(s). */ - for_all_sbusdev(sdev, sbus) { - /* - * The version is coded in the last character - */ - if (!strncmp(sdev->prom_name, "SUNW,DBRI", 9)) { - dprintk(D_GEN, "DBRI: Found %s in SBUS slot %d\n", - sdev->prom_name, sdev->slot); + struct snd_card *card = dev_get_drvdata(&dev->dev); - if (dbri_attach(sdev->prom_node, sdev) == 0) - found++; - } - } + snd_dbri_free(card->private_data); + snd_card_free(card); - return (found > 0) ? 0 : -EIO; + dev_set_drvdata(&dev->dev, NULL); + + return 0; } -static void __exit dbri_exit(void) -{ - struct snd_dbri *this = dbri_list; +static struct of_device_id dbri_match[] = { + { + .name = "SUNW,DBRIe", + }, + { + .name = "SUNW,DBRIf", + }, + {}, +}; - while (this != NULL) { - struct snd_dbri *next = this->next; - struct snd_card *card = this->card; +MODULE_DEVICE_TABLE(of, dbri_match); - snd_dbri_free(this); - snd_card_free(card); - this = next; - } - dbri_list = NULL; +static struct of_platform_driver dbri_sbus_driver = { + .name = "dbri", + .match_table = dbri_match, + .probe = dbri_probe, + .remove = __devexit_p(dbri_remove), +}; + +/* Probe for the dbri chip and then attach the driver. */ +static int __init dbri_init(void) +{ + return of_register_driver(&dbri_sbus_driver, &sbus_bus_type); +} + +static void __exit dbri_exit(void) +{ + of_unregister_driver(&dbri_sbus_driver); } module_init(dbri_init);

2 1

Re: [alsa-devel] RE : M-Audio Audiophile USB, native little endian (S16_LE) ?
by Hakan Lennestal 05 Sep '07

05 Sep '07

> > I don't know if this is already known (?), but the M-Audio > > Audiophile USB seem to be possible to run in native S16_LE > > mode if initialized with device_setup=0x01 (possibly with > > 0x11 also, have not tested that). > > No I wasn't aware of this ! > How did you figure this out ? When initialized with device_setup=0x01 my box only produces noise. With 0x00 (or omitted) it works ok, except for spdif out sometimes not being setup correctly (no signal detect by the amp it is connected to). 0x09 seem to work without problems. Have not tested any other combinations. Using a setup with 0x01 and preparing a raw S16_LE file (with "mpg123 -s") and then playing with "aplay -D hw:2 -f S16_BE -r 44100 -c 2" produced good quality sound. This made me suspect that the device really is operating in native S16_LE. > When you say 'it is possible' to run it in S16_LE do you mean it both > recognizes S16_BE and S16_LE or only S16_LE ? Sorry, a bit unclear. Only S16_LE. > Mine works with S16_BE but I must admit I haven't tested with S16_LE. Ok, this then seem to be something that has changed over time in the firmware (or hw) ? The fw version of my box is 1.01. Regards. /Håkan

4 15

[alsa-devel] [PATCH] Gallant SC-6000 driver (3rd version)
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is port of the Gallant SC-6000 driver from the OSS aedsp16 driver. This card was also sold as AudioExcel DSP 16 and Zoltrix AV302 (Audio Plus True 16). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The two issues pointed by Takashi are fixed. diff -urpN linux-2.6.22.old/sound/isa/Kconfig linux-2.6.22/sound/isa/Kconfig --- linux-2.6.22.old/sound/isa/Kconfig 2007-08-28 23:14:08.000000000 +0200 +++ linux-2.6.22/sound/isa/Kconfig 2007-09-01 18:38:41.000000000 +0200 @@ -191,6 +191,19 @@ config SND_ES18XX To compile this driver as a module, choose M here: the module will be called snd-es18xx. +config SND_SC6000 + tristate "Gallant SC-6000, Audio Excel DSP 16" + depends on SND + select SND_AD1848_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for Gallant SC-6000 card and clones: + Audio Excel DSP 16 and Zoltrix AV302. + + To compile this driver as a module, choose M here: the module + will be called snd-sc6000. + config SND_GUS_SYNTH tristate diff -urpN linux-2.6.22.old/sound/isa/Makefile linux-2.6.22/sound/isa/Makefile --- linux-2.6.22.old/sound/isa/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22/sound/isa/Makefile 2007-09-03 22:09:58.000000000 +0200 @@ -10,6 +10,7 @@ snd-cmi8330-objs := cmi8330.o snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o +snd-sc6000-objs := sc6000.o snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330 obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o +obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o diff -urpN linux-2.6.22.old/sound/isa/sc6000.c linux-2.6.22/sound/isa/sc6000.c --- linux-2.6.22.old/sound/isa/sc6000.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22/sound/isa/sc6000.c 2007-09-04 19:25:04.000000000 +0200 @@ -0,0 +1,630 @@ +/* + * Driver for Gallant SC-6000 soundcard. This card is also known as + * Audio Excel DSP 16 or Zoltrix AV302. + * These cards use CompuMedia ASC-9308 chip + AD1848 codec. + * + * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1(a)wp.pl> + * + * I don't have documentation for this card. I used the driver + * for OSS/Free included in the kernel source as reference. + * + * 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 + */ + +#include <linux/module.h> +#include <linux/isa.h> +#include <asm/dma.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/ad1848.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#include <sound/control.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Krzysztof Helt"); +MODULE_DESCRIPTION("Gallant SC-6000"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000}," + "{AudioExcel, Audio Excel DSP 16}," + "{Zoltrix, AV302}}"); + +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; /* Enable this card */ +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ +static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + /* 0x300, 0x310, 0x320, 0x330 */ +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); +module_param_array(mss_port, long, NULL, 0444); +MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); +module_param_array(dma, int, NULL, 0444); +MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); + +/* + * Commands of SC6000's DSP (SBPRO+special). + * Some of them are COMMAND_xx, in the future they may change. + */ +#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ +#define COMMAND_52 0x52 /* */ +#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ +#define COMMAND_5C 0x5c /* */ +#define COMMAND_60 0x60 /* */ +#define COMMAND_66 0x66 /* */ +#define COMMAND_6C 0x6c /* */ +#define COMMAND_6E 0x6e /* */ +#define COMMAND_88 0x88 /* Unknown command */ +#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ +#define COMMAND_C5 0xc5 /* */ +#define GET_DSP_VERSION 0xe1 /* Get DSP Version */ +#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ + +/* + * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port + * to have the actual I/O port. + * Register permissions are: + * (wo) == Write Only + * (ro) == Read Only + * (w-) == Write + * (r-) == Read + */ +#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ +#define DSP_READ 0x0a /* offset of DSP READ (ro) */ +#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ +#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ +#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ +#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ + +#define PFX "sc6000: " + +/* hardware dependent functions */ + +/* + * sc6000_irq_to_softcfg - Decode irq number into cfg code. + */ +static inline unsigned char sc6000_irq_to_softcfg(int irq) +{ + unsigned char val = 0; + + switch (irq) { + case 5: + val = 0x28; + break; + case 7: + val = 0x8; + break; + case 9: + val = 0x10; + break; + case 10: + val = 0x18; + break; + case 11: + val = 0x20; + break; + default: + break; + } + return val; +} + +/* + * sc6000_dma_to_softcfg - Decode dma number into cfg code. + */ +static inline unsigned char sc6000_dma_to_softcfg(int dma) +{ + unsigned char val = 0; + + switch (dma) { + case 0: + val = 1; + break; + case 1: + val = 2; + break; + case 3: + val = 3; + break; + default: + break; + } + return val; +} + +/* + * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code. + */ +static inline unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) +{ + unsigned char val = 0; + + switch (mpu_irq) { + case 5: + val = 4; + break; + case 7: + val = 0x44; + break; + case 9: + val = 0x84; + break; + case 10: + val = 0xc4; + break; + default: + break; + } + return val; +} + +static __devinit int sc6000_wait_data(int port) +{ + int loop = 1000; + unsigned char val = 0; + + do { + val = inb(port + DSP_DATAVAIL); + if (val & 0x80) + return 0; + cpu_relax(); + } while (loop--); + + return -EAGAIN; +} + +static inline int sc6000_read(int port) +{ + if (sc6000_wait_data(port)) + return -EBUSY; + + return inb(port + DSP_READ); + +} + +static __devinit int sc6000_write(int port, int cmd) +{ + unsigned char val; + int loop = 500000; + + do { + val = inb(port + DSP_STATUS); + /* + * DSP ready to receive data if bit 7 of val == 0 + */ + if (!(val & 0x80)) { + outb(cmd, port + DSP_COMMAND); + return 0; + } + cpu_relax(); + } while (loop--); + + snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd); + + return -EAGAIN; +} + +static int __devinit sc6000_dsp_get_answer(int port, int command, char *data, + int data_len) +{ + int len = 0; + + if (sc6000_write(port, command)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command); + return -EFAULT; + } + + do { + int val = sc6000_read(port); + + if (val < 0) + break; + + data[len++] = val; + + } while (len < data_len); + + /* + * If no more data available, return to the caller, no error if len>0. + * We have no other way to know when the string is finished. + */ + return len ? len : -EIO; +} + +static int __devinit sc6000_dsp_reset(int port) +{ + outb(1, port + DSP_RESET); + udelay(10); + outb(0, port + DSP_RESET); + udelay(20); + if (sc6000_read(port) == 0xaa) + return 0; + return -EBUSY; +} + +/* detection and initialization */ +static int __devinit sc6000_cfg_write(int port, unsigned char softcfg) +{ + + if (sc6000_write(port, WRITE_MDIRQ_CFG)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return -EIO; + } + if (sc6000_write(port, softcfg)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EIO; + } + return 0; +} + +static int __devinit sc6000_setup_board(int port, int config) +{ + int loop = 10; + + do { + if (sc6000_write(port, COMMAND_88)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", + COMMAND_88); + return -EIO; + } + } while ((sc6000_wait_data(port) < 0) && loop--); + + if (sc6000_read(port) < 0) { + snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", + COMMAND_88); + return -EIO; + } + + if (sc6000_cfg_write(port, config)) + return -EFAULT; + + return 0; +} + +static int __devinit sc6000_init_mss(int port, int config, int mss_port, + int mss_config) +{ + if (sc6000_write(port, DSP_INIT_MSS)) { + snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n", + DSP_INIT_MSS); + return -EIO; + } + + msleep(10); + + if (sc6000_cfg_write(port, config)) + return -EIO; + + outb(mss_config, mss_port); + + return 0; +} + +static int __devinit sc6000_init_board(int port, int irq, int dma, + int mss_port, int mpu_irq) +{ + char answer[15]; + char version[2]; + int mss_config = sc6000_irq_to_softcfg(irq) | + sc6000_dma_to_softcfg(dma); + int config = mss_config | + sc6000_mpu_irq_to_softcfg(mpu_irq); + + if (sc6000_dsp_reset(port)) { + snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n"); + return -EBUSY; + } + + if (sc6000_dsp_get_answer(port, GET_DSP_COPYRIGHT, answer, 15) <= 0) { + snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n"); + return -ENODEV; + } + /* + * My SC-6000 card return "SC-6000" in DSPCopyright, so + * if we have something different, we have to be warned. + * Mine returns SC-6000A - KH + */ + if (strncmp("SC-6000", answer, 7)) + snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); + + if (sc6000_dsp_get_answer(port, GET_DSP_VERSION, version, 2) < 2) { + snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n"); + return -ENODEV; + } + printk(KERN_INFO PFX "Detected model: %s version %d.%d\n", + answer, version[1], version[0]); + + /* + * 0x0A == (IRQ 7, DMA 1, MIRQ 0) + */ + if (sc6000_cfg_write(port, 0x0a)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EFAULT; + } + + if (sc6000_setup_board(port, config) < 0) { + snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + if (sc6000_init_mss(port, config, mss_port, + mss_config) < 0) { + snd_printk(KERN_ERR "Can not initialize" + "Microsoft Sound System mode.\n"); + return -ENODEV; + } + + return 0; +} + +static int __devinit snd_sc6000_detect(int dev, int irq, int dma) +{ + int err = -ENODEV; + + snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", + port[dev], irq, dma, + mpu_irq[dev] == SND_AUTO_IRQ ? 0 : mpu_irq[dev]); + +/* + * We must request the port region because these are the I/O + * ports to access card's control registers. + */ + if (!request_region(port[dev], 0x10, "sc-6000 (base)")) { + snd_printk(KERN_ERR + "SC-6000 port I/O port region is already in use.\n"); + return -EBUSY; + } + + err = sc6000_init_board(port[dev], irq, dma, + mss_port[dev], mpu_irq[dev]); + + release_region(port[dev], 0x10); + + return err; +} + +static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +{ + struct snd_card *card = chip->card; + struct snd_ctl_elem_id id1, id2; + int err; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 to FM */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "FM Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + /* reassign AUX1 to CD */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + return 0; +} + +static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify IO port\n"); + return 0; + } + if (mss_port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify MSS port\n"); + return 0; + } + if (port[dev] != 0x220 && port[dev] != 0x240) { + printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n"); + return 0; + } + if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) { + printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n"); + return 0; + } + if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) { + printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]); + return 0; + } + if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) { + printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x30l) != 0x300) { + printk(KERN_ERR PFX "invalid MPU-401 port %lx\n", + mpu_port[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 && + !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) { + printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]); + return 0; + } + return 1; +} + +static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) +{ + static int possible_irqs[] = { 7, 9, 10, 11, -1 }; + static int possible_dmas[] = { 1, 3, 0, -1 }; + int err; + int xirq = irq[dev]; + int xdma = dma[dev]; + struct snd_card *card; + struct snd_ad1848 *chip; + struct snd_opl3 *opl3; + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (!card) + return -ENOMEM; + + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto _err; + } + } + + if (xdma == SNDRV_AUTO_DMA) { + xdma = snd_legacy_find_free_dma(possible_dmas); + if (xdma < 0) { + snd_printk(KERN_ERR "unable to find a free DMA\n"); + err = -EBUSY; + goto _err; + } + } + + err = snd_sc6000_detect(dev, xirq, xdma); + if (err < 0) + goto _err; + + err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, + AD1848_HW_DETECT, &chip); + if (err < 0) + goto _err; + card->private_data = chip; + + err = snd_ad1848_pcm(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR + "error creating new ad1848 PCM device\n"); + goto _err; + } + err = snd_ad1848_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "error creating new ad1848 mixer\n"); + goto _err; + } + err = snd_sc6000_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "the mixer rewrite failed\n"); + goto _err; + } + if (snd_opl3_create(card, + 0x388, 0x388 + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR "no OPL device at 0x%x-0x%x ?\n", + 0x388, 0x388 + 2); + } else { + err = snd_opl3_timer_new(opl3, 0, 1); + if (err < 0) + goto _err; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto _err; + } + + if (mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", + mpu_port[dev]); + } + + strcpy(card->driver, "Gallant SC-6000"); + strcpy(card->shortname, "Gallant SC-6000"); + sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d", + mss_port[dev], xirq, xdma); + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto _err; + + dev_set_drvdata(devptr, card); + return 0; + + _err: + snd_card_free(card); + return err; +} + +static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + dev_set_drvdata(devptr, NULL); + return 0; +} + +static struct isa_driver snd_sc6000_driver = { + .match = snd_sc6000_match, + .probe = snd_sc6000_probe, + .remove = __devexit_p(snd_sc6000_remove), + /* FIXME: suspend/resume */ + .driver = { + .name = "sc6000", + }, +}; + + +static int __init alsa_card_sc6000_init(void) +{ + return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_sc6000_exit(void) +{ + isa_unregister_driver(&snd_sc6000_driver); +} + +module_init(alsa_card_sc6000_init) +module_exit(alsa_card_sc6000_exit)

2 3

[alsa-devel] HDA Conexant: is mic boost possible?
by Pablo N. Hess 05 Sep '07

05 Sep '07

I have to make the sound chip of a Itautec W7635 (= FIC VA250D) work on a Gentoo(-based) distro (I'm now trying Sabayon). It is a HDA-intel (Conexant codec). Using my earphones as a stereo mic, recording with 'arecord -f cd' has always worked perfectly fine, from alsa-driver-1.0.14 on. However, when using a regular mono mic (like the ones that come in common headsets), recording simply does *not* work. Here's the quick info: alsa-driver-1.0.14: playback: ok recording: stereo mic: ok mono mic: no input at all (very high volume already tried) alsa-driver-1.0.15rc1: playback: ok (even better than with alsa-driver-1.0.14) recording: stereo mic: ok (seems better thatn with alsa-driver-1.0.14) mono mic: *very* very low volume, bad quality, lots of low-volume noise, too. Now the paragraphed info: Playback is perfectly ok since alsa-driver-1.0.14, just like recording with an earphone as a stereo mic. With alsa-driver-1.0.14, definitely no mono-mic input works, no matter the settings at alsamixer. With alsa-driver-1.0.15rc1, mono-mic input works very very bad (extremely low volume, terrible quality, lots of clicks and noise). I guess that the mono mic needs a mic boost in alsa-driver-1.0.15rc1. However, with a 'grep -i "mic boost" .../pci/hda/*.c' (alsa-driver-1.0.15rc1 source code), it seems that only patch-realtek.c and patch-analog.c support this feature. Is it even possible to add a mic boost feature to this codec? The output of alsa-info.sh with alsa-driver-1.0.15rc1 is at: http://pastebin.ca/681023 Thanks a lot for any info, and congratulations to all of you for your terrific work with ALSA. Pablo Hess

2 1

[alsa-devel] [PATCH] cs4231-lib: replace common delay loop by function
by Krzysztof Helt 05 Sep '07

05 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch replaces a common delay loop by a function. It also uses ARRAY_SIZE macro for the rates table. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-2.6.23/sound/isa/cs423x/cs4231_lib.c linux-2.6.22/sound/isa/cs423x/cs4231_lib.c --- linux-2.6.23/sound/isa/cs423x/cs4231_lib.c 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22/sound/isa/cs423x/cs4231_lib.c 2007-09-04 09:52:01.000000000 +0200 @@ -74,7 +74,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, .mask = 0, }; @@ -134,41 +134,38 @@ static inline u8 cs4231_inb(struct snd_c return inb(chip->port + offset); } -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, - unsigned char mask, unsigned char value) +static void snd_cs4231_ready(struct snd_cs4231 *chip) { int timeout; - unsigned char tmp; for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) udelay(100); +} + +static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, + unsigned char mask, unsigned char value) +{ + unsigned char tmp = (chip->image[reg] & mask) | value; + + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif - if (chip->calibrate_mute) { - chip->image[reg] &= mask; - chip->image[reg] |= value; - } else { + if (!chip->calibrate_mute) { cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); - mb(); - tmp = (chip->image[reg] & mask) | value; + wmb(); cs4231_outb(chip, CS4231P(REG), tmp); - chip->image[reg] = tmp; mb(); } + chip->image[reg] = tmp; } static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(10); + snd_cs4231_ready(chip); cs4231_outb(chip, CS4231P(REGSEL), chip->mce_bit | reg); cs4231_outb(chip, CS4231P(REG), value); mb(); @@ -176,12 +173,7 @@ static void snd_cs4231_dout(struct snd_c void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); @@ -190,19 +182,11 @@ void snd_cs4231_out(struct snd_cs4231 *c cs4231_outb(chip, CS4231P(REG), value); chip->image[reg] = value; mb(); -#if 0 - printk("codec out - reg 0x%x = 0x%x\n", chip->mce_bit | reg, value); -#endif } unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) { - int timeout; - - for (timeout = 250; - timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("in: auto calibration time out - reg = 0x%x\n", reg); @@ -304,8 +288,7 @@ void snd_cs4231_mce_up(struct snd_cs4231 unsigned long flags; int timeout; - for (timeout = 250; timeout > 0 && (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT); timeout--) - udelay(100); + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (cs4231_inb(chip, CS4231P(REGSEL)) & CS4231_INIT) snd_printk("mce_up - auto calibration time out (0)\n"); @@ -459,11 +442,11 @@ static unsigned char snd_cs4231_get_rate { int i; - for (i = 0; i < 14; i++) + for (i = 0; i < ARRAY_SIZE(rates); i++) if (rate == rates[i]) return freq_bits[i]; // snd_BUG(); - return freq_bits[13]; + return freq_bits[ARRAY_SIZE(rates) - 1]; } static unsigned char snd_cs4231_get_format(struct snd_cs4231 *chip,

2 1

[alsa-devel] How to control the route in smartphone
by Leijin Tang 04 Sep '07

04 Sep '07

Hi all, When the audio codec is used in the smartphone, we get the problem how to define and control the route in the alsa driver. The scenario is only in the phone device. The application need switch the routes based on the different scenes. Example: the coedec is wm9713L. the in port: LINEL，LINER，MONOIN，MIC1，MIC2, ac97_in the out port: HEADSET, SPEAKER, RECEIVER, ac97_out When the phone is coming, we need chose the route LINE to receiver. After the calling and playing the mp3, we should switch the route ac97_in to speaker on. This scenario doesn’t appear in the PC. So how to carry out the function in the alsa architecture is so difficulty. The router names how to define, the alsa-lib how to do the control.There has no design to the phone device in the alsa by now. Does anybody have some ideas? We can do some discussion and resolve the issue.

2 1

[alsa-devel] [PATCH] Gallant SC-6000 driver (2nd version)
by Krzysztof Helt 04 Sep '07

04 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is port of the Gallant SC-6000 driver from the OSS aedsp16 driver. This card was also sold as AudioExcel DSP 16 and Zoltrix AV302 (Audio Plus True 16). --- I ported only SC-6000 part from the OSS driver. PCM, OPL3 and CD playback work. Mixer also works. I haven't tested recording and MPU-401 port. This driver does not support the SC-6600 card. If it does the names inside the driver will reflect it (like "Gallant SC-6x00"). If anybody has the SC-6600 card I can port this part as well, but I do not have hardware to test. diff -urNp linux-2.6.22.old/sound/isa/Kconfig linux-2.6.22/sound/isa/Kconfig --- linux-2.6.22.old/sound/isa/Kconfig 2007-08-28 23:14:08.000000000 +0200 +++ linux-2.6.22/sound/isa/Kconfig 2007-09-01 18:38:41.000000000 +0200 @@ -191,6 +191,19 @@ config SND_ES18XX To compile this driver as a module, choose M here: the module will be called snd-es18xx. +config SND_SC6000 + tristate "Gallant SC-6000, Audio Excel DSP 16" + depends on SND + select SND_AD1848_LIB + select SND_OPL3_LIB + select SND_MPU401_UART + help + Say Y here to include support for Gallant SC-6000 card and clones: + Audio Excel DSP 16 and Zoltrix AV302. + + To compile this driver as a module, choose M here: the module + will be called snd-sc6000. + config SND_GUS_SYNTH tristate diff -urNp linux-2.6.22.old/sound/isa/Makefile linux-2.6.22/sound/isa/Makefile --- linux-2.6.22.old/sound/isa/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.22/sound/isa/Makefile 2007-09-03 22:09:58.000000000 +0200 @@ -10,6 +10,7 @@ snd-cmi8330-objs := cmi8330.o snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o +snd-sc6000-objs := sc6000.o snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330 obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o +obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o diff -urNp linux-2.6.22.old/sound/isa/sc6000.c linux-2.6.22/sound/isa/sc6000.c --- linux-2.6.22.old/sound/isa/sc6000.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.22/sound/isa/sc6000.c 2007-09-03 22:12:32.000000000 +0200 @@ -0,0 +1,630 @@ +/* + * Driver for Gallant SC-6000 soundcard. This card is also known as + * Audio Excel DSP 16 or Zoltrix AV302. + * These cards use CompuMedia ASC-9308 chip + AD1848 codec. + * + * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1(a)wp.pl> + * + * I don't have documentation for this card. I used the driver + * for OSS/Free included in the kernel source as reference. + * + * 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 + */ + +#include <linux/module.h> +#include <linux/isa.h> +#include <asm/dma.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/ad1848.h> +#include <sound/opl3.h> +#include <sound/mpu401.h> +#include <sound/control.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Krzysztof Helt"); +MODULE_DESCRIPTION("Gallant SC-6000"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000}," + "{AudioExcel, Audio Excel DSP 16}," + "{Zoltrix, AV302}}"); + +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; /* Enable this card */ +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ +static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + /* 0x300, 0x310, 0x320, 0x330 */ +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); +module_param_array(mss_port, long, NULL, 0444); +MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); +module_param_array(dma, int, NULL, 0444); +MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); + +/* + * Commands of SC6000's DSP (SBPRO+special). + * Some of them are COMMAND_xx, in the future they may change. + */ +#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ +#define COMMAND_52 0x52 /* */ +#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ +#define COMMAND_5C 0x5c /* */ +#define COMMAND_60 0x60 /* */ +#define COMMAND_66 0x66 /* */ +#define COMMAND_6C 0x6c /* */ +#define COMMAND_6E 0x6e /* */ +#define COMMAND_88 0x88 /* Unknown command */ +#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ +#define COMMAND_C5 0xc5 /* */ +#define GET_DSP_VERSION 0xe1 /* Get DSP Version */ +#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ + +/* + * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port + * to have the actual I/O port. + * Register permissions are: + * (wo) == Write Only + * (ro) == Read Only + * (w-) == Write + * (r-) == Read + */ +#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ +#define DSP_READ 0x0a /* offset of DSP READ (ro) */ +#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ +#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ +#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ +#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ + +#define PFX "sc6000: " + +/* hardware dependent functions */ + +/* + * sc6000_irq_to_softcfg - Decode irq number into cfg code. + */ +static inline unsigned char sc6000_irq_to_softcfg(int irq) +{ + unsigned char val = 0; + + switch (irq) { + case 5: + val = 0x28; + break; + case 7: + val = 0x8; + break; + case 9: + val = 0x10; + break; + case 10: + val = 0x18; + break; + case 11: + val = 0x20; + break; + default: + break; + } + return val; +} + +/* + * sc6000_dma_to_softcfg - Decode dma number into cfg code. + */ +static inline unsigned char sc6000_dma_to_softcfg(int dma) +{ + unsigned char val = 0; + + switch (dma) { + case 0: + val = 1; + break; + case 1: + val = 2; + break; + case 3: + val = 3; + break; + default: + break; + } + return val; +} + +/* + * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code. + */ +static inline unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) +{ + unsigned char val = 0; + + switch (mpu_irq) { + case 5: + val = 4; + break; + case 7: + val = 0x44; + break; + case 9: + val = 0x84; + break; + case 10: + val = 0xc4; + break; + default: + break; + } + return val; +} + +static __devinit int sc6000_wait_data(int port) +{ + int loop = 1000; + unsigned char val = 0; + + do { + val = inb(port + DSP_DATAVAIL); + if (val & 0x80) + return 0; + cpu_relax(); + } while (loop--); + + return -EAGAIN; +} + +static inline int sc6000_read(int port) +{ + if (sc6000_wait_data(port)) + return -EBUSY; + + return inb(port + DSP_READ); + +} + +static __devinit int sc6000_write(int port, int cmd) +{ + unsigned char val; + int loop = 500000; + + do { + val = inb(port + DSP_STATUS); + /* + * DSP ready to receive data if bit 7 of val == 0 + */ + if (!(val & 0x80)) { + outb(cmd, port + DSP_COMMAND); + return 0; + } + cpu_relax(); + } while (loop--); + + snd_printk(KERN_ERR "DSP Command (0x%x) timeout.\n", cmd); + + return -EAGAIN; +} + +static int __devinit sc6000_dsp_get_answer(int port, int command, char *data, + int data_len) +{ + int len = 0; + + if (sc6000_write(port, command)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", command); + return -EFAULT; + } + + do { + int val = sc6000_read(port); + + if (val < 0) + break; + + data[len++] = val; + + } while (len < data_len); + + /* + * If no more data available, return to the caller, no error if len>0. + * We have no other way to know when the string is finished. + */ + return len ? len : -EIO; +} + +static int __devinit sc6000_dsp_reset(int port) +{ + outb(1, port + DSP_RESET); + udelay(10); + outb(0, port + DSP_RESET); + udelay(20); + if (sc6000_read(port) == 0xaa) + return 0; + return -EBUSY; +} + +/* detection and initialization */ +static int __devinit sc6000_cfg_write(int port, unsigned char softcfg) +{ + + if (sc6000_write(port, WRITE_MDIRQ_CFG)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return -EIO; + } + if (sc6000_write(port, softcfg)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EIO; + } + return 0; +} + +static int __devinit sc6000_setup_board(int port, int config) +{ + int loop = 10; + + do { + if (sc6000_write(port, COMMAND_88)) { + snd_printk(KERN_ERR "CMD 0x%x: failed!\n", + COMMAND_88); + return -EIO; + } + } while ((sc6000_wait_data(port) < 0) && loop--); + + if (sc6000_read(port) < 0) { + snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", + COMMAND_88); + return -EIO; + } + + if (sc6000_cfg_write(port, config)) + return -EFAULT; + + return 0; +} + +static int __devinit sc6000_init_mss(int port, int config, int mss_port, + int mss_config) +{ + if (sc6000_write(port, DSP_INIT_MSS)) { + snd_printk(KERN_ERR "sc6000_init_mss [0x%x]: failed!\n", + DSP_INIT_MSS); + return -EIO; + } + + mdelay(10); + + if (sc6000_cfg_write(port, config)) + return -EIO; + + outb(mss_config, mss_port); + + return 0; +} + +static int __devinit sc6000_init_board(int port, int irq, int dma, + int mss_port, int mpu_irq) +{ + char answer[15]; + char version[2]; + int mss_config = sc6000_irq_to_softcfg(irq) | + sc6000_dma_to_softcfg(dma); + int config = mss_config | + sc6000_mpu_irq_to_softcfg(mpu_irq); + + if (sc6000_dsp_reset(port)) { + snd_printk(KERN_ERR "sc6000_dsp_reset: failed!\n"); + return -EBUSY; + } + + if (sc6000_dsp_get_answer(port, GET_DSP_COPYRIGHT, answer, 15) <= 0) { + snd_printk(KERN_ERR "sc6000_dsp_copyright: failed!\n"); + return -ENODEV; + } + /* + * My SC-6000 card return "SC-6000" in DSPCopyright, so + * if we have something different, we have to be warned. + * Mine returns SC-6000A - KH + */ + if (strncmp("SC-6000", answer, 7)) + snd_printk(KERN_WARNING "Warning: non SC-6000 audio card!\n"); + + if (sc6000_dsp_get_answer(port, GET_DSP_VERSION, version, 2) < 2) { + snd_printk(KERN_ERR "sc6000_dsp_version: failed!\n"); + return -ENODEV; + } + printk(KERN_INFO PFX "Detected model: %s version %d.%d\n", + answer, version[1], version[0]); + + /* + * 0x0A == (IRQ 7, DMA 1, MIRQ 0) + */ + if (sc6000_cfg_write(port, 0x0a)) { + snd_printk(KERN_ERR "sc6000_cfg_write: failed!\n"); + return -EFAULT; + } + + if (sc6000_setup_board(port, config) < 0) { + snd_printk(KERN_ERR "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + if (sc6000_init_mss(port, config, mss_port, + mss_config) < 0) { + snd_printk(KERN_ERR "Can not initialize" + "Microsoft Sound System mode.\n"); + return -ENODEV; + } + + return 0; +} + +static int __devinit snd_sc6000_detect(int dev, int irq, int dma) +{ + int err = -ENODEV; + + snd_printd("Initializing BASE[0x%lx] IRQ[%d] DMA[%d] MIRQ[%d]\n", + port[dev], irq, dma, + mpu_irq[dev] == SND_AUTO_IRQ ? 0 : mpu_irq[dev]); + +/* + * We must request the port region because these are the I/O + * ports to access card's control registers. + */ + if (!request_region(port[dev], 0x10, "sc-6000 (base)")) { + snd_printk(KERN_ERR + "SC-6000 port I/O port region is already in use.\n"); + return -EBUSY; + } + + err = sc6000_init_board(port[dev], irq, dma, + mss_port[dev], mpu_irq[dev]); + + release_region(port[dev], 0x10); + + return err; +} + +static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +{ + struct snd_card *card = chip->card; + struct snd_ctl_elem_id id1, id2; + int err; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 to FM */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "FM Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + /* reassign AUX1 to CD */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + return 0; +} + +static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify IO port\n"); + return 0; + } + if (mss_port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify MSS port\n"); + return 0; + } + if (port[dev] != 0x220 && port[dev] != 0x240) { + printk(KERN_ERR PFX "Port must be 0x220 or 0x240\n"); + return 0; + } + if (mss_port[dev] != 0x530 && mss_port[dev] != 0xe80) { + printk(KERN_ERR PFX "MSS port must be 0x530 or 0xe80\n"); + return 0; + } + if (irq[dev] != SNDRV_AUTO_IRQ && !sc6000_irq_to_softcfg(irq[dev])) { + printk(KERN_ERR PFX "invalid IRQ %d\n", irq[dev]); + return 0; + } + if (dma[dev] != SNDRV_AUTO_DMA && !sc6000_dma_to_softcfg(dma[dev])) { + printk(KERN_ERR PFX "invalid DMA %d\n", dma[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + (mpu_port[dev] & ~0x30l) != 0x300) { + printk(KERN_ERR PFX "invalid MPU-401 port %lx\n", + mpu_port[dev]); + return 0; + } + if (mpu_port[dev] != SNDRV_AUTO_PORT && + mpu_irq[dev] != SNDRV_AUTO_IRQ && mpu_irq[dev] != 0 && + !sc6000_mpu_irq_to_softcfg(mpu_irq[dev])) { + printk(KERN_ERR PFX "invalid MPU-401 IRQ %d\n", mpu_irq[dev]); + return 0; + } + return 1; +} + +static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) +{ + static int possible_irqs[] = { 7, 9, 10, 11, -1 }; + static int possible_dmas[] = { 1, 3, 0, -1 }; + int err; + int xirq = irq[dev]; + int xdma = dma[dev]; + struct snd_card *card; + struct snd_ad1848 *chip; + struct snd_opl3 *opl3; + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR "unable to find a free IRQ\n"); + err = -EBUSY; + goto _err; + } + } + + if (xdma == SNDRV_AUTO_DMA) { + xdma = snd_legacy_find_free_dma(possible_dmas); + if (xdma < 0) { + snd_printk(KERN_ERR "unable to find a free DMA\n"); + err = -EBUSY; + goto _err; + } + } + + err = snd_sc6000_detect(dev, xirq, xdma); + if (err < 0) + goto _err; + + err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, + AD1848_HW_DETECT, &chip); + if (err < 0) + goto _err; + card->private_data = chip; + + err = snd_ad1848_pcm(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR + "error creating new ad1848 PCM device\n"); + goto _err; + } + err = snd_ad1848_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "error creating new ad1848 mixer\n"); + goto _err; + } + err = snd_sc6000_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR "the mixer rewrite failed\n"); + goto _err; + } + if (snd_opl3_create(card, + 0x388, 0x388 + 2, + OPL3_HW_AUTO, 0, &opl3) < 0) { + snd_printk(KERN_ERR "no OPL device at 0x%x-0x%x ?\n", + 0x388, 0x388 + 2); + } else { + err = snd_opl3_timer_new(opl3, 0, 1); + if (err < 0) + goto _err; + + err = snd_opl3_hwdep_new(opl3, 0, 1, NULL); + if (err < 0) + goto _err; + } + + if (mpu_port[dev] != SNDRV_AUTO_PORT) { + if (mpu_irq[dev] == SNDRV_AUTO_IRQ) + mpu_irq[dev] = -1; + if (snd_mpu401_uart_new(card, 0, + MPU401_HW_MPU401, + mpu_port[dev], 0, + mpu_irq[dev], + mpu_irq[dev] >= 0 ? IRQF_DISABLED : 0, + NULL) < 0) + snd_printk(KERN_ERR "no MPU-401 device at 0x%lx ?\n", + mpu_port[dev]); + } + + strcpy(card->driver, "Gallant SC-6000"); + strcpy(card->shortname, "Gallant SC-6000"); + sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d", + mss_port[dev], xirq, xdma); + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto _err; + + dev_set_drvdata(devptr, card); + return 0; + + _err: + snd_card_free(card); + return err; +} + +static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + dev_set_drvdata(devptr, NULL); + return 0; +} + +static struct isa_driver snd_sc6000_driver = { + .match = snd_sc6000_match, + .probe = snd_sc6000_probe, + .remove = __devexit_p(snd_sc6000_remove), + /* FIXME: suspend/resume */ + .driver = { + .name = "sc6000", + }, +}; + + +static int __init alsa_card_sc6000_init(void) +{ + return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_sc6000_exit(void) +{ + isa_unregister_driver(&snd_sc6000_driver); +} + +module_init(alsa_card_sc6000_init) +module_exit(alsa_card_sc6000_exit)

2 1

[alsa-devel] [PATCH] cs4231 header split
by Krzysztof Helt 04 Sep '07

04 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch splits the cs4231.h file into two parts: - cs4231-regs.h which contain register constants and macros - cs4231.h which includes the above and contain rest of the definitions This will allow to share register definitions between x86 ISA cs4231 and SPARC cs4231. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- The next step is to use the definitions into the sparc cs4231 driver. If it is not a good idea please reject this patch. diff -uNrp linux-2.6.23.orig/include/sound/cs4231-regs.h linux-2.6.23/include/sound/cs4231-regs.h --- linux-2.6.23.orig/include/sound/cs4231-regs.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.23/include/sound/cs4231-regs.h 2007-09-03 18:28:14.590542832 +0200 @@ -0,0 +1,180 @@ +#ifndef __SOUND_CS4231_REGS_H +#define __SOUND_CS4231_REGS_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)suse.cz> + * Definitions for CS4231 & InterWave chips & compatible 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 CS4231P(x) (c_d_c_CS4231##x) + +#define c_d_c_CS4231REGSEL 0 +#define c_d_c_CS4231REG 1 +#define c_d_c_CS4231STATUS 2 +#define c_d_c_CS4231PIO 3 + +/* codec registers */ + +#define CS4231_LEFT_INPUT 0x00 /* left input control */ +#define CS4231_RIGHT_INPUT 0x01 /* right input control */ +#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ +#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ +#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ +#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ +#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ +#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ +#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ +#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ +#define CS4231_PIN_CTRL 0x0a /* pin control */ +#define CS4231_TEST_INIT 0x0b /* test and initialization */ +#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ +#define CS4231_LOOPBACK 0x0d /* loopback control */ +#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ +#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ +#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ +#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */ +#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ +#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */ +#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */ +#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */ +#define CS4231_TIMER_LOW 0x14 /* timer low byte */ +#define CS4231_TIMER_HIGH 0x15 /* timer high byte */ +#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */ +#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */ +#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */ +#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */ +#define CS4236_EXT_REG 0x17 /* extended register access */ +#define CS4231_IRQ_STATUS 0x18 /* irq status register */ +#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */ +#define CS4231_VERSION 0x19 /* CS4231(A) - version values */ +#define CS4231_MONO_CTRL 0x1a /* mono input/output control */ +#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */ +#define AD1845_PWR_DOWN 0x1b /* power down control */ +#define CS4235_LEFT_MASTER 0x1b /* left master output control */ +#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */ +#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */ +#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */ +#define CS4235_RIGHT_MASTER 0x1d /* right master output control */ +#define CS4231_REC_UPR_CNT 0x1e /* record upper count */ +#define CS4231_REC_LWR_CNT 0x1f /* record lower count */ + +/* definitions for codec register select port - CODECP( REGSEL ) */ + +#define CS4231_INIT 0x80 /* CODEC is initializing */ +#define CS4231_MCE 0x40 /* mode change enable */ +#define CS4231_TRD 0x20 /* transfer request disable */ + +/* definitions for codec status register - CODECP( STATUS ) */ + +#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ + +/* definitions for codec irq status */ + +#define CS4231_PLAYBACK_IRQ 0x10 +#define CS4231_RECORD_IRQ 0x20 +#define CS4231_TIMER_IRQ 0x40 +#define CS4231_ALL_IRQS 0x70 +#define CS4231_REC_UNDERRUN 0x08 +#define CS4231_REC_OVERRUN 0x04 +#define CS4231_PLY_OVERRUN 0x02 +#define CS4231_PLY_UNDERRUN 0x01 + +/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ + +#define CS4231_ENABLE_MIC_GAIN 0x20 + +#define CS4231_MIXS_LINE 0x00 +#define CS4231_MIXS_AUX1 0x40 +#define CS4231_MIXS_MIC 0x80 +#define CS4231_MIXS_ALL 0xc0 + +/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ + +#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ +#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ +#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ +#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ +#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ +#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ +#define CS4231_STEREO 0x10 /* stereo mode */ +/* bits 3-1 define frequency divisor */ +#define CS4231_XTAL1 0x00 /* 24.576 crystal */ +#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ + +/* definitions for interface control register - CS4231_IFACE_CTRL */ + +#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ +#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ +#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ +#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ +#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ +#define CS4231_RECORD_ENABLE 0x02 /* record enable */ +#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ + +/* definitions for pin control register - CS4231_PIN_CTRL */ + +#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ +#define CS4231_XCTL1 0x40 /* external control #1 */ +#define CS4231_XCTL0 0x80 /* external control #0 */ + +/* definitions for test and init register - CS4231_TEST_INIT */ + +#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ +#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ + +/* definitions for misc control register - CS4231_MISC_INFO */ + +#define CS4231_MODE2 0x40 /* MODE 2 */ +#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */ +#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */ + +/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */ + +#define CS4231_DACZ 0x01 /* zero DAC when underrun */ +#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */ +#define CS4231_OLB 0x80 /* output level bit */ + +/* definitions for Extended Registers - CS4236+ */ + +#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f)) +#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8) + +#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */ +#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */ +#define CS4236_LEFT_MIC 0x28 /* left MIC volume */ +#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */ +#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */ +#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */ +#define CS4236_LEFT_FM 0x68 /* left FM volume */ +#define CS4236_RIGHT_FM 0x78 /* right FM volume */ +#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */ +#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */ +#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */ +#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */ +#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */ +#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */ +#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */ +#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */ +#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */ +#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ +#define CS4236_VERSION 0x9c /* chip version and ID */ + +#endif /* __SOUND_CS4231_REGS_H */ diff -uNrp linux-2.6.23.orig/include/sound/cs4231-regs.h~ linux-2.6.23/include/sound/cs4231-regs.h~ --- linux-2.6.23.orig/include/sound/cs4231-regs.h~ 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.23/include/sound/cs4231-regs.h~ 2007-09-03 18:27:55.321444748 +0200 @@ -0,0 +1,184 @@ +#ifndef __SOUND_CS4231_REGS_H +#define __SOUND_CS4231_REGS_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)suse.cz> + * Definitions for CS4231 & InterWave chips & compatible chips + * + * + * 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 + * + */ + +#include "control.h" +#include "pcm.h" +#include "timer.h" + +/* IO ports */ + +#define CS4231P(x) (c_d_c_CS4231##x) + +#define c_d_c_CS4231REGSEL 0 +#define c_d_c_CS4231REG 1 +#define c_d_c_CS4231STATUS 2 +#define c_d_c_CS4231PIO 3 + +/* codec registers */ + +#define CS4231_LEFT_INPUT 0x00 /* left input control */ +#define CS4231_RIGHT_INPUT 0x01 /* right input control */ +#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ +#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ +#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ +#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ +#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ +#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ +#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ +#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ +#define CS4231_PIN_CTRL 0x0a /* pin control */ +#define CS4231_TEST_INIT 0x0b /* test and initialization */ +#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ +#define CS4231_LOOPBACK 0x0d /* loopback control */ +#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ +#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ +#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ +#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */ +#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ +#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */ +#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */ +#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */ +#define CS4231_TIMER_LOW 0x14 /* timer low byte */ +#define CS4231_TIMER_HIGH 0x15 /* timer high byte */ +#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */ +#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */ +#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */ +#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */ +#define CS4236_EXT_REG 0x17 /* extended register access */ +#define CS4231_IRQ_STATUS 0x18 /* irq status register */ +#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */ +#define CS4231_VERSION 0x19 /* CS4231(A) - version values */ +#define CS4231_MONO_CTRL 0x1a /* mono input/output control */ +#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */ +#define AD1845_PWR_DOWN 0x1b /* power down control */ +#define CS4235_LEFT_MASTER 0x1b /* left master output control */ +#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */ +#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */ +#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */ +#define CS4235_RIGHT_MASTER 0x1d /* right master output control */ +#define CS4231_REC_UPR_CNT 0x1e /* record upper count */ +#define CS4231_REC_LWR_CNT 0x1f /* record lower count */ + +/* definitions for codec register select port - CODECP( REGSEL ) */ + +#define CS4231_INIT 0x80 /* CODEC is initializing */ +#define CS4231_MCE 0x40 /* mode change enable */ +#define CS4231_TRD 0x20 /* transfer request disable */ + +/* definitions for codec status register - CODECP( STATUS ) */ + +#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ + +/* definitions for codec irq status */ + +#define CS4231_PLAYBACK_IRQ 0x10 +#define CS4231_RECORD_IRQ 0x20 +#define CS4231_TIMER_IRQ 0x40 +#define CS4231_ALL_IRQS 0x70 +#define CS4231_REC_UNDERRUN 0x08 +#define CS4231_REC_OVERRUN 0x04 +#define CS4231_PLY_OVERRUN 0x02 +#define CS4231_PLY_UNDERRUN 0x01 + +/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ + +#define CS4231_ENABLE_MIC_GAIN 0x20 + +#define CS4231_MIXS_LINE 0x00 +#define CS4231_MIXS_AUX1 0x40 +#define CS4231_MIXS_MIC 0x80 +#define CS4231_MIXS_ALL 0xc0 + +/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ + +#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ +#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ +#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ +#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ +#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ +#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ +#define CS4231_STEREO 0x10 /* stereo mode */ +/* bits 3-1 define frequency divisor */ +#define CS4231_XTAL1 0x00 /* 24.576 crystal */ +#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ + +/* definitions for interface control register - CS4231_IFACE_CTRL */ + +#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ +#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ +#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ +#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ +#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ +#define CS4231_RECORD_ENABLE 0x02 /* record enable */ +#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ + +/* definitions for pin control register - CS4231_PIN_CTRL */ + +#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ +#define CS4231_XCTL1 0x40 /* external control #1 */ +#define CS4231_XCTL0 0x80 /* external control #0 */ + +/* definitions for test and init register - CS4231_TEST_INIT */ + +#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ +#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ + +/* definitions for misc control register - CS4231_MISC_INFO */ + +#define CS4231_MODE2 0x40 /* MODE 2 */ +#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */ +#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */ + +/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */ + +#define CS4231_DACZ 0x01 /* zero DAC when underrun */ +#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */ +#define CS4231_OLB 0x80 /* output level bit */ + +/* definitions for Extended Registers - CS4236+ */ + +#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f)) +#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8) + +#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */ +#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */ +#define CS4236_LEFT_MIC 0x28 /* left MIC volume */ +#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */ +#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */ +#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */ +#define CS4236_LEFT_FM 0x68 /* left FM volume */ +#define CS4236_RIGHT_FM 0x78 /* right FM volume */ +#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */ +#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */ +#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */ +#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */ +#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */ +#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */ +#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */ +#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */ +#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */ +#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ +#define CS4236_VERSION 0x9c /* chip version and ID */ + +#endif /* __SOUND_CS4231_REGS_H */ diff -uNrp linux-2.6.23.orig/include/sound/cs4231.h linux-2.6.23/include/sound/cs4231.h --- linux-2.6.23.orig/include/sound/cs4231.h 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.23/include/sound/cs4231.h 2007-09-03 18:27:20.211443943 +0200 @@ -26,160 +26,7 @@ #include "pcm.h" #include "timer.h" -/* IO ports */ - -#define CS4231P(x) (c_d_c_CS4231##x) - -#define c_d_c_CS4231REGSEL 0 -#define c_d_c_CS4231REG 1 -#define c_d_c_CS4231STATUS 2 -#define c_d_c_CS4231PIO 3 - -/* codec registers */ - -#define CS4231_LEFT_INPUT 0x00 /* left input control */ -#define CS4231_RIGHT_INPUT 0x01 /* right input control */ -#define CS4231_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define CS4231_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define CS4231_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define CS4231_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define CS4231_LEFT_OUTPUT 0x06 /* left output control register */ -#define CS4231_RIGHT_OUTPUT 0x07 /* right output control register */ -#define CS4231_PLAYBK_FORMAT 0x08 /* clock and data format - playback - bits 7-0 MCE */ -#define CS4231_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define CS4231_PIN_CTRL 0x0a /* pin control */ -#define CS4231_TEST_INIT 0x0b /* test and initialization */ -#define CS4231_MISC_INFO 0x0c /* miscellaneaous information */ -#define CS4231_LOOPBACK 0x0d /* loopback control */ -#define CS4231_PLY_UPR_CNT 0x0e /* playback upper base count */ -#define CS4231_PLY_LWR_CNT 0x0f /* playback lower base count */ -#define CS4231_ALT_FEATURE_1 0x10 /* alternate #1 feature enable */ -#define AD1845_AF1_MIC_LEFT 0x10 /* alternate #1 feature + MIC left */ -#define CS4231_ALT_FEATURE_2 0x11 /* alternate #2 feature enable */ -#define AD1845_AF2_MIC_RIGHT 0x11 /* alternate #2 feature + MIC right */ -#define CS4231_LEFT_LINE_IN 0x12 /* left line input control */ -#define CS4231_RIGHT_LINE_IN 0x13 /* right line input control */ -#define CS4231_TIMER_LOW 0x14 /* timer low byte */ -#define CS4231_TIMER_HIGH 0x15 /* timer high byte */ -#define CS4231_LEFT_MIC_INPUT 0x16 /* left MIC input control register (InterWave only) */ -#define AD1845_UPR_FREQ_SEL 0x16 /* upper byte of frequency select */ -#define CS4231_RIGHT_MIC_INPUT 0x17 /* right MIC input control register (InterWave only) */ -#define AD1845_LWR_FREQ_SEL 0x17 /* lower byte of frequency select */ -#define CS4236_EXT_REG 0x17 /* extended register access */ -#define CS4231_IRQ_STATUS 0x18 /* irq status register */ -#define CS4231_LINE_LEFT_OUTPUT 0x19 /* left line output control register (InterWave only) */ -#define CS4231_VERSION 0x19 /* CS4231(A) - version values */ -#define CS4231_MONO_CTRL 0x1a /* mono input/output control */ -#define CS4231_LINE_RIGHT_OUTPUT 0x1b /* right line output control register (InterWave only) */ -#define AD1845_PWR_DOWN 0x1b /* power down control */ -#define CS4235_LEFT_MASTER 0x1b /* left master output control */ -#define CS4231_REC_FORMAT 0x1c /* clock and data format - record - bits 7-0 MCE */ -#define CS4231_PLY_VAR_FREQ 0x1d /* playback variable frequency */ -#define AD1845_CLOCK 0x1d /* crystal clock select and total power down */ -#define CS4235_RIGHT_MASTER 0x1d /* right master output control */ -#define CS4231_REC_UPR_CNT 0x1e /* record upper count */ -#define CS4231_REC_LWR_CNT 0x1f /* record lower count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define CS4231_INIT 0x80 /* CODEC is initializing */ -#define CS4231_MCE 0x40 /* mode change enable */ -#define CS4231_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define CS4231_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for codec irq status */ - -#define CS4231_PLAYBACK_IRQ 0x10 -#define CS4231_RECORD_IRQ 0x20 -#define CS4231_TIMER_IRQ 0x40 -#define CS4231_ALL_IRQS 0x70 -#define CS4231_REC_UNDERRUN 0x08 -#define CS4231_REC_OVERRUN 0x04 -#define CS4231_PLY_OVERRUN 0x02 -#define CS4231_PLY_UNDERRUN 0x01 - -/* definitions for CS4231_LEFT_INPUT and CS4231_RIGHT_INPUT registers */ - -#define CS4231_ENABLE_MIC_GAIN 0x20 - -#define CS4231_MIXS_LINE 0x00 -#define CS4231_MIXS_AUX1 0x40 -#define CS4231_MIXS_MIC 0x80 -#define CS4231_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - CS4231_PLAYBK_FORMAT */ - -#define CS4231_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define CS4231_ALAW_8 0x60 /* 8-bit A-law companded */ -#define CS4231_ULAW_8 0x20 /* 8-bit U-law companded */ -#define CS4231_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define CS4231_LINEAR_16_BIG 0xc0 /* 16-bit twos complement data - big endian */ -#define CS4231_ADPCM_16 0xa0 /* 16-bit ADPCM */ -#define CS4231_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define CS4231_XTAL1 0x00 /* 24.576 crystal */ -#define CS4231_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - CS4231_IFACE_CTRL */ - -#define CS4231_RECORD_PIO 0x80 /* record PIO enable */ -#define CS4231_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define CS4231_CALIB_MODE 0x18 /* calibration mode bits */ -#define CS4231_AUTOCALIB 0x08 /* auto calibrate */ -#define CS4231_SINGLE_DMA 0x04 /* use single DMA channel */ -#define CS4231_RECORD_ENABLE 0x02 /* record enable */ -#define CS4231_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - CS4231_PIN_CTRL */ - -#define CS4231_IRQ_ENABLE 0x02 /* enable IRQ */ -#define CS4231_XCTL1 0x40 /* external control #1 */ -#define CS4231_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - CS4231_TEST_INIT */ - -#define CS4231_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define CS4231_DMA_REQUEST 0x10 /* DMA request in progress */ - -/* definitions for misc control register - CS4231_MISC_INFO */ - -#define CS4231_MODE2 0x40 /* MODE 2 */ -#define CS4231_IW_MODE3 0x6c /* MODE 3 - InterWave enhanced mode */ -#define CS4231_4236_MODE3 0xe0 /* MODE 3 - CS4236+ enhanced mode */ - -/* definitions for alternate feature 1 register - CS4231_ALT_FEATURE_1 */ - -#define CS4231_DACZ 0x01 /* zero DAC when underrun */ -#define CS4231_TIMER_ENABLE 0x40 /* codec timer enable */ -#define CS4231_OLB 0x80 /* output level bit */ - -/* definitions for Extended Registers - CS4236+ */ - -#define CS4236_REG(i23val) (((i23val << 2) & 0x10) | ((i23val >> 4) & 0x0f)) -#define CS4236_I23VAL(reg) ((((reg)&0xf) << 4) | (((reg)&0x10) >> 2) | 0x8) - -#define CS4236_LEFT_LINE 0x08 /* left LINE alternate volume */ -#define CS4236_RIGHT_LINE 0x18 /* right LINE alternate volume */ -#define CS4236_LEFT_MIC 0x28 /* left MIC volume */ -#define CS4236_RIGHT_MIC 0x38 /* right MIC volume */ -#define CS4236_LEFT_MIX_CTRL 0x48 /* synthesis and left input mixer control */ -#define CS4236_RIGHT_MIX_CTRL 0x58 /* right input mixer control */ -#define CS4236_LEFT_FM 0x68 /* left FM volume */ -#define CS4236_RIGHT_FM 0x78 /* right FM volume */ -#define CS4236_LEFT_DSP 0x88 /* left DSP serial port volume */ -#define CS4236_RIGHT_DSP 0x98 /* right DSP serial port volume */ -#define CS4236_RIGHT_LOOPBACK 0xa8 /* right loopback monitor volume */ -#define CS4236_DAC_MUTE 0xb8 /* DAC mute and IFSE enable */ -#define CS4236_ADC_RATE 0xc8 /* indenpendent ADC sample frequency */ -#define CS4236_DAC_RATE 0xd8 /* indenpendent DAC sample frequency */ -#define CS4236_LEFT_MASTER 0xe8 /* left master digital audio volume */ -#define CS4236_RIGHT_MASTER 0xf8 /* right master digital audio volume */ -#define CS4236_LEFT_WAVE 0x0c /* left wavetable serial port volume */ -#define CS4236_RIGHT_WAVE 0x1c /* right wavetable serial port volume */ -#define CS4236_VERSION 0x9c /* chip version and ID */ +#include "cs4231-regs.h" /* defines for codec.mode */

2 1

[alsa-devel] [PATCH] dbri: more cleanups (2nd version)
by Krzysztof Helt 04 Sep '07

04 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch: - removes redundant constant suffices - removes redundant parentheses - removes redundant curly brackets - removes check if a spinlock is locked inside method which is only called with the spinlock locked - moves few functions to the __init section - removes line which appears twice after the previous patch - minor comments improvements Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Takashi, I fixed issues you pointed out. If you want more fixes of coding style, let me know and I'll prepare a separate patch. --- linux-2.6.22/sound/sparc/dbri.c 2007-09-03 20:51:40.000000000 +0200 +++ linux-2.6.23/sound/sparc/dbri.c 2007-09-03 20:53:27.000000000 +0200 @@ -233,12 +233,12 @@ static struct { ****************************************************************************/ /* DBRI main registers */ -#define REG0 0x00UL /* Status and Control */ -#define REG1 0x04UL /* Mode and Interrupt */ -#define REG2 0x08UL /* Parallel IO */ -#define REG3 0x0cUL /* Test */ -#define REG8 0x20UL /* Command Queue Pointer */ -#define REG9 0x24UL /* Interrupt Queue Pointer */ +#define REG0 0x00 /* Status and Control */ +#define REG1 0x04 /* Mode and Interrupt */ +#define REG2 0x08 /* Parallel IO */ +#define REG3 0x0c /* Test */ +#define REG8 0x20 /* Command Queue Pointer */ +#define REG9 0x24 /* Interrupt Queue Pointer */ #define DBRI_NO_CMDS 64 #define DBRI_INT_BLK 64 @@ -565,7 +565,7 @@ struct snd_dbri { /* Translate the ALSA direction into the array index */ #define DBRI_STREAMNO(substream) \ (substream->stream == \ - SNDRV_PCM_STREAM_PLAYBACK? DBRI_PLAY: DBRI_REC) + SNDRV_PCM_STREAM_PLAYBACK ? DBRI_PLAY: DBRI_REC) /* Return a pointer to dbri_streaminfo */ #define DBRI_STREAM(dbri, substream) \ @@ -611,8 +611,8 @@ The list is terminated with a WAIT comma CPU interrupt to signal completion. Since the DBRI can run in parallel with the CPU, several means of -synchronization present themselves. The method implemented here is only -use of the dbri_cmdwait() to wait for execution of batch of sent commands. +synchronization present themselves. The method implemented here uses +the dbri_cmdwait() to wait for execution of batch of sent commands. A circular command buffer is used here. A new command is being added while another can be executed. The scheme works by adding two WAIT commands @@ -648,15 +648,14 @@ static void dbri_cmdwait(struct snd_dbri } spin_unlock_irqrestore(&dbri->lock, flags); - if (maxloops == 0) { + if (maxloops == 0) printk(KERN_ERR "DBRI: Chip never completed command buffer\n"); - } else { + else dprintk(D_CMD, "Chip completed command buffer (%d)\n", MAXLOOPS - maxloops - 1); - } } /* - * Lock the command queue and returns pointer to a space for len cmd words + * Lock the command queue and return pointer to space for len cmd words * It locks the cmdlock spinlock. */ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) @@ -749,7 +748,7 @@ static void dbri_reset(struct snd_dbri * } /* Lock must not be held before calling this */ -static void dbri_initialize(struct snd_dbri *dbri) +static void __init dbri_initialize(struct snd_dbri *dbri) { s32 *cmd; u32 dma_addr; @@ -804,7 +803,7 @@ list ordering, among other things. The here interface closely with the transmit and receive interrupt code. */ -static int pipe_active(struct snd_dbri *dbri, int pipe) +static inline int pipe_active(struct snd_dbri *dbri, int pipe) { return ((pipe >= 0) && (dbri->pipes[pipe].desc != -1)); } @@ -1148,6 +1147,7 @@ static int setup_descs(struct snd_dbri * if (!dbri->dma->desc[desc].ba) break; } + if (desc == DBRI_NO_DESCS) { printk(KERN_ERR "DBRI: setup_descs: No descriptors\n"); return -1; @@ -1308,7 +1308,7 @@ to the DBRI via the CHI interface and fe * Lock must not be held before calling it. */ -static void cs4215_setup_pipes(struct snd_dbri *dbri) +static __init void cs4215_setup_pipes(struct snd_dbri *dbri) { unsigned long flags; @@ -1341,7 +1341,7 @@ static void cs4215_setup_pipes(struct sn dbri_cmdwait(dbri); } -static int cs4215_init_data(struct cs4215 *mm) +static __init int cs4215_init_data(struct cs4215 *mm) { /* * No action, memory resetting only. @@ -1633,7 +1633,7 @@ static int cs4215_prepare(struct snd_dbr /* * */ -static int cs4215_init(struct snd_dbri *dbri) +static __init int cs4215_init(struct snd_dbri *dbri) { u32 reg2 = sbus_readl(dbri->regs + REG2); dprintk(D_MM, "cs4215_init: reg2=0x%x\n", reg2); @@ -1771,13 +1771,10 @@ static void xmit_descs(struct snd_dbri * static void transmission_complete_intr(struct snd_dbri *dbri, int pipe) { - struct dbri_streaminfo *info; - int td; + struct dbri_streaminfo *info = &dbri->stream_info[DBRI_PLAY]; + int td = dbri->pipes[pipe].desc; int status; - info = &dbri->stream_info[DBRI_PLAY]; - - td = dbri->pipes[pipe].desc; while (td >= 0) { if (td >= DBRI_NO_DESCS) { printk(KERN_ERR "DBRI: invalid td on pipe %d\n", pipe); @@ -1798,12 +1795,9 @@ static void transmission_complete_intr(s } /* Notify ALSA */ - if (spin_is_locked(&dbri->lock)) { - spin_unlock(&dbri->lock); - snd_pcm_period_elapsed(info->substream); - spin_lock(&dbri->lock); - } else - snd_pcm_period_elapsed(info->substream); + spin_unlock(&dbri->lock); + snd_pcm_period_elapsed(info->substream); + spin_lock(&dbri->lock); } static void reception_complete_intr(struct snd_dbri *dbri, int pipe) @@ -1830,12 +1824,9 @@ static void reception_complete_intr(stru rd, DBRI_RD_STATUS(status), DBRI_RD_CNT(status)); /* Notify ALSA */ - if (spin_is_locked(&dbri->lock)) { - spin_unlock(&dbri->lock); - snd_pcm_period_elapsed(info->substream); - spin_lock(&dbri->lock); - } else - snd_pcm_period_elapsed(info->substream); + spin_unlock(&dbri->lock); + snd_pcm_period_elapsed(info->substream); + spin_lock(&dbri->lock); } static void dbri_process_one_interrupt(struct snd_dbri *dbri, int x) @@ -1986,10 +1977,10 @@ static irqreturn_t snd_dbri_interrupt(in PCM Interface ****************************************************************************/ static struct snd_pcm_hardware snd_dbri_pcm_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_U8 | @@ -1999,7 +1990,7 @@ static struct snd_pcm_hardware snd_dbri_ .rate_max = 48000, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = (64 * 1024), + .buffer_bytes_max = 64 * 1024, .period_bytes_min = 1, .period_bytes_max = DBRI_TD_MAXCNT, .periods_min = 1, @@ -2266,11 +2257,10 @@ static int snd_cs4215_info_volume(struct uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - if (kcontrol->private_value == DBRI_PLAY) { + if (kcontrol->private_value == DBRI_PLAY) uinfo->value.integer.max = DBRI_MAX_VOLUME; - } else { + else uinfo->value.integer.max = DBRI_MAX_GAIN; - } return 0; } @@ -2304,7 +2294,7 @@ static int snd_cs4215_put_volume(struct info->right_gain = ucontrol->value.integer.value[1]; changed = 1; } - if (changed == 1) { + if (changed) { /* First mute outputs, and wait 1/8000 sec (125 us) * to make sure this takes. This avoids clicking noises. */ @@ -2443,8 +2433,9 @@ static int __init snd_dbri_mixer(struct strcpy(card->mixername, card->shortname); for (idx = 0; idx < ARRAY_SIZE(dbri_controls); idx++) { - if ((err = snd_ctl_add(card, - snd_ctl_new1(&dbri_controls[idx], dbri))) < 0) + err = snd_ctl_add(card, + snd_ctl_new1(&dbri_controls[idx], dbri)); + if (err < 0) return err; } @@ -2485,8 +2476,8 @@ static void dbri_debug_read(struct snd_i "Pipe %d: %s SDP=0x%x desc=%d, " "len=%d next %d\n", pipe, - ((pptr->sdp & D_SDP_TO_SER) ? "output" : - "input"), + (pptr->sdp & D_SDP_TO_SER) ? "output" : + "input", pptr->sdp, pptr->desc, pptr->length, pptr->nextpipe); } @@ -2502,7 +2493,7 @@ void snd_dbri_proc(struct snd_dbri *dbri snd_info_set_text_ops(entry, dbri, dbri_regs_read); #ifdef DBRI_DEBUG - if (! snd_card_proc_new(dbri->card, "debug", &entry)) { + if (!snd_card_proc_new(dbri->card, "debug", &entry)) { snd_info_set_text_ops(entry, dbri, dbri_debug_read); entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ } @@ -2633,11 +2624,12 @@ static int __init dbri_attach(int prom_n } dbri = card->private_data; - if ((err = snd_dbri_pcm(dbri)) < 0) + err = snd_dbri_pcm(dbri); + if (err < 0) goto _err; - if ((err = snd_dbri_mixer(dbri)) < 0) - if ((err = snd_dbri_mixer(dbri)) < 0) + err = snd_dbri_mixer(dbri); + if (err < 0) goto _err; /* /proc file handling */

2 1

[alsa-devel] [PATCH] sun-cs4231: code improvements (2nd version)
by Krzysztof Helt 04 Sep '07

04 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch does some code improvements to make driver (both code and binary) shorter. It also make use of card->private_data pointer to store chip information. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Do I understand correctly, that lines: if ((err = function()) < 0) are passe now? They are so abudant in the alsa code, that I thought it is some kind of template code. I can clean up drivers according to the checkpatch.pl script if it helps (above problem, long lines, etc). diff -urp linux-2.6.23.old/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-2.6.23.old/sound/sparc/cs4231.c 2007-09-02 14:02:32.000000000 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-03 15:56:03.000000000 +0200 @@ -30,17 +30,11 @@ #ifdef CONFIG_SBUS #define SBUS_SUPPORT -#endif - -#ifdef SBUS_SUPPORT #include <asm/sbus.h> #endif #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64) #define EBUS_SUPPORT -#endif - -#ifdef EBUS_SUPPORT #include <linux/pci.h> #include <asm/ebus.h> #endif @@ -339,7 +333,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, }; @@ -389,116 +383,85 @@ static unsigned char snd_cs4231_original static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr) { #ifdef EBUS_SUPPORT - if (cp->flags & CS4231_FLAG_EBUS) { + if (cp->flags & CS4231_FLAG_EBUS) return readb(reg_addr); - } else { + else #endif #ifdef SBUS_SUPPORT return sbus_readb(reg_addr); #endif -#ifdef EBUS_SUPPORT - } -#endif } static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, void __iomem *reg_addr) { #ifdef EBUS_SUPPORT - if (cp->flags & CS4231_FLAG_EBUS) { + if (cp->flags & CS4231_FLAG_EBUS) return writeb(val, reg_addr); - } else { + else #endif #ifdef SBUS_SUPPORT return sbus_writeb(val, reg_addr); #endif -#ifdef EBUS_SUPPORT - } -#endif } /* * Basic I/O functions */ -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, - unsigned char mask, unsigned char value) +static void snd_cs4231_ready(struct snd_cs4231 *chip) { int timeout; - unsigned char tmp; for (timeout = 250; timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); timeout--) udelay(100); -#ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - if (chip->calibrate_mute) { - chip->image[reg] &= mask; - chip->image[reg] |= value; - } else { - __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); - mb(); - tmp = (chip->image[reg] & mask) | value; - __cs4231_writeb(chip, tmp, CS4231P(chip, REG)); - chip->image[reg] = tmp; - mb(); - } } static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); + snd_printdd("out: auto calibration time out - reg = 0x%x, " + "value = 0x%x\n", + reg, value); #endif __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); + wmb(); __cs4231_writeb(chip, value, CS4231P(chip, REG)); mb(); } -static void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +static inline void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, + unsigned char mask, unsigned char value) { - int timeout; + unsigned char tmp = (chip->image[reg] & mask) | value; - for (timeout = 250; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); -#ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); - __cs4231_writeb(chip, value, CS4231P(chip, REG)); + chip->image[reg] = tmp; + if (!chip->calibrate_mute) + snd_cs4231_dout(chip, reg, tmp); +} + +static void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, + unsigned char value) +{ + snd_cs4231_dout(chip, reg, value); chip->image[reg] = value; mb(); } static unsigned char snd_cs4231_in(struct snd_cs4231 *chip, unsigned char reg) { - int timeout; - unsigned char ret; - - for (timeout = 250; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("in: auto calibration time out - reg = 0x%x\n", reg); + snd_printdd("in: auto calibration time out - reg = 0x%x\n", + reg); #endif __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); mb(); - ret = __cs4231_readb(chip, CS4231P(chip, REG)); - return ret; + return __cs4231_readb(chip, CS4231P(chip, REG)); } /* @@ -517,7 +480,7 @@ static void snd_cs4231_busy_wait(struct for (timeout = 500; timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); timeout--) - udelay(1000); + msleep(1); } static void snd_cs4231_mce_up(struct snd_cs4231 *chip) @@ -526,8 +489,7 @@ static void snd_cs4231_mce_up(struct snd int timeout; spin_lock_irqsave(&chip->lock, flags); - for (timeout = 250; timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); timeout--) - udelay(100); + snd_cs4231_ready(chip); #ifdef CONFIG_SND_DEBUG if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) snd_printdd("mce_up - auto calibration time out (0)\n"); @@ -565,8 +527,8 @@ static void snd_cs4231_mce_down(struct s /* calibration process */ - for (timeout = 500; timeout > 0 && (snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0; timeout--) - udelay(100); + snd_cs4231_ready(chip); + snd_cs4231_ready(chip); if ((snd_cs4231_in(chip, CS4231_TEST_INIT) & CS4231_CALIB_IN_PROGRESS) == 0) { snd_printd("cs4231_mce_down - auto calibration time out (1)\n"); spin_unlock_irqrestore(&chip->lock, flags); @@ -1058,11 +1020,6 @@ static int snd_cs4231_playback_hw_params return 0; } -static int snd_cs4231_playback_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - static int snd_cs4231_playback_prepare(struct snd_pcm_substream *substream) { struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); @@ -1100,11 +1057,6 @@ static int snd_cs4231_capture_hw_params( return 0; } -static int snd_cs4231_capture_hw_free(struct snd_pcm_substream *substream) -{ - return snd_pcm_lib_free_pages(substream); -} - static int snd_cs4231_capture_prepare(struct snd_pcm_substream *substream) { struct snd_cs4231 *chip = snd_pcm_substream_chip(substream); @@ -1182,10 +1134,6 @@ static snd_pcm_uframes_t snd_cs4231_capt return bytes_to_frames(substream->runtime, ptr); } -/* - - */ - static int __init snd_cs4231_probe(struct snd_cs4231 *chip) { unsigned long flags; @@ -1356,7 +1304,7 @@ static struct snd_pcm_ops snd_cs4231_pla .close = snd_cs4231_playback_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_cs4231_playback_hw_params, - .hw_free = snd_cs4231_playback_hw_free, + .hw_free = snd_pcm_lib_free_pages, .prepare = snd_cs4231_playback_prepare, .trigger = snd_cs4231_trigger, .pointer = snd_cs4231_playback_pointer, @@ -1367,18 +1315,20 @@ static struct snd_pcm_ops snd_cs4231_cap .close = snd_cs4231_capture_close, .ioctl = snd_pcm_lib_ioctl, .hw_params = snd_cs4231_capture_hw_params, - .hw_free = snd_cs4231_capture_hw_free, + .hw_free = snd_pcm_lib_free_pages, .prepare = snd_cs4231_capture_prepare, .trigger = snd_cs4231_trigger, .pointer = snd_cs4231_capture_pointer, }; -static int __init snd_cs4231_pcm(struct snd_cs4231 *chip) +static int __init snd_cs4231_pcm(struct snd_card *card) { + struct snd_cs4231 *chip = card->private_data; struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "CS4231", 0, 1, 1, &pcm)) < 0) + err = snd_pcm_new(card, "CS4231", 0, 1, 1, &pcm); + if (err < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); @@ -1396,8 +1346,9 @@ static int __init snd_cs4231_pcm(struct return 0; } -static int __init snd_cs4231_timer(struct snd_cs4231 *chip) +static int __init snd_cs4231_timer(struct snd_card *card) { + struct snd_cs4231 *chip = card->private_data; struct snd_timer *timer; struct snd_timer_id tid; int err; @@ -1405,10 +1356,11 @@ static int __init snd_cs4231_timer(struc /* Timer initialization */ tid.dev_class = SNDRV_TIMER_CLASS_CARD; tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; - tid.card = chip->card->number; + tid.card = card->number; tid.device = 0; tid.subdevice = 0; - if ((err = snd_timer_new(chip->card, "CS4231", &tid, &timer)) < 0) + err = snd_timer_new(card, "CS4231", &tid, &timer); + if (err < 0) return err; strcpy(timer->name, "CS4231"); timer->private_data = chip; @@ -1428,9 +1380,7 @@ static int snd_cs4231_info_mux(struct sn static char *texts[4] = { "Line", "CD", "Mic", "Mix" }; - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - snd_assert(chip->card != NULL, return -EINVAL); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 2; uinfo->value.enumerated.items = 4; @@ -1670,21 +1620,19 @@ CS4231_SINGLE("Line Out Switch", 0, CS42 CS4231_SINGLE("Headphone Out Switch", 0, CS4231_PIN_CTRL, 7, 1, 1) }; -static int __init snd_cs4231_mixer(struct snd_cs4231 *chip) +static int __init snd_cs4231_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_cs4231 *chip = card->private_data; int err, idx; snd_assert(chip != NULL && chip->pcm != NULL, return -EINVAL); - card = chip->card; - 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) + err = snd_ctl_add(card, + snd_ctl_new1(&snd_cs4231_controls[idx], chip)); + if (err < 0) return err; } return 0; @@ -1695,6 +1643,7 @@ static int dev; static int __init cs4231_attach_begin(struct snd_card **rcard) { struct snd_card *card; + struct snd_cs4231 *chip; *rcard = NULL; @@ -1706,31 +1655,40 @@ static int __init cs4231_attach_begin(st return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_cs4231)); if (card == NULL) return -ENOMEM; strcpy(card->driver, "CS4231"); strcpy(card->shortname, "Sun CS4231"); + chip = card->private_data; + chip->card = card; + *rcard = card; return 0; } -static int __init cs4231_attach_finish(struct snd_card *card, struct snd_cs4231 *chip) +static int __init cs4231_attach_finish(struct snd_card *card) { + struct snd_cs4231 *chip = card->private_data; int err; - if ((err = snd_cs4231_pcm(chip)) < 0) + err = snd_cs4231_pcm(card); + if (err < 0) goto out_err; - if ((err = snd_cs4231_mixer(chip)) < 0) + err = snd_cs4231_mixer(card); + if (err < 0) goto out_err; - if ((err = snd_cs4231_timer(chip)) < 0) + err = snd_cs4231_timer(card); + if (err < 0) goto out_err; - if ((err = snd_card_register(card)) < 0) + err = snd_card_register(card); + if (err < 0) goto out_err; chip->next = cs4231_list; @@ -1925,23 +1883,16 @@ static struct snd_device_ops snd_cs4231_ static int __init snd_cs4231_sbus_create(struct snd_card *card, struct sbus_dev *sdev, - int dev, - struct snd_cs4231 **rchip) + int dev) { - struct snd_cs4231 *chip; + struct snd_cs4231 *chip = card->private_data; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - spin_lock_init(&chip->lock); spin_lock_init(&chip->c_dma.sbus_info.lock); spin_lock_init(&chip->p_dma.sbus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - chip->card = card; chip->dev_u.sdev = sdev; chip->regs_size = sdev->reg_addrs[0].reg_size; memcpy(&chip->image, &snd_cs4231_original_image, @@ -1992,14 +1943,12 @@ static int __init snd_cs4231_sbus_create return err; } - *rchip = chip; return 0; } static int __init cs4231_sbus_attach(struct sbus_dev *sdev) { struct resource *rp = &sdev->resource[0]; - struct snd_cs4231 *cp; struct snd_card *card; int err; @@ -2013,12 +1962,13 @@ static int __init cs4231_sbus_attach(str (unsigned long long)rp->start, sdev->irqs[0]); - if ((err = snd_cs4231_sbus_create(card, sdev, dev, &cp)) < 0) { + err = snd_cs4231_sbus_create(card, sdev, dev); + if (err < 0) { snd_card_free(card); return err; } - return cs4231_attach_finish(card, cp); + return cs4231_attach_finish(card); } #endif @@ -2105,24 +2055,17 @@ static struct snd_device_ops snd_cs4231_ static int __init snd_cs4231_ebus_create(struct snd_card *card, struct linux_ebus_device *edev, - int dev, - struct snd_cs4231 **rchip) + int dev) { - struct snd_cs4231 *chip; + struct snd_cs4231 *chip = card->private_data; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - spin_lock_init(&chip->lock); spin_lock_init(&chip->c_dma.ebus_info.lock); spin_lock_init(&chip->p_dma.ebus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); chip->flags |= CS4231_FLAG_EBUS; - chip->card = card; chip->dev_u.pdev = edev->bus->self; memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); @@ -2192,14 +2135,12 @@ static int __init snd_cs4231_ebus_create return err; } - *rchip = chip; return 0; } static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) { struct snd_card *card; - struct snd_cs4231 *chip; int err; err = cs4231_attach_begin(&card); @@ -2211,12 +2152,13 @@ static int __init cs4231_ebus_attach(str edev->resource[0].start, edev->irqs[0]); - if ((err = snd_cs4231_ebus_create(card, edev, dev, &chip)) < 0) { + err = snd_cs4231_ebus_create(card, edev, dev); + if (err < 0) { snd_card_free(card); return err; } - return cs4231_attach_finish(card, chip); + return cs4231_attach_finish(card); } #endif

2 1

[alsa-devel] How to control the route in smartphone
by Leijin Tang 04 Sep '07

04 Sep '07

Hi all, When the audio codec is used in the smartphone, we get the problem how to define and control the route in the alsa driver. The scenario is only in the phone device. The application need switch the routes based on the different scene. Example: the coedec is wm9713L. the in port: LINEL，LINER，MONOIN，MIC1，MIC2, ac97_in the out port: HEADSET, SPEAKER, RECEIVER, ac97_out When the phone is coming, we need chose the route LINE to receiver. After the phone and we playing the mp3, We should switch the route ac97_in to speaker on. This scenario doesn’t appear in the PC. So how to carry out the function in the alsa architecture is so difficulty. The router names how to define, the alsa-lib how to do the control. By now, there has no design to the phone device in the alsa. Does anybody have some ideas? We can do some discussion and resolve the issue.

1 0

[alsa-devel] ALSA bug 0002948
by Brian Hinz 04 Sep '07

04 Sep '07

Hi, I'll apologize up front if this post seems a little incoherent, I've been trying to find a solution to this bug for a few months now and this is sort of a collection of observations that I've made and I'm hoping someone might read this and a light will go on... The problem is no sound from a stac9200 on various gateway models, although it's possible to get sound from the mic port, never from the hp or internal stereo speakers. I recently used a windows utility called "regmon" to log the registry events that occurred during a jack cycle on the hp port. I expected to see something like this: ... stsystra.exe:2380 SetValue HKLM\Software\Sigmatel\GlobalState\STSysTray\1.0\JackPresenced SUCCESS 0x0 ... indicating an event on port D, but instead got: ... SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompRatio SUCCESS 0xC8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompAttack SUCCESS 0x26B6 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompRelease SUCCESS 0x26B6 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompThreshDb SUCCESS 0xFFFFFDA8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\MakeupGainDB SUCCESS 0xC8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateRatio SUCCESS 0x258 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateAttack SUCCESS 0x2706 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateRelease SUCCESS 0x2706 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateThreshDb SUCCESS 0xFFFFDCD8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\Flags SUCCESS 0x9 ... SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompRatio SUCCESS 0xE74 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompAttack SUCCESS 0x26AC SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompRelease SUCCESS 0x26AC SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompThreshDb SUCCESS 0x0 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\MakeupGainDB SUCCESS 0x0 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateRatio SUCCESS 0x64 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateAttack SUCCESS 0x2328 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateRelease SUCCESS 0x2328 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateThreshDb SUCCESS 0xFFFFDCD8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\Flags SUCCESS 0xB ... and upon removal: ... SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompRatio SUCCESS 0xC8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompAttack SUCCESS 0x26B6 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompRelease SUCCESS 0x26B6 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\CompThreshDb SUCCESS 0xFFFFFDA8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\MakeupGainDB SUCCESS 0xC8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateRatio SUCCESS 0x258 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateAttack SUCCESS 0x2706 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateRelease SUCCESS 0x2706 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\GateThreshDb SUCCESS 0xFFFFDCD8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Compressor\Flags SUCCESS 0x8 ... SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompRatio SUCCESS 0xE74 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompAttack SUCCESS 0x26AC SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompRelease SUCCESS 0x26AC SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\CompThreshDb SUCCESS 0x0 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\MakeupGainDB SUCCESS 0x0 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateRatio SUCCESS 0x64 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateAttack SUCCESS 0x2328 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateRelease SUCCESS 0x2328 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\GateThreshDb SUCCESS 0xFFFFDCD8 SetValue HKLM\System\CurrentControlSet\Control\Class\{4D36E96C-E325-11CE-BFC1-08002BE10318}\0005\Settings\filter\Rear\Presets\_Initial\Render\Limiter\Flags SUCCESS 0xA ... Notice how the "Flags" key changes? I don't understand much of this, but it seems like the output is being manipulated via digital post-processing rather than re-tasking of the analog pin widgets. I had also been trying to make sense of the following line in the windows driver file: InitVerbs: 03, a0, 45, 01; which I'd guessed earlier could be interpreted as verb 0x3a0 and payload of 0x4501, (amplifier gain/mute, set input amp, port B(?), unmute, gain step 1) which actually makes some sense in this context, since if the flags key is referencing pin widgets the input mux should be unmuted if it's to be used as the limiter. So if I'm actually on the right track here, how would something like this be implemented in ALSA? Via /etc/asoundrc? I tried setting the digital mixer as the default pcm device and now the gnome-volume-applet shows the device as being muted, and is unable to unmute it. Nothing is muted in alsamixer though, so I'm guessing that I didn't do this correctly in any event. Another potentially interesting observation is that in the windows sound control options there is both a master volume control and a "Wave" volume control which seem to be bound together somehow even though they are independent controls. ie: the wave control appears to have the exact same function as the master volume. Lastly, the system beep (which for what it's worth "sounds digital" to my untrained ears), works prior to the snd-hda-intel module being loaded, but not afterwards, even when unloaded. Is there any debugging code aready in alsa that would produce a copy of /proc/asound/card0/codec#0 prior to any codec initialization? If anyone can make some suggestions I'd be quite grateful. Also, if there is any specific debugging results you'd like to see let me know and I'll append them to the bug report. Thanks, -brian

2 2

[alsa-devel] ALSA lib troubles with xine
by Markus Korber 03 Sep '07

03 Sep '07

Hi, I've got problems using ALSA (hg) and xine-lib (1.1.7). I'm trying to use fbxine for playback but ALSA debug logging gives me: ,---- | audio_alsa_out: Audio Device name = default | audio_alsa_out: Number of channels = 2 | Buffer size range from 4096 to 0 <- !!! | Period size range from 0 to 0 <- !!! | Buffer time size 44100 | To choose buffer_size = 4096 | To choose period_size = 512 | was set period_size = 0 | was set buffer_size = 4096 | audio_alsa_out:open pause_resume=1 | ACCESS: RW_INTERLEAVED | FORMAT: S16_LE | SUBFORMAT: STD | SAMPLE_BITS: 16 | FRAME_BITS: 32 | CHANNELS: 2 | RATE: 44100 | PERIOD_TIME: (46439 46440) | PERIOD_SIZE: 2048 | PERIOD_BYTES: 8192 | PERIODS: 2 | BUFFER_TIME: (92879 92880) | BUFFER_SIZE: 4096 | BUFFER_BYTES: 16384 | TICK_TIME: 10000 | audio_alsa_out: Unable to set swparams: Invalid argument `---- So buffer and period sizes seem to be wrong and therefore I get no sound. (audio_alsa_out logging from [1].) Next I've tried to run speaker-test and the ouptut look good: ,---- | [root@(none) ~]# speaker-test | | speaker-test 1.0.15rc1 | | 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 4096 to 4096 | Period size range from 2048 to 2048 | Using max buffer size 4096 | Periods = 4 | was set period_size = 2048 | was set buffer_size = 4096 | 0 - Front Left `---- Both applications are linked against the same ALSA library (dynamic or static linkage doesn't make a difference - speaker-test always works and xine always fails). And both applications use the same instructions (snd_pcm_hw_params_get_period_size_min/max()) AFAICS. Any ideas what I could try next? Thanks. Toolchain: buildroot based arm gcc 4.2.1 with uclibc 0.9.29. However, other applications (mplayer, aplay, alsaplayer) work as well, so I guess it's not toolchain related. [1] http://xine.cvs.sourceforge.net/xine/xine-lib/src/audio_out/audio_alsa_out.c Regards, Markus Korber

4 10

[alsa-devel] [PATCH] STAC92xx : add support for missing features
by Maxim Levitsky 03 Sep '07

03 Sep '07

Hi, As promised, those are patches that extend patch_sigmatel.c driver to add missing functionality They all are runtime tested with my STAC9227, and datasheet tested with others Note that I did my best to ensure that even if a codec doesn't support a feature (and I don't know sbout that), maximum impact will be a new, non-working mixer control STAC927x are almost exactly the same as STAC922x, but have DAC4 1) Center/LFE swap : It can be sometimes useful to swap those channels, since they are on same line. I add that control only in case Center/LFE DAC is there, and supports L/R swap, it should be very safe. 2) Analog loopback: Added for 922x, 927x, and for 9204/5 which has a different verb, that offers same analog loopback 3) Volume Knob: Added for 9204/5, 9220/1/3, 922x 92-7x I made it a master volume 4) Fixed support for codecs that have 2 or 3 ADC: I added separate mixer controls for all 2/3 ADCs, and made PCM capture device sub-streams count to depend on actual number of ADCs I am looking for your comments, Best regards, Maxim Levitsky

3 22

[alsa-devel] firmware compilation
by paul blakeley 03 Sep '07

03 Sep '07

How do I know if I need to download and then compile the firmware source code? Cant find anything in the docs that details this.

2 1

[alsa-devel] [PATCH] sun-cs4231: code improvements
by Krzysztof Helt 03 Sep '07

03 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch does some code improvements to make driver (both code and binary) shorter. It also make use of card->private_data pointer to store chip information. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-2.6.23.old/sound/sparc/cs4231.c linux-2.6.23/sound/sparc/cs4231.c --- linux-2.6.23.old/sound/sparc/cs4231.c 2007-09-02 14:02:32.000000000 +0200 +++ linux-2.6.23/sound/sparc/cs4231.c 2007-09-02 19:11:28.000000000 +0200 @@ -30,17 +30,11 @@ #ifdef CONFIG_SBUS #define SBUS_SUPPORT -#endif - -#ifdef SBUS_SUPPORT #include <asm/sbus.h> #endif #if defined(CONFIG_PCI) && defined(CONFIG_SPARC64) #define EBUS_SUPPORT -#endif - -#ifdef EBUS_SUPPORT #include <linux/pci.h> #include <asm/ebus.h> #endif @@ -339,7 +333,7 @@ static unsigned int rates[14] = { }; static struct snd_pcm_hw_constraint_list hw_constraints_rates = { - .count = 14, + .count = ARRAY_SIZE(rates), .list = rates, }; @@ -389,42 +383,34 @@ static unsigned char snd_cs4231_original static u8 __cs4231_readb(struct snd_cs4231 *cp, void __iomem *reg_addr) { #ifdef EBUS_SUPPORT - if (cp->flags & CS4231_FLAG_EBUS) { + if (cp->flags & CS4231_FLAG_EBUS) return readb(reg_addr); - } else { + else #endif #ifdef SBUS_SUPPORT return sbus_readb(reg_addr); #endif -#ifdef EBUS_SUPPORT - } -#endif } static void __cs4231_writeb(struct snd_cs4231 *cp, u8 val, void __iomem *reg_addr) { #ifdef EBUS_SUPPORT - if (cp->flags & CS4231_FLAG_EBUS) { + if (cp->flags & CS4231_FLAG_EBUS) return writeb(val, reg_addr); - } else { + else #endif #ifdef SBUS_SUPPORT return sbus_writeb(val, reg_addr); #endif -#ifdef EBUS_SUPPORT - } -#endif } /* * Basic I/O functions */ -static void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, - unsigned char mask, unsigned char value) +static void snd_cs4231_ready(struct snd_cs4231 *chip) { int timeout; - unsigned char tmp; for (timeout = 250; timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); @@ -434,50 +420,30 @@ static void snd_cs4231_outm(struct snd_c if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) snd_printdd("outm: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); #endif - if (chip->calibrate_mute) { - chip->image[reg] &= mask; - chip->image[reg] |= value; - } else { - __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); - mb(); - tmp = (chip->image[reg] & mask) | value; - __cs4231_writeb(chip, tmp, CS4231P(chip, REG)); - chip->image[reg] = tmp; - mb(); - } } static void snd_cs4231_dout(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) { - int timeout; - - for (timeout = 250; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); -#ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif + snd_cs4231_ready(chip); __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); + wmb(); __cs4231_writeb(chip, value, CS4231P(chip, REG)); mb(); } -static void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +static inline void snd_cs4231_outm(struct snd_cs4231 *chip, unsigned char reg, + unsigned char mask, unsigned char value) { - int timeout; + unsigned char tmp = (chip->image[reg] & mask) | value; - for (timeout = 250; - timeout > 0 && (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT); - timeout--) - udelay(100); -#ifdef CONFIG_SND_DEBUG - if (__cs4231_readb(chip, CS4231P(chip, REGSEL)) & CS4231_INIT) - snd_printdd("out: auto calibration time out - reg = 0x%x, value = 0x%x\n", reg, value); -#endif - __cs4231_writeb(chip, chip->mce_bit | reg, CS4231P(chip, REGSEL)); - __cs4231_writeb(chip, value, CS4231P(chip, REG)); + chip->image[reg] = tmp; + if (!chip->calibrate_mute) + snd_cs4231_dout(chip, reg, tmp); +} + +static void snd_cs4231_out(struct snd_cs4231 *chip, unsigned char reg, unsigned char value) +{ + snd_cs4231_dout(chip, reg, value); chip->image[reg] = value; mb(); } @@ -1373,12 +1339,13 @@ static struct snd_pcm_ops snd_cs4231_cap .pointer = snd_cs4231_capture_pointer, }; -static int __init snd_cs4231_pcm(struct snd_cs4231 *chip) +static int __init snd_cs4231_pcm(struct snd_card *card) { + struct snd_cs4231 *chip = card->private_data; struct snd_pcm *pcm; int err; - if ((err = snd_pcm_new(chip->card, "CS4231", 0, 1, 1, &pcm)) < 0) + if ((err = snd_pcm_new(card, "CS4231", 0, 1, 1, &pcm)) < 0) return err; snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, &snd_cs4231_playback_ops); @@ -1396,8 +1363,9 @@ static int __init snd_cs4231_pcm(struct return 0; } -static int __init snd_cs4231_timer(struct snd_cs4231 *chip) +static int __init snd_cs4231_timer(struct snd_card *card) { + struct snd_cs4231 *chip = card->private_data; struct snd_timer *timer; struct snd_timer_id tid; int err; @@ -1405,10 +1373,10 @@ static int __init snd_cs4231_timer(struc /* Timer initialization */ tid.dev_class = SNDRV_TIMER_CLASS_CARD; tid.dev_sclass = SNDRV_TIMER_SCLASS_NONE; - tid.card = chip->card->number; + tid.card = card->number; tid.device = 0; tid.subdevice = 0; - if ((err = snd_timer_new(chip->card, "CS4231", &tid, &timer)) < 0) + if ((err = snd_timer_new(card, "CS4231", &tid, &timer)) < 0) return err; strcpy(timer->name, "CS4231"); timer->private_data = chip; @@ -1428,9 +1396,7 @@ static int snd_cs4231_info_mux(struct sn static char *texts[4] = { "Line", "CD", "Mic", "Mix" }; - struct snd_cs4231 *chip = snd_kcontrol_chip(kcontrol); - snd_assert(chip->card != NULL, return -EINVAL); uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = 2; uinfo->value.enumerated.items = 4; @@ -1670,15 +1636,13 @@ CS4231_SINGLE("Line Out Switch", 0, CS42 CS4231_SINGLE("Headphone Out Switch", 0, CS4231_PIN_CTRL, 7, 1, 1) }; -static int __init snd_cs4231_mixer(struct snd_cs4231 *chip) +static int __init snd_cs4231_mixer(struct snd_card *card) { - struct snd_card *card; + struct snd_cs4231 *chip = card->private_data; int err, idx; snd_assert(chip != NULL && chip->pcm != NULL, return -EINVAL); - card = chip->card; - strcpy(card->mixername, chip->pcm->name); for (idx = 0; idx < ARRAY_SIZE(snd_cs4231_controls); idx++) { @@ -1695,6 +1659,7 @@ static int dev; static int __init cs4231_attach_begin(struct snd_card **rcard) { struct snd_card *card; + struct snd_cs4231 *chip; *rcard = NULL; @@ -1706,28 +1671,33 @@ static int __init cs4231_attach_begin(st return -ENOENT; } - card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + card = snd_card_new(index[dev], id[dev], THIS_MODULE, + sizeof(struct snd_cs4231)); if (card == NULL) return -ENOMEM; strcpy(card->driver, "CS4231"); strcpy(card->shortname, "Sun CS4231"); + + chip = card->private_data; + chip->card = card; *rcard = card; return 0; } -static int __init cs4231_attach_finish(struct snd_card *card, struct snd_cs4231 *chip) +static int __init cs4231_attach_finish(struct snd_card *card) { + struct snd_cs4231 *chip = card->private_data; int err; - if ((err = snd_cs4231_pcm(chip)) < 0) + if ((err = snd_cs4231_pcm(card)) < 0) goto out_err; - if ((err = snd_cs4231_mixer(chip)) < 0) + if ((err = snd_cs4231_mixer(card)) < 0) goto out_err; - if ((err = snd_cs4231_timer(chip)) < 0) + if ((err = snd_cs4231_timer(card)) < 0) goto out_err; if ((err = snd_card_register(card)) < 0) @@ -1925,23 +1895,16 @@ static struct snd_device_ops snd_cs4231_ static int __init snd_cs4231_sbus_create(struct snd_card *card, struct sbus_dev *sdev, - int dev, - struct snd_cs4231 **rchip) + int dev) { - struct snd_cs4231 *chip; + struct snd_cs4231 *chip = card->private_data; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - spin_lock_init(&chip->lock); spin_lock_init(&chip->c_dma.sbus_info.lock); spin_lock_init(&chip->p_dma.sbus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); - chip->card = card; chip->dev_u.sdev = sdev; chip->regs_size = sdev->reg_addrs[0].reg_size; memcpy(&chip->image, &snd_cs4231_original_image, @@ -1992,14 +1955,12 @@ static int __init snd_cs4231_sbus_create return err; } - *rchip = chip; return 0; } static int __init cs4231_sbus_attach(struct sbus_dev *sdev) { struct resource *rp = &sdev->resource[0]; - struct snd_cs4231 *cp; struct snd_card *card; int err; @@ -2013,12 +1974,12 @@ static int __init cs4231_sbus_attach(str (unsigned long long)rp->start, sdev->irqs[0]); - if ((err = snd_cs4231_sbus_create(card, sdev, dev, &cp)) < 0) { + if ((err = snd_cs4231_sbus_create(card, sdev, dev)) < 0) { snd_card_free(card); return err; } - return cs4231_attach_finish(card, cp); + return cs4231_attach_finish(card); } #endif @@ -2105,24 +2066,17 @@ static struct snd_device_ops snd_cs4231_ static int __init snd_cs4231_ebus_create(struct snd_card *card, struct linux_ebus_device *edev, - int dev, - struct snd_cs4231 **rchip) + int dev) { - struct snd_cs4231 *chip; + struct snd_cs4231 *chip = card->private_data; int err; - *rchip = NULL; - chip = kzalloc(sizeof(*chip), GFP_KERNEL); - if (chip == NULL) - return -ENOMEM; - spin_lock_init(&chip->lock); spin_lock_init(&chip->c_dma.ebus_info.lock); spin_lock_init(&chip->p_dma.ebus_info.lock); mutex_init(&chip->mce_mutex); mutex_init(&chip->open_mutex); chip->flags |= CS4231_FLAG_EBUS; - chip->card = card; chip->dev_u.pdev = edev->bus->self; memcpy(&chip->image, &snd_cs4231_original_image, sizeof(snd_cs4231_original_image)); @@ -2192,14 +2146,12 @@ static int __init snd_cs4231_ebus_create return err; } - *rchip = chip; return 0; } static int __init cs4231_ebus_attach(struct linux_ebus_device *edev) { struct snd_card *card; - struct snd_cs4231 *chip; int err; err = cs4231_attach_begin(&card); @@ -2211,12 +2163,12 @@ static int __init cs4231_ebus_attach(str edev->resource[0].start, edev->irqs[0]); - if ((err = snd_cs4231_ebus_create(card, edev, dev, &chip)) < 0) { + if ((err = snd_cs4231_ebus_create(card, edev, dev)) < 0) { snd_card_free(card); return err; } - return cs4231_attach_finish(card, chip); + return cs4231_attach_finish(card); } #endif

2 1

[alsa-devel] HDA: STAC92xx: [PATCH v2] : add support for missing features + fix two bugs in new dynamic powersave mode
by Maxim Levitsky 03 Sep '07

03 Sep '07

Hi, This is new version of my patchset to improve STAC driver. I fixed all issues you told me about. Note that I also fixed two bugs in hda driver itself, and this is why I didn't have sound after resume-from ram. Best regards, Maxim Levitsky

2 8

[alsa-devel] [PATCH] dbri: more cleanups
by Krzysztof Helt 03 Sep '07

03 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch: - removes redundant constant suffices - removes redundant parentheses - removes redundant curly brackets - removes check if a spinlock is locked inside method which is only called with the spinlock locked - moves few functions to the __init section - removes line which appears twice after the previous patch - minor comments improvements Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp old/sound/sparc/dbri.c new/sound/sparc/dbri.c --- old/sound/sparc/dbri.c 2007-08-30 10:27:00.000000000 +0200 +++ new/sound/sparc/dbri.c 2007-09-01 22:48:24.947948928 +0200 @@ -233,12 +233,12 @@ static struct { ****************************************************************************/ /* DBRI main registers */ -#define REG0 0x00UL /* Status and Control */ -#define REG1 0x04UL /* Mode and Interrupt */ -#define REG2 0x08UL /* Parallel IO */ -#define REG3 0x0cUL /* Test */ -#define REG8 0x20UL /* Command Queue Pointer */ -#define REG9 0x24UL /* Interrupt Queue Pointer */ +#define REG0 0x00 /* Status and Control */ +#define REG1 0x04 /* Mode and Interrupt */ +#define REG2 0x08 /* Parallel IO */ +#define REG3 0x0c /* Test */ +#define REG8 0x20 /* Command Queue Pointer */ +#define REG9 0x24 /* Interrupt Queue Pointer */ #define DBRI_NO_CMDS 64 #define DBRI_INT_BLK 64 @@ -565,7 +565,7 @@ struct snd_dbri { /* Translate the ALSA direction into the array index */ #define DBRI_STREAMNO(substream) \ (substream->stream == \ - SNDRV_PCM_STREAM_PLAYBACK? DBRI_PLAY: DBRI_REC) + SNDRV_PCM_STREAM_PLAYBACK ? DBRI_PLAY: DBRI_REC) /* Return a pointer to dbri_streaminfo */ #define DBRI_STREAM(dbri, substream) \ @@ -611,8 +611,8 @@ The list is terminated with a WAIT comma CPU interrupt to signal completion. Since the DBRI can run in parallel with the CPU, several means of -synchronization present themselves. The method implemented here is only -use of the dbri_cmdwait() to wait for execution of batch of sent commands. +synchronization present themselves. The method implemented here uses +the dbri_cmdwait() to wait for execution of batch of sent commands. A circular command buffer is used here. A new command is being added while another can be executed. The scheme works by adding two WAIT commands @@ -648,15 +648,14 @@ static void dbri_cmdwait(struct snd_dbri } spin_unlock_irqrestore(&dbri->lock, flags); - if (maxloops == 0) { + if (maxloops == 0) printk(KERN_ERR "DBRI: Chip never completed command buffer\n"); - } else { + else dprintk(D_CMD, "Chip completed command buffer (%d)\n", MAXLOOPS - maxloops - 1); - } } /* - * Lock the command queue and returns pointer to a space for len cmd words + * Lock the command queue and return pointer to space for len cmd words * It locks the cmdlock spinlock. */ static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) @@ -749,7 +748,7 @@ static void dbri_reset(struct snd_dbri * } /* Lock must not be held before calling this */ -static void dbri_initialize(struct snd_dbri *dbri) +static void __init dbri_initialize(struct snd_dbri *dbri) { s32 *cmd; u32 dma_addr; @@ -804,7 +803,7 @@ list ordering, among other things. The here interface closely with the transmit and receive interrupt code. */ -static int pipe_active(struct snd_dbri *dbri, int pipe) +static inline int pipe_active(struct snd_dbri *dbri, int pipe) { return ((pipe >= 0) && (dbri->pipes[pipe].desc != -1)); } @@ -1144,10 +1143,10 @@ static int setup_descs(struct snd_dbri * while (len > 0) { int mylen; - for (; desc < DBRI_NO_DESCS; desc++) { + for (; desc < DBRI_NO_DESCS; desc++) if (!dbri->dma->desc[desc].ba) break; - } + if (desc == DBRI_NO_DESCS) { printk(KERN_ERR "DBRI: setup_descs: No descriptors\n"); return -1; @@ -1308,7 +1307,7 @@ to the DBRI via the CHI interface and fe * Lock must not be held before calling it. */ -static void cs4215_setup_pipes(struct snd_dbri *dbri) +static __init void cs4215_setup_pipes(struct snd_dbri *dbri) { unsigned long flags; @@ -1341,7 +1340,7 @@ static void cs4215_setup_pipes(struct sn dbri_cmdwait(dbri); } -static int cs4215_init_data(struct cs4215 *mm) +static __init int cs4215_init_data(struct cs4215 *mm) { /* * No action, memory resetting only. @@ -1633,7 +1632,7 @@ static int cs4215_prepare(struct snd_dbr /* * */ -static int cs4215_init(struct snd_dbri *dbri) +static __init int cs4215_init(struct snd_dbri *dbri) { u32 reg2 = sbus_readl(dbri->regs + REG2); dprintk(D_MM, "cs4215_init: reg2=0x%x\n", reg2); @@ -1771,13 +1770,10 @@ static void xmit_descs(struct snd_dbri * static void transmission_complete_intr(struct snd_dbri *dbri, int pipe) { - struct dbri_streaminfo *info; - int td; + struct dbri_streaminfo *info = &dbri->stream_info[DBRI_PLAY]; + int td = dbri->pipes[pipe].desc; int status; - info = &dbri->stream_info[DBRI_PLAY]; - - td = dbri->pipes[pipe].desc; while (td >= 0) { if (td >= DBRI_NO_DESCS) { printk(KERN_ERR "DBRI: invalid td on pipe %d\n", pipe); @@ -1798,12 +1794,9 @@ static void transmission_complete_intr(s } /* Notify ALSA */ - if (spin_is_locked(&dbri->lock)) { - spin_unlock(&dbri->lock); - snd_pcm_period_elapsed(info->substream); - spin_lock(&dbri->lock); - } else - snd_pcm_period_elapsed(info->substream); + spin_unlock(&dbri->lock); + snd_pcm_period_elapsed(info->substream); + spin_lock(&dbri->lock); } static void reception_complete_intr(struct snd_dbri *dbri, int pipe) @@ -1830,12 +1823,9 @@ static void reception_complete_intr(stru rd, DBRI_RD_STATUS(status), DBRI_RD_CNT(status)); /* Notify ALSA */ - if (spin_is_locked(&dbri->lock)) { - spin_unlock(&dbri->lock); - snd_pcm_period_elapsed(info->substream); - spin_lock(&dbri->lock); - } else - snd_pcm_period_elapsed(info->substream); + spin_unlock(&dbri->lock); + snd_pcm_period_elapsed(info->substream); + spin_lock(&dbri->lock); } static void dbri_process_one_interrupt(struct snd_dbri *dbri, int x) @@ -1848,13 +1838,12 @@ static void dbri_process_one_interrupt(s int rval = D_INTR_GETRVAL(x); #endif - if (channel == D_INTR_CMD) { + if (channel == D_INTR_CMD) dprintk(D_CMD, "INTR: Command: %-5s Value:%d\n", cmds[command], val); - } else { + else dprintk(D_INT, "INTR: Chan:%d Code:%d Val:%#x\n", channel, code, rval); - } switch (code) { case D_INTR_CMDI: @@ -1986,10 +1975,10 @@ static irqreturn_t snd_dbri_interrupt(in PCM Interface ****************************************************************************/ static struct snd_pcm_hardware snd_dbri_pcm_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), + .info = SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID, .formats = SNDRV_PCM_FMTBIT_MU_LAW | SNDRV_PCM_FMTBIT_A_LAW | SNDRV_PCM_FMTBIT_U8 | @@ -1999,7 +1988,7 @@ static struct snd_pcm_hardware snd_dbri_ .rate_max = 48000, .channels_min = 1, .channels_max = 2, - .buffer_bytes_max = (64 * 1024), + .buffer_bytes_max = 64 * 1024, .period_bytes_min = 1, .period_bytes_max = DBRI_TD_MAXCNT, .periods_min = 1, @@ -2266,11 +2255,10 @@ static int snd_cs4215_info_volume(struct uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; uinfo->count = 2; uinfo->value.integer.min = 0; - if (kcontrol->private_value == DBRI_PLAY) { + if (kcontrol->private_value == DBRI_PLAY) uinfo->value.integer.max = DBRI_MAX_VOLUME; - } else { + else uinfo->value.integer.max = DBRI_MAX_GAIN; - } return 0; } @@ -2304,7 +2292,7 @@ static int snd_cs4215_put_volume(struct info->right_gain = ucontrol->value.integer.value[1]; changed = 1; } - if (changed == 1) { + if (changed) { /* First mute outputs, and wait 1/8000 sec (125 us) * to make sure this takes. This avoids clicking noises. */ @@ -2442,11 +2430,10 @@ static int __init snd_dbri_mixer(struct card = dbri->card; strcpy(card->mixername, card->shortname); - for (idx = 0; idx < ARRAY_SIZE(dbri_controls); idx++) { + for (idx = 0; idx < ARRAY_SIZE(dbri_controls); idx++) if ((err = snd_ctl_add(card, snd_ctl_new1(&dbri_controls[idx], dbri))) < 0) return err; - } for (idx = DBRI_REC; idx < DBRI_NO_STREAMS; idx++) { dbri->stream_info[idx].left_gain = 0; @@ -2478,19 +2465,18 @@ static void dbri_debug_read(struct snd_i int pipe; snd_iprintf(buffer, "debug=%d\n", dbri_debug); - for (pipe = 0; pipe < 32; pipe++) { + for (pipe = 0; pipe < 32; pipe++) if (pipe_active(dbri, pipe)) { struct dbri_pipe *pptr = &dbri->pipes[pipe]; snd_iprintf(buffer, "Pipe %d: %s SDP=0x%x desc=%d, " "len=%d next %d\n", pipe, - ((pptr->sdp & D_SDP_TO_SER) ? "output" : - "input"), + (pptr->sdp & D_SDP_TO_SER) ? "output" : + "input", pptr->sdp, pptr->desc, pptr->length, pptr->nextpipe); } - } } #endif @@ -2502,7 +2488,7 @@ void snd_dbri_proc(struct snd_dbri *dbri snd_info_set_text_ops(entry, dbri, dbri_regs_read); #ifdef DBRI_DEBUG - if (! snd_card_proc_new(dbri->card, "debug", &entry)) { + if (!snd_card_proc_new(dbri->card, "debug", &entry)) { snd_info_set_text_ops(entry, dbri, dbri_debug_read); entry->mode = S_IFREG | S_IRUGO; /* Readable only. */ } @@ -2637,7 +2623,6 @@ static int __init dbri_attach(int prom_n goto _err; if ((err = snd_dbri_mixer(dbri)) < 0) - if ((err = snd_dbri_mixer(dbri)) < 0) goto _err; /* /proc file handling */ @@ -2668,7 +2653,7 @@ static int __init dbri_init(void) int found = 0; /* Probe each SBUS for the DBRI chip(s). */ - for_all_sbusdev(sdev, sbus) { + for_all_sbusdev(sdev, sbus) /* * The version is coded in the last character */ @@ -2679,7 +2664,6 @@ static int __init dbri_init(void) if (dbri_attach(sdev->prom_node, sdev) == 0) found++; } - } return (found > 0) ? 0 : -EIO; }

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[alsa-devel] Problems with left audio channel on nVidia MCP61.
by Steven Young 03 Sep '07

03 Sep '07

Hi, I am having problems with the left audio channel when using the nVidia MCP61 chipset. It first cropped up when I bought a new machine at home, and at the time I figured it was probably faulty hardware. However, I recently acquired another machine with the same chipset, and the problem occurs on it too, so now I'm thinking it might be a driver problem. The problem is that the left audio channel emits a high-pitched squeal whenever the audio device is in use. The right channel works normally. Also, trying to adjust the volume causes all sound to stop working until I uninstall and re-install all the modules relevant for sound. I haven't done anything fancy to try and fix the problem since I don't know a whole lot about ALSA. I'm just using whatever configuration Ubuntu 7.04 magically decided would be best. lspci of one of the machines is appended. Can anyone point me in the direction of what I need to do to fix this? I searched the mailing list archive and couldn't find anything directly relevant. Thanks, Steve. 00:00.0 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a1) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0 Capabilities: [44] HyperTransport: Slave or Primary Interface Capabilities: [dc] HyperTransport: MSI Mapping 00:01.0 ISA bridge: nVidia Corporation MCP61 LPC Bridge (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0 00:01.1 SMBus: nVidia Corporation MCP61 SMBus (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: 66MHz, fast devsel, IRQ 10 I/O ports at dc00 [size=64] I/O ports at 0600 [size=64] I/O ports at 0700 [size=64] Capabilities: [44] Power Management version 2 00:01.2 RAM memory: nVidia Corporation MCP61 Memory Controller (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: 66MHz, fast devsel 00:01.3 Co-processor: nVidia Corporation MCP61 SMU (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: 66MHz, fast devsel Memory at 40100000 (32-bit, non-prefetchable) [disabled] [size=512K] 00:02.0 USB Controller: nVidia Corporation MCP61 USB Controller (rev a2) (prog-if 10 [OHCI]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 16 Memory at dfe7f000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 00:02.1 USB Controller: nVidia Corporation MCP61 USB Controller (rev a2) (prog-if 20 [EHCI]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 19 Memory at dfe7ec00 (32-bit, non-prefetchable) [size=256] Capabilities: [44] Debug port Capabilities: [80] Power Management version 2 00:04.0 PCI bridge: nVidia Corporation MCP61 PCI bridge (rev a1) (prog-if 01 [Subtractive decode]) Flags: bus master, 66MHz, fast devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=64 I/O behind bridge: 0000e000-0000efff Memory behind bridge: dff00000-dfffffff Prefetchable memory behind bridge: 40000000-400fffff Capabilities: [b8] Subsystem: nVidia Corporation Unknown device cb84 Capabilities: [8c] HyperTransport: MSI Mapping 00:05.0 Audio device: nVidia Corporation MCP61 High Definition Audio (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 19 Memory at dfe78000 (32-bit, non-prefetchable) [size=16K] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/0 Enable- Capabilities: [6c] HyperTransport: MSI Mapping 00:06.0 IDE interface: nVidia Corporation MCP61 IDE (rev a2) (prog-if 8a [Master SecP PriP]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0 [virtual] Memory at 000001f0 (32-bit, non-prefetchable) [disabled] [size=8] [virtual] Memory at 000003f0 (type 3, non-prefetchable) [disabled] [size=1] [virtual] Memory at 00000170 (32-bit, non-prefetchable) [disabled] [size=8] [virtual] Memory at 00000370 (type 3, non-prefetchable) [disabled] [size=1] I/O ports at ffa0 [size=16] Capabilities: [44] Power Management version 2 00:07.0 Bridge: nVidia Corporation MCP61 Ethernet (rev a2) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at dfe7d000 (32-bit, non-prefetchable) [size=4K] I/O ports at d480 [size=8] Capabilities: [44] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask+ 64bit+ Queue=0/3 Enable- Capabilities: [6c] HyperTransport: MSI Mapping 00:08.0 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) (prog-if 85 [Master SecO PriO]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 20 I/O ports at d400 [size=8] I/O ports at d080 [size=4] I/O ports at d000 [size=8] I/O ports at cc00 [size=4] I/O ports at c880 [size=16] Memory at dfe7c000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 Capabilities: [b0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/2 Enable- Capabilities: [cc] HyperTransport: MSI Mapping 00:08.1 IDE interface: nVidia Corporation MCP61 SATA Controller (rev a2) (prog-if 85 [Master SecO PriO]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 16 I/O ports at c800 [size=8] I/O ports at c480 [size=4] I/O ports at c400 [size=8] I/O ports at c080 [size=4] I/O ports at c000 [size=16] Memory at dfe73000 (32-bit, non-prefetchable) [size=4K] Capabilities: [44] Power Management version 2 Capabilities: [b0] Message Signalled Interrupts: Mask- 64bit+ Queue=0/2 Enable- Capabilities: [cc] HyperTransport: MSI Mapping 00:09.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=02, subordinate=02, sec-latency=0 Capabilities: [40] Subsystem: nVidia Corporation Unknown device 0000 Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable- Capabilities: [60] HyperTransport: MSI Mapping Capabilities: [80] Express Root Port (Slot+) IRQ 0 00:0b.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=03, subordinate=03, sec-latency=0 Capabilities: [40] Subsystem: nVidia Corporation Unknown device 0000 Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable- Capabilities: [60] HyperTransport: MSI Mapping Capabilities: [80] Express Root Port (Slot+) IRQ 0 00:0c.0 PCI bridge: nVidia Corporation MCP61 PCI Express bridge (rev a2) (prog-if 00 [Normal decode]) Flags: bus master, fast devsel, latency 0 Bus: primary=00, secondary=04, subordinate=04, sec-latency=0 Capabilities: [40] Subsystem: nVidia Corporation Unknown device 0000 Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/1 Enable- Capabilities: [60] HyperTransport: MSI Mapping Capabilities: [80] Express Root Port (Slot+) IRQ 0 00:0d.0 VGA compatible controller: nVidia Corporation GeForce 6100 nForce 430 (rev a2) (prog-if 00 [VGA]) Subsystem: ASUSTeK Computer Inc. Unknown device 8234 Flags: bus master, 66MHz, fast devsel, latency 0, IRQ 17 Memory at de000000 (32-bit, non-prefetchable) [size=16M] Memory at c0000000 (64-bit, prefetchable) [size=256M] Memory at dd000000 (64-bit, non-prefetchable) [size=16M] [virtual] Expansion ROM at dfe40000 [disabled] [size=128K] Capabilities: [48] Power Management version 2 Capabilities: [50] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration Flags: fast devsel Capabilities: [80] HyperTransport: Host or Secondary Interface 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map Flags: fast devsel 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller Flags: fast devsel 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control Flags: fast devsel Capabilities: [f0] #0f [0010] 01:06.0 Ethernet controller: ADMtek NC100 Network Everywhere Fast Ethernet 10/100 (rev 11) Subsystem: Accton Technology Corporation Unknown device 1216 Flags: bus master, medium devsel, latency 64, IRQ 18 I/O ports at e800 [size=256] Memory at dffffc00 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 40000000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 01:07.0 FireWire (IEEE 1394): NEC Corporation uPD72874 IEEE1394 OHCI 1.1 3-port PHY-Link Ctrlr (rev 01) (prog-if 10 [OHCI]) Subsystem: NEC Corporation uPD72874 IEEE1394 OHCI 1.1 3-port PHY-Link Ctrlr Flags: bus master, medium devsel, latency 64, IRQ 21 Memory at dfffe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [60] Power Management version 2

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[alsa-devel] Dealing with 'siamesed' DAC sample rates
by John Utz 03 Sep '07

03 Sep '07

Hello All; The thrice cursed vt1618 has the DAC sample rates regs for the Surround and Center/LFE DAC's (0x2E, 0x30) set to RO and locked to whatever rate that the Front DAC is set to. This was probably a reasonable choice with respect to what the typical end user might want to do, but i have to support content at 44100, not 48K. So, here's the example of setting the front dac and getting the other dac's changed: jutz-gntdv alsa-kernel # echo 0x2c 0xac44> /proc/asound/card0/codec97#0/ac97#0-0+regs jutz-gntdv alsa-kernel # cat /proc/asound/card0/codec97#0/ac97#0-0+regs| grep ac44 0:2c = ac44 0:2e = ac44 0:30 = ac44 The *problem* seems to be that alsa wants to *set* the register even tho the value that's in the register is already the right value (44100=ac44) So, i *think* that i want to change my driver code to poke the value for setting hw:0,1 (should be reg 0x2e) into 0x2c instead? or am i barking up the wrong tree? Here's the problem that i am trying to solve: Here's the output of speaker-test on the front (hw:0,0) and surround (hw:0,1) devices. [root@DMX-116248 jutz]# ./speaker-test -c2 -Dhw:0,0 -r 44100 speaker-test 1.0.14rc2 Playback device is hw:0,0 Stream parameters are 44100Hz, S16_LE, 2 channels Using 16 octaves of pink noise Rate set to 44100Hz (requested 44100Hz) Buffer size range from 16 to 16384 Period size range from 8 to 8192 Using max buffer size 16384 Periods = 4 was set period_size = 4096 was set buffer_size = 16384 0 - Front Left ^X [root@DMX-116248 jutz]# ./speaker-test -c2 -Dhw:0,1 -r 44100 speaker-test 1.0.14rc2 Playback device is hw:0,1 Stream parameters are 44100Hz, S16_LE, 2 channels Using 16 octaves of pink noise ALSA ERROR hw_params: set (RATE) value = 44100 : Invalid argument ACCESS: RW_INTERLEAVED FORMAT: S16_LE SUBFORMAT: STD SAMPLE_BITS: 16 FRAME_BITS: 32 CHANNELS: 2 RATE: 48000 PERIOD_TIME: (166 170667) PERIOD_SIZE: [8 8192] PERIOD_BYTES: [32 32768] PERIODS: [2 127] BUFFER_TIME: (333 341334) BUFFER_SIZE: [16 16384] BUFFER_BYTES: [64 65536] TICK_TIME: 1000 Rate 44100Hz not available for playback: Invalid argument Setting of hwparams failed: Invalid argument [root@DMX-116248 jutz]# Does anybody have any suggestions? tnx! johnu

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[PATCH] add support for analog loopback to STAC9204/9205/922x/927x
by Maxim Levitsky 03 Sep '07

03 Sep '07

The analog loopback routes the sound just before it enters ADC0 to output of DAC0 Signed-off-by: Maxim Levitsky <maximlevitsky(a)gmail.com> --- pci/hda/patch_sigmatel.c | 58 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 58 insertions(+), 0 deletions(-) diff --git a/pci/hda/patch_sigmatel.c b/pci/hda/patch_sigmatel.c index 297f740..c94775c 100644 --- a/pci/hda/patch_sigmatel.c +++ b/pci/hda/patch_sigmatel.c @@ -147,6 +147,7 @@ struct sigmatel_spec { /* i/o switches */ unsigned int io_switch[2]; unsigned int clfe_swap; + unsigned int aloopback; struct hda_pcm pcm_rec[2]; /* PCM information */ @@ -296,6 +297,49 @@ static int stac92xx_mux_enum_put(struct snd_kcontrol *kcontrol, struct snd_ctl_e spec->mux_nids[adc_idx], &spec->cur_mux[adc_idx]); } +#define stac92xx_aloopback_info snd_ctl_boolean_mono_info + +static int stac92xx_aloopback_get(struct snd_kcontrol *kcontrol, + 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; + return 0; +} + +static int stac92xx_aloopback_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; + unsigned int dac_mode; + + if (spec->aloopback == ucontrol->value.integer.value[0]) + return 0; + + spec->aloopback = ucontrol->value.integer.value[0]; + + + dac_mode = snd_hda_codec_read(codec, codec->afg, 0, + kcontrol->private_value & 0xFFFF, 0x0); + + if (spec->aloopback) { + snd_hda_power_up(codec); + dac_mode |= 0x40; + } else { + snd_hda_power_down(codec); + dac_mode &= ~0x40; + } + + snd_hda_codec_write_cache(codec, codec->afg, 0, + kcontrol->private_value >> 16, dac_mode); + + return 1; +} + + static struct hda_verb stac9200_core_init[] = { /* set dac0mux for dac converter */ { 0x07, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -346,6 +390,17 @@ static struct hda_verb stac9205_core_init[] = { .put = stac92xx_mux_enum_put, \ } +#define STAC_ANALOG_LOOPBACK(verb_read,verb_write) \ + { \ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, \ + .name = "Analog Loopback", \ + .count = 1, \ + .info = stac92xx_aloopback_info, \ + .get = stac92xx_aloopback_get, \ + .put = stac92xx_aloopback_put, \ + .private_value = verb_read | (verb_write << 16), \ + } + static struct snd_kcontrol_new stac9200_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0xb, 0, HDA_OUTPUT), @@ -377,6 +432,7 @@ static struct snd_kcontrol_new stac922x_mixer[] = { /* This needs to be generated dynamically based on sequence */ static struct snd_kcontrol_new stac9227_mixer[] = { STAC_INPUT_SOURCE, + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), HDA_CODEC_VOLUME("Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE("Capture Switch", 0x1b, 0x0, HDA_OUTPUT), { } /* end */ @@ -384,6 +440,7 @@ static struct snd_kcontrol_new stac9227_mixer[] = { static struct snd_kcontrol_new stac927x_mixer[] = { STAC_INPUT_SOURCE, + STAC_ANALOG_LOOPBACK(0xFEB, 0x7EB), HDA_CODEC_VOLUME("InMux Capture Volume", 0x15, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("InVol Capture Volume", 0x18, 0x0, HDA_INPUT), HDA_CODEC_MUTE("ADCMux Capture Switch", 0x1b, 0x0, HDA_OUTPUT), @@ -400,6 +457,7 @@ static struct snd_kcontrol_new stac9205_mixer[] = { .put = stac92xx_dmux_enum_put, }, STAC_INPUT_SOURCE, + STAC_ANALOG_LOOPBACK(0xFE0, 0x7E0), HDA_CODEC_VOLUME("InMux Capture Volume", 0x19, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME("InVol Capture Volume", 0x1b, 0x0, HDA_INPUT), HDA_CODEC_MUTE("ADCMux Capture Switch", 0x1d, 0x0, HDA_OUTPUT), -- 1.5.2.3 --=-jwBifUR82u2MUgztB4NU--

1 0

[alsa-devel] [PATCH] Gallant SC-6000 driver
by Krzysztof Helt 03 Sep '07

03 Sep '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This is port of the Gallant SC-6000 driver from the OSS aedsp16 driver. This card was also sold as AudioExcel DSP 16 and Zoltrix AV302 (Audio Plus True 16). --- I ported only SC-6000 part from the OSS driver. PCM and CD output work. Mixer also works. I haven't tested recording. If anybody has the SC-6600 card I can port this part as well, but I do not have hardware to test. BTW. OSS driver does not work. Sound loops during play. diff -urpN linux-2.6.23.orig/sound/isa/Kconfig linux-2.6.23/sound/isa/Kconfig --- linux-2.6.23.orig/sound/isa/Kconfig 2007-08-19 08:35:59.000000000 +0200 +++ linux-2.6.23/sound/isa/Kconfig 2007-09-01 08:59:15.526990235 +0200 @@ -191,6 +191,17 @@ config SND_ES18XX To compile this driver as a module, choose M here: the module will be called snd-es18xx. +config SND_SC6000 + tristate "Gallant SC-6000, Audio Excel DSP 16" + depends on SND + select SND_AD1848_LIB + help + Say Y here to include support for Gallant SC-6000 card and + compatible soundcards: Audio Excel DSP 16 and Zoltrix AV302. + + To compile this driver as a module, choose M here: the module + will be called snd-sc6000. + config SND_GUS_SYNTH tristate diff -urpN linux-2.6.23.orig/sound/isa/Makefile linux-2.6.23/sound/isa/Makefile --- linux-2.6.23.orig/sound/isa/Makefile 2007-07-09 01:32:17.000000000 +0200 +++ linux-2.6.23/sound/isa/Makefile 2007-09-01 08:59:45.872719539 +0200 @@ -10,6 +10,7 @@ snd-cmi8330-objs := cmi8330.o snd-dt019x-objs := dt019x.o snd-es18xx-objs := es18xx.o snd-opl3sa2-objs := opl3sa2.o +snd-sc6000-objs := sc6000.o snd-sgalaxy-objs := sgalaxy.o snd-sscape-objs := sscape.o @@ -21,6 +22,7 @@ obj-$(CONFIG_SND_CMI8330) += snd-cmi8330 obj-$(CONFIG_SND_DT019X) += snd-dt019x.o obj-$(CONFIG_SND_ES18XX) += snd-es18xx.o obj-$(CONFIG_SND_OPL3SA2) += snd-opl3sa2.o +obj-$(CONFIG_SND_SC6000) += snd-sc6000.o obj-$(CONFIG_SND_SGALAXY) += snd-sgalaxy.o obj-$(CONFIG_SND_SSCAPE) += snd-sscape.o diff -urpN linux-2.6.23.orig/sound/isa/sc6000.c linux-2.6.23/sound/isa/sc6000.c --- linux-2.6.23.orig/sound/isa/sc6000.c 1970-01-01 01:00:00.000000000 +0100 +++ linux-2.6.23/sound/isa/sc6000.c 2007-09-01 09:00:48.480287340 +0200 @@ -0,0 +1,566 @@ +/* + * Driver for Gallant SC-6000 soundcard. This card is also known as + * Audio Excel DSP 16 or Zoltrix AV302. + * Copyright (C) 2007 Krzysztof Helt <krzysztof.h1(a)wp.pl> + * + * I don't have documentation for this card. I used the driver + * for OSS/Free included in the kernel source as reference. + * + * 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 + */ + +#include <linux/module.h> +#include <linux/isa.h> +#include <asm/dma.h> +#include <sound/driver.h> +#include <sound/core.h> +#include <sound/ad1848.h> +#include <sound/control.h> +#define SNDRV_LEGACY_FIND_FREE_IRQ +#define SNDRV_LEGACY_FIND_FREE_DMA +#include <sound/initval.h> + +MODULE_AUTHOR("Krzysztof Helt"); +MODULE_DESCRIPTION("Gallant SC-6000"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Gallant, SC-6000}," + "{AudioExcel, Audio Excel DSP 16}," + "{Zoltrix, AV302}}"); + +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; /* Enable this card */ +static long port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x220, 0x240 */ +static int irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 11 */ +static long mss_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; /* 0x530, 0xe80 */ +static long mpu_port[SNDRV_CARDS] = SNDRV_DEFAULT_PORT; + /* 0x300, 0x310, 0x320, 0x330 */ +static int mpu_irq[SNDRV_CARDS] = SNDRV_DEFAULT_IRQ; /* 5, 7, 9, 10, 0 */ +static int dma[SNDRV_CARDS] = SNDRV_DEFAULT_DMA; /* 0, 1, 3 */ + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for sc-6000 based soundcard."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for sc-6000 based soundcard."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable sc-6000 based soundcard."); +module_param_array(port, long, NULL, 0444); +MODULE_PARM_DESC(port, "Port # for sc-6000 driver."); +module_param_array(mss_port, long, NULL, 0444); +MODULE_PARM_DESC(mss_port, "MSS Port # for sc-6000 driver."); +module_param_array(mpu_port, long, NULL, 0444); +MODULE_PARM_DESC(mpu_port, "MPU-401 port # for sc-6000 driver."); +module_param_array(irq, int, NULL, 0444); +MODULE_PARM_DESC(irq, "IRQ # for sc-6000 driver."); +module_param_array(mpu_irq, int, NULL, 0444); +MODULE_PARM_DESC(mpu_irq, "MPU-401 IRQ # for sc-6000 driver."); +module_param_array(dma, int, NULL, 0444); +MODULE_PARM_DESC(dma, "DMA # for sc-6000 driver."); + +/* + * Commands of SC6000's DSP (SBPRO+special). + * Some of them are COMMAND_xx, in the future they may change. + */ +#define WRITE_MDIRQ_CFG 0x50 /* Set M&I&DRQ mask (the real config) */ +#define COMMAND_52 0x52 /* */ +#define READ_HARD_CFG 0x58 /* Read Hardware Config (I/O base etc) */ +#define COMMAND_5C 0x5c /* */ +#define COMMAND_60 0x60 /* */ +#define COMMAND_66 0x66 /* */ +#define COMMAND_6C 0x6c /* */ +#define COMMAND_6E 0x6e /* */ +#define COMMAND_88 0x88 /* */ +#define DSP_INIT_MSS 0x8c /* Enable Microsoft Sound System mode */ +#define COMMAND_C5 0xc5 /* */ +#define GET_DSP_VERSION 0xe1 /* Get DSP Version */ +#define GET_DSP_COPYRIGHT 0xe3 /* Get DSP Copyright */ + +/* + * Offsets of SC6000 DSP I/O ports. The offset is added to base I/O port + * to have the actual I/O port. + * Register permissions are: + * (wo) == Write Only + * (ro) == Read Only + * (w-) == Write + * (r-) == Read + */ +#define DSP_RESET 0x06 /* offset of DSP RESET (wo) */ +#define DSP_READ 0x0a /* offset of DSP READ (ro) */ +#define DSP_WRITE 0x0c /* offset of DSP WRITE (w-) */ +#define DSP_COMMAND 0x0c /* offset of DSP COMMAND (w-) */ +#define DSP_STATUS 0x0c /* offset of DSP STATUS (r-) */ +#define DSP_DATAVAIL 0x0e /* offset of DSP DATA AVAILABLE (ro) */ + +#define PFX "sc-6000: " + +/* hardware dependent functions */ + +/* + * sc6000_irq_to_softcfg - Decode irq number into cfg code. + */ +static inline unsigned char sc6000_irq_to_softcfg(int irq) +{ + unsigned char val = 0; + + switch (irq) { + case 5: + val = 0x28; + break; + case 7: + val = 0x8; + break; + case 9: + val = 0x10; + break; + case 10: + val = 0x18; + break; + case 11: + val = 0x20; + break; + default: + break; + } + return val; +} + +/* + * sc6000_dma_to_softcfg - Decode dma number into cfg code. + */ +static inline unsigned char sc6000_dma_to_softcfg(int dma) +{ + unsigned char val = 0; + + switch (dma) { + case 0: + val = 1; + break; + case 1: + val = 2; + break; + case 3: + val = 3; + break; + default: + break; + } + return val; +} + +/* + * sc6000_mpu_irq_to_softcfg - Decode MPU-401 irq number into cfg code. + */ +static inline unsigned char sc6000_mpu_irq_to_softcfg(int mpu_irq) +{ + unsigned char val = 0; + + switch (mpu_irq) { + case 5: + val = 4; + break; + case 7: + val = 0x44; + break; + case 9: + val = 0x84; + break; + case 10: + val = 0xc4; + break; + default: + break; + } + return val; +} + +static __devinit int sc6000_wait_data(int port) +{ + int loop = 1000; + unsigned char val = 0; + + do { + val = inb(port + DSP_DATAVAIL); + if (val & 0x80) + return 0; + cpu_relax(); + } while (loop--); + + return -EAGAIN; +} + +static inline int sc6000_read(int port) +{ + if (sc6000_wait_data(port)) + return -EBUSY; + + return inb(port + DSP_READ); + +} + +static __devinit int sc6000_write(int port, int cmd) +{ + unsigned char val; + int loop = 500000; + + do { + val = inb(port + DSP_STATUS); + /* + * DSP ready to receive data if bit 7 of val == 0 + */ + if (!(val & 0x80)) { + outb(cmd, port + DSP_COMMAND); + return 0; + } + cpu_relax(); + } while (loop--); + + snd_printk(KERN_ERR PFX "DSP Command (0x%x) timeout.\n", cmd); + + return -EAGAIN; +} + +static int __devinit sc6000_dsp_get_answer(int port, int command, char *data, + int data_len) +{ + int len = 0; + + if (sc6000_write(port, command)) { + snd_printk(KERN_ERR PFX "CMD 0x%x: failed!\n", command); + return -EFAULT; + } + + do { + int val = sc6000_read(port); + + if (val < 0) + break; + + data[len++] = val; + + } while (len < data_len); + + /* + * If no more data available, return to the caller, no error if len>0. + * We have no other way to know when the string is finished. + */ + return len ? len : -ENXIO; +} + +static int __devinit sc6000_dsp_reset(int port) +{ + outb(1, port + DSP_RESET); + udelay(10); + outb(0, port + DSP_RESET); + udelay(20); + if (sc6000_read(port) == 0xaa) + return 0; + return -EBUSY; +} + +/* detection and initialization */ +static int __devinit sc6000_cfg_write(int port, unsigned char softcfg) +{ + + if (sc6000_write(port, WRITE_MDIRQ_CFG)) { + snd_printk(KERN_ERR PFX "CMD 0x%x: failed!\n", WRITE_MDIRQ_CFG); + return -ENXIO; + } + if (sc6000_write(port, softcfg)) { + snd_printk(KERN_ERR PFX "sc6000_cfg_write: failed!\n"); + return -ENXIO; + } + return 0; +} + +static int __devinit sc6000_setup_board(int port, int config) +{ + int loop = 10; + + do { + if (sc6000_write(port, COMMAND_88)) { + snd_printk(KERN_ERR PFX "CMD 0x%x: failed!\n", + COMMAND_88); + return -ENXIO; + } + } while ((sc6000_wait_data(port) < 0) && loop--); + + if (sc6000_read(port) < 0) { + snd_printk(KERN_ERR "sc6000_read after CMD 0x%x: failed\n", + COMMAND_88); + return -ENXIO; + } + + if (sc6000_cfg_write(port, config)) + return -EFAULT; + + return 0; +} + +static int __devinit sc6000_init_mss(int port, int config, int mss_port, + int mss_config) +{ + if (sc6000_write(port, DSP_INIT_MSS)) { + snd_printk(KERN_ERR PFX "sc6000_init_mss [0x%x]: failed!\n", + DSP_INIT_MSS); + return -ENXIO; + } + + mdelay(10); + + if (sc6000_cfg_write(port, config)) + return -ENXIO; + + outb(mss_config, mss_port); + + return 0; +} + +static int __devinit sc6000_init_board(int port, int irq, int dma, + int mss_port, int mpu_irq) +{ + char answer[15]; + int mss_config = sc6000_irq_to_softcfg(irq) | + sc6000_dma_to_softcfg(dma); + int config = mss_config | + sc6000_mpu_irq_to_softcfg(mpu_irq); + + if (sc6000_dsp_reset(port)) { + snd_printk(KERN_ERR PFX "sc6000_dsp_reset: failed!\n"); + return -EBUSY; + } + + if (sc6000_dsp_get_answer(port, GET_DSP_COPYRIGHT, answer, 15) <= 0) { + snd_printk(KERN_ERR PFX "sc6000_dsp_copyright: failed!\n"); + return -ENXIO; + } + /* + * My SC-6000 card return "SC-6000" in DSPCopyright, so + * if we have something different, we have to be warned. + */ + if (strcmp("SC-6000", answer)) + snd_printk(KERN_ERR PFX "Warning: non SC-6000 audio card!\n"); + + if (sc6000_dsp_get_answer(port, GET_DSP_VERSION, answer, 2) < 2) { + snd_printk(KERN_ERR PFX "sc6000_dsp_version: failed!\n"); + return -ENXIO; + } + + /* + * 0x0A == (IRQ 7, DMA 1, MIRQ 0) + */ + if (sc6000_cfg_write(port, 0x0a)) { + snd_printk(KERN_ERR PFX "sc6000_stdcfg: failed!\n"); + return -EFAULT; + } + + if (sc6000_setup_board(port, config) < 0) { + snd_printk(KERN_ERR PFX "sc6000_setup_board: failed!\n"); + return -ENODEV; + } + + if (sc6000_init_mss(port, config, mss_port, + mss_config) < 0) { + snd_printk(KERN_ERR PFX "Can not initialize" + "Microsoft Sound System mode.\n"); + return -ENODEV; + } + + return 0; +} + +static int __devinit snd_sc6000_detect(int dev, int irq, int dma) +{ + int err = -ENODEV; + + snd_printk(KERN_DEBUG PFX "Initializing BASE[0x%lx] IRQ[%d] " + "DMA[%d] MIRQ[%d]\n", + port[dev], irq, dma, mpu_irq[dev]); + +/* + * We must request the port region because these are the I/O + * ports to access card's control registers. + */ + if (!request_region(port[dev], 0x10, "sc-6000 (base)")) { + snd_printk(KERN_ERR PFX + "SC-6000 port I/O port region is already in use.\n"); + return -EBUSY; + } + + err = sc6000_init_board(port[dev], irq, dma, + mss_port[dev], mpu_irq[dev]); + + release_region(port[dev], 0x10); + + return err; +} + +static int __devinit snd_sc6000_mixer(struct snd_ad1848 *chip) +{ + struct snd_card *card = chip->card; + struct snd_ctl_elem_id id1, id2; + int err; + + memset(&id1, 0, sizeof(id1)); + memset(&id2, 0, sizeof(id2)); + id1.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + id2.iface = SNDRV_CTL_ELEM_IFACE_MIXER; + /* reassign AUX0 to FM */ + strcpy(id1.name, "Aux Playback Switch"); + strcpy(id2.name, "FM Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "FM Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + /* reassign AUX1 to CD */ + strcpy(id1.name, "Aux Playback Switch"); id1.index = 1; + strcpy(id2.name, "CD Playback Switch"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + strcpy(id1.name, "Aux Playback Volume"); + strcpy(id2.name, "CD Playback Volume"); + err = snd_ctl_rename_id(card, &id1, &id2); + if (err < 0) + return err; + return 0; +} + +static int __devinit snd_sc6000_match(struct device *devptr, unsigned int dev) +{ + if (!enable[dev]) + return 0; + if (port[dev] == SNDRV_AUTO_PORT) { + printk(KERN_ERR PFX "specify IO port\n"); + return 0; + } + if (mss_port[dev] == SNDRV_AUTO_IRQ) { + printk(KERN_ERR PFX "specify MSS port\n"); + return 0; + } + return 1; +} + +static int __devinit snd_sc6000_probe(struct device *devptr, unsigned int dev) +{ + static int possible_irqs[] = { 7, 9, 10, 11, -1 }; + static int possible_dmas[] = { 1, 3, 0, -1 }; + int err; + int xirq = irq[dev]; + int xdma = dma[dev]; + struct snd_card *card; + struct snd_ad1848 *chip; + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + + if (xirq == SNDRV_AUTO_IRQ) { + xirq = snd_legacy_find_free_irq(possible_irqs); + if (xirq < 0) { + snd_printk(KERN_ERR PFX "unable to find a free IRQ\n"); + err = -EBUSY; + goto _err; + } + } + + if (xdma == SNDRV_AUTO_DMA) { + xdma = snd_legacy_find_free_dma(possible_dmas); + if (xdma < 0) { + snd_printk(KERN_ERR PFX "unable to find a free DMA\n"); + err = -EBUSY; + goto _err; + } + } + + err = snd_sc6000_detect(dev, xirq, xdma); + if (err < 0) + goto _err; + + err = snd_ad1848_create(card, mss_port[dev] + 4, xirq, xdma, + AD1848_HW_DETECT, &chip); + if (err < 0) + goto _err; + card->private_data = chip; + + err = snd_ad1848_pcm(chip, 0, NULL); + if (err < 0) { + snd_printk(KERN_ERR PFX + "error creating new ad1848 PCM device\n"); + goto _err; + } + err = snd_ad1848_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR PFX "error creating new ad1848 mixer\n"); + goto _err; + } + err = snd_sc6000_mixer(chip); + if (err < 0) { + snd_printk(KERN_ERR PFX "the mixer rewrite failed\n"); + goto _err; + } + + strcpy(card->driver, "Gallant SC-6000"); + strcpy(card->shortname, "Gallant SC-6000"); + sprintf(card->longname, "Gallant SC-6000 at 0x%lx, irq %d, dma %d", + mss_port[dev], xirq, xdma); + + snd_card_set_dev(card, devptr); + + err = snd_card_register(card); + if (err < 0) + goto _err; + + dev_set_drvdata(devptr, card); + return 0; + + _err: + snd_card_free(card); + return err; +} + +static int __devexit snd_sc6000_remove(struct device *devptr, unsigned int dev) +{ + snd_card_free(dev_get_drvdata(devptr)); + dev_set_drvdata(devptr, NULL); + return 0; +} + +static struct isa_driver snd_sc6000_driver = { + .match = snd_sc6000_match, + .probe = snd_sc6000_probe, + .remove = __devexit_p(snd_sc6000_remove), + /* FIXME: suspend/resume */ + .driver = { + .name = "sc6000", + }, +}; + + +static int __init alsa_card_sc6000_init(void) +{ + return isa_register_driver(&snd_sc6000_driver, SNDRV_CARDS); +} + +static void __exit alsa_card_sc6000_exit(void) +{ + isa_unregister_driver(&snd_sc6000_driver); +} + +module_init(alsa_card_sc6000_init) +module_exit(alsa_card_sc6000_exit)

2 1

[alsa-devel] Audiophile Delta 24/96 Midi Out continuously sends zeroes (alsa 1.0.14)
by Romain Bossart 03 Sep '07

03 Sep '07

Hi, My audiophile 24/96 Midi Out sends zeroes continuously, so I can't use any instrument. I checked that with amidi by looking at the stream output of a connected instrument (midi out -> instrument -> midi in -> amidi). Zeroes are sent whether or not I try to send (normal) midi messages. I know it used some years ago (it was on win32 at that time). Midi In and the instrument do work perfectly. M-Audio answered me that they don't yet support linux. I'm using alsa 1.0.14 on debian/sid 2.6.22. I cannot tell if this worked with previous versions of alsa. Anyone has an idea? Thank you Romain

2 5

[alsa-devel] [PATCH] Update quirk for Rane SL 1 (aka. Serato Scratch Live)
by Mark Hills 03 Sep '07

03 Sep '07

Update quirk for Rane SL 1 (aka. Serato Scratch Live) Allow the interface's mixer to be used, and give the interface its correct name. Move the quirk out of the 'Novation EMS devices' section to a more appropriate position next to the Stanton DJ interface. Signed-off-by: Mark Hills <mark(a)pogo.org.uk>

2 1

[alsa-devel] [PATCH] Support Rane SL 1 (aka. Serato Scratch Live) interface
by Mark Hills 03 Sep '07

03 Sep '07

Support Rane SL 1 (aka. Serato Scratch Live) interface This quirk ensures the SL 1 interface is detected and can be used. The interface is USB and is supplied with the Serato Scratch Live system for DJs. Signed-off-by: Mark Hills <mark(a)pogo.org.uk> --- alsa-driver-1.0.15rc1/sound/usb/usbquirks.h Thu Aug 30 09:27:00 2007 +++ alsa-driver/sound/usb/usbquirks.h Sun Sep 2 14:55:00 2007 @@ -1748,6 +1748,16 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Rane SL 1 (aka. Serato Scratch Live) device */ +{ + USB_DEVICE(0x13e5, 0x0001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .vendor_name = "Rane", + .product_name = "SL-1", + .ifnum = QUIRK_NO_INTERFACE + } +}, + /* Novation EMS devices */ { USB_DEVICE_VENDOR_SPEC(0x1235, 0x0001),

2 2

[alsa-devel] [PATCH -mm] es18xx: Shut up uninitialized var build warning
by Satyam Sharma 03 Sep '07

03 Sep '07

sound/isa/es18xx.c: In function ‘snd_es18xx_isa_probe’: sound/isa/es18xx.c:2251: warning: ‘err’ may be used uninitialized in this function gcc is a sad, sad compiler. This warning is bogus so let's shut it up. Signed-off-by: Satyam Sharma <satyam(a)infradead.org> --- sound/isa/es18xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) --- linux-2.6.23-rc4-mm1/sound/isa/es18xx.c‾fix 2007-09-02 21:16:54.000000000 +0530 +++ linux-2.6.23-rc4-mm1/sound/isa/es18xx.c 2007-09-02 21:30:13.000000000 +0530 @@ -2248,7 +2248,6 @@ static int __devinit snd_es18xx_isa_prob static int __devinit snd_es18xx_isa_probe(struct device *pdev, unsigned int dev) { - int err; static int possible_irqs[] = {5, 9, 10, 7, 11, 12, -1}; static int possible_dmas[] = {1, 0, 3, 5, -1}; @@ -2276,6 +2275,8 @@ static int __devinit snd_es18xx_isa_prob } else { static unsigned long possible_ports[] = {0x220, 0x240, 0x260, 0x280}; int i; + int uninitialized_var(err); + for (i = 0; i < ARRAY_SIZE(possible_ports); i++) { port[dev] = possible_ports[i]; err = snd_es18xx_isa_probe1(dev, pdev);

2 1

[alsa-devel] capture sound from an app and send it to an other app
by Marco Bnà 01 Sep '07

01 Sep '07

Hello! I'm trying to find help to do something that shoul allow me to capture audio stream outgoing from an application and redirect it to an other as audio stream input. It like to do it for different istances of the application (an instance generates out - the other take input, this many times). how to do so? What i should do? And how? Thanks, Marco

2 1

[alsa-devel] ALSA 1.0.15rc1
by Jaroslav Kysela 01 Sep '07

01 Sep '07

Hi all, ALSA 1.0.15rc1 packages are released. Please, check http://www.alsa-project.org for download and changelog. Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

5 15

Re: [alsa-devel] Altec Lansing FX5051 mixer
by Jeremy Nickurak 01 Sep '07

01 Sep '07

Precisely the same problems with alsa 1.0.15rc1. On Tue, Aug 28, 2007 at 01:43:51AM -0600, Jeremy Nickurak wrote: > On Tue, Aug 28, 2007 at 08:50:00AM +0200, Clemens Ladisch wrote: > > > bmaControls( 0) 0x15 > > > Mute > > > Bass > > > Treble > > > bmaControls( 1) 0x02 > > > Volume > > > bmaControls( 2) 0x02 > > > Volume > > > bmaControls( 3) 0x00 > > > bmaControls( 4) 0x00 > > > bmaControls( 5) 0x00 > > > bmaControls( 6) 0x00 > > > > These descriptors say that there is a two-channel volume control that > > applies to the front channels, but I guess it really affects all > > channels. > > > > Please show the output of "amixer contents" for this card. > > numid=3,iface=MIXER,name='Tone Control - Bass' > ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 > : values=5 > numid=4,iface=MIXER,name='Tone Control - Treble' > ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 > : values=5 > 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 > > > > > > > ... > > > HID Device Descriptor: > > > bLength 9 > > > bDescriptorType 33 > > > bcdHID 1.00 > > > bCountryCode 0 Not supported > > > bNumDescriptors 1 > > > bDescriptorType 34 Report > > > wDescriptorLength 172 > > > Report Descriptor: (length is 172) > > > Item(Global): Usage Page, data= [ 0x0c ] 12 > > > Consumer > > > Item(Local ): Usage, data= [ 0x01 ] 1 > > > Consumer Control > > > Item(Main ): Collection, data= [ 0x01 ] 1 > > > Application > > > Item(Global): Usage Page, data= [ 0x0c ] 12 > > > Consumer > > > Item(Local ): Usage, data= [ 0xe2 ] 226 > > > Mute > > > Item(Local ): Usage, data= [ 0xe9 ] 233 > > > Volume Increment > > > ... > > > > This describes an input device that reports 'key' presses to the > > computer. I guess the control panel and remote control don't have a > > direct effect but expect the computer to explicitly change mixer > > controls when a key is pressed. > > Since it works fine in Windows without any non-ms software, would this > imply that this is somewhat standard behavior? > > > > > However, this shouldn't prevent alsamixer from working. > > > > It might be possible that the mixer controls don't have many steps and > > that the step size used by alsamixer is so small that the device rounds > > the setting back to the old position. Please try PageUp/Down or Home/ > > End instead of Up/Down keys. > > PgUp/PgDown don't do any better on any controls. I seem to be able to > drop the PCM volume to nothing in practice with End, but Home won't > restore it. Up/Down immediately brings it back up to 100%. The LED's on > the control reflect this btw, but alsamixer still reports 98%. The control pad wouldn't be an especially big deal, if only alsamixer did work right. -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

1 0

Re: [alsa-devel] [2/4] 2.6.23-rc4: known regressions
by Michal Piotrowski 01 Sep '07

01 Sep '07

Hi all, Here is a list of some known regressions in 2.6.23-rc4. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions List of Aces Name Regressions fixed since 21-Jun-2007 Adrian Bunk 9 Andi Kleen 5 Linus Torvalds 5 Andrew Morton 4 Hugh Dickins 4 Al Viro 3 Alan Stern 3 Cornelia Huck 3 Jens Axboe 3 Tejun Heo 3 ACPI Subject : 2.6.23-rc4: maxcpus still broken References : http://lkml.org/lkml/2007/8/28/87 Last known good : ? Submitter : Alexey Dobriyan <adobriyan(a)sw.ru> Caused-By : Len Brown <len.brown(a)intel.com> commit 61ec7567db103d537329b0db9a887db570431ff4 Handled-By : Len Brown <len.brown(a)intel.com> Status : problem is being debugged Subject : MCFG bug on hp nx6310 References : http://lkml.org/lkml/2007/8/8/252 Last known good : ? Submitter : Michael Sedkowski <sedmich(a)gmail.com> Caused-By : ? Handled-By : ? Status : unknown Subject : 2.6.23-rc1-git10 hangs on boot - needs "acpi=off" References : http://lkml.org/lkml/2007/8/2/24 Last known good : ? Submitter : Plamen Petrov <plamen.petrov(a)tk.ru.acad.bg> Caused-By : ? Handled-By : Len Brown <len.brown(a)intel.com> Status : problem is being debugged ALSA Subject : Master volume control broken References : http://lkml.org/lkml/2007/8/18/46 Last known good : ? Submitter : Thomas Meyer <thomas(a)m3y3r.de> Caused-By : Ivan N. Zlatev <contact(a)i-nz.net> commit 5d5d3bc3eddf2ad97b2cb090b92580e7fed6cee1 Handled-By : Ivan N. Zlatev <contact(a)i-nz.net> Status : problem is being debugged CPUFREQ Subject : ide problems: 2.6.22-git17 working, 2.6.23-rc1* is not References : http://lkml.org/lkml/2007/7/27/298 http://lkml.org/lkml/2007/7/29/371 Last known good : ? Submitter : dth <dth(a)dth.net> Caused-By : Len Brown <lenb(a)kernel.org> commit f79e3185dd0f8650022518d7624c876d8929061b Handled-By : Len Brown <lenb(a)kernel.org> Status : problem is being debugged Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

2 1

[alsa-devel] azx driver compilation query
by paul blakeley 31 Aug '07

31 Aug '07

Hi, I am trying to compile ALSA 1.0.14 and have compiled it but there are problems. How do you find out which ALSA driver is associated with a chipset? The chipset is ALC888, but when I run the ./configure script what do I pass in to the '--with-cards' parameter? I have compiled with 'hda-intel' but I dont think this correct? Can you help?

2 2

[alsa-devel] [PATCH mod-deps.c] Remove libs from toplevel driver list
by Eliot Blennerhassett 31 Aug '07

31 Aug '07

Add 2 more libraries to list that gets filtered out when building the list of cards for configure. (?Can't this be done automatically somehow?) diff -r 49001ef1e617 utils/mod-deps.c --- a/utils/mod-deps.c Thu Aug 30 10:22:35 2007 +0200 +++ b/utils/mod-deps.c Fri Aug 31 20:02:20 2007 +1200 @@ -169,6 +169,8 @@ static char *no_cards[] = { "SND_VERBOSE_PROCFS", "SND_VERBOSE_PRINTK", "SND_BIT32_EMUL", + "#SND_AD1848_LIB", + "#SND_CS4231_LIB", "#SND_SB_COMMON", "#SND_SB8_DSP", "#SND_SB16_DSP",

2 1

[alsa-devel] Virtual Surround, where to start
by Colin Nickerson 31 Aug '07

31 Aug '07

Hello everyone. I am a programmer but I do not have experience with ALSA. I would like to see if I can add software support for the SRS and other features of my M-Audio Revolution 7.1 card. How could I best start on this task? I am assuming that the card has some hardware acceleration for the SRS algorithms that I will need to access. thanks, colin

3 5

[alsa-devel] no sound with hda-intel, conexant venice chip on hp 2836
by Pieter-Augustijn Van Malleghem 31 Aug '07

31 Aug '07

hi, i've just installed debian unstable on an hp 2386ea. This is the sound card it uses: 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Hewlett-Packard Company Unknown device 30b2 The driver is snd-hda-intel and everything loads fine. Alsamixer does the job equally well. However, no actual sound is produced. It uses the Conexant Venice (cx20549) codec. Nothing special in dmesg. I tried loading the kernel with noacpi, but no changes. I attach alsa-info.txt. Any clues ? Thanks in advance, pieter-augustijn name=root&type=33&description=/tmp/alsa-info.txt&expiry=&s=Submit+Post&content= !!################################ !!ALSA Information Script v 0.4.30 !!################################ !!Script ran on: Wed Aug 15 17:19:13 CEST 2007 !!Linux Distribution !!------------------ Debian GNU/Linux lenny/sid \n \l !!Kernel Information !!------------------ Kernel release: 2.6.22-1-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.14 Library version: Utilities version: 1.0.14 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xd6400000 irq 22 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 103c:30b2 !!HDA-Intel Codec information !!--------------------------- Codec: Conexant CX20549 (Venice) Address: 0 Vendor Id: 0x14f15045 Subsystem Id: 0x103c30b2 Revision Id: 0x100100 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: [0x1e 0x1e] Pincap 0x0810014: OUT EAPD Detect Pin Default 0x95170110: [Fixed] Speaker at Int Top 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: [0x1e 0x1e] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x0221401f: [Jack] HP Out at Ext Front 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: [0x14 0x14] Pincap 0x08113c: IN OUT HP Detect Pin Default 0x62a1902e: [N/A] Mic at Sep Front 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 0x21451130: [Jack] SPDIF Out at Sep Rear Conn = Optical, Color = Black Pin-ctls: 0x00: Connection: 1 0x18 Node 0x14 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081124: IN Detect Pin Default 0x97a701f0: [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 0x54330121: [N/A] CD at Int Right Conn = ATAPI, 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: [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] [0x94 0x94] 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 !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 2007-08-15 16:56 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 24 2007-08-15 16:56 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 2007-08-15 16:56 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 17 2007-08-15 16:56 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 33 2007-08-15 16:03 /dev/snd/timer !!Aplay output !!------------ **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: Conexant Digital [Conexant Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Simple mixer control 'Master',0 Capabilities: pvolume pswitch pswitch-joined Playback channels: Front Left - Front Right Limits: Playback 0 - 43 Mono: Front Left: Playback 30 [70%] [on] Front Right: Playback 30 [70%] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 231 [91%] [-4.80dB] Front Right: Playback 231 [91%] [-4.80dB] Simple mixer control 'LineIn',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Ext Mic',0 Capabilities: volume pswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 23 Front Left: 9 [39%] [13.50dB] Playback [on] Front Right: 9 [39%] [13.50dB] Playback [on] Simple mixer control 'Int Mic',0 Capabilities: volume pswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 23 Front Left: 13 [57%] [19.50dB] Playback [on] Front Right: 13 [57%] [19.50dB] Playback [on] Simple mixer control 'IntMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] !!All Loaded Modules !!------------------ Module snd_hda_intel snd_pcm_oss snd_mixer_oss snd_pcm snd_timer snd soundcore snd_page_alloc arc4 ecb blkcipher ieee80211_crypt_wep nvidia ipv6 button ac battery dm_snapshot dm_mirror dm_mod firewire_sbp2 loop tsdev joydev ipw3945 hci_usb i2c_i801 sdhci ieee80211 ieee80211_crypt psmouse bluetooth i2c_core mmc_core serio_raw firmware_class iTCO_wdt intel_agp evdev ext3 jbd mbcache ide_cd cdrom sd_mod piix generic ide_core e100 ata_generic ahci firewire_ohci firewire_core crc_itu_t mii libata ehci_hcd scsi_mod uhci_hcd thermal processor fan

4 7

[alsa-devel] SIIG-USB SoundWave 7.1 Pro
by John Dey 30 Aug '07

30 Aug '07

Hi, I have the above referenced sound card. It is running on the latest version of CCRMA x86. Jack shows only stereo. How do I configure so that I can run 5 channels. I haven't tried yet to look for directions on line. If someone could provide me with a URL to the most up-to-date instructions I would be willing to give it a try. Thanks. John

2 4

[alsa-devel] This is a nudge. This is only a nudge. In the event of actual emergency...
by Eric S. Raymond 30 Aug '07

30 Aug '07

There is a user community of significant size hanging on your next point release. Users of recent ThinkPads, especially X61 and T61s. ALSA 1.0.14 doesn't cut it for the AD1984 Intel sound card on these machines; your head revision, however, does. This is a nudge to your release manager. Please ship a point release soon so .deb and RPM packages con become available quickly. Owners of many laptops, including my X61, will thank you. -- <a href="http://www.catb.org/~esr/">Eric S. Raymond</a> In every country and in every age, the priest has been hostile to liberty. He is always in alliance with the despot, abetting his abuses in return for protection to his own. -- Thomas Jefferson, 1814

6 7

[alsa-devel] [patch] ALC262 Input source bug on HP_BPC and HP_BPC_D7000
by zhejiang 30 Aug '07

30 Aug '07

Hi, I have a HP BPC D7000 machine and run fedora6 + alsa-1.0.14 on it. I found that when I used the front mic,I needed to choose the "Input source " as "Line" to record. Then I read the source code of alsa-1.0.14 and alsa-hg-20080828. I found that the capture_source items is wrong. I tried to fix and test them. It works well on my machine. When i read the source code, I found that the capture_source items of alc262_HP_capture_source[] are not suitable.In this patch,I fixed them, but i don't have the machine to test them. --- a/alsa-driver-hg20070828/alsa-kernel/pci/hda/patch_realtek.c 2007-08-24 08:00:06.000000000 +0800 +++ b/alsa-driver-hg20070828/alsa-kernel/pci/hda/patch_realtek.c 2007-08-29 12:47:30.000000000 +0800 @@ -7706,13 +7706,23 @@ .num_items = 5, .items = { { "Mic", 0x0 }, - { "Front Mic", 0x3 }, + { "Front Mic", 0x1 }, { "Line", 0x2 }, { "CD", 0x4 }, { "AUX IN", 0x6 }, }, }; +static struct hda_input_mux alc262_HP_D7000_capture_source = { + .num_items = 4, + .items = { + { "Mic", 0x0 }, + { "Front Mic", 0x2 }, + { "Line", 0x1 }, + { "CD", 0x4 }, + }, +}; + /* mute/unmute internal speaker according to the hp jack and mute state */ static void alc262_fujitsu_automute(struct hda_codec *codec, int force) { @@ -8313,7 +8323,7 @@ .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, - .input_mux = &alc262_HP_capture_source, + .input_mux = &alc262_HP_D7000_capture_source, }, [ALC262_HP_BPC_D7000_WL] = { .mixers = { alc262_HP_BPC_WildWest_mixer, @@ -8324,7 +8334,7 @@ .hp_nid = 0x03, .num_channel_mode = ARRAY_SIZE(alc262_modes), .channel_mode = alc262_modes, - .input_mux = &alc262_HP_capture_source, + .input_mux = &alc262_HP_D7000_capture_source, }, [ALC262_BENQ_ED8] = { .mixers = { alc262_base_mixer },

2 1

[alsa-devel] Time for ALSA 1.0.15-rc1?
by Takashi Iwai 29 Aug '07

29 Aug '07

Hi, How about considering about the 1.0.15-rc1 release? People tend to use still the old 1.0.14 release simply because it's THE last release. I'm too tired of answering "try HG version, the bug must have been fixed." We have already tons of changes, especially in the ALSA kernel/driver tree about 25,000 lines changes. That's high enough. Takashi

3 2

[alsa-devel] mic doesn't work with hda-intel/AD1983 driver
by Shuduo Sang 29 Aug '07

29 Aug '07

Hi, I am running ubuntu 7.04 on Dell Optiplex 745 with hda-intel card. The mic does not work now. I tried latest stable alsa driver. The chip is Analog Device AD1983. According to the document of alsa driver, there is no more model parameter can be set. Can anyone give me some information about that? Is it a defect of current driver code? Or something I missed? thanks, Regards, Sang Shuduo

2 4

[alsa-devel] Audiophile Delta 24/96 Midi Out continuously sends zeroes
by Romain Bossart 29 Aug '07

29 Aug '07

Hi, My audiophile 24/96 Midi Out sends zeroes continuously, so I can't use any instrument. I checked that with amidi by looking at the stream output of a connected instrument (midi out -> instrument -> midi in -> amidi). Zeroes are sent whether or not I try to send (normal) midi messages. I know it used some years ago (it was on win32 at that time). Midi In and the instrument do work perfectly. M-Audio answered me that they don't yet support linux. I'm using alsa 1.0.14 on debian/sid 2.6.22. Anyone has an idea? Thank you Romain

1 0

[alsa-devel] Audiophile Delta 24/96 Midi Out continuously sends zeroes
by Romain Bossart 28 Aug '07

28 Aug '07

Hi, My audiophile 24/96 Midi Out sends zeroes continuously, so I can't use any instrument. I checked that with amidi by looking at the stream output of a connected instrument (midi out -> instrument -> midi in -> amidi). Zeroes are sent whether or not I try to send (normal) midi messages. I know it used some years ago (it was on win32 at that time). Midi In and the instrument do work perfectly. M-Audio answered me that they don't yet support linux. I'm using alsa 1.0.14 on debian/sid 2.6.22. Anyone has an idea? Thank you Romain

1 0

Re: [alsa-devel] conexant alsa patch
by Tobin Davis 28 Aug '07

28 Aug '07

On Tue, 2007-08-28 at 15:20 +0200, Takashi Iwai wrote: > At Tue, 28 Aug 2007 00:27:39 +0000, > Hirakendu Das wrote: > > > > Hi. > > > > This regarding the conexant alsa patches authored by you. Firstly, thanks for > > the drivers :). Just wanted to say that if you can kindly look at the following > > minor issue. The master volume control in any mixer frontend (alsamixer or kmix > > etc) doesn't work in the latest patches. It used to work in the old patches. > > Strangely and thankfully, just changing the references of "Master Playback > > Volume" and "Master Playback Switch" in the file patch_conexant.c to simply > > "Playback Volume" and "Playback Switch" as in the old patch seems to make it > > work again as earlier. It works for me on my hp dv5000t notebook with conexant > > 5047. Could you just look into if its the case for all others. (Please see the > > attachments for details.) > > (Its of a bit of cosmetic value for me, as without it, I can't control volume > > using my notebook buttons.) > > PS : Is it true for other sound codecs too ? > > Can you try to build from HG tree? Then you can trace exactly which > commit breaks by bisecting the commits. > > > thanks, > > Takashi Preliminary tests against the 2.6.22 (alsa driver 1.0.14) breaks the automute on my test board (for some reason - still looking at this). I will also try against the tip to see if there are other major issues. -- Tobin Davis <tdavis(a)dsl-only.net>

1 1

[alsa-devel] [patch] kernel-sync
by Trent Piepho 28 Aug '07

28 Aug '07

Here's patch for alsa-kernel to sync bt87x.c with the git kernel, from commit cf784d554fdb94f21671830135dba56b507126cf Or maybe process is to wait for 2.6.23 to be released, and then sync to that? I have some more patch to bt87x, but I'm waiting for this to appear in alsa-kernel so I can base them off it.

2 1

[alsa-devel] echo-cancellation plugin advice?
by Robert Brewer 28 Aug '07

28 Aug '07

I'm writing a small speech input app for my laptop. My app is unusable when other sounds are being generated by the laptop, such as from a music playing app. For this reason, I'd like to have an echo cancellation plugin that would cancel out the music played on the laptop speakers when it is received by the mic. I think this should be doable since the music is playing through alsa. I've done some poking in the plugin documentation, and it's not clear to me how to architect this plugin. Ideally, I would route all my music playing apps through the echo cancelling plugin. I would also route the mic capture stream through the plugin. So the plugin must handle both playback and capture streams. Does the asym plugin hold the answers I need? Does anyone have any thoughts on how I might do this? There are several open source echo cancellation algorithms available, so I'm mainly asking for advice on implementing this as an alsa plugin. Thanks, Rob -- Robert W. Brewer

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Re: [alsa-devel] Altec Lansing FX5051 mixer
by Jeremy Nickurak 28 Aug '07

28 Aug '07

On Tue, Aug 28, 2007 at 01:43:51AM -0600, Jeremy Nickurak wrote: > On Tue, Aug 28, 2007 at 08:50:00AM +0200, Clemens Ladisch wrote: > > > bmaControls( 0) 0x15 > > > Mute > > > Bass > > > Treble > > > bmaControls( 1) 0x02 > > > Volume > > > bmaControls( 2) 0x02 > > > Volume > > > bmaControls( 3) 0x00 > > > bmaControls( 4) 0x00 > > > bmaControls( 5) 0x00 > > > bmaControls( 6) 0x00 > > > > These descriptors say that there is a two-channel volume control that > > applies to the front channels, but I guess it really affects all > > channels. > > > > Please show the output of "amixer contents" for this card. > > numid=3,iface=MIXER,name='Tone Control - Bass' > ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 > : values=5 > numid=4,iface=MIXER,name='Tone Control - Treble' > ; type=INTEGER,access=rw------,values=1,min=0,max=10,step=0 > : values=5 > 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 > > > > > > > ... > > > HID Device Descriptor: > > > bLength 9 > > > bDescriptorType 33 > > > bcdHID 1.00 > > > bCountryCode 0 Not supported > > > bNumDescriptors 1 > > > bDescriptorType 34 Report > > > wDescriptorLength 172 > > > Report Descriptor: (length is 172) > > > Item(Global): Usage Page, data= [ 0x0c ] 12 > > > Consumer > > > Item(Local ): Usage, data= [ 0x01 ] 1 > > > Consumer Control > > > Item(Main ): Collection, data= [ 0x01 ] 1 > > > Application > > > Item(Global): Usage Page, data= [ 0x0c ] 12 > > > Consumer > > > Item(Local ): Usage, data= [ 0xe2 ] 226 > > > Mute > > > Item(Local ): Usage, data= [ 0xe9 ] 233 > > > Volume Increment > > > ... > > > > This describes an input device that reports 'key' presses to the > > computer. I guess the control panel and remote control don't have a > > direct effect but expect the computer to explicitly change mixer > > controls when a key is pressed. > > Since it works fine in Windows without any non-ms software, would this > imply that this is somewhat standard behavior? > > > > > However, this shouldn't prevent alsamixer from working. > > > > It might be possible that the mixer controls don't have many steps and > > that the step size used by alsamixer is so small that the device rounds > > the setting back to the old position. Please try PageUp/Down or Home/ > > End instead of Up/Down keys. > > PgUp/PgDown don't do any better on any controls. I seem to be able to > drop the PCM volume to nothing in practice with End, but Home won't > restore it. Up/Down immediately brings it back up to 100%. The LED's on > the control reflect this btw, but alsamixer still reports 98%. > > -- > Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =- -- Jeremy Nickurak -= Email/Jabber: atrus(a)rifetech.com =-

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[alsa-devel] ALSA sound/core/pcm_lib.c:210: Unexpected hw_pointer value [1] (stream = 0, delta: -4096, max jitter = 8192): wrong interrupt acknowledge?
by Pharaoh . 28 Aug '07

28 Aug '07

I am getting this error at the start of song, I get this only once and the song plays quite well except some periodic minute/unnoticable clicks which occur often if period size is less. Is there any chance of this being the culprit i.e. causing clicks? I used to get the same messages after every buffer size previously, but after fixing my pointer callback this happens only once at the beginning. After reading the code I think that, this might not be the reason behind clicks, since this is just xrun debug thing, but this doesn't correspond to xruns directly, does it? I got the above error message even when there were no xruns. How serious is this issue? After referring web I collected that, this might happen because the period is not processed as fast as it should be. -pharaoh.

2 1

[alsa-devel] Noise at end of audio file
by Lopez Cruz, Misael 27 Aug '07

27 Aug '07

Hi, I listen some noise at the end of playback of an audio file, with some debug I noticed that the noise appears in the last two period of the playback. I tried changing the minimum number of period (it was 16 and changed to 32) but the noise appears even before. Any idea what could be wrong? Misael

2 5

[alsa-devel] Very elementary question regarding period size
by Pharaoh . 27 Aug '07

27 Aug '07

I am testing out my alsa driver for various period and buffer sizes, at some configurations it doesn't work so arises this very basic question. Is every sound card/hardware supposed to work at all the period sizes? The way I have written my driver, makes the DMA processing dependent on period size so the answer is obvious in my case, that it might not work for all period sizes. Does it hold true for all the cards? -pharaoh.

2 2

[alsa-devel] acl888 chipset
by paul blakeley 27 Aug '07

27 Aug '07

Can someone tell me who wrote the driver for the alc888 chipset. I am currently encountering problems with capturing data from the line-in. Think there maybe a problem with the driver.

2 1

[alsa-devel] [1/4] 2.6.23-rc3: known regressions v3
by Michal Piotrowski 25 Aug '07

25 Aug '07

Hi all, Here is a list of some known regressions in 2.6.23-rc3. Feel free to add new regressions/remove fixed etc. http://kernelnewbies.org/known_regressions List of Aces Name Regressions fixed since 21-Jun-2007 Adrian Bunk 9 Andi Kleen 5 Linus Torvalds 5 Andrew Morton 4 Al Viro 3 Alan Stern 3 Cornelia Huck 3 Jens Axboe 3 Tejun Heo 3 Unclassified Subject : 2.6.23-rc3-git1 crash/stuck on VIA CN700 system References : http://lkml.org/lkml/2007/8/20/174 Last known good : ? Submitter : Stefan Becker <Stefan.Becker(a)nokia.com> Caused-By : ? Handled-By : ? Status : unknown Subject : 2.6.23-rc2 cross compile regressions (alpha,xtensa) References : http://lkml.org/lkml/2007/8/6/43 Last known good : alpha: 2.6.22-git8 xtensa: 2.6.22-git6 Submitter : Jan Dittmer <jdi(a)l4x.org> Caused-By : ? Handled-By : xtensa: Christian Zankel <chris(a)zankel.net> Status : unknown Subject : console is messed up after resume from s2ram or switching to console from X References : http://lkml.org/lkml/2007/8/4/6 Last known good : ? Submitter : Jeff Chua <jeff.chua.linux(a)gmail.com> Caused-By : ? Handled-By : H. Peter Anvin <hpa(a)zytor.com> Antonino A. Daplas <adaplas(a)gmail.com> Workaround : "s2ram --force --acpi_sleep 1 --vbe_mode" Status : problem is being debugged Subject : konqueror suddenly vanishing, "konqueror: Fatal IO error: client killed" References : http://lkml.org/lkml/2007/7/22/86 Last known good : ? Submitter : Markus <lists4me(a)web.de> Caused-By : ? Handled-By : Ingo Molnar <mingo(a)elte.hu> Status : problem is being debugged ALSA Subject : Master volume control broken References : http://lkml.org/lkml/2007/8/18/46 Last known good : ? Submitter : Thomas Meyer <thomas(a)m3y3r.de> Caused-By : Ivan N. Zlatev <contact(a)i-nz.net> commit 5d5d3bc3eddf2ad97b2cb090b92580e7fed6cee1 Handled-By : ? Status : unknown Regards, Michal -- LOG http://www.stardust.webpages.pl/log/

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[alsa-devel] Audio quality issue - regular clicks are observed
by Pharaoh . 24 Aug '07

24 Aug '07

Hi While testing the audio driver using aplay in blocking mode, a periodic murmur is observed, this increases with the audio volume and occurs even when there are no issues like xrun etc. Aplay was used in normal read/write mode i.e. no mmap. The song can be heard properly i.e. with no noticable gaps aur hisses but the clicks are audible. I have tried various combinations of period size and buffer size but the quality doesn't seem to improve. I checked the latency between to consecutive calls of snd_pcm_write in aplay and log is as follows. I am having 16 periods of 1K size and from log it is seen that the latency is too much at the buffer boundary, i.e. higlighted and starred line in the logs. I think the clicks are caused due these huge delays since the clicks are also quite periodic. Any thoughts? #aplay -Dhw:0,0 --rate=44100 --format=S16 --channels=2 --period-size=1024 --buffer-size=16384 Learning.wav 1 2 Latency = 0.000000 3 Latency = 0.001805 4 Latency = 0.000741 5 Latency = 0.000674 6 Latency = 0.000643 7 Latency = 0.000668 8 Latency = 0.000625 9 Latency = 0.000611 10 Latency = 0.000647 11 Latency = 0.000627 12 Latency = 0.000626 13 Latency = 0.000635 14 Latency = 0.000631 15 Latency = 0.000616 16 Latency = 0.000645 17 Latency = 0.000634 18 Latency = 0.000958 19 Avg Latency is => 0.000737 20 21 Latency = 0.371909 /********Too much latency***********/ 22 Latency = 0.000767 23 Latency = 0.000701 24 Latency = 0.000696 25 Latency = 0.000657 26 Latency = 0.000677 27 Latency = 0.000667 28 Latency = 0.000673 29 Latency = 0.000674 30 Latency = 0.000627 31 Latency = 0.000675 32 Latency = 0.000880 33 Latency = 0.000726 34 Latency = 0.000682 35 Latency = 0.000705 Latency = 0.000648 37 Avg Latency is => 0.023898 38 39 Latency = 0.361167 /********Too much latency***********/ 40 Latency = 0.000744 41 Latency = 0.000692 42 Latency = 0.000678 43 Latency = 0.000672 44 Latency = 0.000637 45 Latency = 0.000696 46 Latency = 0.000668 47 Latency = 0.000674 48 Latency = 0.000885 49 Latency = 0.000710 50 Latency = 0.000670 51 Latency = 0.000663 52 Latency = 0.000654 53 Latency = 0.000667 54 Latency = 0.000676 55 Avg Latency is => 0.023222 56 57 Latency = 0.361260 /********Too much latency***********/ 58 Latency = 0.000710 59 Latency = 0.000703 60 Latency = 0.000685 61 Latency = 0.000671 62 Latency = 0.000665 63 Latency = 0.000796 64 Latency = 0.000714 65 Latency = 0.000692 66 Latency = 0.000704 67 Latency = 0.000695 -Pharaoh

2 4

[alsa-devel] please test the beta CMI8788 driver
by Clemens Ladisch 24 Aug '07

24 Aug '07

If you have a card based on the C-Media CMI8788 chip (Asound A-8788, Asus Xonar D2, Auzentech X-Meridian, Bgears b-Enspirer, Club3D Theatron DTS, HTOmega Claro, Razer Barracuda AC-1, or Sondigo Inferno), please help testing the new snd-cmi8788 driver. To test it, get the source from the Hg repository, or download this beta package: http://www.informatik.uni-halle.de/~ladischc/alsa-driver-1.0.14-cl1.tar.bz2 To compile, run "./configure", then "make", then, as root, "make install-modules". (If you don't want to compile all the other drivers, use "./configure --with-cards=cmi8788" instead.) What should work: * analog playback (2/4/6/8 channels) and capture * digital playback/capture (device name "plughw:CMI8788,1") * all sample rates (32-192 kHz) Not yet implemented: * front panel playback/capture * capture source selection * SPDIF controls (for AC-3, copyright bit etc.) * MIDI Known bugs: * PCM volume on Asus Xonar doesn't work Unknown bugs: * That's for you to find out! I'm not sure about the channel order. Please try speaker-test -D hw:CMI8788,0 -c 8 -t wav and report if it works correctly. *WARNING*: I don't have the hardware; I cannot run this driver myself. It's quite possible that it crashes your computer or that it causes the speakers to fall onto your cat. ;-) Clemens

1 0

[alsa-devel] Unknown card info
by Jorge Peixoto de Morais Neto 24 Aug '07

24 Aug '07

When I execute sudo modprobe snd_bt87x, I can't capture audio from my tv capture card (mencoder segfaults). It only works when I execute sudo modprobe snd_bt87x load_all=1. Then it seems to work ok. When I issue sudo modprobe snd_bt87x load_all=1, the following lines appear in dmesg: unknown card 0x878-0x1554:0x4011, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> I am obeying the request and mailing this information to you. I have no idea what is the correct digital_rate option; all I know is that the default seems to work ok. I have a cable connecting the sound output of the tv card to the line in of my sound card; it seems that when wathing tv through mplayer this is the mechanism used (and it works ok). However, when recording tv through mencoder, it seems to use snd_bt87x (in fact, it does not work if snd_bt87x is not loaded), and it also works ok. The card option I pass when loading the module bttv is card=139. According to /usr/src/linux/Documentation/video4linux/CARDLIST.bttv, this corresponds to the card Prolink PixelView PlayTV MPEG2 PV-M4900 In fact, there is a sticker in the PCB that reads (if my eyes are not failing me) PixelView PV-M4900 FM.RC The full output of dmesg is attached to this email. If you need any additional information please contact me directly, since I will now unsubscribe to this list. Some possibly useful information: jorge@jorge ~ $ uname -a Linux jorge 2.6.22-gentoo-r3 #1 PREEMPT Tue Aug 14 18:51:06 BRT 2007 i686 AMD Athlon(tm) XP 2600+ AuthenticAMD GNU/Linux jorge@jorge ~ $ cat /etc/modprobe.d/tv options bttv card=139 automute=0 options snd_bt87x load_all=1 jorge@jorge ~ $ arecord -l **** List of CAPTURE Hardware Devices **** card 0: CMI8738MC6 [C-Media PCI CMI8738-MC6], device 0: CMI8738-MC6 [C-Media PCI DAC/ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: CMI8738MC6 [C-Media PCI CMI8738-MC6], device 2: CMI8738-MC6 [C-Media PCI IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Bt878 [Brooktree Bt878], device 0: Bt87x Digital [Bt87x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: Bt878 [Brooktree Bt878], device 1: Bt87x Analog [Bt87x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 jorge@jorge ~ $ sudo lspci Password: 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400/KT600 AGP] Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8235 PCI Bridge 00:07.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) 00:0b.0 Multimedia video controller: Brooktree Corporation Bt878 Video Capture (rev 11) 00:0b.1 Multimedia controller: Brooktree Corporation Bt878 Audio Capture (rev 11) 00:0c.0 Multimedia audio controller: C-Media Electronics Inc CM8738 (rev 10) 00:10.0 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) 00:10.1 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) 00:10.2 USB Controller: VIA Technologies, Inc. VT82xxxxx UHCI USB 1.1 Controller (rev 80) 00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82) 00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge 00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) 01:00.0 VGA compatible controller: nVidia Corporation NV17 [GeForce4 MX 440] (rev a3) -- Software is like sex: it is better when it is free. --Linus Torvalds

3 3

[alsa-devel] Unable to open plughw:0,0 from user program.
by stan 24 Aug '07

24 Aug '07

Hi, I am unable to use plughw:0,0 as a device string while calling err = snd_pcm_open (&alsa_dev, hw_from_default, SND_PCM_STREAM_PLAYBACK, 0); The same call works with hw_from_default == 'default', but fails when it is 'plughw:0,0'. The error is cannot open audio device "plughw:0,0" (Device or resource busy) I am using the latest snapshot of alsa-lib, alsa-lib20070821. I tried debugging but I cannot trace through the macros. As near as I can tell there have been no changes to alsa.conf, and this worked just fine before the latest upgrade alsa package alsa-lib.1.0.14-3 Is this a deliberate change in alsa that is known about, and is there a way to still get the functionality? Thanks.

3 7

[alsa-devel] hw_params function and OSS emulation
by Timur Tabi 24 Aug '07

24 Aug '07

I'm working on an ASoC driver, and I noticed that with OSS emulation enabled, my snd_pcm_ops.hw_params and and snd_pcm_ops.hw_free are called multiple times when an OSS driver uses the OSS emulation. In my case, .hw_params is called *four* times, each time with a different DMA buffer size and number of periods. The problem is that my driver allocates a DMA buffer in my .hw_params function. For now, I have it deallocate the buffers at the top of the function and then allocate new ones based on the new hw_params values. This is really annoying. So I have a few questions: 1) Is there any way this can be fixed? Can't the OSS emulation code figure out what it needs and wait until it's done before it calls .hw_params? As a solution to the DMA buffer deallocate/reallocate hack I'm using, would it be okay to move the actual allocations to snd_pcm_ops.prepare()? My .hw_params function will collect the relevant data and keep them in some private structure. Then when .prepare() is called, I do the actual buffer allocation. 2) Can I assume that .prepare() is called only once? 3) Can I assume that .hw_params() is never called after .prepare() is called? -- Timur Tabi Linux Kernel Developer @ Freescale

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[alsa-devel] Intel-HDA ALC262 (again)
by Andy 23 Aug '07

23 Aug '07

Alas, I spoke too soon. The headphone jack and internal speakers work as expected, until I adjust the main volume (labeled in alsamixer as 'front') at which point both the internal speakers and headphones play! Unplugging the headphones and reconnecting them makes everything OK again. Any ideas? (patch against 2.6.22 that makes my chip mostly work is attached). I guess that alc262_hippo_automute() needs to be called somewhere it isn't, but I couldn't see where. Thanks again! Andy >Yep, I'm happy to report that does indeed fix it! >Thank-you Takashi! :) >Andy. > At Wed, 25 Jul 2007 16:51:25 +0100 (BST), > Andy wrote: > > > > Hello again Takashi, > > > > I put in some printks, and indeed the function > does > > run and it correctly detects that there are no > > headphones present, but the speakers are not > unmuted. > > > > (the 'mute' variable both times is reported as > being > > 0x80 after it is set with snd_hda_codec_amp_read) > > Ah, it means that this intiailization gets called > before restoring the > mixer elements. Since sony model doesn't have the > mixer element for > the NID 0x14 that is linked with 0x15, 0x14 isn't > restored properly in > the end. > > How about the patch below? It's a bit too ad hoc, > but better than > buggy. > ___________________________________________________________ Yahoo! Answers - Got a question? Someone out there knows the answer. Try it now. http://uk.answers.yahoo.com/

2 3

[alsa-devel] I want to improve Sigma-Tel STAC9227 driver / How to implement support for multiple ADC
by Maxim Levitsky 23 Aug '07

23 Aug '07

Hello, I am a user of Sigma-Tel STAC9227 intel hda based chip. I have studied its documentation, and found that alsa driver doesn't support number of its features. 1) it has "analog loop-back" which means it can mix sound just before it enters ADC0 to output of DAC0 (headphones/front). For me it is the most important feature, since I have a tv card with analog output. 2) it has the ability to "swap" left/right channels, and this can be very useful for LFE/center channel 3) It has a VolumeKnob "widget" , that allows to decrease volume of all DACs _after_ their own volumes were applied, and I think I turn it to "master volume" Those things below aren't supported by windows driver ether, but that doesn't mean we can't support that :-) 4) this chip has 3 ADC, each capable recording from different source, but alsa uses only first. 5) I have 5.1 configuration, so DAC1 is unused. It is possible to connect it to "front" mic, and get 7.1 sound card I think about a option "Use front mic as output", for 3Stack configuration 1,2,3 are easy to implement, since those are just mixer switches. But I have a question about 4: Now the driver has buggy support for second ADC, it record from it, but fails to connect it to any input source, and unmute it. I can fix that and trivially add support for third ADC, but current driver makes all ADCs a sub-streams of a single PCM device. I think that sub-streams are meant to be used only by playback PCMs that have hardware support of mixing, so I feel that is is better to make each ADC to be a separate PCM device, with separate mixer controls (enabled/input source/gain), and I want to ask you whenever you agree with me. Also I noticed that dsnoop chokes on record streams with more that one substream, it fails with error, when I try to open second arecord. Best regards, Maxim Levitsky

2 4

Re: [alsa-devel] [Alsa-user] Impossible to make subwoofer work on a Dell XPS M1710w
by Michael Gerdau 23 Aug '07

23 Aug '07

[crossposting to alsa-devel as adviced in ALSA-Configuration.txt] > The model option is _the_ important option for HD-audio driver. > It overrides the configuration preset. As default, the driver reads > the PCI (subsystem) IDs to identify which configuration preset to > use. If not found, the driver does a guess work from the BIOS setup > in most cases. > > Now, since many laptops are shipped with broken BIOS and the guess > work is imperfect, you often face the famous problem -- "sound > doesn't work" (great words that explain everything). Then you can > pass model option for the preset designed for anohter hardware > (e.g. model=thinkpad) to make the driver probe as if it's such a > hardware. If you are lucky, your device is similar with others, and > the driver works with such a preset. If not, we'd need to hack > another preset for matching with your device. > > The list of available model option is found in > ALSA-Configuration.txt. So it seems the Dell XPS M1710 isn't yet listed among the known models and also does not seem to be detected properly (as of hg20070816). 'lspci -v' gives: ... 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 01) Subsystem: Dell Unknown device 01ce Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at efffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) ... 'lspci -nv' gives: ... 00:1b.0 Class 0403: 8086:27d8 (rev 01) Subsystem: 1028:01ce Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at efffc000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 Capabilities: [100] Virtual Channel Capabilities: [130] Unknown (5) ... Anything else I could do to help getting it (fully) supported ? Best, Michael -- Vote against SPAM - see http://www.politik-digital.de/spam/ !tagline Michael Gerdau email: mgerdau(a)tiscali.de GPG-keys available on request or at public keyserver

5 15

[alsa-devel] Tascam US-428 USB Problems
by Michael Bourgeous 23 Aug '07

23 Aug '07

Hello, I hope this reaches the Tascam driver developer... I have a Tascam US-428 which worked great with my Athlon XP, but does not work at all with my new system, an Athlon 64 X2. I've tried using the regular alsa driver, as well as the hwdep audio, and jackd always crashes with a SIGSEGV, with the following message in the kernel log: [ 2538.003798] ALSA /home/nitrogen/src/alsa-driver-1.0.14rc3/usb/usx2y/usbusx2yaudio.c:148: should not be here with counts=42 [ 2538.003806] ALSA /home/nitrogen/src/alsa-driver-1.0.14rc3/usb/usx2y/usbusx2yaudio.c:91: active frame status -70. Most probably some hardware problem. n I did some searching through source code, and found that the frame status of -70 (ECOMM) indicates an OHCI buffer overflow error. As soon as the usx2y driver receives this it bails out of whatever it's doing, and jackd crashes. I've tried changing the usx2y driver to ignore such errors, and while there are audio glitches when such an error occurs, jackd no longer crashes and the audio keeps playing. This leads to question one: Is there any reason to bail out instead of trying to recover from errors? I've tried all the different USB ports on my motherboard, different USB cables, and running through a hub instead of a direct connection. None of these things helped. Though the usx2y message says "Most probably some hardware problem", I think this is something that is software related, as other USB devices have no problems. So here is question two: what kinds of problems, software or hardware, would lead to a buffer overflow error, and can I fix this in the driver code? Thanks, Mike Bourgeous

2 6

Re: [alsa-devel] ESI reply!! (MAYA44 datasheet request)
by Claus Riethmueller 23 Aug '07

23 Aug '07

Hi James, please let me say first that I am not intending this as any personal comment as I am pretty sure you are not the only person with similar thoughts about the discussion. Yet, I have to say that it is exactly the somewhat ignorant attitude that turns us and quite a number of other vendors "off" in supporting the ALSA developer community better. It is what I have refered to in my email to Piotr that he cut & pasted to this mailing list (without asking for permission btw.). I have been following the developments at ALSA since many years now and worked with a number of different developers for different products to get them working properly. Many of our products are supported just fine under ALSA and both the developers and the users seemed to appreciate that fact. Unfortunatly your comments show your obvious lack of understanding about how some companies are working in this industry and what sort of approach is needed by you guys to get better support from us. This is why I used the term "culture crash" in my email. > So, basically, ESI cannot help us. > I am amazed that they can write any sort of driver without some sort of > datasheet. Well ... frankly, it should not be your concern on how we are developing drivers for our hardware designs (which are ours). What you wrote is a cheap comment and I do not see how it could be an appropriate comment if you look at everything what I wrote and not just the introduction. The way (format, style and most important in our case: language) we store our internal company data is no ones concern except mine and the concern of my employees. We are not developing our products with the purpose to share this information with others. It is not part of our business model. We are not a chipset vendor and we are not making designs that other companies are supposed to use or work with. Our data is used inside our company and nowhere else. Every document that goes out to the public must go through a process that is aprooved by a number of people, written by the actual responsible person, in our case partially even translated into another language by someone who is not an expert on the issue and only then could be sent out. Unless we have to do that to achieve something we want to achieve, we simply can't do that. This might be different with other companies you've been in contact with (I saw you developed major parts of the drivers for various Creative devices and can probably guess that this is where main parts of your experience come from) where there are more people available that could get and go through such tasks. I could now add information about company and market sizes, etc. but that would make everything only more complicated (that's the type of discussion that is nice in combination with a few beers in a pub at some late evening ...). What I did in the introduction of my email to Piotr was to mention this fact about our situation without going into any details. I am probably saying more about how our company works than what many other vendors ever would disclose about themselve. I hoped it would help to improve the understanding of "the other side". Your comments are disapointing and puzzle me a little bit, but unfortunatly I have to say that I have expected something like this from my past experience. Luckily not all ALSA contributors / developers share your views as the fact that many of our products (as well as quite a number of products from other vendors) are supported in ALSA at this moment without having the document (you think) you need, speaks for itself I might add. > But I do know it does happen, because I have received datasheets that > were only written years after a particular sound card was first sold. > I would like to stress, that the datasheet does not have to be perfect. > We would be quite happy if it has spelling mistakes in, and some wrongly > labeled registers. > I have received datasheets from other manufactures where some engineer > has just scribbled some notes on the back of a piece of paper. At least you acknowledge that this happens which I think is quite positive. I am not saying that what you mention here resembles what goes on in our company, but it is not that far from the actual situation. If you add that our engineers are in Korea and make their notes mostly in Korea, you get a little bit closer to the real picture. Of course we have documents but they are not ready in any way that I would approove sending them out to the public. > Now, if ESI cannot even provide this, we should highlight ESI in RED in > the sound card matrix. As you can guess, it was this sentence that prompted me to reply directly to the list. In my opinion, your comment is unfair and simply ignoring major parts of what I wrote in my mail to Piotr. I have shown a certain level of commitment and your response basically tells me that you are not even interested ... if more people would share your views, ALSA would have made no development at all in the last 5 years I guess. Our engineers are all happy to answer questions and provide assistance and help if required. As I said, we are happy to be in contact with an individual developer who is actually doing the development. It is a lot less time consuming and for us a lot more simple (actually, I believe it is more simple for the developer on the ALSA end as well but I realize some people might disagree here). I'm sorry if this long mail has distracted a few people from the daily discussion about fixing bugs and improving code, but I am sure there are enough people that understand where I am coming from and what I wrote here. I do not want to start a long discussion here, so I also do not intend to comment on the whole issue further on the mailing list. Contact me directly and we can go from there (as I mentioned in my mail to Piotr). Best regards, Claus Riethmueller Managing Director ESI Audiotechnik GmbH

6 7

[alsa-devel] aplay giving underruns in non blocking mode
by Pharaoh . 22 Aug '07

22 Aug '07

Hi I am tesing my alsa driver for an arm based board. aplay works well when it is used in blocking mode, but I see underruns when I switch to non-blocking mode. What could be the reason? I have tried to remove all latencies in driver by using minimal locking etc, but it doesn't seem to make any difference. Also, the audio quality degenerates gradually in non-blocking mode. -Pharaoh.

1 1

[alsa-devel] ESI reply!! (MAYA44 datasheet request)
by Piotr Makowski 22 Aug '07

22 Aug '07

Hi Piotr, thank you very much for your email. The data sheet and information you request does not exist in any format that we could provide and frankly, we are way too busy with our normal work to prepare something like that in a way that would be of general use and would resemble what you require in your request. I am well aware of the general request and I know the text you sent us pretty well ... I have seen it many times before in various versions, usually copied via cut & paste and sent to us in one way or another ... and actually I am a bit tired of it. Unfortunatly we believe that the text and the way the request is sent to us is not really useful and not tailored at all to the special requirements that are common in this industry (pro audio / higher level sound card manufacturers) and that make this industry very different from your auverage computer component manufacturer. It's my personal experience, following the ALSA developer community for years, that this sort of basic cultural understanding is only slowly changing step-by-step but at a much slower pace than it could happen. Unfortunatly the majority of ALSA developers still seem to be "computer guys" and not "music guys" ... causing all sorts of problems that won't help to resolve the culture crash between the different parties. Most "music" companies work in a different way than typical IT companies. Our business are much smaller, and time of every single employee is extremly critical for us. I guess this all may sound ignorant to you, but please understand that I know what I am talking about to a pretty good extent (you can actually look up my name on the ALSA site where I am listed under contributors - http://www.alsa-project.org/main/index.php/ALSA_contributors - so I have experience on both sides). Again, we are happy to be in contact with the individual developer who is actually doing the development. If you volunteer to do the development yourself and/or if you know someone who would do it, please inform me and we can go from there. Please don't get me wrong, I understand what you want and where you are coming from, but this direct contact with the developer is a basic requirement for us - sorry. Best regards, Claus Riethmueller Managing Director ESI Audiotechnik GmbH

2 2

[alsa-devel] include/mixer.h missing from doxygen
by Rolland Dudemaine 22 Aug '07

22 Aug '07

Hi, include/mixer.h seems to be missing from the doxygen input file list. Should be added to the INPUT list in doc/doxygen.cfg. Btw, I'm pretty new do ALSA, why is the Mixer interface marked legacy in a couple places ? Best Regards, --Rolland Dudemaine

2 1

[alsa-devel] unknown card info [snd-bt87x]
by Vincent Minder 22 Aug '07

22 Aug '07

Hello, After loading the snd-bt87x module on my Fedora 7 x86_64 box, here is what I found in the dmesg output: unknown card 0x878-0xbd11:0x1200, using default rate 32000 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> The card is a rather old and cheap "PINNACLE systems - Studio PCTV rave" The chip on the board is from Conexant: CONEXANT Fusion 878A (chip markings: 25878-13, E32673.1, 0023 KOREA) The card standard is PAL B/G. I think this model supports only MONO audio. It has TV-in, Audio-out (mini jack), Composite-in, S-VIDEO-in. TV playback is working all right (tested with "kdetv" and "tvtime"). I have sound as long as I use the Audio-out connector and bridge it to my motherboard's integrated audio Line-in . However, I didn't manage to use the internal audio capabities of that board. So I have no clue about the "correct digital_rate". (I must admit I am not familiar with these technologies at all.) Here is the output of "arecord -l" regarding this card: $ arecord -l **** List of CAPTURE Hardware Devices **** (...) card 2: Bt878 [Brooktree Bt878], device 0: Bt87x Digital [Bt87x Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 2: Bt878 [Brooktree Bt878], device 1: Bt87x Analog [Bt87x Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 I am using the stock Alsa package of Fedora 7 x86_64 (fully updated distribution): alsa-lib-1.0.14-1.fc7 I hope this report will be useful to you. Sincerely, Vincent Minder (Belgium)

2 1

[alsa-devel] [PATCH] fix cmi8788 config (was: Re: [alsa-cvslog] alsa-lib: cmi8788: add alsa-lib config)
by Thierry Vignaud 22 Aug '07

22 Aug '07

Clemens Ladisch <clemens(a)ladisch.de> writes: > changeset: 2314:3bf9c6dec8c8 > tag: tip > user: cladisch > date: Wed Aug 22 09:42:13 2007 +0200 > files: src/conf/cards/CMI8788.conf > description: > cmi8788: add alsa-lib config > > Add a .conf file to enable dmix/dsnoop and softvol for CMI8788. > > Using dmix helps mask the bug that all audio is forced to 48 kHz. :-) as usual, here's the missing bit:

2 1

[alsa-devel] problems writing pcm driver
by Trent Piepho 22 Aug '07

22 Aug '07

I'm trying to write an ALSA driver, but there are some thing I'm not clear on and it's not working as well as I'd like. Atomicity: trigger, pointer, and ack are atomic with respect to themselves. e.g., two copies of the trigger callback can't be running at the same time. That much is clear. But are they atomic with respect to each other? Can the trigger callback run at the same time as the pointer callback? How about the atomic callbacks with respect to the non-atomic ones? Can trigger run at the same time as prepare? The pointer callback: What exactly is the current hardware position? My hardware has a counter that tells me how many periods have been DMAed into the buffer. Suppose my period size is 256 frames and I'm doing audio capture. If pointer is called before any DMA transfers have completed, do I return 0? Now suppose it's called after one period of audio has been DMAed into the buffer. Do I return 255 or 256? After two periods have been received, should I return 511 or 0? Why is pointer called so many times? Here's a log of callbacks and irqs my driver is getting. I'm using the cycle counter to give them microsecond timestamps. My period is 256 frames at 48kHz, so each period is 5333 us. There are only two periods in the buffer. Time (us) | Callback or IRQ ---------------------------------------------- 0.000 trigger - start DMA 134.513 pointer - count 1, returned 0 160.254 IRQ - not calling snd_pcm_period_elapsed 5313.030 IRQ - called snd_pcm_period_elapsed 5318.808 pointer - count 2, returned 256 IRQ hander finished 5348.309 pointer - count 2, returned 256 10627.526 IRQ - called snd_pcm_period_elapsed 10633.075 pointer - count 3, returned 0 IRQ hander finished 10659.755 pointer - count 3, returned 0 15941.173 IRQ - called snd_pcm_period_elapsed 15946.601 pointer - count 4, returned 256 IRQ hander finished 15969.743 pointer - count 4, returned 256 [and so on] The pointer callback is called only 134 us after the trigger, before the first IRQ or a period of data could possibly be received. Why is this done? To get the position in the buffer to start at? I assumed the data should start at the beginning of the buffer, but is this not the case? You'll notice the first IRQ is received at only 160 us. The hardware generates an IRQ when each period _starts_, not when one ends. So the first IRQ is ignored, the second IRQ tells me the first period has finished, and so on. The hardware counter is the number of periods started; I have to subtract one to get the number of periods finished. When the IRQ handler calls snd_pcm_period_elapsed(), that calls the pointer callback. That's what I expected from what the docs said. But about 30 us after the pointer callback is called the first time, after the irq handler has exited, it's called again. What is the purpose of this? Is it ok that I return the same value as the previous time it was called?

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Re: [alsa-devel] ESI MAYA44 PCI datasheet request
by Ash Willis 22 Aug '07

22 Aug '07

Hi Piotr, In general, I think this is good. Some points though... * I don't think you should demand firmware source. Asking for that might be seen as quite invasive. It might be a bonus to have, but I don't see anyone hacking on it. Their firmware works, It's a driver that we need. Does this card even use firmware? I guess... * Linking to a mirror probably isn't too smart if they think their infos are going to distributed all over the web. Sorry, that was my fault. I'm an idiot. I think the correct link should be http://www-old.alsa-project.org/alsa/ftp/datasheets/ * I would offer a fall back. I would propose that the approach you have mentioned is in their best interest however if they are unwilling to distribute the information freely, then the next best thing would be to find one developer that is able, willing and possesses the hardware. That's not ideal. I would prefer the second approach to no driver at all, as would you, I'm sure. However, I don't speak for anyone else when I say that. * I'm sure they know the ins and outs of open source but you should confirm with them the terms of this arrangement. NDAs are basically incompatible with this whole thing because, usually, releasing the source will breach the NDA. Make sure they know that the driver being produced is open source and licensed under the GNU GPL. Ash > ----- Original Message ----- > From: Piotrek <oponek(a)gmail.com> > To: alsa-devel(a)alsa-project.org, "Pawe³ Wolniewicz" <admin(a)linux-muzyka.ixion.pl> > Subject: [alsa-devel] ESI MAYA44 PCI datasheet request > Date: Tue, 21 Aug 2007 19:50:27 +0200 > > > I would like toshow you the letter that I am going to send to the ESI > company. I am asking you to accept this letter because I am going to send it > on behalf of the ALSA development team. Any suggestions will be welcome. > > > Dear Sir/Madam, > > In reply to the letter sent 14.07.2007: > > 2007/8/14, ESI UK Support <uksupport(a)esi-audio.com>: > > Dear Piotr > > ESI do not provide and have no plans for Linux drivers for our products > However > ESI are happy to provide /share information to an Individual developer > within the Linux community > That is able and capable to develope drivers > and wishes to develope drivers > > Regards > UK Support > ESI Auditechnik > > > > I would like to ask you to share the information about ESI MAYA44 PCI card > needed to write drivers on behalf of the ALSA development team. ALSA team > needs all the information that you give your programmers to create drivers > for the device. We would like to ask you for: > > - a full datasheet containing IO port layout, register descriptions and > commands, > - ESI MAYA44 PCI firmware, > - EIS MAYA44 PCI firmware source code, > - permission for the firmware to be distributed with ALSA, > - source code for currnet windows drivers would be very helpful but isn't > essential. > > There are many peope that are willing to write the driver for MAYA44 and > because of the way Open-Source software and drivers are developed, in the > interest of it's quality, it is important that the information about the > card are accesable for all of the ALSA developers. We ask for your permision > to place that information on the ALSA server for developers. Thanks to that > any qualified person will have acces to it. Many other companies provided > that kind of information to us this way before: > http://ftp.mj2.org/pub/alsa/datasheets/ > > I am sure that adding this card to the list of the suported cards under the > linux operating system will cause the number of satisfied ESI card users to > increase. Thank you for your co-operation. > > > -- > Best Regards, > Piotr Makowski > ALSA Development Team > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > > ----- Original Message ----- > From: Piotrek <oponek(a)gmail.com> > To: alsa-devel(a)alsa-project.org, "Pawe³ Wolniewicz" <admin(a)linux-muzyka.ixion.pl> > Subject: [alsa-devel] ESI MAYA44 PCI datasheet request > Date: Tue, 21 Aug 2007 19:50:27 +0200 > > > I would like toshow you the letter that I am going to send to the ESI > company. I am asking you to accept this letter because I am going to send it > on behalf of the ALSA development team. Any suggestions will be welcome. > > > Dear Sir/Madam, > > In reply to the letter sent 14.07.2007: > > 2007/8/14, ESI UK Support <uksupport(a)esi-audio.com>: > > Dear Piotr > > ESI do not provide and have no plans for Linux drivers for our products > However > ESI are happy to provide /share information to an Individual developer > within the Linux community > That is able and capable to develope drivers > and wishes to develope drivers > > Regards > UK Support > ESI Auditechnik > > > > I would like to ask you to share the information about ESI MAYA44 PCI card > needed to write drivers on behalf of the ALSA development team. ALSA team > needs all the information that you give your programmers to create drivers > for the device. We would like to ask you for: > > - a full datasheet containing IO port layout, register descriptions and > commands, > - ESI MAYA44 PCI firmware, > - EIS MAYA44 PCI firmware source code, > - permission for the firmware to be distributed with ALSA, > - source code for currnet windows drivers would be very helpful but isn't > essential. > > There are many peope that are willing to write the driver for MAYA44 and > because of the way Open-Source software and drivers are developed, in the > interest of it's quality, it is important that the information about the > card are accesable for all of the ALSA developers. We ask for your permision > to place that information on the ALSA server for developers. Thanks to that > any qualified person will have acces to it. Many other companies provided > that kind of information to us this way before: > http://ftp.mj2.org/pub/alsa/datasheets/ > > I am sure that adding this card to the list of the suported cards under the > linux operating system will cause the number of satisfied ESI card users to > increase. Thank you for your co-operation. > > > -- > Best Regards, > Piotr Makowski > ALSA Development Team > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains

3 2

[alsa-devel] ESI MAYA44 PCI datasheet request
by Piotrek 21 Aug '07

21 Aug '07

I would like toshow you the letter that I am going to send to the ESI company. I am asking you to accept this letter because I am going to send it on behalf of the ALSA development team. Any suggestions will be welcome. Dear Sir/Madam, In reply to the letter sent 14.07.2007: 2007/8/14, ESI UK Support <uksupport(a)esi-audio.com>: Dear Piotr ESI do not provide and have no plans for Linux drivers for our products However ESI are happy to provide /share information to an Individual developer within the Linux community That is able and capable to develope drivers and wishes to develope drivers Regards UK Support ESI Auditechnik I would like to ask you to share the information about ESI MAYA44 PCI card needed to write drivers on behalf of the ALSA development team. ALSA team needs all the information that you give your programmers to create drivers for the device. We would like to ask you for: - a full datasheet containing IO port layout, register descriptions and commands, - ESI MAYA44 PCI firmware, - EIS MAYA44 PCI firmware source code, - permission for the firmware to be distributed with ALSA, - source code for currnet windows drivers would be very helpful but isn't essential. There are many peope that are willing to write the driver for MAYA44 and because of the way Open-Source software and drivers are developed, in the interest of it's quality, it is important that the information about the card are accesable for all of the ALSA developers. We ask for your permision to place that information on the ALSA server for developers. Thanks to that any qualified person will have acces to it. Many other companies provided that kind of information to us this way before: http://ftp.mj2.org/pub/alsa/datasheets/ I am sure that adding this card to the list of the suported cards under the linux operating system will cause the number of satisfied ESI card users to increase. Thank you for your co-operation. -- Best Regards, Piotr Makowski ALSA Development Team

1 0

[alsa-devel] HDA - Add more support for Dell systems.
by Tobin Davis 21 Aug '07

21 Aug '07

This patch adds support for more Dell systems with Sigmatel/IDT codecs. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] ESI MAYA44 PCI - ESI wants to collaborate!
by Piotrek 21 Aug '07

21 Aug '07

Referring to the lack of the official ALSA support of ESI MAYA44 PCI (in fact, the card does not work) I would like to ask I there are any plans to get this card working with ALSA? I would be thankful to be able to run my soundcard properly under Linux. There are also a lot of people that would appreciate your work and are waiting for maya44 pci drivers for linux: http://www.esiforum.com/show.asp?db=esi&t=esi1&mode=3&msg=793 I wrote a letter to the UK ESI SUPPORT asking if they were going to write drivers for their products for Linux systems as well. There is a reply: " Dear Piotr ESI do not provide and have no plans for Linux drivers for our products However ESI are happy to provide /share information to an Individual developer within the Linux community That is able and capable to develope drivers and wishes to develope drivers Regards UK Support ESI Auditechnik " It means that ESI will collaborate with developers who would like to write such driver and share the documentation. Again the question: Are there any plans to make ESI MAYA44 PCI card working with ALSA? I'm not a programmer but I will help you as much as I could. -- Best Regards, Piotr Makowski

4 4

[alsa-devel] Intel HDA / AD1986A problems
by Peter Skensved 21 Aug '07

21 Aug '07

Configuration : Lenovo 3000 N100 laptop with Intel HDA / Anlog Devices AD1986A soundcard running CentOS5.0 with 2.6.18-8.1.8.el5 kernel ( uses 1.0.12rc1 based alsa driver ). The driver works with the model=laptop-eapd option except for a few things : 1) Plugging in an external headset will not mute the internal speakers ( has to be done manually in alsa-mixer ) 2) The PC speaker ( bell ) appears to be permanently muted. 3) The external microphone input does not appear to be working no matter what I do. 4) When recording from the internal microphone the level is very, very low and almost drowns in static noise. Any easy way to fix the above ? Number 4 in particular ? I've tinkered with backporting what I thought were relevant differences between the latest patch_analog.c and the one used in the kernel I have (alsa-driver 1.0.12rc1 ) but I see no difference. Any other way to tweak or reconfigure the hardware ? Enable microphone boost ?? Looking at /proc/asound/card0/codec#0 makes me believe that the hardware has several unused or permanently disabled features. This particular laptop has a model number which differs slightly from the one listed in patch_analog.c ( 0768-6KU instead of 0768-4JU ). Not sure if is relevant. And finally a ( somewhat ) related question : Is it possible to get the sound card to work with the slmodem package to drive the built in modem ??? ============================================================================================ Partial output from lspci -nv is attached below together with stuff from /proc/asound : 00:1b.0 0403: 8086:27d8 (rev 02) Subsystem: 17aa:2066 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 233 Region 0: Memory at b0000000 (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: 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) more /proc/asound/cards 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xb0000000 irq 233 more /proc/asound/devices 2: : timer 3: : sequencer 4: [ 0- 0]: digital audio playback 5: [ 0- 0]: digital audio capture 6: [ 0] : control more /proc/asound/modules 0 snd_hda_intel more /proc/asound/card0/codec#0 Codec: Analog Devices AD1986A Address: 0 Vendor Id: 0x11d41986 Subsystem Id: 0x17aa2066 Revision Id: 0x100500 Default PCM: rates 0x07f, bits 0x0e, types 0x1 Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Node 0x02 [Audio Output] wcaps 0x30311: Stereo Digital PCM: rates 0x060, bits 0x02, types 0x5 Connection: 2 0x01* 0x06 Node 0x03 [Audio Output] wcaps 0x44d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1d 0x1d] Power: 0x0 Node 0x04 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Power: 0x0 Node 0x05 [Audio Output] wcaps 0x40d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Power: 0x0 Node 0x06 [Audio Input] wcaps 0x100511: Stereo PCM: rates 0x07f, bits 0x06, types 0x1 Power: 0x0 Connection: 1 0x12 Node 0x07 [Audio Mixer] wcaps 0x200101: Stereo Connection: 8 0x03 0x09 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x08 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x07 Node 0x09 [Audio Mixer] wcaps 0x20010e: Mono Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80] [0x80] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80] Connection: 2 0x04 0x05 Node 0x0a [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x07* 0x04 0x05 Node 0x0b [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x07* 0x04 Node 0x0c [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x04* 0x07 Node 0x0d [Audio Selector] wcaps 0x300101: Stereo Connection: 2 0x05* 0x08 Node 0x0e [Audio Selector] wcaps 0x300100: Mono Connection: 2 0x08* 0x11 Node 0x0f [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 8 0x1f* 0x20 0x1d 0x1d 0x27 0x28 0x29 0x2a Node 0x10 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x20* 0x1c 0x1f Node 0x11 [Audio Selector] wcaps 0x300941: Stereo Connection: 2 0x0f* 0x2b Node 0x12 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x8f 0x8f] Connection: 8 0x11 0x22 0x00 0x21 0x10 0x07* 0x08 0x23 Node 0x13 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x1d 0x1d] Connection: 1 0x11 Node 0x14 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Connection: 1 0x23 Node 0x15 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x22 Node 0x16 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Connection: 1 0x21 Node 0x17 [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x15 0x15] Connection: 1 0x10 Node 0x18 [Audio Selector] wcaps 0x30010c: Mono Amp-Out Amp-Out caps: ofs=0x0f, nsteps=0x0f, stepsize=0x0b, mute=1 Amp-Out vals: [0x8f] Connection: 2 0x19* 0x24 Node 0x19 [Beep Generator Widget] wcaps 0x700000: Mono Node 0x1a [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x06 0x06] Pincap 0x081f: OUT HP Detect Pin Default 0x02214021: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green Pin-ctls: 0xc0: OUT HP Connection: 1 0x0a Node 0x1b [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x06 0x06] Pincap 0x081001f: OUT HP EAPD Detect Pin Default 0x81114011: [Fixed] Speaker at Ext Rear Conn = 1/8, Color = Green Pin-ctls: 0x40: OUT Connection: 1 0x0b Node 0x1c [Pin Complex] wcaps 0x400185: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0837: IN OUT Detect Pin Default 0x41013012: [N/A] Line Out at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x40: OUT Connection: 1 0x0c Node 0x1d [Pin Complex] wcaps 0x400985: Stereo Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x081737: IN OUT Detect Pin Default 0x41019015: [N/A] Line Out at Ext Rear Conn = 1/8, Color = Pink Pin-ctls: 0x40: OUT Connection: 1 0x0d Node 0x1e [Pin Complex] wcaps 0x400104: Mono Amp-Out Amp-Out caps: ofs=0x1f, nsteps=0x1f, stepsize=0x05, mute=1 Amp-Out vals: [0x80] Pincap 0x0810: OUT Pin Default 0x501700f0: [N/A] Speaker at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x40: OUT Connection: 1 0x0e Node 0x1f [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0xc2a190f0: [Both] Mic at Ext Front Conn = 1/8, Color = Pink Pin-ctls: 0x24: IN Node 0x20 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x081727: IN Detect Pin Default 0x418130f0: [N/A] Line In at Ext Rear Conn = 1/8, Color = Blue Pin-ctls: 0x20: IN Node 0x21 [Pin Complex] wcaps 0x400081: Stereo Pincap 0x0827: IN Detect Pin Default 0x509700f0: [N/A] Aux at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x22 [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x903300f0: [Fixed] CD at Int N/A Conn = ATAPI, Color = Unknown Pin-ctls: 0x20: IN Node 0x23 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90b70130: [Fixed] Telephony at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x24 [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x90f701f0: [Fixed] Other at Int N/A Conn = Analog, Color = Unknown Pin-ctls: 0x20: IN Node 0x25 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x0810: OUT Pin Default 0x014511f0: [Jack] SPDIF Out at Ext Rear Conn = Optical, Color = Black Pin-ctls: 0x40: OUT Connection: 1 0x02 Node 0x26 [Power Widget] wcaps 0x500500: Mono Power: 0x0 Connection: 8 0x07* 0x08 0x13 0x14 0x15 0x16 0x17 0x18 Node 0x27 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x1d Node 0x28 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1f 0x20 Node 0x29 [Audio Mixer] wcaps 0x200101: Stereo Connection: 2 0x1d 0x20 Node 0x2a [Audio Mixer] wcaps 0x200101: Stereo Connection: 3 0x1f 0x1d 0x20 Node 0x2b [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x0f more /proc/asound/card0/codec#1 Codec: Generic 11c1 ID 1040 Address: 1 Vendor Id: 0x11c11040 Subsystem Id: 0x11c10001 Revision Id: 0x100200 more /proc/asound/card0/pcm0c/info card: 0 device: 0 subdevice: 0 stream: CAPTURE id: AD198x Analog name: AD198x Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 more /proc/asound/card0/pcm0p/info card: 0 device: 0 subdevice: 0 stream: PLAYBACK id: AD198x Analog name: AD198x Analog subname: subdevice #0 class: 0 subclass: 0 subdevices_count: 1 subdevices_avail: 1 amixer scontents Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 12 [39%] [on] Front Right: Playback 12 [39%] [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 29 [94%] [on] Front Right: Playback 29 [94%] [on] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [on] Front Left: Playback 29 [94%] [on] Front Right: Playback 29 [94%] [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 15 [100%] [on] Front Right: Capture 15 [100%] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Internal Mic',0 Capabilities: pvolume pswitch cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Mono Limits: Playback 0 - 31 Mono: Capture [off] Front Left: Playback 15 [48%] [on] Front Right: Playback 15 [48%] [on] ======================================================================================= Peter Skensved Email : peter at SNO dot Phy dot QueensU dot CA Dept. of Physics, Queen's University, Kingston, Ontario, Canada

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Re: [alsa-devel] ESI MAYA44 PCI - ESI wants to collaborate!
by Ash Willis 21 Aug '07

21 Aug '07

Hi Piotrek, Thanks for being so active in trying to improve ALSA support. I've written a couple of these drivers too but someone better than me should do this ;) especially if ESI is only willing to discuss this matter with one developer. Besides, I don't fancy paying £70 for the opportunity. The people at ESI will know what they need to provide: A full datasheet containing things like IO port layout, register descriptions and commands. Source code for current Windows drivers would be helpful but not essential. If the card needs firmware, then we'll need that too. It's also not essential that the source code for the firmware is provided but I think that's probably preferred as well. In either case, ESI will need to give permission for the firmware to be distributed for it to be well supported by ALSA. Otherwise, users will be responsible for extracting the firmware from their Windows drivers, and that's just not nice. Like I said, ESI will know will they need to provide: that is to say all the information they gave to their programmers to let them create the current drivers for this device. It is clear by the ESI response that they are willing to release the information on the provision that a good driver will be produced thereby making them look good. That is fair enough. I would suggest to ESI, however, that to truly create the maximum possibility of this happening, they could allow the datasheet to be held on the ALSA server. By doing this, the relevant information would be available to any developer who wanted to create the driver or make further improvements. You can point them to the address below if they don't think that sounds like something a sane company would do. http://ftp.mj2.org/pub/alsa/datasheets/ Anyway, good luck and my advice would be to try and get the info first and find a developer later. I don't know if ESI will do that though. There will always be people, floating around, who will code this up, but there's no guaranty that the company will always be as happy and friendly as they were to you :) Ash > ----- Original Message ----- > From: Piotrek <oponek(a)gmail.com> > To: "Adrian McMenamin" <adrianmcmenamin(a)gmail.com> > Subject: Re: [alsa-devel] ESI MAYA44 PCI - ESI wants to collaborate! > Date: Mon, 20 Aug 2007 22:53:53 +0200 > > > 2007/8/20, Adrian McMenamin <adrianmcmenamin(a)gmail.com>: > > > > On 20/08/07, Takashi Iwai <tiwai(a)suse.de> wrote: > > > At Sat, 18 Aug 2007 09:28:34 +0200, > > > Piotrek wrote: > > > > > > > > Referring to the lack of the official ALSA support of ESI MAYA44 > > PCI > > > > (in fact, the card does not work) I would like to ask I there are any > > plans > > > > to get this card working with ALSA? I would be thankful to be able to > > run my > > > > soundcard properly under Linux. There are also a lot of people that > > would > > > > appreciate your work and are waiting for maya44 pci drivers for linux: > > > > http://www.esiforum.com/show.asp?db=esi&t=esi1&mode=3&msg=793 > > > > I wrote a letter to the UK ESI SUPPORT asking if they were going > > to > > > > write drivers for their products for Linux systems as well. There is a > > > > reply: > > > > > > > > " > > > > Dear Piotr > > > > > > > > ESI do not provide and have no plans for Linux drivers for our > > products > > > > However > > > > ESI are happy to provide /share information to an Individual developer > > > > within the Linux community > > > > That is able and capable to develope drivers > > > > and wishes to develope drivers > > > > > > > > Regards > > > > UK Support > > > > ESI Auditechnik > > > > " > > > > > > > > It means that ESI will collaborate with developers who would like to > > write > > > > such driver and share the documentation. > > > > > > > > Again the question: Are there any plans to make ESI MAYA44 PCI card > > working > > > > with ALSA? > > > > > > AFAIK, no plan. We don't have this hardware for developing. > > > > > > > > > > I have no idea what this card is like, but if they would be prepared > > to send me one, I'd be happy to take the project on. I've written one > > alsa driver - that will shortly be in the stable kernel - so can point > > to that. > > > > Of course, other developers will probably be able to do the job at a > > greater speed - but I would be happy to take it on as a pro bono > > project. > > > > I am in the UK. > > > > Thank you all for your replies. I think we should wait a while for the > response of the rest of the ALSA dev team and then we would be able to > choose the best person for this quest and decide what to do with it > together. > However, we need to > specify what kind of information do we exactly need form ESI before we can > start talking with them. Please write your suggestions. > When we will know what we need from ESI we can start working on the > drivers. I will be glad to be able to test your work. > > -- > Regards, > Piotr Makowski > _______________________________________________ > Alsa-devel mailing list > Alsa-devel(a)alsa-project.org > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel > -- We've Got Your Name at http://www.mail.com ! Get a FREE E-mail Account Today - Choose From 100+ Domains

1 0

[alsa-devel] Alsa-HOWTO
by Wallace, Brooke 21 Aug '07

21 Aug '07

Anyone know what happend to this howto? Seems like it had some good information that I can no longer find on the Offical ALSA Wiki... The URL was: http://www.alsa-project.org/alsa-doc/alsa-howto/c1706.htm <http://www.alsa-project.org/alsa-doc/alsa-howto/c1706.htm> before the switch to the new wiki was completed. -Brooke

2 1

[alsa-devel] Getting a driver to work
by Adrian McMenamin 20 Aug '07

20 Aug '07

Apologies for asking this old chestnut ... but I have upgraded my busybox/buildroot setup on my Dreamcast and now the AICA driver fails - looks pretty much like a failure to load the firmware. Information on the internet seems rather mixed up and out of date - some of it referring to 2.2 kernels etc... What is needed to load firmware - is it udev or hotplug (or is hotplug obsolete), and what role do the alsa libs have in all this? I went through all this two years ago but have forgotten again :( Thanks Adrian

2 2

[alsa-devel] VIA82XX on MSI K78MM Front Channel Problem
by Daniel Persch 20 Aug '07

20 Aug '07

Dear Developers, I'm using the on-board AC97 in 6-channel mode with analog output. However everything works fine except front channels: - front right is playing front left - front left is mute In stereo mode: - front right is playing front left - center is playing front right I was wandering whether someone could tell me where in the source I can play with the channel-settings/slots or if there is a solution for my issue. Or should I create a Bug-Report? Regards in advance, Daniel lspci output for the sound card: 00:11.5 0401: 1106:3059 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 60) Alsamodule: via82xx

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[alsa-devel] [PATCH] dbri: driver cleanup
by Krzysztof Helt 20 Aug '07

20 Aug '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch fixes white spaces, spelling and formatting to conform closer to the coding standard of the kernel. It contains few fixes pointed out by the checkpatch.pl script. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- In case this email is mangled, the original patch is attached. diff -urp linux-2.6.23.2/sound/sparc/dbri.c linux-2.6. 23/sound/sparc/dbri.c --- linux-2.6.23.2/sound/sparc/dbri.c 2007-07-09 01:32:16. 000000000 +0200 +++ linux-2.6.23/sound/sparc/dbri.c 2007-08-17 10:48:31. 000000000 +0200 @@ -8,18 +8,18 @@ * Copyright (C) 1997 Rudolf Koenig (rfkoenig(a)immd4.informatik. uni-erlangen.de) * Copyright (C) 1998, 1999 Brent Baccala (baccala(a)freesoft.org) * - * This is the lowlevel driver for the DBRI & MMCODEC duo used for ISDN & AUDIO - * on Sun SPARCstation 10, 20, LX and Voyager models. + * This is the low level driver for the DBRI & MMCODEC duo used for ISDN & AUDIO + * on Sun SPARCStation 10, 20, LX and Voyager models. * * - DBRI: AT&T T5900FX Dual Basic Rates ISDN Interface. It is a 32 channel * data time multiplexer with ISDN support (aka T7259) * Interfaces: SBus,ISDN NT & TE, CHI, 4 bits parallel. * CHI: (spelled ki) Concentration Highway Interface (AT&T or Intel bus ?). * Documentation: - * - "STP 4000SBus Dual Basic Rate ISDN (DBRI) Tranceiver" from + * - "STP 4000SBus Dual Basic Rate ISDN (DBRI) Transceiver" from * Sparc Technology Business (courtesy of Sun Support) * - Data sheet of the T7903, a newer but very similar ISA bus equivalent - * available from the Lucent (formarly AT&T microelectronics) home + * available from the Lucent (formerly AT&T microelectronics) home * page. * - http://www.freesoft.org/Linux/DBRI/ * - MMCODEC: Crystal Semiconductor CS4215 16 bit Multimedia Audio Codec @@ -27,21 +27,21 @@ * Documentation: from the Crystal Semiconductor home page. * * The DBRI is a 32 pipe machine, each pipe can transfer some bits between - * memory and a serial device (long pipes, nr 0-15) or between two serial - * devices (short pipes, nr 16-31), or simply send a fixed data to a serial + * memory and a serial device (long pipes, no. 0-15) or between two serial + * devices (short pipes, no. 16-31), or simply send a fixed data to a serial * device (short pipes). - * A timeslot defines the bit-offset and nr of bits read from a serial device. + * A timeslot defines the bit-offset and no. of bits read from a serial device. * The timeslots are linked to 6 circular lists, one for each direction for * each serial device (NT,TE,CHI). A timeslot is associated to 1 or 2 pipes * (the second one is a monitor/tee pipe, valid only for serial input). * * The mmcodec is connected via the CHI bus and needs the data & some - * parameters (volume, output selection) timemultiplexed in 8 byte + * parameters (volume, output selection) time multiplexed in 8 byte * chunks. It also has a control mode, which serves for audio format setting. * * Looking at the CS4215 data sheet it is easy to set up 2 or 4 codecs on - * the same CHI bus, so I thought perhaps it is possible to use the onboard - * & the speakerbox codec simultanously, giving 2 (not very independent :-) + * the same CHI bus, so I thought perhaps it is possible to use the on-board + * & the speakerbox codec simultaneously, giving 2 (not very independent :-) * audio devices. But the SUN HW group decided against it, at least on my * LX the speakerbox connector has at least 1 pin missing and 1 wrongly * connected. @@ -56,6 +56,8 @@ #include <sound/driver.h> #include <linux/interrupt.h> #include <linux/delay.h> +#include <linux/irq.h> +#include <linux/io.h> #include <sound/core.h> #include <sound/pcm.h> @@ -64,8 +66,6 @@ #include <sound/control.h> #include <sound/initval.h> -#include <asm/irq.h> -#include <asm/io.h> #include <asm/sbus.h> #include <asm/atomic.h> @@ -76,7 +76,8 @@ MODULE_SUPPORTED_DEVICE("{{Sun,DBRI}}"); 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 */ +/* Enable this card */ +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; module_param_array(index, int, NULL, 0444); MODULE_PARM_DESC(index, "Index value for Sun DBRI soundcard."); @@ -104,7 +105,7 @@ static char *cmds[] = { "SSP", "CHI", "NT", "TE", "CDEC", "TEST", "CDM", "RESRV" }; -#define dprintk(a, x...) if(dbri_debug & a) printk(KERN_DEBUG x) +#define dprintk(a, x...) if (dbri_debug & a) printk(KERN_DEBUG x) #else #define dprintk(a, x...) do { } while (0) @@ -131,7 +132,7 @@ struct cs4215 { }; /* - * Control mode first + * Control mode first */ /* Time Slot 1, Status register */ @@ -219,7 +220,7 @@ static struct { /* Time Slot 7, Input Setting */ #define CS4215_LG(v) v /* Left Gain Setting 0xf: 22.5 dB */ #define CS4215_IS (1<<4) /* Input Select: 1=Microphone, 0=Line */ -#define CS4215_OVR (1<<5) /* 1: Overrange condition occurred */ +#define CS4215_OVR (1<<5) /* 1: Over range condition occurred */ #define CS4215_PIO0 (1<<6) /* Parallel I/O 0 */ #define CS4215_PIO1 (1<<7) @@ -285,7 +286,7 @@ struct dbri_pipe { /* Per stream (playback or record) information */ struct dbri_streaminfo { struct snd_pcm_substream *substream; - u32 dvma_buffer; /* Device view of Alsa DMA buffer */ + u32 dvma_buffer; /* Device view of ALSA DMA buffer */ int size; /* Size of DMA buffer */ size_t offset; /* offset in user buffer */ int pipe; /* Data pipe used */ @@ -341,11 +342,11 @@ struct snd_dbri { /* DBRI Reg1 - Mode and Interrupt Register - defines. (Page 18) */ #define D_LITTLE_END (1<<8) /* Byte Order */ #define D_BIG_END (0<<8) /* Byte Order */ -#define D_MRR (1<<4) /* Multiple Error Ack on SBus (readonly) */ -#define D_MLE (1<<3) /* Multiple Late Error on SBus (readonly) */ -#define D_LBG (1<<2) /* Lost Bus Grant on SBus (readonly) */ -#define D_MBE (1<<1) /* Burst Error on SBus (readonly) */ -#define D_IR (1<<0) /* Interrupt Indicator (readonly) */ +#define D_MRR (1<<4) /* Multiple Error Ack on SBus (read only) */ +#define D_MLE (1<<3) /* Multiple Late Error on SBus (read only) */ +#define D_LBG (1<<2) /* Lost Bus Grant on SBus (read only) */ +#define D_MBE (1<<1) /* Burst Error on SBus (read only) */ +#define D_IR (1<<0) /* Interrupt Indicator (read only) */ /* DBRI Reg2 - Parallel IO Register - defines. (Page 18) */ #define D_ENPIO3 (1<<7) /* Enable Pin 3 */ @@ -376,11 +377,11 @@ struct snd_dbri { #define D_CDM 0xe /* CHI Data mode command */ /* Special bits for some commands */ -#define D_PIPE(v) ((v)<<0) /* Pipe Nr: 0-15 long, 16-21 short */ +#define D_PIPE(v) ((v)<<0) /* Pipe No.: 0-15 long, 16-21 short */ /* Setup Data Pipe */ /* IRM */ -#define D_SDP_2SAME (1<<18) /* Report 2nd time in a row value rcvd */ +#define D_SDP_2SAME (1<<18) /* Report 2nd time in a row value received */ #define D_SDP_CHANGE (2<<18) /* Report any changes */ #define D_SDP_EVERY (3<<18) /* Report any changes */ #define D_SDP_EOL (1<<17) /* EOL interrupt enable */ @@ -419,7 +420,7 @@ struct snd_dbri { #define D_TS_NONCONTIG (3<<10) /* Non contiguous mode */ #define D_TS_ANCHOR (7<<10) /* Starting short pipes */ #define D_TS_MON(v) ((v)<<5) /* Monitor Pipe */ -#define D_TS_NEXT(v) ((v)<<0) /* Pipe Nr: 0-15 long, 16-21 short */ +#define D_TS_NEXT(v) ((v)<<0) /* Pipe no.: 0-15 long, 16-21 short */ /* Concentration Highway Interface Modes */ #define D_CHI_CHICM(v) ((v)<<16) /* Clock mode */ @@ -435,7 +436,7 @@ struct snd_dbri { #define D_NT_NBF (1<<16) /* Number of bad frames to loose framing */ #define D_NT_IRM_IMM (1<<15) /* Interrupt Report & Mask: Immediate */ #define D_NT_IRM_EN (1<<14) /* Interrupt Report & Mask: Enable */ -#define D_NT_ISNT (1<<13) /* Configfure interface as NT */ +#define D_NT_ISNT (1<<13) /* Configure interface as NT */ #define D_NT_FT (1<<12) /* Fixed Timing */ #define D_NT_EZ (1<<11) /* Echo Channel is Zeros */ #define D_NT_IFA (1<<10) /* Inhibit Final Activation */ @@ -455,7 +456,7 @@ struct snd_dbri { #define D_TEST_RAM(v) ((v)<<16) /* RAM Pointer */ #define D_TEST_SIZE(v) ((v)<<11) /* */ #define D_TEST_ROMONOFF 0x5 /* Toggle ROM opcode monitor on/off */ -#define D_TEST_PROC 0x6 /* MicroProcessor test */ +#define D_TEST_PROC 0x6 /* Microprocessor test */ #define D_TEST_SER 0x7 /* Serial-Controller test */ #define D_TEST_RAMREAD 0x8 /* Copy from Ram to system memory */ #define D_TEST_RAMWRITE 0x9 /* Copy into Ram from system memory */ @@ -464,12 +465,12 @@ struct snd_dbri { #define D_TEST_DUMP 0xe /* ROM Dump */ /* CHI Data Mode */ -#define D_CDM_THI (1<<8) /* Transmit Data on CHIDR Pin */ -#define D_CDM_RHI (1<<7) /* Receive Data on CHIDX Pin */ -#define D_CDM_RCE (1<<6) /* Receive on Rising Edge of CHICK */ -#define D_CDM_XCE (1<<2) /* Transmit Data on Rising Edge of CHICK */ -#define D_CDM_XEN (1<<1) /* Transmit Highway Enable */ -#define D_CDM_REN (1<<0) /* Receive Highway Enable */ +#define D_CDM_THI (1 << 8) /* Transmit Data on CHIDR Pin */ +#define D_CDM_RHI (1 << 7) /* Receive Data on CHIDX Pin */ +#define D_CDM_RCE (1 << 6) /* Receive on Rising Edge of CHICK */ +#define D_CDM_XCE (1 << 2) /* Transmit Data on Rising Edge of CHICK */ +#define D_CDM_XEN (1 << 1) /* Transmit Highway Enable */ +#define D_CDM_REN (1 << 0) /* Receive Highway Enable */ /* The Interrupts */ #define D_INTR_BRDY 1 /* Buffer Ready for processing */ @@ -493,9 +494,9 @@ struct snd_dbri { #define D_INTR_CHI 36 #define D_INTR_CMD 38 -#define D_INTR_GETCHAN(v) (((v)>>24) & 0x3f) -#define D_INTR_GETCODE(v) (((v)>>20) & 0xf) -#define D_INTR_GETCMD(v) (((v)>>16) & 0xf) +#define D_INTR_GETCHAN(v) (((v) >> 24) & 0x3f) +#define D_INTR_GETCODE(v) (((v) >> 20) & 0xf) +#define D_INTR_GETCMD(v) (((v) >> 16) & 0xf) #define D_INTR_GETVAL(v) ((v) & 0xffff) #define D_INTR_GETRVAL(v) ((v) & 0xfffff) @@ -533,41 +534,42 @@ struct snd_dbri { #define D_P_31 31 /* */ /* Transmit descriptor defines */ -#define DBRI_TD_F (1<<31) /* End of Frame */ -#define DBRI_TD_D (1<<30) /* Do not append CRC */ -#define DBRI_TD_CNT(v) ((v)<<16) /* Number of valid bytes in the buffer */ -#define DBRI_TD_B (1<<15) /* Final interrupt */ -#define DBRI_TD_M (1<<14) /* Marker interrupt */ -#define DBRI_TD_I (1<<13) /* Transmit Idle Characters */ -#define DBRI_TD_FCNT(v) (v) /* Flag Count */ -#define DBRI_TD_UNR (1<<3) /* Underrun: transmitter is out of data */ -#define DBRI_TD_ABT (1<<2) /* Abort: frame aborted */ -#define DBRI_TD_TBC (1<<0) /* Transmit buffer Complete */ -#define DBRI_TD_STATUS(v) ((v)&0xff) /* Transmit status */ - /* Maximum buffer size per TD: almost 8Kb */ +#define DBRI_TD_F (1 << 31) /* End of Frame */ +#define DBRI_TD_D (1 << 30) /* Do not append CRC */ +#define DBRI_TD_CNT(v) ((v) << 16) /* Number of valid bytes in the buffer */ +#define DBRI_TD_B (1 << 15) /* Final interrupt */ +#define DBRI_TD_M (1 << 14) /* Marker interrupt */ +#define DBRI_TD_I (1 << 13) /* Transmit Idle Characters */ +#define DBRI_TD_FCNT(v) (v) /* Flag Count */ +#define DBRI_TD_UNR (1 << 3) /* Underrun: transmitter is out of data */ +#define DBRI_TD_ABT (1 << 2) /* Abort: frame aborted */ +#define DBRI_TD_TBC (1 << 0) /* Transmit buffer Complete */ +#define DBRI_TD_STATUS(v) ((v) & 0xff) /* Transmit status */ + /* Maximum buffer size per TD: almost 8KB */ #define DBRI_TD_MAXCNT ((1 << 13) - 4) /* Receive descriptor defines */ -#define DBRI_RD_F (1<<31) /* End of Frame */ -#define DBRI_RD_C (1<<30) /* Completed buffer */ -#define DBRI_RD_B (1<<15) /* Final interrupt */ -#define DBRI_RD_M (1<<14) /* Marker interrupt */ -#define DBRI_RD_BCNT(v) (v) /* Buffer size */ -#define DBRI_RD_CRC (1<<7) /* 0: CRC is correct */ -#define DBRI_RD_BBC (1<<6) /* 1: Bad Byte received */ -#define DBRI_RD_ABT (1<<5) /* Abort: frame aborted */ -#define DBRI_RD_OVRN (1<<3) /* Overrun: data lost */ -#define DBRI_RD_STATUS(v) ((v)&0xff) /* Receive status */ -#define DBRI_RD_CNT(v) (((v)>>16)&0x1fff) /* Valid bytes in the buffer */ +#define DBRI_RD_F (1 << 31) /* End of Frame */ +#define DBRI_RD_C (1 << 30) /* Completed buffer */ +#define DBRI_RD_B (1 << 15) /* Final interrupt */ +#define DBRI_RD_M (1 << 14) /* Marker interrupt */ +#define DBRI_RD_BCNT(v) (v) /* Buffer size */ +#define DBRI_RD_CRC (1 << 7) /* 0: CRC is correct */ +#define DBRI_RD_BBC (1 << 6) /* 1: Bad Byte received */ +#define DBRI_RD_ABT (1 << 5) /* Abort: frame aborted */ +#define DBRI_RD_OVRN (1 << 3) /* Overrun: data lost */ +#define DBRI_RD_STATUS(v) ((v) & 0xff) /* Receive status */ +#define DBRI_RD_CNT(v) (((v) >> 16) & 0x1fff) /* Valid bytes in the buffer */ /* stream_info[] access */ /* Translate the ALSA direction into the array index */ #define DBRI_STREAMNO(substream) \ - (substream->stream == \ + (substream->stream == \ SNDRV_PCM_STREAM_PLAYBACK? DBRI_PLAY: DBRI_REC) /* Return a pointer to dbri_streaminfo */ -#define DBRI_STREAM(dbri, substream) &dbri->stream_info[DBRI_STREAMNO(substream)] +#define DBRI_STREAM(dbri, substream) \ + &dbri->stream_info[DBRI_STREAMNO(substream)] static struct snd_dbri *dbri_list; /* All DBRI devices */ @@ -612,18 +614,18 @@ Since the DBRI can run in parallel with synchronization present themselves. The method implemented here is only use of the dbri_cmdwait() to wait for execution of batch of sent commands. -A circular command buffer is used here. A new command is being added +A circular command buffer is used here. A new command is being added while another can be executed. The scheme works by adding two WAIT commands after each sent batch of commands. When the next batch is prepared it is added after the WAIT commands then the WAITs are replaced with single JUMP -command to the new batch. The the DBRI is forced to reread the last WAIT -command (replaced by the JUMP by then). If the DBRI is still executing +command to the new batch. The the DBRI is forced to reread the last WAIT +command (replaced by the JUMP by then). If the DBRI is still executing previous commands the request to reread the WAIT command is ignored. Every time a routine wants to write commands to the DBRI, it must -first call dbri_cmdlock() and get pointer to a free space in -dbri->dma->cmd buffer. After this, the commands can be written to -the buffer, and dbri_cmdsend() is called with the final pointer value +first call dbri_cmdlock() and get pointer to a free space in +dbri->dma->cmd buffer. After this, the commands can be written to +the buffer, and dbri_cmdsend() is called with the final pointer value to send them to the DBRI. */ @@ -657,7 +659,7 @@ static void dbri_cmdwait(struct snd_dbri * Lock the command queue and returns pointer to a space for len cmd words * It locks the cmdlock spinlock. */ -static s32 *dbri_cmdlock(struct snd_dbri * dbri, int len) +static s32 *dbri_cmdlock(struct snd_dbri *dbri, int len) { /* Space for 2 WAIT cmds (replaced later by 1 JUMP cmd) */ len += 2; @@ -680,7 +682,7 @@ static s32 *dbri_cmdlock(struct snd_dbri * * Lock must be held before calling this. */ -static void dbri_cmdsend(struct snd_dbri * dbri, s32 * cmd,int len) +static void dbri_cmdsend(struct snd_dbri *dbri, s32 *cmd, int len) { s32 tmp, addr; static int wait_id = 0; @@ -700,16 +702,17 @@ static void dbri_cmdsend(struct snd_dbri s32 *ptr; for (ptr = dbri->cmdptr; ptr < cmd+2; ptr++) - dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr); + dprintk(D_CMD, "cmd: %lx:%08x\n", + (unsigned long)ptr, *ptr); } else { s32 *ptr = dbri->cmdptr; dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr); ptr++; dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr); - for (ptr = dbri->dma->cmd; ptr < cmd+2; ptr++) { - dprintk(D_CMD, "cmd: %lx:%08x\n", (unsigned long)ptr, *ptr); - } + for (ptr = dbri->dma->cmd; ptr < cmd+2; ptr++) + dprintk(D_CMD, "cmd: %lx:%08x\n", + (unsigned long)ptr, *ptr); } #endif @@ -723,7 +726,7 @@ static void dbri_cmdsend(struct snd_dbri } /* Lock must be held when calling this */ -static void dbri_reset(struct snd_dbri * dbri) +static void dbri_reset(struct snd_dbri *dbri) { int i; u32 tmp; @@ -746,7 +749,7 @@ static void dbri_reset(struct snd_dbri * } /* Lock must not be held before calling this */ -static void dbri_initialize(struct snd_dbri * dbri) +static void dbri_initialize(struct snd_dbri *dbri) { s32 *cmd; u32 dma_addr; @@ -763,7 +766,7 @@ static void dbri_initialize(struct snd_d spin_lock_init(&dbri->cmdlock); /* - * Initialize the interrupt ringbuffer. + * Initialize the interrupt ring buffer. */ dma_addr = dbri->dma_dvma + dbri_dma_off(intr, 0); dbri->dma->intr[0] = dma_addr; @@ -801,7 +804,7 @@ list ordering, among other things. The here interface closely with the transmit and receive interrupt code. */ -static int pipe_active(struct snd_dbri * dbri, int pipe) +static int pipe_active(struct snd_dbri *dbri, int pipe) { return ((pipe >= 0) && (dbri->pipes[pipe].desc != -1)); } @@ -811,20 +814,22 @@ static int pipe_active(struct snd_dbri * * Called on an in-use pipe to clear anything being transmitted or received * Lock must be held before calling this. */ -static void reset_pipe(struct snd_dbri * dbri, int pipe) +static void reset_pipe(struct snd_dbri *dbri, int pipe) { int sdp; int desc; s32 *cmd; if (pipe < 0 || pipe > DBRI_MAX_PIPE) { - printk(KERN_ERR "DBRI: reset_pipe called with illegal pipe number\n"); + printk(KERN_ERR "DBRI: reset_pipe called with " + "illegal pipe number\n"); return; } sdp = dbri->pipes[pipe].sdp; if (sdp == 0) { - printk(KERN_ERR "DBRI: reset_pipe called on uninitialized pipe\n"); + printk(KERN_ERR "DBRI: reset_pipe called " + "on uninitialized pipe\n"); return; } @@ -835,9 +840,10 @@ static void reset_pipe(struct snd_dbri * dbri_cmdsend(dbri, cmd, 3); desc = dbri->pipes[pipe].first_desc; - if ( desc >= 0) + if (desc >= 0) do { - dbri->dma->desc[desc].nda = dbri->dma->desc[desc].ba = 0; + dbri->dma->desc[desc].ba = 0; + dbri->dma->desc[desc].nda = 0; desc = dbri->next_desc[desc]; } while (desc != -1 && desc != dbri->pipes[pipe]. first_desc); @@ -848,15 +854,17 @@ static void reset_pipe(struct snd_dbri * /* * Lock must be held before calling this. */ -static void setup_pipe(struct snd_dbri * dbri, int pipe, int sdp) +static void setup_pipe(struct snd_dbri *dbri, int pipe, int sdp) { if (pipe < 0 || pipe > DBRI_MAX_PIPE) { - printk(KERN_ERR "DBRI: setup_pipe called with illegal pipe number\n"); + printk(KERN_ERR "DBRI: setup_pipe called " + "with illegal pipe number\n"); return; } if ((sdp & 0xf800) != sdp) { - printk(KERN_ERR "DBRI: setup_pipe called with strange SDP value\n"); + printk(KERN_ERR "DBRI: setup_pipe called " + "with strange SDP value\n"); /* sdp &= 0xf800; */ } @@ -877,25 +885,26 @@ static void setup_pipe(struct snd_dbri * /* * Lock must be held before calling this. */ -static void link_time_slot(struct snd_dbri * dbri, int pipe, +static void link_time_slot(struct snd_dbri *dbri, int pipe, int prevpipe, int nextpipe, int length, int cycle) { s32 *cmd; int val; - if (pipe < 0 || pipe > DBRI_MAX_PIPE + if (pipe < 0 || pipe > DBRI_MAX_PIPE || prevpipe < 0 || prevpipe > DBRI_MAX_PIPE || nextpipe < 0 || nextpipe > DBRI_MAX_PIPE) { - printk(KERN_ERR + printk(KERN_ERR "DBRI: link_time_slot called with illegal pipe number\n"); return; } - if (dbri->pipes[pipe].sdp == 0 + if (dbri->pipes[pipe].sdp == 0 || dbri->pipes[prevpipe].sdp == 0 || dbri->pipes[nextpipe].sdp == 0) { - printk(KERN_ERR "DBRI: link_time_slot called on uninitialized pipe\n"); + printk(KERN_ERR "DBRI: link_time_slot called " + "on uninitialized pipe\n"); return; } @@ -935,17 +944,17 @@ static void link_time_slot(struct snd_db /* * Lock must be held before calling this. */ -static void unlink_time_slot(struct snd_dbri * dbri, int pipe, +static void unlink_time_slot(struct snd_dbri *dbri, int pipe, enum in_or_out direction, int prevpipe, int nextpipe) { s32 *cmd; int val; - if (pipe < 0 || pipe > DBRI_MAX_PIPE + if (pipe < 0 || pipe > DBRI_MAX_PIPE || prevpipe < 0 || prevpipe > DBRI_MAX_PIPE || nextpipe < 0 || nextpipe > DBRI_MAX_PIPE) { - printk(KERN_ERR + printk(KERN_ERR "DBRI: unlink_time_slot called with illegal pipe number\n"); return; } @@ -985,7 +994,7 @@ static void unlink_time_slot(struct snd_ * * Lock must not be held before calling it. */ -static void xmit_fixed(struct snd_dbri * dbri, int pipe, unsigned int data) +static void xmit_fixed(struct snd_dbri *dbri, int pipe, unsigned int data) { s32 *cmd; unsigned long flags; @@ -996,7 +1005,8 @@ static void xmit_fixed(struct snd_dbri * } if (D_SDP_MODE(dbri->pipes[pipe].sdp) == 0) { - printk(KERN_ERR "DBRI: xmit_fixed: Uninitialized pipe % d\n", pipe); + printk(KERN_ERR "DBRI: xmit_fixed: " + "Uninitialized pipe %d\n", pipe); return; } @@ -1006,7 +1016,8 @@ static void xmit_fixed(struct snd_dbri * } if (!(dbri->pipes[pipe].sdp & D_SDP_TO_SER)) { - printk(KERN_ERR "DBRI: xmit_fixed: Called on receive pipe %d\n", pipe); + printk(KERN_ERR "DBRI: xmit_fixed: Called on receive pipe %d\n", + pipe); return; } @@ -1028,20 +1039,23 @@ static void xmit_fixed(struct snd_dbri * } -static void recv_fixed(struct snd_dbri * dbri, int pipe, volatile __u32 * ptr) +static void recv_fixed(struct snd_dbri *dbri, int pipe, volatile __u32 *ptr) { if (pipe < 16 || pipe > DBRI_MAX_PIPE) { - printk(KERN_ERR "DBRI: recv_fixed called with illegal pipe number\n"); + printk(KERN_ERR "DBRI: recv_fixed called with " + "illegal pipe number\n"); return; } if (D_SDP_MODE(dbri->pipes[pipe].sdp) != D_SDP_FIXED) { - printk(KERN_ERR "DBRI: recv_fixed called on non-fixed pipe %d\n", pipe); + printk(KERN_ERR "DBRI: recv_fixed called on " + "non-fixed pipe %d\n", pipe); return; } if (dbri->pipes[pipe].sdp & D_SDP_TO_SER) { - printk(KERN_ERR "DBRI: recv_fixed called on transmit pipe %d\n", pipe); + printk(KERN_ERR "DBRI: recv_fixed called on " + "transmit pipe %d\n", pipe); return; } @@ -1064,7 +1078,7 @@ static void recv_fixed(struct snd_dbri * * * Lock must be held before calling this. */ -static int setup_descs(struct snd_dbri * dbri, int streamno, unsigned int period) +static int setup_descs(struct snd_dbri *dbri, int streamno, unsigned int period) { struct dbri_streaminfo *info = &dbri->stream_info[streamno] ; __u32 dvma_buffer; @@ -1089,21 +1103,23 @@ static int setup_descs(struct snd_dbri * if (streamno == DBRI_PLAY) { if (!(dbri->pipes[info->pipe].sdp & D_SDP_TO_SER)) { - printk(KERN_ERR "DBRI: setup_descs: Called on receive pipe %d\n", - info->pipe); + printk(KERN_ERR "DBRI: setup_descs: " + "Called on receive pipe %d\n", info->pipe); return -2; } } else { if (dbri->pipes[info->pipe].sdp & D_SDP_TO_SER) { - printk(KERN_ERR + printk(KERN_ERR "DBRI: setup_descs: Called on transmit pipe %d\n", info->pipe); return -2; } - /* Should be able to queue multiple buffers to receive on a pipe */ + /* Should be able to queue multiple buffers + * to receive on a pipe + */ if (pipe_active(dbri, info->pipe)) { - printk(KERN_ERR "DBRI: recv_on_pipe: Called on active pipe %d\n", - info->pipe); + printk(KERN_ERR "DBRI: recv_on_pipe: " + "Called on active pipe %d\n", info->pipe); return -2; } @@ -1113,11 +1129,13 @@ static int setup_descs(struct snd_dbri * /* Free descriptors if pipe has any */ desc = dbri->pipes[info->pipe].first_desc; - if ( desc >= 0) + if (desc >= 0) do { - dbri->dma->desc[desc].nda = dbri->dma->desc[desc].ba = 0; + dbri->dma->desc[desc].ba = 0; + dbri->dma->desc[desc].nda = 0; desc = dbri->next_desc[desc]; - } while (desc != -1 && desc != dbri->pipes[info->pipe]. first_desc); + } while (desc != -1 && + desc != dbri->pipes[info->pipe].first_desc); dbri->pipes[info->pipe].desc = -1; dbri->pipes[info->pipe].first_desc = -1; @@ -1150,8 +1168,7 @@ static int setup_descs(struct snd_dbri * if (streamno == DBRI_PLAY) { dbri->dma->desc[desc].word1 = DBRI_TD_CNT(mylen); dbri->dma->desc[desc].word4 = 0; - dbri->dma->desc[desc].word1 |= - DBRI_TD_F | DBRI_TD_B; + dbri->dma->desc[desc].word1 |= DBRI_TD_F | DBRI_TD_B; } else { dbri->dma->desc[desc].word1 = 0; dbri->dma->desc[desc].word4 = @@ -1172,7 +1189,8 @@ static int setup_descs(struct snd_dbri * } if (first_desc == -1 || last_desc == -1) { - printk(KERN_ERR "DBRI: setup_descs: Not enough descriptors available\n"); + printk(KERN_ERR "DBRI: setup_descs: " + " Not enough descriptors available\n"); return -1; } @@ -1183,14 +1201,14 @@ static int setup_descs(struct snd_dbri * dbri->pipes[info->pipe].desc = first_desc; #ifdef DBRI_DEBUG - for (desc = first_desc; desc != -1; ) { + for (desc = first_desc; desc != -1;) { dprintk(D_DESC, "DESC %d: %08x %08x %08x %08x\n", desc, dbri->dma->desc[desc].word1, dbri->dma->desc[desc].ba, dbri->dma->desc[desc].nda, dbri->dma->desc[desc]. word4); desc = dbri->next_desc[desc]; - if ( desc == first_desc ) + if (desc == first_desc) break; } #endif @@ -1213,7 +1231,8 @@ enum master_or_slave { CHImaster, CHIsla /* * Lock must not be held before calling it. */ -static void reset_chi(struct snd_dbri * dbri, enum master_or_slave master_or_slave, +static void reset_chi(struct snd_dbri *dbri, + enum master_or_slave master_or_slave, int bits_per_frame) { s32 *cmd; @@ -1222,7 +1241,7 @@ static void reset_chi(struct snd_dbri * /* Set CHI Anchor: Pipe 16 */ cmd = dbri_cmdlock(dbri, 4); - val = D_DTS_VO | D_DTS_VI | D_DTS_INS + val = D_DTS_VO | D_DTS_VI | D_DTS_INS | D_DTS_PRVIN(16) | D_PIPE(16) | D_DTS_PRVOUT(16); *(cmd++) = DBRI_CMD(D_DTS, 0, val); *(cmd++) = D_TS_ANCHOR | D_TS_NEXT(16); @@ -1246,15 +1265,16 @@ static void reset_chi(struct snd_dbri * } else { /* Setup DBRI for CHI Master - generate clock, FS * - * BPF = bits per 8 kHz frame - * 12.288 MHz / CHICM_divisor = clock rate - * FD = 1 - drive CHIFS on rising edge of CHICK + * BPF = bits per 8 kHz frame + * 12.288 MHz / CHICM_divisor = clock rate + * FD = 1 - drive CHIFS on rising edge of CHICK */ int clockrate = bits_per_frame * 8; int divisor = 12288 / clockrate; if (divisor > 255 || divisor * clockrate != 12288) - printk(KERN_ERR "DBRI: illegal bits_per_frame in setup_chi\n"); + printk(KERN_ERR "DBRI: illegal bits_per_frame " + "in setup_chi\n"); *(cmd++) = DBRI_CMD(D_CHI, 0, D_CHI_CHICM(divisor) | D_CHI_FD | D_CHI_BPF(bits_per_frame)); @@ -1288,7 +1308,7 @@ to the DBRI via the CHI interface and fe * Lock must not be held before calling it. */ -static void cs4215_setup_pipes(struct snd_dbri * dbri) +static void cs4215_setup_pipes(struct snd_dbri *dbri) { unsigned long flags; @@ -1303,9 +1323,9 @@ static void cs4215_setup_pipes(struct sn * not relevant for us (only for doublechecking). * * Control mode: - * Pipe 17: Send timeslots 1-4 (slots 5-8 are readonly) + * Pipe 17: Send timeslots 1-4 (slots 5-8 are read only) * Pipe 18: Receive timeslot 1 (clb). - * Pipe 19: Receive timeslot 7 (version). + * Pipe 19: Receive timeslot 7 (version). */ setup_pipe(dbri, 4, D_SDP_MEM | D_SDP_TO_SER | D_SDP_MSB); @@ -1355,7 +1375,7 @@ static int cs4215_init_data(struct cs421 return 0; } -static void cs4215_setdata(struct snd_dbri * dbri, int muted) +static void cs4215_setdata(struct snd_dbri *dbri, int muted) { if (muted) { dbri->mm.data[0] |= 63; @@ -1387,7 +1407,7 @@ static void cs4215_setdata(struct snd_db /* * Set the CS4215 to data mode. */ -static void cs4215_open(struct snd_dbri * dbri) +static void cs4215_open(struct snd_dbri *dbri) { int data_width; u32 tmp; @@ -1452,7 +1472,7 @@ static void cs4215_open(struct snd_dbri /* * Send the control information (i.e. audio format) */ -static int cs4215_setctrl(struct snd_dbri * dbri) +static int cs4215_setctrl(struct snd_dbri *dbri) { int i, val; u32 tmp; @@ -1502,9 +1522,9 @@ static int cs4215_setctrl(struct snd_dbr /* * Control mode: - * Pipe 17: Send timeslots 1-4 (slots 5-8 are readonly) + * Pipe 17: Send timeslots 1-4 (slots 5-8 are read only) * Pipe 18: Receive timeslot 1 (clb). - * Pipe 19: Receive timeslot 7 (version). + * Pipe 19: Receive timeslot 7 (version). */ link_time_slot(dbri, 17, 16, 16, 32, dbri->mm.offset); @@ -1522,9 +1542,9 @@ static int cs4215_setctrl(struct snd_dbr sbus_writel(tmp, dbri->regs + REG0); spin_unlock_irqrestore(&dbri->lock, flags); - for (i = 10; ((dbri->mm.status & 0xe4) != 0x20); --i) { + for (i = 10; ((dbri->mm.status & 0xe4) != 0x20); --i) msleep_interruptible(1); - } + if (i == 0) { dprintk(D_MM, "CS4215 didn't respond to CLB (0x%02x)\n", dbri->mm.status); @@ -1556,7 +1576,7 @@ static int cs4215_setctrl(struct snd_dbr * As part of the process we resend the settings for the data * timeslots as well. */ -static int cs4215_prepare(struct snd_dbri * dbri, unsigned int rate, +static int cs4215_prepare(struct snd_dbri *dbri, unsigned int rate, snd_pcm_format_t format, unsigned int channels) { int freq_idx; @@ -1613,7 +1633,7 @@ static int cs4215_prepare(struct snd_dbr /* * */ -static int cs4215_init(struct snd_dbri * dbri) +static int cs4215_init(struct snd_dbri *dbri) { u32 reg2 = sbus_readl(dbri->regs + REG2); dprintk(D_MM, "cs4215_init: reg2=0x%x\n", reg2); @@ -1674,7 +1694,7 @@ interrupts are disabled. /* xmit_descs() * - * Starts transmiting the current TD's for recording/playing. + * Starts transmitting the current TD's for recording/playing. * For playback, ALSA has filled the DMA memory with new data (we hope). */ static void xmit_descs(struct snd_dbri *dbri) @@ -1701,7 +1721,8 @@ static void xmit_descs(struct snd_dbri * *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + dbri_dma_off(desc, first_td); + *(cmd++) = dbri->dma_dvma + + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); /* Reset our admin of the pipe. */ @@ -1722,7 +1743,8 @@ static void xmit_descs(struct snd_dbri * *(cmd++) = DBRI_CMD(D_SDP, 0, dbri->pipes[info->pipe].sdp | D_SDP_P | D_SDP_EVERY | D_SDP_C); - *(cmd++) = dbri->dma_dvma + dbri_dma_off(desc, first_td); + *(cmd++) = dbri->dma_dvma + + dbri_dma_off(desc, first_td); dbri_cmdsend(dbri, cmd, 2); /* Reset our admin of the pipe. */ @@ -1747,7 +1769,7 @@ static void xmit_descs(struct snd_dbri * * */ -static void transmission_complete_intr(struct snd_dbri * dbri, int pipe) +static void transmission_complete_intr(struct snd_dbri *dbri, int pipe) { struct dbri_streaminfo *info; int td; @@ -1763,9 +1785,8 @@ static void transmission_complete_intr(s } status = DBRI_TD_STATUS(dbri->dma->desc[td].word4); - if (!(status & DBRI_TD_TBC)) { + if (!(status & DBRI_TD_TBC)) break; - } dprintk(D_INT, "TD %d, status 0x%02x\n", td, status); @@ -1785,7 +1806,7 @@ static void transmission_complete_intr(s snd_pcm_period_elapsed(info->substream); } -static void reception_complete_intr(struct snd_dbri * dbri, int pipe) +static void reception_complete_intr(struct snd_dbri *dbri, int pipe) { struct dbri_streaminfo *info; int rd = dbri->pipes[pipe].desc; @@ -1817,7 +1838,7 @@ static void reception_complete_intr(stru snd_pcm_period_elapsed(info->substream); } -static void dbri_process_one_interrupt(struct snd_dbri * dbri, int x) +static void dbri_process_one_interrupt(struct snd_dbri *dbri, int x) { int val = D_INTR_GETVAL(x); int channel = D_INTR_GETCHAN(x); @@ -1889,7 +1910,7 @@ static void dbri_process_one_interrupt(s * right now). Non-zero words require processing and are handed off * to dbri_process_one_interrupt AFTER advancing the pointer. */ -static void dbri_process_interrupt_buffer(struct snd_dbri * dbri) +static void dbri_process_interrupt_buffer(struct snd_dbri *dbri) { s32 x; @@ -1965,15 +1986,15 @@ static irqreturn_t snd_dbri_interrupt(in PCM Interface ***************************************************************** ***********/ static struct snd_pcm_hardware snd_dbri_pcm_hw = { - .info = (SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_BLOCK_TRANSFER | - SNDRV_PCM_INFO_MMAP_VALID), - .formats = SNDRV_PCM_FMTBIT_MU_LAW | - SNDRV_PCM_FMTBIT_A_LAW | - SNDRV_PCM_FMTBIT_U8 | - SNDRV_PCM_FMTBIT_S16_BE, - .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_5512, + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_MU_LAW | + SNDRV_PCM_FMTBIT_A_LAW | + SNDRV_PCM_FMTBIT_U8 | + SNDRV_PCM_FMTBIT_S16_BE, + .rates = SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_5512, .rate_min = 5512, .rate_max = 48000, .channels_min = 1, @@ -2011,7 +2032,8 @@ static int snd_hw_rule_channels(struct s snd_interval_any(&ch); if (!(f->bits[0] & SNDRV_PCM_FMTBIT_S16_BE)) { - ch.min = ch.max = 1; + ch.min = 1; + ch.max = 1; ch.integer = 1; return snd_interval_refine(c, &ch); } @@ -2035,14 +2057,14 @@ static int snd_dbri_open(struct snd_pcm_ info->pipe = -1; spin_unlock_irqrestore(&dbri->lock, flags); - snd_pcm_hw_rule_add(runtime,0,SNDRV_PCM_HW_PARAM_CHANNELS, + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_hw_rule_format, NULL, SNDRV_PCM_HW_PARAM_FORMAT, -1); - snd_pcm_hw_rule_add(runtime,0,SNDRV_PCM_HW_PARAM_FORMAT, - snd_hw_rule_channels, NULL, + snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, + snd_hw_rule_channels, NULL, SNDRV_PCM_HW_PARAM_CHANNELS, -1); - + cs4215_open(dbri); return 0; @@ -2145,7 +2167,7 @@ static int snd_dbri_prepare(struct snd_p spin_lock_irq(&dbri->lock); info->offset = 0; - /* Setup the all the transmit/receive desciptors to cover the + /* Setup the all the transmit/receive descriptors to cover the * whole DMA buffer. */ ret = setup_descs(dbri, DBRI_STREAMNO(substream), @@ -2205,7 +2227,7 @@ static struct snd_pcm_ops snd_dbri_ops = .pointer = snd_dbri_pointer, }; -static int __devinit snd_dbri_pcm(struct snd_dbri * dbri) +static int __devinit snd_dbri_pcm(struct snd_dbri *dbri) { struct snd_pcm *pcm; int err; @@ -2228,9 +2250,8 @@ static int __devinit snd_dbri_pcm(struct if ((err = snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, snd_dma_continuous_data(GFP_KERNEL), - 64 * 1024, 64 * 1024)) < 0) { + 64 * 1024, 64 * 1024)) < 0) return err; - } return 0; } @@ -2271,7 +2292,8 @@ static int snd_cs4215_put_volume(struct struct snd_ctl_elem_value *ucontrol) { struct snd_dbri *dbri = snd_kcontrol_chip(kcontrol); - struct dbri_streaminfo *info = &dbri->stream_info[kcontrol->private_value]; + struct dbri_streaminfo *info = + &dbri->stream_info[kcontrol->private_value]; int changed = 0; if (info->left_gain != ucontrol->value.integer.value[0]) { @@ -2316,18 +2338,16 @@ static int snd_cs4215_get_single(struct int invert = (kcontrol->private_value >> 24) & 1; snd_assert(dbri != NULL, return -EINVAL); - if (elem < 4) { + if (elem < 4) ucontrol->value.integer.value[0] = (dbri->mm.data[elem] >> shift) & mask; - } else { + else ucontrol->value.integer.value[0] = (dbri->mm.ctrl[elem - 4] >> shift) & mask; - } - if (invert == 1) { + if (invert == 1) ucontrol->value.integer.value[0] = mask - ucontrol->value.integer.value[0]; - } return 0; } @@ -2378,11 +2398,12 @@ static int snd_cs4215_put_single(struct timeslots. Shift is the bit offset in the timeslot, mask defines the number of bits. invert is a boolean for use with attenuation. */ -#define CS4215_SINGLE(xname, entry, shift, mask, invert) \ -{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \ - .info = snd_cs4215_info_single, \ - .get = snd_cs4215_get_single, .put = snd_cs4215_put_single, \ - .private_value = entry | (shift << 8) | (mask << 16) | (invert << 24) }, +#define CS4215_SINGLE(xname, entry, shift, mask, invert) \ +{ .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \ + .info = snd_cs4215_info_single, \ + .get = snd_cs4215_get_single, .put = snd_cs4215_put_single, \ + .private_value = (entry) | ((shift) << 8) | ((mask) << 16) | \ + ((invert) << 24) }, static struct snd_kcontrol_new dbri_controls[] __devinitdata = { { @@ -2411,7 +2432,7 @@ static struct snd_kcontrol_new dbri_cont CS4215_SINGLE("Mic boost", 4, 4, 1, 1) }; -static int __init snd_dbri_mixer(struct snd_dbri * dbri) +static int __init snd_dbri_mixer(struct snd_dbri *dbri) { struct snd_card *card; int idx, err; @@ -2438,7 +2459,8 @@ static int __init snd_dbri_mixer(struct /**************************************************************** ************ /proc interface ***************************************************************** ***********/ -static void dbri_regs_read(struct snd_info_entry * entry, struct snd_info_buffer *buffer) +static void dbri_regs_read(struct snd_info_entry *entry, + struct snd_info_buffer *buffer) { struct snd_dbri *dbri = entry->private_data; @@ -2449,7 +2471,7 @@ static void dbri_regs_read(struct snd_in } #ifdef DBRI_DEBUG -static void dbri_debug_read(struct snd_info_entry * entry, +static void dbri_debug_read(struct snd_info_entry *entry, struct snd_info_buffer *buffer) { struct snd_dbri *dbri = entry->private_data; @@ -2463,7 +2485,8 @@ static void dbri_debug_read(struct snd_i "Pipe %d: %s SDP=0x%x desc=%d, " "len=%d next %d\n", pipe, - ((pptr->sdp & D_SDP_TO_SER) ? "output" : "input") , + ((pptr->sdp & D_SDP_TO_SER) ? "output" : + "input"), pptr->sdp, pptr->desc, pptr->length, pptr->nextpipe); } @@ -2471,11 +2494,11 @@ static void dbri_debug_read(struct snd_i } #endif -void snd_dbri_proc(struct snd_dbri * dbri) +void snd_dbri_proc(struct snd_dbri *dbri) { struct snd_info_entry *entry; - if (! snd_card_proc_new(dbri->card, "regs", &entry)) + if (!snd_card_proc_new(dbri->card, "regs", &entry)) snd_info_set_text_ops(entry, dbri, dbri_regs_read); #ifdef DBRI_DEBUG @@ -2491,7 +2514,7 @@ void snd_dbri_proc(struct snd_dbri * dbr **************************** Initialization ******************************** ***************************************************************** *********** */ -static void snd_dbri_free(struct snd_dbri * dbri); +static void snd_dbri_free(struct snd_dbri *dbri); static int __init snd_dbri_create(struct snd_card *card, struct sbus_dev *sdev, @@ -2547,7 +2570,7 @@ static int __init snd_dbri_create(struct return 0; } -static void snd_dbri_free(struct snd_dbri * dbri) +static void snd_dbri_free(struct snd_dbri *dbri) { dprintk(D_GEN, "snd_dbri_free\n"); dbri_reset(dbri); @@ -2587,7 +2610,8 @@ static int __init dbri_attach(int prom_n err = prom_getproperty(prom_node, "intr", (char *)&irq, sizeof(irq)); if (err < 0) { - printk(KERN_ERR "DBRI-%d: Firmware node lacks IRQ property.\n", dev); + printk(KERN_ERR "DBRI-%d: Firmware node lacks IRQ property.\n", + dev); return -ENODEV; } @@ -2613,12 +2637,14 @@ static int __init dbri_attach(int prom_n goto _err; if ((err = snd_dbri_mixer(dbri)) < 0) + if ((err = snd_dbri_mixer(dbri)) < 0) goto _err; /* /proc file handling */ snd_dbri_proc(dbri); - if ((err = snd_card_register(card)) < 0) + err = snd_card_register(card); + if (err < 0) goto _err; printk(KERN_INFO "audio%d at %p (irq %d) is DBRI(%c) +CS4215(%d)\n", @@ -2628,7 +2654,7 @@ static int __init dbri_attach(int prom_n return 0; - _err: +_err: snd_dbri_free(dbri); snd_card_free(card); return err; ---------------------------------------------------- Producenci filmu SEZON NA MISIA przedstawiają surfujące pingwiny w nowej, animowanej komedii NA FALI - w kinach od 10 sierpnia! http://klik.wp.pl/?adr=http%3A%2F%2Fadv.reklama.wp.pl%2Fas%2Fnafali_zwiastu…

2 1

[alsa-devel] snd-hda-intel, AD 1988b: SPDIF output not working correctly, volume low for analog output
by Matthias Dahl 20 Aug '07

20 Aug '07

Hello everyone. I very recently upgraded my system to a Asus Crosshair (AM2, nforce 590 sli) board because it seemed just good quality. Unfortunately the AD1988b chip on it is causing me some trouble. 1] SPDIF output When I use it for dts/dolby passthrough, it works just fine even though from time to time spdif just hangs- not during playback but when I start the playback. (analog output is unaffected) Meaning that either the player is hanging or playback starts but I hear no sound. After that only a reboot solves the problem... restarting alsa has not effect. Also when the passthrough stops or during playback of anything else besides a dts/dolby stream, I hear a low pitch beating (usually in combination with a very high pitch sound) in the background. I double checked everything to be sure that my other equipment was fine and not the cause of this. 2] analog output I am unable to get it as loud as with the win drivers without overdriving it. The volume is okay... but not on par with the win drivers. That's especially true for the SPDIF output. Increasing the SPDIF volume usually results in worse audio quality over spdif and increased "noise" (see above). Like said earlier, I checked with the win drivers, and everything (audio wise) works just fine there. So it's not a hardware fault. (relief -g-) My guess would be that the AD 1988b is quite a bit different to program from its 1988 predecessors as it contains dts live and stuff. Just a guess... I have attached some hopefully useful informations. If you need anything else, please let me know. I really need that spdif working, so I am more than willing to test just about anything. :-) Last but not leat, I have tried the latest official alsa drivers as well as a mercurial snapshot from yesterday. (which I am currently using) Both show the same problems. Thanks for any help in advance, matthew.

4 10

[alsa-devel] Voip Application with ALSA OSS Emulation ==> voice delay
by Akio 20 Aug '07

20 Aug '07

Dear Sir: Could you do me a favor? Right now, we are using a arm box with linux kernel 2.6.14 to develop a voip solution. After the rtp connection was connected at first 20 secs, there is no delay. As the timing passing by, the delay will going longer and longer. After 16 houres, the delay will up to 12 secs. Because the alsa lib application can't detect any alsa device, we have no choice but only alsa oss emulation data path to using audio device. after tracing sound/core/oss/pcm_oss.c. we find some things. 1.in snd_pcm_oss_write() and in snd_pcm_oss_read(), there is a little slight difference. in snd_pcm_oss_write(), before and after snd_pcm_oss_write1() there are using up() and down() to protect snd_pcm_oss_write1(). but in snd_pcm_oss_read(), there is no protection. why doesn't use up() and down() to protect snd_pcm_oss_read1()? 2.after using printk to tracing runtime->oss.xxx fields in snd_pcm_oss_read1(), we got the data positions seem correct. and alignment very well. what should we have to check? where do we have to check? 3.in snd_pcm_oss_capture_position_fixup(), why using the following condition to check, if (*delay <= (snd_pcm_sframes_t)runtime->buffer_size)? why not using if (*delay <= (snd_pcm_sframes_t)runtime->period_size)? because the following wrapping, frames = (*delay - runtime->buffer_size) + runtime->period_size - 1; frames /= runtime->period_size; frames *= runtime->period_size; using runtime->period_size, not runtime->buffer_size. 4.why using snd_pcm_oss_capture_position_fixup() in snd_pcm_oss_read3() not in snd_pcm_oss_read()? 5.the following url seem counter the same problems. http://blackfin.uclinux.org/gf/project/uclinux-dist/forum/? action=ForumBrowse&forum_id=39&_forum_action= ForumMessageBrowse&thread_id=22524 We will check the write direction to see the runtime->oss.xxx fields will act as the same read or not. Thanks in advance. Best Regards, Akio

2 2

[alsa-devel] MAYA44 PCI
by Piotrek 19 Aug '07

19 Aug '07

Dear Developers, I tried to run my ESI MAYA44 PCI with alsa compiled with ice1724 driver, but I can't run alsamixer because of the error: alsamixer: function snd_ctl_open failed for default: No such device. Card does not work. Any suggestions how to configure it or maybe it's more difficult to get it working? I have reported an error on alsa bugtracker: https://bugtrack.alsa-project.org/alsa-bug/view.php?id=3303 -- Best Regards, Piotr Makowski

1 0

[alsa-devel] Noisy audio - ALSA Configuration
by Iovescu, Magdalena 16 Aug '07

16 Aug '07

Does anyone have examples or documentation on how to configure periods, frames and buffers for various sample rates? I am writing a program to route a mp3 decoder output to ALSA on an ARM processor, and I get noisy audio. For a CD quality stream (S16_LE, 44100 kHz, Stereo, 128 bps) I am using an output buffer of 44100 bytes, a period of 44100/(channels*2). With a buffer this big, I do not get underruns, but all the writes are short writes. I have not been able to find the right configuration to avoid short writes, and more importantly, to get good quality audio. What do 'short writes' (I am using writei) really mean? Is this a performance issue in the driver, or is mainly because of buffer setup? I would really appreciate any help with this. Mag

2 2

Re: [alsa-devel] Recent code changes for HD-audio on HG repo
by Nicola Fagnani 16 Aug '07

16 Aug '07

As you said, the patch didn't solve the problem. To test audio I used this simple command: aplay < /bin/bash but if there is no audio output for a while, executing it produces no noise. When I said the sound goes mute, I meant there is no sound, not that some audio channel is muted. If I set the power_save_controller=0 module option the sound works always, so these are the details of my hardware: lspci -v 00:1b.0 Audio device: Intel Corporation 82801G (ICH7 Family) High Definition Audio Controller (rev 02) Subsystem: Realtek Semiconductor Co., Ltd. Unknown device 0885 Flags: bus master, fast devsel, latency 0, IRQ 21 Memory at 52400000 (64-bit, non-prefetchable) [size=16K] Capabilities: [50] Power Management version 2 Capabilities: [60] Message Signalled Interrupts: Mask- 64bit+ Queue=0/0 Enable- Capabilities: [70] Express Unknown type IRQ 0 cat /proc/asound/card0/codec#0 Codec: Realtek ALC885 Address: 0 Vendor Id: 0x10ec0885 Subsystem Id: 0x106b1000 Revision Id: 0x100103 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 [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x00 0x00] PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 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: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] 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] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] 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] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] 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 0x08373c: IN OUT HP Detect Pin Default 0x012b4050: [Jack] HP Out at Ext Rear Conn = Comb, Color = Green 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: [0x80 0x80] Pincap 0x08373c: IN OUT HP Detect Pin Default 0x018b3020: [Jack] Line In at Ext Rear Conn = Comb, Color = Blue Pin-ctls: 0x20: IN 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 0x083c: IN OUT HP Detect Pin Default 0x400000f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown 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 0x083c: IN OUT HP Detect Pin Default 0x400000f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown 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: [0x00 0x00] Pincap 0x08373c: IN OUT HP Detect Pin Default 0x90100140: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT 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 0x08373c: IN OUT HP Detect Pin Default 0x90a00110: [Fixed] Mic at Int N/A Conn = Unknown, Color = Unknown 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: [0x00 0x00] Pincap 0x08373c: IN OUT HP Detect Pin Default 0x90100141: [Fixed] Speaker at Int N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x40: OUT 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 0x08373c: IN OUT HP Detect Pin Default 0x400000f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x0820: IN Pin Default 0x400000f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x0820: IN Pin Default 0x400000f0: [N/A] Line Out at Ext N/A Conn = Unknown, Color = Unknown Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x0810: OUT Pin Default 0x014be060: [Jack] SPDIF Out at Ext Rear Conn = Comb, Color = White Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x0820: IN Pin Default 0x01cbe030: [Jack] SPDIF In at Ext Rear Conn = Comb, Color = White Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, 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] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, 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] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, 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] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b 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=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x25 0x0b amixer Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 44 [69%] [-20.00dB] [on] Front Right: Playback 44 [69%] [-20.00dB] [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 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined Playback channels: Mono Capture channels: Mono Mono: Playback [off] Capture [off] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [on] Front Right: Capture 0 [0%] [-16.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [on] Front Right: Capture 0 [0%] [-16.00dB] [on] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [on] Front Right: Capture 0 [0%] [-16.00dB] [on] 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' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: enum Items: 'Mic' 'Front Mic' 'Line' 'CD' Item0: 'Mic' If this isn't enough, let me know. Thanks Nicola Fagnani

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Re: [alsa-devel] Noisy audio - ALSA Configuration
by Iovescu, Magdalena 16 Aug '07

16 Aug '07

>> Does anyone have examples or documentation on how to configure periods, >> frames and buffers for various sample rates? I am writing a program to >> route a mp3 decoder output to ALSA on an ARM processor, and I get noisy >> audio. For a CD quality stream (S16_LE, 44100 kHz, Stereo, 128 bps) I am >> using an output buffer of 44100 bytes, a period of 44100/(channels*2). With >>Be careful about the distinction between bytes and frames (i.e. >>a frame of stereo S16 is 4 bytes) >Start with the period >1024 frames = 4096 bytes >then make your buffer at least 2x this: >8192 bytes > (I'm not an app developer, so others may jump in with a better >explanation) >Did you look at the code in alsa-lib/test? Thanks Eliot, I will try this buffer size. I did look at alsa tests, but I was not too convinced, as when I run those apps I still got plenty of short writes. Would someone please explain why do these 'short writes' occur? >> a buffer this big, I do not get underruns, but I do get all short writes. I >> have not been able to find the right configuration to avoid short writes, >> and more importantly, to get good quality audio. What do 'short writes' (I >> am using writei) really mean? Is this a performance issue in the driver, or > is mainly because of buffer setup? I would really appreciate any help with >> this. >> Magdalena

1 0

[alsa-devel] Writing a new sound driver
by Michael Bourgeous 16 Aug '07

16 Aug '07

I am the developer of the work-in-progress MyHD/TL880 Linux driver. The TL880 chip can act as a sound input and output device, but it doesn't seem to support DMA for sound. Is it possible to write a sound driver for ALSA that uses a polling mechanism? Also, how would I integrate this driver with my core driver for accessing the TL880? The chip's features span four separate kernel subsystems: v4l, dvb, sound, and fb. I guess my main question is this: what is the bare minimum of code and hardware functionality required to write a sound driver? Thanks, Mike Bourgeous http://myhd.sourceforge.net/

5 5

[alsa-devel] Noisy audio - ALSA Configuration
by Magdalena Iovescu 16 Aug '07

16 Aug '07

2 1

[alsa-devel] Recent code changes for HD-audio on HG repo
by nicfagn 15 Aug '07

15 Aug '07

I tested the alsa-driver-hg20070814 snapshot on my system and after the power save timeout (10 secs) the sound stops working. I am on an iMac 24''(2006) with hda and realtek ALC885 and Ubuntu Gutsy with a 2.6.22-9 kernel. If I can help in some way, let me know. Thanks

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[alsa-devel] [PATCH 1/2] Xlinx ML403 AC97 Controller Reference device driver
by Joachim Förster 15 Aug '07

15 Aug '07

From: Joachim Foerster <JOFT(a)gmx.de> Add ALSA support for the opb_ac97_controller_ref_v1_00_a ip core found in Xilinx' ML403 reference design. Known issue: Currently this driver hits a WARN_ON_ONCE(1) statement in kernel/irq/resend.c (line 70). According to Linus (http://lkml.org/lkml/2007/8/5/5) this may be ignored, right? I haven't had a look into this "problem" yet. (Patch for Linus' master branch, date 2007/08/08) This patchset _will_ be published on http://www.esic-solutions.com/support.html soon (like the first version of the driver (tar file), but this may take some days ...). Signed-off-by: Joachim Foerster <JOFT(a)gmx.de> --- sound/ppc/Kconfig | 13 + sound/ppc/Makefile | 2 + sound/ppc/ml403_ac97cr.c | 1274 +++++++++++++++++++++++++++++++++++++++++++++ sound/ppc/pcm-indirect2.h | 658 +++++++++++++++++++++++ 4 files changed, 1947 insertions(+), 0 deletions(-) create mode 100644 sound/ppc/ml403_ac97cr.c create mode 100644 sound/ppc/pcm-indirect2.h diff --git a/sound/ppc/Kconfig b/sound/ppc/Kconfig index cacb0b1..cd492bf 100644 --- a/sound/ppc/Kconfig +++ b/sound/ppc/Kconfig @@ -52,4 +52,17 @@ config SND_PS3_DEFAULT_START_DELAY int "Startup delay time in ms" depends on SND_PS3 default "2000" + +config SND_ML403_AC97CR + tristate "Xilinx ML403 AC97 Controller Reference" + depends on SND && XILINX_VIRTEX + select SND_AC97_CODEC + help + Say Y here to include support for the + opb_ac97_controller_ref_v1_00_a ip core found in Xilinx' ML403 + reference design. + + To compile this driver as a module, choose M here: the module + will be called snd-ml403_ac97cr. + endmenu diff --git a/sound/ppc/Makefile b/sound/ppc/Makefile index eacee2d..827f2f5 100644 --- a/sound/ppc/Makefile +++ b/sound/ppc/Makefile @@ -4,7 +4,9 @@ # snd-powermac-objs := powermac.o pmac.o awacs.o burgundy.o daca.o tumbler.o keywest.o beep.o +snd-ml403_ac97cr-objs := ml403_ac97cr.o # Toplevel Module Dependency obj-$(CONFIG_SND_POWERMAC) += snd-powermac.o obj-$(CONFIG_SND_PS3) += snd_ps3.o +obj-$(CONFIG_SND_ML403_AC97CR) += snd-ml403_ac97cr.o diff --git a/sound/ppc/ml403_ac97cr.c b/sound/ppc/ml403_ac97cr.c new file mode 100644 index 0000000..99791d7 --- /dev/null +++ b/sound/ppc/ml403_ac97cr.c @@ -0,0 +1,1274 @@ + +/* ALSA driver for Xilinx ML403 AC97 Controller Reference + * IP: opb_ac97_controller_ref_v1_00_a (EDK 8.1i) + * IP: opb_ac97_controller_ref_v1_00_a (EDK 9.1i) + * + * Copyright (c) by 2007 Joachim Foerster <JOFT(a)gmx.de> + * + * 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 + * + */ + +#include <sound/driver.h> +#include <linux/init.h> +#include <linux/moduleparam.h> + +#include <linux/platform_device.h> + +#include <linux/ioport.h> +#include <asm/io.h> +#include <linux/interrupt.h> + +/* HZ */ +#include <linux/param.h> +/* jiffies, time_*() */ +#include <linux/jiffies.h> +/* schedule_timeout*() */ +#include <linux/sched.h> +/* spin_lock*() */ +#include <linux/spinlock.h> + +/* snd_printk(), snd_printd() */ +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/initval.h> +#include <sound/ac97_codec.h> + + +#define SND_ML403_AC97CR_DRIVER "ml403_ac97cr" + +MODULE_AUTHOR("Joachim Foerster <JOFT(a)gmx.de>"); +MODULE_DESCRIPTION("Xilinx ML403 AC97 Controller Reference"); +MODULE_LICENSE("GPL"); +MODULE_SUPPORTED_DEVICE("{{Xilinx,ML403 AC97 Controller Reference}}"); +MODULE_VERSION("0.0.1-pre2"); + +static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; +static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR; +static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE; + +module_param_array(index, int, NULL, 0444); +MODULE_PARM_DESC(index, "Index value for ML403 AC97 Controller Reference."); +module_param_array(id, charp, NULL, 0444); +MODULE_PARM_DESC(id, "ID string for ML403 AC97 Controller Reference."); +module_param_array(enable, bool, NULL, 0444); +MODULE_PARM_DESC(enable, "Enable this ML403 AC97 Controller Reference."); + +/* Special feature options */ +/*#define CODEC_WRITE_CHECK_RAF*/ /* don't return after a write to a codec + * register, while RAF bit is not set + */ +/* Debug options for code which may be removed completely in a final version */ +#ifdef CONFIG_SND_DEBUG +/*#define CODEC_STAT*/ /* turn on some minimal "statistics" + * about codec register usage + */ +#define SND_PCM_INDIRECT2_STAT /* turn on some "statistics" about the + * process of copying bytes from the + * intermediate buffer to the hardware + * fifo and the other way round + */ +#endif + +#include "pcm-indirect2.h" + +/* Definition of a "level/facility dependent" printk(); may be removed + * completely in a final version + */ +#undef PDEBUG +#ifdef CONFIG_SND_DEBUG +/* "facilities" for PDEBUG */ +#define UNKNOWN (1<<0) +#define CODEC_SUCCESS (1<<1) +#define CODEC_FAKE (1<<2) +#define INIT_INFO (1<<3) +#define INIT_FAILURE (1<<4) +#define WORK_INFO (1<<5) +#define WORK_FAILURE (1<<6) + +#define PDEBUG_FACILITIES (UNKNOWN | INIT_FAILURE | WORK_FAILURE) + +#define PDEBUG(fac, fmt, args...) if (fac & PDEBUG_FACILITIES) \ + snd_printd(KERN_DEBUG SND_ML403_AC97CR_DRIVER ": " fmt, ##args) +#else +#define PDEBUG(fac, fmt, args...) /* nothing */ +#endif + + + +/* Defines for "waits"/timeouts (portions of HZ=250 on arch/ppc by default) */ +#define CODEC_TIMEOUT_ON_INIT 5 /* timeout for checking for codec + * readiness (after insmod) + */ +#ifndef CODEC_WRITE_CHECK_RAF +#define CODEC_WAIT_AFTER_WRITE 100 /* general, static wait after a write + * access to a codec register, may be + * 0 to completely remove wait + */ +#else +#define CODEC_TIMEOUT_AFTER_WRITE 5 /* timeout after a write access to a + * codec register, if RAF bit is used + */ +#endif +#define CODEC_TIMEOUT_AFTER_READ 5 /* timeout after a read access to a + * codec register (checking RAF bit) + */ + +/* Infrastructure for codec register shadowing */ +#define LM4550_REG_DONEREAD (1<<0) /* read register once, value should be the + * same currently in the register + */ +#define LM4550_REG_NOSAVE (1<<1) /* values written to this register will + * not be saved in the register + */ +#define LM4550_REG_NOSHADOW (1<<2) /* don't do register shadowing, use plain + * hardware access + */ +#define LM4550_REG_READONLY (1<<3) /* register is read only */ +#define LM4550_REG_FAKEPROBE (1<<4) /* fake write _and_ read actions during + * probe() correctly + */ +#define LM4550_REG_FAKEREAD (1<<5) /* fake read access, always return default + * value + */ +#define LM4550_REG_ALLFAKE (LM4550_REG_FAKEREAD | LM4550_REG_FAKEPROBE) + +struct lm4550_reg { + u16 reg; + u16 value; + u16 flag; + u16 wmask; + u16 def; +}; + +struct lm4550_reg lm4550_regfile[64] = { + {.reg = 0x00, + .flag = LM4550_REG_NOSAVE | LM4550_REG_FAKEREAD, + .def = 0x0D50}, + {.reg = 0x02, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8000}, + {.reg = 0x04, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8000}, + {.reg = 0x06, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x801F, + .def = 0x8000}, + {}, + {.reg = 0x0A, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x801E, + .def = 0x0}, + {.reg = 0x0C, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x801F, + .def = 0x8008}, + {.reg = 0x0E, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x805F, + .def = 0x8008}, + {.reg = 0x10, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + {.reg = 0x12, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + {.reg = 0x14, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + {.reg = 0x16, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8808}, + {.reg = 0x18, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x9F1F, + .def = 0x8008}, + {.reg = 0x1A, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x707, + .def = 0x0}, + {.reg = 0x1C, + .flag = LM4550_REG_FAKEPROBE, + .wmask = 0x8F0F, + .def = 0x8000}, + {}, + {.reg = 0x20, + .flag = LM4550_REG_FAKEPROBE, + .def = 0x0, + .wmask = 0xA380}, + {.reg = 0x22, + .flag = LM4550_REG_FAKEREAD | LM4550_REG_READONLY, + .def = 0x0101}, + {}, + {.reg = 0x26, + .flag = LM4550_REG_NOSHADOW | LM4550_REG_NOSAVE, + .wmask = 0xFF00}, /* may not write ones to REF/ANL/DAC/ADC bits + * FIXME: Is this ok? + */ + {.reg = 0x28, + .flag = LM4550_REG_FAKEREAD | LM4550_REG_READONLY, + .def = 0x0201}, /* primary codec */ + {.reg = 0x2A, + .flag = LM4550_REG_NOSHADOW | LM4550_REG_NOSAVE, + .wmask = 0x1}, + {.reg = 0x2C, + .flag = LM4550_REG_FAKEPROBE, + .def = 0xBB80, + .wmask = 0xFFFF}, + {}, {}, + {.reg = 0x32, + .flag = LM4550_REG_FAKEPROBE, + .def = 0xBB80, + .wmask = 0xFFFF}, + {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, + {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, {}, + {.reg = 0x7C, + .flag = LM4550_REG_READONLY | LM4550_REG_FAKEREAD, + .def = 0x4E53}, + {.reg = 0x7E, + .flag = LM4550_REG_READONLY | LM4550_REG_FAKEREAD, + .def = 0x4350} +}; + +#define LM4550_RF_OK(reg) ((lm4550_regfile[reg / 2].reg != 0) \ + || (lm4550_regfile[reg / 2].flag != 0)) +#define LM4550_RF_FLAG(reg) lm4550_regfile[reg / 2].flag +#define LM4550_RF_VAL(reg) lm4550_regfile[reg / 2].value +#define LM4550_RF_DEF(reg) lm4550_regfile[reg / 2].def +#define LM4550_RF_WMASK(reg) lm4550_regfile[reg / 2].wmask + +static void lm4550_regfile_init(void) +{ + int i; + for (i = 0; i < 128; i = i + 2) + if (LM4550_RF_FLAG(i) & LM4550_REG_FAKEPROBE) + LM4550_RF_VAL(i) = LM4550_RF_DEF(i); +} + +static void lm4550_regfile_write_values_after_init(struct snd_ac97 *ac97) +{ + int i; + for (i = 0; i < 128; i = i + 2) + if ((LM4550_RF_FLAG(i) & LM4550_REG_FAKEPROBE) && + (LM4550_RF_VAL(i) != LM4550_RF_DEF(i))) { + PDEBUG(CODEC_FAKE, "lm4550_regfile_write_values_after_" + "init(): reg=0x%x value=0x%x / %d is different " + "from def=0x%x / %d\n", + i, LM4550_RF_VAL(i), LM4550_RF_VAL(i), + LM4550_RF_DEF(i), LM4550_RF_DEF(i)); + snd_ac97_write(ac97, i, LM4550_RF_VAL(i)); + LM4550_RF_FLAG(i) |= LM4550_REG_DONEREAD; + } +} + + +/* direct registers */ +#define CR_REG(ml403_ac97cr,x) ((ml403_ac97cr)->port + CR_REG_##x) + +#define CR_REG_PLAYFIFO 0x00 +#define CR_PLAYDATA(a) ((a) & 0xFFFF) + +#define CR_REG_RECFIFO 0x04 +#define CR_RECDATA(a) ((a) & 0xFFFF) + +#define CR_REG_STATUS 0x08 +#define CR_RECOVER (1<<7) +#define CR_PLAYUNDER (1<<6) +#define CR_CODECREADY (1<<5) +#define CR_RAF (1<<4) +#define CR_RECEMPTY (1<<3) +#define CR_RECFULL (1<<2) +#define CR_PLAYHALF (1<<1) +#define CR_PLAYFULL (1<<0) + +#define CR_REG_RESETFIFO 0x0C +#define CR_RECRESET (1<<1) +#define CR_PLAYRESET (1<<0) + +#define CR_REG_CODEC_ADDR 0x10 +/* UG082 says: + * #define CR_CODEC_ADDR(a) ((a) << 1) + * #define CR_CODEC_READ (1<<0) + * #define CR_CODEC_WRITE (0<<0) + */ +/* RefDesign example says: */ +#define CR_CODEC_ADDR(a) ((a) << 0) +#define CR_CODEC_READ (1<<7) +#define CR_CODEC_WRITE (0<<7) + +#define CR_REG_CODEC_DATAREAD 0x14 +#define CR_CODEC_DATAREAD(v) ((v) & 0xFFFF) + +#define CR_REG_CODEC_DATAWRITE 0x18 +#define CR_CODEC_DATAWRITE(v) ((v) & 0xFFFF) + +#define CR_FIFO_SIZE 32 + +struct snd_ml403_ac97cr { + spinlock_t reg_lock; + + int irq; /* for playback */ + int enable_irq; /* for playback */ + + int capture_irq; + int enable_capture_irq; + + struct resource *res_port; + void *port; + + struct snd_ac97 *ac97; + int ac97_fake; +#ifdef CODEC_STAT + int ac97_read; + int ac97_write; +#endif + + struct platform_device *pfdev; + struct snd_card *card; + struct snd_pcm *pcm; + struct snd_pcm_substream *playback_substream; + struct snd_pcm_substream *capture_substream; + + struct snd_pcm_indirect2 ind_rec; /* for playback */ + struct snd_pcm_indirect2 capture_ind2_rec; +}; + +static struct snd_pcm_hardware snd_ml403_ac97cr_playback = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = CR_FIFO_SIZE/2, + .period_bytes_max = (64*1024), + .periods_min = 2, + .periods_max = (128*1024)/(CR_FIFO_SIZE/2), + .fifo_size = 0, +}; + +static struct snd_pcm_hardware snd_ml403_ac97cr_capture = { + .info = (SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_MMAP_VALID), + .formats = SNDRV_PCM_FMTBIT_S16_BE, + .rates = (SNDRV_PCM_RATE_CONTINUOUS | + SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_KNOT), + .rate_min = 4000, + .rate_max = 48000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = (128*1024), + .period_bytes_min = CR_FIFO_SIZE/2, + .period_bytes_max = (64*1024), + .periods_min = 2, + .periods_max = (128*1024)/(CR_FIFO_SIZE/2), + .fifo_size = 0, +}; + +static size_t +snd_ml403_ac97cr_playback_ind2_zero(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int copied_words = 0; + u32 full = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + spin_lock(&ml403_ac97cr->reg_lock); + while ((full = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_PLAYFULL)) != CR_PLAYFULL) { + out_be32(CR_REG(ml403_ac97cr, PLAYFIFO), 0); + copied_words++; + } + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static size_t +snd_ml403_ac97cr_playback_ind2_copy(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, size_t bytes) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + u16 *src; + int copied_words = 0; + u32 full = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + src = substream->runtime->dma_area + rec->sw_data; + + spin_lock(&ml403_ac97cr->reg_lock); + while (((full = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_PLAYFULL)) != CR_PLAYFULL) && (bytes > 1)) { + out_be32(CR_REG(ml403_ac97cr, PLAYFIFO), + CR_PLAYDATA(src[copied_words])); + copied_words++; + bytes = bytes - 2; + } + if (full != CR_PLAYFULL) + rec->hw_ready = 1; + else + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static size_t +snd_ml403_ac97cr_capture_ind2_null(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int copied_words = 0; + u32 empty = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + spin_lock(&ml403_ac97cr->reg_lock); + while ((empty = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RECEMPTY)) != CR_RECEMPTY) { + volatile u32 trash; + + trash = CR_RECDATA(in_be32(CR_REG(ml403_ac97cr, RECFIFO))); + /* Hmmmm, really necessary? */ + ++trash; + copied_words++; + } + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static size_t +snd_ml403_ac97cr_capture_ind2_copy(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, size_t bytes) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + u16 *dst; + int copied_words = 0; + u32 empty = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + dst = substream->runtime->dma_area + rec->sw_data; + + spin_lock(&ml403_ac97cr->reg_lock); + while (((empty = (in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RECEMPTY)) != CR_RECEMPTY) && (bytes > 1)) { + dst[copied_words] = CR_RECDATA(in_be32(CR_REG(ml403_ac97cr, + RECFIFO))); + copied_words++; + bytes = bytes - 2; + } + if (empty != CR_RECEMPTY) + rec->hw_ready = 1; + else + rec->hw_ready = 0; + spin_unlock(&ml403_ac97cr->reg_lock); + + return (size_t) (copied_words * 2); +} + +static snd_pcm_uframes_t +snd_ml403_ac97cr_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_indirect2 *ind2_rec = NULL; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + if (substream == ml403_ac97cr->playback_substream) + ind2_rec = &ml403_ac97cr->ind_rec; + if (substream == ml403_ac97cr->capture_substream) + ind2_rec = &ml403_ac97cr->capture_ind2_rec; + + if (ind2_rec != NULL) + return snd_pcm_indirect2_pointer(substream, ind2_rec); + return (snd_pcm_uframes_t) 0; +} + +static int +snd_ml403_ac97cr_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int err = 0; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + if (substream == ml403_ac97cr->playback_substream) { + PDEBUG(WORK_INFO, "trigger(playback): START\n"); + ml403_ac97cr->ind_rec.hw_ready = 1; + + /* clear play FIFO */ + out_be32(CR_REG(ml403_ac97cr, RESETFIFO), CR_PLAYRESET); + + /* enable play irq */ + ml403_ac97cr->enable_irq = 1; + enable_irq(ml403_ac97cr->irq); + } + if (substream == ml403_ac97cr->capture_substream) { + PDEBUG(WORK_INFO, "trigger(capture): START\n"); + ml403_ac97cr->capture_ind2_rec.hw_ready = 0; + + /* clear record FIFO */ + out_be32(CR_REG(ml403_ac97cr, RESETFIFO), CR_RECRESET); + + /* enable record irq */ + ml403_ac97cr->enable_capture_irq = 1; + enable_irq(ml403_ac97cr->capture_irq); + } + break; + case SNDRV_PCM_TRIGGER_STOP: + if (substream == ml403_ac97cr->playback_substream) { + PDEBUG(WORK_INFO, "trigger(playback): STOP\n"); + ml403_ac97cr->ind_rec.hw_ready = 0; +#ifdef SND_PCM_INDIRECT2_STAT + snd_pcm_indirect2_stat(substream, + &ml403_ac97cr->ind_rec); +#endif + /* disable play irq */ + disable_irq_nosync(ml403_ac97cr->irq); + ml403_ac97cr->enable_irq = 0; + } + if (substream == ml403_ac97cr->capture_substream) { + PDEBUG(WORK_INFO, "trigger(capture): STOP\n"); + ml403_ac97cr->capture_ind2_rec.hw_ready = 0; +#ifdef SND_PCM_INDIRECT2_STAT + snd_pcm_indirect2_stat(substream, + &ml403_ac97cr->capture_ind2_rec); +#endif + /* disable capture irq */ + disable_irq_nosync(ml403_ac97cr->capture_irq); + ml403_ac97cr->enable_capture_irq = 0; + } + break; + default: + err = -EINVAL; + break; + } + PDEBUG(WORK_INFO, "trigger(): (done)\n"); + return err; +} + +static int snd_ml403_ac97cr_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + if (substream == ml403_ac97cr->playback_substream) { + PDEBUG(WORK_INFO, + "prepare(): period_bytes=%d, minperiod_bytes=%d\n", + snd_pcm_lib_period_bytes(substream), CR_FIFO_SIZE / 2); + + /* set sampling rate */ + snd_ac97_set_rate(ml403_ac97cr->ac97, AC97_PCM_FRONT_DAC_RATE, + runtime->rate); + PDEBUG(WORK_INFO, "prepare(): rate=%d\n", runtime->rate); + + /* init struct for intermediate buffer */ + memset(&ml403_ac97cr->ind_rec, 0, + sizeof(struct snd_pcm_indirect2)); + ml403_ac97cr->ind_rec.hw_buffer_size = CR_FIFO_SIZE; + ml403_ac97cr->ind_rec.sw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + ml403_ac97cr->ind_rec.min_periods = -1; + ml403_ac97cr->ind_rec.min_multiple = + snd_pcm_lib_period_bytes(substream) / (CR_FIFO_SIZE / 2); + PDEBUG(WORK_INFO, "prepare(): hw_buffer_size=%d, " + "sw_buffer_size=%d, min_multiple=%d\n", + CR_FIFO_SIZE, ml403_ac97cr->ind_rec.sw_buffer_size, + ml403_ac97cr->ind_rec.min_multiple); + } + if (substream == ml403_ac97cr->capture_substream) { + PDEBUG(WORK_INFO, "prepare(capture): period_bytes=%d, " + "minperiod_bytes=%d\n", + snd_pcm_lib_period_bytes(substream), CR_FIFO_SIZE / 2); + + /* set sampling rate */ + snd_ac97_set_rate(ml403_ac97cr->ac97, AC97_PCM_LR_ADC_RATE, + runtime->rate); + PDEBUG(WORK_INFO, "prepare(capture): rate=%d\n", runtime->rate); + + /* init struct for intermediate buffer */ + memset(&ml403_ac97cr->capture_ind2_rec, 0, + sizeof(struct snd_pcm_indirect2)); + ml403_ac97cr->capture_ind2_rec.hw_buffer_size = CR_FIFO_SIZE; + ml403_ac97cr->capture_ind2_rec.sw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + ml403_ac97cr->capture_ind2_rec.min_multiple = + snd_pcm_lib_period_bytes(substream) / (CR_FIFO_SIZE / 2); + PDEBUG(WORK_INFO, "prepare(capture): hw_buffer_size=%d, " + "sw_buffer_size=%d, min_multiple=%d\n", CR_FIFO_SIZE, + ml403_ac97cr->capture_ind2_rec.sw_buffer_size, + ml403_ac97cr->capture_ind2_rec.min_multiple); + } + return 0; +} + +static int snd_ml403_ac97cr_hw_free(struct snd_pcm_substream *substream) +{ + PDEBUG(WORK_INFO, "hw_free()\n"); + return snd_pcm_lib_free_pages(substream); +} + +static int +snd_ml403_ac97cr_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + PDEBUG(WORK_INFO, "hw_params(): desired buffer bytes=%d, desired " + "period bytes=%d\n", + params_buffer_bytes(hw_params), params_period_bytes(hw_params)); + /* check period bytes, has to be multiple of CR_FIFO_SIZE / 2, don't + * know if ALSA takes multiples of period_bytes_min _only_ ...?! + */ + if (params_period_bytes(hw_params) % (CR_FIFO_SIZE / 2) != 0) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "hw_params(): period bytes (%d) are not a multiple " + "of %d bytes!\n", + params_period_bytes(hw_params), CR_FIFO_SIZE / 2); + return -EINVAL; + } + return snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); +} + +static int snd_ml403_ac97cr_playback_open(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + PDEBUG(WORK_INFO, "open(playback)\n"); + ml403_ac97cr->playback_substream = substream; + runtime->hw = snd_ml403_ac97cr_playback; + return 0; +} + +static int snd_ml403_ac97cr_capture_open(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct snd_pcm_runtime *runtime; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + + PDEBUG(WORK_INFO, "open(capture)\n"); + ml403_ac97cr->capture_substream = substream; + runtime->hw = snd_ml403_ac97cr_capture; + return 0; +} + +static int snd_ml403_ac97cr_playback_close(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + PDEBUG(WORK_INFO, "close(playback)\n"); + ml403_ac97cr->playback_substream = NULL; + return 0; +} + +static int snd_ml403_ac97cr_capture_close(struct snd_pcm_substream *substream) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + + ml403_ac97cr = snd_pcm_substream_chip(substream); + + PDEBUG(WORK_INFO, "close(capture)\n"); + ml403_ac97cr->capture_substream = NULL; + return 0; +} + +struct snd_pcm_ops snd_ml403_ac97cr_playback_ops = { + .open = snd_ml403_ac97cr_playback_open, + .close = snd_ml403_ac97cr_playback_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ml403_ac97cr_hw_params, + .hw_free = snd_ml403_ac97cr_hw_free, + .prepare = snd_ml403_ac97cr_pcm_prepare, + .trigger = snd_ml403_ac97cr_pcm_trigger, + .pointer = snd_ml403_ac97cr_pcm_pointer, +}; + +static struct snd_pcm_ops snd_ml403_ac97cr_capture_ops = { + .open = snd_ml403_ac97cr_capture_open, + .close = snd_ml403_ac97cr_capture_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_ml403_ac97cr_hw_params, + .hw_free = snd_ml403_ac97cr_hw_free, + .prepare = snd_ml403_ac97cr_pcm_prepare, + .trigger = snd_ml403_ac97cr_pcm_trigger, + .pointer = snd_ml403_ac97cr_pcm_pointer, +}; + +irqreturn_t snd_ml403_ac97cr_irq(int irq, void *dev_id, struct pt_regs *regs) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + struct platform_device *pfdev; + int cmp_irq; + + if ((ml403_ac97cr = (struct snd_ml403_ac97cr *)dev_id) == NULL) + return IRQ_NONE; + + pfdev = ml403_ac97cr->pfdev; + + /* playback interrupt */ + cmp_irq = platform_get_irq(pfdev, 0); + if (irq == cmp_irq) { + if (ml403_ac97cr->enable_irq) { + snd_pcm_indirect2_playback_interrupt( + ml403_ac97cr->playback_substream, + &ml403_ac97cr->ind_rec, + snd_ml403_ac97cr_playback_ind2_copy, + snd_ml403_ac97cr_playback_ind2_zero); + } else + goto __disable_irq; + } else { + /* record interrupt */ + cmp_irq = platform_get_irq(pfdev, 1); + if (irq == cmp_irq) { + if (ml403_ac97cr->enable_capture_irq) { + snd_pcm_indirect2_capture_interrupt( + ml403_ac97cr->capture_substream, + &ml403_ac97cr->capture_ind2_rec, + snd_ml403_ac97cr_capture_ind2_copy, + snd_ml403_ac97cr_capture_ind2_null); + } else { + goto __disable_irq; + } + } else { + return IRQ_NONE; + } + } + return IRQ_HANDLED; + + __disable_irq: + PDEBUG(INIT_INFO, "irq(): irq %d is meant to be disabled! So, now try " + "to disable it _really_!\n", irq); + disable_irq_nosync(irq); + return IRQ_HANDLED; +} + +static unsigned short +snd_ml403_ac97cr_codec_read(struct snd_ac97 *ac97, unsigned short reg) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data; +#ifdef CODEC_STAT + u32 stat, rafaccess = 0; +#endif + unsigned long end_time; + u16 value = 0; + + if (!LM4550_RF_OK(reg)) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "access to unknown/unused codec register 0x%x " + "ignored!\n", reg); + return 0; + } + /* check if we can fake/answer this access from our shadow register */ + if ((LM4550_RF_FLAG(reg) & + (LM4550_REG_DONEREAD | LM4550_REG_ALLFAKE)) && + !(LM4550_RF_FLAG(reg) & LM4550_REG_NOSHADOW)) { + if (LM4550_RF_FLAG(reg) & LM4550_REG_FAKEREAD) { + PDEBUG(CODEC_FAKE, "codec_read(): faking read from " + "reg=0x%x, val=0x%x / %d\n", + reg, LM4550_RF_DEF(reg), LM4550_RF_DEF(reg)); + return LM4550_RF_DEF(reg); + } else if ((LM4550_RF_FLAG(reg) & LM4550_REG_FAKEPROBE) && + ml403_ac97cr->ac97_fake) { + PDEBUG(CODEC_FAKE, "codec_read(): faking read from " + "reg=0x%x, val=0x%x / %d (probe)\n", + reg, LM4550_RF_VAL(reg), LM4550_RF_VAL(reg)); + return LM4550_RF_VAL(reg); + } else { +#ifdef CODEC_STAT + PDEBUG(CODEC_FAKE, "codec_read(): read access " + "answered by shadow register 0x%x (value=0x%x " + "/ %d) (cw=%d cr=%d)\n", + reg, LM4550_RF_VAL(reg), LM4550_RF_VAL(reg), + ml403_ac97cr->ac97_write, + ml403_ac97cr->ac97_read); +#else + PDEBUG(CODEC_FAKE, "codec_read(): read access " + "answered by shadow register 0x%x (value=0x%x " + "/ %d)\n", + reg, LM4550_RF_VAL(reg), LM4550_RF_VAL(reg)); +#endif + return LM4550_RF_VAL(reg); + } + } + /* if we are here, we _have_ to access the codec really, no faking */ + spin_lock(&ml403_ac97cr->reg_lock); +#ifdef CODEC_STAT + ml403_ac97cr->ac97_read++; +#endif + out_be32(CR_REG(ml403_ac97cr, CODEC_ADDR), + CR_CODEC_ADDR(reg) | CR_CODEC_READ); + end_time = jiffies + (HZ / CODEC_TIMEOUT_AFTER_READ); + do { +#ifdef CODEC_STAT + rafaccess++; + if (((stat = in_be32(CR_REG(ml403_ac97cr, STATUS))) & + CR_RAF) == CR_RAF) { + value = CR_CODEC_DATAREAD( + in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD))); + PDEBUG(CODEC_SUCCESS, "codec_read(): (done) reg=0x%x, " + "value=0x%x / %d (STATUS=0x%x)\n", + reg, value, value, stat); +#else + if ((in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RAF) == CR_RAF) { + value = CR_CODEC_DATAREAD( + in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD))); + PDEBUG(CODEC_SUCCESS, "codec_read(): (done) " + "reg=0x%x, value=0x%x / %d\n", + reg, value, value); +#endif + lm4550_regfile[reg / 2].value = value; + lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD; + spin_unlock(&ml403_ac97cr->reg_lock); + return value; + } + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); + value = + CR_CODEC_DATAREAD(in_be32(CR_REG(ml403_ac97cr, CODEC_DATAREAD))); +#ifdef CODEC_STAT + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec read! " + "(reg=0x%x, last STATUS=0x%x, DATAREAD=0x%x / %d, %d) " + "(cw=%d, cr=%d)\n", + reg, stat, value, value, rafaccess, ml403_ac97cr->ac97_write, + ml403_ac97cr->ac97_read); +#else + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec read! (reg=0x%x, DATAREAD=0x%x / %d)\n", + reg, value, value); +#endif + /* BUG: This is PURE speculation! But after _most_ read timeouts the + * value in the register is ok! + */ + lm4550_regfile[reg / 2].value = value; + lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD; + spin_unlock(&ml403_ac97cr->reg_lock); + return value; +} + +static void +snd_ml403_ac97cr_codec_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data; + +#ifdef CODEC_STAT + u32 stat, rafaccess = 0; +#endif +#ifdef CODEC_WRITE_CHECK_RAF + unsigned long end_time; +#endif + + if (!LM4550_RF_OK(reg)) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "access to unknown/unused codec register 0x%x " + "ignored!\n", reg); + return; + } + if (LM4550_RF_FLAG(reg) & LM4550_REG_READONLY) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "write access to read only codec register 0x%x " + "ignored!\n", reg); + return; + } + if ((val & LM4550_RF_WMASK(reg)) != val) { + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "write access to codec register 0x%x with bad value " + "0x%x / %d!\n", + reg, val, val); + val = val & LM4550_RF_WMASK(reg); + } + if (((LM4550_RF_FLAG(reg) & LM4550_REG_FAKEPROBE) && + ml403_ac97cr->ac97_fake) && + !(LM4550_RF_FLAG(reg) & LM4550_REG_NOSHADOW)) { + PDEBUG(CODEC_FAKE, "codec_write(): faking write to reg=0x%x, " + "val=0x%x / %d\n", reg, val, val); + LM4550_RF_VAL(reg) = (val & LM4550_RF_WMASK(reg)); + return; + } + spin_lock(&ml403_ac97cr->reg_lock); +#ifdef CODEC_STAT + ml403_ac97cr->ac97_write++; +#endif + out_be32(CR_REG(ml403_ac97cr, CODEC_DATAWRITE), + CR_CODEC_DATAWRITE(val)); + out_be32(CR_REG(ml403_ac97cr, CODEC_ADDR), + CR_CODEC_ADDR(reg) | CR_CODEC_WRITE); +#ifdef CODEC_WRITE_CHECK_RAF + /* check CR_CODEC_RAF bit to see if write access to register is done; + * loop until bit is set or timeout happens + */ + end_time = jiffies + HZ / CODEC_TIMEOUT_AFTER_WRITE; + do { +#ifdef CODEC_STAT + rafaccess++; + if (((stat = in_be32(CR_REG(ml403_ac97cr, STATUS))) & + CR_RAF) == CR_RAF) { +#else + if ((in_be32(CR_REG(ml403_ac97cr, STATUS)) & + CR_RAF) == CR_RAF) { +#endif + PDEBUG(CODEC_SUCCESS, "codec_write(): (done) reg=0x%x, " + "value=%d / 0x%x\n", reg, val, val); + if (!(LM4550_RF_FLAG(reg) & LM4550_REG_NOSHADOW) && + !(LM4550_RF_FLAG(reg) & LM4550_REG_NOSAVE)) + lm4550_regfile[reg / 2].value = val; + lm4550_regfile[reg / 2].flag |= LM4550_REG_DONEREAD; + spin_unlock(&ml403_ac97cr->reg_lock); + return; + } + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); +#ifdef CODEC_STAT + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec write " + "(reg=0x%x, val=0x%x / %d, last STATUS=0x%x, %d) " + "(cw=%d, cr=%d)\n", + reg, val, val, stat, rafaccess, ml403_ac97cr->ac97_write, + ml403_ac97cr->ac97_read); +#else + snd_printk(KERN_WARNING SND_ML403_AC97CR_DRIVER ": " + "timeout while codec write (reg=0x%x, val=0x%x / %d)\n", + reg, val, val); +#endif +#else /* CODEC_WRITE_CHECK_RAF */ +#if CODEC_WAIT_AFTER_WRITE > 0 + /* officially, in AC97 spec there is no possibility for a AC97 + * controller to determine, if write access is done or not - so: How + * is Xilinx able to provide a RAF bit for write access? + * => very strange, thus just don't check RAF bit (compare with + * Xilinx's example app in EDK 8.1i) and wait + */ + schedule_timeout_uninterruptible(HZ / CODEC_WAIT_AFTER_WRITE); +#endif + PDEBUG(CODEC_SUCCESS, "codec_write(): (done) reg=0x%x, value=%d / 0x%x " + "(no RAF check)\n", + reg, val, val); +#endif + spin_unlock(&ml403_ac97cr->reg_lock); + return; +} + +static int snd_ml403_ac97cr_chip_init(struct snd_ml403_ac97cr *ml403_ac97cr) +{ + unsigned long end_time; + PDEBUG(INIT_INFO, "chip_init():\n"); + end_time = jiffies + HZ / CODEC_TIMEOUT_ON_INIT; + do { + if (in_be32(CR_REG(ml403_ac97cr, STATUS)) & CR_CODECREADY) { + /* clear both hardware FIFOs */ + out_be32(CR_REG(ml403_ac97cr, RESETFIFO), + CR_RECRESET | CR_PLAYRESET); + PDEBUG(INIT_INFO, "chip_init(): (done)\n"); + return 0; + } + schedule_timeout_uninterruptible(1); + } while (time_after(end_time, jiffies)); + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "timeout while waiting for codec, " + "not ready!\n"); + return -EBUSY; +} + +static int snd_ml403_ac97cr_free(struct snd_ml403_ac97cr *ml403_ac97cr) +{ + PDEBUG(INIT_INFO, "free():\n"); + /* irq release */ + if (ml403_ac97cr->irq >= 0) + free_irq(ml403_ac97cr->irq, ml403_ac97cr); + if (ml403_ac97cr->capture_irq >= 0) + free_irq(ml403_ac97cr->capture_irq, ml403_ac97cr); + /* give back "port" */ + if (ml403_ac97cr->port != NULL) + iounmap(ml403_ac97cr->port); + kfree(ml403_ac97cr); + PDEBUG(INIT_INFO, "free(): (done)\n"); + return 0; +} + +static int snd_ml403_ac97cr_dev_free(struct snd_device *snddev) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = snddev->device_data; + PDEBUG(INIT_INFO, "dev_free():\n"); + return snd_ml403_ac97cr_free(ml403_ac97cr); +} + +static int __init +snd_ml403_ac97cr_create(struct snd_card *card, struct platform_device *pfdev, + struct snd_ml403_ac97cr **rml403_ac97cr) +{ + struct snd_ml403_ac97cr *ml403_ac97cr; + int err; + static struct snd_device_ops ops = { + .dev_free = snd_ml403_ac97cr_dev_free, + }; + struct resource *resource; + int irq; + + *rml403_ac97cr = NULL; + ml403_ac97cr = kzalloc(sizeof(*ml403_ac97cr), GFP_KERNEL); + if (ml403_ac97cr == NULL) { + return -ENOMEM; + } + spin_lock_init(&ml403_ac97cr->reg_lock); + ml403_ac97cr->card = card; + ml403_ac97cr->pfdev = pfdev; + ml403_ac97cr->irq = -1; + ml403_ac97cr->enable_irq = 0; + ml403_ac97cr->capture_irq = -1; + ml403_ac97cr->enable_capture_irq = 0; + ml403_ac97cr->port = NULL; + ml403_ac97cr->res_port = NULL; + + PDEBUG(INIT_INFO, "Trying to reserve resources now ...\n"); + resource = platform_get_resource(pfdev, IORESOURCE_MEM, 0); + /* get "port" */ + ml403_ac97cr->port = ioremap_nocache(resource->start, + (resource->end) - + (resource->start) + 1); + if (ml403_ac97cr->port == NULL) { + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "unable to remap memory region (%x to %x)\n", + resource->start, resource->end); + snd_ml403_ac97cr_free(ml403_ac97cr); + return -EBUSY; + } + snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": " + "remap controller memory region to " + "0x%x done\n", (unsigned int)ml403_ac97cr->port); + /* get irq */ + irq = platform_get_irq(pfdev, 0); + if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "unable to grab IRQ %d\n", + irq); + snd_ml403_ac97cr_free(ml403_ac97cr); + return -EBUSY; + } + ml403_ac97cr->irq = irq; + snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": " + "request (playback) irq %d done\n", + ml403_ac97cr->irq); + irq = platform_get_irq(pfdev, 1); + if (request_irq(irq, snd_ml403_ac97cr_irq, IRQF_DISABLED, + pfdev->dev.bus_id, (void *)ml403_ac97cr)) { + snd_printk(KERN_ERR SND_ML403_AC97CR_DRIVER ": " + "unable to grab IRQ %d\n", + irq); + snd_ml403_ac97cr_free(ml403_ac97cr); + return -EBUSY; + } + ml403_ac97cr->capture_irq = irq; + snd_printk(KERN_INFO SND_ML403_AC97CR_DRIVER ": " + "request (capture) irq %d done\n", + ml403_ac97cr->capture_irq); + + if ((err = snd_ml403_ac97cr_chip_init(ml403_ac97cr)) < 0) { + snd_ml403_ac97cr_free(ml403_ac97cr); + return err; + } + + err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, ml403_ac97cr, &ops); + if (err < 0) { + PDEBUG(INIT_FAILURE, "probe(): snd_device_new() failed!\n"); + snd_ml403_ac97cr_free(ml403_ac97cr); + return err; + } + + snd_card_set_dev(card, &pfdev->dev); + + *rml403_ac97cr = ml403_ac97cr; + return 0; +} + +static void snd_ml403_ac97cr_mixer_free(struct snd_ac97 *ac97) +{ + struct snd_ml403_ac97cr *ml403_ac97cr = ac97->private_data; + PDEBUG(INIT_INFO, "mixer_free():\n"); + ml403_ac97cr->ac97 = NULL; + PDEBUG(INIT_INFO, "mixer_free(): (done)\n"); +} + +static int __init snd_ml403_ac97cr_mixer(struct snd_ml403_ac97cr *ml403_ac97cr) +{ + struct snd_ac97_bus *bus; + struct snd_ac97_template ac97; + int err; + static struct snd_ac97_bus_ops ops = { + .write = snd_ml403_ac97cr_codec_write, + .read = snd_ml403_ac97cr_codec_read, + }; + PDEBUG(INIT_INFO, "mixer():\n"); + if ((err = snd_ac97_bus(ml403_ac97cr->card, 0, &ops, NULL, &bus)) < 0) + return err; + + memset(&ac97, 0, sizeof(ac97)); + ml403_ac97cr->ac97_fake = 1; + lm4550_regfile_init(); +#ifdef CODEC_STAT + ml403_ac97cr->ac97_read = 0; + ml403_ac97cr->ac97_write = 0; +#endif + ac97.private_data = ml403_ac97cr; + ac97.private_free = snd_ml403_ac97cr_mixer_free; + ac97.scaps = AC97_SCAP_AUDIO | AC97_SCAP_SKIP_MODEM | + AC97_SCAP_NO_SPDIF; + err = snd_ac97_mixer(bus, &ac97, &ml403_ac97cr->ac97); + ml403_ac97cr->ac97_fake = 0; + lm4550_regfile_write_values_after_init(ml403_ac97cr->ac97); + PDEBUG(INIT_INFO, "mixer(): (done) snd_ac97_mixer()=%d\n", err); + return err; +} + +static int __init +snd_ml403_ac97cr_pcm(struct snd_ml403_ac97cr *ml403_ac97cr, int device, + struct snd_pcm **rpcm) +{ + struct snd_pcm *pcm; + int err; + + if (rpcm) + *rpcm = NULL; + if ((err = snd_pcm_new(ml403_ac97cr->card, "ML403AC97CR/1", device, 1, + 1, &pcm)) < 0) + return err; + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_ml403_ac97cr_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_ml403_ac97cr_capture_ops); + pcm->private_data = ml403_ac97cr; + pcm->info_flags = 0; + strcpy(pcm->name, "ML403AC97CR DAC/ADC"); + ml403_ac97cr->pcm = pcm; + + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + 64 * 1024, + 128 * 1024); + if (rpcm) + *rpcm = pcm; + return 0; +} + +static int __init snd_ml403_ac97cr_probe(struct platform_device *pfdev) +{ + struct snd_card *card; + struct snd_ml403_ac97cr *ml403_ac97cr = NULL; + int err; + int dev = pfdev->id; + + if (dev >= SNDRV_CARDS) + return -ENODEV; + if (!enable[dev]) + return -ENOENT; + + card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0); + if (card == NULL) + return -ENOMEM; + if ((err = snd_ml403_ac97cr_create(card, pfdev, &ml403_ac97cr)) < 0) { + PDEBUG(INIT_FAILURE, "probe(): create failed!\n"); + snd_card_free(card); + return err; + } + PDEBUG(INIT_INFO, "probe(): create done\n"); + card->private_data = ml403_ac97cr; + if ((err = snd_ml403_ac97cr_mixer(ml403_ac97cr)) < 0) { + snd_card_free(card); + return err; + } + PDEBUG(INIT_INFO, "probe(): mixer done\n"); + if ((err = snd_ml403_ac97cr_pcm(ml403_ac97cr, 0, NULL)) < 0) { + snd_card_free(card); + return err; + } + PDEBUG(INIT_INFO, "probe(): PCM done\n"); + strcpy(card->driver, SND_ML403_AC97CR_DRIVER); + strcpy(card->shortname, "ML403 AC97 Controller Reference"); + sprintf(card->longname, "%s %s at 0x%lx, irq %i & %i, device %i", + card->shortname, card->driver, + (unsigned long)ml403_ac97cr->port, ml403_ac97cr->irq, + ml403_ac97cr->capture_irq, dev + 1); + + if ((err = snd_card_register(card)) < 0) { + snd_card_free(card); + return err; + } + platform_set_drvdata(pfdev, card); + PDEBUG(INIT_INFO, "probe(): (done)\n"); + return 0; +} + +static int snd_ml403_ac97cr_remove(struct platform_device *pfdev) +{ + snd_card_free(platform_get_drvdata(pfdev)); + platform_set_drvdata(pfdev, NULL); + return 0; +} + +static struct platform_driver snd_ml403_ac97cr_driver = { + .probe = snd_ml403_ac97cr_probe, + .remove = snd_ml403_ac97cr_remove, + .driver = { + .name = SND_ML403_AC97CR_DRIVER, + }, +}; + +static int __init alsa_card_ml403_ac97cr_init(void) +{ + return platform_driver_register(&snd_ml403_ac97cr_driver); +} + +static void __exit alsa_card_ml403_ac97cr_exit(void) +{ + platform_driver_unregister(&snd_ml403_ac97cr_driver); +} + +module_init(alsa_card_ml403_ac97cr_init) +module_exit(alsa_card_ml403_ac97cr_exit) diff --git a/sound/ppc/pcm-indirect2.h b/sound/ppc/pcm-indirect2.h new file mode 100644 index 0000000..c7fe74f --- /dev/null +++ b/sound/ppc/pcm-indirect2.h @@ -0,0 +1,658 @@ +/* + * Helper functions for indirect PCM data transfer to a simple FIFO in + * hardware (small, no possibility to read "hardware io position", + * updating position done by interrupt, ...) + * + * Copyright (c) by 2007 Joachim Foerster <JOFT(a)gmx.de> + * + * Based on "pcm-indirect.h" (alsa-driver-1.0.13) by + * + * Copyright (c) by Takashi Iwai <tiwai(a)suse.de> + * Jaroslav Kysela <perex(a)suse.cz> + * + * 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., 675 Mass Ave, Cambridge, MA 02139, USA. + */ + +#ifndef __SOUND_PCM_INDIRECT2_H +#define __SOUND_PCM_INDIRECT2_H + +#include <sound/core.h> +#include <sound/pcm.h> + +/* jiffies */ +#include <linux/jiffies.h> + +struct snd_pcm_indirect2 { + unsigned int hw_buffer_size; /* Byte size of hardware buffer */ + int hw_ready; /* playback: 1 = hw fifo has room left, + * 0 = hw fifo is full + */ + unsigned int min_multiple; + int min_periods; /* counts number of min. periods until + * min_multiple is reached + */ + int min_period_count; /* counts bytes to count number of + * min. periods + */ + + unsigned int sw_buffer_size; /* Byte size of software buffer */ + + /* sw_data: position in intermediate buffer, where we will read (or + * write) from/to next time (to transfer data to/from HW) + */ + unsigned int sw_data; /* Offset to next dst (or src) in sw + * ring buffer + */ + /* easiest case (playback): + * sw_data is nearly the same as ~ runtime->control->appl_ptr, with the + * exception that sw_data is "behind" by the number if bytes ALSA wrote + * to the intermediate buffer last time. + * A call to ack() callback synchronizes both indirectly. + */ + + /* We have no real sw_io pointer here. Usually sw_io is pointing to the + * current playback/capture position _inside_ the hardware. Devices + * with plain FIFOs often have no possibility to publish this position. + * So we say: if sw_data is updated, that means bytes were copied to + * the hardware, we increase sw_io by that amount, because there have + * to be as much bytes which were played. So sw_io will stay behind + * sw_data all the time and has to converge to sw_data at the end of + * playback. + */ + unsigned int sw_io; /* Current software pointer in bytes */ + + /* sw_ready: number of bytes ALSA copied to the intermediate buffer, so + * it represents the number of bytes which wait for transfer to the HW + */ + int sw_ready; /* Bytes ready to be transferred to/from hw */ + + /* appl_ptr: last known position of ALSA (where ALSA is going to write + * next time into the intermediate buffer + */ + snd_pcm_uframes_t appl_ptr; /* Last seen appl_ptr */ + + unsigned int bytes2hw; + int check_alignment; + +#ifdef SND_PCM_INDIRECT2_STAT + unsigned int zeros2hw; + unsigned int mul_elapsed; + unsigned int mul_elapsed_real; + unsigned long firstbytetime; + unsigned long lastbytetime; + unsigned long firstzerotime; + unsigned int byte_sizes[64]; + unsigned int zero_sizes[64]; + unsigned int min_adds[8]; + unsigned int mul_adds[8]; + unsigned int zero_times[3750]; /* = 15s */ + unsigned int zero_times_saved; + unsigned int zero_times_notsaved; + unsigned int irq_occured; + unsigned int pointer_calls; + unsigned int lastdifftime; +#endif +}; + +typedef size_t(*snd_pcm_indirect2_copy_t) (struct snd_pcm_substream * substream, + struct snd_pcm_indirect2 * rec, + size_t bytes); +typedef size_t(*snd_pcm_indirect2_zero_t) (struct snd_pcm_substream * substream, + struct snd_pcm_indirect2 * rec); + +#ifdef SND_PCM_INDIRECT2_STAT +static inline void snd_pcm_indirect2_stat(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int i, j, k; + int seconds = (rec->lastbytetime - rec->firstbytetime) / HZ; + + snd_printk(KERN_DEBUG "STAT: mul_elapsed: %u, mul_elapsed_real: %d, " + "irq_occured: %d\n", + rec->mul_elapsed, rec->mul_elapsed_real, rec->irq_occured); + snd_printk(KERN_DEBUG "STAT: min_multiple: %d (irqs/period)\n", + rec->min_multiple); + snd_printk(KERN_DEBUG "STAT: firstbytetime: %lu, lastbytetime: %lu, " + "firstzerotime: %lu\n", + rec->firstbytetime, rec->lastbytetime, rec->firstzerotime); + snd_printk(KERN_DEBUG "STAT: bytes2hw: %u Bytes => (by runtime->rate) " + "length: %d s\n", + rec->bytes2hw, rec->bytes2hw / 2 / 2 / runtime->rate); + snd_printk(KERN_DEBUG "STAT: (by measurement) length: %d => " + "rate: %d Bytes/s = %d Frames/s|Hz\n", + seconds, rec->bytes2hw / seconds, + rec->bytes2hw / 2 / 2 / seconds); + snd_printk(KERN_DEBUG + "STAT: zeros2hw: %u = %d ms ~ %d * %d zero copies\n", + rec->zeros2hw, ((rec->zeros2hw / 2 / 2) * 1000) / + runtime->rate, + rec->zeros2hw / (rec->hw_buffer_size / 2), + (rec->hw_buffer_size / 2)); + snd_printk(KERN_DEBUG "STAT: pointer_calls: %u, lastdifftime: %u\n", + rec->pointer_calls, rec->lastdifftime); + snd_printk(KERN_DEBUG "STAT: sw_io: %d, sw_data: %d\n", rec->sw_io, + rec->sw_data); + snd_printk(KERN_DEBUG "STAT: byte_sizes[]:\n"); + k = 0; + for (j = 0; j < 8; j++) { + for (i = j * 8; i < (j + 1) * 8; i++) + if (rec->byte_sizes[i] != 0) { + snd_printk(KERN_DEBUG "%u: %u", + i, rec->byte_sizes[i]); + k++; + } + if (((k % 8) == 0) && (k != 0)) { + snd_printk(KERN_DEBUG "\n"); + k = 0; + } + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG "STAT: zero_sizes[]:\n"); + for (j = 0; j < 8; j++) { + k = 0; + for (i = j * 8; i < (j + 1) * 8; i++) + if (rec->zero_sizes[i] != 0) + snd_printk(KERN_DEBUG "%u: %u", + i, rec->zero_sizes[i]); + else + k++; + if (!k) + snd_printk(KERN_DEBUG "\n"); + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG "STAT: min_adds[]:\n"); + for (j = 0; j < 8; j++) { + if (rec->min_adds[j] != 0) + snd_printk(KERN_DEBUG "%u: %u", j, rec->min_adds[j]); + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG "STAT: mul_adds[]:\n"); + for (j = 0; j < 8; j++) { + if (rec->mul_adds[j] != 0) + snd_printk(KERN_DEBUG "%u: %u", j, rec->mul_adds[j]); + } + snd_printk(KERN_DEBUG "\n"); + snd_printk(KERN_DEBUG + "STAT: zero_times_saved: %d, zero_times_notsaved: %d\n", + rec->zero_times_saved, rec->zero_times_notsaved); + /* snd_printk(KERN_DEBUG "STAT: zero_times[]\n"); + i = 0; + for (j = 0; j < 3750; j++) { + if (rec->zero_times[j] != 0) { + snd_printk(KERN_DEBUG "%u: %u", j, rec->zero_times[j]); + i++; + } + if (((i % 8) == 0) && (i != 0)) + snd_printk(KERN_DEBUG "\n"); + } + snd_printk(KERN_DEBUG "\n"); */ + return; +} +#endif + +/* + * _internal_ helper function for playback/capture transfer function + */ +static inline void +snd_pcm_indirect2_increase_min_periods(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + int isplay, int iscopy, + unsigned int bytes) +{ + if (rec->min_periods >= 0) { + if (iscopy) { + rec->sw_io += bytes; + if (rec->sw_io >= rec->sw_buffer_size) + rec->sw_io -= rec->sw_buffer_size; + } else if (isplay) { + /* If application does not write data in multiples of + * a period, move sw_data to the next correctly aligned + * position, so that sw_io can converge to it (in the + * next step). + */ + if (!rec->check_alignment) { + if (rec->bytes2hw % + snd_pcm_lib_period_bytes(substream)) { + unsigned bytes2hw_aligned = + (1 + + (rec->bytes2hw / + snd_pcm_lib_period_bytes + (substream))) * + snd_pcm_lib_period_bytes(substream); + rec->sw_data = + bytes2hw_aligned % + rec->sw_buffer_size; +#ifdef SND_PCM_INDIRECT2_STAT + snd_printk(KERN_DEBUG + "STAT: @re-align: aligned " + "bytes2hw to next period " + "size boundary: %d " + "(instead of %d)\n", + bytes2hw_aligned, + rec->bytes2hw); + snd_printk(KERN_DEBUG + "STAT: @re-align: sw_data " + "moves to: %d\n", + rec->sw_data); +#endif + } + rec->check_alignment = 1; + } + /* We are at the end and are copying zero into the fifo. + * Now, we have to make sure that sw_io is increased + * until the position of sw_data: Filling the fifo with + * the first zeros means, the last bytes were played. + */ + if (rec->sw_io != rec->sw_data) { + unsigned int diff; + if (rec->sw_data > rec->sw_io) + diff = rec->sw_data - rec->sw_io; + else + diff = + (rec->sw_buffer_size - rec->sw_io) + + rec->sw_data; + if (bytes >= diff) + rec->sw_io = rec->sw_data; + else { + rec->sw_io += bytes; + if (rec->sw_io >= rec->sw_buffer_size) + rec->sw_io -= + rec->sw_buffer_size; + } + } + } + rec->min_period_count += bytes; + if (rec->min_period_count >= (rec->hw_buffer_size / 2)) { + rec->min_periods += + (rec->min_period_count / (rec->hw_buffer_size / 2)); +#ifdef SND_PCM_INDIRECT2_STAT + if ((rec->min_period_count / + (rec->hw_buffer_size / 2)) > 7) + snd_printk(KERN_DEBUG + "STAT: more than 7 (%d) min_adds at " + "once - too big to save!\n", + (rec->min_period_count / + (rec->hw_buffer_size / 2))); + else + rec->min_adds[(rec->min_period_count / + (rec->hw_buffer_size / 2))]++; +#endif + rec->min_period_count = + (rec->min_period_count % (rec->hw_buffer_size / 2)); + } + } else if (isplay && iscopy) + rec->min_periods = 0; +} + +/* + * helper function for playback/capture pointer callback + */ +static inline snd_pcm_uframes_t +snd_pcm_indirect2_pointer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec) +{ +#ifdef SND_PCM_INDIRECT2_STAT + rec->pointer_calls++; +#endif + return bytes_to_frames(substream->runtime, rec->sw_io); +} + +/* + * helper function for playback ack callback + */ +static inline void +snd_pcm_indirect2_playback_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t zero) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr; + + /* runtime->control->appl_ptr: position where ALSA will write next time + * rec->appl_ptr: position where ALSA was last time + * diff: obviously ALSA wrote that much bytes into the intermediate + * buffer since we checked last time + */ + snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr; + + if (diff) { +#ifdef SND_PCM_INDIRECT2_STAT + rec->lastdifftime = jiffies; +#endif + if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) + diff += runtime->boundary; + /* number of bytes "added" by ALSA increases the number of bytes + * which are ready to "be transfered to HW"/"played" + * Then, set rec->appl_ptr to not count bytes twice next time. + */ + rec->sw_ready += (int)frames_to_bytes(runtime, diff); + rec->appl_ptr = appl_ptr; + } + if (rec->hw_ready && (rec->sw_ready <= 0)) { + unsigned int bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstzerotime == 0) { + rec->firstzerotime = jiffies; + snd_printk(KERN_DEBUG + "STAT: @firstzerotime: mul_elapsed: %d, " + "min_period_count: %d\n", + rec->mul_elapsed, rec->min_period_count); + snd_printk(KERN_DEBUG + "STAT: @firstzerotime: sw_io: %d, " + "sw_data: %d, appl_ptr: %u\n", + rec->sw_io, rec->sw_data, + (unsigned int)appl_ptr); + } + if ((jiffies - rec->firstzerotime) < 3750) { + rec->zero_times[(jiffies - rec->firstzerotime)]++; + rec->zero_times_saved++; + } else + rec->zero_times_notsaved++; +#endif + bytes = zero(substream, rec); + +#ifdef SND_PCM_INDIRECT2_STAT + rec->zeros2hw += bytes; + if (bytes < 64) + rec->zero_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: %d zero Bytes copied to hardware at " + "once - too big to save!\n", + bytes); +#endif + snd_pcm_indirect2_increase_min_periods(substream, rec, 1, 0, + bytes); + return; + } + while (rec->hw_ready && (rec->sw_ready > 0)) { + /* sw_to_end: max. number of bytes that can be read/take from + * the current position (sw_data) in _one_ step + */ + unsigned int sw_to_end = rec->sw_buffer_size - rec->sw_data; + + /* bytes: number of bytes we have available (for reading) */ + unsigned int bytes = rec->sw_ready; + + if (sw_to_end < bytes) { + bytes = sw_to_end; + } + if (!bytes) + break; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstbytetime == 0) + rec->firstbytetime = jiffies; + rec->lastbytetime = jiffies; +#endif + /* copy bytes from intermediate buffer position sw_data to the + * HW and return number of bytes actually written + * Furthermore, set hw_ready to 0, if the fifo isn't empty + * now => more could be transfered to fifo + */ + bytes = copy(substream, rec, bytes); + rec->bytes2hw += bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (bytes < 64) + rec->byte_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: %d Bytes copied to hardware at once " + "- too big to save!\n", + bytes); +#endif + /* increase sw_data by the number of actually written bytes + * (= number of taken bytes from intermediate buffer) + */ + rec->sw_data += bytes; + if (rec->sw_data == rec->sw_buffer_size) + rec->sw_data = 0; + /* now sw_data is the position where ALSA is going to write + * in the intermediate buffer next time = position we are going + * to read from next time + */ + + snd_pcm_indirect2_increase_min_periods(substream, rec, 1, 1, + bytes); + + /* we read bytes from intermediate buffer, so we need to say + * that the number of bytes ready for transfer are decreased + * now + */ + rec->sw_ready -= bytes; + } + return; +} + +/* + * helper function for playback interrupt routine + */ +static inline void +snd_pcm_indirect2_playback_interrupt(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t zero) +{ +#ifdef SND_PCM_INDIRECT2_STAT + rec->irq_occured++; +#endif + /* hardware played some bytes, so there is room again (in fifo) */ + rec->hw_ready = 1; + + /* don't call ack() now, instead call transfer() function directly + * (normally called by ack() ) + */ + snd_pcm_indirect2_playback_transfer(substream, rec, copy, zero); + + if (rec->min_periods >= rec->min_multiple) { +#ifdef SND_PCM_INDIRECT2_STAT + if ((rec->min_periods / rec->min_multiple) > 7) + snd_printk(KERN_DEBUG + "STAT: more than 7 (%d) mul_adds - too big " + "to save!\n", + (rec->min_periods / rec->min_multiple)); + else + rec->mul_adds[(rec->min_periods / rec->min_multiple)]++; + rec->mul_elapsed_real += (rec->min_periods / rec->min_multiple); + rec->mul_elapsed++; +#endif + rec->min_periods = 0; + snd_pcm_period_elapsed(substream); + } +} + +/* + * helper function for capture ack callback + */ +static inline void +snd_pcm_indirect2_capture_transfer(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t null) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + snd_pcm_uframes_t appl_ptr = runtime->control->appl_ptr; + snd_pcm_sframes_t diff = appl_ptr - rec->appl_ptr; + + if (diff) { +#ifdef SND_PCM_INDIRECT2_STAT + rec->lastdifftime = jiffies; +#endif + if (diff < -(snd_pcm_sframes_t) (runtime->boundary / 2)) + diff += runtime->boundary; + rec->sw_ready -= frames_to_bytes(runtime, diff); + rec->appl_ptr = appl_ptr; + } + /* if hardware has something, but the intermediate buffer is full + * => skip contents of buffer + */ + if (rec->hw_ready && (rec->sw_ready >= (int)rec->sw_buffer_size)) { + unsigned int bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstzerotime == 0) { + rec->firstzerotime = jiffies; + snd_printk(KERN_DEBUG "STAT: (capture) @firstzerotime: " + "mul_elapsed: %d, min_period_count: %d\n", + rec->mul_elapsed, rec->min_period_count); + snd_printk(KERN_DEBUG "STAT: (capture) @firstzerotime: " + "sw_io: %d, sw_data: %d, appl_ptr: %u\n", + rec->sw_io, rec->sw_data, + (unsigned int)appl_ptr); + } + if ((jiffies - rec->firstzerotime) < 3750) { + rec->zero_times[(jiffies - rec->firstzerotime)]++; + rec->zero_times_saved++; + } else + rec->zero_times_notsaved++; +#endif + bytes = null(substream, rec); + +#ifdef SND_PCM_INDIRECT2_STAT + rec->zeros2hw += bytes; + if (bytes < 64) + rec->zero_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: (capture) %d zero Bytes copied to " + "hardware at once - too big to save!\n", + bytes); +#endif + snd_pcm_indirect2_increase_min_periods(substream, rec, 0, 0, + bytes); + /* report an overrun */ + rec->sw_io = SNDRV_PCM_POS_XRUN; + return; + } + while (rec->hw_ready && (rec->sw_ready < (int)rec->sw_buffer_size)) { + /* sw_to_end: max. number of bytes that we can write to the + * intermediate buffer (until it's end) + */ + size_t sw_to_end = rec->sw_buffer_size - rec->sw_data; + + /* bytes: max. number of bytes, which may be copied to the + * intermediate buffer without overflow (in _one_ step) + */ + size_t bytes = rec->sw_buffer_size - rec->sw_ready; + + /* limit number of bytes (for transfer) by available room in + * the intermediate buffer + */ + if (sw_to_end < bytes) + bytes = sw_to_end; + if (!bytes) + break; + +#ifdef SND_PCM_INDIRECT2_STAT + if (rec->firstbytetime == 0) + rec->firstbytetime = jiffies; + rec->lastbytetime = jiffies; +#endif + /* copy bytes from the intermediate buffer (position sw_data) + * to the HW at most and return number of bytes actually copied + * from HW + * Furthermore, set hw_ready to 0, if the fifo is empty now. + */ + bytes = copy(substream, rec, bytes); + rec->bytes2hw += bytes; + +#ifdef SND_PCM_INDIRECT2_STAT + if (bytes < 64) + rec->byte_sizes[bytes]++; + else + snd_printk(KERN_DEBUG + "STAT: (capture) %d Bytes copied to " + "hardware at once - too big to save!\n", + bytes); +#endif + /* increase sw_data by the number of actually copied bytes from + * HW + */ + rec->sw_data += bytes; + if (rec->sw_data == rec->sw_buffer_size) + rec->sw_data = 0; + + snd_pcm_indirect2_increase_min_periods(substream, rec, 0, 1, + bytes); + + /* number of bytes in the intermediate buffer, which haven't + * been fetched by ALSA yet. + */ + rec->sw_ready += bytes; + } + return; +} + +/* + * helper function for capture interrupt routine + */ +static inline void +snd_pcm_indirect2_capture_interrupt(struct snd_pcm_substream *substream, + struct snd_pcm_indirect2 *rec, + snd_pcm_indirect2_copy_t copy, + snd_pcm_indirect2_zero_t null) +{ +#ifdef SND_PCM_INDIRECT2_STAT + rec->irq_occured++; +#endif + /* hardware recorded some bytes, so there is something to read from the + * record fifo: + */ + rec->hw_ready = 1; + + /* don't call ack() now, instead call transfer() function directly + * (normally called by ack() ) + */ + snd_pcm_indirect2_capture_transfer(substream, rec, copy, null); + + if (rec->min_periods >= rec->min_multiple) { + +#ifdef SND_PCM_INDIRECT2_STAT + if ((rec->min_periods / rec->min_multiple) > 7) + snd_printk(KERN_DEBUG + "STAT: more than 7 (%d) mul_adds - " + "too big to save!\n", + (rec->min_periods / rec->min_multiple)); + else + rec->mul_adds[(rec->min_periods / rec->min_multiple)]++; + rec->mul_elapsed_real += (rec->min_periods / rec->min_multiple); + rec->mul_elapsed++; + + if (!(rec->mul_elapsed % 4)) { + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned int appl_ptr = + frames_to_bytes(runtime, + (unsigned int)runtime->control-> + appl_ptr) % rec->sw_buffer_size; + int diff = rec->sw_data - appl_ptr; + if (diff < 0) + diff += rec->sw_buffer_size; + snd_printk(KERN_DEBUG + "STAT: mul_elapsed: %d, sw_data: %u, " + "appl_ptr (bytes): %u, diff: %d\n", + rec->mul_elapsed, rec->sw_data, appl_ptr, + diff); + } +#endif + rec->min_periods = 0; + snd_pcm_period_elapsed(substream); + } +} + +#endif /* __SOUND_PCM_INDIRECT2_H */ -- 1.5.2.4

5 13

[alsa-devel] ALSA bugs and Launchpad
by Gavin Panella 15 Aug '07

15 Aug '07

Hi, I previously sent a similar message to webmaster(a)alsa-project.org, so apologies if you've seen this before. We've been trying to get Launchpad (*) to watch bugs within your Mantis bug tracker, but we can't get a full CSV export from your installation. I'm guessing it's been disabled or restricted somehow, so I want to make a case for letting Launchpad get at it. [* I work for Canonical Ltd. (http://www.canonical.com/) on the Launchpad.net site, specifically on the Bug Tracker (a.k.a. Malone). You can verify my signature and status as a Launchpad Developer at https://launchpad.net/~allenap.] I'll explain from the beginning, so you might want to skip a few paragraphs if you already know something about the Launchpad Bug Tracker. One feature of Launchpad is that users can add upstream dependencies to a bug report. For example, a display bug in Ubuntu could be filed on Launchpad, but be related to a bug in the core X.Org server project: https://bugs.launchpad.net/ubuntu/+source/xorg/+bug/38939 To add a dependency, a Launchpadder clicks "Also affects: Upstream" on the bug report, then enters the URL of the upstream bug report. Launchpad recognises this URL and will then "watch" the bug: once a day, Launchpad will fetch the status of the upstream bug and record it. Any changes are sent to bug subscribers, who might be the maintainers for that package in Ubuntu and who can then update the package with the fix. [Note that Launchpad only tracks the status of the upstream bug. If a Launchpadder wants to get involved in fixing the upstream bug then Launchpad will direct them to the upstream bug tracker.] We've recently added support for Mantis BT: Support Mantis as a remote bugtracker (and add ALSA's bugtracker) https://bugs.launchpad.net/malone/+bug/32266 There's also some info here: https://help.launchpad.net/Mantis The current approach we take is simple: the code logs in to a Mantis instance as a guest and fetches a complete CSV export (via csv_export.php). Many Mantis instances return a complete export of all their bugs to the guest user, but the ALSA bug tracker returns nothing unless the search is narrowed. This approach is not perfect (*) but it does work right now. If you're happy to do so, we'd be very grateful if you could let us fetch the full export while we're working on improvements. This will, amongst other things, help people working on Ubuntu be more aware of what's going on in the ALSA world. [* Taking a full export is crude (we get a full export even though we might only be watching a single bug, we only record the status and resolution anyway, and CSV has a plate-full of problems out-of-the- box) so we're already planning improvements, like narrowing searches, page scraping, writing extensions for Mantis, or using MantisConnect. Above all, we need to play nicely with external bug trackers.] Gavin. -- Gavin Panella, Europe/Luxembourg Tel: +352 26687161; GSM: +352 691739359 Skype: gavinpanella; Jabber: gavinpanella(a)gmail.com

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[alsa-devel] Why is the body of my msg stripped when posted to the list?
by Magdalena Iovescu 14 Aug '07

14 Aug '07

3 2

[alsa-devel] emu10k1 ALSA/jack problem
by jamesmstone＠gmail.com 14 Aug '07

14 Aug '07

Dear All, I am currently working on an audio program (aldrin) which uses libbzub driver http://trac.zeitherrschaft.org/zzub/browser/trunk/src/libzzub/driver.cpp for accessing the sound card. Libzzub, in turn uses rtaudio version 3.03, which is part of the libzzub source tree. The problem arises when I am running a jack server using my sblive, but specifying the default interface (hw:0). On startup, a vast number of xruns are generated (>200) and jack either falls over, or the computer almost completely locks up. This appears to be caused by rtaudio probing the alsa devices whilst jack is running. Interestingly, if I specify a sub-device (hw:0,3), or the integrated sound card (intel), no xruns are generated. I am wondering if this problem is arising because of a bug in the emu10k1 driver, or a problem with the way rtaudio probes alsa devices, or because it is a bad idea to probe alsa devices while they are being used by jack??? Best wishes, James

2 1

[alsa-devel] Noisy audio
by Magdalena Iovescu 14 Aug '07

14 Aug '07

1 0

[alsa-devel] Current alsa module list discrepancies
by Eliot Blennerhassett 14 Aug '07

14 Aug '07

Hello all, As part of getting the wiki up to date I have been comparing three lists of driver modules 1) List given by "configure --help" for --with-cards option 2) List of modules installed on my system when I make install alsa-driver 3) Current list of Module-* pages in the wiki The results of these comparisons is here. http://www.alsa-project.org/main/index.php/Matrix_Talk:Main Question is, are there some drivers in these lists that should be listed in "configure --help" (loopback -> aloop at least) Should all modules (even helpers) be documented in the wiki? Particularly helper modules that have parameters, to explain what the parameters are for. regards Eliot

2 1

[alsa-devel] Problem with CME UF5 MIDI keyboard master volume slider
by Damon Chaplin 13 Aug '07

13 Aug '07

Hi, I have a CME UF5 MIDI keyboard (USB), which mostly works fine with ALSA. However if I move the Master Volume slider I stop getting MIDI events - I have to disconnect the snd_seq port and reconnect it to get MIDI events working again. Any ideas what the problem is? Or how to track it down? Also, I wanted to try the rawmidi test app, but what would the device ID be? Damon

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[alsa-devel] HDA - Add support for more Dell systems
by Tobin Davis 13 Aug '07

13 Aug '07

This patch adds more support for Dell systems with Stac9205 codecs. Tested against a couple of different systems (with different pin configs), but the others should also work. Also cleaned up some of the 9205 patch code. Ignore previous email, as I missed a few vendor ID replacements. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

2 1

[alsa-devel] Recent code changes for HD-audio on HG repo
by Claude Michel 13 Aug '07

13 Aug '07

Hi, I have done come test of the new suspend/resume capabilities. However suspend failed to correctly work on my laptop (dell d630). You must note that there is a few trouble issues with sound on this laptop: the snd_hda_intel driver correctly detect a STAC9205 and get the default pin configuration from the bios. So far, so good ... however, pin B is by default configured as line out and is working as such. But, it does not correctly work with an external microphone as it need a vref. So, instead of using the bios settings, I use the windows default settings. This is not perfectly working : I usually get some glinches that disappear after a logout/login procedure. Nothing significant appears from a diff between /proc/asound/card0/codec0 before and after this procedure ... Now let talk about the suspend failure. The suspend procedure (which was previously working) produce a bug (I was not able to catch the message at this stage. This is a partial and hand written report ..): Bug: unable to handle kernel NULL Pointer deferred at virtual address 0000 .... Call trace ... <f9546d24> hda_set_power_state+0x34/0x150 [snd_hda_intel] <f9547768> hda_call_code_suspend+0x39/0x60 [snd_hda_intel] <f955477b7> snd_hda_suspend+0x27/0x50 [snd_hda_intel]<f954603c> azx_suspend+0x60/0xd0 [snd_hda_intel] <c023b29ac> pci_device_suspend+0x2c/0x70 <c0f29effd> suspend_devide+0xdd/0x110 .... By the way, I am running mandriva 2007.1 with kernel 2.6.17-14mdv. cpjm

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[alsa-devel] HDA - Add support for more Dell systems
by Tobin Davis 12 Aug '07

12 Aug '07

This patch adds more support for Dell systems with Stac9205 codecs. Tested against a couple of different systems (with different pin configs), but the others should also work. Also cleaned up some of the 9205 patch code. Signed off by Tobin Davis <tdavis(a)dsl-only.net>

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[alsa-devel] Intel HDA pin-config values (patch_sigmatel.c) - Dell Inspiron 1721 (also 1521?) mic
by Thorsten Riess 11 Aug '07

11 Aug '07

Hi, I am wondering where the values of ref9205_pin_configs in patch_sigmatel come from. The SigmaTel documentation gives different reference values for the 9205. The set of values I use at the moment is static unsigned int ref9205_pin_configs[12] = { 0x02214020, 0x02a19041, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x40000100, 0x10a000f0, 0x50a000f0, 0x01441030, 0x01c41030 }; (the first two are actually from the SigmaTel document, the other just mute the remaining in/outs) These numbers actually work quite well for the Dell Inspiron 1721 (and probably for the 1521), even the mic input works - although the quality could be better and the input levels have to be adjusted carefully. Can anybody confirm this? Thorsten.

2 1

[alsa-devel] ALSA WWW moved to Mediawiki
by Jaroslav Kysela 11 Aug '07

11 Aug '07

Hi all, today, I moved www.alsa-project.org to the second ALSA server with Mediawiki. The official URL is: http://www.alsa-project.org/main/index.php/Main_Page or http://www.alsa-project.org (for links from other sites) I also created templates to keep actual release versions in one page. The nice "Download a package" box on main page was taken from mediawiki.org. I'm also working on new box with news for the front page generating news from Main_Page_News. Last change is the modification of Special:Prefixindex (new special name is Special:ALSAPrefixindex) to remove 'Vendor-' and 'Module-' (or any other) prefixes to increase readability of Matrix:Main page. I would like to thank you to all wiki contributors. Jaroslav ----- Jaroslav Kysela <perex(a)suse.cz> Linux Kernel Sound Maintainer ALSA Project, SUSE Labs

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[alsa-devel] Where is simple mixer control "Surround" defined for ac97 or via82xx? I cant find it anywhere in ac97....
by John L. Utz III 10 Aug '07

10 Aug '07

The Surround controls exists: amixer sget Surround Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-46.50dB] [off] Front Right: Playback 0 [0%] [-46.50dB] [off] amixer controls | grep Surround numid=20,iface=MIXER,name='Surround Playback Switch' numid=21,iface=MIXER,name='Surround Playback Volume' But for the life of me, i cant find the code that implements it! There are many implementations inside of the hda directory, but those dont have anything todo with non hda things, the only item in the ac97 directory is for AD18XX: ac97_codec.c:958:AD18XX_PCM_VOLUME("Surround Playback Volume", 1) This shouldnt have anything to do with VIA 1618 stuff, right? tnx! johnu

3 4

Re: [alsa-devel] Audigy: recording level much too low (microphone)
by Zbigniew Baniewski 10 Aug '07

10 Aug '07

On Fri, Aug 10, 2007 at 07:53:27PM +0200, Peter Zubaj wrote: > Replace 0x0000 with value you want. 0x0000 is gain value - 0x0000 is > minimum and I think it is current default - 0 dB > 0x0101 is 1.5 dB > 0x0202 is 3 dB > .... > 0x0f0f is max - 22.5 dB > step is 1.5 dB Yes! Thanks a lot! :) I realize, it's a "workaround" rather than real solution, but this made Audigy suitable for VOIP (I tried the highest value of 0x0f0f). I've got a feeling, it could be even set in release this way for all the Audigy owners, until the final solution of DSP-related issues will take place, because it doesn't seem to spoil anything (although I didn't made much testing of all the available options yet). Or it could be switched on by some special option: like at "configuration" stage before compilation, or perhaps even as the additional module parameter. -- pozdrawiam / regards Zbigniew Baniewski

1 0

[alsa-devel] why would an scontrol think that it should have 2 items instead of just 1?
by John L. Utz III 10 Aug '07

10 Aug '07

Hello The VT1618 codec has a register value that turns on the Surround Side Output (it's off by default) I have implemented the control, but I am getting a really weird result, i have a bogus 'Item1' that shouldnt exist: jutz-gntdv alsa-driver # amixer sset Side On Simple mixer control 'Side',0 Capabilities: pvolume pswitch enum Items: 'On' 'Off' Item0: 'On' Item1: 'On' jutz-gntdv alsa-driver # amixer sset Side Off Simple mixer control 'Side',0 Capabilities: pvolume pswitch enum Items: 'On' 'Off' Item0: 'Off' Item1: 'On' This wasnt always there, it just showed up during development. Is the number of items cached in a temp file somewhere that i just need to remove? Here's the mixer element: { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = "Side", .info = snd_ac97_vt1618_surside_info, .get = snd_ac97_vt1618_surside_get, .put = snd_ac97_vt1618_surside_put }, Here's the function impls: /* surround side power up or down */ static int snd_ac97_vt1618_surside_info(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_info *uinfo) { static const char* texts[] = {"On", "Off"}; return ac97_enum_text_info(kcontrol, uinfo, texts, 2); } static int snd_ac97_vt1618_surside_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); ucontrol->value.enumerated.item[0] = (snd_ac97_read(pac97, 0x5c) & 0x0008) >> 3; return 0; } static int snd_ac97_vt1618_surside_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { //0x5c, 3, 1, struct snd_ac97 *pac97 = snd_kcontrol_chip(kcontrol); snd_ac97_update_bits(pac97, 0x5c, 0x0008, ucontrol->value.enumerated.item[0] << 3); return 0; }

2 2

[alsa-devel] [Regression] snd-hda-intel no longer loads as of alsa-driver-hg20070728
by Michael Gerdau 10 Aug '07

10 Aug '07

Hi list, snd-hda-intel does no longer load on my Dell XPS M1710 using alsa-driver-hg20070728. However it works without apparent problems with alsa-driver-hg20070727. I'm running 2.6.22.1-cfs-v19 on a SuSE 10.2 x86_64. Find attached an excerpt from /var/log/messages first loading hg20070727 followed by hg20070728. If you need additional info drop me a note. Best, Michael -- Technosis GmbH, Geschäftsführer: Michael Gerdau, Tobias Dittmar Sitz Hamburg; HRB 89145 Amtsgericht Hamburg Vote against SPAM - see http://www.politik-digital.de/spam/ Michael Gerdau email: mgd(a)technosis.de GPG-keys available on request or at public keyserver

2 5

[alsa-devel] Recent code changes for HD-audio on HG repo
by Takashi Iwai 10 Aug '07

10 Aug '07

Hi, right now I committed a series of patches for HD-audio driver to HG repository. There are two big changes: - The suspend/resume code is rewritten Now the driver uses cached register values to restore the last status. It's much cleaner than the previous code, and should work better. But, I couldn't test it on all existing laptops. So, please test and report any positive/negative feedbacks. - New POWER_SAVE kconfig The driver has POWER_SAVE confiugration. When enabled (it's so if you build from alsa-driver tree), the driver turns off the codec and the controller chips when not used, and turn on again dynamically on demand. Good for laptops, good for mother earth! :) The feature can be enabled/diabled even on the fly via "power_save" module option of snd-hda-intel. For disabling it, write 0 to /sys/modules/snd_hda_intel/parameters/power_save for disabling it, or pass power_save=0 in modprobe option (or its configuration). Again, please give any positive/negative feedbacks, as I couldn't test many devices. thanks, Takashi

1 0

[alsa-devel] Need help with alsa dev wiki migration
by Eliot Blennerhassett 10 Aug '07

10 Aug '07

Greetings, Anybody who can figure out where to put the pages in the ToDo section of this page please step forward now... http://bugtrack.alsa-project.org/main/index.php/DriversInDevelopment The remaining pages in todo here are users home pages, would be great if those users would migrate them. http://bugtrack.alsa-project.org/main/index.php/Developer_Zone Other than these, the contents of the alsa developers wiki has been copied to the alsa wiki. Other eyes checking that this has been done satisfactorily would be appreciated thanks Eliot

2 1

[alsa-devel] pcm_empty.c in doxygen
by Rolland Dudemaine 10 Aug '07

10 Aug '07

Hi, It looks like the doxygen-generated documentation lacks the pcm/pcm_empty.c bit. Since the default plugin uses empty as plugin, it's useful to have docs for it. Here is a diff to apply to solve this. --- doc/doxygen.cfg.old 2007-08-10 13:14:00.000000000 +0200 +++ doc/doxygen.cfg 2007-08-10 13:14:38.000000000 +0200 @@ -72,6 +72,7 @@ ../src/pcm/pcm_ioplug.c \ ../src/pcm/pcm_misc.c \ ../src/pcm/pcm_simple.c \ + ../src/pcm/pcm_empty.c \ ../src/rawmidi \ ../src/timer \ ../src/hwdep \ Best Regards, --Rolland Dudemaine

2 1

[alsa-devel] Bug 305
by chris＠idlelion.net 10 Aug '07

10 Aug '07

I tested Fedora 8 on my ThinkPad 600X. The 600X has a CS4264, and F8 uses a 2.6.23 kernel (I forget the version at the moment). The sound does not work after resume. How should I check whether the fix for this issue is included in a kernel? Version number? Thanks, Chris

2 5

[alsa-devel] Audigy: recording level much too low (microphone)
by Zbigniew Baniewski 10 Aug '07

10 Aug '07

Hallo, I've got a problem with Audigy (it's SB0230 release) soundcard: the recording level, when taking the sound from microphone, is much too low. I've sent a mail to "users" list, so here I'm just pointing to that mail: http://www.mail-archive.com/alsa-user@lists.sourceforge.net/msg20235.html ...and so I won't repeat it all here. (I made a mistake in the subject there, writing about "Audigy2") My card is seen by 1.0.14 ALSA as ( cat /proc/asound/cards ): 0 [Audigy ]: Audigy - Audigy 1 [Unknown] Audigy 1 [Unknown] (rev.3, serial:0x5a1102) at 0xc400, irq 11 As I understand, there is some variable (or register?), which controls that recording level for microphone input - perhaps someone of the developers could tell me, which variable is it, and which file should I search? In such way perhaps I could fix it on my own - just by setting higher max. level, or in similar way. But searching all that files on my own could take serious amount of time, I'm afraid. Or perhaps there's another way to fix it (some special settings)? -- pozdrawiam / regards Zbigniew Baniewski

2 1

[alsa-devel] [PATCH] ad1848-lib Makefile cleanup
by Krzysztof Helt 10 Aug '07

10 Aug '07

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> This patch uses the Kconfig parameter SND_AD1848_LIB instead of mentioning each driver that requires the ad1848-lib separately in the Makefile. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- diff -urp linux-2.6.23.old/sound/isa/ad1848/Makefile linux-2.6.23/sound/isa/ad1848/Makefile --- linux-2.6.23.old/sound/isa/ad1848/Makefile 2007-08-04 22:12:08.000000000 +0200 +++ linux-2.6.23/sound/isa/ad1848/Makefile 2007-08-08 18:53:20.000000000 +0200 @@ -7,9 +7,7 @@ snd-ad1848-lib-objs := ad1848_lib.o snd-ad1848-objs := ad1848.o # Toplevel Module Dependency -obj-$(CONFIG_SND_CMI8330) += snd-ad1848-lib.o -obj-$(CONFIG_SND_SGALAXY) += snd-ad1848-lib.o obj-$(CONFIG_SND_AD1848) += snd-ad1848.o snd-ad1848-lib.o -obj-$(CONFIG_SND_OPTI92X_AD1848) += snd-ad1848-lib.o +obj-$(CONFIG_SND_AD1848_LIB) += snd-ad1848-lib.o obj-m := $(sort $(obj-m))

2 5

[alsa-devel] C-Media 8788
by Pekka.Panula＠sofor.fi 10 Aug '07

10 Aug '07

Hi Any inside info about C-Media 8788 support, i see that repository contains some sort of driver for it, but what state is it, is it working as wiki says its about 35% done. Happy to know also if its going for Linux 2.6.23 kernel? I think lot of people would be happy to get it to kernel soon as possible as it seems Creative has serious delays about x-fi driver (i wonder if there will be any Linux support at all, not looking good) and many people are thinking about Razer barracuda AC-1 or Auzentech X-Meridean which uses C-Media 8788 chipset. Terveisin/Regards, Pekka Panula, Net Servant Oy - A Sofor company pekka.panula(a)sofor.fi

2 1

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

10 Aug '07

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

2 2

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

10 Aug '07

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

2 1

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

09 Aug '07

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

2 1

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

09 Aug '07

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

2 1

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

09 Aug '07

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

3 2

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

09 Aug '07

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

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

09 Aug '07

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

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

08 Aug '07

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

3 2

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

08 Aug '07

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

3 4

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

08 Aug '07

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

2 1

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

08 Aug '07

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

2 4

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

07 Aug '07

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

2 2

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

07 Aug '07

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

1 1

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

07 Aug '07

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

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

07 Aug '07

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

2 1

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

07 Aug '07

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

2 1

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

07 Aug '07

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

2 3

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

06 Aug '07

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

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

06 Aug '07

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

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

06 Aug '07

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

2 1

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

06 Aug '07

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

3 2

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

05 Aug '07

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

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

04 Aug '07

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

1 0

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

04 Aug '07

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

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

04 Aug '07

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

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

03 Aug '07

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

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

03 Aug '07

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

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