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

16 Jun '08

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

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

16 Jun '08

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Separate AD1848 registers definition into a new header. The new header can be used by compatible CS4231 chip driver. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- This patch is a start combining ad1848_lib and cs4231_lib functionality. My approach is to make the cs4231_lib module use the ad1848_lib module. This should allow removing most of identical functions from the cs4231_lib and use the functions from the ad1848_lib directly. If this approach is not a good idea, please drop the patch. My current plan is to reuse the ad1848 register constants in the cs423x code and remove duplicated cs4231 constants. This should make the ad1848_lib and cs4231_lib easier to compare for differences. The next step is to use the snd_cs4231 structure instead of the snd_ad1848 structure inside the ad1848_lib. This will make functions from the ad1848_lib taking the same arguments as the functions in the cs4231_lib. The last step would be to export needed functions from the ad1848_lib and use them directly in the cs4231_lib. This should reduce the size of the cs4231_lib while preserving smaller the ad1848_lib for driver which uses only ad1848 functionality. Kind regards, Krzysztof diff -urpN linux-ref/include/sound/ad1848-regs.h linux-mm/include/sound/ad1848-regs.h --- linux-ref/include/sound/ad1848-regs.h 1970-01-01 01:00:00.000000000 +0100 +++ linux-mm/include/sound/ad1848-regs.h 2008-06-15 07:00:08.041351281 +0200 @@ -0,0 +1,104 @@ +#ifndef __SOUND_AD1848_REGS_H +#define __SOUND_AD1848_REGS_H + +/* + * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> + * Definitions for AD1847/AD1848/CS4248 chips registers + * + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA + * + */ + +/* IO ports */ + +#define AD1848P(chip, x) ((chip)->port + c_d_c_AD1848##x) + +#define c_d_c_AD1848REGSEL 0 +#define c_d_c_AD1848REG 1 +#define c_d_c_AD1848STATUS 2 +#define c_d_c_AD1848PIO 3 + +/* codec registers */ + +#define AD1848_LEFT_INPUT 0x00 /* left input control */ +#define AD1848_RIGHT_INPUT 0x01 /* right input control */ +#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ +#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ +#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ +#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ +#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ +#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ +#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ +#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ +#define AD1848_PIN_CTRL 0x0a /* pin control */ +#define AD1848_TEST_INIT 0x0b /* test and initialization */ +#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ +#define AD1848_LOOPBACK 0x0d /* loopback control */ +#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ +#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ + +/* definitions for codec register select port - CODECP( REGSEL ) */ + +#define AD1848_INIT 0x80 /* CODEC is initializing */ +#define AD1848_MCE 0x40 /* mode change enable */ +#define AD1848_TRD 0x20 /* transfer request disable */ + +/* definitions for codec status register - CODECP( STATUS ) */ + +#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ + +/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ + +#define AD1848_ENABLE_MIC_GAIN 0x20 + +#define AD1848_MIXS_LINE1 0x00 +#define AD1848_MIXS_AUX1 0x40 +#define AD1848_MIXS_LINE2 0x80 +#define AD1848_MIXS_ALL 0xc0 + +/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ + +#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ +#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ +#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ +#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ +#define AD1848_STEREO 0x10 /* stereo mode */ +/* bits 3-1 define frequency divisor */ +#define AD1848_XTAL1 0x00 /* 24.576 crystal */ +#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ + +/* definitions for interface control register - AD1848_IFACE_CTRL */ + +#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ +#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ +#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ +#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ +#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ +#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ +#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ + +/* definitions for pin control register - AD1848_PIN_CTRL */ + +#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ +#define AD1848_XCTL1 0x40 /* external control #1 */ +#define AD1848_XCTL0 0x80 /* external control #0 */ + +/* definitions for test and init register - AD1848_TEST_INIT */ + +#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ +#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ + +#endif /* __SOUND_AD1848_REGS_H */ diff -urpN linux-ref/include/sound/ad1848.h linux-mm/include/sound/ad1848.h --- linux-ref/include/sound/ad1848.h 2008-06-15 06:58:15.293013657 +0200 +++ linux-mm/include/sound/ad1848.h 2008-06-15 00:14:36.728628406 +0200 @@ -25,84 +25,7 @@ #include "pcm.h" #include <linux/interrupt.h> -/* IO ports */ - -#define AD1848P( chip, x ) ( (chip) -> port + c_d_c_AD1848##x ) - -#define c_d_c_AD1848REGSEL 0 -#define c_d_c_AD1848REG 1 -#define c_d_c_AD1848STATUS 2 -#define c_d_c_AD1848PIO 3 - -/* codec registers */ - -#define AD1848_LEFT_INPUT 0x00 /* left input control */ -#define AD1848_RIGHT_INPUT 0x01 /* right input control */ -#define AD1848_AUX1_LEFT_INPUT 0x02 /* left AUX1 input control */ -#define AD1848_AUX1_RIGHT_INPUT 0x03 /* right AUX1 input control */ -#define AD1848_AUX2_LEFT_INPUT 0x04 /* left AUX2 input control */ -#define AD1848_AUX2_RIGHT_INPUT 0x05 /* right AUX2 input control */ -#define AD1848_LEFT_OUTPUT 0x06 /* left output control register */ -#define AD1848_RIGHT_OUTPUT 0x07 /* right output control register */ -#define AD1848_DATA_FORMAT 0x08 /* clock and data format - playback/capture - bits 7-0 MCE */ -#define AD1848_IFACE_CTRL 0x09 /* interface control - bits 7-2 MCE */ -#define AD1848_PIN_CTRL 0x0a /* pin control */ -#define AD1848_TEST_INIT 0x0b /* test and initialization */ -#define AD1848_MISC_INFO 0x0c /* miscellaneous information */ -#define AD1848_LOOPBACK 0x0d /* loopback control */ -#define AD1848_DATA_UPR_CNT 0x0e /* playback/capture upper base count */ -#define AD1848_DATA_LWR_CNT 0x0f /* playback/capture lower base count */ - -/* definitions for codec register select port - CODECP( REGSEL ) */ - -#define AD1848_INIT 0x80 /* CODEC is initializing */ -#define AD1848_MCE 0x40 /* mode change enable */ -#define AD1848_TRD 0x20 /* transfer request disable */ - -/* definitions for codec status register - CODECP( STATUS ) */ - -#define AD1848_GLOBALIRQ 0x01 /* IRQ is active */ - -/* definitions for AD1848_LEFT_INPUT and AD1848_RIGHT_INPUT registers */ - -#define AD1848_ENABLE_MIC_GAIN 0x20 - -#define AD1848_MIXS_LINE1 0x00 -#define AD1848_MIXS_AUX1 0x40 -#define AD1848_MIXS_LINE2 0x80 -#define AD1848_MIXS_ALL 0xc0 - -/* definitions for clock and data format register - AD1848_PLAYBK_FORMAT */ - -#define AD1848_LINEAR_8 0x00 /* 8-bit unsigned data */ -#define AD1848_ALAW_8 0x60 /* 8-bit A-law companded */ -#define AD1848_ULAW_8 0x20 /* 8-bit U-law companded */ -#define AD1848_LINEAR_16 0x40 /* 16-bit twos complement data - little endian */ -#define AD1848_STEREO 0x10 /* stereo mode */ -/* bits 3-1 define frequency divisor */ -#define AD1848_XTAL1 0x00 /* 24.576 crystal */ -#define AD1848_XTAL2 0x01 /* 16.9344 crystal */ - -/* definitions for interface control register - AD1848_IFACE_CTRL */ - -#define AD1848_CAPTURE_PIO 0x80 /* capture PIO enable */ -#define AD1848_PLAYBACK_PIO 0x40 /* playback PIO enable */ -#define AD1848_CALIB_MODE 0x18 /* calibration mode bits */ -#define AD1848_AUTOCALIB 0x08 /* auto calibrate */ -#define AD1848_SINGLE_DMA 0x04 /* use single DMA channel */ -#define AD1848_CAPTURE_ENABLE 0x02 /* capture enable */ -#define AD1848_PLAYBACK_ENABLE 0x01 /* playback enable */ - -/* definitions for pin control register - AD1848_PIN_CTRL */ - -#define AD1848_IRQ_ENABLE 0x02 /* enable IRQ */ -#define AD1848_XCTL1 0x40 /* external control #1 */ -#define AD1848_XCTL0 0x80 /* external control #0 */ - -/* definitions for test and init register - AD1848_TEST_INIT */ - -#define AD1848_CALIB_IN_PROGRESS 0x20 /* auto calibrate in progress */ -#define AD1848_DMA_REQUEST 0x10 /* DMA request in progress */ +#include "ad1848-regs.h" /* defines for codec.mode */ ---------------------------------------------------------------------- Tania telefonia internetowa! 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[alsa-devel] Problem regarding CA0106
by Travis Place 16 Jun '08

16 Jun '08

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

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

15 Jun '08

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

15 Jun '08

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

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

14 Jun '08

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

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

14 Jun '08

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

4 26

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

14 Jun '08

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

4 9

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

13 Jun '08

Added support for new family of IDT codecs. --- Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index a4f44a0..115cee6 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -349,7 +349,7 @@ static hda_nid_t stac92hd73xx_pin_nids[13] = { static hda_nid_t stac92hd71bxx_pin_nids[10] = { 0x0a, 0x0b, 0x0c, 0x0d, 0x0e, - 0x0f, 0x14, 0x18, 0x19, 0x1e, + 0x14, 0x18, 0x19, 0x1e, 0x0f }; static hda_nid_t stac927x_pin_nids[14] = { @@ -641,15 +641,21 @@ static struct hda_verb stac92hd71bxx_analog_core_init[] = { { 0x28, AC_VERB_SET_VOLUME_KNOB_CONTROL, 0xff}, /* connect headphone jack to dac1 */ { 0x0a, AC_VERB_SET_CONNECT_SEL, 0x01}, - /* connect ports 0d and 0f to audio mixer */ - { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x2}, - { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* connect port 0d to audio mixer */ + { 0x0d, AC_VERB_SET_CONNECT_SEL, 0x02}, /* unmute dac0 input in audio mixer */ { 0x17, AC_VERB_SET_AMP_GAIN_MUTE, 0x701f}, - /* unmute right and left channels for nodes 0x0a, 0xd, 0x0f */ + /* unmute right and left channels for nodes 0x0a, 0x0d */ { 0x0a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, { 0x0d, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, + {} +}; + +static struct hda_verb stac92hd71bxx_enable_portf[] = { + /* connect port 0f to audio mixer */ + { 0x0f, AC_VERB_SET_CONNECT_SEL, 0x02}, + { 0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, /* Speaker */ + /* unmute right and left channels for nodes 0x0f */ { 0x0f, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, {} }; @@ -1313,20 +1319,20 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { static unsigned int ref92hd71bxx_pin_configs[10] = { 0x02214030, 0x02a19040, 0x01a19020, 0x01014010, - 0x0181302e, 0x01114010, 0x01019020, 0x90a000f0, - 0x90a000f0, 0x01452050, + 0x0181302e, 0x01019020, 0x90a000f0, 0x90a000f0, + 0x01452050, 0x01114010, }; -static unsigned int dell_m4_1_pin_configs[13] = { +static unsigned int dell_m4_1_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x40f000f0, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x90a000f0, - 0x40f000f0, 0x4f0000f0, + 0x23a1902e, 0x40f000f0, 0x90a000f0, 0x40f000f0, + 0x4f0000f0, 0x23014250, }; -static unsigned int dell_m4_2_pin_configs[13] = { +static unsigned int dell_m4_2_pin_configs[10] = { 0x0421101f, 0x04a11221, 0x90a70330, 0x90170110, - 0x23a1902e, 0x23014250, 0x40f000f0, 0x40f000f0, - 0x40f000f0, 0x044413b0, + 0x23a1902e, 0x40f000f0, 0x40f000f0, 0x40f000f0, + 0x044413b0, 0x23014250, }; static unsigned int *stac92hd71bxx_brd_tbl[STAC_92HD71BXX_MODELS] = { @@ -2633,7 +2639,7 @@ static int stac92xx_auto_create_dmic_input_ctls(struct hda_codec *codec, 0, AC_VERB_GET_CONFIG_DEFAULT, 0); - if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) + if (def_conf && (def_conf == AC_JACK_PORT_NONE)) continue; nid = spec->dmic_nids[i]; @@ -3103,13 +3109,14 @@ static int stac92xx_init(struct hda_codec *codec) 0, AC_VERB_GET_PIN_WIDGET_CONTROL, 0); int def_conf = snd_hda_codec_read(codec, spec->pwr_nids[i], 0, AC_VERB_GET_CONFIG_DEFAULT, 0); + def_conf = get_defcfg_connect(def_conf); /* outputs are only ports capable of power management * any attempts on powering down a input port cause the * referenced VREF to act quirky. */ if (pinctl & AC_PINCTL_IN_EN) continue; - if (get_defcfg_connect(def_conf) != AC_JACK_PORT_FIXED) + if (def_conf && def_conf != AC_JACK_PORT_FIXED) continue; enable_pin_detect(codec, spec->pwr_nids[i], event | i); codec->patch_ops.unsol_event(codec, (event | i) << 26); @@ -3614,12 +3621,39 @@ static int patch_stac92hd71bxx(struct hda_codec *codec) codec->spec = spec; spec->num_pins = ARRAY_SIZE(stac92hd71bxx_pin_nids); + spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pin_nids = stac92hd71bxx_pin_nids; spec->board_config = snd_hda_check_board_config(codec, STAC_92HD71BXX_MODELS, stac92hd71bxx_models, stac92hd71bxx_cfg_tbl); again: + switch (codec->vendor_id) { + case 0x111d76b6: /* 4 Port without Analog Mixer */ + case 0x111d76b7: + case 0x111d76b4: /* 6 Port without Analog Mixer */ + case 0x111d76b5: + spec->mixer = stac92hd71bxx_mixer; + spec->init = stac92hd71bxx_core_init; + break; + case 0x111d7603: /* 6 Port with Analog Mixer */ + case 0x111d7608: + /* no output amps */ + spec->num_pwrs = 0; + /* fallthru */ + default: + spec->mixer = stac92hd71bxx_analog_mixer; + spec->init = stac92hd71bxx_analog_core_init; + } + + if (codec->vendor_id != 0x111d7608) + snd_hda_sequence_write(codec, stac92hd71bxx_enable_portf); + else { + /* skip the VSW on port f */ + spec->num_pins--; + stac92xx_set_config_reg(codec, 0xf, 0x40f000f0); + } + if (spec->board_config < 0) { snd_printdd(KERN_INFO "hda_codec: Unknown model for" " STAC92HD71BXX, using BIOS defaults\n"); @@ -3634,19 +3668,6 @@ again: stac92xx_set_config_regs(codec); } - switch (codec->vendor_id) { - case 0x111d76b6: /* 4 Port without Analog Mixer */ - case 0x111d76b7: - case 0x111d76b4: /* 6 Port without Analog Mixer */ - case 0x111d76b5: - spec->mixer = stac92hd71bxx_mixer; - spec->init = stac92hd71bxx_core_init; - break; - default: - spec->mixer = stac92hd71bxx_analog_mixer; - spec->init = stac92hd71bxx_analog_core_init; - } - spec->aloopback_mask = 0x20; spec->aloopback_shift = 0; @@ -3666,7 +3687,6 @@ again: spec->num_dmics = STAC92HD71BXX_NUM_DMICS; spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); - spec->num_pwrs = ARRAY_SIZE(stac92hd71bxx_pwr_nids); spec->pwr_nids = stac92hd71bxx_pwr_nids; spec->multiout.num_dacs = 1; @@ -4306,10 +4326,11 @@ struct hda_codec_preset snd_hda_preset_sigmatel[] = { { .id = 0x838476a5, .name = "STAC9255D", .patch = patch_stac9205 }, { .id = 0x838476a6, .name = "STAC9254", .patch = patch_stac9205 }, { .id = 0x838476a7, .name = "STAC9254D", .patch = patch_stac9205 }, + { .id = 0x111d7603, .name = "92HD75B3X5", .patch = patch_stac92hd71bxx}, + { .id = 0x111d7608, .name = "92HD75B2X5", .patch = patch_stac92hd71bxx}, { .id = 0x111d7674, .name = "92HD73D1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7675, .name = "92HD73C1X5", .patch = patch_stac92hd73xx }, { .id = 0x111d7676, .name = "92HD73E1X5", .patch = patch_stac92hd73xx }, - { .id = 0x111d7608, .name = "92HD71BXX", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b0, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b1, .name = "92HD71B8X", .patch = patch_stac92hd71bxx }, { .id = 0x111d76b2, .name = "92HD71B7X", .patch = patch_stac92hd71bxx },

2 3

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

13 Jun '08

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

2 2